From fc85d9d0b3ba8f8934963c760af98fc38029d9da Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Thu, 9 Nov 2023 15:58:59 +0200 Subject: [PATCH 001/337] ASoC: SOF: imx8m: Add DAI driver entry for MICFIL PDM MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This will allow creating of PDM DAI links. Reviewed-by: Péter Ujfalusi Reviewed-by: Iuliana Prodan Signed-off-by: Daniel Baluta Link: https://lore.kernel.org/r/20231109135900.88310-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8m.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 2680f061ba42..f088fd1a672b 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -313,6 +313,13 @@ static struct snd_soc_dai_driver imx8m_dai[] = { .channels_max = 32, }, }, +{ + .name = "micfil", + .capture = { + .channels_min = 1, + .channels_max = 8, + }, +}, }; static int imx8m_dsp_set_power_state(struct snd_sof_dev *sdev, From 89ef42088b3ba884a007ad10bd89ce8a81b9dedd Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Thu, 9 Nov 2023 15:59:00 +0200 Subject: [PATCH 002/337] ASoC: SOF: Add support for configuring PDM interface from topology MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Currently we only support configuration for number of channels and sample rate. Reviewed-by: Péter Ujfalusi Reviewed-by: Iuliana Prodan Signed-off-by: Daniel Baluta Link: https://lore.kernel.org/r/20231109135900.88310-3-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- include/sound/sof/dai-imx.h | 7 +++++ include/sound/sof/dai.h | 2 ++ include/uapi/sound/sof/tokens.h | 4 +++ sound/soc/sof/ipc3-pcm.c | 11 ++++++++ sound/soc/sof/ipc3-topology.c | 46 +++++++++++++++++++++++++++++++++ sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 5 ++++ 7 files changed, 76 insertions(+) diff --git a/include/sound/sof/dai-imx.h b/include/sound/sof/dai-imx.h index ca8325353d41..6bc987bd4761 100644 --- a/include/sound/sof/dai-imx.h +++ b/include/sound/sof/dai-imx.h @@ -51,4 +51,11 @@ struct sof_ipc_dai_sai_params { uint16_t tdm_slot_width; uint16_t reserved2; /* alignment */ } __packed; + +/* MICFIL Configuration Request - SOF_IPC_DAI_MICFIL_CONFIG */ +struct sof_ipc_dai_micfil_params { + uint32_t pdm_rate; + uint32_t pdm_ch; +} __packed; + #endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 3041f5805b7b..4773a5f616a4 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -88,6 +88,7 @@ enum sof_ipc_dai_type { SOF_DAI_AMD_HS, /**< Amd HS */ SOF_DAI_AMD_SP_VIRTUAL, /**< AMD ACP SP VIRTUAL */ SOF_DAI_AMD_HS_VIRTUAL, /**< AMD ACP HS VIRTUAL */ + SOF_DAI_IMX_MICFIL, /** < i.MX MICFIL PDM */ }; /* general purpose DAI configuration */ @@ -117,6 +118,7 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_acpdmic_params acpdmic; struct sof_ipc_dai_acp_params acphs; struct sof_ipc_dai_mtk_afe_params afe; + struct sof_ipc_dai_micfil_params micfil; }; } __packed; diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index 453cab2a1209..0fb39780f9bd 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -213,4 +213,8 @@ #define SOF_TKN_AMD_ACPI2S_CH 1701 #define SOF_TKN_AMD_ACPI2S_TDM_MODE 1702 +/* MICFIL PDM */ +#define SOF_TKN_IMX_MICFIL_RATE 2000 +#define SOF_TKN_IMX_MICFIL_CH 2001 + #endif diff --git a/sound/soc/sof/ipc3-pcm.c b/sound/soc/sof/ipc3-pcm.c index 2d0addcbc819..330f04bcd75d 100644 --- a/sound/soc/sof/ipc3-pcm.c +++ b/sound/soc/sof/ipc3-pcm.c @@ -384,6 +384,17 @@ static int sof_ipc3_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, dev_dbg(component->dev, "AMD_DMIC channels_min: %d channels_max: %d\n", channels->min, channels->max); break; + case SOF_DAI_IMX_MICFIL: + rate->min = private->dai_config->micfil.pdm_rate; + rate->max = private->dai_config->micfil.pdm_rate; + channels->min = private->dai_config->micfil.pdm_ch; + channels->max = private->dai_config->micfil.pdm_ch; + + dev_dbg(component->dev, + "MICFIL PDM rate_min: %d rate_max: %d\n", rate->min, rate->max); + dev_dbg(component->dev, "MICFIL PDM channels_min: %d channels_max: %d\n", + channels->min, channels->max); + break; default: dev_err(component->dev, "Invalid DAI type %d\n", private->dai_config->type); break; diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index ba4ef290b634..7a4932c152a9 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -286,6 +286,16 @@ static const struct sof_topology_token acpi2s_tokens[] = { offsetof(struct sof_ipc_dai_acp_params, tdm_mode)}, }; +/* MICFIL PDM */ +static const struct sof_topology_token micfil_pdm_tokens[] = { + {SOF_TKN_IMX_MICFIL_RATE, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_micfil_params, pdm_rate)}, + {SOF_TKN_IMX_MICFIL_CH, + SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_dai_micfil_params, pdm_ch)}, +}; + /* Core tokens */ static const struct sof_topology_token core_tokens[] = { {SOF_TKN_COMP_CORE_ID, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, @@ -322,6 +332,8 @@ static const struct sof_token_info ipc3_token_list[SOF_TOKEN_COUNT] = { [SOF_AFE_TOKENS] = {"AFE tokens", afe_tokens, ARRAY_SIZE(afe_tokens)}, [SOF_ACPDMIC_TOKENS] = {"ACPDMIC tokens", acpdmic_tokens, ARRAY_SIZE(acpdmic_tokens)}, [SOF_ACPI2S_TOKENS] = {"ACPI2S tokens", acpi2s_tokens, ARRAY_SIZE(acpi2s_tokens)}, + [SOF_MICFIL_TOKENS] = {"MICFIL PDM tokens", + micfil_pdm_tokens, ARRAY_SIZE(micfil_pdm_tokens)}, }; /** @@ -1136,6 +1148,37 @@ static int sof_link_esai_load(struct snd_soc_component *scomp, struct snd_sof_da return 0; } +static int sof_link_micfil_load(struct snd_soc_component *scomp, struct snd_sof_dai_link *slink, + struct sof_ipc_dai_config *config, struct snd_sof_dai *dai) +{ + struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; + struct sof_dai_private_data *private = dai->private; + u32 size = sizeof(*config); + int ret; + + /* handle master/slave and inverted clocks */ + sof_dai_set_format(hw_config, config); + + config->hdr.size = size; + + /* parse the required set of MICFIL PDM tokens based on num_hw_cfgs */ + ret = sof_update_ipc_object(scomp, &config->micfil, SOF_MICFIL_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; + + dev_info(scomp->dev, "MICFIL PDM config dai_index %d channel %d rate %d\n", + config->dai_index, config->micfil.pdm_ch, config->micfil.pdm_rate); + + dai->number_configs = 1; + dai->current_config = 0; + private->dai_config = kmemdup(config, size, GFP_KERNEL); + if (!private->dai_config) + return -ENOMEM; + + return 0; +} + static int sof_link_acp_dmic_load(struct snd_soc_component *scomp, struct snd_sof_dai_link *slink, struct sof_ipc_dai_config *config, struct snd_sof_dai *dai) { @@ -1559,6 +1602,9 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget) case SOF_DAI_IMX_ESAI: ret = sof_link_esai_load(scomp, slink, config, dai); break; + case SOF_DAI_IMX_MICFIL: + ret = sof_link_micfil_load(scomp, slink, config, dai); + break; case SOF_DAI_AMD_BT: ret = sof_link_acp_bt_load(scomp, slink, config, dai); break; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 5d5eeb1a1a6f..99c940b22538 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -275,6 +275,7 @@ enum sof_tokens { SOF_GAIN_TOKENS, SOF_ACPDMIC_TOKENS, SOF_ACPI2S_TOKENS, + SOF_MICFIL_TOKENS, /* this should be the last */ SOF_TOKEN_COUNT, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index a3a3af252259..9f717366cddc 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -296,6 +296,7 @@ static const struct sof_dai_types sof_dais[] = { {"AFE", SOF_DAI_MEDIATEK_AFE}, {"ACPSP_VIRTUAL", SOF_DAI_AMD_SP_VIRTUAL}, {"ACPHS_VIRTUAL", SOF_DAI_AMD_HS_VIRTUAL}, + {"MICFIL", SOF_DAI_IMX_MICFIL}, }; @@ -1960,6 +1961,10 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_ACPI2S_TOKENS; num_tuples += token_list[SOF_ACPI2S_TOKENS].count; break; + case SOF_DAI_IMX_MICFIL: + token_id = SOF_MICFIL_TOKENS; + num_tuples += token_list[SOF_MICFIL_TOKENS].count; + break; default: break; } From 59fff33e9d923779b68fe46f39262256caba71d6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:46:56 +0100 Subject: [PATCH 003/337] ASoC: Intel: avs: da7219: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-2-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/da7219.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 6060894954df..c018f84fe025 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -277,16 +277,24 @@ static int avs_da7219_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_da7219_driver_ids[] = { + { + .name = "avs_da7219", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_da7219_driver_ids); + static struct platform_driver avs_da7219_driver = { .probe = avs_da7219_probe, .driver = { .name = "avs_da7219", .pm = &snd_soc_pm_ops, }, + .id_table = avs_da7219_driver_ids, }; module_platform_driver(avs_da7219_driver); MODULE_AUTHOR("Cezary Rojewski "); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_da7219"); From deb8dcad7bc3505881e2c0680d8f01f23fcbba98 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:46:57 +0100 Subject: [PATCH 004/337] ASoC: Intel: avs: dmic: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-3-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/dmic.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c index c270646faf86..ba2bc7f689eb 100644 --- a/sound/soc/intel/avs/boards/dmic.c +++ b/sound/soc/intel/avs/boards/dmic.c @@ -77,15 +77,23 @@ static int avs_dmic_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_dmic_driver_ids[] = { + { + .name = "avs_dmic", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_dmic_driver_ids); + static struct platform_driver avs_dmic_driver = { .probe = avs_dmic_probe, .driver = { .name = "avs_dmic", .pm = &snd_soc_pm_ops, }, + .id_table = avs_dmic_driver_ids, }; module_platform_driver(avs_dmic_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_dmic"); From 9441450e171fc90aa3fceadae9728acf8bd0726a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:46:58 +0100 Subject: [PATCH 005/337] ASoC: Intel: avs: es8336: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-4-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/es8336.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/es8336.c b/sound/soc/intel/avs/boards/es8336.c index f972ef64d284..1090082e7d5b 100644 --- a/sound/soc/intel/avs/boards/es8336.c +++ b/sound/soc/intel/avs/boards/es8336.c @@ -307,15 +307,23 @@ static int avs_es8336_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_es8336_driver_ids[] = { + { + .name = "avs_es8336", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_es8336_driver_ids); + static struct platform_driver avs_es8336_driver = { .probe = avs_es8336_probe, .driver = { .name = "avs_es8336", .pm = &snd_soc_pm_ops, }, + .id_table = avs_es8336_driver_ids, }; module_platform_driver(avs_es8336_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_es8336"); From 9a872caede56d20564caf30d7ea7cf61b66f4060 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:46:59 +0100 Subject: [PATCH 006/337] ASoC: Intel: avs: hdaudio: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-5-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/hdaudio.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index 8876558f19a1..844a918f9a81 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -218,12 +218,21 @@ static int avs_hdaudio_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_hdaudio_driver_ids[] = { + { + .name = "avs_hdaudio", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_hdaudio_driver_ids); + static struct platform_driver avs_hdaudio_driver = { .probe = avs_hdaudio_probe, .driver = { .name = "avs_hdaudio", .pm = &snd_soc_pm_ops, }, + .id_table = avs_hdaudio_driver_ids, }; module_platform_driver(avs_hdaudio_driver) @@ -231,4 +240,3 @@ module_platform_driver(avs_hdaudio_driver) MODULE_DESCRIPTION("Intel HD-Audio machine driver"); MODULE_AUTHOR("Cezary Rojewski "); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_hdaudio"); From 8267213c54db078db091b778e904ec1af8fc6ee6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:00 +0100 Subject: [PATCH 007/337] ASoC: Intel: avs: i2s_test: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-6-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/i2s_test.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c index 3d03e1eed3a9..28f254eb0d03 100644 --- a/sound/soc/intel/avs/boards/i2s_test.c +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -185,15 +185,23 @@ static int avs_i2s_test_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_i2s_test_driver_ids[] = { + { + .name = "avs_i2s_test", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_i2s_test_driver_ids); + static struct platform_driver avs_i2s_test_driver = { .probe = avs_i2s_test_probe, .driver = { .name = "avs_i2s_test", .pm = &snd_soc_pm_ops, }, + .id_table = avs_i2s_test_driver_ids, }; module_platform_driver(avs_i2s_test_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_i2s_test"); From f1e9f4f5e9e5506edbd17b33065ec8c8a9e6caab Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:01 +0100 Subject: [PATCH 008/337] ASoC: Intel: avs: max98357a: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-7-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/max98357a.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index 6ba7b6564279..a83b95f25129 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -135,15 +135,23 @@ static int avs_max98357a_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_max98357a_driver_ids[] = { + { + .name = "avs_max98357a", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_max98357a_driver_ids); + static struct platform_driver avs_max98357a_driver = { .probe = avs_max98357a_probe, .driver = { .name = "avs_max98357a", .pm = &snd_soc_pm_ops, }, + .id_table = avs_max98357a_driver_ids, }; module_platform_driver(avs_max98357a_driver) MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_max98357a"); From 8e660f303230741ecaab561a30e123e7dc76abda Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:02 +0100 Subject: [PATCH 009/337] ASoC: Intel: avs: max98373: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-8-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/max98373.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c index cc7dfdf72083..3b980a025e6f 100644 --- a/sound/soc/intel/avs/boards/max98373.c +++ b/sound/soc/intel/avs/boards/max98373.c @@ -192,15 +192,23 @@ static int avs_max98373_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_max98373_driver_ids[] = { + { + .name = "avs_max98373", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_max98373_driver_ids); + static struct platform_driver avs_max98373_driver = { .probe = avs_max98373_probe, .driver = { .name = "avs_max98373", .pm = &snd_soc_pm_ops, }, + .id_table = avs_max98373_driver_ids, }; module_platform_driver(avs_max98373_driver) MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_max98373"); From c3ff01859c31408eadfd607352d4f87e52096371 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:03 +0100 Subject: [PATCH 010/337] ASoC: Intel: avs: max98927: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-9-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/max98927.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/max98927.c b/sound/soc/intel/avs/boards/max98927.c index fb0175f37d61..86dd2b228df3 100644 --- a/sound/soc/intel/avs/boards/max98927.c +++ b/sound/soc/intel/avs/boards/max98927.c @@ -189,15 +189,23 @@ static int avs_max98927_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_max98927_driver_ids[] = { + { + .name = "avs_max98927", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_max98927_driver_ids); + static struct platform_driver avs_max98927_driver = { .probe = avs_max98927_probe, .driver = { .name = "avs_max98927", .pm = &snd_soc_pm_ops, }, + .id_table = avs_max98927_driver_ids, }; module_platform_driver(avs_max98927_driver) MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_max98927"); From c94643c2b416afc473541943291e61c453846a6d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:04 +0100 Subject: [PATCH 011/337] ASoC: Intel: avs: nau8825: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-10-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/nau8825.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index d98b5deb78c9..1c1e2083f474 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -294,15 +294,23 @@ static int avs_nau8825_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_nau8825_driver_ids[] = { + { + .name = "avs_nau8825", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_nau8825_driver_ids); + static struct platform_driver avs_nau8825_driver = { .probe = avs_nau8825_probe, .driver = { .name = "avs_nau8825", .pm = &snd_soc_pm_ops, }, + .id_table = avs_nau8825_driver_ids, }; module_platform_driver(avs_nau8825_driver) MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_nau8825"); From 4a5403e3a75ddc24941f754c14c3299cc0e28fe8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:05 +0100 Subject: [PATCH 012/337] ASoC: Intel: avs: probe: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-11-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/probe.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/probe.c b/sound/soc/intel/avs/boards/probe.c index 411acaee74f9..a9469b5ecb40 100644 --- a/sound/soc/intel/avs/boards/probe.c +++ b/sound/soc/intel/avs/boards/probe.c @@ -50,15 +50,23 @@ static int avs_probe_mb_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_probe_mb_driver_ids[] = { + { + .name = "avs_probe_mb", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_probe_mb_driver_ids); + static struct platform_driver avs_probe_mb_driver = { .probe = avs_probe_mb_probe, .driver = { .name = "avs_probe_mb", .pm = &snd_soc_pm_ops, }, + .id_table = avs_probe_mb_driver_ids, }; module_platform_driver(avs_probe_mb_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_probe_mb"); From 54c830fd4e38261adffa51bb22c44a2e33d803ba Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:06 +0100 Subject: [PATCH 013/337] ASoC: Intel: avs: rt274: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-12-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt274.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c index 157183b1de24..bfcb8845fd15 100644 --- a/sound/soc/intel/avs/boards/rt274.c +++ b/sound/soc/intel/avs/boards/rt274.c @@ -257,15 +257,23 @@ static int avs_rt274_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_rt274_driver_ids[] = { + { + .name = "avs_rt274", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_rt274_driver_ids); + static struct platform_driver avs_rt274_driver = { .probe = avs_rt274_probe, .driver = { .name = "avs_rt274", .pm = &snd_soc_pm_ops, }, + .id_table = avs_rt274_driver_ids, }; module_platform_driver(avs_rt274_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_rt274"); From 027ab0cab18071fa3476ffe7bb69ade551515ce6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:07 +0100 Subject: [PATCH 014/337] ASoC: Intel: avs: rt286: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-13-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt286.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c index 131237471e3e..28d7d86b1cc9 100644 --- a/sound/soc/intel/avs/boards/rt286.c +++ b/sound/soc/intel/avs/boards/rt286.c @@ -228,15 +228,23 @@ static int avs_rt286_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_rt286_driver_ids[] = { + { + .name = "avs_rt286", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_rt286_driver_ids); + static struct platform_driver avs_rt286_driver = { .probe = avs_rt286_probe, .driver = { .name = "avs_rt286", .pm = &snd_soc_pm_ops, }, + .id_table = avs_rt286_driver_ids, }; module_platform_driver(avs_rt286_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_rt286"); From 3d4021f30abdb7cef51a901adc0dfc4d4ee98e9d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:08 +0100 Subject: [PATCH 015/337] ASoC: Intel: avs: rt298: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-14-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt298.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c index ea32a7690c8a..80f490b9e118 100644 --- a/sound/soc/intel/avs/boards/rt298.c +++ b/sound/soc/intel/avs/boards/rt298.c @@ -247,15 +247,23 @@ static int avs_rt298_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_rt298_driver_ids[] = { + { + .name = "avs_rt298", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_rt298_driver_ids); + static struct platform_driver avs_rt298_driver = { .probe = avs_rt298_probe, .driver = { .name = "avs_rt298", .pm = &snd_soc_pm_ops, }, + .id_table = avs_rt298_driver_ids, }; module_platform_driver(avs_rt298_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_rt298"); From 389f3c6c7ed89d53ecd83e629b7e529630cdb96c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:09 +0100 Subject: [PATCH 016/337] ASoC: Intel: avs: rt5514: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-15-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt5514.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt5514.c b/sound/soc/intel/avs/boards/rt5514.c index ad486a52e5e3..60105f453ae2 100644 --- a/sound/soc/intel/avs/boards/rt5514.c +++ b/sound/soc/intel/avs/boards/rt5514.c @@ -173,15 +173,23 @@ static int avs_rt5514_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_rt5514_driver_ids[] = { + { + .name = "avs_rt5514", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_rt5514_driver_ids); + static struct platform_driver avs_rt5514_driver = { .probe = avs_rt5514_probe, .driver = { .name = "avs_rt5514", .pm = &snd_soc_pm_ops, }, + .id_table = avs_rt5514_driver_ids, }; module_platform_driver(avs_rt5514_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_rt5514"); From 5f249523d3fcf1fe28e567a17a3053ea7ff899a0 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:10 +0100 Subject: [PATCH 017/337] ASoC: Intel: avs: rt5663: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-16-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt5663.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt5663.c b/sound/soc/intel/avs/boards/rt5663.c index 3effd789a45e..b4762c2a7bf2 100644 --- a/sound/soc/intel/avs/boards/rt5663.c +++ b/sound/soc/intel/avs/boards/rt5663.c @@ -246,15 +246,23 @@ static int avs_rt5663_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_rt5663_driver_ids[] = { + { + .name = "avs_rt5663", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_rt5663_driver_ids); + static struct platform_driver avs_rt5663_driver = { .probe = avs_rt5663_probe, .driver = { .name = "avs_rt5663", .pm = &snd_soc_pm_ops, }, + .id_table = avs_rt5663_driver_ids, }; module_platform_driver(avs_rt5663_driver); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_rt5663"); From ba096fc618254918056061ecef32aa77c2fcaf84 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:11 +0100 Subject: [PATCH 018/337] ASoC: Intel: avs: rt5682: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-17-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/rt5682.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 84e850c0b085..243f979fda98 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -322,16 +322,24 @@ static int avs_rt5682_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_rt5682_driver_ids[] = { + { + .name = "avs_rt5682", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_rt5682_driver_ids); + static struct platform_driver avs_rt5682_driver = { .probe = avs_rt5682_probe, .driver = { .name = "avs_rt5682", .pm = &snd_soc_pm_ops, }, + .id_table = avs_rt5682_driver_ids, }; module_platform_driver(avs_rt5682_driver) MODULE_AUTHOR("Cezary Rojewski "); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_rt5682"); From ca5abf5d2e1c3860382e0e33599e969cb9c9b42b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 2 Nov 2023 13:47:12 +0100 Subject: [PATCH 019/337] ASoC: Intel: avs: ssm4567: Add proper id_table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add id_table and use it instead of alias to load module. Suggested-by: Krzysztof Kozlowski Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231102124712.2549327-18-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/ssm4567.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index 6bcab9deae5c..4a0e136835ff 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -181,15 +181,23 @@ static int avs_ssm4567_probe(struct platform_device *pdev) return devm_snd_soc_register_card(dev, card); } +static const struct platform_device_id avs_ssm4567_driver_ids[] = { + { + .name = "avs_ssm4567", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, avs_ssm4567_driver_ids); + static struct platform_driver avs_ssm4567_driver = { .probe = avs_ssm4567_probe, .driver = { .name = "avs_ssm4567", .pm = &snd_soc_pm_ops, }, + .id_table = avs_ssm4567_driver_ids, }; module_platform_driver(avs_ssm4567_driver) MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:avs_ssm4567"); From 1fb1a7c4a6328aff97eeca513fe7239099c13016 Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Tue, 7 Nov 2023 11:52:29 +0800 Subject: [PATCH 020/337] ASoC: dt-bindings: nau8821: Add DMIC slew rate. Add input with DMIC slew rate controls. Signed-off-by: Seven Lee Acked-by: Conor Dooley Link: https://lore.kernel.org/r/20231107035230.1241683-2-wtli@nuvoton.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index 3e54abd4ca74..3380b6aa9542 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -89,6 +89,14 @@ properties: $ref: /schemas/types.yaml#/definitions/uint32 default: 3072000 + nuvoton,dmic-slew-rate: + description: The range 0 to 7 represents the speed of DMIC slew rate. + The lowest value 0 means the slowest rate and the highest value + 7 means the fastest rate. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + nuvoton,left-input-single-end: description: Enable left input with single-ended settings if set. For the headset mic application, the single-ended control is @@ -127,6 +135,7 @@ examples: nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <0>; nuvoton,dmic-clk-threshold = <3072000>; + nuvoton,dmic-slew-rate= <0>; #sound-dai-cells = <0>; }; }; From 91d1a18b6381abd7a0137449fe345924072e4a32 Mon Sep 17 00:00:00 2001 From: Seven Lee Date: Tue, 7 Nov 2023 11:52:30 +0800 Subject: [PATCH 021/337] ASoC: nau8821: Add slew rate controls. The patch supports DMIC clock slew rate controls. Signed-off-by: Seven Lee Link: https://lore.kernel.org/r/20231107035230.1241683-3-wtli@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8821.c | 7 +++++++ sound/soc/codecs/nau8821.h | 3 +++ 2 files changed, 10 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 6e1b6b26298a..012e347e6391 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -1738,6 +1738,10 @@ static int nau8821_read_device_properties(struct device *dev, &nau8821->dmic_clk_threshold); if (ret) nau8821->dmic_clk_threshold = 3072000; + ret = device_property_read_u32(dev, "nuvoton,dmic-slew-rate", + &nau8821->dmic_slew_rate); + if (ret) + nau8821->dmic_slew_rate = 0; return 0; } @@ -1797,6 +1801,9 @@ static void nau8821_init_regs(struct nau8821 *nau8821) NAU8821_ADC_SYNC_DOWN_MASK, NAU8821_ADC_SYNC_DOWN_64); regmap_update_bits(regmap, NAU8821_R2C_DAC_CTRL1, NAU8821_DAC_OVERSAMPLE_MASK, NAU8821_DAC_OVERSAMPLE_64); + regmap_update_bits(regmap, NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_SLEW_MASK, nau8821->dmic_slew_rate << + NAU8821_DMIC_SLEW_SFT); if (nau8821->left_input_single_end) { regmap_update_bits(regmap, NAU8821_R6B_PGA_MUTE, NAU8821_MUTE_MICNL_EN, NAU8821_MUTE_MICNL_EN); diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h index 00a888ed07ce..62eaad130b2e 100644 --- a/sound/soc/codecs/nau8821.h +++ b/sound/soc/codecs/nau8821.h @@ -236,6 +236,8 @@ #define NAU8821_DMIC_SRC_MASK (0x3 << NAU8821_DMIC_SRC_SFT) #define NAU8821_CLK_DMIC_SRC (0x2 << NAU8821_DMIC_SRC_SFT) #define NAU8821_DMIC_EN_SFT 0 +#define NAU8821_DMIC_SLEW_SFT 8 +#define NAU8821_DMIC_SLEW_MASK (0x7 << NAU8821_DMIC_SLEW_SFT) /* GPIO12_CTRL (0x1a) */ #define NAU8821_JKDET_PULL_UP (0x1 << 11) /* 0 - pull down, 1 - pull up */ @@ -573,6 +575,7 @@ struct nau8821 { int jack_eject_debounce; int fs; int dmic_clk_threshold; + int dmic_slew_rate; int key_enable; }; From ee09084fbf9fdda6bf0179bcdca52196d0cde8a1 Mon Sep 17 00:00:00 2001 From: Zhu Ning Date: Wed, 1 Nov 2023 15:27:00 +0800 Subject: [PATCH 022/337] ASoC: codecs: ES8326: Add chip version flag The new chip version requires the addition of a new clock table. We determine which clock table to choose based on the version. Newer versions of the chip have fewer processes to go through in the headset detection, so the version flag is used to skip them. Signed-off-by: Zhu Ning Link: https://lore.kernel.org/r/20231101072702.91316-2-zhuning0077@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 174 +++++++++++++++++++++++--------------- 1 file changed, 107 insertions(+), 67 deletions(-) mode change 100644 => 100755 sound/soc/codecs/es8326.c diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c old mode 100644 new mode 100755 index 6c263086c44d..7400c3d8af23 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -198,77 +198,108 @@ struct _coeff_div { /* codec hifi mclk clock divider coefficients */ /* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ -static const struct _coeff_div coeff_div[] = { - {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, - {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, - {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, - {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, - {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, - {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, - {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, - {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, +static const struct _coeff_div coeff_div_v0[] = { + {64, 8000, 512000, 0x60, 0x01, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x33, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {64, 44100, 2822400, 0xE0, 0x00, 0x03, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0xE0, 0x00, 0x03, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {128, 8000, 1024000, 0x60, 0x00, 0x33, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {128, 44100, 5644800, 0xE0, 0x01, 0x03, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0xE0, 0x01, 0x03, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, - {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, - {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, - {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, - {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, - {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, - {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, - {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, - {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {192, 32000, 6144000, 0xE0, 0x02, 0x03, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {256, 8000, 2048000, 0x60, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x01, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {256, 44100, 11289600, 0xE0, 0x00, 0x30, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0xE0, 0x00, 0x30, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {384, 32000, 12288000, 0xE0, 0x05, 0x03, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0xE9, 0x04, 0x0F, 0x6d, 0x4A, 0x0A, 0x1F, 0x1F}, - {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, - {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, - {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, - {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, - {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, - {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, - {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, - - {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, - {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, - {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, - {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, - {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, - {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, - - {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, - {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, - {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, - {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {500, 48000, 24000000, 0xF8, 0x04, 0x3F, 0x6D, 0x4A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x01, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {768, 32000, 24576000, 0xE0, 0x02, 0x30, 0x2D, 0x4A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x30, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, - {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, - {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, - - {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, - {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, - {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, - {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, - {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, - {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, - {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, - }; -static inline int get_coeff(int mclk, int rate) +static const struct _coeff_div coeff_div_v3[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x8A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x8A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x8A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x8A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x8A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x8A, 0x0A, 0x1F, 0x1F}, + + {64, 8000, 512000, 0x60, 0x00, 0x35, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x3F}, + {64, 44100, 2822400, 0xE0, 0x00, 0x31, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0xE0, 0x00, 0x31, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x8A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0xB8, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27}, + + {128, 8000, 1024000, 0x60, 0x00, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {128, 44100, 5644800, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0xE0, 0x00, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x8A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x8A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, + {192, 32000, 6144000, 0xE0, 0x02, 0x31, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0xCA, 0x1B, 0x1F, 0x3F}, + + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0xCA, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x31, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {256, 44100, 11289600, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0xE0, 0x00, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x8A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {384, 32000, 12288000, 0xE0, 0x02, 0x01, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x8A, 0x0A, 0x1F, 0x1F}, + + {400, 48000, 19200000, 0xE4, 0x04, 0x35, 0x6d, 0xCA, 0x0A, 0x1F, 0x1F}, + {500, 48000, 24000000, 0xF8, 0x04, 0x3F, 0x6D, 0xCA, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x30, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {512, 44100, 22579200, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0xE0, 0x00, 0x00, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {768, 32000, 24576000, 0xE0, 0x02, 0x30, 0x2D, 0xCA, 0x0A, 0x1F, 0x1F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x8A, 0x0A, 0x1F, 0x1F}, + + {1024, 8000, 8192000, 0x60, 0x00, 0x30, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x8A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x8A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x8A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x8A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x8A, 0x0A, 0x27, 0x27}, +}; + +static inline int get_coeff(int mclk, int rate, int array, + const struct _coeff_div *coeff_div) { int i; - for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + for (i = 0; i < array; i++) { if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) return i; } @@ -276,6 +307,7 @@ static inline int get_coeff(int mclk, int rate) return -EINVAL; } + static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -333,11 +365,19 @@ static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + const struct _coeff_div *coeff_div; struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); u8 srate = 0; - int coeff; + int coeff, array; - coeff = get_coeff(es8326->sysclk, params_rate(params)); + if (es8326->version == 0) { + coeff_div = coeff_div_v0; + array = ARRAY_SIZE(coeff_div_v0); + } else { + coeff_div = coeff_div_v3; + array = ARRAY_SIZE(coeff_div_v3); + } + coeff = get_coeff(es8326->sysclk, params_rate(params), array, coeff_div); /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -594,7 +634,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) iface = snd_soc_component_read(comp, ES8326_HPDET_STA); dev_dbg(comp->dev, "gpio flag %#04x", iface); - if (es8326->jack_remove_retry == 1) { + if ((es8326->jack_remove_retry == 1) && (es8326->version != ES8326_VERSION_B)) { if (iface & ES8326_HPINSERT_FLAG) es8326->jack_remove_retry = 2; else @@ -628,7 +668,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ - if (es8326->jack_remove_retry == 0) { + if ((es8326->jack_remove_retry == 0) && (es8326->version != ES8326_VERSION_B)) { es8326->jack_remove_retry = 1; dev_dbg(comp->dev, "remove event check, invert HPJACK_POL, cnt = %d\n", es8326->jack_remove_retry); From fc702b2c04d778d7e3a4091ebe54a86c5d0a0d96 Mon Sep 17 00:00:00 2001 From: Zhu Ning Date: Wed, 1 Nov 2023 15:27:01 +0800 Subject: [PATCH 023/337] ASoC: codecs: ES8326: Changing initialisation and broadcasting New chip versions require new initialisation and playback processes. Changing the initialisation and playback process for better results. The old chip versions are going to work well with the new sequences. We've tested this with version_v0 and version_v3 chips under the new sequence and they both pass. Signed-off-by: Zhu Ning Link: https://lore.kernel.org/r/20231101072702.91316-3-zhuning0077@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 51 ++++++++++++++++++++++----------------- 1 file changed, 29 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 7400c3d8af23..772788811197 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -132,6 +132,11 @@ static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, + 4, 7, 0, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, + 0, 7, 0, 0), + SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), }; @@ -156,6 +161,9 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { {"LHPMIX", NULL, "Left DAC"}, {"RHPMIX", NULL, "Right DAC"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "LHPMIX"}, {"HPOR", NULL, "RHPMIX"}, }; @@ -307,7 +315,6 @@ static inline int get_coeff(int mclk, int rate, int array, return -EINVAL; } - static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -449,8 +456,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } - regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa0); - regmap_write(es8326->regmap, ES8326_HP_VOL, 0x80); + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa1); + regmap_write(es8326->regmap, ES8326_HP_VOL, 0x91); regmap_write(es8326->regmap, ES8326_HP_CAL, ES8326_HP_ON); regmap_update_bits(es8326->regmap, ES8326_DAC_MUTE, ES8326_MUTE_MASK, ~(ES8326_MUTE)); @@ -470,8 +477,6 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, if (ret) return ret; - regmap_write(es8326->regmap, ES8326_RESET, 0x9f); - msleep(20); regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x00); regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, @@ -480,19 +485,21 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_PGA_PDN, 0x40); regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); - regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); + + regmap_update_bits(es8326->regmap, ES8326_RESET, + ES8326_CSM_ON, ES8326_CSM_ON); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - break; - case SND_SOC_BIAS_OFF: - clk_disable_unprepare(es8326->mclk); regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(es8326->mclk); + break; } return 0; @@ -762,13 +769,15 @@ static int es8326_calibrate(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_CLK_DIV1, 0x01); regmap_write(es8326->regmap, ES8326_CLK_DLL, 0x30); regmap_write(es8326->regmap, ES8326_CLK_MUX, 0xed); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL, 0x08); regmap_write(es8326->regmap, ES8326_CLK_TRI, 0xc1); regmap_write(es8326->regmap, ES8326_DAC_MUTE, 0x03); regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7f); - regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x33); + regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x03); regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x88); - regmap_write(es8326->regmap, ES8326_HP_VOL, 0x80); + usleep_range(15000, 20000); regmap_write(es8326->regmap, ES8326_HP_OFFSET_CAL, 0x8c); + usleep_range(15000, 20000); regmap_write(es8326->regmap, ES8326_RESET, 0xc0); usleep_range(15000, 20000); @@ -806,27 +815,27 @@ static int es8326_resume(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_RESET, 0x1f); regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); usleep_range(10000, 15000); - regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0x88); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xe9); + regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0x4b); /* set headphone default type and detect pin */ - regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x81); + regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83); regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05); + regmap_write(es8326->regmap, ES8326_HP_MISC, 0x30); /* set internal oscillator as clock source of headpone cp */ - regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC, 0x84); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC, 0x89); regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_ON); /* clock manager reset release */ regmap_write(es8326->regmap, ES8326_RESET, 0x17); /* set headphone detection as half scan mode */ - regmap_write(es8326->regmap, ES8326_HP_MISC, 0x08); + regmap_write(es8326->regmap, ES8326_HP_MISC, 0x30); regmap_write(es8326->regmap, ES8326_PULLUP_CTL, 0x00); /* enable headphone driver */ regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa7); usleep_range(2000, 5000); - regmap_write(es8326->regmap, ES8326_HP_DRIVER_REF, 0xab); - usleep_range(2000, 5000); - regmap_write(es8326->regmap, ES8326_HP_DRIVER_REF, 0xbb); - usleep_range(2000, 5000); + regmap_write(es8326->regmap, ES8326_HP_DRIVER_REF, 0xa3); + regmap_write(es8326->regmap, ES8326_HP_DRIVER_REF, 0xb3); regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xa1); regmap_write(es8326->regmap, ES8326_CLK_INV, 0x00); @@ -840,9 +849,6 @@ static int es8326_resume(struct snd_soc_component *component) /* set ADC and DAC in low power mode */ regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); - /* force micbias on */ - regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0x4f); - regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x08); regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F); /* select vdda as micbias source */ regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x23); @@ -870,6 +876,7 @@ static int es8326_resume(struct snd_soc_component *component) ((es8326->version == ES8326_VERSION_B) ? (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol) : (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol | 0x04))); + regmap_write(es8326->regmap, ES8326_HP_VOL, 0x11); es8326->jack_remove_retry = 0; es8326->hp = 0; From 8a81491adbd9b25a648704c9825adaefb0c31868 Mon Sep 17 00:00:00 2001 From: Zhu Ning Date: Wed, 1 Nov 2023 15:27:02 +0800 Subject: [PATCH 024/337] ASoC: codecs: ES8326: Changing the headset detection time The old headset detection time is not enough for the new chip version. An error occurs with the old detection time.According to tests, 400ms is the best detection time that does not trigger an error. The delay time after the trigger is reduced by 300ms to keep the whole detection time unchanged. Signed-off-by: Zhu Ning Link: https://lore.kernel.org/r/20231101072702.91316-4-zhuning0077@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 772788811197..fa890f6205e2 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -691,14 +691,14 @@ static void es8326_jack_detect_handler(struct work_struct *work) if (es8326->hp == 0) { dev_dbg(comp->dev, "First insert, start OMTP/CTIA type check\n"); /* - * set auto-check mode, then restart jack_detect_work after 100ms. + * set auto-check mode, then restart jack_detect_work after 400ms. * Don't report jack status. */ regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); queue_delayed_work(system_wq, &es8326->jack_detect_work, - msecs_to_jiffies(100)); + msecs_to_jiffies(400)); es8326->hp = 1; goto exit; } @@ -748,7 +748,7 @@ static irqreturn_t es8326_irq(int irq, void *dev_id) msecs_to_jiffies(10)); else queue_delayed_work(system_wq, &es8326->jack_detect_work, - msecs_to_jiffies(600)); + msecs_to_jiffies(300)); out: return IRQ_HANDLED; From ab475966455ce285c2c9978a3e3bfe97d75ff8d4 Mon Sep 17 00:00:00 2001 From: Trevor Wu Date: Fri, 3 Nov 2023 17:54:30 +0800 Subject: [PATCH 025/337] ASoC: SOF: mediatek: mt8195: clean up unused code Since there are some variables that are no longer being used, we remove the code that was implemented for those variables. Signed-off-by: Trevor Wu Reviewed-by: Yaochun Hung Reviewed-by: AngeloGioacchino Del Regno Tested-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20231103095433.10475-2-trevor.wu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8195/mt8195.c | 49 -------------------------- 1 file changed, 49 deletions(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index cac0a085f60a..8ee7ee246344 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -96,29 +96,6 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) struct mtk_adsp_chip_info *adsp = data; int ret; - mem_region = of_parse_phandle(dev->of_node, "memory-region", 0); - if (!mem_region) { - dev_err(dev, "no dma memory-region phandle\n"); - return -ENODEV; - } - - ret = of_address_to_resource(mem_region, 0, &res); - of_node_put(mem_region); - if (ret) { - dev_err(dev, "of_address_to_resource dma failed\n"); - return ret; - } - - dev_dbg(dev, "DMA %pR\n", &res); - - adsp->pa_shared_dram = (phys_addr_t)res.start; - adsp->shared_size = resource_size(&res); - if (adsp->pa_shared_dram & DRAM_REMAP_MASK) { - dev_err(dev, "adsp shared dma memory(%#x) is not 4K-aligned\n", - (u32)adsp->pa_shared_dram); - return -EINVAL; - } - ret = of_reserved_mem_device_init(dev); if (ret) { dev_err(dev, "of_reserved_mem_device_init failed\n"); @@ -238,26 +215,6 @@ static int adsp_memory_remap_init(struct device *dev, struct mtk_adsp_chip_info return 0; } -static int adsp_shared_base_ioremap(struct platform_device *pdev, void *data) -{ - struct device *dev = &pdev->dev; - struct mtk_adsp_chip_info *adsp = data; - - /* remap shared-dram base to be non-cachable */ - adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, - adsp->shared_size); - if (!adsp->shared_dram) { - dev_err(dev, "failed to ioremap base %pa size %#x\n", - adsp->shared_dram, adsp->shared_size); - return -ENOMEM; - } - - dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", - adsp->shared_dram, &adsp->pa_shared_dram, adsp->shared_size); - - return 0; -} - static int mt8195_run(struct snd_sof_dev *sdev) { u32 adsp_bootup_addr; @@ -338,12 +295,6 @@ static int mt8195_dsp_probe(struct snd_sof_dev *sdev) } priv->adsp->va_dram = sdev->bar[SOF_FW_BLK_TYPE_SRAM]; - ret = adsp_shared_base_ioremap(pdev, priv->adsp); - if (ret) { - dev_err(sdev->dev, "adsp_shared_base_ioremap fail!\n"); - goto err_adsp_sram_power_off; - } - sdev->bar[DSP_REG_BAR] = priv->adsp->va_cfgreg; sdev->mmio_bar = SOF_FW_BLK_TYPE_SRAM; From a4de5a345cf76c6f294a906e11c2fb675a46fd3d Mon Sep 17 00:00:00 2001 From: Trevor Wu Date: Fri, 3 Nov 2023 17:54:31 +0800 Subject: [PATCH 026/337] ASoC: SOF: mediatek: mt8186: clean up unused code Since there are some variables that are no longer being used, we remove the code that was implemented for those variables. Signed-off-by: Trevor Wu Reviewed-by: Yaochun Hung Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20231103095433.10475-3-trevor.wu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8186/mt8186.c | 49 -------------------------- 1 file changed, 49 deletions(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index b69fa788b16f..0d2d7d697de0 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -96,29 +96,6 @@ static int platform_parse_resource(struct platform_device *pdev, void *data) struct mtk_adsp_chip_info *adsp = data; int ret; - mem_region = of_parse_phandle(dev->of_node, "memory-region", 0); - if (!mem_region) { - dev_err(dev, "no dma memory-region phandle\n"); - return -ENODEV; - } - - ret = of_address_to_resource(mem_region, 0, &res); - of_node_put(mem_region); - if (ret) { - dev_err(dev, "of_address_to_resource dma failed\n"); - return ret; - } - - dev_dbg(dev, "DMA %pR\n", &res); - - adsp->pa_shared_dram = (phys_addr_t)res.start; - adsp->shared_size = resource_size(&res); - if (adsp->pa_shared_dram & DRAM_REMAP_MASK) { - dev_err(dev, "adsp shared dma memory(%#x) is not 4K-aligned\n", - (u32)adsp->pa_shared_dram); - return -EINVAL; - } - ret = of_reserved_mem_device_init(dev); if (ret) { dev_err(dev, "of_reserved_mem_device_init failed\n"); @@ -248,26 +225,6 @@ static int adsp_memory_remap_init(struct snd_sof_dev *sdev, struct mtk_adsp_chip return 0; } -static int adsp_shared_base_ioremap(struct platform_device *pdev, void *data) -{ - struct device *dev = &pdev->dev; - struct mtk_adsp_chip_info *adsp = data; - - /* remap shared-dram base to be non-cachable */ - adsp->shared_dram = devm_ioremap(dev, adsp->pa_shared_dram, - adsp->shared_size); - if (!adsp->shared_dram) { - dev_err(dev, "failed to ioremap base %pa size %#x\n", - adsp->shared_dram, adsp->shared_size); - return -ENOMEM; - } - - dev_dbg(dev, "shared-dram vbase=%p, phy addr :%pa, size=%#x\n", - adsp->shared_dram, &adsp->pa_shared_dram, adsp->shared_size); - - return 0; -} - static int mt8186_run(struct snd_sof_dev *sdev) { u32 adsp_bootup_addr; @@ -324,12 +281,6 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) priv->adsp->va_dram = sdev->bar[SOF_FW_BLK_TYPE_SRAM]; - ret = adsp_shared_base_ioremap(pdev, priv->adsp); - if (ret) { - dev_err(sdev->dev, "adsp_shared_base_ioremap fail!\n"); - return ret; - } - sdev->bar[DSP_REG_BAR] = priv->adsp->va_cfgreg; sdev->bar[DSP_SECREG_BAR] = priv->adsp->va_secreg; sdev->bar[DSP_BUSREG_BAR] = priv->adsp->va_busreg; From a08ee9d00e3120e2e77a3fc093ba29070a3979be Mon Sep 17 00:00:00 2001 From: Trevor Wu Date: Fri, 3 Nov 2023 17:54:32 +0800 Subject: [PATCH 027/337] ASoC: SOF: mediatek: remove unused variables To prevent confusion on the follow-up platform, it is necessary to remove any unused variables within the struct mtk_adsp_chip_info. Signed-off-by: Trevor Wu Reviewed-by: Yaochun Hung Reviewed-by: AngeloGioacchino Del Regno Link: https://lore.kernel.org/r/20231103095433.10475-4-trevor.wu@mediatek.com Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/adsp_helper.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/sof/mediatek/adsp_helper.h b/sound/soc/sof/mediatek/adsp_helper.h index d41e904e6614..35527567962e 100644 --- a/sound/soc/sof/mediatek/adsp_helper.h +++ b/sound/soc/sof/mediatek/adsp_helper.h @@ -15,17 +15,13 @@ struct mtk_adsp_chip_info { phys_addr_t pa_sram; phys_addr_t pa_dram; /* adsp dram physical base */ - phys_addr_t pa_shared_dram; /* adsp dram physical base */ phys_addr_t pa_cfgreg; u32 sramsize; u32 dramsize; u32 cfgregsize; - u32 shared_size; void __iomem *va_sram; /* corresponding to pa_sram */ void __iomem *va_dram; /* corresponding to pa_dram */ void __iomem *va_cfgreg; - void __iomem *shared_sram; /* part of va_sram */ - void __iomem *shared_dram; /* part of va_dram */ phys_addr_t adsp_bootup_addr; int dram_offset; /*dram offset between system and dsp view*/ From bb341f75a05238ccd35b1ec1eb1849a3955eebb3 Mon Sep 17 00:00:00 2001 From: Keguang Zhang Date: Mon, 6 Nov 2023 20:51:03 +0800 Subject: [PATCH 028/337] ASoC: sti-uniperf: Use default pcm_config instead The sti-uniperf pcm_config is the same as the default pcm_config. Since commit 43556516fffe ("ASoC: soc-generic-dmaengine-pcm: Use default config when none is given"), passing a NULL pointer could let this driver use the default config. Signed-off-by: Keguang Zhang Link: https://lore.kernel.org/r/20231106-sti-uniperf-v1-1-b2d8749cfa2e@gmail.com Signed-off-by: Mark Brown --- sound/soc/sti/sti_uniperif.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index 2c21a86421e6..ba824f14a39c 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -461,10 +461,6 @@ static int sti_uniperiph_cpu_dai_of(struct device_node *node, return 0; } -static const struct snd_dmaengine_pcm_config dmaengine_pcm_config = { - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, -}; - static int sti_uniperiph_probe(struct platform_device *pdev) { struct sti_uniperiph_data *priv; @@ -493,8 +489,7 @@ static int sti_uniperiph_probe(struct platform_device *pdev) if (ret < 0) return ret; - return devm_snd_dmaengine_pcm_register(&pdev->dev, - &dmaengine_pcm_config, 0); + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } static struct platform_driver sti_uniperiph_driver = { From 79323dc80318aab6c2d05abdbcf88f09b50522c5 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Thu, 26 Oct 2023 13:35:49 +0200 Subject: [PATCH 029/337] ASoC: dapm: Simplify widget clone MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit New DAPM widgets are created based on a provided template. When cloning the data, the name and stream name also need to be cloned. Currently the data and the names are initialized in different places. Simplify the code by having entire initialization in one place. Reviewed-by: Cezary Rojewski Signed-off-by: Amadeusz Sławiński Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20231026113549.1897368-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 30 ++++++++++++++---------------- 1 file changed, 14 insertions(+), 16 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3844f777c87b..407a26af3eaf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -320,7 +320,8 @@ EXPORT_SYMBOL_GPL(dapm_mark_endpoints_dirty); /* create a new dapm widget */ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( - const struct snd_soc_dapm_widget *_widget) + const struct snd_soc_dapm_widget *_widget, + const char *prefix) { struct snd_soc_dapm_widget *w; @@ -328,13 +329,19 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( if (!w) return NULL; - /* - * w->name is duplicated in caller, but w->sname isn't. - * Duplicate it here if defined - */ + if (prefix) + w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, _widget->name); + else + w->name = kstrdup_const(_widget->name, GFP_KERNEL); + if (!w->name) { + kfree(w); + return NULL; + } + if (_widget->sname) { w->sname = kstrdup_const(_widget->sname, GFP_KERNEL); if (!w->sname) { + kfree_const(w->name); kfree(w); return NULL; } @@ -3629,20 +3636,12 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, { enum snd_soc_dapm_direction dir; struct snd_soc_dapm_widget *w; - const char *prefix; int ret = -ENOMEM; - if ((w = dapm_cnew_widget(widget)) == NULL) + w = dapm_cnew_widget(widget, soc_dapm_prefix(dapm)); + if (!w) goto cnew_failed; - prefix = soc_dapm_prefix(dapm); - if (prefix) - w->name = kasprintf(GFP_KERNEL, "%s %s", prefix, widget->name); - else - w->name = kstrdup_const(widget->name, GFP_KERNEL); - if (!w->name) - goto name_failed; - switch (w->id) { case snd_soc_dapm_regulator_supply: w->regulator = devm_regulator_get(dapm->dev, widget->name); @@ -3767,7 +3766,6 @@ request_failed: dev_err_probe(dapm->dev, ret, "ASoC: Failed to request %s\n", w->name); kfree_const(w->name); -name_failed: kfree_const(w->sname); kfree(w); cnew_failed: From 60b4a86cc6b83ccd1e4fb6a74f28a127b8025bce Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Tue, 31 Oct 2023 19:29:32 +0530 Subject: [PATCH 030/337] ASoC: amd: acp: Fix for indentation issue Fix indentation issue reported in acp70_pcm_resume() function. Fixes: e84db124cb21 (ASoC: amd: acp: Add pci legacy driver support for acp7.0 platform") Signed-off-by: Syed Saba Kareem Link: https://lore.kernel.org/r/20231031135949.1064581-1-Syed.SabaKareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp70.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index dd384c966ae9..1d699c96f619 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -229,8 +229,8 @@ static int __maybe_unused acp70_pcm_resume(struct device *dev) } } } - spin_unlock(&adata->acp_lock); - return 0; + spin_unlock(&adata->acp_lock); + return 0; } static const struct dev_pm_ops acp70_dma_pm_ops = { From 970f88ad0026ba8427e319b1da932c161723db82 Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Tue, 31 Oct 2023 19:29:33 +0530 Subject: [PATCH 031/337] ASoC: amd: acp: correct the format order Correct the formats order for dai driver structures. Signed-off-by: Syed Saba Kareem Link: https://lore.kernel.org/r/20231031135949.1064581-2-Syed.SabaKareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp70.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index 1d699c96f619..0d7cdd4017e5 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -52,8 +52,8 @@ static struct snd_soc_dai_driver acp70_dai[] = { .playback = { .stream_name = "I2S SP Playback", .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rate_min = 8000, @@ -62,8 +62,8 @@ static struct snd_soc_dai_driver acp70_dai[] = { .capture = { .stream_name = "I2S SP Capture", .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 2, .rate_min = 8000, @@ -77,8 +77,8 @@ static struct snd_soc_dai_driver acp70_dai[] = { .playback = { .stream_name = "I2S BT Playback", .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rate_min = 8000, @@ -87,8 +87,8 @@ static struct snd_soc_dai_driver acp70_dai[] = { .capture = { .stream_name = "I2S BT Capture", .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 2, .rate_min = 8000, @@ -102,8 +102,8 @@ static struct snd_soc_dai_driver acp70_dai[] = { .playback = { .stream_name = "I2S HS Playback", .rates = SNDRV_PCM_RATE_8000_96000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rate_min = 8000, @@ -112,8 +112,8 @@ static struct snd_soc_dai_driver acp70_dai[] = { .capture = { .stream_name = "I2S HS Capture", .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, .channels_max = 8, .rate_min = 8000, From 6d02f355c3d2fe86f503793e4df09898e9f9e703 Mon Sep 17 00:00:00 2001 From: Rob Herring Date: Wed, 1 Nov 2023 09:09:22 -0500 Subject: [PATCH 032/337] ASoC: dt-bindings: Simplify port schema The use of 'definitions' is not necessary and also problematic because the dtschema tools don't process 'definitions' resulting in this spurious warning: Documentation/devicetree/bindings/sound/renesas,rsnd.example.dtb: sound@ec500000: port:endpoint: Unevaluated properties are not allowed ('phandle' was unexpected) from schema $id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# Fix this by moving the port schema to #/properties/port and referencing that directly from the 'ports' schema. Really, a binding should just always use 'ports' if multiple ports are possible. There's no benefit to supporting both forms. However, it appears there are already lots of users of this one with a single 'port' node. Signed-off-by: Rob Herring Link: https://lore.kernel.org/r/20231101140923.16344-2-robh@kernel.org Signed-off-by: Mark Brown --- .../bindings/sound/renesas,rsnd.yaml | 28 ++++++++----------- 1 file changed, 12 insertions(+), 16 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 13a5a0a10fe6..1174205286d4 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -9,20 +9,6 @@ title: Renesas R-Car Sound Driver maintainers: - Kuninori Morimoto -definitions: - port-def: - $ref: audio-graph-port.yaml#/definitions/port-base - unevaluatedProperties: false - patternProperties: - "^endpoint(@[0-9a-f]+)?": - $ref: audio-graph-port.yaml#/definitions/endpoint-base - properties: - playback: - $ref: /schemas/types.yaml#/definitions/phandle-array - capture: - $ref: /schemas/types.yaml#/definitions/phandle-array - unevaluatedProperties: false - properties: compatible: @@ -125,7 +111,17 @@ properties: # ports is below port: - $ref: "#/definitions/port-def" + $ref: audio-graph-port.yaml#/definitions/port-base + unevaluatedProperties: false + patternProperties: + "^endpoint(@[0-9a-f]+)?": + $ref: audio-graph-port.yaml#/definitions/endpoint-base + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + unevaluatedProperties: false rcar_sound,dvc: description: DVC subnode. @@ -269,7 +265,7 @@ patternProperties: unevaluatedProperties: false patternProperties: '^port(@[0-9a-f]+)?$': - $ref: "#/definitions/port-def" + $ref: "#/properties/port" required: - compatible From 8df735701a7051825254ec7a12a661307bb7bdc1 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sat, 11 Nov 2023 18:37:39 +0100 Subject: [PATCH 033/337] ASoC: tegra: convert not to use dma_request_slave_channel() dma_request_slave_channel() is deprecated. dma_request_chan() should be used directly instead. Switch to the preferred function and update the error handling accordingly. Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/b78685e4103f12931ddb09c1654bc6b04b640868.1699724240.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 142e8d4eefd5..42acb56543db 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -98,8 +98,8 @@ int tegra_pcm_open(struct snd_soc_component *component, return ret; } - chan = dma_request_slave_channel(cpu_dai->dev, dmap->chan_name); - if (!chan) { + chan = dma_request_chan(cpu_dai->dev, dmap->chan_name); + if (IS_ERR(chan)) { dev_err(cpu_dai->dev, "dmaengine request slave channel failed! (%s)\n", dmap->chan_name); From fc213b8d4466d1fcf39ab760979be4eac2a67635 Mon Sep 17 00:00:00 2001 From: Weidong Wang Date: Thu, 9 Nov 2023 17:37:07 +0800 Subject: [PATCH 034/337] ASoC: codecs: Modify the bin file parsing method Modify the aw88395_lib file so that the bin file parsing is no longer related to the chip id of the chip. Adopt the bin file data type "prof_data_type" as the differentiation between different chip bin file parsing methods. Since the chip id macro for the aw88399 is no longer defined in aw88395_reg.h, define the chip id for the aw88399 in aw88399.h Signed-off-by: Weidong Wang Link: https://lore.kernel.org/r/20231109093708.13155-1-wangweidong.a@awinic.com Signed-off-by: Mark Brown --- sound/soc/codecs/aw88395/aw88395_device.h | 1 + sound/soc/codecs/aw88395/aw88395_lib.c | 124 +++++++--------------- sound/soc/codecs/aw88395/aw88395_reg.h | 3 - sound/soc/codecs/aw88399.c | 1 - sound/soc/codecs/aw88399.h | 1 + 5 files changed, 42 insertions(+), 88 deletions(-) diff --git a/sound/soc/codecs/aw88395/aw88395_device.h b/sound/soc/codecs/aw88395/aw88395_device.h index 791c8c106557..0f750f654f3e 100644 --- a/sound/soc/codecs/aw88395/aw88395_device.h +++ b/sound/soc/codecs/aw88395/aw88395_device.h @@ -146,6 +146,7 @@ struct aw_device { unsigned int channel; unsigned int fade_step; + unsigned int prof_data_type; struct i2c_client *i2c; struct device *dev; diff --git a/sound/soc/codecs/aw88395/aw88395_lib.c b/sound/soc/codecs/aw88395/aw88395_lib.c index 9ebe7c510109..f25f6e0d4428 100644 --- a/sound/soc/codecs/aw88395/aw88395_lib.c +++ b/sound/soc/codecs/aw88395/aw88395_lib.c @@ -11,7 +11,6 @@ #include #include "aw88395_lib.h" #include "aw88395_device.h" -#include "aw88395_reg.h" #define AW88395_CRC8_POLYNOMIAL 0x8C DECLARE_CRC8_TABLE(aw_crc8_table); @@ -456,14 +455,6 @@ static int aw_dev_parse_reg_bin_with_hdr(struct aw_device *aw_dev, goto parse_bin_failed; } - if (aw_dev->chip_id == AW88261_CHIP_ID) { - if (aw_bin->header_info[0].valid_data_len % 4) { - dev_err(aw_dev->dev, "bin data len get error!"); - ret = -EINVAL; - goto parse_bin_failed; - } - } - prof_desc->sec_desc[AW88395_DATA_TYPE_REG].data = data + aw_bin->header_info[0].valid_data_addr; prof_desc->sec_desc[AW88395_DATA_TYPE_REG].len = @@ -528,7 +519,7 @@ static int aw_dev_parse_dev_type(struct aw_device *aw_dev, cfg_dde[i].dev_profile); return -EINVAL; } - + aw_dev->prof_data_type = cfg_dde[i].data_type; ret = aw_dev_parse_data_by_sec_type(aw_dev, prof_hdr, &cfg_dde[i], &all_prof_info->prof_desc[cfg_dde[i].dev_profile]); if (ret < 0) { @@ -564,6 +555,7 @@ static int aw_dev_parse_dev_default_type(struct aw_device *aw_dev, cfg_dde[i].dev_profile); return -EINVAL; } + aw_dev->prof_data_type = cfg_dde[i].data_type; ret = aw_dev_parse_data_by_sec_type(aw_dev, prof_hdr, &cfg_dde[i], &all_prof_info->prof_desc[cfg_dde[i].dev_profile]); if (ret < 0) { @@ -582,7 +574,7 @@ static int aw_dev_parse_dev_default_type(struct aw_device *aw_dev, return 0; } -static int aw88261_dev_cfg_get_valid_prof(struct aw_device *aw_dev, +static int aw_dev_cfg_get_reg_valid_prof(struct aw_device *aw_dev, struct aw_all_prof_info *all_prof_info) { struct aw_prof_desc *prof_desc = all_prof_info->prof_desc; @@ -624,7 +616,7 @@ static int aw88261_dev_cfg_get_valid_prof(struct aw_device *aw_dev, return 0; } -static int aw88395_dev_cfg_get_valid_prof(struct aw_device *aw_dev, +static int aw_dev_cfg_get_multiple_valid_prof(struct aw_device *aw_dev, struct aw_all_prof_info *all_prof_info) { struct aw_prof_desc *prof_desc = all_prof_info->prof_desc; @@ -703,26 +695,20 @@ static int aw_dev_load_cfg_by_hdr(struct aw_device *aw_dev, goto exit; } - switch (aw_dev->chip_id) { - case AW88395_CHIP_ID: - case AW88399_CHIP_ID: - ret = aw88395_dev_cfg_get_valid_prof(aw_dev, all_prof_info); - if (ret < 0) - goto exit; + switch (aw_dev->prof_data_type) { + case ACF_SEC_TYPE_MULTIPLE_BIN: + ret = aw_dev_cfg_get_multiple_valid_prof(aw_dev, all_prof_info); break; - case AW88261_CHIP_ID: - case AW87390_CHIP_ID: - ret = aw88261_dev_cfg_get_valid_prof(aw_dev, all_prof_info); - if (ret < 0) - goto exit; + case ACF_SEC_TYPE_HDR_REG: + ret = aw_dev_cfg_get_reg_valid_prof(aw_dev, all_prof_info); break; default: - dev_err(aw_dev->dev, "valid prof unsupported"); + dev_err(aw_dev->dev, "unsupport data type\n"); ret = -EINVAL; break; } - - aw_dev->prof_info.prof_name_list = profile_name; + if (!ret) + aw_dev->prof_info.prof_name_list = profile_name; exit: devm_kfree(aw_dev->dev, all_prof_info); @@ -791,39 +777,23 @@ static int aw_get_dev_scene_count_v1(struct aw_device *aw_dev, struct aw_contain struct aw_cfg_dde_v1 *cfg_dde = (struct aw_cfg_dde_v1 *)(aw_cfg->data + cfg_hdr->hdr_offset); unsigned int i; - int ret; - switch (aw_dev->chip_id) { - case AW88395_CHIP_ID: - case AW88399_CHIP_ID: - for (i = 0; i < cfg_hdr->ddt_num; ++i) { - if ((cfg_dde[i].data_type == ACF_SEC_TYPE_MULTIPLE_BIN) && - (aw_dev->chip_id == cfg_dde[i].chip_id) && - (aw_dev->i2c->adapter->nr == cfg_dde[i].dev_bus) && - (aw_dev->i2c->addr == cfg_dde[i].dev_addr)) - (*scene_num)++; - } - ret = 0; - break; - case AW88261_CHIP_ID: - case AW87390_CHIP_ID: - for (i = 0; i < cfg_hdr->ddt_num; ++i) { - if (((cfg_dde[i].data_type == ACF_SEC_TYPE_REG) || - (cfg_dde[i].data_type == ACF_SEC_TYPE_HDR_REG)) && - (aw_dev->chip_id == cfg_dde[i].chip_id) && - (aw_dev->i2c->adapter->nr == cfg_dde[i].dev_bus) && - (aw_dev->i2c->addr == cfg_dde[i].dev_addr)) - (*scene_num)++; - } - ret = 0; - break; - default: - dev_err(aw_dev->dev, "unsupported device"); - ret = -EINVAL; - break; + for (i = 0; i < cfg_hdr->ddt_num; ++i) { + if (((cfg_dde[i].data_type == ACF_SEC_TYPE_REG) || + (cfg_dde[i].data_type == ACF_SEC_TYPE_HDR_REG) || + (cfg_dde[i].data_type == ACF_SEC_TYPE_MULTIPLE_BIN)) && + (aw_dev->chip_id == cfg_dde[i].chip_id) && + (aw_dev->i2c->adapter->nr == cfg_dde[i].dev_bus) && + (aw_dev->i2c->addr == cfg_dde[i].dev_addr)) + (*scene_num)++; } - return ret; + if ((*scene_num) == 0) { + dev_err(aw_dev->dev, "failed to obtain scene, scenu_num = %d\n", (*scene_num)); + return -EINVAL; + } + + return 0; } static int aw_get_default_scene_count_v1(struct aw_device *aw_dev, @@ -834,37 +804,23 @@ static int aw_get_default_scene_count_v1(struct aw_device *aw_dev, struct aw_cfg_dde_v1 *cfg_dde = (struct aw_cfg_dde_v1 *)(aw_cfg->data + cfg_hdr->hdr_offset); unsigned int i; - int ret; - switch (aw_dev->chip_id) { - case AW88395_CHIP_ID: - case AW88399_CHIP_ID: - for (i = 0; i < cfg_hdr->ddt_num; ++i) { - if ((cfg_dde[i].data_type == ACF_SEC_TYPE_MULTIPLE_BIN) && - (aw_dev->chip_id == cfg_dde[i].chip_id) && - (aw_dev->channel == cfg_dde[i].dev_index)) - (*scene_num)++; - } - ret = 0; - break; - case AW88261_CHIP_ID: - case AW87390_CHIP_ID: - for (i = 0; i < cfg_hdr->ddt_num; ++i) { - if (((cfg_dde[i].data_type == ACF_SEC_TYPE_REG) || - (cfg_dde[i].data_type == ACF_SEC_TYPE_HDR_REG)) && - (aw_dev->chip_id == cfg_dde[i].chip_id) && - (aw_dev->channel == cfg_dde[i].dev_index)) - (*scene_num)++; - } - ret = 0; - break; - default: - dev_err(aw_dev->dev, "unsupported device"); - ret = -EINVAL; - break; + + for (i = 0; i < cfg_hdr->ddt_num; ++i) { + if (((cfg_dde[i].data_type == ACF_SEC_TYPE_MULTIPLE_BIN) || + (cfg_dde[i].data_type == ACF_SEC_TYPE_REG) || + (cfg_dde[i].data_type == ACF_SEC_TYPE_HDR_REG)) && + (aw_dev->chip_id == cfg_dde[i].chip_id) && + (aw_dev->channel == cfg_dde[i].dev_index)) + (*scene_num)++; } - return ret; + if ((*scene_num) == 0) { + dev_err(aw_dev->dev, "failed to obtain scene, scenu_num = %d\n", (*scene_num)); + return -EINVAL; + } + + return 0; } static int aw_dev_parse_scene_count_v1(struct aw_device *aw_dev, diff --git a/sound/soc/codecs/aw88395/aw88395_reg.h b/sound/soc/codecs/aw88395/aw88395_reg.h index ede7deab6a9c..e64f24e97150 100644 --- a/sound/soc/codecs/aw88395/aw88395_reg.h +++ b/sound/soc/codecs/aw88395/aw88395_reg.h @@ -95,10 +95,7 @@ #define AW88395_TM_REG (0x7C) enum aw88395_id { - AW88399_CHIP_ID = 0x2183, AW88395_CHIP_ID = 0x2049, - AW88261_CHIP_ID = 0x2113, - AW87390_CHIP_ID = 0x76, }; #define AW88395_REG_MAX (0x7D) diff --git a/sound/soc/codecs/aw88399.c b/sound/soc/codecs/aw88399.c index 54f8457e8497..9fcb805bf971 100644 --- a/sound/soc/codecs/aw88399.c +++ b/sound/soc/codecs/aw88399.c @@ -15,7 +15,6 @@ #include #include "aw88399.h" #include "aw88395/aw88395_device.h" -#include "aw88395/aw88395_reg.h" static const struct regmap_config aw88399_remap_config = { .val_bits = 16, diff --git a/sound/soc/codecs/aw88399.h b/sound/soc/codecs/aw88399.h index 4f391099d0f2..5e9cdf725d3d 100644 --- a/sound/soc/codecs/aw88399.h +++ b/sound/soc/codecs/aw88399.h @@ -491,6 +491,7 @@ #define AW88399_CRC_FW_BASE_ADDR (0x4C0) #define AW88399_ACF_FILE "aw88399_acf.bin" #define AW88399_DEV_SYSST_CHECK_MAX (10) +#define AW88399_CHIP_ID 0x2183 #define AW88399_I2C_NAME "aw88399" From c479f4989486b79cb92f0ee3b2ffcfc77fc3e443 Mon Sep 17 00:00:00 2001 From: Sebastian Reichel Date: Thu, 9 Nov 2023 19:44:43 +0100 Subject: [PATCH 035/337] dt-bindings: es8328: convert to DT schema format Convert the binding to DT schema format. Note, that "IPVDD-supply" got fixed to be "HPVDD-supply" during the conversion. This was obviously a typo in the old binding. The old binding example, DT files, chip datasheet and Linux driver use HPVDD. Signed-off-by: Sebastian Reichel Reviewed-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231109184453.108676-1-sebastian.reichel@collabora.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/es8328.txt | 38 --------- .../bindings/sound/everest,es8328.yaml | 77 +++++++++++++++++++ 2 files changed, 77 insertions(+), 38 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es8328.yaml diff --git a/Documentation/devicetree/bindings/sound/es8328.txt b/Documentation/devicetree/bindings/sound/es8328.txt deleted file mode 100644 index 33fbf058c997..000000000000 --- a/Documentation/devicetree/bindings/sound/es8328.txt +++ /dev/null @@ -1,38 +0,0 @@ -Everest ES8328 audio CODEC - -This device supports both I2C and SPI. - -Required properties: - - - compatible : Should be "everest,es8328" or "everest,es8388" - - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V - - AVDD-supply : Regulator providing analog supply voltage 3.3V - - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V - - IPVDD-supply : Regulator providing analog output voltage 3.3V - - clocks : A 22.5792 or 11.2896 MHz clock - - reg : the I2C address of the device for I2C, the chip select number for SPI - -Pins on the device (for linking into audio routes): - - * LOUT1 - * LOUT2 - * ROUT1 - * ROUT2 - * LINPUT1 - * RINPUT1 - * LINPUT2 - * RINPUT2 - * Mic Bias - - -Example: - -codec: es8328@11 { - compatible = "everest,es8328"; - DVDD-supply = <®_3p3v>; - AVDD-supply = <®_3p3v>; - PVDD-supply = <®_3p3v>; - HPVDD-supply = <®_3p3v>; - clocks = <&clks 169>; - reg = <0x11>; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8328.yaml b/Documentation/devicetree/bindings/sound/everest,es8328.yaml new file mode 100644 index 000000000000..a0f4670fa38c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8328.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8328.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8328 audio CODEC + +description: + Everest Audio Codec, which can be connected via I2C or SPI. + Pins on the device (for linking into audio routes) are + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * LINPUT1 + * RINPUT1 + * LINPUT2 + * RINPUT2 + * Mic Bias + +maintainers: + - David Yang + +properties: + compatible: + enum: + - everest,es8328 + - everest,es8388 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: A 22.5792 or 11.2896 MHz clock + + DVDD-supply: + description: Regulator providing digital core supply voltage 1.8 - 3.6V + + AVDD-supply: + description: Regulator providing analog supply voltage 3.3V + + PVDD-supply: + description: Regulator providing digital IO supply voltage 1.8 - 3.6V + + HPVDD-supply: + description: Regulator providing analog output voltage 3.3V + +required: + - compatible + - clocks + - DVDD-supply + - AVDD-supply + - PVDD-supply + - HPVDD-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8328: codec@11 { + compatible = "everest,es8328"; + reg = <0x11>; + AVDD-supply = <®_3p3v>; + DVDD-supply = <®_3p3v>; + HPVDD-supply = <®_3p3v>; + PVDD-supply = <®_3p3v>; + clocks = <&clks 169>; + }; + }; From c239b79315167d3e9b4b1e537b00e1ff5b87a317 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 6 Nov 2023 19:04:22 +0100 Subject: [PATCH 036/337] ASoC: dt-bindings: qcom,sm8250: add SM8550 sound card Add sound card for SM8550, which as of now looks fully compatible with SM8450. Signed-off-by: Krzysztof Kozlowski Acked-by: Rob Herring Link: https://lore.kernel.org/r/20231106180422.170492-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index e082a4fe095d..ec641fa2cd4b 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -21,6 +21,10 @@ properties: - lenovo,yoga-c630-sndcard - qcom,db845c-sndcard - const: qcom,sdm845-sndcard + - items: + - enum: + - qcom,sm8550-sndcard + - const: qcom,sm8450-sndcard - enum: - qcom,apq8016-sbc-sndcard - qcom,msm8916-qdsp6-sndcard From dc29d3d253f1f3513a916f0b4271569223860c71 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 7 Nov 2023 11:16:10 +0100 Subject: [PATCH 037/337] ASoC: dt-bindings: use "soundwire" as controller's node name in examples Soundwire Devicetree bindings expect the Soundwire controller device node to be named just "soundwire". Correct examples, so the incorrect code will not be re-used. Reported-by: Neil Armstrong Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231107101610.13728-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/adi,max98363.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,max98363.yaml b/Documentation/devicetree/bindings/sound/adi,max98363.yaml index a844b63f3930..c388cda56011 100644 --- a/Documentation/devicetree/bindings/sound/adi,max98363.yaml +++ b/Documentation/devicetree/bindings/sound/adi,max98363.yaml @@ -39,7 +39,7 @@ unevaluatedProperties: false examples: - | - soundwire-controller@3250000 { + soundwire@3250000 { #address-cells = <2>; #size-cells = <0>; reg = <0x3250000 0x2000>; diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml index ba572a7f4f3c..8e462cdf0018 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml @@ -52,7 +52,7 @@ examples: - | #include - soundwire-controller@3250000 { + soundwire@3250000 { #address-cells = <2>; #size-cells = <0>; reg = <0x3250000 0x2000>; diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml index e6723c9e312a..d717017b0fdb 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -48,7 +48,7 @@ examples: - | #include - soundwire-controller { + soundwire { #address-cells = <2>; #size-cells = <0>; From 601cc04c9d73ed728b29eaddb29ae13b48b03ce7 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 30 Oct 2023 12:03:51 +0300 Subject: [PATCH 038/337] ASoC: amd: acp: remove unnecessary NULL check The list iterator in a list_for_each_entry() loop can never be NULL. Remove the check and pull the code in a tab. Signed-off-by: Dan Carpenter Link: https://lore.kernel.org/r/e376a712-e0c6-446f-9e0b-c444dd795cbb@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp63.c | 22 ++++++++++------------ 1 file changed, 10 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp/acp63.c b/sound/soc/amd/acp/acp63.c index b871a216a6af..4d342441a650 100644 --- a/sound/soc/amd/acp/acp63.c +++ b/sound/soc/amd/acp/acp63.c @@ -283,18 +283,16 @@ static int __maybe_unused acp63_pcm_resume(struct device *dev) spin_lock(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { - if (stream) { - substream = stream->substream; - if (substream && substream->runtime) { - buf_in_frames = (substream->runtime->buffer_size); - buf_size = frames_to_bytes(substream->runtime, buf_in_frames); - config_pte_for_stream(adata, stream); - config_acp_dma(adata, stream, buf_size); - if (stream->dai_id) - restore_acp_i2s_params(substream, adata, stream); - else - restore_acp_pdm_params(substream, adata); - } + substream = stream->substream; + if (substream && substream->runtime) { + buf_in_frames = (substream->runtime->buffer_size); + buf_size = frames_to_bytes(substream->runtime, buf_in_frames); + config_pte_for_stream(adata, stream); + config_acp_dma(adata, stream, buf_size); + if (stream->dai_id) + restore_acp_i2s_params(substream, adata, stream); + else + restore_acp_pdm_params(substream, adata); } } spin_unlock(&adata->acp_lock); From a1321811985bed1b110d224a6c7ce1b967ee7607 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 13 Nov 2023 15:14:29 +0000 Subject: [PATCH 039/337] ASoC: cs42l43: Add missing static from runtime PM ops Fixes: 2b59332ead54 ("ASoC: cs42l43: Use new-style PM runtime macros") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202311091824.5z6PROGZ-lkp@intel.com/ Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20231113151429.1554139-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index d62c9f26c632..5c98343ebf71 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2251,8 +2251,8 @@ static int cs42l43_codec_runtime_resume(struct device *dev) return 0; } -DEFINE_RUNTIME_DEV_PM_OPS(cs42l43_codec_pm_ops, NULL, - cs42l43_codec_runtime_resume, NULL); +static DEFINE_RUNTIME_DEV_PM_OPS(cs42l43_codec_pm_ops, NULL, + cs42l43_codec_runtime_resume, NULL); static const struct platform_device_id cs42l43_codec_id_table[] = { { "cs42l43-codec", }, From a55ea47bb8743ee044550c5dfdd3cb3147602e05 Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Mon, 13 Nov 2023 06:59:07 -0500 Subject: [PATCH 040/337] ASoC: Intel: sof_rt5682: add mtl_rt5650 support RT5650 is I2S codec integrated with HP and SPK. The HW board connects SoC I2S to RT5650 codec as below: I2S0: ALC5650 aif1 for Speaker I2S2: ALC5650 aif2 for Headphone Reviewed-by: Bard Liao Signed-off-by: Mac Chiang Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231113115907.18539-1-mac.chiang@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 12 ++++++++++++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 06ad15af46de..9723479f43da 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1147,6 +1147,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "mtl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(2) | + SOF_RT5682_SSP_AMP(0) | + SOF_RT5682_NUM_HDMIDEV(3) | + SOF_BT_OFFLOAD_SSP(1) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 301b8142d554..af4224bff718 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -40,6 +40,11 @@ static const struct snd_soc_acpi_codecs mtl_lt6911_hdmi = { .codecs = {"INTC10B0"} }; +static const struct snd_soc_acpi_codecs mtl_rt5650_amp = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { { .comp_ids = &mtl_rt5682_rt5682s_hp, @@ -77,6 +82,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "mtl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &mtl_rt5650_amp, + .sof_tplg_filename = "sof-mtl-rt5650.tplg", + }, /* place amp-only boards in the end of table */ { .id = "INTC10B0", From d3534684ada99ef8c0899eb28c62b4462483ee19 Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Mon, 13 Nov 2023 18:03:42 +0530 Subject: [PATCH 041/337] ASoC: amd: acp: add Kconfig options for acp7.0 based platform driver ACP7.0 based platform legacy drivers can be built by selecting necessary kernel config option. This patch enables build support of the same. Signed-off-by: Syed Saba Kareem Link: https://lore.kernel.org/r/20231113123345.2196504-1-Syed.SabaKareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 12 ++++++++++++ sound/soc/amd/acp/Makefile | 2 ++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 5fb322212938..c8ac0027f741 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -73,6 +73,18 @@ config SND_AMD_ASOC_ACP63 Say Y if you want to enable AUDIO on ACP6.3 If unsure select "N". +config SND_AMD_ASOC_ACP70 + tristate "AMD ACP ASOC Acp7.0 Support" + depends on X86 && PCI + depends on ACPI + select SND_SOC_AMD_ACP_PCM + select SND_SOC_AMD_ACP_I2S + select SND_SOC_AMD_ACP_PDM + help + This option enables Acp7.0 PDM support on AMD platform. + Say Y if you want to enable AUDIO on ACP7.0 + If unsure select "N". + config SND_SOC_AMD_MACH_COMMON tristate depends on X86 && PCI && I2C diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index dd85700f1c5f..ff5f7893b81e 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -15,6 +15,7 @@ snd-acp-pci-objs := acp-pci.o snd-acp-renoir-objs := acp-renoir.o snd-acp-rembrandt-objs := acp-rembrandt.o snd-acp63-objs := acp63.o +snd-acp70-objs := acp70.o #machine specific driver snd-acp-mach-objs := acp-mach-common.o @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SOC_AMD_ACP_PCI) += snd-acp-pci.o obj-$(CONFIG_SND_AMD_ASOC_RENOIR) += snd-acp-renoir.o obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o obj-$(CONFIG_SND_AMD_ASOC_ACP63) += snd-acp63.o +obj-$(CONFIG_SND_AMD_ASOC_ACP70) += snd-acp70.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o From 577d71544871b075a25a09e4c5aa31008850c0a8 Mon Sep 17 00:00:00 2001 From: Jack Yu Date: Mon, 13 Nov 2023 10:37:15 +0000 Subject: [PATCH 042/337] ASoC: rt5682s: Add LDO output selection for dacref Add LDO output selection for dacref. Signed-off-by: Jack Yu Link: https://lore.kernel.org/r/62cad4e51c044108bad872ab349e36f8@realtek.com Signed-off-by: Mark Brown --- include/sound/rt5682s.h | 8 ++++++++ sound/soc/codecs/rt5682s.c | 23 +++++++++++++++++++++++ sound/soc/codecs/rt5682s.h | 7 +++++++ 3 files changed, 38 insertions(+) diff --git a/include/sound/rt5682s.h b/include/sound/rt5682s.h index 66ca0c75b914..006e6003d11c 100644 --- a/include/sound/rt5682s.h +++ b/include/sound/rt5682s.h @@ -31,6 +31,13 @@ enum rt5682s_dai_clks { RT5682S_DAI_NUM_CLKS, }; +enum { + RT5682S_LDO_1_607V, + RT5682S_LDO_1_5V, + RT5682S_LDO_1_406V, + RT5682S_LDO_1_731V, +}; + struct rt5682s_platform_data { enum rt5682s_dmic1_data_pin dmic1_data_pin; enum rt5682s_dmic1_clk_pin dmic1_clk_pin; @@ -38,6 +45,7 @@ struct rt5682s_platform_data { unsigned int dmic_clk_rate; unsigned int dmic_delay; unsigned int amic_delay; + unsigned int ldo_dacref; bool dmic_clk_driving_high; const char *dai_clk_names[RT5682S_DAI_NUM_CLKS]; diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index c261c33c4be7..3322056bbb3b 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2971,6 +2971,8 @@ static int rt5682s_parse_dt(struct rt5682s_priv *rt5682s, struct device *dev) &rt5682s->pdata.dmic_delay); device_property_read_u32(dev, "realtek,amic-delay-ms", &rt5682s->pdata.amic_delay); + device_property_read_u32(dev, "realtek,ldo-sel", + &rt5682s->pdata.ldo_dacref); if (device_property_read_string_array(dev, "clock-output-names", rt5682s->pdata.dai_clk_names, @@ -3250,6 +3252,27 @@ static int rt5682s_i2c_probe(struct i2c_client *i2c) break; } + /* LDO output voltage control */ + switch (rt5682s->pdata.ldo_dacref) { + case RT5682S_LDO_1_607V: + break; + case RT5682S_LDO_1_5V: + regmap_update_bits(rt5682s->regmap, RT5682S_BIAS_CUR_CTRL_7, + RT5682S_LDO_DACREF_MASK, RT5682S_LDO_DACREF_1_5V); + break; + case RT5682S_LDO_1_406V: + regmap_update_bits(rt5682s->regmap, RT5682S_BIAS_CUR_CTRL_7, + RT5682S_LDO_DACREF_MASK, RT5682S_LDO_DACREF_1_406V); + break; + case RT5682S_LDO_1_731V: + regmap_update_bits(rt5682s->regmap, RT5682S_BIAS_CUR_CTRL_7, + RT5682S_LDO_DACREF_MASK, RT5682S_LDO_DACREF_1_731V); + break; + default: + dev_warn(&i2c->dev, "invalid LDO output setting.\n"); + break; + } + INIT_DELAYED_WORK(&rt5682s->jack_detect_work, rt5682s_jack_detect_handler); INIT_DELAYED_WORK(&rt5682s->jd_check_work, rt5682s_jd_check_handler); diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 1d79d432d0d8..67f42898de96 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1263,6 +1263,13 @@ #define RT5682S_JDH_NO_PLUG (0x1 << 4) #define RT5682S_JDH_PLUG (0x0 << 4) +/* Bias current control 7 (0x0110) */ +#define RT5682S_LDO_DACREF_MASK (0x3 << 4) +#define RT5682S_LDO_DACREF_1_607V (0x0 << 4) +#define RT5682S_LDO_DACREF_1_5V (0x1 << 4) +#define RT5682S_LDO_DACREF_1_406V (0x2 << 4) +#define RT5682S_LDO_DACREF_1_731V (0x3 << 4) + /* Charge Pump Internal Register1 (0x0125) */ #define RT5682S_CP_CLK_HP_MASK (0x3 << 4) #define RT5682S_CP_CLK_HP_100KHZ (0x0 << 4) From 459956b17dd5cba06b0f2b75772497e46a59468b Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Thu, 16 Nov 2023 11:03:57 +0530 Subject: [PATCH 043/337] ASoC: amd: acp: add missing SND_SOC_AMD_ACP_LEGACY_COMMON flag for ACP70 add missing dependent SND_SOC_AMD_ACP_LEGACY_COMMON flag for ACP70 platform. Signed-off-by: Syed Saba Kareem Link: https://lore.kernel.org/r/20231116053405.2574081-1-Syed.SabaKareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index c8ac0027f741..84c963241dc5 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -80,6 +80,7 @@ config SND_AMD_ASOC_ACP70 select SND_SOC_AMD_ACP_PCM select SND_SOC_AMD_ACP_I2S select SND_SOC_AMD_ACP_PDM + select SND_SOC_AMD_ACP_LEGACY_COMMON help This option enables Acp7.0 PDM support on AMD platform. Say Y if you want to enable AUDIO on ACP7.0 From 7d562ac331ddc4f798e6185a858bc98c22ee7d1b Mon Sep 17 00:00:00 2001 From: Lad Prabhakar Date: Wed, 15 Nov 2023 21:33:58 +0000 Subject: [PATCH 044/337] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/Five SoC The SSI block on the RZ/Five SoC is identical to one found on the RZ/G2UL SoC. "renesas,r9a07g043-ssi" compatible string will be used on the RZ/Five SoC so to make this clear and to keep this file consistent, update the comment to include RZ/Five SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback on RZ/Five SoC. Signed-off-by: Lad Prabhakar Reviewed-by: Geert Uytterhoeven Acked-by: Conor Dooley Link: https://lore.kernel.org/r/20231115213358.33400-1-prabhakar.mahadev-lad.rj@bp.renesas.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 3b5ae45eee4a..8b9695f5decc 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -16,7 +16,7 @@ properties: compatible: items: - enum: - - renesas,r9a07g043-ssi # RZ/G2UL + - renesas,r9a07g043-ssi # RZ/G2UL and RZ/Five - renesas,r9a07g044-ssi # RZ/G2{L,LC} - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi From 552206add94dd7977bad32c37eba16e23756a0f9 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 17 Nov 2023 14:13:41 +0000 Subject: [PATCH 045/337] ASoC: cs43130: Store device in private struct and use it more consistently Also remove one unnecessary debug print Signed-off-by: Maciej Strozek Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231117141344.64320-5-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 151 ++++++++++++++++++------------------- sound/soc/codecs/cs43130.h | 1 + 2 files changed, 75 insertions(+), 77 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index d8ec325b9cc9..cdf26f9a72a3 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -238,7 +238,7 @@ static int cs43130_pll_config(struct snd_soc_component *component) struct cs43130_private *cs43130 = snd_soc_component_get_drvdata(component); const struct cs43130_pll_params *pll_entry; - dev_dbg(component->dev, "cs43130->mclk = %u, cs43130->mclk_int = %u\n", + dev_dbg(cs43130->dev, "cs43130->mclk = %u, cs43130->mclk_int = %u\n", cs43130->mclk, cs43130->mclk_int); pll_entry = cs43130_get_pll_table(cs43130->mclk, cs43130->mclk_int); @@ -303,7 +303,7 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int cs43130->mclk = freq_in; break; default: - dev_err(component->dev, + dev_err(cs43130->dev, "unsupported pll input reference clock:%d\n", freq_in); return -EINVAL; } @@ -316,13 +316,13 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int cs43130->mclk_int = freq_out; break; default: - dev_err(component->dev, + dev_err(cs43130->dev, "unsupported pll output ref clock: %u\n", freq_out); return -EINVAL; } ret = cs43130_pll_config(component); - dev_dbg(component->dev, "cs43130->pll_bypass = %d", cs43130->pll_bypass); + dev_dbg(cs43130->dev, "cs43130->pll_bypass = %d", cs43130->pll_bypass); return ret; } @@ -346,7 +346,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, mclk_int_decoded = CS43130_MCLK_24P5; break; default: - dev_err(component->dev, "Invalid MCLK INT freq: %u\n", cs43130->mclk_int); + dev_err(cs43130->dev, "Invalid MCLK INT freq: %u\n", cs43130->mclk_int); return -EINVAL; } @@ -370,7 +370,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); if (ret == 0) { - dev_err(component->dev, "Timeout waiting for XTAL_READY interrupt\n"); + dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); return -ETIMEDOUT; } } @@ -406,7 +406,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); if (ret == 0) { - dev_err(component->dev, "Timeout waiting for XTAL_READY interrupt\n"); + dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); return -ETIMEDOUT; } } @@ -422,7 +422,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_PLL_RDY_INT_MASK, 1 << CS43130_PLL_RDY_INT_SHIFT); if (ret == 0) { - dev_err(component->dev, "Timeout waiting for PLL_READY interrupt\n"); + dev_err(cs43130->dev, "Timeout waiting for PLL_READY interrupt\n"); return -ETIMEDOUT; } @@ -453,7 +453,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, 1 << CS43130_PDN_PLL_SHIFT); break; default: - dev_err(component->dev, "Invalid MCLK source value\n"); + dev_err(cs43130->dev, "Invalid MCLK source value\n"); return -EINVAL; } @@ -804,7 +804,7 @@ static int cs43130_dsd_hw_params(struct snd_pcm_substream *substream, dsd_speed = 1; break; default: - dev_err(component->dev, "Rate(%u) not supported\n", + dev_err(cs43130->dev, "Rate(%u) not supported\n", params_rate(params)); return -EINVAL; } @@ -875,7 +875,7 @@ static int cs43130_hw_params(struct snd_pcm_substream *substream, dsd_speed = 1; break; default: - dev_err(component->dev, "Rate(%u) not supported\n", + dev_err(cs43130->dev, "Rate(%u) not supported\n", params_rate(params)); return -EINVAL; } @@ -892,7 +892,7 @@ static int cs43130_hw_params(struct snd_pcm_substream *substream, regmap_write(cs43130->regmap, CS43130_SP_SRATE, rate_map->val); break; default: - dev_err(component->dev, "Invalid DAI (%d)\n", dai->id); + dev_err(cs43130->dev, "Invalid DAI (%d)\n", dai->id); return -EINVAL; } @@ -916,21 +916,21 @@ static int cs43130_hw_params(struct snd_pcm_substream *substream, if (!sclk) { /* at this point, SCLK must be set */ - dev_err(component->dev, "SCLK freq is not set\n"); + dev_err(cs43130->dev, "SCLK freq is not set\n"); return -EINVAL; } bitwidth_sclk = (sclk / params_rate(params)) / params_channels(params); if (bitwidth_sclk < bitwidth_dai) { - dev_err(component->dev, "Format not supported: SCLK freq is too low\n"); + dev_err(cs43130->dev, "Format not supported: SCLK freq is too low\n"); return -EINVAL; } - dev_dbg(component->dev, + dev_dbg(cs43130->dev, "sclk = %u, fs = %d, bitwidth_dai = %u\n", sclk, params_rate(params), bitwidth_dai); - dev_dbg(component->dev, + dev_dbg(cs43130->dev, "bitwidth_sclk = %u, num_ch = %u\n", bitwidth_sclk, params_channels(params)); @@ -1189,7 +1189,7 @@ static int cs43130_dsd_event(struct snd_soc_dapm_widget *w, } break; default: - dev_err(component->dev, "Invalid event = 0x%x\n", event); + dev_err(cs43130->dev, "Invalid event = 0x%x\n", event); return -EINVAL; } return 0; @@ -1246,7 +1246,7 @@ static int cs43130_pcm_event(struct snd_soc_dapm_widget *w, } break; default: - dev_err(component->dev, "Invalid event = 0x%x\n", event); + dev_err(cs43130->dev, "Invalid event = 0x%x\n", event); return -EINVAL; } return 0; @@ -1322,7 +1322,7 @@ static int cs43130_dac_event(struct snd_soc_dapm_widget *w, } break; default: - dev_err(component->dev, "Invalid DAC event = 0x%x\n", event); + dev_err(cs43130->dev, "Invalid DAC event = 0x%x\n", event); return -EINVAL; } return 0; @@ -1360,7 +1360,7 @@ static int cs43130_hpin_event(struct snd_soc_dapm_widget *w, ARRAY_SIZE(hpin_postpmu_seq)); break; default: - dev_err(component->dev, "Invalid HPIN event = 0x%x\n", event); + dev_err(cs43130->dev, "Invalid HPIN event = 0x%x\n", event); return -EINVAL; } return 0; @@ -1479,7 +1479,7 @@ static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM; break; default: - dev_err(component->dev, "unsupported mode\n"); + dev_err(cs43130->dev, "unsupported mode\n"); return -EINVAL; } @@ -1497,12 +1497,12 @@ static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_DSP_B; break; default: - dev_err(component->dev, + dev_err(cs43130->dev, "unsupported audio format\n"); return -EINVAL; } - dev_dbg(component->dev, "dai_id = %d, dai_mode = %u, dai_format = %u\n", + dev_dbg(cs43130->dev, "dai_id = %d, dai_mode = %u, dai_format = %u\n", codec_dai->id, cs43130->dais[codec_dai->id].dai_mode, cs43130->dais[codec_dai->id].dai_format); @@ -1523,11 +1523,11 @@ static int cs43130_dsd_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) cs43130->dais[codec_dai->id].dai_mode = SND_SOC_DAIFMT_CBM_CFM; break; default: - dev_err(component->dev, "Unsupported DAI format.\n"); + dev_err(cs43130->dev, "Unsupported DAI format.\n"); return -EINVAL; } - dev_dbg(component->dev, "dai_mode = 0x%x\n", + dev_dbg(cs43130->dev, "dai_mode = 0x%x\n", cs43130->dais[codec_dai->id].dai_mode); return 0; @@ -1540,7 +1540,7 @@ static int cs43130_set_sysclk(struct snd_soc_dai *codec_dai, struct cs43130_private *cs43130 = snd_soc_component_get_drvdata(component); cs43130->dais[codec_dai->id].sclk = freq; - dev_dbg(component->dev, "dai_id = %d, sclk = %u\n", codec_dai->id, + dev_dbg(cs43130->dev, "dai_id = %d, sclk = %u\n", codec_dai->id, cs43130->dais[codec_dai->id].sclk); return 0; @@ -1630,7 +1630,7 @@ static int cs43130_component_set_sysclk(struct snd_soc_component *component, { struct cs43130_private *cs43130 = snd_soc_component_get_drvdata(component); - dev_dbg(component->dev, "clk_id = %d, source = %d, freq = %d, dir = %d\n", + dev_dbg(cs43130->dev, "clk_id = %d, source = %d, freq = %d, dir = %d\n", clk_id, source, freq, dir); switch (freq) { @@ -1639,14 +1639,14 @@ static int cs43130_component_set_sysclk(struct snd_soc_component *component, cs43130->mclk = freq; break; default: - dev_err(component->dev, "Invalid MCLK INT freq: %u\n", freq); + dev_err(cs43130->dev, "Invalid MCLK INT freq: %u\n", freq); return -EINVAL; } if (source == CS43130_MCLK_SRC_EXT) { cs43130->pll_bypass = true; } else { - dev_err(component->dev, "Invalid MCLK source\n"); + dev_err(cs43130->dev, "Invalid MCLK source\n"); return -EINVAL; } @@ -1933,7 +1933,6 @@ static int cs43130_update_hpload(unsigned int msk, int ac_idx, unsigned int reg; u32 addr; u16 impedance; - struct snd_soc_component *component = cs43130->component; switch (msk) { case CS43130_HPLOAD_DC_INT: @@ -1963,7 +1962,7 @@ static int cs43130_update_hpload(unsigned int msk, int ac_idx, else cs43130->hpload_dc[HP_RIGHT] = impedance; - dev_dbg(component->dev, "HP DC impedance (Ch %u): %u\n", !left_ch, + dev_dbg(cs43130->dev, "HP DC impedance (Ch %u): %u\n", !left_ch, impedance); } else { if (left_ch) @@ -1971,7 +1970,7 @@ static int cs43130_update_hpload(unsigned int msk, int ac_idx, else cs43130->hpload_ac[ac_idx][HP_RIGHT] = impedance; - dev_dbg(component->dev, "HP AC (%u Hz) impedance (Ch %u): %u\n", + dev_dbg(cs43130->dev, "HP AC (%u Hz) impedance (Ch %u): %u\n", cs43130->ac_freq[ac_idx], !left_ch, impedance); } @@ -1985,7 +1984,6 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, int ret; unsigned int msk; u16 ac_reg_val; - struct snd_soc_component *component = cs43130->component; reinit_completion(&cs43130->hpload_evt); @@ -2008,17 +2006,17 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, msecs_to_jiffies(1000)); regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); if (!ret) { - dev_err(component->dev, "Timeout waiting for HPLOAD interrupt\n"); + dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt\n"); return -1; } - dev_dbg(component->dev, "HP load stat: %x, INT_MASK_4: %x\n", + dev_dbg(cs43130->dev, "HP load stat: %x, INT_MASK_4: %x\n", cs43130->hpload_stat, msk); if ((cs43130->hpload_stat & (CS43130_HPLOAD_NO_DC_INT | CS43130_HPLOAD_UNPLUG_INT | CS43130_HPLOAD_OOR_INT)) || !(cs43130->hpload_stat & rslt_msk)) { - dev_dbg(component->dev, "HP load measure failed\n"); + dev_dbg(cs43130->dev, "HP load measure failed\n"); return -1; } @@ -2129,9 +2127,9 @@ static void cs43130_imp_meas(struct work_struct *wk) snd_soc_jack_report(&cs43130->jack, CS43130_JACK_HEADPHONE, CS43130_JACK_MASK); - dev_dbg(component->dev, "Set HP output control. DC threshold\n"); + dev_dbg(cs43130->dev, "Set HP output control. DC threshold\n"); for (i = 0; i < CS43130_DC_THRESHOLD; i++) - dev_dbg(component->dev, "DC threshold[%d]: %u.\n", i, + dev_dbg(cs43130->dev, "DC threshold[%d]: %u.\n", i, cs43130->dc_threshold[i]); cs43130_set_hv(cs43130->regmap, cs43130->hpload_dc[HP_LEFT], @@ -2165,7 +2163,6 @@ exit: static irqreturn_t cs43130_irq_thread(int irq, void *data) { struct cs43130_private *cs43130 = (struct cs43130_private *)data; - struct snd_soc_component *component = cs43130->component; unsigned int stickies[CS43130_NUM_INT]; unsigned int irq_occurrence = 0; unsigned int masks[CS43130_NUM_INT]; @@ -2183,8 +2180,6 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) for (j = 0; j < 8; j++) irq_occurrence += (stickies[i] >> j) & 1; } - dev_dbg(component->dev, "number of interrupts occurred (%u)\n", - irq_occurrence); if (!irq_occurrence) return IRQ_NONE; @@ -2201,7 +2196,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[3] & CS43130_HPLOAD_NO_DC_INT) { cs43130->hpload_stat = stickies[3]; - dev_err(component->dev, + dev_err(cs43130->dev, "DC load has not completed before AC load (%x)\n", cs43130->hpload_stat); complete(&cs43130->hpload_evt); @@ -2210,7 +2205,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[3] & CS43130_HPLOAD_UNPLUG_INT) { cs43130->hpload_stat = stickies[3]; - dev_err(component->dev, "HP unplugged during measurement (%x)\n", + dev_err(cs43130->dev, "HP unplugged during measurement (%x)\n", cs43130->hpload_stat); complete(&cs43130->hpload_evt); return IRQ_HANDLED; @@ -2218,7 +2213,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[3] & CS43130_HPLOAD_OOR_INT) { cs43130->hpload_stat = stickies[3]; - dev_err(component->dev, "HP load out of range (%x)\n", + dev_err(cs43130->dev, "HP load out of range (%x)\n", cs43130->hpload_stat); complete(&cs43130->hpload_evt); return IRQ_HANDLED; @@ -2226,7 +2221,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[3] & CS43130_HPLOAD_AC_INT) { cs43130->hpload_stat = stickies[3]; - dev_dbg(component->dev, "HP AC load measurement done (%x)\n", + dev_dbg(cs43130->dev, "HP AC load measurement done (%x)\n", cs43130->hpload_stat); complete(&cs43130->hpload_evt); return IRQ_HANDLED; @@ -2234,7 +2229,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[3] & CS43130_HPLOAD_DC_INT) { cs43130->hpload_stat = stickies[3]; - dev_dbg(component->dev, "HP DC load measurement done (%x)\n", + dev_dbg(cs43130->dev, "HP DC load measurement done (%x)\n", cs43130->hpload_stat); complete(&cs43130->hpload_evt); return IRQ_HANDLED; @@ -2242,7 +2237,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[3] & CS43130_HPLOAD_ON_INT) { cs43130->hpload_stat = stickies[3]; - dev_dbg(component->dev, "HP load state machine on done (%x)\n", + dev_dbg(cs43130->dev, "HP load state machine on done (%x)\n", cs43130->hpload_stat); complete(&cs43130->hpload_evt); return IRQ_HANDLED; @@ -2250,19 +2245,19 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[3] & CS43130_HPLOAD_OFF_INT) { cs43130->hpload_stat = stickies[3]; - dev_dbg(component->dev, "HP load state machine off done (%x)\n", + dev_dbg(cs43130->dev, "HP load state machine off done (%x)\n", cs43130->hpload_stat); complete(&cs43130->hpload_evt); return IRQ_HANDLED; } if (stickies[0] & CS43130_XTAL_ERR_INT) { - dev_err(component->dev, "Crystal err: clock is not running\n"); + dev_err(cs43130->dev, "Crystal err: clock is not running\n"); return IRQ_HANDLED; } if (stickies[0] & CS43130_HP_UNPLUG_INT) { - dev_dbg(component->dev, "HP unplugged\n"); + dev_dbg(cs43130->dev, "HP unplugged\n"); cs43130->hpload_done = false; snd_soc_jack_report(&cs43130->jack, 0, CS43130_JACK_MASK); return IRQ_HANDLED; @@ -2271,7 +2266,7 @@ static irqreturn_t cs43130_irq_thread(int irq, void *data) if (stickies[0] & CS43130_HP_PLUG_INT) { if (cs43130->dc_meas && !cs43130->hpload_done && !work_busy(&cs43130->work)) { - dev_dbg(component->dev, "HP load queue work\n"); + dev_dbg(cs43130->dev, "HP load queue work\n"); queue_work(cs43130->wq, &cs43130->work); } @@ -2303,19 +2298,19 @@ static int cs43130_probe(struct snd_soc_component *component) ret = snd_soc_card_jack_new(card, "Headphone", CS43130_JACK_MASK, &cs43130->jack); if (ret < 0) { - dev_err(component->dev, "Cannot create jack\n"); + dev_err(cs43130->dev, "Cannot create jack\n"); return ret; } cs43130->hpload_done = false; if (cs43130->dc_meas) { - ret = sysfs_create_groups(&component->dev->kobj, hpload_groups); + ret = sysfs_create_groups(&cs43130->dev->kobj, hpload_groups); if (ret) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); if (!cs43130->wq) { - sysfs_remove_groups(&component->dev->kobj, hpload_groups); + sysfs_remove_groups(&cs43130->dev->kobj, hpload_groups); return -ENOMEM; } INIT_WORK(&cs43130->work, cs43130_imp_meas); @@ -2391,7 +2386,7 @@ static int cs43130_handle_device_data(struct i2c_client *i2c_client, cs43130->xtal_ibias = CS43130_XTAL_IBIAS_15UA; break; default: - dev_err(&i2c_client->dev, + dev_err(cs43130->dev, "Invalid cirrus,xtal-ibias value: %d\n", val); return -EINVAL; } @@ -2426,6 +2421,8 @@ static int cs43130_i2c_probe(struct i2c_client *client) if (!cs43130) return -ENOMEM; + cs43130->dev = &client->dev; + i2c_set_clientdata(client, cs43130); cs43130->regmap = devm_regmap_init_i2c(client, &cs43130_regmap); @@ -2434,7 +2431,7 @@ static int cs43130_i2c_probe(struct i2c_client *client) return ret; } - if (client->dev.of_node) { + if (cs43130->dev->of_node) { ret = cs43130_handle_device_data(client, cs43130); if (ret != 0) return ret; @@ -2442,21 +2439,21 @@ static int cs43130_i2c_probe(struct i2c_client *client) for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++) cs43130->supplies[i].supply = cs43130_supply_names[i]; - ret = devm_regulator_bulk_get(&client->dev, + ret = devm_regulator_bulk_get(cs43130->dev, ARRAY_SIZE(cs43130->supplies), cs43130->supplies); if (ret != 0) { - dev_err(&client->dev, "Failed to request supplies: %d\n", ret); + dev_err(cs43130->dev, "Failed to request supplies: %d\n", ret); return ret; } ret = regulator_bulk_enable(ARRAY_SIZE(cs43130->supplies), cs43130->supplies); if (ret != 0) { - dev_err(&client->dev, "Failed to enable supplies: %d\n", ret); + dev_err(cs43130->dev, "Failed to enable supplies: %d\n", ret); return ret; } - cs43130->reset_gpio = devm_gpiod_get_optional(&client->dev, + cs43130->reset_gpio = devm_gpiod_get_optional(cs43130->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(cs43130->reset_gpio)) { ret = PTR_ERR(cs43130->reset_gpio); @@ -2470,7 +2467,7 @@ static int cs43130_i2c_probe(struct i2c_client *client) devid = cirrus_read_device_id(cs43130->regmap, CS43130_DEVID_AB); if (devid < 0) { ret = devid; - dev_err(&client->dev, "Failed to read device ID: %d\n", ret); + dev_err(cs43130->dev, "Failed to read device ID: %d\n", ret); goto err; } @@ -2481,7 +2478,7 @@ static int cs43130_i2c_probe(struct i2c_client *client) case CS43198_CHIP_ID: break; default: - dev_err(&client->dev, + dev_err(cs43130->dev, "CS43130 Device ID %X. Expected ID %X, %X, %X or %X\n", devid, CS43130_CHIP_ID, CS4399_CHIP_ID, CS43131_CHIP_ID, CS43198_CHIP_ID); @@ -2492,11 +2489,11 @@ static int cs43130_i2c_probe(struct i2c_client *client) cs43130->dev_id = devid; ret = regmap_read(cs43130->regmap, CS43130_REV_ID, ®); if (ret < 0) { - dev_err(&client->dev, "Get Revision ID failed\n"); + dev_err(cs43130->dev, "Get Revision ID failed\n"); goto err; } - dev_info(&client->dev, + dev_info(cs43130->dev, "Cirrus Logic CS43130 (%x), Revision: %02X\n", devid, reg & 0xFF); @@ -2506,21 +2503,21 @@ static int cs43130_i2c_probe(struct i2c_client *client) init_completion(&cs43130->pll_rdy); init_completion(&cs43130->hpload_evt); - ret = devm_request_threaded_irq(&client->dev, client->irq, + ret = devm_request_threaded_irq(cs43130->dev, client->irq, NULL, cs43130_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs43130", cs43130); if (ret != 0) { - dev_err(&client->dev, "Failed to request IRQ: %d\n", ret); + dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); goto err; } cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; - pm_runtime_set_autosuspend_delay(&client->dev, 100); - pm_runtime_use_autosuspend(&client->dev); - pm_runtime_set_active(&client->dev); - pm_runtime_enable(&client->dev); + pm_runtime_set_autosuspend_delay(cs43130->dev, 100); + pm_runtime_use_autosuspend(cs43130->dev); + pm_runtime_set_active(cs43130->dev); + pm_runtime_enable(cs43130->dev); switch (cs43130->dev_id) { case CS43130_CHIP_ID: @@ -2556,11 +2553,11 @@ static int cs43130_i2c_probe(struct i2c_client *client) break; } - ret = devm_snd_soc_register_component(&client->dev, + ret = devm_snd_soc_register_component(cs43130->dev, &soc_component_dev_cs43130, cs43130_dai, ARRAY_SIZE(cs43130_dai)); if (ret < 0) { - dev_err(&client->dev, + dev_err(cs43130->dev, "snd_soc_register_component failed with ret = %d\n", ret); goto err; } @@ -2598,15 +2595,15 @@ static void cs43130_i2c_remove(struct i2c_client *client) cancel_work_sync(&cs43130->work); flush_workqueue(cs43130->wq); - device_remove_file(&client->dev, &dev_attr_hpload_dc_l); - device_remove_file(&client->dev, &dev_attr_hpload_dc_r); - device_remove_file(&client->dev, &dev_attr_hpload_ac_l); - device_remove_file(&client->dev, &dev_attr_hpload_ac_r); + device_remove_file(cs43130->dev, &dev_attr_hpload_dc_l); + device_remove_file(cs43130->dev, &dev_attr_hpload_dc_r); + device_remove_file(cs43130->dev, &dev_attr_hpload_ac_l); + device_remove_file(cs43130->dev, &dev_attr_hpload_ac_r); } gpiod_set_value_cansleep(cs43130->reset_gpio, 0); - pm_runtime_disable(&client->dev); + pm_runtime_disable(cs43130->dev); regulator_bulk_disable(CS43130_NUM_SUPPLIES, cs43130->supplies); } diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index 90e8895275e7..2f5ec3888103 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -500,6 +500,7 @@ struct cs43130_dai { }; struct cs43130_private { + struct device *dev; struct snd_soc_component *component; struct regmap *regmap; struct regulator_bulk_data supplies[CS43130_NUM_SUPPLIES]; From ce7944b73e7729dc702b6741cff2b26886bb723c Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 17 Nov 2023 14:13:42 +0000 Subject: [PATCH 046/337] ASoC: cs43130: Add handling of ACPI Signed-off-by: Maciej Strozek Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231117141344.64320-6-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 44 ++++++++++++++++++++++++-------------- 1 file changed, 28 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index cdf26f9a72a3..a51d77947964 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -16,7 +16,7 @@ #include #include #include -#include +#include #include #include #include @@ -2362,14 +2362,12 @@ static const u16 cs43130_dc_threshold[CS43130_DC_THRESHOLD] = { 120, }; -static int cs43130_handle_device_data(struct i2c_client *i2c_client, - struct cs43130_private *cs43130) +static int cs43130_handle_device_data(struct cs43130_private *cs43130) { - struct device_node *np = i2c_client->dev.of_node; unsigned int val; int i; - if (of_property_read_u32(np, "cirrus,xtal-ibias", &val) < 0) { + if (device_property_read_u32(cs43130->dev, "cirrus,xtal-ibias", &val) < 0) { /* Crystal is unused. System clock is used for external MCLK */ cs43130->xtal_ibias = CS43130_XTAL_UNUSED; return 0; @@ -2391,18 +2389,18 @@ static int cs43130_handle_device_data(struct i2c_client *i2c_client, return -EINVAL; } - cs43130->dc_meas = of_property_read_bool(np, "cirrus,dc-measure"); - cs43130->ac_meas = of_property_read_bool(np, "cirrus,ac-measure"); + cs43130->dc_meas = device_property_read_bool(cs43130->dev, "cirrus,dc-measure"); + cs43130->ac_meas = device_property_read_bool(cs43130->dev, "cirrus,ac-measure"); - if (of_property_read_u16_array(np, "cirrus,ac-freq", cs43130->ac_freq, - CS43130_AC_FREQ) < 0) { + if (!device_property_read_u16_array(cs43130->dev, "cirrus,ac-freq", cs43130->ac_freq, + CS43130_AC_FREQ)) { for (i = 0; i < CS43130_AC_FREQ; i++) cs43130->ac_freq[i] = cs43130_ac_freq[i]; } - if (of_property_read_u16_array(np, "cirrus,dc-threshold", + if (!device_property_read_u16_array(cs43130->dev, "cirrus,dc-threshold", cs43130->dc_threshold, - CS43130_DC_THRESHOLD) < 0) { + CS43130_DC_THRESHOLD)) { for (i = 0; i < CS43130_DC_THRESHOLD; i++) cs43130->dc_threshold[i] = cs43130_dc_threshold[i]; } @@ -2431,11 +2429,12 @@ static int cs43130_i2c_probe(struct i2c_client *client) return ret; } - if (cs43130->dev->of_node) { - ret = cs43130_handle_device_data(client, cs43130); + if (dev_fwnode(cs43130->dev)) { + ret = cs43130_handle_device_data(cs43130); if (ret != 0) return ret; } + for (i = 0; i < ARRAY_SIZE(cs43130->supplies); i++) cs43130->supplies[i].supply = cs43130_supply_names[i]; @@ -2666,6 +2665,7 @@ static const struct dev_pm_ops cs43130_runtime_pm = { NULL) }; +#if IS_ENABLED(CONFIG_OF) static const struct of_device_id cs43130_of_match[] = { {.compatible = "cirrus,cs43130",}, {.compatible = "cirrus,cs4399",}, @@ -2675,6 +2675,17 @@ static const struct of_device_id cs43130_of_match[] = { }; MODULE_DEVICE_TABLE(of, cs43130_of_match); +#endif + +#if IS_ENABLED(CONFIG_ACPI) +static const struct acpi_device_id cs43130_acpi_match[] = { + { "CSC4399", 0 }, + {} +}; + +MODULE_DEVICE_TABLE(acpi, cs43130_acpi_match); +#endif + static const struct i2c_device_id cs43130_i2c_id[] = { {"cs43130", 0}, @@ -2688,9 +2699,10 @@ MODULE_DEVICE_TABLE(i2c, cs43130_i2c_id); static struct i2c_driver cs43130_i2c_driver = { .driver = { - .name = "cs43130", - .of_match_table = cs43130_of_match, - .pm = &cs43130_runtime_pm, + .name = "cs43130", + .of_match_table = of_match_ptr(cs43130_of_match), + .acpi_match_table = ACPI_PTR(cs43130_acpi_match), + .pm = &cs43130_runtime_pm, }, .id_table = cs43130_i2c_id, .probe = cs43130_i2c_probe, From 9158221bf2aa5f7bfb916452c079b2fe63ca76e8 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 17 Nov 2023 14:13:44 +0000 Subject: [PATCH 047/337] ASoC: cs43130: Add switch to control normal and alt hp inputs Make sure these inputs are mutually exclusive as recommended by the datasheet Signed-off-by: Maciej Strozek Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231117141344.64320-8-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index a51d77947964..845611afed85 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1366,7 +1366,15 @@ static int cs43130_hpin_event(struct snd_soc_dapm_widget *w, return 0; } +static const char * const bypass_mux_text[] = { + "Internal", + "Alternative", +}; +static SOC_ENUM_SINGLE_DECL(bypass_enum, SND_SOC_NOPM, 0, bypass_mux_text); +static const struct snd_kcontrol_new bypass_ctrl = SOC_DAPM_ENUM("Switch", bypass_enum); + static const struct snd_soc_dapm_widget digital_hp_widgets[] = { + SND_SOC_DAPM_MUX("Bypass Switch", SND_SOC_NOPM, 0, 0, &bypass_ctrl), SND_SOC_DAPM_OUTPUT("HPOUTA"), SND_SOC_DAPM_OUTPUT("HPOUTB"), @@ -1419,13 +1427,13 @@ static const struct snd_soc_dapm_route digital_hp_routes[] = { {"DSD", NULL, "XSPIN DSD"}, {"HiFi DAC", NULL, "ASPIN PCM"}, {"HiFi DAC", NULL, "DSD"}, - {"HPOUTA", NULL, "HiFi DAC"}, - {"HPOUTB", NULL, "HiFi DAC"}, + {"Bypass Switch", "Internal", "HiFi DAC"}, + {"HPOUTA", NULL, "Bypass Switch"}, + {"HPOUTB", NULL, "Bypass Switch"}, }; static const struct snd_soc_dapm_route analog_hp_routes[] = { - {"HPOUTA", NULL, "Analog Playback"}, - {"HPOUTB", NULL, "Analog Playback"}, + {"Bypass Switch", "Alternative", "Analog Playback"}, }; static struct snd_soc_dapm_route all_hp_routes[ From 5a77457232fa0453da4d3d5a7d530129944aeeb5 Mon Sep 17 00:00:00 2001 From: Philipp Stanner Date: Thu, 2 Nov 2023 20:03:10 +0100 Subject: [PATCH 048/337] ALSA: wavefront: copy userspace array safely wavefront_fx.c utilizes memdup_user() to copy a userspace array. This does not check for an overflow. Use the new wrapper memdup_array_user() to copy the array more safely. Suggested-by: Dave Airlie Signed-off-by: Philipp Stanner Link: https://lore.kernel.org/r/20231102190309.50891-2-pstanner@redhat.com Signed-off-by: Takashi Iwai --- sound/isa/wavefront/wavefront_fx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index 3c21324b2a0e..0273b7dfaf12 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -191,9 +191,9 @@ snd_wavefront_fx_ioctl (struct snd_hwdep *sdev, struct file *file, "> 512 bytes to FX\n"); return -EIO; } - page_data = memdup_user((unsigned char __user *) - r.data[3], - r.data[2] * sizeof(short)); + page_data = memdup_array_user((unsigned char __user *) + r.data[3], + r.data[2], sizeof(short)); if (IS_ERR(page_data)) return PTR_ERR(page_data); pd = page_data; From cac15dc25f416972f8dd0b6a8d74daa79dc3a998 Mon Sep 17 00:00:00 2001 From: Lucas Tanure Date: Sun, 19 Nov 2023 10:45:14 +0000 Subject: [PATCH 049/337] ASoC: fsl_mqs: Remove duplicate linux/of.h header Remove linux/of.h as is included more than once. Reported by make includecheck. Signed-off-by: Lucas Tanure Acked-by: Shengjiu Wang Link: https://lore.kernel.org/r/20231119104514.25536-1-tanure@linux.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_mqs.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index f2d74ec05cdf..86704ba5f6f0 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -10,7 +10,6 @@ #include #include #include -#include #include #include #include From af524e9dcb43f5914cecb3a3f4b79081d2e3f7f8 Mon Sep 17 00:00:00 2001 From: David Lin Date: Mon, 20 Nov 2023 16:42:28 +0800 Subject: [PATCH 050/337] ASoC: nau8810: Fix incorrect type in assignment and cast to restricted __be16 This issue is reproduced when W=1 build in compiler gcc-12. The following are sparse warnings: sound/soc/codecs/nau8810.c:183:25: sparse: warning: incorrect type in assignment sound/soc/codecs/nau8810.c:183:25: sparse: expected int sound/soc/codecs/nau8810.c:183:25: sparse: got restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 This issue is not still actively checked by kernel test robot. Actually, it is same with nau8822's sparse warnings issue. Signed-off-by: David Lin Link: https://lore.kernel.org/r/20231120084227.1766633-1-CTLIN0@nuvoton.com Signed-off-by: Mark Brown --- sound/soc/codecs/nau8810.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 47f000cd4d99..97a54059474c 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -169,6 +169,7 @@ static int nau8810_eq_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *params = (void *)kcontrol->private_value; int i, reg, reg_val; u16 *val; + __be16 tmp; val = (u16 *)ucontrol->value.bytes.data; reg = NAU8810_REG_EQ1; @@ -177,8 +178,8 @@ static int nau8810_eq_get(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - reg_val = cpu_to_be16(reg_val); - memcpy(val + i, ®_val, sizeof(reg_val)); + tmp = cpu_to_be16(reg_val); + memcpy(val + i, &tmp, sizeof(tmp)); } return 0; @@ -201,6 +202,7 @@ static int nau8810_eq_put(struct snd_kcontrol *kcontrol, void *data; u16 *val, value; int i, reg, ret; + __be16 *tmp; data = kmemdup(ucontrol->value.bytes.data, params->max, GFP_KERNEL | GFP_DMA); @@ -213,7 +215,8 @@ static int nau8810_eq_put(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - value = be16_to_cpu(*(val + i)); + tmp = (__be16 *)(val + i); + value = be16_to_cpup(tmp); ret = regmap_write(nau8810->regmap, reg + i, value); if (ret) { dev_err(component->dev, "EQ configuration fail, register: %x ret: %d\n", From 27c69d7da1084af0b8b3a20ef9ff01e9eda5270c Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Tue, 21 Nov 2023 13:25:11 +0800 Subject: [PATCH 051/337] ASoC: dt-bindings: sound-card-common: List sound widgets ignoring system suspend Add a property to list audio sound widgets which are marked ignoring system suspend. Paths between these endpoints are still active over suspend of the main application processor that the current operating system is running. Signed-off-by: Chancel Liu Link: https://lore.kernel.org/r/20231121052512.20235-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sound-card-common.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sound-card-common.yaml b/Documentation/devicetree/bindings/sound/sound-card-common.yaml index 3a941177f684..721950f65748 100644 --- a/Documentation/devicetree/bindings/sound/sound-card-common.yaml +++ b/Documentation/devicetree/bindings/sound/sound-card-common.yaml @@ -17,6 +17,13 @@ properties: pair of strings, the first being the connection's sink, the second being the connection's source. + ignore-suspend-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of audio sound widgets which are marked ignoring system suspend. + Paths between these endpoints are still active over suspend of the main + application processor that the current operating system is running. + model: $ref: /schemas/types.yaml#/definitions/string description: User specified audio sound card name From 5d9f746ca64c3ebfba3b650dbc4b0de705c83f3b Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Tue, 21 Nov 2023 13:25:12 +0800 Subject: [PATCH 052/337] ASoC: imx-rpmsg: Force codec power on in low power audio mode Low power audio mode requires binding codec still power on while Acore enters into suspend so Mcore can continue playback music. ASoC machine driver acquires DAPM endpoints through reading "ignore-suspend-widgets" property from DT and then forces the path between these endpoints ignoring suspend. If the rpmsg sound card is in low power audio mode, the suspend/resume callback of binding codec is overridden to disable the suspend/resume. Signed-off-by: Chancel Liu Link: https://lore.kernel.org/r/20231121052512.20235-2-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-rpmsg.c | 61 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 59 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index a0c5c35817dd..e5bd63dab10c 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -2,9 +2,8 @@ // Copyright 2017-2020 NXP #include -#include +#include #include -#include #include #include #include @@ -21,8 +20,11 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; unsigned long sysclk; + bool lpa; }; +static struct dev_pm_ops lpa_pm; + static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), @@ -39,6 +41,58 @@ static int imx_rpmsg_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (data->lpa) { + struct snd_soc_component *codec_comp; + struct device_node *codec_np; + struct device_driver *codec_drv; + struct device *codec_dev = NULL; + + codec_np = data->dai.codecs->of_node; + if (codec_np) { + struct platform_device *codec_pdev; + struct i2c_client *codec_i2c; + + codec_i2c = of_find_i2c_device_by_node(codec_np); + if (codec_i2c) + codec_dev = &codec_i2c->dev; + if (!codec_dev) { + codec_pdev = of_find_device_by_node(codec_np); + if (codec_pdev) + codec_dev = &codec_pdev->dev; + } + } + if (codec_dev) { + codec_comp = snd_soc_lookup_component_nolocked(codec_dev, NULL); + if (codec_comp) { + int i, num_widgets; + const char *widgets; + struct snd_soc_dapm_context *dapm; + + num_widgets = of_property_count_strings(data->card.dev->of_node, + "ignore-suspend-widgets"); + for (i = 0; i < num_widgets; i++) { + of_property_read_string_index(data->card.dev->of_node, + "ignore-suspend-widgets", + i, &widgets); + dapm = snd_soc_component_get_dapm(codec_comp); + snd_soc_dapm_ignore_suspend(dapm, widgets); + } + } + codec_drv = codec_dev->driver; + if (codec_drv->pm) { + memcpy(&lpa_pm, codec_drv->pm, sizeof(lpa_pm)); + lpa_pm.suspend = NULL; + lpa_pm.resume = NULL; + lpa_pm.freeze = NULL; + lpa_pm.thaw = NULL; + lpa_pm.poweroff = NULL; + lpa_pm.restore = NULL; + codec_drv->pm = &lpa_pm; + } + put_device(codec_dev); + } + } + if (!data->sysclk) return 0; @@ -138,6 +192,9 @@ static int imx_rpmsg_probe(struct platform_device *pdev) goto fail; } + if (of_property_read_bool(np, "fsl,enable-lpa")) + data->lpa = true; + data->card.dev = &pdev->dev; data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; From 26257869672fd4a06a60c2da841e15fb2cb47bbe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Nov 2023 16:41:25 +0100 Subject: [PATCH 053/337] ALSA: hda: Refer to correct stream index at loops In a couple of loops over the all streams, we check the bitmap against the loop counter. A more correct reference would be, however, the index of each stream, instead. This patch corrects the check of bitmaps to the stream index. Note that this change doesn't fix anything for now; all existing drivers set up the stream indices properly, hence the loop count is always equal with the stream index. That said, this change is only for consistency. Link: https://lore.kernel.org/r/20231121154125.4888-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/hdac_stream.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index 6ce24e248f8e..610ea7a33cd8 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -671,17 +671,15 @@ void snd_hdac_stream_timecounter_init(struct hdac_stream *azx_dev, struct hdac_stream *s; bool inited = false; u64 cycle_last = 0; - int i = 0; list_for_each_entry(s, &bus->stream_list, list) { - if (streams & (1 << i)) { + if ((streams & (1 << s->index))) { azx_timecounter_init(s, inited, cycle_last); if (!inited) { inited = true; cycle_last = s->tc.cycle_last; } } - i++; } snd_pcm_gettime(runtime, &runtime->trigger_tstamp); @@ -726,14 +724,13 @@ void snd_hdac_stream_sync(struct hdac_stream *azx_dev, bool start, unsigned int streams) { struct hdac_bus *bus = azx_dev->bus; - int i, nwait, timeout; + int nwait, timeout; struct hdac_stream *s; for (timeout = 5000; timeout; timeout--) { nwait = 0; - i = 0; list_for_each_entry(s, &bus->stream_list, list) { - if (!(streams & (1 << i++))) + if (!(streams & (1 << s->index))) continue; if (start) { From 67c7666fe808c3a7af3cc6f9d0a3dd3acfd26115 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Tue, 21 Nov 2023 14:07:51 +0200 Subject: [PATCH 054/337] ASoC: doc: Fix undefined SND_SOC_DAPM_NOPM argument The virtual widget example makes use of an undefined SND_SOC_DAPM_NOPM argument passed to SND_SOC_DAPM_MIXER(). Replace with the correct SND_SOC_NOPM definition. Signed-off-by: Cristian Ciocaltea Link: https://lore.kernel.org/r/20231121120751.77355-1-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- Documentation/sound/soc/dapm.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/soc/dapm.rst b/Documentation/sound/soc/dapm.rst index 8e44107933ab..c3154ce6e1b2 100644 --- a/Documentation/sound/soc/dapm.rst +++ b/Documentation/sound/soc/dapm.rst @@ -234,7 +234,7 @@ corresponding soft power control. In this case it is necessary to create a virtual widget - a widget with no control bits e.g. :: - SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), This can be used to merge to signal paths together in software. From 9996cd782a602f2542e110e2a4035dd6627bd520 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Wed, 22 Nov 2023 18:19:59 +0800 Subject: [PATCH 055/337] ASoC: dt-bindings: fsl,mqs: Convert format to json-schema Convert NXP medium quality sound (MQS) device tree binding documentation to json-schema. Signed-off-by: Chancel Liu Reviewed-by: Rob Herring Link: https://lore.kernel.org/r/20231122101959.30264-4-chancel.liu@nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,mqs.txt | 36 ------ .../devicetree/bindings/sound/fsl,mqs.yaml | 105 ++++++++++++++++++ 2 files changed, 105 insertions(+), 36 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt deleted file mode 100644 index d66284b8bef2..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,mqs.txt +++ /dev/null @@ -1,36 +0,0 @@ -fsl,mqs audio CODEC - -Required properties: - - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" - "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs", "fsl,imx93-mqs". - - clocks : A list of phandles + clock-specifiers, one for each entry in - clock-names - - clock-names : "mclk" - must required. - "core" - required if compatible is "fsl,imx8qm-mqs", it - is for register access. - - gpr : A phandle of General Purpose Registers in IOMUX Controller. - Required if compatible is "fsl,imx6sx-mqs". - -Required if compatible is "fsl,imx8qm-mqs": - - power-domains: A phandle of PM domain provider node. - - reg: Offset and length of the register set for the device. - -Example: - -mqs: mqs { - compatible = "fsl,imx6sx-mqs"; - gpr = <&gpr>; - clocks = <&clks IMX6SX_CLK_SAI1>; - clock-names = "mclk"; - status = "disabled"; -}; - -mqs: mqs@59850000 { - compatible = "fsl,imx8qm-mqs"; - reg = <0x59850000 0x10000>; - clocks = <&clk IMX8QM_AUD_MQS_IPG>, - <&clk IMX8QM_AUD_MQS_HMCLK>; - clock-names = "core", "mclk"; - power-domains = <&pd_mqs0>; - status = "disabled"; -}; diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.yaml b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml new file mode 100644 index 000000000000..8b33353a80ca --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml @@ -0,0 +1,105 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,mqs.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Medium Quality Sound (MQS) + +maintainers: + - Shengjiu Wang + - Chancel Liu + +description: | + Medium quality sound (MQS) is used to generate medium quality audio + via a standard GPIO in the pinmux, allowing the user to connect + stereo speakers or headphones to a power amplifier without an + additional DAC chip. + +properties: + compatible: + enum: + - fsl,imx6sx-mqs + - fsl,imx8qm-mqs + - fsl,imx8qxp-mqs + - fsl,imx93-mqs + + clocks: + minItems: 1 + maxItems: 2 + + clock-names: + minItems: 1 + maxItems: 2 + + gpr: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle to the General Purpose Register (GPR) node + + reg: + maxItems: 1 + + power-domains: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - compatible + - clocks + - clock-names + +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx8qm-mqs + - fsl,imx8qxp-mqs + then: + properties: + clocks: + items: + - description: Master clock + - description: Clock for register access + clock-names: + items: + - const: mclk + - const: core + required: + - reg + - power-domains + else: + properties: + clocks: + items: + - description: Master clock + clock-names: + items: + - const: mclk + required: + - gpr + +additionalProperties: false + +examples: + - | + #include + mqs0: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + }; + + - | + #include + mqs1: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&mqs0_lpcg 0>, <&mqs0_lpcg 1>; + clock-names = "mclk", "core"; + power-domains = <&pd IMX_SC_R_MQS_0>; + }; From b1cea462a79316bd619173f1ded8b28202b5ce3a Mon Sep 17 00:00:00 2001 From: Michael Ellerman Date: Wed, 22 Nov 2023 17:27:11 +1100 Subject: [PATCH 056/337] ASoC: fsl: mpc8610_hpcd: Remove unused driver The mpc8610_hpcd.c driver depends on CONFIG_MPC8610_HPCD which was removed in commit 248667f8bbde ("powerpc: drop HPCD/MPC8610 evaluation platform support"). That makes the driver unbuildable and unusable, so remove it. Depends-on: 248667f8bbde ("powerpc: drop HPCD/MPC8610 evaluation platform support") Signed-off-by: Michael Ellerman Link: https://lore.kernel.org/r/20231122062712.2250426-1-mpe@ellerman.id.au Signed-off-by: Mark Brown --- MAINTAINERS | 1 - sound/soc/fsl/Kconfig | 13 - sound/soc/fsl/Makefile | 4 - sound/soc/fsl/mpc8610_hpcd.c | 451 ----------------------------------- 4 files changed, 469 deletions(-) delete mode 100644 sound/soc/fsl/mpc8610_hpcd.c diff --git a/MAINTAINERS b/MAINTAINERS index 97f51d5ec1cf..3943ad3ca346 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8581,7 +8581,6 @@ L: linuxppc-dev@lists.ozlabs.org S: Maintained F: sound/soc/fsl/fsl* F: sound/soc/fsl/imx* -F: sound/soc/fsl/mpc8610_hpcd.c FREESCALE SOC SOUND QMC DRIVER M: Herve Codina diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 725c530a3636..bec2aa561930 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -183,19 +183,6 @@ config SND_SOC_POWERPC_QMC_AUDIO comment "SoC Audio support for Freescale PPC boards:" -config SND_SOC_MPC8610_HPCD - tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" - # I2C is necessary for the CS4270 driver - depends on MPC8610_HPCD && I2C - select SND_SOC_FSL_SSI - select SND_SOC_FSL_UTILS - select SND_SOC_POWERPC_DMA - select SND_SOC_CS4270 - select SND_SOC_CS4270_VD33_ERRATA - default y if MPC8610_HPCD - help - Say Y if you want to enable audio on the Freescale MPC8610 HPCD. - config SND_SOC_P1022_DS tristate "ALSA SoC support for the Freescale P1022 DS board" # I2C is necessary for the WM8776 driver diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8db7e97d0bd5..b45eda80c196 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -1,8 +1,4 @@ # SPDX-License-Identifier: GPL-2.0 -# MPC8610 HPCD Machine Support -snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o -obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o - # P1022 DS Machine Support snd-soc-p1022-ds-objs := p1022_ds.o obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c deleted file mode 100644 index 52fb9e7bcca4..000000000000 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ /dev/null @@ -1,451 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// -// Freescale MPC8610HPCD ALSA SoC Machine driver -// -// Author: Timur Tabi -// -// Copyright 2007-2010 Freescale Semiconductor, Inc. - -#include -#include -#include -#include -#include -#include -#include - -#include "fsl_dma.h" -#include "fsl_ssi.h" -#include "fsl_utils.h" - -/* There's only one global utilities register */ -static phys_addr_t guts_phys; - -/** - * mpc8610_hpcd_data: machine-specific ASoC device data - * - * This structure contains data for a single sound platform device on an - * MPC8610 HPCD. Some of the data is taken from the device tree. - */ -struct mpc8610_hpcd_data { - struct snd_soc_dai_link dai[2]; - struct snd_soc_card card; - unsigned int dai_format; - unsigned int codec_clk_direction; - unsigned int cpu_clk_direction; - unsigned int clk_frequency; - unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ - unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ - unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ -}; - -/** - * mpc8610_hpcd_machine_probe: initialize the board - * - * This function is used to initialize the board-specific hardware. - * - * Here we program the DMACR and PMUXCR registers. - */ -static int mpc8610_hpcd_machine_probe(struct snd_soc_card *card) -{ - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Program the signal routing between the SSI and the DMA */ - guts_set_dmacr(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], - CCSR_GUTS_DMACR_DEV_SSI); - guts_set_dmacr(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], - CCSR_GUTS_DMACR_DEV_SSI); - - guts_set_pmuxcr_dma(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], 0); - guts_set_pmuxcr_dma(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_SSI); - break; - case 1: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_SSI); - break; - } - - iounmap(guts); - - return 0; -} - -/** - * mpc8610_hpcd_startup: program the board with various hardware parameters - * - * This function takes board-specific information, like clock frequencies - * and serial data formats, and passes that information to the codec and - * transport drivers. - */ -static int mpc8610_hpcd_startup(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct mpc8610_hpcd_data *machine_data = - container_of(rtd->card, struct mpc8610_hpcd_data, card); - struct device *dev = rtd->card->dev; - int ret = 0; - - /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(snd_soc_rtd_to_codec(rtd, 0), machine_data->dai_format); - if (ret < 0) { - dev_err(dev, "could not set codec driver audio format\n"); - return ret; - } - - /* - * Tell the codec driver what the MCLK frequency is, and whether it's - * a slave or master. - */ - ret = snd_soc_dai_set_sysclk(snd_soc_rtd_to_codec(rtd, 0), 0, - machine_data->clk_frequency, - machine_data->codec_clk_direction); - if (ret < 0) { - dev_err(dev, "could not set codec driver clock params\n"); - return ret; - } - - return 0; -} - -/** - * mpc8610_hpcd_machine_remove: Remove the sound device - * - * This function is called to remove the sound device for one SSI. We - * de-program the DMACR and PMUXCR register. - */ -static int mpc8610_hpcd_machine_remove(struct snd_soc_card *card) -{ - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - struct ccsr_guts __iomem *guts; - - guts = ioremap(guts_phys, sizeof(struct ccsr_guts)); - if (!guts) { - dev_err(card->dev, "could not map global utilities\n"); - return -ENOMEM; - } - - /* Restore the signal routing */ - - guts_set_dmacr(guts, machine_data->dma_id[0], - machine_data->dma_channel_id[0], 0); - guts_set_dmacr(guts, machine_data->dma_id[1], - machine_data->dma_channel_id[1], 0); - - switch (machine_data->ssi_id) { - case 0: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI1_MASK, CCSR_GUTS_PMUXCR_SSI1_LA); - break; - case 1: - clrsetbits_be32(&guts->pmuxcr, - CCSR_GUTS_PMUXCR_SSI2_MASK, CCSR_GUTS_PMUXCR_SSI2_LA); - break; - } - - iounmap(guts); - - return 0; -} - -/** - * mpc8610_hpcd_ops: ASoC machine driver operations - */ -static const struct snd_soc_ops mpc8610_hpcd_ops = { - .startup = mpc8610_hpcd_startup, -}; - -/** - * mpc8610_hpcd_probe: platform probe function for the machine driver - * - * Although this is a machine driver, the SSI node is the "master" node with - * respect to audio hardware connections. Therefore, we create a new ASoC - * device for each new SSI node that has a codec attached. - */ -static int mpc8610_hpcd_probe(struct platform_device *pdev) -{ - struct device *dev = pdev->dev.parent; - /* ssi_pdev is the platform device for the SSI node that probed us */ - struct platform_device *ssi_pdev = to_platform_device(dev); - struct device_node *np = ssi_pdev->dev.of_node; - struct device_node *codec_np = NULL; - struct mpc8610_hpcd_data *machine_data; - struct snd_soc_dai_link_component *comp; - int ret; - const char *sprop; - const u32 *iprop; - - /* Find the codec node for this SSI. */ - codec_np = of_parse_phandle(np, "codec-handle", 0); - if (!codec_np) { - dev_err(dev, "invalid codec node\n"); - return -EINVAL; - } - - machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) { - ret = -ENOMEM; - goto error_alloc; - } - - comp = devm_kzalloc(&pdev->dev, 6 * sizeof(*comp), GFP_KERNEL); - if (!comp) { - ret = -ENOMEM; - goto error_alloc; - } - - machine_data->dai[0].cpus = &comp[0]; - machine_data->dai[0].codecs = &comp[1]; - machine_data->dai[0].platforms = &comp[2]; - - machine_data->dai[0].num_cpus = 1; - machine_data->dai[0].num_codecs = 1; - machine_data->dai[0].num_platforms = 1; - - machine_data->dai[1].cpus = &comp[3]; - machine_data->dai[1].codecs = &comp[4]; - machine_data->dai[1].platforms = &comp[5]; - - machine_data->dai[1].num_cpus = 1; - machine_data->dai[1].num_codecs = 1; - machine_data->dai[1].num_platforms = 1; - - machine_data->dai[0].cpus->dai_name = dev_name(&ssi_pdev->dev); - machine_data->dai[0].ops = &mpc8610_hpcd_ops; - - /* ASoC core can match codec with device node */ - machine_data->dai[0].codecs->of_node = codec_np; - - /* The DAI name from the codec (snd_soc_dai_driver.name) */ - machine_data->dai[0].codecs->dai_name = "cs4270-hifi"; - - /* We register two DAIs per SSI, one for playback and the other for - * capture. Currently, we only support codecs that have one DAI for - * both playback and capture. - */ - memcpy(&machine_data->dai[1], &machine_data->dai[0], - sizeof(struct snd_soc_dai_link)); - - /* Get the device ID */ - iprop = of_get_property(np, "cell-index", NULL); - if (!iprop) { - dev_err(&pdev->dev, "cell-index property not found\n"); - ret = -EINVAL; - goto error; - } - machine_data->ssi_id = be32_to_cpup(iprop); - - /* Get the serial format and clock direction. */ - sprop = of_get_property(np, "fsl,mode", NULL); - if (!sprop) { - dev_err(&pdev->dev, "fsl,mode property not found\n"); - ret = -EINVAL; - goto error; - } - - if (strcasecmp(sprop, "i2s-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - - /* In i2s-slave mode, the codec has its own clock source, so we - * need to get the frequency from the device tree and pass it to - * the codec driver. - */ - iprop = of_get_property(codec_np, "clock-frequency", NULL); - if (!iprop || !*iprop) { - dev_err(&pdev->dev, "codec bus-frequency " - "property is missing or invalid\n"); - ret = -EINVAL; - goto error; - } - machine_data->clk_frequency = be32_to_cpup(iprop); - } else if (strcasecmp(sprop, "i2s-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "lj-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "lj-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "rj-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "ac97-slave") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBP_CFP; - machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; - } else if (strcasecmp(sprop, "ac97-master") == 0) { - machine_data->dai_format = - SND_SOC_DAIFMT_AC97 | SND_SOC_DAIFMT_CBC_CFC; - machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; - machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else { - dev_err(&pdev->dev, - "unrecognized fsl,mode property '%s'\n", sprop); - ret = -EINVAL; - goto error; - } - - if (!machine_data->clk_frequency) { - dev_err(&pdev->dev, "unknown clock frequency\n"); - ret = -EINVAL; - goto error; - } - - /* Find the playback DMA channel to use. */ - machine_data->dai[0].platforms->name = machine_data->platform_name[0]; - ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", - &machine_data->dai[0], - &machine_data->dma_channel_id[0], - &machine_data->dma_id[0]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); - goto error; - } - - /* Find the capture DMA channel to use. */ - machine_data->dai[1].platforms->name = machine_data->platform_name[1]; - ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", - &machine_data->dai[1], - &machine_data->dma_channel_id[1], - &machine_data->dma_id[1]); - if (ret) { - dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); - goto error; - } - - /* Initialize our DAI data structure. */ - machine_data->dai[0].stream_name = "playback"; - machine_data->dai[1].stream_name = "capture"; - machine_data->dai[0].name = machine_data->dai[0].stream_name; - machine_data->dai[1].name = machine_data->dai[1].stream_name; - - machine_data->card.probe = mpc8610_hpcd_machine_probe; - machine_data->card.remove = mpc8610_hpcd_machine_remove; - machine_data->card.name = pdev->name; /* The platform driver name */ - machine_data->card.owner = THIS_MODULE; - machine_data->card.dev = &pdev->dev; - machine_data->card.num_links = 2; - machine_data->card.dai_link = machine_data->dai; - - /* Register with ASoC */ - ret = snd_soc_register_card(&machine_data->card); - if (ret) { - dev_err(&pdev->dev, "could not register card\n"); - goto error; - } - - of_node_put(codec_np); - - return 0; - -error: - kfree(machine_data); -error_alloc: - of_node_put(codec_np); - return ret; -} - -/** - * mpc8610_hpcd_remove: remove the platform device - * - * This function is called when the platform device is removed. - */ -static void mpc8610_hpcd_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct mpc8610_hpcd_data *machine_data = - container_of(card, struct mpc8610_hpcd_data, card); - - snd_soc_unregister_card(card); - kfree(machine_data); -} - -static struct platform_driver mpc8610_hpcd_driver = { - .probe = mpc8610_hpcd_probe, - .remove_new = mpc8610_hpcd_remove, - .driver = { - /* The name must match 'compatible' property in the device tree, - * in lowercase letters. - */ - .name = "snd-soc-mpc8610hpcd", - }, -}; - -/** - * mpc8610_hpcd_init: machine driver initialization. - * - * This function is called when this module is loaded. - */ -static int __init mpc8610_hpcd_init(void) -{ - struct device_node *guts_np; - struct resource res; - - pr_info("Freescale MPC8610 HPCD ALSA SoC machine driver\n"); - - /* Get the physical address of the global utilities registers */ - guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts"); - if (of_address_to_resource(guts_np, 0, &res)) { - pr_err("mpc8610-hpcd: missing/invalid global utilities node\n"); - of_node_put(guts_np); - return -EINVAL; - } - guts_phys = res.start; - of_node_put(guts_np); - - return platform_driver_register(&mpc8610_hpcd_driver); -} - -/** - * mpc8610_hpcd_exit: machine driver exit - * - * This function is called when this driver is unloaded. - */ -static void __exit mpc8610_hpcd_exit(void) -{ - platform_driver_unregister(&mpc8610_hpcd_driver); -} - -module_init(mpc8610_hpcd_init); -module_exit(mpc8610_hpcd_exit); - -MODULE_AUTHOR("Timur Tabi "); -MODULE_DESCRIPTION("Freescale MPC8610 HPCD ALSA SoC machine driver"); -MODULE_LICENSE("GPL v2"); From fa91703dc2e010e48a230dc92967cb5ae23f8680 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Thu, 23 Nov 2023 09:06:58 +0000 Subject: [PATCH 057/337] ASoC: cs43130: Allow driver to work without IRQ connection Add a polling mechanism that will keep the driver operational even in absence of physical IRQ connection. If IRQ line is detected, the driver will continue working as usual, in case of missing IRQ line it will fallback to the polling mechanism introduced in this change. This will support users which choose not to connect an IRQ line as it is not critical to part's operation. Signed-off-by: Maciej Strozek Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231123090658.10418-1-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 77 ++++++++++++++++++++++++++------------ sound/soc/codecs/cs43130.h | 1 + 2 files changed, 55 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 845611afed85..4f16baf4eafb 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -326,6 +326,34 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int return ret; } +static int cs43130_wait_for_completion(struct cs43130_private *cs43130, struct completion *to_poll, + int time) +{ + int stickies, offset, flag, ret; + + if (cs43130->has_irq_line) { + ret = wait_for_completion_timeout(to_poll, msecs_to_jiffies(time)); + if (ret == 0) + return -ETIMEDOUT; + else + return 0; // Discard number of jiffies left till timeout and return success + } + + if (to_poll == &cs43130->xtal_rdy) { + offset = 0; + flag = CS43130_XTAL_RDY_INT; + } else if (to_poll == &cs43130->pll_rdy) { + offset = 0; + flag = CS43130_PLL_RDY_INT; + } else { + return -EINVAL; + } + + return regmap_read_poll_timeout(cs43130->regmap, CS43130_INT_STATUS_1 + offset, + stickies, (stickies & flag), + 1000, time * 1000); +} + static int cs43130_change_clksrc(struct snd_soc_component *component, enum cs43130_mclk_src_sel src) { @@ -364,14 +392,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -400,14 +427,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for XTAL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for XTAL_READY interrupt: %d\n", ret); + return ret; } } @@ -416,14 +442,13 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_PLL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_PLL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->pll_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->pll_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_PLL_RDY_INT_MASK, 1 << CS43130_PLL_RDY_INT_SHIFT); - if (ret == 0) { - dev_err(cs43130->dev, "Timeout waiting for PLL_READY interrupt\n"); - return -ETIMEDOUT; + if (ret) { + dev_err(cs43130->dev, "Error waiting for PLL_READY interrupt: %d\n", ret); + return ret; } regmap_update_bits(cs43130->regmap, CS43130_SYS_CLK_CTL_1, @@ -2015,7 +2040,7 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); if (!ret) { dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt\n"); - return -1; + return -ETIMEDOUT; } dev_dbg(cs43130->dev, "HP load stat: %x, INT_MASK_4: %x\n", @@ -2510,13 +2535,19 @@ static int cs43130_i2c_probe(struct i2c_client *client) init_completion(&cs43130->pll_rdy); init_completion(&cs43130->hpload_evt); - ret = devm_request_threaded_irq(cs43130->dev, client->irq, - NULL, cs43130_irq_thread, - IRQF_ONESHOT | IRQF_TRIGGER_LOW, - "cs43130", cs43130); - if (ret != 0) { - dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); - goto err; + if (!client->irq) { + dev_dbg(cs43130->dev, "IRQ not found, will poll instead\n"); + cs43130->has_irq_line = 0; + } else { + ret = devm_request_threaded_irq(cs43130->dev, client->irq, + NULL, cs43130_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs43130", cs43130); + if (ret != 0) { + dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); + goto err; + } + cs43130->has_irq_line = 1; } cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index 2f5ec3888103..694286b78d03 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -507,6 +507,7 @@ struct cs43130_private { struct gpio_desc *reset_gpio; unsigned int dev_id; /* codec device ID */ int xtal_ibias; + bool has_irq_line; /* shared by both DAIs */ struct mutex clk_mutex; From 52be2c4926831f7858c25701950afe9c1879f71f Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Fri, 24 Nov 2023 09:50:30 +0000 Subject: [PATCH 058/337] ASoC: cs43130: Allow configuration of bit clock and frame inversion Signed-off-by: Maciej Strozek Link: https://lore.kernel.org/r/20231124095030.24539-1-mstrozek@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 42 ++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/cs43130.h | 1 + 2 files changed, 41 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 4f16baf4eafb..0f3ead84665f 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -620,6 +620,27 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, return -EINVAL; } + switch (cs43130->dais[dai_id].dai_invert) { + case SND_SOC_DAIFMT_NB_NF: + sclk_edge = 1; + lrck_edge = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + sclk_edge = 0; + lrck_edge = 0; + break; + case SND_SOC_DAIFMT_NB_IF: + sclk_edge = 1; + lrck_edge = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + sclk_edge = 0; + lrck_edge = 1; + break; + default: + return -EINVAL; + } + switch (cs43130->dais[dai_id].dai_mode) { case SND_SOC_DAIFMT_CBS_CFS: dai_mode_val = 0; @@ -632,8 +653,6 @@ static int cs43130_set_sp_fmt(int dai_id, unsigned int bitwidth_sclk, } frm_size = bitwidth_sclk * params_channels(params); - sclk_edge = 1; - lrck_edge = 0; loc_ch1 = 0; loc_ch2 = bitwidth_sclk * (params_channels(params) - 1); @@ -1516,6 +1535,25 @@ static int cs43130_pcm_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_NB_NF; + break; + case SND_SOC_DAIFMT_IB_NF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_IB_NF; + break; + case SND_SOC_DAIFMT_NB_IF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_NB_IF; + break; + case SND_SOC_DAIFMT_IB_IF: + cs43130->dais[codec_dai->id].dai_invert = SND_SOC_DAIFMT_IB_IF; + break; + default: + dev_err(cs43130->dev, "Unsupported invert mode 0x%x\n", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: cs43130->dais[codec_dai->id].dai_format = SND_SOC_DAIFMT_I2S; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index 694286b78d03..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -497,6 +497,7 @@ struct cs43130_dai { unsigned int sclk; unsigned int dai_format; unsigned int dai_mode; + unsigned int dai_invert; }; struct cs43130_private { From 29b0b68f25ae6f9454c3e1c31b054595af0a80fc Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 24 Nov 2023 09:38:03 +0100 Subject: [PATCH 059/337] ASoC: dt-bindings: correct white-spaces in examples Use only one and exactly one space around '=' in DTS example. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231124083803.12773-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml | 4 ++-- .../devicetree/bindings/sound/qcom,wcd938x-sdw.yaml | 4 ++-- Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml | 4 ++-- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 8 ++++---- .../devicetree/bindings/sound/ti,tlv320aic32x4.yaml | 2 +- 6 files changed, 12 insertions(+), 12 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index 3380b6aa9542..054b53954ac3 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -135,7 +135,7 @@ examples: nuvoton,jack-insert-debounce = <7>; nuvoton,jack-eject-debounce = <0>; nuvoton,dmic-clk-threshold = <3072000>; - nuvoton,dmic-slew-rate= <0>; + nuvoton,dmic-slew-rate = <0>; #sound-dai-cells = <0>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml index 4df59f3b7b01..beb0ff0245b0 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd934x.yaml @@ -201,9 +201,9 @@ examples: - | codec@1,0{ compatible = "slim217,250"; - reg = <1 0>; + reg = <1 0>; reset-gpios = <&tlmm 64 0>; - slim-ifc-dev = <&wcd9340_ifd>; + slim-ifc-dev = <&wcd9340_ifd>; #sound-dai-cells = <1>; interrupt-parent = <&tlmm>; interrupts = <54 4>; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml index b430dd3e1841..7b31bf93f1a1 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml @@ -51,7 +51,7 @@ examples: reg = <0x03210000 0x2000>; wcd938x_rx: codec@0,4 { compatible = "sdw20217010d00"; - reg = <0 4>; + reg = <0 4>; qcom,rx-port-mapping = <1 2 3 4 5>; }; }; @@ -62,7 +62,7 @@ examples: reg = <0x03230000 0x2000>; wcd938x_tx: codec@0,3 { compatible = "sdw20217010d00"; - reg = <0 3>; + reg = <0 3>; qcom,tx-port-mapping = <2 3 4 5>; }; }; diff --git a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml index 018565793a3e..adbfa67f88ed 100644 --- a/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml @@ -137,7 +137,7 @@ examples: reg = <0x03210000 0x2000>; wcd938x_rx: codec@0,4 { compatible = "sdw20217010d00"; - reg = <0 4>; + reg = <0 4>; qcom,rx-port-mapping = <1 2 3 4 5>; }; }; @@ -148,7 +148,7 @@ examples: reg = <0x03230000 0x2000>; wcd938x_tx: codec@0,3 { compatible = "sdw20217010d00"; - reg = <0 3>; + reg = <0 3>; qcom,tx-port-mapping = <2 3 4 5>; }; }; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 1174205286d4..0d7a6b576d88 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -497,19 +497,19 @@ examples: rcar_sound,dai { dai0 { playback = <&ssi5>, <&src5>; - capture = <&ssi6>; + capture = <&ssi6>; }; dai1 { playback = <&ssi3>; }; dai2 { - capture = <&ssi4>; + capture = <&ssi4>; }; dai3 { playback = <&ssi7>; }; dai4 { - capture = <&ssi8>; + capture = <&ssi8>; }; }; @@ -523,7 +523,7 @@ examples: frame-master = <&rsnd_endpoint0>; playback = <&ssi0>, <&src0>, <&dvc0>; - capture = <&ssi1>, <&src1>, <&dvc1>; + capture = <&ssi1>, <&src1>, <&dvc1>; }; }; }; diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml index a7cc9aa34468..4783e6dbb5c4 100644 --- a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml @@ -90,7 +90,7 @@ examples: ldoin-supply = <®_3v3>; clocks = <&clks 201>; clock-names = "mclk"; - aic32x4-gpio-func= < + aic32x4-gpio-func = < 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ From 5980bda0a998a6ee6afd83b97a482a40c1c68076 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 24 Nov 2023 17:08:50 +0200 Subject: [PATCH 060/337] ASoC: SOF: ipc4-topology: Helper to find an swidget by module/instance id The sof_ipc4_find_swidget_by_ids() can be used to find the swidget of a module instance. The lookup parameters are the module_id and the instance_id. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231124150853.18648-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-priv.h | 3 +++ sound/soc/sof/ipc4-topology.c | 20 ++++++++++++++++++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index 9e69b7c29117..fea93b693f4d 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -115,6 +115,9 @@ int sof_ipc4_reload_fw_libraries(struct snd_sof_dev *sdev); struct sof_ipc4_fw_module *sof_ipc4_find_module_by_uuid(struct snd_sof_dev *sdev, const guid_t *uuid); +struct snd_sof_widget *sof_ipc4_find_swidget_by_ids(struct snd_sof_dev *sdev, + u32 module_id, int instance_id); + struct sof_ipc4_base_module_cfg; void sof_ipc4_update_cpc_from_manifest(struct snd_sof_dev *sdev, struct sof_ipc4_fw_module *fw_module, diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index b24a64377f68..8ab9c42069ee 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -167,6 +167,26 @@ static const struct sof_token_info ipc4_token_list[SOF_TOKEN_COUNT] = { [SOF_SRC_TOKENS] = {"SRC tokens", src_tokens, ARRAY_SIZE(src_tokens)}, }; +struct snd_sof_widget *sof_ipc4_find_swidget_by_ids(struct snd_sof_dev *sdev, + u32 module_id, int instance_id) +{ + struct snd_sof_widget *swidget; + + list_for_each_entry(swidget, &sdev->widget_list, list) { + struct sof_ipc4_fw_module *fw_module = swidget->module_info; + + /* Only active module instances have valid instance_id */ + if (!swidget->use_count) + continue; + + if (fw_module && fw_module->man4_module_entry.id == module_id && + swidget->instance_id == instance_id) + return swidget; + } + + return NULL; +} + static void sof_ipc4_dbg_audio_format(struct device *dev, struct sof_ipc4_pin_format *pin_fmt, int num_formats) { From 1a307538c9cc274b3191b9a1380bbceece262626 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 24 Nov 2023 17:08:51 +0200 Subject: [PATCH 061/337] ASoC: SOF: ipc4: Add data struct for module notification message from firmware With the module notification message the information about the notification is provided via the mailbox with the sof_ipc4_notify_module_data struct. It contains the module and instance id of the sender of the notification, the event_id and optionally additional data which is module and event specific. At the same time add definitions to identify ALSA kcontrol change notification. These notifications use standardized event_id, modules must follow this if they support such notifications: upper 16 bit: 0xA15A as a magic identification value lower 16 bit: param_id of the changed control Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231124150853.18648-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/ipc4/header.h | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/include/sound/sof/ipc4/header.h b/include/sound/sof/ipc4/header.h index 574a9d581f88..2c81d6dde577 100644 --- a/include/sound/sof/ipc4/header.h +++ b/include/sound/sof/ipc4/header.h @@ -532,6 +532,35 @@ struct sof_ipc4_notify_resource_data { #define SOF_IPC4_DEBUG_SLOT_TELEMETRY 0x4c455400 #define SOF_IPC4_DEBUG_SLOT_BROKEN 0x44414544 +/** + * struct sof_ipc4_notify_module_data - payload for module notification + * @instance_id: instance ID of the originator module of the notification + * @module_id: module ID of the originator of the notification + * @event_id: module specific event id + * @event_data_size: Size of the @event_data (if any) in bytes + * @event_data: Optional notification data, module and notification dependent + */ +struct sof_ipc4_notify_module_data { + uint16_t instance_id; + uint16_t module_id; + uint32_t event_id; + uint32_t event_data_size; + uint8_t event_data[]; +} __packed __aligned(4); + +/* + * ALSA kcontrol change notification + * + * The event_id of struct sof_ipc4_notify_module_data is divided into two u16: + * upper u16: magic number for ALSA kcontrol types: 0xA15A + * lower u16: param_id of the control, which is the type of the control + * The event_data contains the struct sof_ipc4_control_msg_payload of the control + * which sent the notification. + */ +#define SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_MAGIC_MASK GENMASK(31, 16) +#define SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_MAGIC_VAL 0xA15A0000 +#define SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_PARAMID_MASK GENMASK(15, 0) + /** @}*/ #endif From f5eb9945cf9c17eb016aa64c7de13875f259ea07 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 24 Nov 2023 17:08:52 +0200 Subject: [PATCH 062/337] ASoC: SOF: ipc4-control: Implement control update for switch/enum controls Implement the sof_ipc_tplg_control_ops.update function to support a control change notification from the firmware on switch or enum control types. Based on the module notification message content, look up the swidget, then the scontrol which was the source of the notification then if the message contains the changed values update the cached values. If only a notification without values received, marked the control as dirty and on next read access fetch the new values from the firmware. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231124150853.18648-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-control.c | 179 +++++++++++++++++++++++++++++++++++ 1 file changed, 179 insertions(+) diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 938efaceb81c..3d2a35f27a87 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -240,6 +240,50 @@ sof_ipc4_set_generic_control_data(struct snd_sof_dev *sdev, return ret; } +static void sof_ipc4_refresh_generic_control(struct snd_sof_control *scontrol) +{ + struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; + struct snd_soc_component *scomp = scontrol->scomp; + struct sof_ipc4_control_msg_payload *data; + struct sof_ipc4_msg *msg = &cdata->msg; + size_t data_size; + unsigned int i; + int ret; + + if (!scontrol->comp_data_dirty) + return; + + if (!pm_runtime_active(scomp->dev)) + return; + + data_size = struct_size(data, chanv, scontrol->num_channels); + data = kmalloc(data_size, GFP_KERNEL); + if (!data) + return; + + data->id = cdata->index; + data->num_elems = scontrol->num_channels; + msg->data_ptr = data; + msg->data_size = data_size; + + scontrol->comp_data_dirty = false; + ret = sof_ipc4_set_get_kcontrol_data(scontrol, false, true); + msg->data_ptr = NULL; + msg->data_size = 0; + if (!ret) { + for (i = 0; i < scontrol->num_channels; i++) { + cdata->chanv[i].channel = data->chanv[i].channel; + cdata->chanv[i].value = data->chanv[i].value; + } + } else { + dev_err(scomp->dev, "Failed to read control data for %s\n", + scontrol->name); + scontrol->comp_data_dirty = true; + } + + kfree(data); +} + static bool sof_ipc4_switch_put(struct snd_sof_control *scontrol, struct snd_ctl_elem_value *ucontrol) { @@ -290,6 +334,8 @@ static int sof_ipc4_switch_get(struct snd_sof_control *scontrol, struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; unsigned int i; + sof_ipc4_refresh_generic_control(scontrol); + /* read back each channel */ for (i = 0; i < scontrol->num_channels; i++) ucontrol->value.integer.value[i] = cdata->chanv[i].value; @@ -347,6 +393,8 @@ static int sof_ipc4_enum_get(struct snd_sof_control *scontrol, struct sof_ipc4_control_data *cdata = scontrol->ipc_control_data; unsigned int i; + sof_ipc4_refresh_generic_control(scontrol); + /* read back each channel */ for (i = 0; i < scontrol->num_channels; i++) ucontrol->value.enumerated.item[i] = cdata->chanv[i].value; @@ -601,6 +649,136 @@ sof_ipc4_volsw_setup(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget, return sof_ipc4_set_volume_data(sdev, swidget, scontrol, false); } +#define PARAM_ID_FROM_EXTENSION(_ext) (((_ext) & SOF_IPC4_MOD_EXT_MSG_PARAM_ID_MASK) \ + >> SOF_IPC4_MOD_EXT_MSG_PARAM_ID_SHIFT) + +static void sof_ipc4_control_update(struct snd_sof_dev *sdev, void *ipc_message) +{ + struct sof_ipc4_msg *ipc4_msg = ipc_message; + struct sof_ipc4_notify_module_data *ndata = ipc4_msg->data_ptr; + struct sof_ipc4_control_msg_payload *msg_data; + struct sof_ipc4_control_data *cdata; + struct snd_soc_dapm_widget *widget; + struct snd_sof_control *scontrol; + struct snd_sof_widget *swidget; + struct snd_kcontrol *kc = NULL; + bool scontrol_found = false; + u32 event_param_id; + int i, type; + + if (ndata->event_data_size < sizeof(*msg_data)) { + dev_err(sdev->dev, + "%s: Invalid event data size for module %u.%u: %u\n", + __func__, ndata->module_id, ndata->instance_id, + ndata->event_data_size); + return; + } + + event_param_id = ndata->event_id & SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_PARAMID_MASK; + switch (event_param_id) { + case SOF_IPC4_SWITCH_CONTROL_PARAM_ID: + type = SND_SOC_TPLG_TYPE_MIXER; + break; + case SOF_IPC4_ENUM_CONTROL_PARAM_ID: + type = SND_SOC_TPLG_TYPE_ENUM; + break; + default: + dev_err(sdev->dev, + "%s: Invalid control type for module %u.%u: %u\n", + __func__, ndata->module_id, ndata->instance_id, + event_param_id); + return; + } + + /* Find the swidget based on ndata->module_id and ndata->instance_id */ + swidget = sof_ipc4_find_swidget_by_ids(sdev, ndata->module_id, + ndata->instance_id); + if (!swidget) { + dev_err(sdev->dev, "%s: Failed to find widget for module %u.%u\n", + __func__, ndata->module_id, ndata->instance_id); + return; + } + + /* Find the scontrol which is the source of the notification */ + msg_data = (struct sof_ipc4_control_msg_payload *)ndata->event_data; + list_for_each_entry(scontrol, &sdev->kcontrol_list, list) { + if (scontrol->comp_id == swidget->comp_id) { + u32 local_param_id; + + cdata = scontrol->ipc_control_data; + /* + * The scontrol's param_id is stored in the IPC message + * template's extension + */ + local_param_id = PARAM_ID_FROM_EXTENSION(cdata->msg.extension); + if (local_param_id == event_param_id && + msg_data->id == cdata->index) { + scontrol_found = true; + break; + } + } + } + + if (!scontrol_found) { + dev_err(sdev->dev, + "%s: Failed to find control on widget %s: %u:%u\n", + __func__, swidget->widget->name, ndata->event_id & 0xffff, + msg_data->id); + return; + } + + if (msg_data->num_elems) { + /* + * The message includes the updated value/data, update the + * control's local cache using the received notification + */ + for (i = 0; i < msg_data->num_elems; i++) { + u32 channel = msg_data->chanv[i].channel; + + if (channel >= scontrol->num_channels) { + dev_warn(sdev->dev, + "Invalid channel index for %s: %u\n", + scontrol->name, i); + + /* + * Mark the scontrol as dirty to force a refresh + * on next read + */ + scontrol->comp_data_dirty = true; + break; + } + + cdata->chanv[channel].value = msg_data->chanv[i].value; + } + } else { + /* + * Mark the scontrol as dirty because the value/data is changed + * in firmware, forcing a refresh on next read access + */ + scontrol->comp_data_dirty = true; + } + + /* + * Look up the ALSA kcontrol of the scontrol to be able to send a + * notification to user space + */ + widget = swidget->widget; + for (i = 0; i < widget->num_kcontrols; i++) { + /* skip non matching types or non matching indexes within type */ + if (widget->dobj.widget.kcontrol_type[i] == type && + widget->kcontrol_news[i].index == cdata->index) { + kc = widget->kcontrols[i]; + break; + } + } + + if (!kc) + return; + + snd_ctl_notify_one(swidget->scomp->card->snd_card, + SNDRV_CTL_EVENT_MASK_VALUE, kc, 0); +} + /* set up all controls for the widget */ static int sof_ipc4_widget_kcontrol_setup(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) { @@ -674,6 +852,7 @@ const struct sof_ipc_tplg_control_ops tplg_ipc4_control_ops = { .bytes_ext_put = sof_ipc4_bytes_ext_put, .bytes_ext_get = sof_ipc4_bytes_ext_get, .bytes_ext_volatile_get = sof_ipc4_bytes_ext_volatile_get, + .update = sof_ipc4_control_update, .widget_kcontrol_setup = sof_ipc4_widget_kcontrol_setup, .set_up_volume_table = sof_ipc4_set_up_volume_table, }; From 0ff23d460718641c80c8054425256391dca1ac7d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 24 Nov 2023 17:08:53 +0200 Subject: [PATCH 063/337] ASoC: SOF: ipc4: Handle ALSA kcontrol change notification from firmware The control change notification is sent as module notification with a standardized event_id (higher 16 bit is 0xA15A). Add generic code to handle the module notification and invoke the control update callback if the notification is an ALSA kcontrol change message. Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231124150853.18648-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4.c | 57 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index 8441f4ae4065..a9d9800d2fcc 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -78,6 +78,9 @@ static const struct sof_ipc4_fw_status { {165, "Reserved (ADSP_IPC_PIPELINE_ALREADY_EXISTS removed)"}, }; +typedef void (*ipc4_notification_handler)(struct snd_sof_dev *sdev, + struct sof_ipc4_msg *msg); + static int sof_ipc4_check_reply_status(struct snd_sof_dev *sdev, u32 status) { int i, ret; @@ -610,9 +613,55 @@ static int ipc4_fw_ready(struct snd_sof_dev *sdev, struct sof_ipc4_msg *ipc4_msg return sof_ipc4_init_msg_memory(sdev); } +static void sof_ipc4_module_notification_handler(struct snd_sof_dev *sdev, + struct sof_ipc4_msg *ipc4_msg) +{ + struct sof_ipc4_notify_module_data *data = ipc4_msg->data_ptr; + + /* + * If the notification includes additional, module specific data, then + * we need to re-allocate the buffer and re-read the whole payload, + * including the event_data + */ + if (data->event_data_size) { + void *new; + int ret; + + ipc4_msg->data_size += data->event_data_size; + + new = krealloc(ipc4_msg->data_ptr, ipc4_msg->data_size, GFP_KERNEL); + if (!new) { + ipc4_msg->data_size -= data->event_data_size; + return; + } + + /* re-read the whole payload */ + ipc4_msg->data_ptr = new; + ret = snd_sof_ipc_msg_data(sdev, NULL, ipc4_msg->data_ptr, + ipc4_msg->data_size); + if (ret < 0) { + dev_err(sdev->dev, + "Failed to read the full module notification: %d\n", + ret); + return; + } + data = ipc4_msg->data_ptr; + } + + /* Handle ALSA kcontrol notification */ + if ((data->event_id & SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_MAGIC_MASK) == + SOF_IPC4_NOTIFY_MODULE_EVENTID_ALSA_MAGIC_VAL) { + const struct sof_ipc_tplg_ops *tplg_ops = sdev->ipc->ops->tplg; + + if (tplg_ops->control->update) + tplg_ops->control->update(sdev, ipc4_msg); + } +} + static void sof_ipc4_rx_msg(struct snd_sof_dev *sdev) { struct sof_ipc4_msg *ipc4_msg = sdev->ipc->msg.rx_data; + ipc4_notification_handler handler_func = NULL; size_t data_size = 0; int err; @@ -648,6 +697,10 @@ static void sof_ipc4_rx_msg(struct snd_sof_dev *sdev) case SOF_IPC4_NOTIFY_EXCEPTION_CAUGHT: snd_sof_dsp_panic(sdev, 0, true); break; + case SOF_IPC4_NOTIFY_MODULE_NOTIFICATION: + data_size = sizeof(struct sof_ipc4_notify_module_data); + handler_func = sof_ipc4_module_notification_handler; + break; default: dev_dbg(sdev->dev, "Unhandled DSP message: %#x|%#x\n", ipc4_msg->primary, ipc4_msg->extension); @@ -663,6 +716,10 @@ static void sof_ipc4_rx_msg(struct snd_sof_dev *sdev) snd_sof_ipc_msg_data(sdev, NULL, ipc4_msg->data_ptr, ipc4_msg->data_size); } + /* Handle notifications with payload */ + if (handler_func) + handler_func(sdev, ipc4_msg); + sof_ipc4_log_header(sdev->dev, "ipc rx done ", ipc4_msg, true); if (data_size) { From 45cc50d13433a62f23b7b4af380497aae5e8ddc7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Nov 2023 01:28:27 +0000 Subject: [PATCH 064/337] ASoC: makes CPU/Codec channel connection map more generic Current ASoC CPU:Codec = N:M connection is using connection mapping idea, but it is used for N < M case only. We want to use it for any case. By this patch, not only N:M connection, but all existing connection (1:1, 1:N, N:N) will use same connection mapping. Then, because it will use default mapping, no conversion patch is needed to exising drivers. More over, CPU:Codec = N:M (N > M) also supported in the same time. ch_maps array will has CPU/Codec index by this patch. Image CPU0 <---> Codec0 CPU1 <-+-> Codec1 CPU2 <-/ ch_map ch_map[0].cpu = 0 ch_map[0].codec = 0 ch_map[1].cpu = 1 ch_map[1].codec = 1 ch_map[2].cpu = 2 ch_map[2].codec = 1 Link: https://lore.kernel.org/r/87fs6wuszr.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/878r7yqeo4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Kuninori Morimoto Reviewed-by: Bard Liao Tested-by: Pierre-Louis Bossart Tested-by: Jerome Brunet Link: https://lore.kernel.org/r/87ttpq4f2c.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 56 ++++++++++++++++++- sound/soc/intel/boards/sof_sdw.c | 28 ++++------ sound/soc/soc-core.c | 95 +++++++++++++++++++++++++++++++- sound/soc/soc-dapm.c | 45 ++++----------- sound/soc/soc-pcm.c | 44 +++++---------- 5 files changed, 185 insertions(+), 83 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7792c393e238..f3803c2dc349 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -655,8 +655,45 @@ struct snd_soc_dai_link_component { struct of_phandle_args *dai_args; }; -struct snd_soc_dai_link_codec_ch_map { - unsigned int connected_cpu_id; +/* + * [dai_link->ch_maps Image sample] + * + *------------------------- + * CPU0 <---> Codec0 + * + * ch-map[0].cpu = 0 ch-map[0].codec = 0 + * + *------------------------- + * CPU0 <---> Codec0 + * CPU1 <---> Codec1 + * CPU2 <---> Codec2 + * + * ch-map[0].cpu = 0 ch-map[0].codec = 0 + * ch-map[1].cpu = 1 ch-map[1].codec = 1 + * ch-map[2].cpu = 2 ch-map[2].codec = 2 + * + *------------------------- + * CPU0 <---> Codec0 + * CPU1 <-+-> Codec1 + * CPU2 <-/ + * + * ch-map[0].cpu = 0 ch-map[0].codec = 0 + * ch-map[1].cpu = 1 ch-map[1].codec = 1 + * ch-map[2].cpu = 2 ch-map[2].codec = 1 + * + *------------------------- + * CPU0 <---> Codec0 + * CPU1 <-+-> Codec1 + * \-> Codec2 + * + * ch-map[0].cpu = 0 ch-map[0].codec = 0 + * ch-map[1].cpu = 1 ch-map[1].codec = 1 + * ch-map[2].cpu = 1 ch-map[2].codec = 2 + * + */ +struct snd_soc_dai_link_ch_map { + unsigned int cpu; + unsigned int codec; unsigned int ch_mask; }; @@ -688,7 +725,9 @@ struct snd_soc_dai_link { struct snd_soc_dai_link_component *codecs; unsigned int num_codecs; - struct snd_soc_dai_link_codec_ch_map *codec_ch_maps; + /* num_ch_maps = max(num_cpu, num_codecs) */ + struct snd_soc_dai_link_ch_map *ch_maps; + /* * You MAY specify the link's platform/PCM/DMA driver, either by * device name, or by DT/OF node, but not both. Some forms of link @@ -775,6 +814,10 @@ struct snd_soc_dai_link { #endif }; +static inline int snd_soc_link_num_ch_map(struct snd_soc_dai_link *link) { + return max(link->num_cpus, link->num_codecs); +} + static inline struct snd_soc_dai_link_component* snd_soc_link_to_cpu(struct snd_soc_dai_link *link, int n) { return &(link)->cpus[n]; @@ -808,6 +851,12 @@ snd_soc_link_to_platform(struct snd_soc_dai_link *link, int n) { ((cpu) = snd_soc_link_to_cpu(link, i)); \ (i)++) +#define for_each_link_ch_maps(link, i, ch_map) \ + for ((i) = 0; \ + ((i) < snd_soc_link_num_ch_map(link) && \ + ((ch_map) = link->ch_maps + i)); \ + (i)++) + /* * Sample 1 : Single CPU/Codec/Platform * @@ -1163,6 +1212,7 @@ struct snd_soc_pcm_runtime { ((i) < (rtd)->dai_link->num_cpus + (rtd)->dai_link->num_codecs) && \ ((dai) = (rtd)->dais[i]); \ (i)++) +#define for_each_rtd_ch_maps(rtd, i, ch_maps) for_each_link_ch_maps(rtd->dai_link, i, ch_maps) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3312ad8a563b..2faf7372bad0 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -570,16 +570,14 @@ int sdw_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai_link_ch_map *ch_maps; int ch = params_channels(params); - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; unsigned int ch_mask; int num_codecs; int step; int i; - int j; - if (!rtd->dai_link->codec_ch_maps) + if (!rtd->dai_link->ch_maps) return 0; /* Identical data will be sent to all codecs in playback */ @@ -605,13 +603,9 @@ int sdw_hw_params(struct snd_pcm_substream *substream, * link has more than one codec DAIs. Set codec channel mask and * ASoC will set the corresponding channel numbers for each cpu dai. */ - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (rtd->dai_link->codec_ch_maps[j].connected_cpu_id != i) - continue; - rtd->dai_link->codec_ch_maps[j].ch_mask = ch_mask << (j * step); - } - } + for_each_link_ch_maps(rtd->dai_link, i, ch_maps) + ch_maps->ch_mask = ch_mask << (i * step); + return 0; } @@ -1350,15 +1344,17 @@ static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, return 0; } -static void set_dailink_map(struct snd_soc_dai_link_codec_ch_map *sdw_codec_ch_maps, +static void set_dailink_map(struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps, int codec_num, int cpu_num) { int step; int i; step = codec_num / cpu_num; - for (i = 0; i < codec_num; i++) - sdw_codec_ch_maps[i].connected_cpu_id = i / step; + for (i = 0; i < codec_num; i++) { + sdw_codec_ch_maps[i].cpu = i / step; + sdw_codec_ch_maps[i].codec = i; + } } static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; @@ -1453,7 +1449,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, *ignore_pch_dmic = true; for_each_pcm_streams(stream) { - struct snd_soc_dai_link_codec_ch_map *sdw_codec_ch_maps; + struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps; char *name, *cpu_name; int playback, capture; static const char * const sdw_stream_name[] = { @@ -1530,7 +1526,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *link_index, dai_links[*link_index].nonatomic = true; set_dailink_map(sdw_codec_ch_maps, codec_num, cpu_dai_num); - dai_links[*link_index].codec_ch_maps = sdw_codec_ch_maps; + dai_links[*link_index].ch_maps = sdw_codec_ch_maps; ret = set_codec_init_func(card, adr_link, dai_links + (*link_index)++, playback, group_id, adr_index, dai_index); if (ret < 0) { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b2bd45e87bc3..4ca3319a8e19 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1015,6 +1015,94 @@ component_dai_empty: return -EINVAL; } +#define MAX_DEFAULT_CH_MAP_SIZE 7 +static struct snd_soc_dai_link_ch_map default_ch_map_sync[MAX_DEFAULT_CH_MAP_SIZE] = { + { .cpu = 0, .codec = 0 }, + { .cpu = 1, .codec = 1 }, + { .cpu = 2, .codec = 2 }, + { .cpu = 3, .codec = 3 }, + { .cpu = 4, .codec = 4 }, + { .cpu = 5, .codec = 5 }, + { .cpu = 6, .codec = 6 }, +}; +static struct snd_soc_dai_link_ch_map default_ch_map_1cpu[MAX_DEFAULT_CH_MAP_SIZE] = { + { .cpu = 0, .codec = 0 }, + { .cpu = 0, .codec = 1 }, + { .cpu = 0, .codec = 2 }, + { .cpu = 0, .codec = 3 }, + { .cpu = 0, .codec = 4 }, + { .cpu = 0, .codec = 5 }, + { .cpu = 0, .codec = 6 }, +}; +static struct snd_soc_dai_link_ch_map default_ch_map_1codec[MAX_DEFAULT_CH_MAP_SIZE] = { + { .cpu = 0, .codec = 0 }, + { .cpu = 1, .codec = 0 }, + { .cpu = 2, .codec = 0 }, + { .cpu = 3, .codec = 0 }, + { .cpu = 4, .codec = 0 }, + { .cpu = 5, .codec = 0 }, + { .cpu = 6, .codec = 0 }, +}; +static int snd_soc_compensate_channel_connection_map(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_ch_map *ch_maps; + int i; + + /* + * dai_link->ch_maps indicates how CPU/Codec are connected. + * It will be a map seen from a larger number of DAI. + * see + * soc.h :: [dai_link->ch_maps Image sample] + */ + + /* it should have ch_maps if connection was N:M */ + if (dai_link->num_cpus > 1 && dai_link->num_codecs > 1 && + dai_link->num_cpus != dai_link->num_codecs && !dai_link->ch_maps) { + dev_err(card->dev, "need to have ch_maps when N:M connction (%s)", + dai_link->name); + return -EINVAL; + } + + /* do nothing if it has own maps */ + if (dai_link->ch_maps) + goto sanity_check; + + /* check default map size */ + if (dai_link->num_cpus > MAX_DEFAULT_CH_MAP_SIZE || + dai_link->num_codecs > MAX_DEFAULT_CH_MAP_SIZE) { + dev_err(card->dev, "soc-core.c needs update default_connection_maps"); + return -EINVAL; + } + + /* Compensate missing map for ... */ + if (dai_link->num_cpus == dai_link->num_codecs) + dai_link->ch_maps = default_ch_map_sync; /* for 1:1 or N:N */ + else if (dai_link->num_cpus < dai_link->num_codecs) + dai_link->ch_maps = default_ch_map_1cpu; /* for 1:N */ + else + dai_link->ch_maps = default_ch_map_1codec; /* for N:1 */ + +sanity_check: + dev_dbg(card->dev, "dai_link %s\n", dai_link->stream_name); + for_each_link_ch_maps(dai_link, i, ch_maps) { + if ((ch_maps->cpu >= dai_link->num_cpus) || + (ch_maps->codec >= dai_link->num_codecs)) { + dev_err(card->dev, + "unexpected dai_link->ch_maps[%d] index (cpu(%d/%d) codec(%d/%d))", + i, + ch_maps->cpu, dai_link->num_cpus, + ch_maps->codec, dai_link->num_codecs); + return -EINVAL; + } + + dev_dbg(card->dev, " [%d] cpu%d <-> codec%d\n", + i, ch_maps->cpu, ch_maps->codec); + } + + return 0; +} + /** * snd_soc_remove_pcm_runtime - Remove a pcm_runtime from card * @card: The ASoC card to which the pcm_runtime has @@ -1121,8 +1209,13 @@ int snd_soc_add_pcm_runtimes(struct snd_soc_card *card, int num_dai_link) { for (int i = 0; i < num_dai_link; i++) { - int ret = snd_soc_add_pcm_runtime(card, dai_link + i); + int ret; + ret = snd_soc_compensate_channel_connection_map(card, dai_link + i); + if (ret < 0) + return ret; + + ret = snd_soc_add_pcm_runtime(card, dai_link + i); if (ret < 0) return ret; } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 407a26af3eaf..bffeea80277f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4436,11 +4436,14 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; - int i; /* for each BE DAI link... */ for_each_card_rtds(card, rtd) { + struct snd_soc_dai_link_ch_map *ch_maps; + int i; + /* * dynamic FE links have no fixed DAI mapping. * CODEC<->CODEC links have no direct connection. @@ -4448,39 +4451,15 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) if (rtd->dai_link->dynamic) continue; - if (rtd->dai_link->num_cpus == 1) { - for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_connect_dai_pair(card, rtd, codec_dai, - snd_soc_rtd_to_cpu(rtd, 0)); - } else if (rtd->dai_link->num_codecs == rtd->dai_link->num_cpus) { - for_each_rtd_codec_dais(rtd, i, codec_dai) - dapm_connect_dai_pair(card, rtd, codec_dai, - snd_soc_rtd_to_cpu(rtd, i)); - } else if (rtd->dai_link->num_codecs > rtd->dai_link->num_cpus) { - int cpu_id; + /* + * see + * soc.h :: [dai_link->ch_maps Image sample] + */ + for_each_rtd_ch_maps(rtd, i, ch_maps) { + cpu_dai = snd_soc_rtd_to_cpu(rtd, ch_maps->cpu); + codec_dai = snd_soc_rtd_to_codec(rtd, ch_maps->codec); - if (!rtd->dai_link->codec_ch_maps) { - dev_err(card->dev, "%s: no codec channel mapping table provided\n", - __func__); - continue; - } - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - cpu_id = rtd->dai_link->codec_ch_maps[i].connected_cpu_id; - if (cpu_id >= rtd->dai_link->num_cpus) { - dev_err(card->dev, - "%s: dai_link %s cpu_id %d too large, num_cpus is %d\n", - __func__, rtd->dai_link->name, cpu_id, - rtd->dai_link->num_cpus); - continue; - } - dapm_connect_dai_pair(card, rtd, codec_dai, - snd_soc_rtd_to_cpu(rtd, cpu_id)); - } - } else { - dev_err(card->dev, - "%s: codec number %d < cpu number %d is not supported\n", - __func__, rtd->dai_link->num_codecs, rtd->dai_link->num_cpus); + dapm_connect_dai_pair(card, rtd, codec_dai, cpu_dai); } } } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 323e4d7b6adf..c20573aeb756 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1050,6 +1050,7 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, } for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + struct snd_soc_dai_link_ch_map *ch_maps; unsigned int ch_mask = 0; int j; @@ -1063,22 +1064,20 @@ static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, /* copy params for each cpu */ tmp_params = *params; - if (!rtd->dai_link->codec_ch_maps) - goto hw_params; /* * construct cpu channel mask by combining ch_mask of each * codec which maps to the cpu. + * see + * soc.h :: [dai_link->ch_maps Image sample] */ - for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (rtd->dai_link->codec_ch_maps[j].connected_cpu_id == i) - ch_mask |= rtd->dai_link->codec_ch_maps[j].ch_mask; - } + for_each_rtd_ch_maps(rtd, j, ch_maps) + if (ch_maps->cpu == i) + ch_mask |= ch_maps->ch_mask; /* fixup cpu channel number */ if (ch_mask) soc_pcm_codec_params_fixup(&tmp_params, ch_mask); -hw_params: ret = snd_soc_dai_hw_params(cpu_dai, substream, &tmp_params); if (ret < 0) goto out; @@ -2826,35 +2825,20 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, } } } else { + struct snd_soc_dai_link_ch_map *ch_maps; struct snd_soc_dai *codec_dai; /* Adapt stream for codec2codec links */ int cpu_capture = snd_soc_get_stream_cpu(dai_link, SNDRV_PCM_STREAM_CAPTURE); int cpu_playback = snd_soc_get_stream_cpu(dai_link, SNDRV_PCM_STREAM_PLAYBACK); - for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (dai_link->num_cpus == 1) { - cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); - } else if (dai_link->num_cpus == dai_link->num_codecs) { - cpu_dai = snd_soc_rtd_to_cpu(rtd, i); - } else if (rtd->dai_link->num_codecs > rtd->dai_link->num_cpus) { - int cpu_id; - - if (!rtd->dai_link->codec_ch_maps) { - dev_err(rtd->card->dev, "%s: no codec channel mapping table provided\n", - __func__); - return -EINVAL; - } - - cpu_id = rtd->dai_link->codec_ch_maps[i].connected_cpu_id; - cpu_dai = snd_soc_rtd_to_cpu(rtd, cpu_id); - } else { - dev_err(rtd->card->dev, - "%s codec number %d < cpu number %d is not supported\n", - __func__, rtd->dai_link->num_codecs, - rtd->dai_link->num_cpus); - return -EINVAL; - } + /* + * see + * soc.h :: [dai_link->ch_maps Image sample] + */ + for_each_rtd_ch_maps(rtd, i, ch_maps) { + cpu_dai = snd_soc_rtd_to_cpu(rtd, ch_maps->cpu); + codec_dai = snd_soc_rtd_to_codec(rtd, ch_maps->codec); if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) From 912eb415631140c93ff5f05378411fec8e6a537f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Nov 2023 01:28:58 +0000 Subject: [PATCH 065/337] ASoC: audio-graph-card2: use better image for Multi connection 1st port on Multi ports is for paired CPU/Codec, and the 2nd or later port are for Multi Elements. This patch indicates its image to easy to understand. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87sf5a4f1i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 52 +++++++++++++-------------- 1 file changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 7146611df730..c564f630abf6 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -83,32 +83,32 @@ Multi-CPU/Codec ************************************ -It has connection part (= X) and list part (= y). -links indicates connection part of CPU side (= A). +It has link connection part (= X,x) and list part (= A,B,a,b). +"links" is connection part of CPU side (= @). - +-+ (A) +-+ - CPU1 --(y) | | <-(X)--(X)-> | | (y)-- Codec1 - CPU2 --(y) | | | | (y)-- Codec2 - +-+ +-+ + +----+ +---+ + CPU1 --|A X| <-@----> |x a|-- Codec1 + CPU2 --|B | | b|-- Codec2 + +----+ +---+ - sound { - compatible = "audio-graph-card2"; + sound { + compatible = "audio-graph-card2"; -(A) links = <&mcpu>; +(@) links = <&mcpu>; - multi { - ports@0 { -(X) (A) mcpu: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; -(y) port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; -(y) port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; - }; - ports@1 { -(X) port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; -(y) port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; -(y) port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; - }; + multi { + ports@0 { +(@) mcpu: port@0 { mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; // (X) to pair + port@1 { mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; // (A) Multi Element + port@2 { mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; // (B) Multi Element + }; + ports@1 { + port@0 { mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; // (x) to pair + port@1 { mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; // (a) Multi Element + port@2 { mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; // (b) Multi Element }; }; + }; CPU { ports { @@ -328,9 +328,9 @@ static struct device_node *graph_get_next_multi_ep(struct device_node **port) /* * multi { * ports { - * => lnk: port@0 { ... }; - * port@1 { ep { ... = rep0 } }; - * port@2 { ep { ... = rep1 } }; + * => lnk: port@0 { ... }; // to pair + * port@1 { ep { ... = rep0 } }; // Multi Element + * port@2 { ep { ... = rep1 } }; // Multi Element * ... * }; * }; @@ -920,9 +920,9 @@ static int graph_counter(struct device_node *lnk) * * multi { * ports { - * => lnk: port@0 { ... }; - * port@1 { ... }; - * port@2 { ... }; + * => lnk: port@0 { ... }; // to pair + * port@1 { ... }; // Multi Element + * port@2 { ... }; // Multi Element * ... * }; * }; From e2de6808df4ad5faa6106f7a80617921fdf5dff5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Nov 2023 01:29:23 +0000 Subject: [PATCH 066/337] ASoC: audio-graph-card2: add CPU:Codec = N:M support Now ASoC is supporting CPU:Codec = N:M support. This patch enables it on Audio Graph Card2. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87r0ku4f0t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 237 ++++++++++++++++++++++---- 1 file changed, 208 insertions(+), 29 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index c564f630abf6..d9e10308a508 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -504,38 +504,201 @@ static int __graph_parse_node(struct simple_util_priv *priv, return 0; } +static int graph_parse_node_multi_nm(struct snd_soc_dai_link *dai_link, + int *nm_idx, int cpu_idx, + struct device_node *mcpu_port) +{ + /* + * +---+ +---+ + * | X|<-@------->|x | + * | | | | + * cpu0 <--|A 1|<--------->|4 a|-> codec0 + * cpu1 <--|B 2|<-----+--->|5 b|-> codec1 + * cpu2 <--|C 3|<----/ +---+ + * +---+ + * + * multi { + * ports { + * port@0 { mcpu_top_ep {... = mcodec_ep; }; }; // (X) to pair + * port@1 { mcpu0_ep { ... = cpu0_ep; }; // (A) Multi Element + * mcpu0_ep_0 { ... = mcodec0_ep_0; }; }; // (1) connected Codec + * port@2 { mcpu1_ep { ... = cpu1_ep; }; // (B) Multi Element + * mcpu1_ep_0 { ... = mcodec1_ep_0; }; }; // (2) connected Codec + * port@3 { mcpu2_ep { ... = cpu2_ep; }; // (C) Multi Element + * mcpu2_ep_0 { ... = mcodec1_ep_1; }; }; // (3) connected Codec + * }; + * + * ports { + * port@0 { mcodec_top_ep {... = mcpu_ep; }; }; // (x) to pair + * port@1 { mcodec0_ep { ... = codec0_ep; }; // (a) Multi Element + * mcodec0_ep_0 { ... = mcpu0_ep_0; }; }; // (4) connected CPU + * port@2 { mcodec1_ep { ... = codec1_ep; }; // (b) Multi Element + * mcodec1_ep_0 { ... = mcpu1_ep_0; }; // (5) connected CPU + * mcodec1_ep_1 { ... = mcpu2_ep_0; }; }; // (5) connected CPU + * }; + * }; + */ + struct device_node *mcpu_ep = port_to_endpoint(mcpu_port); + struct device_node *mcpu_ep_n = mcpu_ep; + struct device_node *mcpu_port_top = of_get_next_child(of_get_parent(mcpu_port), NULL); + struct device_node *mcpu_ep_top = port_to_endpoint(mcpu_port_top); + struct device_node *mcodec_ep_top = of_graph_get_remote_endpoint(mcpu_ep_top); + struct device_node *mcodec_port_top = of_get_parent(mcodec_ep_top); + struct device_node *mcodec_ports = of_get_parent(mcodec_port_top); + int nm_max = max(dai_link->num_cpus, dai_link->num_codecs); + int ret = -EINVAL; + + if (cpu_idx > dai_link->num_cpus) + goto mcpu_err; + + while (1) { + struct device_node *mcodec_ep_n; + struct device_node *mcodec_port_i; + struct device_node *mcodec_port; + int codec_idx; + + if (*nm_idx > nm_max) + break; + + mcpu_ep_n = of_get_next_child(mcpu_port, mcpu_ep_n); + if (!mcpu_ep_n) { + ret = 0; + break; + } + + mcodec_ep_n = of_graph_get_remote_endpoint(mcpu_ep_n); + mcodec_port = of_get_parent(mcodec_ep_n); + + if (mcodec_ports != of_get_parent(mcodec_port)) + goto mcpu_err; + + codec_idx = 0; + mcodec_port_i = of_get_next_child(mcodec_ports, NULL); + while (1) { + if (codec_idx > dai_link->num_codecs) + goto mcodec_err; + + mcodec_port_i = of_get_next_child(mcodec_ports, mcodec_port_i); + + if (!mcodec_port_i) + goto mcodec_err; + + if (mcodec_port_i == mcodec_port) + break; + + codec_idx++; + } + + dai_link->ch_maps[*nm_idx].cpu = cpu_idx; + dai_link->ch_maps[*nm_idx].codec = codec_idx; + + (*nm_idx)++; + + of_node_put(mcodec_port_i); +mcodec_err: + of_node_put(mcodec_port); + of_node_put(mcpu_ep_n); + of_node_put(mcodec_ep_n); + } +mcpu_err: + of_node_put(mcpu_ep); + of_node_put(mcpu_port_top); + of_node_put(mcpu_ep_top); + of_node_put(mcodec_ep_top); + of_node_put(mcodec_port_top); + of_node_put(mcodec_ports); + + return ret; +} + +static int graph_parse_node_multi(struct simple_util_priv *priv, + enum graph_type gtype, + struct device_node *port, + struct link_info *li, int is_cpu) +{ + struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct device *dev = simple_priv_to_dev(priv); + struct device_node *ep; + int ret = -ENOMEM; + int nm_idx = 0; + int nm_max = max(dai_link->num_cpus, dai_link->num_codecs); + + /* + * create ch_maps if CPU:Codec = N:M + * DPCM is out of scope + */ + if (gtype != GRAPH_DPCM && !dai_link->ch_maps && + dai_link->num_cpus > 1 && dai_link->num_codecs > 1 && + dai_link->num_cpus != dai_link->num_codecs) { + + dai_link->ch_maps = devm_kcalloc(dev, nm_max, + sizeof(struct snd_soc_dai_link_ch_map), GFP_KERNEL); + if (!dai_link->ch_maps) + goto multi_err; + } + + for (int idx = 0;; idx++) { + /* + * multi { + * ports { + * port@0 { ... }; // to pair + * port@1 { mcpu1_ep { ... = cpu1_ep };}; // Multi Element + * port@2 { mcpu2_ep { ... = cpu2_ep };}; // Multi Element + * }; + * }; + * + * cpu { + * ports { + * port@0 { cpu1_ep { ... = mcpu1_ep };}; + * }; + * }; + */ + ep = graph_get_next_multi_ep(&port); + if (!ep) + break; + + ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, idx); + of_node_put(ep); + if (ret < 0) + goto multi_err; + + /* CPU:Codec = N:M */ + if (is_cpu && dai_link->ch_maps) { + ret = graph_parse_node_multi_nm(dai_link, &nm_idx, idx, port); + if (ret < 0) + goto multi_err; + } + } + + if (is_cpu && dai_link->ch_maps && (nm_idx != nm_max)) + ret = -EINVAL; + +multi_err: + return ret; +} + +static int graph_parse_node_single(struct simple_util_priv *priv, + enum graph_type gtype, + struct device_node *port, + struct link_info *li, int is_cpu) +{ + struct device_node *ep = port_to_endpoint(port); + int ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); + + of_node_put(ep); + + return ret; +} + static int graph_parse_node(struct simple_util_priv *priv, enum graph_type gtype, struct device_node *port, struct link_info *li, int is_cpu) { - struct device_node *ep; - int ret = 0; - - if (graph_lnk_is_multi(port)) { - int idx; - - of_node_get(port); - - for (idx = 0;; idx++) { - ep = graph_get_next_multi_ep(&port); - if (!ep) - break; - - ret = __graph_parse_node(priv, gtype, ep, - li, is_cpu, idx); - of_node_put(ep); - if (ret < 0) - break; - } - } else { - /* Single CPU / Codec */ - ep = port_to_endpoint(port); - ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); - } - - return ret; + if (graph_lnk_is_multi(port)) + return graph_parse_node_multi(priv, gtype, port, li, is_cpu); + else + return graph_parse_node_single(priv, gtype, port, li, is_cpu); } static void graph_parse_daifmt(struct device_node *node, @@ -929,8 +1092,24 @@ static int graph_counter(struct device_node *lnk) * * ignore first lnk part */ - if (graph_lnk_is_multi(lnk)) - return of_graph_get_endpoint_count(of_get_parent(lnk)) - 1; + if (graph_lnk_is_multi(lnk)) { + struct device_node *ports = of_get_parent(lnk); + struct device_node *port = NULL; + int cnt = 0; + + /* + * CPU/Codec = N:M case has many endpoints. + * We can't use of_graph_get_endpoint_count() here + */ + while(1) { + port = of_get_next_child(ports, port); + if (!port) + break; + cnt++; + } + + return cnt - 1; + } /* * Single CPU / Codec */ From a706366f93c37c6649acfe15a1ef9a80e25bace4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Nov 2023 01:30:09 +0000 Subject: [PATCH 067/337] ASoC: audio-graph-card2-custom-sample: Add connection image Audio Graph Card2 is supporting many type of Sound connections, but thus it is very difficult to understand how these are connected. To support well understanding, adds each connection images and indicates each settings are for where. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87pm0e4ezi.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- .../audio-graph-card2-custom-sample.dtsi | 181 ++++++++++++++---- 1 file changed, 142 insertions(+), 39 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 8acaa2ddb335..7e2cd9cc3fa8 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -58,7 +58,7 @@ * | |-> codec13 * +-+ * - * [Multi-CPU/Codec] + * [Multi-CPU/Codec-0] * +-+ +-+ * cpu1 <--| |<-@--------->| |-> codec1 * cpu2 <--| | | |-> codec2 @@ -144,11 +144,14 @@ */ &cpu0 - /* [Semi-Multi] */ + /* + * [Semi-Multi] + * cpu7/codec12/codec13 + */ &sm0 /* - * [Multi-CPU/Codec]: cpu side only + * [Multi-CPU/Codec-0]: cpu side only * cpu1/cpu2/codec1/codec2 */ &mcpu0 @@ -182,64 +185,115 @@ #address-cells = <1>; #size-cells = <0>; + /* + * [Multi-CPU-0] + * + * +---+ +---+ + * cpu1 <--|A X|<-@------->|x a|-> codec1 + * cpu2 <--|B | | b|-> codec2 + * +---+ +---+ + */ ports@0 { reg = <0>; #address-cells = <1>; #size-cells = <0>; - /* [Multi-CPU] */ - mcpu0: port@0 { reg = <0>; mcpu0_ep: endpoint { remote-endpoint = <&mcodec0_ep>; }; }; - port@1 { reg = <1>; mcpu1_ep: endpoint { remote-endpoint = <&cpu1_ep>; }; }; - port@2 { reg = <2>; mcpu2_ep: endpoint { remote-endpoint = <&cpu2_ep>; }; }; + mcpu0: port@0 { reg = <0>; mcpu00_ep: endpoint { remote-endpoint = <&mcodec00_ep>; };};/* (X) to pair */ + port@1 { reg = <1>; mcpu01_ep: endpoint { remote-endpoint = <&cpu1_ep>; };};/* (A) Multi Element */ + port@2 { reg = <2>; mcpu02_ep: endpoint { remote-endpoint = <&cpu2_ep>; };};/* (B) Multi Element */ }; - /* [Multi-Codec] */ + /* + * [Multi-Codec-0] + * + * +---+ +---+ + * cpu1 <--|A X|<-@------->|x a|-> codec1 + * cpu2 <--|B | | b|-> codec2 + * +---+ +---+ + */ ports@1 { reg = <1>; #address-cells = <1>; #size-cells = <0>; - port@0 { reg = <0>; mcodec0_ep: endpoint { remote-endpoint = <&mcpu0_ep>; }; }; - port@1 { reg = <1>; mcodec1_ep: endpoint { remote-endpoint = <&codec1_ep>; }; }; - port@2 { reg = <2>; mcodec2_ep: endpoint { remote-endpoint = <&codec2_ep>; }; }; + port@0 { reg = <0>; mcodec00_ep: endpoint { remote-endpoint = <&mcpu00_ep>; };};/* (x) to pair */ + port@1 { reg = <1>; mcodec01_ep: endpoint { remote-endpoint = <&codec1_ep>; };};/* (a) Multi Element */ + port@2 { reg = <2>; mcodec02_ep: endpoint { remote-endpoint = <&codec2_ep>; };};/* (b) Multi Element */ }; - /* [DPCM-Multi]::BE */ + /* + * [DPCM-Multi]::BE + * + * FE BE + * **** +---+ + * cpu5 <-@--* *-----@--->|x a|-> codec4 + * cpu6 <-@--* * | b|-> codec5 + * **** +---+ + */ ports@2 { reg = <2>; #address-cells = <1>; #size-cells = <0>; - port@0 { reg = <0>; mbe_ep: endpoint { remote-endpoint = <&be10_ep>; }; }; - port@1 { reg = <1>; mbe1_ep: endpoint { remote-endpoint = <&codec4_ep>; }; }; - port@2 { reg = <2>; mbe2_ep: endpoint { remote-endpoint = <&codec5_ep>; }; }; + port@0 { reg = <0>; mbe_ep: endpoint { remote-endpoint = <&be10_ep>; };};/* (x) to pair */ + port@1 { reg = <1>; mbe1_ep: endpoint { remote-endpoint = <&codec4_ep>; };};/* (a) Multi Element */ + port@2 { reg = <2>; mbe2_ep: endpoint { remote-endpoint = <&codec5_ep>; };};/* (b) Multi Element */ }; - /* [Codec2Codec-Multi]::CPU */ + /* + * [Codec2Codec-Multi]::CPU + * + * +---+ + * +-@->|X A|-> codec8 + * | | B|-> codec9 + * | +---+ + * | +---+ + * +--->|x a|-> codec10 + * | b|-> codec11 + * +---+ + */ ports@3 { reg = <3>; #address-cells = <1>; #size-cells = <0>; - port@0 { reg = <0>; mc2c0_ep: endpoint { remote-endpoint = <&c2cmf_ep>; }; }; - port@1 { reg = <1>; mc2c00_ep: endpoint { remote-endpoint = <&codec8_ep>; }; }; - port@2 { reg = <2>; mc2c01_ep: endpoint { remote-endpoint = <&codec9_ep>; }; }; + port@0 { reg = <0>; mc2c0_ep: endpoint { remote-endpoint = <&c2cmf_ep>; };};/* (X) to pair */ + port@1 { reg = <1>; mc2c00_ep: endpoint { remote-endpoint = <&codec8_ep>; };};/* (A) Multi Element */ + port@2 { reg = <2>; mc2c01_ep: endpoint { remote-endpoint = <&codec9_ep>; };};/* (B) Multi Element */ }; - /* [Codec2Codec-Multi]::Codec */ + /* + * [Codec2Codec-Multi]::Codec + * + * +---+ + * +-@->|X A|-> codec8 + * | | B|-> codec9 + * | +---+ + * | +---+ + * +--->|x a|-> codec10 + * | b|-> codec11 + * +---+ + */ ports@4 { reg = <4>; #address-cells = <1>; #size-cells = <0>; - port@0 { reg = <0>; mc2c1_ep: endpoint { remote-endpoint = <&c2cmb_ep>; }; }; - port@1 { reg = <1>; mc2c10_ep: endpoint { remote-endpoint = <&codec10_ep>; }; }; - port@2 { reg = <2>; mc2c11_ep: endpoint { remote-endpoint = <&codec11_ep>; }; }; + port@0 { reg = <0>; mc2c1_ep: endpoint { remote-endpoint = <&c2cmb_ep>; };};/* (x) to pair */ + port@1 { reg = <1>; mc2c10_ep: endpoint { remote-endpoint = <&codec10_ep>; };};/* (a) Multi Element */ + port@2 { reg = <2>; mc2c11_ep: endpoint { remote-endpoint = <&codec11_ep>; };};/* (b) Multi Element */ }; - /* [Semi-Multi] */ + /* + * [Semi-Multi] + * + * +---+ + * cpu7 <-@------->|X A|-> codec12 + * | B|-> codec13 + * +---+ + */ ports@5 { reg = <5>; #address-cells = <1>; #size-cells = <0>; - port@0 { reg = <0>; smcodec0_ep: endpoint { remote-endpoint = <&cpu7_ep>; }; }; - port@1 { reg = <1>; smcodec1_ep: endpoint { remote-endpoint = <&codec12_ep>; }; }; - port@2 { reg = <2>; smcodec2_ep: endpoint { remote-endpoint = <&codec13_ep>; }; }; + port@0 { reg = <0>; smcodec0_ep: endpoint { remote-endpoint = <&cpu7_ep>; };};/* (X) to pair */ + port@1 { reg = <1>; smcodec1_ep: endpoint { remote-endpoint = <&codec12_ep>; };};/* (A) Multi Element */ + port@2 { reg = <2>; smcodec2_ep: endpoint { remote-endpoint = <&codec13_ep>; };};/* (B) Multi Element */ }; }; @@ -252,11 +306,27 @@ #address-cells = <1>; #size-cells = <0>; - /* [DPCM]::FE */ + /* + * [DPCM]::FE + * + * FE BE + * **** + * cpu3 <-@(fe00)--* *--(be0)@--> codec3 + * cpu4 <-@(fe01)--* * (44.1kHz) + * **** + */ fe00: port@0 { reg = <0>; fe00_ep: endpoint { remote-endpoint = <&cpu3_ep>; }; }; fe01: port@1 { reg = <1>; fe01_ep: endpoint { remote-endpoint = <&cpu4_ep>; }; }; - /* [DPCM-Multi]::FE */ + /* + * [DPCM-Multi]::FE + * + * FE BE + * **** +-+ + * cpu5 <-@(fe10)--* *---(be1)@-->| |-> codec4 + * cpu6 <-@(fe11)--* * | |-> codec5 + * **** +-+ + */ fe10: port@2 { reg = <2>; fe10_ep: endpoint { remote-endpoint = <&cpu5_ep>; }; }; fe11: port@3 { reg = <3>; fe11_ep: endpoint { remote-endpoint = <&cpu6_ep>; }; }; }; @@ -266,10 +336,26 @@ #address-cells = <1>; #size-cells = <0>; - /* [DPCM]::BE */ + /* + * [DPCM]::BE + * + * FE BE + * **** + * cpu3 <-@(fe00)--* *--(be0)@--> codec3 + * cpu4 <-@(fe01)--* * (44.1kHz) + * **** + */ be0: port@0 { reg = <0>; be00_ep: endpoint { remote-endpoint = <&codec3_ep>; }; }; - /* [DPCM-Multi]::BE */ + /* + * [DPCM-Multi]::BE + * + * FE BE + * **** +-+ + * cpu5 <-@(fe10)--* *---(be1)@-->| |-> codec4 + * cpu6 <-@(fe11)--* * | |-> codec5 + * **** +-+ + */ be1: port@1 { reg = <1>; be10_ep: endpoint { remote-endpoint = <&mbe_ep>; }; }; }; }; @@ -277,7 +363,13 @@ codec2codec { #address-cells = <1>; #size-cells = <0>; - /* [Codec2Codec] */ + /* + * [Codec2Codec] + * + * +-@(c2c)-> codec6 + * | + * +--------> codec7 + */ ports@0 { reg = <0>; @@ -289,7 +381,18 @@ port@1 { reg = <1>; c2cb_ep: endpoint { remote-endpoint = <&codec7_ep>; }; }; }; - /* [Codec2Codec-Multi] */ + /* + * [Codec2Codec-Multi] + * + * +-+ + * +-@(c2c_m)-->| |-> codec8 + * | | |-> codec9 + * | +-+ + * | +-+ + * +----------->| |-> codec10 + * | |-> codec11 + * +-+ + */ ports@1 { reg = <1>; @@ -323,9 +426,9 @@ /* [Normal] */ cpu0: port@0 { reg = <0>; cpu0_ep: endpoint { remote-endpoint = <&codec0_ep>; }; }; - /* [Multi-CPU] */ - port@1 { reg = <1>; cpu1_ep: endpoint { remote-endpoint = <&mcpu1_ep>; }; }; - port@2 { reg = <2>; cpu2_ep: endpoint { remote-endpoint = <&mcpu2_ep>; }; }; + /* [Multi-CPU-0] */ + port@1 { reg = <1>; cpu1_ep: endpoint { remote-endpoint = <&mcpu01_ep>; }; }; + port@2 { reg = <2>; cpu2_ep: endpoint { remote-endpoint = <&mcpu02_ep>; }; }; /* [DPCM]::FE */ port@3 { reg = <3>; cpu3_ep: endpoint { remote-endpoint = <&fe00_ep>; }; }; @@ -363,9 +466,9 @@ /* [Normal] */ port@0 { reg = <0>; codec0_ep: endpoint { remote-endpoint = <&cpu0_ep>; }; }; - /* [Multi-Codec] */ - port@1 { reg = <1>; codec1_ep: endpoint { remote-endpoint = <&mcodec1_ep>; }; }; - port@2 { reg = <2>; codec2_ep: endpoint { remote-endpoint = <&mcodec2_ep>; }; }; + /* [Multi-Codec-0] */ + port@1 { reg = <1>; codec1_ep: endpoint { remote-endpoint = <&mcodec01_ep>; }; }; + port@2 { reg = <2>; codec2_ep: endpoint { remote-endpoint = <&mcodec02_ep>; }; }; /* [DPCM]::BE */ port@3 { From 792846d9daa876186196b66dc496a2ba8ddd7535 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 13 Nov 2023 01:31:04 +0000 Subject: [PATCH 068/337] ASoC: audio-graph-card2-custom-sample: add CPU/Codec = N:M sample Now ASoC is supporting CPU/Codec = N:M connection, add its sample settings. One note here is that it has many type of samples, it reached to maximum of sound minor number. Therefore, new sample is disabled so far. If you want to try it, you need to disable some other one instead. Signed-off-by: Kuninori Morimoto Link: https://lore.kernel.org/r/87o7fy4ey0.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- .../audio-graph-card2-custom-sample.dtsi | 199 ++++++++++++++++++ 1 file changed, 199 insertions(+) diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi index 7e2cd9cc3fa8..9efd31206c9b 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.dtsi +++ b/sound/soc/generic/audio-graph-card2-custom-sample.dtsi @@ -64,6 +64,26 @@ * cpu2 <--| | | |-> codec2 * +-+ +-+ * + * [Multi-CPU/Codec-1] + * + * +-+ +-+ + * | |<-@--------->| | + * | | | | + * cpu8 <--| |<----------->| |-> codec14 + * cpu9 <--| |<---+------->| |-> codec15 + * +-+ \------>| |-> codec16 + * +-+ + * + * [Multi-CPU/Codec-2] + * + * +-+ +-+ + * | |<-@--------->| | + * | | | | + * cpu10 <-| |<----------->| |-> codec17 + * cpu11 <-| |<-----+----->| |-> codec18 + * cpu12 <-| |<----/ +-+ + * +-+ + * * [DPCM] * * CPU3/CPU4 are converting rate to 44100 @@ -156,6 +176,26 @@ */ &mcpu0 + /* + * [Multi-CPU/Codec-1]: cpu side only + * cpu8/cpu9/codec14/codec15/codec16 + * + * Because it will reach to the maximum of sound minor number, + * disable it so far. + * If you want to try it, please disable some other one instead. + */ + //&mcpu1 + + /* + * [Multi-CPU/Codec-2]: cpu side only + * cpu10/cpu11/cpu12/codec17/codec18 + * + * Because it will reach to the maximum of sound minor number, + * disable it so far. + * If you want to try it, please disable some other one instead. + */ + //&mcpu2 + /* * [DPCM]: both FE / BE * cpu3/cpu4/codec3 @@ -295,6 +335,150 @@ port@1 { reg = <1>; smcodec1_ep: endpoint { remote-endpoint = <&codec12_ep>; };};/* (A) Multi Element */ port@2 { reg = <2>; smcodec2_ep: endpoint { remote-endpoint = <&codec13_ep>; };};/* (B) Multi Element */ }; + + /* + * [Multi-CPU-1] + * + * +---+ +---+ + * | X|<-@------->|x | + * | | | | + * cpu8 <--|A 1|<--------->|3 a|-> codec14 + * cpu9 <--|B 2|<---+----->|4 b|-> codec15 + * +---+ \---->|5 c|-> codec16 + * +---+ + */ + ports@6 { + reg = <6>; + #address-cells = <1>; + #size-cells = <0>; + mcpu1: port@0 { reg = <0>; mcpu10_ep: endpoint { remote-endpoint = <&mcodec10_ep>; };}; /* (X) to pair */ + port@1 { + #address-cells = <1>; + #size-cells = <0>; + reg = <1>; + mcpu11_ep: endpoint@0 { reg = <0>; remote-endpoint = <&cpu8_ep>; }; /* (A) Multi Element */ + mcpu11_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcodec11_ep_0>; }; /* (1) connected Codec */ + }; + port@2 { + #address-cells = <1>; + #size-cells = <0>; + reg = <2>; + mcpu12_ep: endpoint@0 { reg = <0>; remote-endpoint = <&cpu9_ep>; }; /* (B) Multi Element */ + mcpu12_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcodec12_ep_0>; }; /* (2) connected Codec */ + mcpu12_ep_1: endpoint@2 { reg = <2>; remote-endpoint = <&mcodec13_ep_0>; }; /* (2) connected Codec */ + }; + }; + + /* + * [Multi-Codec-1] + * + * +---+ +---+ + * | X|<-@------->|x | + * | | | | + * cpu8 <--|A 1|<--------->|3 a|-> codec14 + * cpu9 <--|B 2|<---+----->|4 b|-> codec15 + * +---+ \---->|5 c|-> codec16 + * +---+ + */ + ports@7 { + reg = <7>; + #address-cells = <1>; + #size-cells = <0>; + port@0 { reg = <0>; mcodec10_ep: endpoint { remote-endpoint = <&mcpu10_ep>; };}; /* (x) to pair */ + port@1 { + #address-cells = <1>; + #size-cells = <0>; + reg = <1>; + mcodec11_ep: endpoint@0 { reg = <0>; remote-endpoint = <&codec14_ep>; }; /* (a) Multi Element */ + mcodec11_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcpu11_ep_0>; }; /* (3) connected CPU */ + }; + port@2 { + #address-cells = <1>; + #size-cells = <0>; + reg = <2>; + mcodec12_ep: endpoint@0 { reg = <0>; remote-endpoint = <&codec15_ep>; }; /* (b) Multi Element */ + mcodec12_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcpu12_ep_0>; }; /* (4) connected CPU */ + }; + port@3 { + #address-cells = <1>; + #size-cells = <0>; + reg = <3>; + mcodec13_ep: endpoint@0 { reg = <0>; remote-endpoint = <&codec16_ep>; }; /* (c) Multi Element */ + mcodec13_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcpu12_ep_1>; }; /* (5) connected CPU */ + }; + }; + + /* + * [Multi-CPU-2] + * + * +---+ +---+ + * | X|<-@------->|x | + * | | | | + * cpu10 <-|A 1|<--------->|4 a|-> codec17 + * cpu11 <-|B 2|<-----+--->|5 b|-> codec18 + * cpu12 <-|C 3|<----/ +---+ + * +---+ + */ + ports@8 { + reg = <8>; + #address-cells = <1>; + #size-cells = <0>; + mcpu2: port@0 { reg = <0>; mcpu20_ep: endpoint { remote-endpoint = <&mcodec20_ep>; };}; /* (X) to pair */ + port@1 { + #address-cells = <1>; + #size-cells = <0>; + reg = <1>; + mcpu21_ep: endpoint@0 { reg = <0>; remote-endpoint = <&cpu10_ep>; }; /* (A) Multi Element */ + mcpu21_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcodec21_ep_0>; }; /* (1) connected Codec */ + }; + port@2 { + #address-cells = <1>; + #size-cells = <0>; + reg = <2>; + mcpu22_ep: endpoint@0 { reg = <0>; remote-endpoint = <&cpu11_ep>; }; /* (B) Multi Element */ + mcpu22_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcodec22_ep_0>; }; /* (2) connected Codec */ + }; + port@3 { + #address-cells = <1>; + #size-cells = <0>; + reg = <3>; + mcpu23_ep: endpoint@0 { reg = <0>; remote-endpoint = <&cpu12_ep>; }; /* (C) Multi Element */ + mcpu23_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcodec22_ep_1>; }; /* (3) connected Codec */ + }; + }; + + /* + * [Multi-Codec-2] + * + * +---+ +---+ + * | X|<-@------->|x | + * | | | | + * cpu10 <-|A 1|<--------->|4 a|-> codec17 + * cpu11 <-|B 2|<-----+--->|5 b|-> codec18 + * cpu12 <-|C 3|<----/ +---+ + * +---+ + */ + ports@9 { + reg = <9>; + #address-cells = <1>; + #size-cells = <0>; + port@0 { reg = <0>; mcodec20_ep: endpoint { remote-endpoint = <&mcpu20_ep>; };}; /* (x) to pair */ + port@1 { + #address-cells = <1>; + #size-cells = <0>; + reg = <1>; + mcodec21_ep: endpoint@0 { reg = <0>; remote-endpoint = <&codec17_ep>; }; /* (a) Multi Element */ + mcodec21_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcpu21_ep_0>; }; /* (4) connected CPU */ + }; + port@2 { + #address-cells = <1>; + #size-cells = <0>; + reg = <2>; + mcodec22_ep: endpoint@0 { reg = <0>; remote-endpoint = <&codec18_ep>; }; /* (b) Multi Element */ + mcodec22_ep_0: endpoint@1 { reg = <1>; remote-endpoint = <&mcpu22_ep_0>; }; /* (5) connected CPU */ + mcodec22_ep_1: endpoint@2 { reg = <2>; remote-endpoint = <&mcpu23_ep_0>; }; /* (5) connected CPU */ + }; + }; }; dpcm { @@ -440,6 +624,14 @@ /* [Semi-Multi] */ sm0: port@7 { reg = <7>; cpu7_ep: endpoint { remote-endpoint = <&smcodec0_ep>; }; }; + + /* [Multi-CPU-1] */ + port@8 { reg = <8>; cpu8_ep: endpoint { remote-endpoint = <&mcpu11_ep>; }; }; + port@9 { reg = <9>; cpu9_ep: endpoint { remote-endpoint = <&mcpu12_ep>; }; }; + /* [Multi-CPU-2] */ + port@a { reg = <10>; cpu10_ep: endpoint { remote-endpoint = <&mcpu21_ep>; }; }; + port@b { reg = <11>; cpu11_ep: endpoint { remote-endpoint = <&mcpu22_ep>; }; }; + port@c { reg = <12>; cpu12_ep: endpoint { remote-endpoint = <&mcpu23_ep>; }; }; }; }; @@ -498,6 +690,13 @@ port@c { reg = <12>; codec12_ep: endpoint { remote-endpoint = <&smcodec1_ep>; }; }; port@d { reg = <13>; codec13_ep: endpoint { remote-endpoint = <&smcodec2_ep>; }; }; + /* [Multi-Codec-1] */ + port@e { reg = <14>; codec14_ep: endpoint { remote-endpoint = <&mcodec11_ep>; }; }; + port@f { reg = <15>; codec15_ep: endpoint { remote-endpoint = <&mcodec12_ep>; }; }; + port@10 { reg = <16>; codec16_ep: endpoint { remote-endpoint = <&mcodec13_ep>; }; }; + /* [Multi-Codec-2] */ + port@11 { reg = <17>; codec17_ep: endpoint { remote-endpoint = <&mcodec21_ep>; }; }; + port@12 { reg = <18>; codec18_ep: endpoint { remote-endpoint = <&mcodec22_ep>; }; }; }; }; }; From 076357cd57c294fb185ac452b9ce5536b2853839 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sun, 19 Nov 2023 19:19:14 +0100 Subject: [PATCH 069/337] ASoC: sh: convert not to use dma_request_slave_channel() dma_request_slave_channel() is deprecated. dma_request_chan() should be used directly instead. Switch to the preferred function and update the error handling accordingly. Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/1a837f96f056fa3dcb02a77afa5892d40b354cb1.1700417934.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/sh/fsi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 2ef47aa2c778..84601ba43b7d 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1379,7 +1379,9 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct dev io->chan = dma_request_channel(mask, shdma_chan_filter, (void *)io->dma_id); #else - io->chan = dma_request_slave_channel(dev, is_play ? "tx" : "rx"); + io->chan = dma_request_chan(dev, is_play ? "tx" : "rx"); + if (IS_ERR(io->chan)) + io->chan = NULL; #endif if (io->chan) { struct dma_slave_config cfg = {}; From d5070d0c10326e09276c34568b9a19fb9a727b6e Mon Sep 17 00:00:00 2001 From: Yong Zhi Date: Mon, 27 Nov 2023 12:52:35 +0200 Subject: [PATCH 070/337] ASoC: SOF: Intel: mtl: call dsp dump when boot retry fails MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Call snd_sof_dsp_dbg_dump() with the same flags/dump_msg as used in function hda_loader.c/cl_dsp_init(). Reviewed-by: Péter Ujfalusi Signed-off-by: Yong Zhi Signed-off-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127105235.30071-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 254dbbeac1d0..3ef9e5c37028 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -440,7 +440,8 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; const struct sof_intel_dsp_desc *chip = hda->desc; unsigned int status; - u32 ipc_hdr; + u32 ipc_hdr, flags; + char *dump_msg; int ret; /* step 1: purge FW request */ @@ -493,8 +494,18 @@ int mtl_dsp_cl_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) return 0; err: - snd_sof_dsp_dbg_dump(sdev, "MTL DSP init fail", 0); + flags = SOF_DBG_DUMP_PCI | SOF_DBG_DUMP_MBOX | SOF_DBG_DUMP_OPTIONAL; + + /* after max boot attempts make sure that the dump is printed */ + if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) + flags &= ~SOF_DBG_DUMP_OPTIONAL; + + dump_msg = kasprintf(GFP_KERNEL, "Boot iteration failed: %d/%d", + hda->boot_iteration, HDA_FW_BOOT_ATTEMPTS); + snd_sof_dsp_dbg_dump(sdev, dump_msg, flags); mtl_dsp_core_power_down(sdev, SOF_DSP_PRIMARY_CORE); + + kfree(dump_msg); return ret; } From 9b3cd8ebb19edd92300932b68fd6941eaf1852d0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 27 Nov 2023 12:43:13 +0200 Subject: [PATCH 071/337] ASoC: SOF: Intel: Use existing helpers to change GPROCEN and PIE bits Instead of directly changing the GPROCEN/PIE bits in PPCTL we should use the existing helper hda_dsp_ctrl_ppcap_enable() and hda_dsp_ctrl_ppcap_int_enable() helpers for clarity. Reviewed-by: Ranjani Sridharan Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231127104313.16661-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 5 ++--- sound/soc/sof/intel/hda.c | 6 ++---- 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 46fb2d1425e9..1805cf754beb 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -510,9 +510,8 @@ cleanup: return chip_info->init_core_mask; /* disable DSP */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, - SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_GPROCEN, 0); + hda_dsp_ctrl_ppcap_enable(sdev, false); + return ret; } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 744c0dd5766d..fe4ae349dad5 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1350,8 +1350,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) if (!sdev->dspless_mode_selected) { /* disable DSP IRQ */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_PIE, 0); + hda_dsp_ctrl_ppcap_int_enable(sdev, false); } /* disable CIE and GIE interrupts */ @@ -1366,8 +1365,7 @@ void hda_dsp_remove(struct snd_sof_dev *sdev) chip->power_down_dsp(sdev); /* disable DSP */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - SOF_HDA_PPCTL_GPROCEN, 0); + hda_dsp_ctrl_ppcap_enable(sdev, false); skip_disable_dsp: free_irq(sdev->ipc_irq, sdev); From 5c0e047ab629bcb5efa94de63fcdc75c9fe69516 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 27 Nov 2023 15:34:42 +0200 Subject: [PATCH 072/337] ASoC: Intel: sof_sdw: Make use of dev_err_probe() The devm_snd_soc_register_card() can return with -EPROBE_DEFER and in that case the driver should not print an error message. Closes: https://github.com/thesofproject/linux/issues/4668 Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Chao Song Link: https://lore.kernel.org/r/20231127133448.18449-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3312ad8a563b..199490f88653 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1946,7 +1946,7 @@ static int mc_probe(struct platform_device *pdev) /* Register the card */ ret = devm_snd_soc_register_card(card->dev, card); if (ret) { - dev_err(card->dev, "snd_soc_register_card failed %d\n", ret); + dev_err_probe(card->dev, ret, "snd_soc_register_card failed %d\n", ret); mc_dailink_exit_loop(card); return ret; } From fd8ff49d35f917c65383642c8f8f20656734f3ad Mon Sep 17 00:00:00 2001 From: Chao Song Date: Mon, 27 Nov 2023 15:34:43 +0200 Subject: [PATCH 073/337] ASoC: Intel: sof_sdw: remove unused function declaration MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The functions sof_sdw_rt712_sdca_init() and sof_sdw_rt712_sdca_exit() declared in header file are never implemented and used, remove them. Signed-off-by: Chao Song Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127133448.18449-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_common.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index e6b98523b4e7..9528f147b719 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -138,12 +138,6 @@ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); /* RT712-SDCA support */ -int sof_sdw_rt712_sdca_init(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *link, - struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, - bool playback); -int sof_sdw_rt712_sdca_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); int sof_sdw_rt712_spk_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, From def127feaa8a9d8684ac41139638b079e6e637e2 Mon Sep 17 00:00:00 2001 From: Chao Song Date: Mon, 27 Nov 2023 15:34:44 +0200 Subject: [PATCH 074/337] ASoC: Intel: sof_sdw: Add rt722 support RT722 is a multi-function codec which supports headset, amp, and dmic functions. Each function provides a DAI which can be used in different dailinks. The RT711 supports up to 3 SoundWire lanes, but that should not have any impact in the machine driver: the lanes are allocated and controlled by the manager and bandwidth allocation algorithm. Signed-off-by: Chao Song Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127133448.18449-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 3 +- sound/soc/intel/boards/sof_sdw.c | 30 ++++++ sound/soc/intel/boards/sof_sdw_common.h | 12 +++ sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 97 +++++++++++++++++++ .../boards/sof_sdw_rt_sdca_jack_common.c | 8 ++ 6 files changed, 150 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/boards/sof_sdw_rt722_sdca.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 9e427f00deac..99ebe48216ea 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -686,6 +686,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_RT712_SDCA_DMIC_SDW select SND_SOC_RT715_SDW select SND_SOC_RT715_SDCA_SDW + select SND_SOC_RT722_SDCA_SDW select SND_SOC_RT1308_SDW select SND_SOC_RT1308 select SND_SOC_RT1316_SDW diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 943bf8b80e01..bbf796a5f7ba 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -41,9 +41,10 @@ snd-soc-sof-sdw-objs += sof_sdw.o \ sof_sdw_rt5682.o sof_sdw_rt700.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ sof_sdw_rt712_sdca.o sof_sdw_rt715.o \ - sof_sdw_rt715_sdca.o sof_sdw_dmic.o \ + sof_sdw_rt715_sdca.o sof_sdw_rt722_sdca.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ + sof_sdw_dmic.o \ sof_sdw_hdmi.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 199490f88653..77a9ce401103 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -860,6 +860,36 @@ static struct sof_sdw_codec_info codec_info_list[] = { }, .dai_num = 1, }, + { + .part_id = 0x722, + .version_id = 3, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt722-sdca-aif1", + .dai_type = SOF_SDW_DAI_TYPE_JACK, + .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .init = sof_sdw_rt_sdca_jack_init, + .exit = sof_sdw_rt_sdca_jack_exit, + }, + { + .direction = {true, false}, + .dai_name = "rt722-sdca-aif2", + .dai_type = SOF_SDW_DAI_TYPE_AMP, + /* No feedback capability is provided by rt722-sdca codec driver*/ + .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .init = sof_sdw_rt722_spk_init, + }, + { + .direction = {false, true}, + .dai_name = "rt722-sdca-aif3", + .dai_type = SOF_SDW_DAI_TYPE_MIC, + .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .init = sof_sdw_rt722_sdca_dmic_init, + }, + }, + .dai_num = 3, + }, { .part_id = 0x8373, .dais = { diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 9528f147b719..f16456945edb 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -183,6 +183,18 @@ int sof_sdw_rt715_sdca_init(struct snd_soc_card *card, struct sof_sdw_codec_info *info, bool playback); +/* RT722-SDCA support */ +int sof_sdw_rt722_spk_init(struct snd_soc_card *card, + const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); +int sof_sdw_rt722_sdca_dmic_init(struct snd_soc_card *card, + const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback); + /* MAXIM codec support */ int sof_sdw_maxim_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c new file mode 100644 index 000000000000..fe3a2bff95bc --- /dev/null +++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c @@ -0,0 +1,97 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2023 Intel Corporation + +/* + * sof_sdw_rt722_sdca - Helpers to handle RT722-SDCA from generic machine driver + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "sof_sdw_common.h" + +static const struct snd_soc_dapm_widget rt722_spk_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route rt722_spk_map[] = { + { "Speaker", NULL, "rt722 SPK" }, +}; + +static const struct snd_kcontrol_new rt722_spk_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int rt722_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s spk:rt722", + card->components); + if (!card->components) + return -ENOMEM; + + ret = snd_soc_add_card_controls(card, rt722_spk_controls, + ARRAY_SIZE(rt722_spk_controls)); + if (ret) { + dev_err(card->dev, "failed to add rt722 spk controls: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_new_controls(&card->dapm, rt722_spk_widgets, + ARRAY_SIZE(rt722_spk_widgets)); + if (ret) { + dev_err(card->dev, "failed to add rt722 spk widgets: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, rt722_spk_map, ARRAY_SIZE(rt722_spk_map)); + if (ret) + dev_err(rtd->dev, "failed to add rt722 spk map: %d\n", ret); + + return ret; +} + +int sof_sdw_rt722_spk_init(struct snd_soc_card *card, + const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + dai_links->init = rt722_spk_init; + + return 0; +} + +static int rt722_sdca_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); + struct snd_soc_component *component = codec_dai->component; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + "%s mic:%s", + card->components, component->name_prefix); + if (!card->components) + return -ENOMEM; + + return 0; +} + +int sof_sdw_rt722_sdca_dmic_init(struct snd_soc_card *card, + const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + dai_links->init = rt722_sdca_dmic_rtd_init; + + return 0; +} diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 65bbcee88d6d..e430be7681d2 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -63,6 +63,11 @@ static const struct snd_soc_dapm_route rt713_sdca_map[] = { { "rt713 MIC2", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt722_sdca_map[] = { + { "Headphone", NULL, "rt722 HP" }, + { "rt722 MIC2", NULL, "Headset Mic" }, +}; + static const struct snd_kcontrol_new rt_sdca_jack_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -117,6 +122,9 @@ static int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd) } else if (strstr(component->name_prefix, "rt713")) { ret = snd_soc_dapm_add_routes(&card->dapm, rt713_sdca_map, ARRAY_SIZE(rt713_sdca_map)); + } else if (strstr(component->name_prefix, "rt722")) { + ret = snd_soc_dapm_add_routes(&card->dapm, rt722_sdca_map, + ARRAY_SIZE(rt722_sdca_map)); } else { dev_err(card->dev, "%s is not supported\n", component->name_prefix); return -EINVAL; From 817178e7674bd8ca35344b2212a3105ed75559e5 Mon Sep 17 00:00:00 2001 From: Mac Chiang Date: Mon, 27 Nov 2023 15:34:45 +0200 Subject: [PATCH 075/337] ASoC: Intel: soc-acpi: rt713+rt1316, no sdw-dmic config This is additional HW board: rt713+rt1316 without rt713-dmic configuration: SDW0: rt713 audio jack SDW1: rt1316 spk_amp_l SDW2: rt1316 spk_amp_r Signed-off-by: Mac Chiang Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127133448.18449-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 25 +++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 301b8142d554..61f2b2645496 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -422,6 +422,25 @@ static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12_rt1713_l3[] = {} }; +static const struct snd_soc_acpi_link_adr mtl_rt713_l0_rt1316_l12[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt713_0_single_adr), + .adr_d = rt713_0_single_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt1316_1_group2_adr), + .adr_d = rt1316_1_group2_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1316_2_group2_adr), + .adr_d = rt1316_2_group2_adr, + }, + {} +}; + static const struct snd_soc_acpi_adr_device mx8363_2_adr[] = { { .adr = 0x000230019F836300ull, @@ -507,6 +526,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt713-l0-rt1316-l12-rt1713-l3.tplg", }, + { + .link_mask = GENMASK(2, 0), + .links = mtl_rt713_l0_rt1316_l12, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt713-l0-rt1316-l12.tplg", + }, { .link_mask = BIT(3) | BIT(0), .links = mtl_712_only, From faca26b6ca90b220cba787ff7c6a05e99528731c Mon Sep 17 00:00:00 2001 From: Chao Song Date: Mon, 27 Nov 2023 15:34:46 +0200 Subject: [PATCH 076/337] ASoC: Intel: soc-acpi: add Gen4.1 SDCA board support for LNL RVP This patch adds support for LNL RVP with Realtek Gen4.1 SDCA codec board, the codec layout is: SDW0: RT711 Headphone SDW1: RT714 DMIC SDW2: RT1316 Speaker SDW3: RT1316 Speaker Signed-off-by: Chao Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127133448.18449-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 71 +++++++++++++++++++ 1 file changed, 71 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index 9f35b77deb11..5897bb6b28b8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -22,6 +22,20 @@ static const struct snd_soc_acpi_endpoint single_endpoint = { .group_id = 0, }; +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, +}; + static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { { .adr = 0x000030025D071101ull, @@ -31,6 +45,33 @@ static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt1316_2_group1_adr[] = { + { + .adr = 0x000230025D131601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "rt1316-1" + } +}; + +static const struct snd_soc_acpi_adr_device rt1316_3_group1_adr[] = { + { + .adr = 0x000331025D131601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "rt1316-2" + } +}; + +static const struct snd_soc_acpi_adr_device rt714_1_adr[] = { + { + .adr = 0x000130025D071401ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt714" + } +}; + static const struct snd_soc_acpi_link_adr lnl_rvp[] = { { .mask = BIT(0), @@ -40,6 +81,30 @@ static const struct snd_soc_acpi_link_adr lnl_rvp[] = { {} }; +static const struct snd_soc_acpi_link_adr lnl_3_in_1_sdca[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(rt1316_2_group1_adr), + .adr_d = rt1316_2_group1_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt1316_3_group1_adr), + .adr_d = rt1316_3_group1_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt714_1_adr), + .adr_d = rt714_1_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { /* mockup tests need to be first */ @@ -61,6 +126,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt715-rt711-rt1308-mono.tplg", }, + { + .link_mask = GENMASK(3, 0), + .links = lnl_3_in_1_sdca, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-rt711-l0-rt1316-l23-rt714-l1.tplg", + }, { .link_mask = BIT(0), .links = lnl_rvp, From 2112aa034907c428785e1a5730927181276ee45b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 17 Nov 2023 13:05:55 +0100 Subject: [PATCH 077/337] ALSA: pcm: Introduce MSBITS subformat interface Improve granularity of format selection for S32/U32 formats by adding constants representing 20, 24 and MAX most significant bits. The MAX means the maximum number of significant bits which can the physical format hold. For 32-bit formats, MAX is related to 32 bits. For 8-bit formats, MAX is related to 8 bits etc. As there is only one user currently (format S32_LE), subformat is represented by a simple u32 and stores flags only for that one user alone. The approach of subformat being part of struct snd_pcm_hardware is a compromise between ALSA and ASoC allowing for hw_params-intersection code to be alloc/free-less while not adding any new responsibilities to ASoC runtime structures. Acked-by: Mark Brown Signed-off-by: Jaroslav Kysela Co-developed-by: Cezary Rojewski Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20231117120610.1755254-2-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 7 ++++ include/uapi/sound/asound.h | 7 ++-- sound/core/pcm.c | 3 ++ sound/core/pcm_native.c | 55 +++++++++++++++++++++++++++++-- tools/include/uapi/sound/asound.h | 7 ++-- 5 files changed, 73 insertions(+), 6 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2a815373dac1..cc175c623dac 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -32,6 +32,7 @@ struct snd_pcm_hardware { unsigned int info; /* SNDRV_PCM_INFO_* */ u64 formats; /* SNDRV_PCM_FMTBIT_* */ + u32 subformats; /* for S32_LE, SNDRV_PCM_SUBFMTBIT_* */ unsigned int rates; /* SNDRV_PCM_RATE_* */ unsigned int rate_min; /* min rate */ unsigned int rate_max; /* max rate */ @@ -217,6 +218,12 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE #endif +#define _SNDRV_PCM_SUBFMTBIT(fmt) BIT((__force int)SNDRV_PCM_SUBFORMAT_##fmt) +#define SNDRV_PCM_SUBFMTBIT_STD _SNDRV_PCM_SUBFMTBIT(STD) +#define SNDRV_PCM_SUBFMTBIT_MSBITS_MAX _SNDRV_PCM_SUBFMTBIT(MSBITS_MAX) +#define SNDRV_PCM_SUBFMTBIT_MSBITS_20 _SNDRV_PCM_SUBFMTBIT(MSBITS_20) +#define SNDRV_PCM_SUBFMTBIT_MSBITS_24 _SNDRV_PCM_SUBFMTBIT(MSBITS_24) + struct snd_pcm_file { struct snd_pcm_substream *substream; int no_compat_mmap; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index f9939da41122..d5b9cfbd9cea 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -142,7 +142,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 15) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 16) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,7 +267,10 @@ typedef int __bitwise snd_pcm_format_t; typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_SUBFORMAT_STD ((__force snd_pcm_subformat_t) 0) -#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_STD +#define SNDRV_PCM_SUBFORMAT_MSBITS_MAX ((__force snd_pcm_subformat_t) 1) +#define SNDRV_PCM_SUBFORMAT_MSBITS_20 ((__force snd_pcm_subformat_t) 2) +#define SNDRV_PCM_SUBFORMAT_MSBITS_24 ((__force snd_pcm_subformat_t) 3) +#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_MSBITS_24 #define SNDRV_PCM_INFO_MMAP 0x00000001 /* hardware supports mmap */ #define SNDRV_PCM_INFO_MMAP_VALID 0x00000002 /* period data are valid during transfer */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 20bb2d7c8d4b..c4bc15f048b6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -265,6 +265,9 @@ static const char * const snd_pcm_access_names[] = { static const char * const snd_pcm_subformat_names[] = { SUBFORMAT(STD), + SUBFORMAT(MSBITS_MAX), + SUBFORMAT(MSBITS_20), + SUBFORMAT(MSBITS_24), }; static const char * const snd_pcm_tstamp_mode_names[] = { diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f610b08f5a2b..f5ff00f99788 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -479,6 +479,7 @@ static int fixup_unreferenced_params(struct snd_pcm_substream *substream, { const struct snd_interval *i; const struct snd_mask *m; + struct snd_mask *m_rw; int err; if (!params->msbits) { @@ -487,6 +488,22 @@ static int fixup_unreferenced_params(struct snd_pcm_substream *substream, params->msbits = snd_interval_value(i); } + if (params->msbits) { + m = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); + if (snd_mask_single(m)) { + snd_pcm_format_t format = (__force snd_pcm_format_t)snd_mask_min(m); + + if (snd_pcm_format_linear(format) && + snd_pcm_format_width(format) != params->msbits) { + m_rw = hw_param_mask(params, SNDRV_PCM_HW_PARAM_SUBFORMAT); + snd_mask_reset(m_rw, + (__force unsigned)SNDRV_PCM_SUBFORMAT_MSBITS_MAX); + if (snd_mask_empty(m_rw)) + return -EINVAL; + } + } + } + if (!params->rate_den) { i = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE); if (snd_interval_single(i)) { @@ -2483,6 +2500,41 @@ static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } +static int snd_pcm_hw_rule_subformats(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_mask *sfmask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_SUBFORMAT); + struct snd_mask *fmask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + u32 *subformats = rule->private; + snd_pcm_format_t f; + struct snd_mask m; + + snd_mask_none(&m); + /* All PCMs support at least the default STD subformat. */ + snd_mask_set(&m, (__force unsigned)SNDRV_PCM_SUBFORMAT_STD); + + pcm_for_each_format(f) { + if (!snd_mask_test(fmask, (__force unsigned)f)) + continue; + + if (f == SNDRV_PCM_FORMAT_S32_LE && *subformats) + m.bits[0] |= *subformats; + else if (snd_pcm_format_linear(f)) + snd_mask_set(&m, (__force unsigned)SNDRV_PCM_SUBFORMAT_MSBITS_MAX); + } + + return snd_mask_refine(sfmask, &m); +} + +static int snd_pcm_hw_constraint_subformats(struct snd_pcm_runtime *runtime, + unsigned int cond, u32 *subformats) +{ + return snd_pcm_hw_rule_add(runtime, cond, -1, + snd_pcm_hw_rule_subformats, (void *)subformats, + SNDRV_PCM_HW_PARAM_SUBFORMAT, + SNDRV_PCM_HW_PARAM_FORMAT, -1); +} + static int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -2634,8 +2686,7 @@ static int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; - err = snd_pcm_hw_constraint_mask(runtime, SNDRV_PCM_HW_PARAM_SUBFORMAT, - PARAM_MASK_BIT(SNDRV_PCM_SUBFORMAT_STD)); + err = snd_pcm_hw_constraint_subformats(runtime, 0, &hw->subformats); if (err < 0) return err; diff --git a/tools/include/uapi/sound/asound.h b/tools/include/uapi/sound/asound.h index f9939da41122..d5b9cfbd9cea 100644 --- a/tools/include/uapi/sound/asound.h +++ b/tools/include/uapi/sound/asound.h @@ -142,7 +142,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 15) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 16) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,7 +267,10 @@ typedef int __bitwise snd_pcm_format_t; typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_SUBFORMAT_STD ((__force snd_pcm_subformat_t) 0) -#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_STD +#define SNDRV_PCM_SUBFORMAT_MSBITS_MAX ((__force snd_pcm_subformat_t) 1) +#define SNDRV_PCM_SUBFORMAT_MSBITS_20 ((__force snd_pcm_subformat_t) 2) +#define SNDRV_PCM_SUBFORMAT_MSBITS_24 ((__force snd_pcm_subformat_t) 3) +#define SNDRV_PCM_SUBFORMAT_LAST SNDRV_PCM_SUBFORMAT_MSBITS_24 #define SNDRV_PCM_INFO_MMAP 0x00000001 /* hardware supports mmap */ #define SNDRV_PCM_INFO_MMAP_VALID 0x00000002 /* period data are valid during transfer */ From a7fc8b862fd5952be791a870ef8ef56016e83ff4 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:05:56 +0100 Subject: [PATCH 078/337] ALSA: hda: Honor subformat when querying PCMs Update mechanism for querying supported PCMs to allow for granular format selection when container size is 32 bits. Currently always the highest bit depth is selected, regardless of how many actual formats codec in question supports. Acked-by: Mark Brown Co-developed-by: Jaroslav Kysela Signed-off-by: Jaroslav Kysela Signed-off-by: Cezary Rojewski Link: https://lore.kernel.org/r/20231117120610.1755254-3-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/hda_codec.h | 5 ++-- include/sound/hdaudio.h | 3 ++- sound/hda/hdac_device.c | 47 ++++++++++++++++++++---------------- sound/pci/hda/hda_codec.c | 2 ++ sound/pci/hda/patch_hdmi.c | 1 + sound/soc/codecs/hdac_hdmi.c | 3 ++- 6 files changed, 36 insertions(+), 25 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 5497dc9c396a..9c94ba7c183d 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -141,6 +141,7 @@ struct hda_pcm_stream { hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */ u32 rates; /* supported rates */ u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */ + u32 subformats; /* for S32_LE format, SNDRV_PCM_SUBFMTBIT_* */ unsigned int maxbps; /* supported max. bit per sample */ const struct snd_pcm_chmap_elem *chmap; /* chmap to override */ struct hda_pcm_ops ops; @@ -448,8 +449,8 @@ void __snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid, #define snd_hda_codec_cleanup_stream(codec, nid) \ __snd_hda_codec_cleanup_stream(codec, nid, 0) -#define snd_hda_query_supported_pcm(codec, nid, ratesp, fmtsp, bpsp) \ - snd_hdac_query_supported_pcm(&(codec)->core, nid, ratesp, fmtsp, bpsp) +#define snd_hda_query_supported_pcm(codec, nid, ratesp, fmtsp, subfmtp, bpsp) \ + snd_hdac_query_supported_pcm(&(codec)->core, nid, ratesp, fmtsp, subfmtp, bpsp) #define snd_hda_is_supported_format(codec, nid, fmt) \ snd_hdac_is_supported_format(&(codec)->core, nid, fmt) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index dd7c87bbc613..7cfea9e7f6e4 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -146,7 +146,8 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, unsigned int maxbps, unsigned short spdif_ctls); int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, - u32 *ratesp, u64 *formatsp, unsigned int *bpsp); + u32 *ratesp, u64 *formatsp, u32 *subformatsp, + unsigned int *bpsp); bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid, unsigned int format); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index bbf7bcdb449a..cde4d5c33771 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -817,15 +817,17 @@ static unsigned int query_stream_param(struct hdac_device *codec, hda_nid_t nid) * @nid: NID to query * @ratesp: the pointer to store the detected rate bitflags * @formatsp: the pointer to store the detected formats + * @subformatsp: the pointer to store the detected subformats for S32_LE format * @bpsp: the pointer to store the detected format widths * - * Queries the supported PCM rates and formats. The NULL @ratesp, @formatsp - * or @bsps argument is ignored. + * Queries the supported PCM rates and formats. The NULL @ratesp, @formatsp, + * @subformatsp or @bpsp argument is ignored. * * Returns 0 if successful, otherwise a negative error code. */ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, - u32 *ratesp, u64 *formatsp, unsigned int *bpsp) + u32 *ratesp, u64 *formatsp, u32 *subformatsp, + unsigned int *bpsp) { unsigned int i, val, wcaps; @@ -848,9 +850,10 @@ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, *ratesp = rates; } - if (formatsp || bpsp) { - u64 formats = 0; + if (formatsp || subformatsp || bpsp) { unsigned int streams, bps; + u32 subformats = 0; + u64 formats = 0; streams = query_stream_param(codec, nid); if (!streams) @@ -866,24 +869,24 @@ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, formats |= SNDRV_PCM_FMTBIT_S16_LE; bps = 16; } - if (wcaps & AC_WCAP_DIGITAL) { - if (val & AC_SUPPCM_BITS_32) - formats |= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE; - if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24)) - formats |= SNDRV_PCM_FMTBIT_S32_LE; - if (val & AC_SUPPCM_BITS_24) - bps = 24; - else if (val & AC_SUPPCM_BITS_20) - bps = 20; - } else if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24| - AC_SUPPCM_BITS_32)) { + if (val & AC_SUPPCM_BITS_20) { formats |= SNDRV_PCM_FMTBIT_S32_LE; - if (val & AC_SUPPCM_BITS_32) + subformats |= SNDRV_PCM_SUBFMTBIT_MSBITS_20; + bps = 20; + } + if (val & AC_SUPPCM_BITS_24) { + formats |= SNDRV_PCM_FMTBIT_S32_LE; + subformats |= SNDRV_PCM_SUBFMTBIT_MSBITS_24; + bps = 24; + } + if (val & AC_SUPPCM_BITS_32) { + if (wcaps & AC_WCAP_DIGITAL) { + formats |= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE; + } else { + formats |= SNDRV_PCM_FMTBIT_S32_LE; + subformats |= SNDRV_PCM_SUBFMTBIT_MSBITS_MAX; bps = 32; - else if (val & AC_SUPPCM_BITS_24) - bps = 24; - else if (val & AC_SUPPCM_BITS_20) - bps = 20; + } } } #if 0 /* FIXME: CS4206 doesn't work, which is the only codec supporting float */ @@ -911,6 +914,8 @@ int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, } if (formatsp) *formatsp = formats; + if (subformatsp) + *subformatsp = subformats; if (bpsp) *bpsp = bps; } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 01718b1fc9a7..12f02cdc9659 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3163,6 +3163,7 @@ static int set_pcm_default_values(struct hda_codec *codec, err = snd_hda_query_supported_pcm(codec, info->nid, info->rates ? NULL : &info->rates, info->formats ? NULL : &info->formats, + info->subformats ? NULL : &info->subformats, info->maxbps ? NULL : &info->maxbps); if (err < 0) return err; @@ -3757,6 +3758,7 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, snd_hda_query_supported_pcm(codec, mout->dig_out_nid, &mout->spdif_rates, &mout->spdif_formats, + NULL, &mout->spdif_maxbps); } mutex_lock(&codec->spdif_mutex); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1cde2a69bdb4..687b8b8fd7ac 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1977,6 +1977,7 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) err = snd_hda_query_supported_pcm(codec, cvt_nid, &per_cvt->rates, &per_cvt->formats, + NULL, &per_cvt->maxbps); if (err < 0) return err; diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index b9c5ffbfb5ba..3e4f632d8665 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -670,6 +670,7 @@ hdac_hdmi_query_cvt_params(struct hdac_device *hdev, struct hdac_hdmi_cvt *cvt) err = snd_hdac_query_supported_pcm(hdev, cvt->nid, &cvt->params.rates, &cvt->params.formats, + NULL, &cvt->params.maxbps); if (err < 0) dev_err(&hdev->dev, @@ -1577,7 +1578,7 @@ static int hdac_hdmi_create_dais(struct hdac_device *hdev, list_for_each_entry(cvt, &hdmi->cvt_list, head) { ret = snd_hdac_query_supported_pcm(hdev, cvt->nid, - &rates, &formats, &bps); + &rates, &formats, NULL, &bps); if (ret) return ret; From 4a6ba09e892aab49fda8464e4f9b244a17fd75b2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:05:57 +0100 Subject: [PATCH 079/337] ASoC: pcm: Honor subformat when configuring runtime Subformat options are ignored when setting up hardware parameters and assigning PCM stream capabilities. Account for them to allow for granular format selection. As there is only one user currently (format S32_LE), subformat is represented by a simple u32 and stores flags only for that one user alone. Such approach allows for alloc/free-less code until there are more users on the horizon. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-4-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 10 ++++++++++ 2 files changed, 11 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 7792c393e238..ccc31bc41e92 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -620,6 +620,7 @@ enum snd_soc_trigger_order { struct snd_soc_pcm_stream { const char *stream_name; u64 formats; /* SNDRV_PCM_FMTBIT_* */ + u32 subformats; /* for S32_LE format, SNDRV_PCM_SUBFMTBIT_* */ unsigned int rates; /* SNDRV_PCM_RATE_* */ unsigned int rate_min; /* min rate */ unsigned int rate_max; /* max rate */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 323e4d7b6adf..9d688917cce2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -554,6 +554,12 @@ static void soc_pcm_hw_update_format(struct snd_pcm_hardware *hw, hw->formats &= p->formats; } +static void soc_pcm_hw_update_subformat(struct snd_pcm_hardware *hw, + struct snd_soc_pcm_stream *p) +{ + hw->subformats &= p->subformats; +} + /** * snd_soc_runtime_calc_hw() - Calculate hw limits for a PCM stream * @rtd: ASoC PCM runtime @@ -592,6 +598,7 @@ int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, soc_pcm_hw_update_chan(hw, cpu_stream); soc_pcm_hw_update_rate(hw, cpu_stream); soc_pcm_hw_update_format(hw, cpu_stream); + soc_pcm_hw_update_subformat(hw, cpu_stream); } cpu_chan_min = hw->channels_min; cpu_chan_max = hw->channels_max; @@ -613,6 +620,7 @@ int snd_soc_runtime_calc_hw(struct snd_soc_pcm_runtime *rtd, soc_pcm_hw_update_chan(hw, codec_stream); soc_pcm_hw_update_rate(hw, codec_stream); soc_pcm_hw_update_format(hw, codec_stream); + soc_pcm_hw_update_subformat(hw, codec_stream); } /* Verify both a valid CPU DAI and a valid CODEC DAI were found */ @@ -1713,6 +1721,7 @@ static void dpcm_runtime_setup_fe(struct snd_pcm_substream *substream) soc_pcm_hw_update_rate(hw, cpu_stream); soc_pcm_hw_update_chan(hw, cpu_stream); soc_pcm_hw_update_format(hw, cpu_stream); + soc_pcm_hw_update_subformat(hw, cpu_stream); } } @@ -1750,6 +1759,7 @@ static void dpcm_runtime_setup_be_format(struct snd_pcm_substream *substream) codec_stream = snd_soc_dai_get_pcm_stream(dai, stream); soc_pcm_hw_update_format(hw, codec_stream); + soc_pcm_hw_update_subformat(hw, codec_stream); } } } From d24f1a090d3f2a5c86a8782afeda71c2d2539966 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:05:58 +0100 Subject: [PATCH 080/337] ALSA: hda: Upgrade stream-format infrastructure Introduce a set of functions that ultimately facilite SDxFMT-related calculations in atomic manner: First, introduce snd_pcm_subformat_width() and snd_pcm_hw_params_bits() helpers that separate the base functionality from the HDAudio-specific one. snd_hdac_format_normalize() - format converter. S20_LE, S24_LE and their unsigned and BE friends are invalid from HDAudio perspective but still can be specified as function argument due to compatibility reasons. snd_hdac_stream_format_bits() - obtain just the bits-per-sample value. Does not ignore subformat and msbits parameters. snd_hdac_stream_format() and snd_hdac_spdif_stream_format() - obtain the SDxFMT value given the audio format parameters. The former is stripped away of spdif-related information. Useful for users that do not care about them. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-5-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 5 ++ include/sound/pcm_params.h | 2 + sound/core/pcm_lib.c | 34 ++++++++++ sound/hda/hdac_device.c | 124 +++++++++++++++++++++++++++++++++++++ 4 files changed, 165 insertions(+) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 7cfea9e7f6e4..b00c631c4053 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -140,6 +140,11 @@ int snd_hdac_get_connections(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns); int snd_hdac_get_sub_nodes(struct hdac_device *codec, hda_nid_t nid, hda_nid_t *start_id); +unsigned int snd_hdac_stream_format_bits(snd_pcm_format_t format, snd_pcm_subformat_t subformat, + unsigned int maxbits); +unsigned int snd_hdac_stream_format(unsigned int channels, unsigned int bits, unsigned int rate); +unsigned int snd_hdac_spdif_stream_format(unsigned int channels, unsigned int bits, + unsigned int rate, unsigned short spdif_ctls); unsigned int snd_hdac_calc_stream_format(unsigned int rate, unsigned int channels, snd_pcm_format_t format, diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index ba184f49f7e1..fbf35df6e5cf 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -362,6 +362,8 @@ static inline int params_physical_width(const struct snd_pcm_hw_params *p) return snd_pcm_format_physical_width(params_format(p)); } +int snd_pcm_hw_params_bits(const struct snd_pcm_hw_params *p); + static inline void params_set_format(struct snd_pcm_hw_params *p, snd_pcm_format_t fmt) { diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a11cd7d6295f..41103e5c43ce 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1706,6 +1706,40 @@ int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, } EXPORT_SYMBOL(snd_pcm_hw_param_last); +/** + * snd_pcm_hw_params_bits - Get the number of bits per the sample. + * @p: hardware parameters + * + * Return: The number of bits per sample based on the format, + * subformat and msbits the specified hw params has. + */ +int snd_pcm_hw_params_bits(const struct snd_pcm_hw_params *p) +{ + snd_pcm_subformat_t subformat = params_subformat(p); + snd_pcm_format_t format = params_format(p); + + switch (format) { + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_U32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + case SNDRV_PCM_FORMAT_U32_BE: + switch (subformat) { + case SNDRV_PCM_SUBFORMAT_MSBITS_20: + return 20; + case SNDRV_PCM_SUBFORMAT_MSBITS_24: + return 24; + case SNDRV_PCM_SUBFORMAT_MSBITS_MAX: + case SNDRV_PCM_SUBFORMAT_STD: + default: + break; + } + fallthrough; + default: + return snd_pcm_format_width(format); + } +} +EXPORT_SYMBOL(snd_pcm_hw_params_bits); + static int snd_pcm_lib_ioctl_reset(struct snd_pcm_substream *substream, void *arg) { diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index cde4d5c33771..556bd24f4014 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -13,6 +13,7 @@ #include #include #include +#include #include "local.h" static void setup_fg_nodes(struct hdac_device *codec); @@ -725,6 +726,129 @@ static const struct hda_rate_tbl rate_bits[] = { { 0 } /* terminator */ }; +static snd_pcm_format_t snd_hdac_format_normalize(snd_pcm_format_t format) +{ + switch (format) { + case SNDRV_PCM_FORMAT_S20_LE: + case SNDRV_PCM_FORMAT_S24_LE: + return SNDRV_PCM_FORMAT_S32_LE; + + case SNDRV_PCM_FORMAT_U20_LE: + case SNDRV_PCM_FORMAT_U24_LE: + return SNDRV_PCM_FORMAT_U32_LE; + + case SNDRV_PCM_FORMAT_S20_BE: + case SNDRV_PCM_FORMAT_S24_BE: + return SNDRV_PCM_FORMAT_S32_BE; + + case SNDRV_PCM_FORMAT_U20_BE: + case SNDRV_PCM_FORMAT_U24_BE: + return SNDRV_PCM_FORMAT_U32_BE; + + default: + return format; + } +} + +/** + * snd_hdac_stream_format_bits - obtain bits per sample value. + * @format: the PCM format. + * @subformat: the PCM subformat. + * @maxbits: the maximum bits per sample. + * + * Return: The number of bits per sample. + */ +unsigned int snd_hdac_stream_format_bits(snd_pcm_format_t format, snd_pcm_subformat_t subformat, + unsigned int maxbits) +{ + struct snd_pcm_hw_params params; + unsigned int bits; + + memset(¶ms, 0, sizeof(params)); + + params_set_format(¶ms, snd_hdac_format_normalize(format)); + snd_mask_set(hw_param_mask(¶ms, SNDRV_PCM_HW_PARAM_SUBFORMAT), + (__force unsigned int)subformat); + + bits = snd_pcm_hw_params_bits(¶ms); + if (maxbits) + return min(bits, maxbits); + return bits; +} +EXPORT_SYMBOL_GPL(snd_hdac_stream_format_bits); + +/** + * snd_hdac_stream_format - convert format parameters to SDxFMT value. + * @channels: the number of channels. + * @bits: bits per sample. + * @rate: the sample rate. + * + * Return: The format bitset or zero if invalid. + */ +unsigned int snd_hdac_stream_format(unsigned int channels, unsigned int bits, unsigned int rate) +{ + unsigned int val = 0; + int i; + + for (i = 0; rate_bits[i].hz; i++) { + if (rate_bits[i].hz == rate) { + val = rate_bits[i].hda_fmt; + break; + } + } + + if (!rate_bits[i].hz) + return 0; + + if (channels == 0 || channels > 8) + return 0; + val |= channels - 1; + + switch (bits) { + case 8: + val |= AC_FMT_BITS_8; + break; + case 16: + val |= AC_FMT_BITS_16; + break; + case 20: + val |= AC_FMT_BITS_20; + break; + case 24: + val |= AC_FMT_BITS_24; + break; + case 32: + val |= AC_FMT_BITS_32; + break; + default: + return 0; + } + + return val; +} +EXPORT_SYMBOL_GPL(snd_hdac_stream_format); + +/** + * snd_hdac_spdif_stream_format - convert format parameters to SDxFMT value. + * @channels: the number of channels. + * @bits: bits per sample. + * @rate: the sample rate. + * @spdif_ctls: HD-audio SPDIF status bits (0 if irrelevant). + * + * Return: The format bitset or zero if invalid. + */ +unsigned int snd_hdac_spdif_stream_format(unsigned int channels, unsigned int bits, + unsigned int rate, unsigned short spdif_ctls) +{ + unsigned int val = snd_hdac_stream_format(channels, bits, rate); + + if (val && spdif_ctls & AC_DIG1_NONAUDIO) + val |= AC_FMT_TYPE_NON_PCM; + + return val; +} +EXPORT_SYMBOL_GPL(snd_hdac_spdif_stream_format); + /** * snd_hdac_calc_stream_format - calculate the format bitset * @rate: the sample rate From 61b52df4b64fa1c8860cdfbc548cd0bbe7310082 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:05:59 +0100 Subject: [PATCH 081/337] ALSA: hda: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-6-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_controller.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index c42e9ffff9db..3e7bfeee84fd 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -151,7 +151,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = to_hda_pcm_stream(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int format_val, stream_tag; + unsigned int format_val, stream_tag, bits; int err; struct hda_spdif_out *spdif = snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid); @@ -165,11 +165,9 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) } snd_hdac_stream_reset(azx_stream(azx_dev)); - format_val = snd_hdac_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hinfo->maxbps, - ctls); + bits = snd_hdac_stream_format_bits(runtime->format, SNDRV_PCM_SUBFORMAT_STD, hinfo->maxbps); + + format_val = snd_hdac_spdif_stream_format(runtime->channels, bits, runtime->rate, ctls); if (!format_val) { dev_err(chip->card->dev, "invalid format_val, rate=%d, ch=%d, format=%d\n", From 67ea58daab010ea7a622a89392cc9a6190f8b9c1 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:00 +0100 Subject: [PATCH 082/337] ALSA: hda/hdmi: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-7-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 687b8b8fd7ac..dff2d7221982 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1655,7 +1655,6 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, #define I915_SILENT_RATE 48000 #define I915_SILENT_CHANNELS 2 -#define I915_SILENT_FORMAT SNDRV_PCM_FORMAT_S16_LE #define I915_SILENT_FORMAT_BITS 16 #define I915_SILENT_FMT_MASK 0xf @@ -1668,8 +1667,8 @@ static void silent_stream_enable_i915(struct hda_codec *codec, per_pin->dev_id, I915_SILENT_RATE); /* trigger silent stream generation in hw */ - format = snd_hdac_calc_stream_format(I915_SILENT_RATE, I915_SILENT_CHANNELS, - I915_SILENT_FORMAT, I915_SILENT_FORMAT_BITS, 0); + format = snd_hdac_stream_format(I915_SILENT_CHANNELS, I915_SILENT_FORMAT_BITS, + I915_SILENT_RATE); snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, I915_SILENT_FMT_MASK, I915_SILENT_FMT_MASK, format); usleep_range(100, 200); From 0d41f0c07f19de7fb5790cbe462dddb68af60fa8 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:01 +0100 Subject: [PATCH 083/337] ALSA: hda/ca0132: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-8-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0132.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 748a3c40966e..aa312441604f 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3022,8 +3022,7 @@ static int dma_convert_to_hda_format(struct hda_codec *codec, { unsigned int format_val; - format_val = snd_hdac_calc_stream_format(sample_rate, - channels, SNDRV_PCM_FORMAT_S32_LE, 32, 0); + format_val = snd_hdac_stream_format(channels, 32, sample_rate); if (hda_format) *hda_format = (unsigned short)format_val; From d5c00ab2f508176e8d8bd976d8b2207c5f0a932b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:02 +0100 Subject: [PATCH 084/337] ASoC: codecs: hda: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. While at it, complete PCM stream initialization with subformat options. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-9-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/codecs/hda-dai.c | 7 +++++-- sound/soc/codecs/hda.c | 2 ++ 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hda-dai.c b/sound/soc/codecs/hda-dai.c index 5371ff086261..7bd7ddcd810f 100644 --- a/sound/soc/codecs/hda-dai.c +++ b/sound/soc/codecs/hda-dai.c @@ -76,13 +76,16 @@ static int hda_codec_dai_prepare(struct snd_pcm_substream *substream, struct snd struct hdac_stream *stream; struct hda_codec *codec; unsigned int format; + unsigned int bits; int ret; codec = dev_to_hda_codec(dai->dev); stream = substream->runtime->private_data; stream_info = snd_soc_dai_get_dma_data(dai, substream); - format = snd_hdac_calc_stream_format(runtime->rate, runtime->channels, runtime->format, - runtime->sample_bits, 0); + + bits = snd_hdac_stream_format_bits(runtime->format, runtime->subformat, + stream_info->maxbps); + format = snd_hdac_stream_format(runtime->channels, bits, runtime->rate); ret = snd_hda_codec_prepare(codec, stream_info, stream->stream_tag, format, substream); if (ret < 0) { diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index d57b043d6bfe..d2117e36ddd1 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -52,6 +52,7 @@ static int hda_codec_create_dais(struct hda_codec *codec, int pcm_count, stream->channels_max = pcm->stream[dir].channels_max; stream->rates = pcm->stream[dir].rates; stream->formats = pcm->stream[dir].formats; + stream->subformats = pcm->stream[dir].subformats; stream->sig_bits = pcm->stream[dir].maxbps; capture_dais: @@ -71,6 +72,7 @@ capture_dais: stream->channels_max = pcm->stream[dir].channels_max; stream->rates = pcm->stream[dir].rates; stream->formats = pcm->stream[dir].formats; + stream->subformats = pcm->stream[dir].subformats; stream->sig_bits = pcm->stream[dir].maxbps; } From 0bb0af123b0dbae3e3fad2166122777cf40abae4 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:03 +0100 Subject: [PATCH 085/337] ASoC: codecs: hdac_hda: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-10-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/codecs/hdac_hda.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 355f30779a34..0c589e46574d 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -215,18 +215,16 @@ static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, struct hdac_hda_priv *hda_pvt; unsigned int format_val; unsigned int maxbps; + unsigned int bits; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) maxbps = dai->driver->playback.sig_bits; else maxbps = dai->driver->capture.sig_bits; + bits = snd_hdac_stream_format_bits(params_format(params), SNDRV_PCM_SUBFORMAT_STD, maxbps); hda_pvt = snd_soc_component_get_drvdata(component); - format_val = snd_hdac_calc_stream_format(params_rate(params), - params_channels(params), - params_format(params), - maxbps, - 0); + format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params)); if (!format_val) { dev_err(dai->dev, "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", From cbc4ebb346162376377ee0a1074fc72d39cfbb51 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:04 +0100 Subject: [PATCH 086/337] ASoC: codecs: hdac_hdmi: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-11-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/codecs/hdac_hdmi.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3e4f632d8665..b3b11225d483 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -468,13 +468,14 @@ static int hdac_hdmi_set_hw_params(struct snd_pcm_substream *substream, struct hdac_hdmi_priv *hdmi = snd_soc_dai_get_drvdata(dai); struct hdac_hdmi_dai_port_map *dai_map; struct hdac_hdmi_pcm *pcm; + unsigned int bits; int format; dai_map = &hdmi->dai_map[dai->id]; - format = snd_hdac_calc_stream_format(params_rate(hparams), - params_channels(hparams), params_format(hparams), - dai->driver->playback.sig_bits, 0); + bits = snd_hdac_stream_format_bits(params_format(hparams), SNDRV_PCM_SUBFORMAT_STD, + dai->driver->playback.sig_bits); + format = snd_hdac_stream_format(params_channels(hparams), bits, params_rate(hparams)); pcm = hdac_hdmi_get_pcm_from_cvt(hdmi, dai_map->cvt); if (!pcm) From 71fd0fbaf61e6d1a0ae63dc1845a1d7a193cd32b Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:05 +0100 Subject: [PATCH 087/337] ASoC: Intel Skylake: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-12-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/intel/skylake/skl-pcm.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index d0c02e8a6785..4d52e3b35850 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -127,6 +127,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) unsigned int format_val; struct hdac_stream *hstream; struct hdac_ext_stream *stream; + unsigned int bits; int err; hstream = snd_hdac_get_stream(bus, params->stream, @@ -137,8 +138,9 @@ int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) stream = stream_to_hdac_ext_stream(hstream); snd_hdac_ext_stream_decouple(bus, stream, true); - format_val = snd_hdac_calc_stream_format(params->s_freq, - params->ch, params->format, params->host_bps, 0); + bits = snd_hdac_stream_format_bits(params->format, SNDRV_PCM_SUBFORMAT_STD, + params->host_bps); + format_val = snd_hdac_stream_format(params->ch, bits, params->s_freq); dev_dbg(dev, "format_val=%d, rate=%d, ch=%d, format=%d\n", format_val, params->s_freq, params->ch, params->format); @@ -165,6 +167,7 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) struct hdac_ext_stream *stream; struct hdac_ext_link *link; unsigned char stream_tag; + unsigned int bits; hstream = snd_hdac_get_stream(bus, params->stream, params->link_dma_id + 1); @@ -173,8 +176,10 @@ int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) stream = stream_to_hdac_ext_stream(hstream); snd_hdac_ext_stream_decouple(bus, stream, true); - format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, - params->format, params->link_bps, 0); + + bits = snd_hdac_stream_format_bits(params->format, SNDRV_PCM_SUBFORMAT_STD, + params->link_bps); + format_val = snd_hdac_stream_format(params->ch, bits, params->s_freq); dev_dbg(dev, "format_val=%d, rate=%d, ch=%d, format=%d\n", format_val, params->s_freq, params->ch, params->format); From 176d13881137b064dc65e282e2c19821c11e60a2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:06 +0100 Subject: [PATCH 088/337] ASoC: SOF: Intel: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-13-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/sof/intel/hda-dai-ops.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 87935554b1e4..55ce75db23e5 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -208,14 +208,16 @@ static unsigned int hda_calc_stream_format(struct snd_sof_dev *sdev, struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); unsigned int link_bps; unsigned int format_val; + unsigned int bits; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) link_bps = codec_dai->driver->playback.sig_bits; else link_bps = codec_dai->driver->capture.sig_bits; - format_val = snd_hdac_calc_stream_format(params_rate(params), params_channels(params), - params_format(params), link_bps, 0); + bits = snd_hdac_stream_format_bits(params_format(params), SNDRV_PCM_SUBFORMAT_STD, + link_bps); + format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params)); dev_dbg(sdev->dev, "format_val=%#x, rate=%d, ch=%d, format=%d\n", format_val, params_rate(params), params_channels(params), params_format(params)); @@ -238,11 +240,11 @@ static unsigned int generic_calc_stream_format(struct snd_sof_dev *sdev, struct snd_pcm_hw_params *params) { unsigned int format_val; + unsigned int bits; - format_val = snd_hdac_calc_stream_format(params_rate(params), params_channels(params), - params_format(params), - params_physical_width(params), - 0); + bits = snd_hdac_stream_format_bits(params_format(params), SNDRV_PCM_SUBFORMAT_STD, + params_physical_width(params)); + format_val = snd_hdac_stream_format(params_channels(params), bits, params_rate(params)); dev_dbg(sdev->dev, "format_val=%#x, rate=%d, ch=%d, format=%d\n", format_val, params_rate(params), params_channels(params), params_format(params)); @@ -258,6 +260,7 @@ static unsigned int dmic_calc_stream_format(struct snd_sof_dev *sdev, snd_pcm_format_t format; unsigned int channels; unsigned int width; + unsigned int bits; channels = params_channels(params); format = params_format(params); @@ -269,10 +272,8 @@ static unsigned int dmic_calc_stream_format(struct snd_sof_dev *sdev, width = 32; } - format_val = snd_hdac_calc_stream_format(params_rate(params), channels, - format, - width, - 0); + bits = snd_hdac_stream_format_bits(format, SNDRV_PCM_SUBFORMAT_STD, width); + format_val = snd_hdac_stream_format(channels, bits, params_rate(params)); dev_dbg(sdev->dev, "format_val=%#x, rate=%d, ch=%d, format=%d\n", format_val, params_rate(params), channels, format); From 615d13cb4f3e499fb5c270a7db66917286f7d0ec Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:07 +0100 Subject: [PATCH 089/337] ASoC: Intel: avs: Switch to new stream-format interface To provide option for selecting different bit-per-sample than just the maximum one, use the new format calculation mechanism. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-14-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/intel/avs/loader.c | 4 ++-- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/avs/pcm.c | 19 ++++++++++++++----- sound/soc/intel/avs/probes.c | 3 +-- 4 files changed, 18 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/avs/loader.c b/sound/soc/intel/avs/loader.c index 65dd8f140fc1..e83ce6a35755 100644 --- a/sound/soc/intel/avs/loader.c +++ b/sound/soc/intel/avs/loader.c @@ -371,7 +371,7 @@ int avs_hda_load_basefw(struct avs_dev *adev, struct firmware *fw) hstream = hdac_stream(estream); /* code loading performed with default format */ - sdfmt = snd_hdac_calc_stream_format(48000, 1, SNDRV_PCM_FORMAT_S32_LE, 32, 0); + sdfmt = snd_hdac_stream_format(1, 32, 48000); ret = snd_hdac_dsp_prepare(hstream, sdfmt, fw->size, &dmab); if (ret < 0) goto release_stream; @@ -438,7 +438,7 @@ int avs_hda_load_library(struct avs_dev *adev, struct firmware *lib, u32 id) stream = hdac_stream(estream); /* code loading performed with default format */ - sdfmt = snd_hdac_calc_stream_format(48000, 1, SNDRV_PCM_FORMAT_S32_LE, 32, 0); + sdfmt = snd_hdac_stream_format(1, 32, 48000); ret = snd_hdac_dsp_prepare(stream, sdfmt, lib->size, &dmab); if (ret < 0) goto release_stream; diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index aa8b50b931c3..3aa16ee8d34c 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -87,7 +87,7 @@ static bool avs_test_hw_params(struct snd_pcm_hw_params *params, return (params_rate(params) == fmt->sampling_freq && params_channels(params) == fmt->num_channels && params_physical_width(params) == fmt->bit_depth && - params_width(params) == fmt->valid_bit_depth); + snd_pcm_hw_params_bits(params) == fmt->valid_bit_depth); } static struct avs_tplg_path * diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 463dbba18426..f586a804099a 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -335,20 +335,25 @@ static int avs_dai_hda_be_prepare(struct snd_pcm_substream *substream, struct sn { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_pcm_runtime *runtime = substream->runtime; - struct hdac_ext_stream *link_stream = runtime->private_data; + struct snd_soc_pcm_stream *stream_info; + struct hdac_ext_stream *link_stream; struct hdac_ext_link *link; struct hda_codec *codec; struct hdac_bus *bus; unsigned int format_val; + unsigned int bits; int ret; + link_stream = runtime->private_data; if (link_stream->link_prepared) return 0; codec = dev_to_hda_codec(snd_soc_rtd_to_codec(rtd, 0)->dev); bus = &codec->bus->core; - format_val = snd_hdac_calc_stream_format(runtime->rate, runtime->channels, runtime->format, - runtime->sample_bits, 0); + stream_info = snd_soc_dai_get_pcm_stream(dai, substream->stream); + bits = snd_hdac_stream_format_bits(runtime->format, runtime->subformat, + stream_info->sig_bits); + format_val = snd_hdac_stream_format(runtime->channels, bits, runtime->rate); snd_hdac_ext_stream_reset(link_stream); snd_hdac_ext_stream_setup(link_stream, format_val); @@ -600,10 +605,12 @@ static int avs_dai_fe_hw_free(struct snd_pcm_substream *substream, struct snd_so static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_stream *stream_info; struct avs_dma_data *data; struct avs_dev *adev = to_avs_dev(dai->dev); struct hdac_ext_stream *host_stream; unsigned int format_val; + unsigned int bits; int ret; data = snd_soc_dai_get_dma_data(dai, substream); @@ -614,8 +621,10 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so snd_hdac_stream_reset(hdac_stream(host_stream)); - format_val = snd_hdac_calc_stream_format(runtime->rate, runtime->channels, runtime->format, - runtime->sample_bits, 0); + stream_info = snd_soc_dai_get_pcm_stream(dai, substream->stream); + bits = snd_hdac_stream_format_bits(runtime->format, runtime->subformat, + stream_info->sig_bits); + format_val = snd_hdac_stream_format(runtime->channels, bits, runtime->rate); ret = snd_hdac_stream_set_params(hdac_stream(host_stream), format_val); if (ret < 0) diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index bdc6b30dc009..817e543036f2 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -140,8 +140,7 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream, bps = snd_pcm_format_physical_width(format); if (bps < 0) return bps; - format_val = snd_hdac_calc_stream_format(params->codec.sample_rate, params->codec.ch_out, - format, bps, 0); + format_val = snd_hdac_stream_format(params->codec.ch_out, bps, params->codec.sample_rate); ret = snd_hdac_stream_set_params(hdac_stream(host_stream), format_val); if (ret < 0) return ret; From dfd6ba6813dd0fdea54de58e7b80ca304ce3fca5 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:08 +0100 Subject: [PATCH 090/337] ALSA: hda: Drop snd_hdac_calc_stream_format() There are no users of the function. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-15-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- include/sound/hdaudio.h | 5 ---- sound/hda/hdac_device.c | 61 ----------------------------------------- 2 files changed, 66 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b00c631c4053..62a3cd154ff2 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -145,11 +145,6 @@ unsigned int snd_hdac_stream_format_bits(snd_pcm_format_t format, snd_pcm_subfor unsigned int snd_hdac_stream_format(unsigned int channels, unsigned int bits, unsigned int rate); unsigned int snd_hdac_spdif_stream_format(unsigned int channels, unsigned int bits, unsigned int rate, unsigned short spdif_ctls); -unsigned int snd_hdac_calc_stream_format(unsigned int rate, - unsigned int channels, - snd_pcm_format_t format, - unsigned int maxbps, - unsigned short spdif_ctls); int snd_hdac_query_supported_pcm(struct hdac_device *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, u32 *subformatsp, unsigned int *bpsp); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 556bd24f4014..7f7b67fe1b65 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -849,67 +849,6 @@ unsigned int snd_hdac_spdif_stream_format(unsigned int channels, unsigned int bi } EXPORT_SYMBOL_GPL(snd_hdac_spdif_stream_format); -/** - * snd_hdac_calc_stream_format - calculate the format bitset - * @rate: the sample rate - * @channels: the number of channels - * @format: the PCM format (SNDRV_PCM_FORMAT_XXX) - * @maxbps: the max. bps - * @spdif_ctls: HD-audio SPDIF status bits (0 if irrelevant) - * - * Calculate the format bitset from the given rate, channels and th PCM format. - * - * Return zero if invalid. - */ -unsigned int snd_hdac_calc_stream_format(unsigned int rate, - unsigned int channels, - snd_pcm_format_t format, - unsigned int maxbps, - unsigned short spdif_ctls) -{ - int i; - unsigned int val = 0; - - for (i = 0; rate_bits[i].hz; i++) - if (rate_bits[i].hz == rate) { - val = rate_bits[i].hda_fmt; - break; - } - if (!rate_bits[i].hz) - return 0; - - if (channels == 0 || channels > 8) - return 0; - val |= channels - 1; - - switch (snd_pcm_format_width(format)) { - case 8: - val |= AC_FMT_BITS_8; - break; - case 16: - val |= AC_FMT_BITS_16; - break; - case 20: - case 24: - case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; - break; - default: - return 0; - } - - if (spdif_ctls & AC_DIG1_NONAUDIO) - val |= AC_FMT_TYPE_NON_PCM; - - return val; -} -EXPORT_SYMBOL_GPL(snd_hdac_calc_stream_format); - static unsigned int query_pcm_param(struct hdac_device *codec, hda_nid_t nid) { unsigned int val = 0; From c93c604e93af7f6085df5b7cad4d96f538ece68d Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:09 +0100 Subject: [PATCH 091/337] ASoC: Intel: avs: Kill S24_LE format Eliminate all occurrences of S24_LE within PCM code, both HOST and LINK side. Replace those with MSBITS subformats to allow for granular selection when S32_LE is the format of choice. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-16-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/intel/avs/pcm.c | 20 +++++++++++++++----- sound/soc/intel/avs/topology.c | 13 ++++++++++++- 2 files changed, 27 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index f586a804099a..b6c48f88ca85 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -1073,8 +1073,10 @@ static const struct snd_pcm_hardware avs_pcm_hardware = { SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | + SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, .buffer_bytes_max = AZX_MAX_BUF_SIZE, .period_bytes_min = 128, .period_bytes_max = AZX_MAX_BUF_SIZE / 2, @@ -1225,8 +1227,10 @@ static const struct snd_soc_dai_driver i2s_dai_template = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | + SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, .capture = { .channels_min = 1, @@ -1234,8 +1238,10 @@ static const struct snd_soc_dai_driver i2s_dai_template = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_KNOT, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | + SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, }; @@ -1310,16 +1316,20 @@ static const struct snd_soc_dai_driver hda_cpu_dai = { .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | + SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, .capture = { .channels_min = 1, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | + SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX, }, }; diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index c74e9d622e4c..778236d3fd28 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1514,8 +1514,16 @@ static int avs_dai_load(struct snd_soc_component *comp, int index, struct snd_soc_dai_driver *dai_drv, struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - if (pcm) + u32 fe_subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | + SNDRV_PCM_SUBFMTBIT_MSBITS_24 | + SNDRV_PCM_SUBFMTBIT_MSBITS_MAX; + + if (pcm) { dai_drv->ops = &avs_dai_fe_ops; + dai_drv->capture.subformats = fe_subformats; + dai_drv->playback.subformats = fe_subformats; + } + return 0; } @@ -1534,6 +1542,9 @@ static int avs_link_load(struct snd_soc_component *comp, int index, struct snd_s /* Open LINK (BE) pipes last and close them first to prevent xruns. */ link->trigger[0] = SND_SOC_DPCM_TRIGGER_PRE; link->trigger[1] = SND_SOC_DPCM_TRIGGER_PRE; + } else { + /* Do not ignore codec capabilities. */ + link->dpcm_merged_format = 1; } return 0; From f8ccb133c9866b17a44f8e1a2478beb1631a366d Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Fri, 17 Nov 2023 13:06:10 +0100 Subject: [PATCH 092/337] ASoC: Intel: avs: Unhardcode HDAudio BE DAI drivers description To not expose more than in fact is supported by the codec, update CPU DAI initialization procedure to rely on codec capabilities instead of hardcoding them. This includes subformat which is currently ignored. As capabilities for HDMI streams are initialized on PCM open, leave it as is for now. Acked-by: Mark Brown Signed-off-by: Cezary Rojewski Acked-by: Jaroslav Kysela Link: https://lore.kernel.org/r/20231117120610.1755254-17-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai --- sound/soc/intel/avs/pcm.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index b6c48f88ca85..4dfc5a1ebb7c 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -17,6 +17,7 @@ #include "avs.h" #include "path.h" #include "topology.h" +#include "../../codecs/hda.h" struct avs_dma_data { struct avs_tplg_path_template *template; @@ -1406,6 +1407,15 @@ static int avs_component_hda_probe(struct snd_soc_component *component) ret = -ENOMEM; goto exit; } + + if (!hda_codec_is_display(codec)) { + dais[i].playback.formats = pcm->stream[0].formats; + dais[i].playback.subformats = pcm->stream[0].subformats; + dais[i].playback.rates = pcm->stream[0].rates; + dais[i].playback.channels_min = pcm->stream[0].channels_min; + dais[i].playback.channels_max = pcm->stream[0].channels_max; + dais[i].playback.sig_bits = pcm->stream[0].maxbps; + } } if (pcm->stream[1].substreams) { @@ -1416,6 +1426,15 @@ static int avs_component_hda_probe(struct snd_soc_component *component) ret = -ENOMEM; goto exit; } + + if (!hda_codec_is_display(codec)) { + dais[i].capture.formats = pcm->stream[1].formats; + dais[i].capture.subformats = pcm->stream[1].subformats; + dais[i].capture.rates = pcm->stream[1].rates; + dais[i].capture.channels_min = pcm->stream[1].channels_min; + dais[i].capture.channels_max = pcm->stream[1].channels_max; + dais[i].capture.sig_bits = pcm->stream[1].maxbps; + } } dai = snd_soc_register_dai(component, &dais[i], false); From ed99878462ccc143395987faebda33c50529b116 Mon Sep 17 00:00:00 2001 From: Chao Song Date: Mon, 27 Nov 2023 15:34:48 +0200 Subject: [PATCH 093/337] ASoC: Intel: soc-acpi-intel-mtl-match: Add rt722 support This patch adds match table for rt722 codec on link 0. RT722 is a multi-function codec, three endpoints are created for its headset, amp and dmic functions. Signed-off-by: Chao Song Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127133448.18449-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-mtl-match.c | 49 +++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c index 61f2b2645496..5a90ebdbf799 100644 --- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c @@ -123,6 +123,31 @@ static const struct snd_soc_acpi_endpoint rt712_endpoints[] = { }, }; +/* + * RT722 is a multi-function codec, three endpoints are created for + * its headset, amp and dmic functions. + */ +static const struct snd_soc_acpi_endpoint rt722_endpoints[] = { + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_endpoint spk_2_endpoint = { .num = 0, .aggregated = 1, @@ -164,6 +189,15 @@ static const struct snd_soc_acpi_adr_device rt1712_3_single_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt722_0_single_adr[] = { + { + .adr = 0x000030025d072201ull, + .num_endpoints = ARRAY_SIZE(rt722_endpoints), + .endpoints = rt722_endpoints, + .name_prefix = "rt722" + } +}; + static const struct snd_soc_acpi_adr_device rt713_0_single_adr[] = { { .adr = 0x000031025D071301ull, @@ -355,6 +389,15 @@ static const struct snd_soc_acpi_link_adr mtl_rvp[] = { {} }; +static const struct snd_soc_acpi_link_adr mtl_rt722_only[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt722_0_single_adr), + .adr_d = rt722_0_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr mtl_3_in_1_sdca[] = { { .mask = BIT(0), @@ -556,6 +599,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-mtl-rt711-l0-rt1316-l23-rt714-l1.tplg", }, + { + .link_mask = BIT(0), + .links = mtl_rt722_only, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-mtl-rt722-l0.tplg", + }, { .link_mask = BIT(0), .links = mtl_rvp, From 9425039741e84120fc936fbb5a9a7a1410fd9409 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:28 +0200 Subject: [PATCH 094/337] ASoC: Intel: sof_ssp_amp: remove dead code This patch fixes a dead code problem when calculating BE ID for each HDMI-In link. Signed-off-by: Brent Lu Reviewed-by: Balamurugan C Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 137ba64254bc..22f37cf3a2ad 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -124,6 +124,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, /* the topology supports HDMI-IN uses fixed BE ID for DAI links */ fixed_be = true; + be_id = HDMI_IN_BE_ID; for (i = 1; i <= num_of_hdmi_ssp; i++) { int port = (i == 1 ? (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_1_SSP_MASK) >> SOF_HDMI_CAPTURE_1_SSP_SHIFT : @@ -138,7 +139,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-HDMI", port); if (!links[id].name) return NULL; - links[id].id = fixed_be ? (HDMI_IN_BE_ID + i - 1) : id; + links[id].id = be_id; links[id].codecs = &snd_soc_dummy_dlc; links[id].num_codecs = 1; links[id].platforms = platform_component; @@ -147,6 +148,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, links[id].no_pcm = 1; links[id].num_cpus = 1; id++; + be_id++; } } From 06dea47be6d3d0ad3ef5f7c9dd0947d1c825e0ff Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 27 Nov 2023 17:26:29 +0200 Subject: [PATCH 095/337] ASoC: Intel: sof_maxim_common: add else between 2 if test if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) and if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) can't be true at the same time. Add an else to clarify it. Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Reviewed-by: Chao Song Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_maxim_common.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index 3c00afc32805..f64124077ca7 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -67,8 +67,7 @@ static int max_98373_hw_params(struct snd_pcm_substream *substream, if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) { /* DEV0 tdm slot configuration */ snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 32); - } - if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { + } else if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { /* DEV1 tdm slot configuration */ snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 32); } From 007d9a638b877f71a0ef28a906525904e44f6aa2 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 27 Nov 2023 17:26:30 +0200 Subject: [PATCH 096/337] ASoC: Intel: sof_maxim_common: check return value snd_soc_dai_set_tdm_slot() could return error. Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Reviewed-by: Chao Song Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_maxim_common.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index f64124077ca7..cf2974718271 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -61,15 +61,21 @@ static int max_98373_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; + int ret = 0; int j; for_each_rtd_codec_dais(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX_98373_DEV0_NAME)) { /* DEV0 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 32); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 32); } else if (!strcmp(codec_dai->component->name, MAX_98373_DEV1_NAME)) { /* DEV1 tdm slot configuration */ - snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 32); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 32); + } + if (ret < 0) { + dev_err(codec_dai->dev, "fail to set tdm slot, ret %d\n", + ret); + return ret; } } return 0; From 65b2df10a1e6264dd199c88623a9e4bbf8849c9b Mon Sep 17 00:00:00 2001 From: Chao Song Date: Mon, 27 Nov 2023 17:26:31 +0200 Subject: [PATCH 097/337] ASoC: Intel: cht_bsw_rt5672: check return value Set codec sysclk could fail and return error, add error check for it. Signed-off-by: Chao Song Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index f6da24f3c466..8cf0b33cc02e 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -93,8 +93,12 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, * when codec is runtime suspended. Codec needs clock for jack * detection and button press. */ - snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, - 48000 * 512, SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK, + 48000 * 512, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } if (ctx->mclk) clk_disable_unprepare(ctx->mclk); From 93f74ebf3d7623b8b647d2ce5f2e23545f9702df Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:32 +0200 Subject: [PATCH 098/337] ASoC: Intel: ssp-common: get codec name function Add a helper function to get codec name string from codec type enum value. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_common.c | 21 +++++++++++++++++++++ sound/soc/intel/boards/sof_ssp_common.h | 1 + 2 files changed, 22 insertions(+) diff --git a/sound/soc/intel/boards/sof_ssp_common.c b/sound/soc/intel/boards/sof_ssp_common.c index 41a258e45a61..96072790e9c0 100644 --- a/sound/soc/intel/boards/sof_ssp_common.c +++ b/sound/soc/intel/boards/sof_ssp_common.c @@ -96,6 +96,27 @@ enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev) } EXPORT_SYMBOL_NS(sof_ssp_detect_amp_type, SND_SOC_INTEL_SOF_SSP_COMMON); +const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(codecs); i++) { + if (codecs[i].codec_type != codec_type) + continue; + + return codecs[i].name; + } + for (i = 0; i < ARRAY_SIZE(amps); i++) { + if (amps[i].codec_type != codec_type) + continue; + + return amps[i].name; + } + + return NULL; +} +EXPORT_SYMBOL_NS(sof_ssp_get_codec_name, SND_SOC_INTEL_SOF_SSP_COMMON); + MODULE_DESCRIPTION("ASoC Intel SOF Common Machine Driver Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_ssp_common.h b/sound/soc/intel/boards/sof_ssp_common.h index e3fd6fb1db1c..6d827103479b 100644 --- a/sound/soc/intel/boards/sof_ssp_common.h +++ b/sound/soc/intel/boards/sof_ssp_common.h @@ -67,5 +67,6 @@ enum sof_ssp_codec { enum sof_ssp_codec sof_ssp_detect_codec_type(struct device *dev); enum sof_ssp_codec sof_ssp_detect_amp_type(struct device *dev); +const char *sof_ssp_get_codec_name(enum sof_ssp_codec codec_type); #endif /* __SOF_SSP_COMMON_H */ From e111dece012f29707da634c341d3f3277bd94528 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:33 +0200 Subject: [PATCH 099/337] ASoC: Intel: board_helpers: support codec link initialization Add a helper function for machine drivers to initialize headphone codec DAI links. The function will initialize common fields and let caller to initialize codec-specific fields like codec, init, exit, and ops fields. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 51 ++++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 6 +++ 2 files changed, 57 insertions(+) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index ce2967850c2d..5ee53c781b37 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -3,6 +3,7 @@ // Copyright(c) 2023 Intel Corporation. All rights reserved. #include +#include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" #include "sof_board_helpers.h" @@ -86,6 +87,55 @@ static struct snd_soc_dai_link_component platform_component[] = { } }; +int sof_intel_board_set_codec_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + enum sof_ssp_codec codec_type, int ssp_codec) +{ + struct snd_soc_dai_link_component *cpus; + + dev_dbg(dev, "link %d: codec %s, ssp %d\n", be_id, + sof_ssp_get_codec_name(codec_type), ssp_codec); + + /* link name */ + link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_codec); + if (!link->name) + return -ENOMEM; + + /* cpus */ + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!cpus) + return -ENOMEM; + + if (soc_intel_is_byt() || soc_intel_is_cht()) { + /* backward-compatibility for BYT/CHT boards */ + cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "ssp%d-port", + ssp_codec); + } else { + cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", + ssp_codec); + } + if (!cpus->dai_name) + return -ENOMEM; + + link->cpus = cpus; + link->num_cpus = 1; + + /* codecs - caller to handle */ + + /* platforms */ + link->platforms = platform_component; + link->num_platforms = ARRAY_SIZE(platform_component); + + link->id = be_id; + link->no_pcm = 1; + link->dpcm_capture = 1; + link->dpcm_playback = 1; + + return 0; +} +EXPORT_SYMBOL_NS(sof_intel_board_set_codec_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); + int sof_intel_board_set_dmic_link(struct device *dev, struct snd_soc_dai_link *link, int be_id, enum sof_dmic_be_type be_type) @@ -202,3 +252,4 @@ MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_SSP_COMMON); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index df99f576c1d8..7392d639672d 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -30,6 +30,7 @@ struct sof_rt5682_private { * @amp_type: type of speaker amplifier * @dmic_be_num: number of Intel PCH DMIC BE link * @hdmi_num: number of Intel HDMI BE link + * @ssp_codec: ssp port number of headphone BE link * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -42,6 +43,8 @@ struct sof_card_private { int dmic_be_num; int hdmi_num; + int ssp_codec; + union { struct sof_rt5682_private rt5682; }; @@ -54,6 +57,9 @@ enum sof_dmic_be_type { int sof_intel_board_card_late_probe(struct snd_soc_card *card); +int sof_intel_board_set_codec_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + enum sof_ssp_codec codec_type, int ssp_codec); int sof_intel_board_set_dmic_link(struct device *dev, struct snd_soc_dai_link *link, int be_id, enum sof_dmic_be_type be_type); From f46f07fe264a26daf8d6a5e16535229ee48108a7 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:34 +0200 Subject: [PATCH 100/337] ASoC: Intel: sof_cs42l42: use common module for codec link Use intel_board module for headphone codec DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 60 ++++++++-------------------- 1 file changed, 16 insertions(+), 44 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 1f760fc4cab2..30e78c20ce6e 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -245,45 +245,6 @@ devm_err: return ret; } -static int create_hp_codec_dai_links(struct device *dev, - struct snd_soc_dai_link *links, - struct snd_soc_dai_link_component *cpus, - int *id, int ssp_codec) -{ - /* codec SSP */ - links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", - ssp_codec); - if (!links[*id].name) - goto devm_err; - - links[*id].id = *id; - links[*id].codecs = cs42l42_component; - links[*id].num_codecs = ARRAY_SIZE(cs42l42_component); - links[*id].platforms = platform_component; - links[*id].num_platforms = ARRAY_SIZE(platform_component); - links[*id].init = sof_cs42l42_init; - links[*id].exit = sof_cs42l42_exit; - links[*id].ops = &sof_cs42l42_ops; - links[*id].dpcm_playback = 1; - links[*id].dpcm_capture = 1; - links[*id].no_pcm = 1; - links[*id].cpus = &cpus[*id]; - links[*id].num_cpus = 1; - - links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_codec); - if (!links[*id].cpus->dai_name) - goto devm_err; - - (*id)++; - - return 0; - -devm_err: - return -ENOMEM; -} - static int create_bt_offload_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -350,12 +311,23 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, switch (link_type) { case LINK_HP: - ret = create_hp_codec_dai_links(dev, links, cpus, &id, ssp_codec); - if (ret < 0) { + ret = sof_intel_board_set_codec_link(dev, &links[id], id, + CODEC_CS42L42, + ssp_codec); + if (ret) { dev_err(dev, "fail to create hp codec dai links, ret %d\n", ret); goto devm_err; } + + /* codec-specific fields */ + links[id].codecs = cs42l42_component; + links[id].num_codecs = ARRAY_SIZE(cs42l42_component); + links[id].init = sof_cs42l42_init; + links[id].exit = sof_cs42l42_exit; + links[id].ops = &sof_cs42l42_ops; + + id++; break; case LINK_SPK: ret = create_spk_amp_dai_links(dev, links, cpus, &id, @@ -440,7 +412,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_bt, ssp_amp, ssp_codec; + int ret, ssp_bt, ssp_amp; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -475,7 +447,7 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; - ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK; + ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK; /* compute number of dai links */ sof_audio_card_cs42l42.num_links = 1 + ctx->dmic_be_num + ctx->hdmi_num; @@ -486,7 +458,7 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ssp_codec, ssp_amp, ssp_bt, + ctx->ssp_codec, ssp_amp, ssp_bt, ctx->dmic_be_num, ctx->hdmi_num, ctx->hdmi.idisp_codec); if (!dai_links) From 99f7422805c96d56a093dbe71beca658b793d0cb Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:35 +0200 Subject: [PATCH 101/337] ASoC: Intel: sof_nau8825: use common module for codec link Use intel_board module for headphone codec DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 29 ++++++++-------------------- 1 file changed, 8 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index dc2821a012d4..3aeed23c8d0d 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -226,30 +226,17 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, goto devm_err; /* codec SSP */ - links[id].name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d-Codec", ssp_codec); - if (!links[id].name) - goto devm_err; + ret = sof_intel_board_set_codec_link(dev, &links[id], id, CODEC_NAU8825, + ssp_codec); + if (ret) + return NULL; - links[id].id = id; + /* codec-specific fields */ links[id].codecs = nau8825_component; links[id].num_codecs = ARRAY_SIZE(nau8825_component); - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); links[id].init = sof_nau8825_codec_init; links[id].exit = sof_nau8825_codec_exit; links[id].ops = &sof_nau8825_ops; - links[id].dpcm_playback = 1; - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_codec); - if (!links[id].cpus->dai_name) - goto devm_err; id++; @@ -368,7 +355,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_amp, ssp_codec; + int ret, ssp_amp; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -396,7 +383,7 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> SOF_NAU8825_SSP_AMP_SHIFT; - ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; + ctx->ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; /* compute number of dai links */ sof_audio_card_nau8825.num_links = 1 + ctx->dmic_be_num + ctx->hdmi_num; @@ -408,7 +395,7 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_nau8825.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ssp_codec, ssp_amp, + ctx->ssp_codec, ssp_amp, ctx->dmic_be_num, ctx->hdmi_num, ctx->hdmi.idisp_codec); if (!dai_links) From 84c280af16b72fc202fbd9dcecadb6e256956c41 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:36 +0200 Subject: [PATCH 102/337] ASoC: Intel: sof_rt5682: use common module for codec link Use intel_board module for headphone codec DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 38 ++++++++--------------------- 1 file changed, 10 insertions(+), 28 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 9723479f43da..8adc82892e2c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -591,13 +591,12 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, goto devm_err; /* codec SSP */ - links[id].name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d-Codec", ssp_codec); - if (!links[id].name) - goto devm_err; - - links[id].id = id; + ret = sof_intel_board_set_codec_link(dev, &links[id], id, codec_type, + ssp_codec); + if (ret) + return NULL; + /* codec-specific fields */ switch (codec_type) { case CODEC_RT5650: links[id].codecs = &rt5650_components[0]; @@ -616,23 +615,11 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, return NULL; } - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); links[id].init = sof_rt5682_codec_init; links[id].exit = sof_rt5682_codec_exit; links[id].ops = &sof_rt5682_ops; - links[id].dpcm_playback = 1; - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - if (is_legacy_cpu) { - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "ssp%d-port", - ssp_codec); - if (!links[id].cpus->dai_name) - goto devm_err; - } else { + + if (!is_legacy_cpu) { /* * Currently, On SKL+ platforms MCLK will be turned off in sof * runtime suspended, and it will go into runtime suspended @@ -643,11 +630,6 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, * It can be removed once we can control MCLK by driver. */ links[id].ignore_pmdown_time = 1; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_codec); - if (!links[id].cpus->dai_name) - goto devm_err; } id++; @@ -819,7 +801,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_amp, ssp_codec; + int ret, ssp_amp; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -888,7 +870,7 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >> SOF_RT5682_SSP_AMP_SHIFT; - ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK; + ctx->ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK; /* compute number of dai links */ sof_audio_card_rt5682.num_links = 1 + ctx->dmic_be_num + ctx->hdmi_num; @@ -905,7 +887,7 @@ static int sof_audio_probe(struct platform_device *pdev) SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT); dai_links = sof_card_dai_links_create(&pdev->dev, ctx->codec_type, - ctx->amp_type, ssp_codec, ssp_amp, + ctx->amp_type, ctx->ssp_codec, ssp_amp, ctx->dmic_be_num, ctx->hdmi_num, ctx->hdmi.idisp_codec, ctx->rt5682.is_legacy_cpu); From ba0c7c328762d78573d3cac4adad9ea9e695f244 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:37 +0200 Subject: [PATCH 103/337] ASoC: Intel: board_helpers: support amp link initialization Add a helper function for machine drivers to initialize speaker amplifier DAI link. The function will initialize common fields and let caller to initialize codec-specific fields like codec, init, exit, and ops fields. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-11-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 42 ++++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 5 +++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 5ee53c781b37..515634db0a4d 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -248,6 +248,48 @@ int sof_intel_board_set_intel_hdmi_link(struct device *dev, } EXPORT_SYMBOL_NS(sof_intel_board_set_intel_hdmi_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); +int sof_intel_board_set_ssp_amp_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + enum sof_ssp_codec amp_type, int ssp_amp) +{ + struct snd_soc_dai_link_component *cpus; + + dev_dbg(dev, "link %d: ssp amp %s, ssp %d\n", be_id, + sof_ssp_get_codec_name(amp_type), ssp_amp); + + /* link name */ + link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_amp); + if (!link->name) + return -ENOMEM; + + /* cpus */ + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!cpus) + return -ENOMEM; + + cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_amp); + if (!cpus->dai_name) + return -ENOMEM; + + link->cpus = cpus; + link->num_cpus = 1; + + /* codecs - caller to handle */ + + /* platforms */ + link->platforms = platform_component; + link->num_platforms = ARRAY_SIZE(platform_component); + + link->id = be_id; + link->no_pcm = 1; + link->dpcm_capture = 1; /* feedback stream or firmware-generated echo reference */ + link->dpcm_playback = 1; + + return 0; +} +EXPORT_SYMBOL_NS(sof_intel_board_set_ssp_amp_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); + MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 7392d639672d..17922f3e17a5 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -31,6 +31,7 @@ struct sof_rt5682_private { * @dmic_be_num: number of Intel PCH DMIC BE link * @hdmi_num: number of Intel HDMI BE link * @ssp_codec: ssp port number of headphone BE link + * @ssp_amp: ssp port number of speaker BE link * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -44,6 +45,7 @@ struct sof_card_private { int hdmi_num; int ssp_codec; + int ssp_amp; union { struct sof_rt5682_private rt5682; @@ -66,5 +68,8 @@ int sof_intel_board_set_dmic_link(struct device *dev, int sof_intel_board_set_intel_hdmi_link(struct device *dev, struct snd_soc_dai_link *link, int be_id, int hdmi_id, bool idisp_codec); +int sof_intel_board_set_ssp_amp_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + enum sof_ssp_codec amp_type, int ssp_amp); #endif /* __SOF_INTEL_BOARD_HELPERS_H */ From 8739841805744bd1a1d12e2485dc5d0f1d880f64 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:38 +0200 Subject: [PATCH 104/337] ASoC: Intel: sof_cs42l42: use common module for amp link Use intel_board module for speaker amplifier DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-12-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 98 +++++++++------------------- 1 file changed, 32 insertions(+), 66 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 30e78c20ce6e..a8252079d696 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -189,62 +189,6 @@ static struct snd_soc_dai_link_component cs42l42_component[] = { } }; -static int create_spk_amp_dai_links(struct device *dev, - struct snd_soc_dai_link *links, - struct snd_soc_dai_link_component *cpus, - int *id, enum sof_ssp_codec amp_type, - int ssp_amp) -{ - int ret = 0; - - /* speaker amp */ - if (amp_type == CODEC_NONE) - return 0; - - links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", - ssp_amp); - if (!links[*id].name) { - ret = -ENOMEM; - goto devm_err; - } - - links[*id].id = *id; - - switch (amp_type) { - case CODEC_MAX98357A: - max_98357a_dai_link(&links[*id]); - break; - case CODEC_MAX98360A: - max_98360a_dai_link(&links[*id]); - break; - default: - dev_err(dev, "invalid amp type %d\n", amp_type); - return -EINVAL; - } - - links[*id].platforms = platform_component; - links[*id].num_platforms = ARRAY_SIZE(platform_component); - links[*id].dpcm_playback = 1; - /* firmware-generated echo reference */ - links[*id].dpcm_capture = 1; - - links[*id].no_pcm = 1; - links[*id].cpus = &cpus[*id]; - links[*id].num_cpus = 1; - - links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", ssp_amp); - if (!links[*id].cpus->dai_name) { - ret = -ENOMEM; - goto devm_err; - } - - (*id)++; - -devm_err: - return ret; -} - static int create_bt_offload_dai_links(struct device *dev, struct snd_soc_dai_link *links, struct snd_soc_dai_link_component *cpus, @@ -330,12 +274,33 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, id++; break; case LINK_SPK: - ret = create_spk_amp_dai_links(dev, links, cpus, &id, - amp_type, ssp_amp); - if (ret < 0) { - dev_err(dev, "fail to create spk amp dai links, ret %d\n", - ret); - goto devm_err; + if (amp_type != CODEC_NONE) { + ret = sof_intel_board_set_ssp_amp_link(dev, + &links[id], + id, + amp_type, + ssp_amp); + if (ret) { + dev_err(dev, "fail to create spk amp dai links, ret %d\n", + ret); + goto devm_err; + } + + /* codec-specific fields */ + switch (amp_type) { + case CODEC_MAX98357A: + max_98357a_dai_link(&links[id]); + break; + case CODEC_MAX98360A: + max_98360a_dai_link(&links[id]); + break; + default: + dev_err(dev, "invalid amp type %d\n", + amp_type); + goto devm_err; + } + + id++; } break; case LINK_DMIC: @@ -412,7 +377,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_bt, ssp_amp; + int ret, ssp_bt; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -444,7 +409,7 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> SOF_CS42L42_SSP_BT_SHIFT; - ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> + ctx->ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> SOF_CS42L42_SSP_AMP_SHIFT; ctx->ssp_codec = sof_cs42l42_quirk & SOF_CS42L42_SSP_CODEC_MASK; @@ -458,8 +423,9 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_cs42l42.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ctx->ssp_codec, ssp_amp, ssp_bt, - ctx->dmic_be_num, ctx->hdmi_num, + ctx->ssp_codec, ctx->ssp_amp, + ssp_bt, ctx->dmic_be_num, + ctx->hdmi_num, ctx->hdmi.idisp_codec); if (!dai_links) return -ENOMEM; From adf711655ba21c5d54d52b79aa8df87fbb39df6b Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:39 +0200 Subject: [PATCH 105/337] ASoC: Intel: sof_nau8825: use common module for amp link Use intel_board module for speaker amplifier DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-13-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 31 +++++++--------------------- 1 file changed, 8 insertions(+), 23 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 3aeed23c8d0d..cc2a11b3de97 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -273,13 +273,12 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, /* speaker amp */ if (amp_type != CODEC_NONE) { - links[id].name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d-Codec", ssp_amp); - if (!links[id].name) - goto devm_err; - - links[id].id = id; + ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], id, + amp_type, ssp_amp); + if (ret) + return NULL; + /* codec-specific fields */ switch (amp_type) { case CODEC_MAX98360A: max_98360a_dai_link(&links[id]); @@ -304,20 +303,6 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, return NULL; } - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_playback = 1; - /* feedback stream or firmware-generated echo reference */ - links[id].dpcm_capture = 1; - - links[id].no_pcm = 1; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_amp); - if (!links[id].cpus->dai_name) - goto devm_err; id++; } @@ -355,7 +340,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_amp; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -380,7 +365,7 @@ static int sof_audio_probe(struct platform_device *pdev) if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> + ctx->ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> SOF_NAU8825_SSP_AMP_SHIFT; ctx->ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; @@ -395,7 +380,7 @@ static int sof_audio_probe(struct platform_device *pdev) sof_audio_card_nau8825.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ctx->ssp_codec, ssp_amp, + ctx->ssp_codec, ctx->ssp_amp, ctx->dmic_be_num, ctx->hdmi_num, ctx->hdmi.idisp_codec); if (!dai_links) From e45cd972a50d2212ad232af33de970fb6923aaf6 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:40 +0200 Subject: [PATCH 106/337] ASoC: Intel: sof_rt5682: use common module for amp link Use intel_board module for speaker amplifier DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 43 +++++++---------------------- 1 file changed, 10 insertions(+), 33 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 8adc82892e2c..777d1db5c6ad 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -666,13 +666,12 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, /* speaker amp */ if (amp_type != CODEC_NONE) { - links[id].name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d-Codec", ssp_amp); - if (!links[id].name) - goto devm_err; - - links[id].id = id; + ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], id, + amp_type, ssp_amp); + if (ret) + return NULL; + /* codec-specific fields */ switch (amp_type) { case CODEC_MAX98357A: max_98357a_dai_link(&links[id]); @@ -713,29 +712,6 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, return NULL; } - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_playback = 1; - /* feedback stream or firmware-generated echo reference */ - links[id].dpcm_capture = 1; - - links[id].no_pcm = 1; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - if (is_legacy_cpu) { - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "ssp%d-port", - ssp_amp); - if (!links[id].cpus->dai_name) - goto devm_err; - - } else { - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_amp); - if (!links[id].cpus->dai_name) - goto devm_err; - } id++; } @@ -801,7 +777,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_amp; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -867,7 +843,7 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); - ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >> + ctx->ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >> SOF_RT5682_SSP_AMP_SHIFT; ctx->ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK; @@ -887,8 +863,9 @@ static int sof_audio_probe(struct platform_device *pdev) SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT); dai_links = sof_card_dai_links_create(&pdev->dev, ctx->codec_type, - ctx->amp_type, ctx->ssp_codec, ssp_amp, - ctx->dmic_be_num, ctx->hdmi_num, + ctx->amp_type, ctx->ssp_codec, + ctx->ssp_amp, ctx->dmic_be_num, + ctx->hdmi_num, ctx->hdmi.idisp_codec, ctx->rt5682.is_legacy_cpu); if (!dai_links) From 5cdc7a82594eaad27e98dcd6fbd72592edbc0f39 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:41 +0200 Subject: [PATCH 107/337] ASoC: Intel: sof_ssp_amp: use common module for amp link Use intel_board module for speaker amplifier DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-15-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 27 ++++++++------------------- 1 file changed, 8 insertions(+), 19 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 22f37cf3a2ad..8e478d1cc875 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -154,12 +154,13 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, /* codec SSP */ if (amp_type != CODEC_NONE) { - links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_codec); - if (!links[id].name) + be_id = fixed_be ? SPK_BE_ID : id; + ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], be_id, + amp_type, ssp_codec); + if (ret) return NULL; - links[id].id = fixed_be ? SPK_BE_ID : id; - + /* codec-specific fields */ switch (amp_type) { case CODEC_CS35L41: cs35l41_set_dai_link(&links[id]); @@ -172,18 +173,6 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, return NULL; } - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_playback = 1; - /* feedback from amplifier or firmware-generated echo reference */ - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_codec); - if (!links[id].cpus->dai_name) - return NULL; - id++; } @@ -256,7 +245,7 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_codec; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -272,7 +261,7 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) else ctx->dmic_be_num = 0; - ssp_codec = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; + ctx->ssp_amp = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; /* set number of dai links */ sof_ssp_amp_card.num_links = ctx->dmic_be_num; @@ -303,7 +292,7 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) sof_ssp_amp_card.num_links++; dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ssp_codec, ctx->dmic_be_num, + ctx->ssp_amp, ctx->dmic_be_num, ctx->hdmi_num, ctx->hdmi.idisp_codec); if (!dai_links) From 823404815fcdf32950b2223ed2c665f077d6b98f Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:42 +0200 Subject: [PATCH 108/337] ASoC: Intel: sof_ssp_amp: rename function parameter Rename the parameter 'ssp_codec' of sof_card_dai_links_create() since it's the port number of speaker amplifier. No functional change in this commit. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-16-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 8e478d1cc875..c463bc698c10 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -98,7 +98,7 @@ static struct snd_soc_dai_link_component platform_component[] = { static struct snd_soc_dai_link * sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, - int ssp_codec, int dmic_be_num, int hdmi_num, + int ssp_amp, int dmic_be_num, int hdmi_num, bool idisp_codec) { struct snd_soc_dai_link_component *cpus; @@ -156,7 +156,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, if (amp_type != CODEC_NONE) { be_id = fixed_be ? SPK_BE_ID : id; ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], be_id, - amp_type, ssp_codec); + amp_type, ssp_amp); if (ret) return NULL; From 53d8df6d3f0a1d6fd520865fb12c11868fe6cf81 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:43 +0200 Subject: [PATCH 109/337] ASoC: Intel: board_helpers: support BT offload link initialization Add a helper function for machine drivers to initialize BT offload DAI link. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-17-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 43 ++++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 8 ++++ 2 files changed, 51 insertions(+) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 515634db0a4d..335c660561d5 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -290,6 +290,49 @@ int sof_intel_board_set_ssp_amp_link(struct device *dev, } EXPORT_SYMBOL_NS(sof_intel_board_set_ssp_amp_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); +int sof_intel_board_set_bt_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + int ssp_bt) +{ + struct snd_soc_dai_link_component *cpus; + + dev_dbg(dev, "link %d: bt offload, ssp %d\n", be_id, ssp_bt); + + /* link name */ + link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", ssp_bt); + if (!link->name) + return -ENOMEM; + + /* cpus */ + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!cpus) + return -ENOMEM; + + cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_bt); + if (!cpus->dai_name) + return -ENOMEM; + + link->cpus = cpus; + link->num_cpus = 1; + + /* codecs */ + link->codecs = &snd_soc_dummy_dlc; + link->num_codecs = 1; + + /* platforms */ + link->platforms = platform_component; + link->num_platforms = ARRAY_SIZE(platform_component); + + link->id = be_id; + link->no_pcm = 1; + link->dpcm_capture = 1; + link->dpcm_playback = 1; + + return 0; +} +EXPORT_SYMBOL_NS(sof_intel_board_set_bt_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); + MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 17922f3e17a5..49a7bfa12a6d 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -32,6 +32,8 @@ struct sof_rt5682_private { * @hdmi_num: number of Intel HDMI BE link * @ssp_codec: ssp port number of headphone BE link * @ssp_amp: ssp port number of speaker BE link + * @ssp_bt: ssp port number of BT offload BE link + * @bt_offload_present: true to create BT offload BE link * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -46,6 +48,9 @@ struct sof_card_private { int ssp_codec; int ssp_amp; + int ssp_bt; + + bool bt_offload_present; union { struct sof_rt5682_private rt5682; @@ -71,5 +76,8 @@ int sof_intel_board_set_intel_hdmi_link(struct device *dev, int sof_intel_board_set_ssp_amp_link(struct device *dev, struct snd_soc_dai_link *link, int be_id, enum sof_ssp_codec amp_type, int ssp_amp); +int sof_intel_board_set_bt_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + int ssp_bt); #endif /* __SOF_INTEL_BOARD_HELPERS_H */ From 117445d7655945b0a601a1247842029d0325c7e4 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:44 +0200 Subject: [PATCH 110/337] ASoC: Intel: sof_cs42l42: use common module for BT offload link Use intel_board module for BT offload DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-18-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_cs42l42.c | 79 +++++++--------------------- 1 file changed, 19 insertions(+), 60 deletions(-) diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index a8252079d696..c2442bf8ced0 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -138,13 +138,6 @@ static const struct snd_soc_ops sof_cs42l42_ops = { .hw_params = sof_cs42l42_hw_params, }; -static struct snd_soc_dai_link_component platform_component[] = { - { - /* name might be overridden during probe */ - .name = "0000:00:1f.3" - } -}; - static int sof_card_late_probe(struct snd_soc_card *card) { return sof_intel_board_card_late_probe(card); @@ -189,52 +182,11 @@ static struct snd_soc_dai_link_component cs42l42_component[] = { } }; -static int create_bt_offload_dai_links(struct device *dev, - struct snd_soc_dai_link *links, - struct snd_soc_dai_link_component *cpus, - int *id, int ssp_bt) -{ - /* bt offload */ - if (!(sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT)) - return 0; - - links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", - ssp_bt); - if (!links[*id].name) - goto devm_err; - - links[*id].id = *id; - links[*id].codecs = &snd_soc_dummy_dlc; - links[*id].num_codecs = 1; - links[*id].platforms = platform_component; - links[*id].num_platforms = ARRAY_SIZE(platform_component); - - links[*id].dpcm_playback = 1; - links[*id].dpcm_capture = 1; - links[*id].no_pcm = 1; - links[*id].cpus = &cpus[*id]; - links[*id].num_cpus = 1; - - links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_bt); - if (!links[*id].cpus->dai_name) - goto devm_err; - - (*id)++; - - return 0; - -devm_err: - return -ENOMEM; -} - static struct snd_soc_dai_link * sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, int ssp_codec, int ssp_amp, int ssp_bt, int dmic_be_num, int hdmi_num, bool idisp_codec) { - struct snd_soc_dai_link_component *cpus; struct snd_soc_dai_link *links; int ret; int id = 0; @@ -243,9 +195,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, links = devm_kcalloc(dev, sof_audio_card_cs42l42.num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL); - cpus = devm_kcalloc(dev, sof_audio_card_cs42l42.num_links, - sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); - if (!links || !cpus) + if (!links) goto devm_err; link_seq = (sof_cs42l42_quirk & SOF_CS42L42_DAILINK_MASK) >> SOF_CS42L42_DAILINK_SHIFT; @@ -350,11 +300,17 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, } break; case LINK_BT: - ret = create_bt_offload_dai_links(dev, links, cpus, &id, ssp_bt); - if (ret < 0) { - dev_err(dev, "fail to create bt offload dai links, ret %d\n", - ret); - goto devm_err; + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) { + ret = sof_intel_board_set_bt_link(dev, + &links[id], id, + ssp_bt); + if (ret) { + dev_err(dev, "fail to create bt offload dai links, ret %d\n", + ret); + goto devm_err; + } + + id++; } break; case LINK_NONE: @@ -377,7 +333,7 @@ static int sof_audio_probe(struct platform_device *pdev) struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; - int ret, ssp_bt; + int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -406,7 +362,8 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_cs42l42_quirk = %lx\n", sof_cs42l42_quirk); - ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> + /* port number of peripherals attached to ssp interface */ + ctx->ssp_bt = (sof_cs42l42_quirk & SOF_CS42L42_SSP_BT_MASK) >> SOF_CS42L42_SSP_BT_SHIFT; ctx->ssp_amp = (sof_cs42l42_quirk & SOF_CS42L42_SSP_AMP_MASK) >> @@ -419,12 +376,14 @@ static int sof_audio_probe(struct platform_device *pdev) if (ctx->amp_type != CODEC_NONE) sof_audio_card_cs42l42.num_links++; - if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) + if (sof_cs42l42_quirk & SOF_BT_OFFLOAD_PRESENT) { + ctx->bt_offload_present = true; sof_audio_card_cs42l42.num_links++; + } dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, ctx->ssp_codec, ctx->ssp_amp, - ssp_bt, ctx->dmic_be_num, + ctx->ssp_bt, ctx->dmic_be_num, ctx->hdmi_num, ctx->hdmi.idisp_codec); if (!dai_links) From 87ddfdc9dc37032492704b51ab468bda1262d155 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:45 +0200 Subject: [PATCH 111/337] ASoC: Intel: sof_nau8825: use common module for BT offload link Use intel_board module for BT offload DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-19-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 42 +++++++++------------------- 1 file changed, 13 insertions(+), 29 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index cc2a11b3de97..32a43d1b5846 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -138,13 +138,6 @@ static struct snd_soc_ops sof_nau8825_ops = { .hw_params = sof_nau8825_hw_params, }; -static struct snd_soc_dai_link_component platform_component[] = { - { - /* name might be overridden during probe */ - .name = "0000:00:1f.3" - } -}; - static int sof_card_late_probe(struct snd_soc_card *card) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); @@ -212,7 +205,6 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, int ssp_codec, int ssp_amp, int dmic_be_num, int hdmi_num, bool idisp_codec) { - struct snd_soc_dai_link_component *cpus; struct snd_soc_dai_link *links; int i; int id = 0; @@ -220,9 +212,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, links = devm_kcalloc(dev, sof_audio_card_nau8825.num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL); - cpus = devm_kcalloc(dev, sof_audio_card_nau8825.num_links, - sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); - if (!links || !cpus) + if (!links) goto devm_err; /* codec SSP */ @@ -311,23 +301,11 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, int port = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> SOF_BT_OFFLOAD_SSP_SHIFT; - links[id].id = id; - links[id].cpus = &cpus[id]; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", port); - if (!links[id].cpus->dai_name) - goto devm_err; - links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); - if (!links[id].name) - goto devm_err; - links[id].codecs = &snd_soc_dummy_dlc; - links[id].num_codecs = 1; - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_playback = 1; - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].num_cpus = 1; + ret = sof_intel_board_set_bt_link(dev, &links[id], id, port); + if (ret) + return NULL; + + id++; } return links; @@ -365,6 +343,10 @@ static int sof_audio_probe(struct platform_device *pdev) if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; + /* port number of peripherals attached to ssp interface */ + ctx->ssp_bt = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + ctx->ssp_amp = (sof_nau8825_quirk & SOF_NAU8825_SSP_AMP_MASK) >> SOF_NAU8825_SSP_AMP_SHIFT; @@ -376,8 +358,10 @@ static int sof_audio_probe(struct platform_device *pdev) if (ctx->amp_type != CODEC_NONE) sof_audio_card_nau8825.num_links++; - if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + ctx->bt_offload_present = true; sof_audio_card_nau8825.num_links++; + } dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, ctx->ssp_codec, ctx->ssp_amp, From 3d5b77b9bee016cd1363f3856a3616eaed9026b8 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:46 +0200 Subject: [PATCH 112/337] ASoC: Intel: sof_rt5682: use common module for BT offload link Use intel_board module for BT offload DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-20-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 30 ++++++++++++----------------- 1 file changed, 12 insertions(+), 18 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 777d1db5c6ad..22dd85129a51 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -720,23 +720,11 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, int port = (sof_rt5682_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> SOF_BT_OFFLOAD_SSP_SHIFT; - links[id].id = id; - links[id].cpus = &cpus[id]; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", port); - if (!links[id].cpus->dai_name) - goto devm_err; - links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); - if (!links[id].name) - goto devm_err; - links[id].codecs = &snd_soc_dummy_dlc; - links[id].num_codecs = 1; - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_playback = 1; - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].num_cpus = 1; + ret = sof_intel_board_set_bt_link(dev, &links[id], id, port); + if (ret) + return NULL; + + id++; } /* HDMI-In SSP */ @@ -843,6 +831,10 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); + /* port number of peripherals attached to ssp interface */ + ctx->ssp_bt = (sof_rt5682_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + ctx->ssp_amp = (sof_rt5682_quirk & SOF_RT5682_SSP_AMP_MASK) >> SOF_RT5682_SSP_AMP_SHIFT; @@ -854,8 +846,10 @@ static int sof_audio_probe(struct platform_device *pdev) if (ctx->amp_type != CODEC_NONE) sof_audio_card_rt5682.num_links++; - if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + ctx->bt_offload_present = true; sof_audio_card_rt5682.num_links++; + } if (sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) sof_audio_card_rt5682.num_links += From dc3d7dcb04eae6af5b9c882dc30eede75bbd7fe7 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:47 +0200 Subject: [PATCH 113/337] ASoC: Intel: sof_ssp_amp: use common module for BT offload link Use intel_board module for BT offload DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-21-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 31 ++++++++++------------------ 1 file changed, 11 insertions(+), 20 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index c463bc698c10..72505f6a6501 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -215,29 +215,14 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, int port = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> SOF_BT_OFFLOAD_SSP_SHIFT; - links[id].id = id; - links[id].cpus = &cpus[id]; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", port); - if (!links[id].cpus->dai_name) - goto devm_err; - links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); - if (!links[id].name) - goto devm_err; - links[id].codecs = &snd_soc_dummy_dlc; - links[id].num_codecs = 1; - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_playback = 1; - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].num_cpus = 1; + ret = sof_intel_board_set_bt_link(dev, &links[id], id, port); + if (ret) + return NULL; + id++; } return links; -devm_err: - return NULL; } static int sof_ssp_amp_probe(struct platform_device *pdev) @@ -261,6 +246,10 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) else ctx->dmic_be_num = 0; + /* port number of peripherals attached to ssp interface */ + ctx->ssp_bt = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> + SOF_BT_OFFLOAD_SSP_SHIFT; + ctx->ssp_amp = sof_ssp_amp_quirk & SOF_AMPLIFIER_SSP_MASK; /* set number of dai links */ @@ -288,8 +277,10 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) ctx->hdmi_num = 0; } - if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) + if (sof_ssp_amp_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + ctx->bt_offload_present = true; sof_ssp_amp_card.num_links++; + } dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, ctx->ssp_amp, ctx->dmic_be_num, From 9fadbf3f11c378fa307517f20bd793eb9773a15e Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:48 +0200 Subject: [PATCH 114/337] ASoC: Intel: sof_ssp_amp: simplify HDMI-In quirks Use bit mask to handle SSP port number of HDMI-In devices to simplify the code. No functional change in this patch. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Balamurugan C Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-22-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 70 +++++++++++----------------- 1 file changed, 26 insertions(+), 44 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 72505f6a6501..eddefa69355d 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -27,21 +27,10 @@ #define SOF_AMPLIFIER_SSP_MASK (GENMASK(3, 0)) /* HDMI capture*/ -#define SOF_SSP_HDMI_CAPTURE_PRESENT BIT(4) -#define SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT 5 -#define SOF_NO_OF_HDMI_CAPTURE_SSP_MASK (GENMASK(6, 5)) -#define SOF_NO_OF_HDMI_CAPTURE_SSP(quirk) \ - (((quirk) << SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT) & SOF_NO_OF_HDMI_CAPTURE_SSP_MASK) - -#define SOF_HDMI_CAPTURE_1_SSP_SHIFT 7 -#define SOF_HDMI_CAPTURE_1_SSP_MASK (GENMASK(9, 7)) -#define SOF_HDMI_CAPTURE_1_SSP(quirk) \ - (((quirk) << SOF_HDMI_CAPTURE_1_SSP_SHIFT) & SOF_HDMI_CAPTURE_1_SSP_MASK) - -#define SOF_HDMI_CAPTURE_2_SSP_SHIFT 10 -#define SOF_HDMI_CAPTURE_2_SSP_MASK (GENMASK(12, 10)) -#define SOF_HDMI_CAPTURE_2_SSP(quirk) \ - (((quirk) << SOF_HDMI_CAPTURE_2_SSP_SHIFT) & SOF_HDMI_CAPTURE_2_SSP_MASK) +#define SOF_HDMI_CAPTURE_SSP_MASK_SHIFT 4 +#define SOF_HDMI_CAPTURE_SSP_MASK_MASK (GENMASK(9, 4)) +#define SOF_HDMI_CAPTURE_SSP_MASK(quirk) \ + (((quirk) << SOF_HDMI_CAPTURE_SSP_MASK_SHIFT) & SOF_HDMI_CAPTURE_SSP_MASK_MASK) /* HDMI playback */ #define SOF_HDMI_PLAYBACK_PRESENT BIT(13) @@ -108,6 +97,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, int ret; bool fixed_be = false; int be_id; + unsigned long ssp_mask_hdmi_in; links = devm_kcalloc(dev, sof_ssp_amp_card.num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL); @@ -117,20 +107,17 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, return NULL; /* HDMI-In SSP */ - if (sof_ssp_amp_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT) { - int num_of_hdmi_ssp = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_CAPTURE_SSP_MASK) >> - SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; + ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> + SOF_HDMI_CAPTURE_SSP_MASK_SHIFT; + + if (ssp_mask_hdmi_in) { + int port = 0; /* the topology supports HDMI-IN uses fixed BE ID for DAI links */ fixed_be = true; be_id = HDMI_IN_BE_ID; - for (i = 1; i <= num_of_hdmi_ssp; i++) { - int port = (i == 1 ? (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_1_SSP_MASK) >> - SOF_HDMI_CAPTURE_1_SSP_SHIFT : - (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_2_SSP_MASK) >> - SOF_HDMI_CAPTURE_2_SSP_SHIFT); - + for_each_set_bit(port, &ssp_mask_hdmi_in, 32) { links[id].cpus = &cpus[id]; links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); @@ -231,6 +218,7 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; int ret; + unsigned long ssp_mask_hdmi_in; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -246,7 +234,10 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) else ctx->dmic_be_num = 0; - /* port number of peripherals attached to ssp interface */ + /* port number/mask of peripherals attached to ssp interface */ + ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> + SOF_HDMI_CAPTURE_SSP_MASK_SHIFT; + ctx->ssp_bt = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> SOF_BT_OFFLOAD_SSP_SHIFT; @@ -258,9 +249,8 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) if (ctx->amp_type != CODEC_NONE) sof_ssp_amp_card.num_links++; - if (sof_ssp_amp_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT) - sof_ssp_amp_card.num_links += (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_CAPTURE_SSP_MASK) >> - SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; + if (ssp_mask_hdmi_in) + sof_ssp_amp_card.num_links += hweight32(ssp_mask_hdmi_in); if (sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT) { ctx->hdmi_num = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_PLAYBACK_MASK) >> @@ -325,10 +315,8 @@ static const struct platform_device_id board_ids[] = { { .name = "tgl_rt1308_hdmi_ssp", .driver_data = (kernel_ulong_t)(SOF_AMPLIFIER_SSP(2) | - SOF_NO_OF_HDMI_CAPTURE_SSP(2) | - SOF_HDMI_CAPTURE_1_SSP(1) | - SOF_HDMI_CAPTURE_2_SSP(5) | - SOF_SSP_HDMI_CAPTURE_PRESENT), + SOF_HDMI_CAPTURE_SSP_MASK(0x22)), + /* SSP 1 and SSP 5 are used for HDMI IN */ }, { .name = "adl_cs35l41", @@ -340,28 +328,22 @@ static const struct platform_device_id board_ids[] = { }, { .name = "adl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_NO_OF_HDMI_CAPTURE_SSP(2) | - SOF_HDMI_CAPTURE_1_SSP(0) | - SOF_HDMI_CAPTURE_2_SSP(2) | - SOF_SSP_HDMI_CAPTURE_PRESENT | + .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | + /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_NO_OF_HDMI_PLAYBACK(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "rpl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_NO_OF_HDMI_CAPTURE_SSP(2) | - SOF_HDMI_CAPTURE_1_SSP(0) | - SOF_HDMI_CAPTURE_2_SSP(2) | - SOF_SSP_HDMI_CAPTURE_PRESENT | + .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | + /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_NO_OF_HDMI_PLAYBACK(3) | SOF_HDMI_PLAYBACK_PRESENT), }, { .name = "mtl_lt6911_hdmi_ssp", - .driver_data = (kernel_ulong_t)(SOF_NO_OF_HDMI_CAPTURE_SSP(2) | - SOF_HDMI_CAPTURE_1_SSP(0) | - SOF_HDMI_CAPTURE_2_SSP(2) | - SOF_SSP_HDMI_CAPTURE_PRESENT | + .driver_data = (kernel_ulong_t)(SOF_HDMI_CAPTURE_SSP_MASK(0x5) | + /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_NO_OF_HDMI_PLAYBACK(3) | SOF_HDMI_PLAYBACK_PRESENT), }, From f7c015add5b1a6a94fb39420ac0a885c8d7b63ad Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:49 +0200 Subject: [PATCH 115/337] ASoC: Intel: board_helpers: support HDMI-In link initialization Add a helper function for machine drivers to initialize HDMI-In DAI link. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Balamurugan C Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-23-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 42 ++++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 5 +++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 335c660561d5..a1dba1f45669 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -333,6 +333,48 @@ int sof_intel_board_set_bt_link(struct device *dev, } EXPORT_SYMBOL_NS(sof_intel_board_set_bt_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); +int sof_intel_board_set_hdmi_in_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + int ssp_hdmi) +{ + struct snd_soc_dai_link_component *cpus; + + dev_dbg(dev, "link %d: hdmi-in, ssp %d\n", be_id, ssp_hdmi); + + /* link name */ + link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-HDMI", ssp_hdmi); + if (!link->name) + return -ENOMEM; + + /* cpus */ + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!cpus) + return -ENOMEM; + + cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_hdmi); + if (!cpus->dai_name) + return -ENOMEM; + + link->cpus = cpus; + link->num_cpus = 1; + + /* codecs */ + link->codecs = &snd_soc_dummy_dlc; + link->num_codecs = 1; + + /* platforms */ + link->platforms = platform_component; + link->num_platforms = ARRAY_SIZE(platform_component); + + link->id = be_id; + link->no_pcm = 1; + link->dpcm_capture = 1; + + return 0; +} +EXPORT_SYMBOL_NS(sof_intel_board_set_hdmi_in_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); + MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index 49a7bfa12a6d..a4bae62a4ed5 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -33,6 +33,7 @@ struct sof_rt5682_private { * @ssp_codec: ssp port number of headphone BE link * @ssp_amp: ssp port number of speaker BE link * @ssp_bt: ssp port number of BT offload BE link + * @ssp_mask_hdmi_in: ssp port mask of HDMI-IN BE link * @bt_offload_present: true to create BT offload BE link * @rt5682: private data for rt5682 machine driver */ @@ -49,6 +50,7 @@ struct sof_card_private { int ssp_codec; int ssp_amp; int ssp_bt; + unsigned long ssp_mask_hdmi_in; bool bt_offload_present; @@ -79,5 +81,8 @@ int sof_intel_board_set_ssp_amp_link(struct device *dev, int sof_intel_board_set_bt_link(struct device *dev, struct snd_soc_dai_link *link, int be_id, int ssp_bt); +int sof_intel_board_set_hdmi_in_link(struct device *dev, + struct snd_soc_dai_link *link, int be_id, + int ssp_hdmi); #endif /* __SOF_INTEL_BOARD_HELPERS_H */ From 426cbd0de2da2553de9c8a2366f956807f94d5b4 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:50 +0200 Subject: [PATCH 116/337] ASoC: Intel: sof_rt5682: use common module for HDMI-In link Use intel_board module for HDMI-In DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Balamurugan C Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-24-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 37 +++++++---------------------- 1 file changed, 9 insertions(+), 28 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 22dd85129a51..d353ad758c60 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -463,13 +463,6 @@ static struct snd_soc_ops sof_rt5682_ops = { .hw_params = sof_rt5682_hw_params, }; -static struct snd_soc_dai_link_component platform_component[] = { - { - /* name might be overridden during probe */ - .name = "0000:00:1f.3" - } -}; - static int sof_card_late_probe(struct snd_soc_card *card) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); @@ -576,18 +569,14 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, int ssp_amp, int dmic_be_num, int hdmi_num, bool idisp_codec, bool is_legacy_cpu) { - struct snd_soc_dai_link_component *cpus; struct snd_soc_dai_link *links; int i; int id = 0; int ret; - int hdmi_id_offset = 0; links = devm_kcalloc(dev, sof_audio_card_rt5682.num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL); - cpus = devm_kcalloc(dev, sof_audio_card_rt5682.num_links, - sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); - if (!links || !cpus) + if (!links) goto devm_err; /* codec SSP */ @@ -735,22 +724,11 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, int port = 0; for_each_set_bit(port, &hdmi_in_ssp, 32) { - links[id].cpus = &cpus[id]; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", port); - if (!links[id].cpus->dai_name) + ret = sof_intel_board_set_hdmi_in_link(dev, &links[id], + id, port); + if (ret) return NULL; - links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-HDMI", port); - if (!links[id].name) - return NULL; - links[id].id = id + hdmi_id_offset; - links[id].codecs = &snd_soc_dummy_dlc; - links[id].num_codecs = 1; - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].num_cpus = 1; + id++; } } @@ -831,7 +809,10 @@ static int sof_audio_probe(struct platform_device *pdev) dev_dbg(&pdev->dev, "sof_rt5682_quirk = %lx\n", sof_rt5682_quirk); - /* port number of peripherals attached to ssp interface */ + /* port number/mask of peripherals attached to ssp interface */ + ctx->ssp_mask_hdmi_in = (sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) >> + SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; + ctx->ssp_bt = (sof_rt5682_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> SOF_BT_OFFLOAD_SSP_SHIFT; From ee2486d5219e9f724372084d5926070697e84836 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:51 +0200 Subject: [PATCH 117/337] ASoC: Intel: sof_ssp_amp: use common module for HDMI-In link Use intel_board module for HDMI-In DAI link initialization. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Balamurugan C Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-25-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 38 ++++++---------------------- 1 file changed, 8 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index eddefa69355d..ee2e813bf4c0 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -71,13 +71,6 @@ static struct snd_soc_card sof_ssp_amp_card = { .late_probe = sof_card_late_probe, }; -static struct snd_soc_dai_link_component platform_component[] = { - { - /* name might be overridden during probe */ - .name = "0000:00:1f.3" - } -}; - /* BE ID defined in sof-tgl-rt1308-hdmi-ssp.m4 */ #define HDMI_IN_BE_ID 0 #define SPK_BE_ID 2 @@ -90,7 +83,6 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, int ssp_amp, int dmic_be_num, int hdmi_num, bool idisp_codec) { - struct snd_soc_dai_link_component *cpus; struct snd_soc_dai_link *links; int i; int id = 0; @@ -101,9 +93,7 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, links = devm_kcalloc(dev, sof_ssp_amp_card.num_links, sizeof(struct snd_soc_dai_link), GFP_KERNEL); - cpus = devm_kcalloc(dev, sof_ssp_amp_card.num_links, - sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); - if (!links || !cpus) + if (!links) return NULL; /* HDMI-In SSP */ @@ -118,22 +108,11 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, be_id = HDMI_IN_BE_ID; for_each_set_bit(port, &ssp_mask_hdmi_in, 32) { - links[id].cpus = &cpus[id]; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", port); - if (!links[id].cpus->dai_name) + ret = sof_intel_board_set_hdmi_in_link(dev, &links[id], + be_id, port); + if (ret) return NULL; - links[id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-HDMI", port); - if (!links[id].name) - return NULL; - links[id].id = be_id; - links[id].codecs = &snd_soc_dummy_dlc; - links[id].num_codecs = 1; - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].num_cpus = 1; + id++; be_id++; } @@ -218,7 +197,6 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; int ret; - unsigned long ssp_mask_hdmi_in; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -235,7 +213,7 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) ctx->dmic_be_num = 0; /* port number/mask of peripherals attached to ssp interface */ - ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> + ctx->ssp_mask_hdmi_in = (sof_ssp_amp_quirk & SOF_HDMI_CAPTURE_SSP_MASK_MASK) >> SOF_HDMI_CAPTURE_SSP_MASK_SHIFT; ctx->ssp_bt = (sof_ssp_amp_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> @@ -249,8 +227,8 @@ static int sof_ssp_amp_probe(struct platform_device *pdev) if (ctx->amp_type != CODEC_NONE) sof_ssp_amp_card.num_links++; - if (ssp_mask_hdmi_in) - sof_ssp_amp_card.num_links += hweight32(ssp_mask_hdmi_in); + if (ctx->ssp_mask_hdmi_in) + sof_ssp_amp_card.num_links += hweight32(ctx->ssp_mask_hdmi_in); if (sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT) { ctx->hdmi_num = (sof_ssp_amp_quirk & SOF_NO_OF_HDMI_PLAYBACK_MASK) >> From a6881210f175ae309721b3c76f54eebc8a258339 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:52 +0200 Subject: [PATCH 118/337] ASoC: Intel: board_helpers: support DAI link array generation Add a helper function for machine drivers to initialize dai_link and num_links of a snd_soc_card structure. Machine driver needs to initialize sof_card_private structure in driver probe function then board_helpers module will create entire DAI link array for this board. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-26-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 152 +++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 7 + 2 files changed, 159 insertions(+) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index a1dba1f45669..4f2cb8e52971 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -375,6 +375,158 @@ int sof_intel_board_set_hdmi_in_link(struct device *dev, } EXPORT_SYMBOL_NS(sof_intel_board_set_hdmi_in_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); +static int calculate_num_links(struct sof_card_private *ctx) +{ + int num_links = 0; + + /* headphone codec */ + if (ctx->codec_type != CODEC_NONE) + num_links++; + + /* dmic01 and dmic16k */ + if (ctx->dmic_be_num > 0) + num_links++; + + if (ctx->dmic_be_num > 1) + num_links++; + + /* idisp HDMI */ + num_links += ctx->hdmi_num; + + /* speaker amp */ + if (ctx->amp_type != CODEC_NONE) + num_links++; + + /* BT audio offload */ + if (ctx->bt_offload_present) + num_links++; + + /* HDMI-In */ + num_links += hweight32(ctx->ssp_mask_hdmi_in); + + return num_links; +} + +int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) +{ + struct snd_soc_dai_link *links; + int num_links; + int i; + int idx = 0; + int ret; + int ssp_hdmi_in = 0; + + num_links = calculate_num_links(ctx); + + links = devm_kcalloc(dev, num_links, sizeof(struct snd_soc_dai_link), + GFP_KERNEL); + if (!links) + return -ENOMEM; + + /* headphone codec */ + if (ctx->codec_type != CODEC_NONE) { + ret = sof_intel_board_set_codec_link(dev, &links[idx], idx, + ctx->codec_type, + ctx->ssp_codec); + if (ret) { + dev_err(dev, "fail to set codec link, ret %d\n", ret); + return ret; + } + + ctx->codec_link = &links[idx]; + idx++; + } + + /* dmic01 and dmic16k */ + if (ctx->dmic_be_num > 0) { + /* at least we have dmic01 */ + ret = sof_intel_board_set_dmic_link(dev, &links[idx], idx, + SOF_DMIC_01); + if (ret) { + dev_err(dev, "fail to set dmic01 link, ret %d\n", ret); + return ret; + } + + idx++; + } + + if (ctx->dmic_be_num > 1) { + /* set up 2 BE links at most */ + ret = sof_intel_board_set_dmic_link(dev, &links[idx], idx, + SOF_DMIC_16K); + if (ret) { + dev_err(dev, "fail to set dmic16k link, ret %d\n", ret); + return ret; + } + + idx++; + } + + /* idisp HDMI */ + for (i = 1; i <= ctx->hdmi_num; i++) { + ret = sof_intel_board_set_intel_hdmi_link(dev, &links[idx], idx, + i, + ctx->hdmi.idisp_codec); + if (ret) { + dev_err(dev, "fail to set hdmi link, ret %d\n", ret); + return ret; + } + + idx++; + } + + /* speaker amp */ + if (ctx->amp_type != CODEC_NONE) { + ret = sof_intel_board_set_ssp_amp_link(dev, &links[idx], idx, + ctx->amp_type, + ctx->ssp_amp); + if (ret) { + dev_err(dev, "fail to set amp link, ret %d\n", ret); + return ret; + } + + ctx->amp_link = &links[idx]; + idx++; + } + + /* BT audio offload */ + if (ctx->bt_offload_present) { + ret = sof_intel_board_set_bt_link(dev, &links[idx], idx, + ctx->ssp_bt); + if (ret) { + dev_err(dev, "fail to set bt link, ret %d\n", ret); + return ret; + } + + idx++; + } + + /* HDMI-In */ + for_each_set_bit(ssp_hdmi_in, &ctx->ssp_mask_hdmi_in, 32) { + ret = sof_intel_board_set_hdmi_in_link(dev, &links[idx], idx, + ssp_hdmi_in); + if (ret) { + dev_err(dev, "fail to set hdmi-in link, ret %d\n", ret); + return ret; + } + + idx++; + } + + if (idx != num_links) { + dev_err(dev, "link number mismatch, idx %d, num_links %d\n", idx, + num_links); + return -EINVAL; + } + + card->dai_link = links; + card->num_links = num_links; + + return 0; +} +EXPORT_SYMBOL_NS(sof_intel_board_set_dai_link, SND_SOC_INTEL_SOF_BOARD_HELPERS); + MODULE_DESCRIPTION("ASoC Intel SOF Machine Driver Board Helpers"); MODULE_AUTHOR("Brent Lu "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index a4bae62a4ed5..3b36058118ca 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -35,6 +35,8 @@ struct sof_rt5682_private { * @ssp_bt: ssp port number of BT offload BE link * @ssp_mask_hdmi_in: ssp port mask of HDMI-IN BE link * @bt_offload_present: true to create BT offload BE link + * @codec_link: pointer to headset codec dai link + * @amp_link: pointer to speaker amplifier dai link * @rt5682: private data for rt5682 machine driver */ struct sof_card_private { @@ -54,6 +56,9 @@ struct sof_card_private { bool bt_offload_present; + struct snd_soc_dai_link *codec_link; + struct snd_soc_dai_link *amp_link; + union { struct sof_rt5682_private rt5682; }; @@ -65,6 +70,8 @@ enum sof_dmic_be_type { }; int sof_intel_board_card_late_probe(struct snd_soc_card *card); +int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx); int sof_intel_board_set_codec_link(struct device *dev, struct snd_soc_dai_link *link, int be_id, From 5da85a3523b5acd3a00347cae32ae2ffe4f4ac8a Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:53 +0200 Subject: [PATCH 119/337] ASoC: Intel: sof_nau8825: use common module for DAI link generation Use intel_board module to generate DAI link array and update num_links field in snd_soc_card structure. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-27-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 166 ++++++++------------------- 1 file changed, 49 insertions(+), 117 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 32a43d1b5846..15ea6732ff94 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -200,123 +200,67 @@ static struct snd_soc_dai_link_component nau8825_component[] = { } }; -static struct snd_soc_dai_link * -sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec amp_type, - int ssp_codec, int ssp_amp, int dmic_be_num, - int hdmi_num, bool idisp_codec) +static int +sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) { - struct snd_soc_dai_link *links; - int i; - int id = 0; int ret; - links = devm_kcalloc(dev, sof_audio_card_nau8825.num_links, - sizeof(struct snd_soc_dai_link), GFP_KERNEL); - if (!links) - goto devm_err; - - /* codec SSP */ - ret = sof_intel_board_set_codec_link(dev, &links[id], id, CODEC_NAU8825, - ssp_codec); + ret = sof_intel_board_set_dai_link(dev, card, ctx); if (ret) - return NULL; + return ret; - /* codec-specific fields */ - links[id].codecs = nau8825_component; - links[id].num_codecs = ARRAY_SIZE(nau8825_component); - links[id].init = sof_nau8825_codec_init; - links[id].exit = sof_nau8825_codec_exit; - links[id].ops = &sof_nau8825_ops; - - id++; - - /* dmic */ - if (dmic_be_num > 0) { - /* at least we have dmic01 */ - ret = sof_intel_board_set_dmic_link(dev, &links[id], id, - SOF_DMIC_01); - if (ret) - return NULL; - - id++; + if (!ctx->codec_link) { + dev_err(dev, "codec link not available"); + return -EINVAL; } - if (dmic_be_num > 1) { - /* set up 2 BE links at most */ - ret = sof_intel_board_set_dmic_link(dev, &links[id], id, - SOF_DMIC_16K); - if (ret) - return NULL; + /* codec-specific fields for headphone codec */ + ctx->codec_link->codecs = nau8825_component; + ctx->codec_link->num_codecs = ARRAY_SIZE(nau8825_component); + ctx->codec_link->init = sof_nau8825_codec_init; + ctx->codec_link->exit = sof_nau8825_codec_exit; + ctx->codec_link->ops = &sof_nau8825_ops; - id++; + if (ctx->amp_type == CODEC_NONE) + return 0; + + if (!ctx->amp_link) { + dev_err(dev, "amp link not available"); + return -EINVAL; } - /* HDMI */ - for (i = 1; i <= hdmi_num; i++) { - ret = sof_intel_board_set_intel_hdmi_link(dev, &links[id], id, - i, idisp_codec); - if (ret) - return NULL; - - id++; + /* codec-specific fields for speaker amplifier */ + switch (ctx->amp_type) { + case CODEC_MAX98360A: + max_98360a_dai_link(ctx->amp_link); + break; + case CODEC_MAX98373: + ctx->amp_link->codecs = max_98373_components; + ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components); + ctx->amp_link->init = max_98373_spk_codec_init; + ctx->amp_link->ops = &max_98373_ops; + break; + case CODEC_NAU8318: + nau8318_set_dai_link(ctx->amp_link); + break; + case CODEC_RT1015P: + sof_rt1015p_dai_link(ctx->amp_link); + break; + case CODEC_RT1019P: + sof_rt1019p_dai_link(ctx->amp_link); + break; + default: + dev_err(dev, "invalid amp type %d\n", ctx->amp_type); + return -EINVAL; } - /* speaker amp */ - if (amp_type != CODEC_NONE) { - ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], id, - amp_type, ssp_amp); - if (ret) - return NULL; - - /* codec-specific fields */ - switch (amp_type) { - case CODEC_MAX98360A: - max_98360a_dai_link(&links[id]); - break; - case CODEC_MAX98373: - links[id].codecs = max_98373_components; - links[id].num_codecs = ARRAY_SIZE(max_98373_components); - links[id].init = max_98373_spk_codec_init; - links[id].ops = &max_98373_ops; - break; - case CODEC_NAU8318: - nau8318_set_dai_link(&links[id]); - break; - case CODEC_RT1015P: - sof_rt1015p_dai_link(&links[id]); - break; - case CODEC_RT1019P: - sof_rt1019p_dai_link(&links[id]); - break; - default: - dev_err(dev, "invalid amp type %d\n", amp_type); - return NULL; - } - - id++; - } - - /* BT audio offload */ - if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { - int port = (sof_nau8825_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ret = sof_intel_board_set_bt_link(dev, &links[id], id, port); - if (ret) - return NULL; - - id++; - } - - return links; -devm_err: - return NULL; + return 0; } static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; int ret; @@ -352,25 +296,13 @@ static int sof_audio_probe(struct platform_device *pdev) ctx->ssp_codec = sof_nau8825_quirk & SOF_NAU8825_SSP_CODEC_MASK; - /* compute number of dai links */ - sof_audio_card_nau8825.num_links = 1 + ctx->dmic_be_num + ctx->hdmi_num; - - if (ctx->amp_type != CODEC_NONE) - sof_audio_card_nau8825.num_links++; - - if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + if (sof_nau8825_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) ctx->bt_offload_present = true; - sof_audio_card_nau8825.num_links++; - } - dai_links = sof_card_dai_links_create(&pdev->dev, ctx->amp_type, - ctx->ssp_codec, ctx->ssp_amp, - ctx->dmic_be_num, ctx->hdmi_num, - ctx->hdmi.idisp_codec); - if (!dai_links) - return -ENOMEM; - - sof_audio_card_nau8825.dai_link = dai_links; + /* update dai_link */ + ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_nau8825, ctx); + if (ret) + return ret; /* update codec_conf */ switch (ctx->amp_type) { From 8fa1116e1cae4858ab69d31da36f1fae9113c29d Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 27 Nov 2023 17:26:54 +0200 Subject: [PATCH 120/337] ASoC: Intel: sof_rt5682: use common module for DAI link generation Use intel_board module to generate DAI link array and update num_links field in snd_soc_card structure. Signed-off-by: Brent Lu Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231127152654.28204-28-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 256 +++++++++------------------- 1 file changed, 82 insertions(+), 174 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index d353ad758c60..cd50f26d1edb 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -563,52 +563,45 @@ static struct snd_soc_dai_link_component rt5650_components[] = { } }; -static struct snd_soc_dai_link * -sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, - enum sof_ssp_codec amp_type, int ssp_codec, - int ssp_amp, int dmic_be_num, int hdmi_num, - bool idisp_codec, bool is_legacy_cpu) +static int +sof_card_dai_links_create(struct device *dev, struct snd_soc_card *card, + struct sof_card_private *ctx) { - struct snd_soc_dai_link *links; - int i; - int id = 0; int ret; - links = devm_kcalloc(dev, sof_audio_card_rt5682.num_links, - sizeof(struct snd_soc_dai_link), GFP_KERNEL); - if (!links) - goto devm_err; - - /* codec SSP */ - ret = sof_intel_board_set_codec_link(dev, &links[id], id, codec_type, - ssp_codec); + ret = sof_intel_board_set_dai_link(dev, card, ctx); if (ret) - return NULL; + return ret; - /* codec-specific fields */ - switch (codec_type) { - case CODEC_RT5650: - links[id].codecs = &rt5650_components[0]; - links[id].num_codecs = 1; - break; - case CODEC_RT5682: - links[id].codecs = rt5682_component; - links[id].num_codecs = ARRAY_SIZE(rt5682_component); - break; - case CODEC_RT5682S: - links[id].codecs = rt5682s_component; - links[id].num_codecs = ARRAY_SIZE(rt5682s_component); - break; - default: - dev_err(dev, "invalid codec type %d\n", codec_type); - return NULL; + if (!ctx->codec_link) { + dev_err(dev, "codec link not available"); + return -EINVAL; } - links[id].init = sof_rt5682_codec_init; - links[id].exit = sof_rt5682_codec_exit; - links[id].ops = &sof_rt5682_ops; + /* codec-specific fields for headphone codec */ + switch (ctx->codec_type) { + case CODEC_RT5650: + ctx->codec_link->codecs = &rt5650_components[0]; + ctx->codec_link->num_codecs = 1; + break; + case CODEC_RT5682: + ctx->codec_link->codecs = rt5682_component; + ctx->codec_link->num_codecs = ARRAY_SIZE(rt5682_component); + break; + case CODEC_RT5682S: + ctx->codec_link->codecs = rt5682s_component; + ctx->codec_link->num_codecs = ARRAY_SIZE(rt5682s_component); + break; + default: + dev_err(dev, "invalid codec type %d\n", ctx->codec_type); + return -EINVAL; + } - if (!is_legacy_cpu) { + ctx->codec_link->init = sof_rt5682_codec_init; + ctx->codec_link->exit = sof_rt5682_codec_exit; + ctx->codec_link->ops = &sof_rt5682_ops; + + if (!ctx->rt5682.is_legacy_cpu) { /* * Currently, On SKL+ platforms MCLK will be turned off in sof * runtime suspended, and it will go into runtime suspended @@ -618,130 +611,64 @@ sof_card_dai_links_create(struct device *dev, enum sof_ssp_codec codec_type, * avoid the noise. * It can be removed once we can control MCLK by driver. */ - links[id].ignore_pmdown_time = 1; - } - id++; - - /* dmic */ - if (dmic_be_num > 0) { - /* at least we have dmic01 */ - ret = sof_intel_board_set_dmic_link(dev, &links[id], id, - SOF_DMIC_01); - if (ret) - return NULL; - - id++; + ctx->codec_link->ignore_pmdown_time = 1; } - if (dmic_be_num > 1) { - /* set up 2 BE links at most */ - ret = sof_intel_board_set_dmic_link(dev, &links[id], id, - SOF_DMIC_16K); - if (ret) - return NULL; + if (ctx->amp_type == CODEC_NONE) + return 0; - id++; + if (!ctx->amp_link) { + dev_err(dev, "amp link not available"); + return -EINVAL; } - /* HDMI */ - for (i = 1; i <= hdmi_num; i++) { - ret = sof_intel_board_set_intel_hdmi_link(dev, &links[id], id, - i, idisp_codec); - if (ret) - return NULL; - - id++; + /* codec-specific fields for speaker amplifier */ + switch (ctx->amp_type) { + case CODEC_MAX98357A: + max_98357a_dai_link(ctx->amp_link); + break; + case CODEC_MAX98360A: + max_98360a_dai_link(ctx->amp_link); + break; + case CODEC_MAX98373: + ctx->amp_link->codecs = max_98373_components; + ctx->amp_link->num_codecs = ARRAY_SIZE(max_98373_components); + ctx->amp_link->init = max_98373_spk_codec_init; + ctx->amp_link->ops = &max_98373_ops; + break; + case CODEC_MAX98390: + max_98390_dai_link(dev, ctx->amp_link); + break; + case CODEC_RT1011: + sof_rt1011_dai_link(ctx->amp_link); + break; + case CODEC_RT1015: + sof_rt1015_dai_link(ctx->amp_link); + break; + case CODEC_RT1015P: + sof_rt1015p_dai_link(ctx->amp_link); + break; + case CODEC_RT1019P: + sof_rt1019p_dai_link(ctx->amp_link); + break; + case CODEC_RT5650: + /* use AIF2 to support speaker pipeline */ + ctx->amp_link->codecs = &rt5650_components[1]; + ctx->amp_link->num_codecs = 1; + ctx->amp_link->init = rt5650_spk_init; + ctx->amp_link->ops = &sof_rt5682_ops; + break; + default: + dev_err(dev, "invalid amp type %d\n", ctx->amp_type); + return -EINVAL; } - /* speaker amp */ - if (amp_type != CODEC_NONE) { - ret = sof_intel_board_set_ssp_amp_link(dev, &links[id], id, - amp_type, ssp_amp); - if (ret) - return NULL; - - /* codec-specific fields */ - switch (amp_type) { - case CODEC_MAX98357A: - max_98357a_dai_link(&links[id]); - break; - case CODEC_MAX98360A: - max_98360a_dai_link(&links[id]); - break; - case CODEC_MAX98373: - links[id].codecs = max_98373_components; - links[id].num_codecs = ARRAY_SIZE(max_98373_components); - links[id].init = max_98373_spk_codec_init; - links[id].ops = &max_98373_ops; - break; - case CODEC_MAX98390: - max_98390_dai_link(dev, &links[id]); - break; - case CODEC_RT1011: - sof_rt1011_dai_link(&links[id]); - break; - case CODEC_RT1015: - sof_rt1015_dai_link(&links[id]); - break; - case CODEC_RT1015P: - sof_rt1015p_dai_link(&links[id]); - break; - case CODEC_RT1019P: - sof_rt1019p_dai_link(&links[id]); - break; - case CODEC_RT5650: - /* use AIF2 to support speaker pipeline */ - links[id].codecs = &rt5650_components[1]; - links[id].num_codecs = 1; - links[id].init = rt5650_spk_init; - links[id].ops = &sof_rt5682_ops; - break; - default: - dev_err(dev, "invalid amp type %d\n", amp_type); - return NULL; - } - - id++; - } - - /* BT audio offload */ - if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { - int port = (sof_rt5682_quirk & SOF_BT_OFFLOAD_SSP_MASK) >> - SOF_BT_OFFLOAD_SSP_SHIFT; - - ret = sof_intel_board_set_bt_link(dev, &links[id], id, port); - if (ret) - return NULL; - - id++; - } - - /* HDMI-In SSP */ - if (sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) { - unsigned long hdmi_in_ssp = (sof_rt5682_quirk & - SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) >> - SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT; - int port = 0; - - for_each_set_bit(port, &hdmi_in_ssp, 32) { - ret = sof_intel_board_set_hdmi_in_link(dev, &links[id], - id, port); - if (ret) - return NULL; - - id++; - } - } - - return links; -devm_err: - return NULL; + return 0; } static int sof_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct snd_soc_dai_link *dai_links; struct sof_card_private *ctx; int ret; @@ -821,32 +748,13 @@ static int sof_audio_probe(struct platform_device *pdev) ctx->ssp_codec = sof_rt5682_quirk & SOF_RT5682_SSP_CODEC_MASK; - /* compute number of dai links */ - sof_audio_card_rt5682.num_links = 1 + ctx->dmic_be_num + ctx->hdmi_num; - - if (ctx->amp_type != CODEC_NONE) - sof_audio_card_rt5682.num_links++; - - if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) { + if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) ctx->bt_offload_present = true; - sof_audio_card_rt5682.num_links++; - } - if (sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) - sof_audio_card_rt5682.num_links += - hweight32((sof_rt5682_quirk & SOF_SSP_HDMI_CAPTURE_PRESENT_MASK) >> - SOF_NO_OF_HDMI_CAPTURE_SSP_SHIFT); - - dai_links = sof_card_dai_links_create(&pdev->dev, ctx->codec_type, - ctx->amp_type, ctx->ssp_codec, - ctx->ssp_amp, ctx->dmic_be_num, - ctx->hdmi_num, - ctx->hdmi.idisp_codec, - ctx->rt5682.is_legacy_cpu); - if (!dai_links) - return -ENOMEM; - - sof_audio_card_rt5682.dai_link = dai_links; + /* update dai_link */ + ret = sof_card_dai_links_create(&pdev->dev, &sof_audio_card_rt5682, ctx); + if (ret) + return ret; /* update codec_conf */ switch (ctx->amp_type) { From 8c91ca76f44804868d12aed20ebdbc2f89aa7d60 Mon Sep 17 00:00:00 2001 From: Kamil Duljas Date: Thu, 16 Nov 2023 23:01:03 +0100 Subject: [PATCH 121/337] ASoC: SOF: icp3-dtrace: Fix wrong kfree() usage trace_filter_parse() allocs memory for *out and when -ENOMEM is returned, caller function, dfsentry_trace_filter_write() trying to freed this memory. After this patch, the memory is freed in trace_filter_parse() before -EINVAL returned. In caller function removed kfree(elms) from error label Signed-off-by: Kamil Duljas Link: https://lore.kernel.org/r/20231116220102.2097-2-kamil.duljas@gmail.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-dtrace.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3-dtrace.c b/sound/soc/sof/ipc3-dtrace.c index 0dca139322f3..93b189c2d2ee 100644 --- a/sound/soc/sof/ipc3-dtrace.c +++ b/sound/soc/sof/ipc3-dtrace.c @@ -137,6 +137,7 @@ static int trace_filter_parse(struct snd_sof_dev *sdev, char *string, dev_err(sdev->dev, "Parsing filter entry '%s' failed with %d\n", entry, entry_len); + kfree(*out); return -EINVAL; } } @@ -208,13 +209,13 @@ static ssize_t dfsentry_trace_filter_write(struct file *file, const char __user ret = ipc3_trace_update_filter(sdev, num_elems, elems); if (ret < 0) { dev_err(sdev->dev, "Filter update failed: %d\n", ret); + kfree(elems); goto error; } } ret = count; error: kfree(string); - kfree(elems); return ret; } From 014fdeb0d747304111cfecf93df4407c1a0c80db Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:15 +0200 Subject: [PATCH 122/337] ASoC: SOF: Move sof_of_machine_select() to sof-of-dev.c from sof-audio.c Move the sof_of_machine_select() function to sof-of-dev.c file and provide an inline stub in case of non OF builds. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.c | 22 ---------------------- sound/soc/sof/sof-of-dev.c | 23 +++++++++++++++++++++++ sound/soc/sof/sof-of-dev.h | 9 +++++++++ 3 files changed, 32 insertions(+), 22 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 563fe6f7789f..2198bb8e92c3 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -990,28 +990,6 @@ int sof_dai_get_bclk(struct snd_soc_pcm_runtime *rtd) } EXPORT_SYMBOL(sof_dai_get_bclk); -static struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) -{ - struct snd_sof_pdata *sof_pdata = sdev->pdata; - const struct sof_dev_desc *desc = sof_pdata->desc; - struct snd_sof_of_mach *mach = desc->of_machines; - - if (!mach) - return NULL; - - for (; mach->compatible; mach++) { - if (of_machine_is_compatible(mach->compatible)) { - sof_pdata->tplg_filename = mach->sof_tplg_filename; - if (mach->fw_filename) - sof_pdata->fw_filename = mach->fw_filename; - - return mach; - } - } - - return NULL; -} - /* * SOF Driver enumeration. */ diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index c6be8a91e74b..432b511bf8c4 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -41,6 +41,29 @@ static void sof_of_probe_complete(struct device *dev) pm_runtime_enable(dev); } +struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *sof_pdata = sdev->pdata; + const struct sof_dev_desc *desc = sof_pdata->desc; + struct snd_sof_of_mach *mach = desc->of_machines; + + if (!mach) + return NULL; + + for (; mach->compatible; mach++) { + if (of_machine_is_compatible(mach->compatible)) { + sof_pdata->tplg_filename = mach->sof_tplg_filename; + if (mach->fw_filename) + sof_pdata->fw_filename = mach->fw_filename; + + return mach; + } + } + + return NULL; +} +EXPORT_SYMBOL(sof_of_machine_select); + int sof_of_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; diff --git a/sound/soc/sof/sof-of-dev.h b/sound/soc/sof/sof-of-dev.h index b6cc70595f3b..547e358a37e3 100644 --- a/sound/soc/sof/sof-of-dev.h +++ b/sound/soc/sof/sof-of-dev.h @@ -16,6 +16,15 @@ struct snd_sof_of_mach { const char *sof_tplg_filename; }; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_OF_DEV) +struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev); +#else +static inline struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) +{ + return NULL; +} +#endif + extern const struct dev_pm_ops sof_of_pm; int sof_of_probe(struct platform_device *pdev); From 3bc3477915587440035f192c89a1bf9a4360abb3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:16 +0200 Subject: [PATCH 123/337] ASoC: SOF: Move sof_machine_* functions from sof-audio.c to core.c Relocate the machine handling functions from sof-audio.c to core.c to maintain code separation. While doing the move, drop the redundant IS_ERR_OR_NULL(plat_data->pdev_mach) check from sof_machine_unregister() Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 95 +++++++++++++++++++++++++++++++++++++ sound/soc/sof/sof-audio.c | 98 --------------------------------------- sound/soc/sof/sof-priv.h | 2 - 3 files changed, 95 insertions(+), 100 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index d7b090224f1b..a87522ea4301 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -13,6 +13,7 @@ #include #include #include "sof-priv.h" +#include "sof-of-dev.h" #include "ops.h" #define CREATE_TRACE_POINTS @@ -143,6 +144,66 @@ void sof_set_fw_state(struct snd_sof_dev *sdev, enum sof_fw_state new_state) } EXPORT_SYMBOL(sof_set_fw_state); +/* SOF Driver enumeration */ +static int sof_machine_check(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *sof_pdata = sdev->pdata; + const struct sof_dev_desc *desc = sof_pdata->desc; + struct snd_soc_acpi_mach *mach; + + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE)) { + const struct snd_sof_of_mach *of_mach; + + if (IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC_DEBUG_SUPPORT) && + sof_debug_check_flag(SOF_DBG_FORCE_NOCODEC)) + goto nocodec; + + /* find machine */ + mach = snd_sof_machine_select(sdev); + if (mach) { + sof_pdata->machine = mach; + + if (sof_pdata->subsystem_id_set) { + mach->mach_params.subsystem_vendor = sof_pdata->subsystem_vendor; + mach->mach_params.subsystem_device = sof_pdata->subsystem_device; + mach->mach_params.subsystem_id_set = true; + } + + snd_sof_set_mach_params(mach, sdev); + return 0; + } + + of_mach = sof_of_machine_select(sdev); + if (of_mach) { + sof_pdata->of_machine = of_mach; + return 0; + } + + if (!IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC)) { + dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); + return -ENODEV; + } + } else { + dev_warn(sdev->dev, "Force to use nocodec mode\n"); + } + +nocodec: + /* select nocodec mode */ + dev_warn(sdev->dev, "Using nocodec machine driver\n"); + mach = devm_kzalloc(sdev->dev, sizeof(*mach), GFP_KERNEL); + if (!mach) + return -ENOMEM; + + mach->drv_name = "sof-nocodec"; + if (!sof_pdata->tplg_filename) + sof_pdata->tplg_filename = desc->nocodec_tplg_filename; + + sof_pdata->machine = mach; + snd_sof_set_mach_params(mach, sdev); + + return 0; +} + /* * FW Boot State Transition Diagram * @@ -527,6 +588,40 @@ int snd_sof_device_shutdown(struct device *dev) } EXPORT_SYMBOL(snd_sof_device_shutdown); +/* Machine driver registering and unregistering */ +int sof_machine_register(struct snd_sof_dev *sdev, void *pdata) +{ + struct snd_sof_pdata *plat_data = pdata; + const char *drv_name; + const void *mach; + int size; + + drv_name = plat_data->machine->drv_name; + mach = plat_data->machine; + size = sizeof(*plat_data->machine); + + /* register machine driver, pass machine info as pdata */ + plat_data->pdev_mach = + platform_device_register_data(sdev->dev, drv_name, + PLATFORM_DEVID_NONE, mach, size); + if (IS_ERR(plat_data->pdev_mach)) + return PTR_ERR(plat_data->pdev_mach); + + dev_dbg(sdev->dev, "created machine %s\n", + dev_name(&plat_data->pdev_mach->dev)); + + return 0; +} +EXPORT_SYMBOL(sof_machine_register); + +void sof_machine_unregister(struct snd_sof_dev *sdev, void *pdata) +{ + struct snd_sof_pdata *plat_data = pdata; + + platform_device_unregister(plat_data->pdev_mach); +} +EXPORT_SYMBOL(sof_machine_unregister); + MODULE_AUTHOR("Liam Girdwood"); MODULE_DESCRIPTION("Sound Open Firmware (SOF) Core"); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 2198bb8e92c3..de40a5e227f4 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -11,7 +11,6 @@ #include #include #include "sof-audio.h" -#include "sof-of-dev.h" #include "ops.h" static bool is_virtual_widget(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget *widget, @@ -989,100 +988,3 @@ int sof_dai_get_bclk(struct snd_soc_pcm_runtime *rtd) return sof_dai_get_clk(rtd, SOF_DAI_CLK_INTEL_SSP_BCLK); } EXPORT_SYMBOL(sof_dai_get_bclk); - -/* - * SOF Driver enumeration. - */ -int sof_machine_check(struct snd_sof_dev *sdev) -{ - struct snd_sof_pdata *sof_pdata = sdev->pdata; - const struct sof_dev_desc *desc = sof_pdata->desc; - struct snd_soc_acpi_mach *mach; - - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_FORCE_NOCODEC_MODE)) { - const struct snd_sof_of_mach *of_mach; - - if (IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC_DEBUG_SUPPORT) && - sof_debug_check_flag(SOF_DBG_FORCE_NOCODEC)) - goto nocodec; - - /* find machine */ - mach = snd_sof_machine_select(sdev); - if (mach) { - sof_pdata->machine = mach; - - if (sof_pdata->subsystem_id_set) { - mach->mach_params.subsystem_vendor = sof_pdata->subsystem_vendor; - mach->mach_params.subsystem_device = sof_pdata->subsystem_device; - mach->mach_params.subsystem_id_set = true; - } - - snd_sof_set_mach_params(mach, sdev); - return 0; - } - - of_mach = sof_of_machine_select(sdev); - if (of_mach) { - sof_pdata->of_machine = of_mach; - return 0; - } - - if (!IS_ENABLED(CONFIG_SND_SOC_SOF_NOCODEC)) { - dev_err(sdev->dev, "error: no matching ASoC machine driver found - aborting probe\n"); - return -ENODEV; - } - } else { - dev_warn(sdev->dev, "Force to use nocodec mode\n"); - } - -nocodec: - /* select nocodec mode */ - dev_warn(sdev->dev, "Using nocodec machine driver\n"); - mach = devm_kzalloc(sdev->dev, sizeof(*mach), GFP_KERNEL); - if (!mach) - return -ENOMEM; - - mach->drv_name = "sof-nocodec"; - if (!sof_pdata->tplg_filename) - sof_pdata->tplg_filename = desc->nocodec_tplg_filename; - - sof_pdata->machine = mach; - snd_sof_set_mach_params(mach, sdev); - - return 0; -} -EXPORT_SYMBOL(sof_machine_check); - -int sof_machine_register(struct snd_sof_dev *sdev, void *pdata) -{ - struct snd_sof_pdata *plat_data = pdata; - const char *drv_name; - const void *mach; - int size; - - drv_name = plat_data->machine->drv_name; - mach = plat_data->machine; - size = sizeof(*plat_data->machine); - - /* register machine driver, pass machine info as pdata */ - plat_data->pdev_mach = - platform_device_register_data(sdev->dev, drv_name, - PLATFORM_DEVID_NONE, mach, size); - if (IS_ERR(plat_data->pdev_mach)) - return PTR_ERR(plat_data->pdev_mach); - - dev_dbg(sdev->dev, "created machine %s\n", - dev_name(&plat_data->pdev_mach->dev)); - - return 0; -} -EXPORT_SYMBOL(sof_machine_register); - -void sof_machine_unregister(struct snd_sof_dev *sdev, void *pdata) -{ - struct snd_sof_pdata *plat_data = pdata; - - if (!IS_ERR_OR_NULL(plat_data->pdev_mach)) - platform_device_unregister(plat_data->pdev_mach); -} -EXPORT_SYMBOL(sof_machine_unregister); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index f4185012eb69..faa8a19ed737 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -814,8 +814,6 @@ int sof_stream_pcm_open(struct snd_sof_dev *sdev, int sof_stream_pcm_close(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); -int sof_machine_check(struct snd_sof_dev *sdev); - /* SOF client support */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_CLIENT) int sof_client_dev_register(struct snd_sof_dev *sdev, const char *name, u32 id, From 1162d267eabd6392d0d07bc88b75056e88042fc0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:17 +0200 Subject: [PATCH 124/337] ASoC: SOF: Add placeholder for platform IPC type and path overrides Add a struct sof_loadable_file_profile which can be filled by platforms (sof-acpi-dev.c, sof-of-dev.c and sof-acpi-dev.c) to be able to use common, generic code to handle path customization. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/include/sound/sof.h b/include/sound/sof.h index 268d0ca0f69f..05213bb515a3 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -57,6 +57,18 @@ enum sof_ipc_type { SOF_IPC_TYPE_COUNT }; +struct sof_loadable_file_profile { + enum sof_ipc_type ipc_type; + + const char *fw_path; + const char *fw_path_postfix; + const char *fw_name; + const char *fw_lib_path; + const char *fw_lib_path_postfix; + const char *tplg_path; + const char *tplg_name; +}; + /* * SOF Platform data. */ @@ -86,6 +98,9 @@ struct snd_sof_pdata { /* descriptor */ const struct sof_dev_desc *desc; + /* platform's preferred IPC type and path overrides */ + struct sof_loadable_file_profile ipc_file_profile_base; + /* firmware and topology filenames */ const char *fw_filename_prefix; const char *fw_filename; From a07625dcaf994ab3c5a6a456624719df05ed8ad6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:18 +0200 Subject: [PATCH 125/337] ASoC: SOF: sof-acpi-dev: Save the default IPC type and path overrides Store the default IPC type and the firmware and topology path overrides to ipc_file_profile_base Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-acpi-dev.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/sof-acpi-dev.c b/sound/soc/sof/sof-acpi-dev.c index 84a4a0a3318e..87c0c2edc4c9 100644 --- a/sound/soc/sof/sof-acpi-dev.c +++ b/sound/soc/sof/sof-acpi-dev.c @@ -87,6 +87,10 @@ int sof_acpi_probe(struct platform_device *pdev, const struct sof_dev_desc *desc else sof_pdata->tplg_filename_prefix = desc->default_tplg_path[SOF_IPC_TYPE_3]; + sof_pdata->ipc_file_profile_base.ipc_type = desc->ipc_default; + sof_pdata->ipc_file_profile_base.fw_path = fw_path; + sof_pdata->ipc_file_profile_base.tplg_path = tplg_path; + /* set callback to be called on successful device probe to enable runtime_pm */ sof_pdata->sof_probe_complete = sof_acpi_probe_complete; From 396016d56da4ad30fe9ff736e44d9e1457484805 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:19 +0200 Subject: [PATCH 126/337] ASoC: SOF: sof-of-dev: Save the default IPC type and path overrides Store the default IPC type and the firmware and topology path overrides to ipc_file_profile_base Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-of-dev.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 432b511bf8c4..7b58f45790f7 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -99,6 +99,10 @@ int sof_of_probe(struct platform_device *pdev) else sof_pdata->tplg_filename_prefix = desc->default_tplg_path[SOF_IPC_TYPE_3]; + sof_pdata->ipc_file_profile_base.ipc_type = desc->ipc_default; + sof_pdata->ipc_file_profile_base.fw_path = fw_path; + sof_pdata->ipc_file_profile_base.tplg_path = tplg_path; + /* set callback to be called on successful device probe to enable runtime_pm */ sof_pdata->sof_probe_complete = sof_of_probe_complete; From 59ddeae037b81303063bcf62b70fb33841b3f89e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:20 +0200 Subject: [PATCH 127/337] ASoC: SOF: sof-pci-dev: Save the default IPC type and path overrides Store the default IPC type and the overrides to ipc_file_profile_base Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-7-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-pci-dev.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 64b326e3ef85..becc85b27d51 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -190,6 +190,7 @@ static void sof_pci_probe_complete(struct device *dev) int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { + struct sof_loadable_file_profile *path_override; struct device *dev = &pci->dev; const struct sof_dev_desc *desc = (const struct sof_dev_desc *)pci_id->driver_data; @@ -334,6 +335,20 @@ int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) sof_pdata->tplg_filename = sof_dmi_override_tplg_name; } + path_override = &sof_pdata->ipc_file_profile_base; + path_override->ipc_type = sof_pdata->ipc_type; + path_override->fw_path = fw_path; + path_override->fw_name = fw_filename; + path_override->fw_lib_path = lib_path; + path_override->tplg_path = tplg_path; + path_override->tplg_name = sof_pdata->tplg_filename; + + if (dmi_check_system(community_key_platforms) && + sof_dmi_use_community_key) { + path_override->fw_path_postfix = "community"; + path_override->fw_lib_path_postfix = "community"; + } + /* set callback to be called on successful device probe to enable runtime_pm */ sof_pdata->sof_probe_complete = sof_pci_probe_complete; From b1a4ee9fd5a2dfb0f23abe58377f816915ec14ba Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:21 +0200 Subject: [PATCH 128/337] ASoC: SOF: core: Implement firmware, topology path setup in core Use the information stored in ipc_file_profile_base by platforms to construct the paths, filenames that are going to be used to load the firmware and topology files. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-8-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Makefile | 3 +- sound/soc/sof/core.c | 70 +++++++++++++++-- sound/soc/sof/fw-file-profile.c | 130 ++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 7 ++ 4 files changed, 203 insertions(+), 7 deletions(-) create mode 100644 sound/soc/sof/fw-file-profile.c diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index ef6fd43d0b72..3624124575af 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -1,7 +1,8 @@ # SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ - control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o + control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o\ + fw-file-profile.o # IPC implementations ifneq ($(CONFIG_SND_SOC_SOF_IPC3),) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index a87522ea4301..f1a083de9f9e 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -204,6 +204,67 @@ nocodec: return 0; } +static int sof_select_ipc_and_paths(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + struct sof_loadable_file_profile *base_profile = &plat_data->ipc_file_profile_base; + struct sof_loadable_file_profile out_profile; + struct device *dev = sdev->dev; + int ret; + + /* check IPC support */ + if (!(BIT(base_profile->ipc_type) & plat_data->desc->ipc_supported_mask)) { + dev_err(dev, + "ipc_type %d is not supported on this platform, mask is %#x\n", + base_profile->ipc_type, plat_data->desc->ipc_supported_mask); + return -EINVAL; + } + + if (base_profile->ipc_type != plat_data->desc->ipc_default) + dev_info(dev, + "Module parameter used, overriding default IPC %d to %d\n", + plat_data->desc->ipc_default, base_profile->ipc_type); + + if (base_profile->fw_path) + dev_dbg(dev, "Module parameter used, changed fw path to %s\n", + base_profile->fw_path); + else if (base_profile->fw_path_postfix) + dev_dbg(dev, "Path postfix appended to default fw path: %s\n", + base_profile->fw_path_postfix); + + if (base_profile->fw_lib_path) + dev_dbg(dev, "Module parameter used, changed fw_lib path to %s\n", + base_profile->fw_lib_path); + else if (base_profile->fw_lib_path_postfix) + dev_dbg(dev, "Path postfix appended to default fw_lib path: %s\n", + base_profile->fw_lib_path_postfix); + + if (base_profile->fw_name) + dev_dbg(dev, "Module parameter used, changed fw filename to %s\n", + base_profile->fw_name); + + if (base_profile->tplg_path) + dev_dbg(dev, "Module parameter used, changed tplg path to %s\n", + base_profile->tplg_path); + + if (base_profile->tplg_name) + dev_dbg(dev, "Module parameter used, changed tplg name to %s\n", + base_profile->tplg_name); + + ret = sof_create_ipc_file_profile(sdev, base_profile, &out_profile); + if (ret) + return ret; + + plat_data->ipc_type = out_profile.ipc_type; + plat_data->fw_filename = out_profile.fw_name; + plat_data->fw_filename_prefix = out_profile.fw_path; + plat_data->fw_lib_prefix = out_profile.fw_lib_path; + plat_data->tplg_filename_prefix = out_profile.tplg_path; + plat_data->tplg_filename = out_profile.tplg_name; + + return 0; +} + /* * FW Boot State Transition Diagram * @@ -442,12 +503,9 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) } } - /* check IPC support */ - if (!(BIT(plat_data->ipc_type) & plat_data->desc->ipc_supported_mask)) { - dev_err(dev, "ipc_type %d is not supported on this platform, mask is %#x\n", - plat_data->ipc_type, plat_data->desc->ipc_supported_mask); - return -EINVAL; - } + ret = sof_select_ipc_and_paths(sdev); + if (ret) + return ret; /* init ops, if necessary */ ret = sof_ops_init(sdev); diff --git a/sound/soc/sof/fw-file-profile.c b/sound/soc/sof/fw-file-profile.c new file mode 100644 index 000000000000..58b55516049e --- /dev/null +++ b/sound/soc/sof/fw-file-profile.c @@ -0,0 +1,130 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2023 Intel Corporation. All rights reserved. +// + +#include +#include "sof-priv.h" + +static int +sof_file_profile_for_ipc_type(struct device *dev, + const struct sof_dev_desc *desc, + struct sof_loadable_file_profile *base_profile, + struct sof_loadable_file_profile *out_profile) +{ + enum sof_ipc_type ipc_type = base_profile->ipc_type; + bool fw_lib_path_allocated = false; + bool fw_path_allocated = false; + int ret = 0; + + /* firmware path */ + if (base_profile->fw_path) { + out_profile->fw_path = base_profile->fw_path; + } else if (base_profile->fw_path_postfix) { + out_profile->fw_path = devm_kasprintf(dev, GFP_KERNEL, "%s/%s", + desc->default_fw_path[ipc_type], + base_profile->fw_path_postfix); + if (!out_profile->fw_path) + return -ENOMEM; + + fw_path_allocated = true; + } else { + out_profile->fw_path = desc->default_fw_path[ipc_type]; + } + + /* firmware filename */ + if (base_profile->fw_name) + out_profile->fw_name = base_profile->fw_name; + else + out_profile->fw_name = desc->default_fw_filename[ipc_type]; + + + /* firmware library path */ + if (base_profile->fw_lib_path) { + out_profile->fw_lib_path = base_profile->fw_lib_path; + } else if (desc->default_lib_path[ipc_type]) { + if (base_profile->fw_lib_path_postfix) { + out_profile->fw_lib_path = devm_kasprintf(dev, + GFP_KERNEL, "%s/%s", + desc->default_lib_path[ipc_type], + base_profile->fw_lib_path_postfix); + if (!out_profile->fw_lib_path) { + ret = -ENOMEM; + goto out; + } + + fw_lib_path_allocated = true; + } else { + out_profile->fw_lib_path = desc->default_lib_path[ipc_type]; + } + } + + if (base_profile->fw_path_postfix) + out_profile->fw_path_postfix = base_profile->fw_path_postfix; + + if (base_profile->fw_lib_path_postfix) + out_profile->fw_lib_path_postfix = base_profile->fw_lib_path_postfix; + + /* topology path */ + if (base_profile->tplg_path) + out_profile->tplg_path = base_profile->tplg_path; + else + out_profile->tplg_path = desc->default_tplg_path[ipc_type]; + + /* topology name */ + if (base_profile->tplg_name) + out_profile->tplg_name = base_profile->tplg_name; + + out_profile->ipc_type = ipc_type; + +out: + if (ret) { + /* Free up path strings created with devm_kasprintf */ + if (fw_path_allocated) + devm_kfree(dev, out_profile->fw_path); + if (fw_lib_path_allocated) + devm_kfree(dev, out_profile->fw_lib_path); + + memset(out_profile, 0, sizeof(*out_profile)); + } + + return ret; +} + +static void sof_print_profile_info(struct device *dev, + struct sof_loadable_file_profile *profile) +{ + dev_info(dev, "Firmware paths/files for ipc type %d:\n", profile->ipc_type); + + dev_info(dev, " Firmware file: %s/%s\n", profile->fw_path, profile->fw_name); + if (profile->fw_lib_path) + dev_info(dev, " Firmware lib path: %s\n", profile->fw_lib_path); + if (profile->tplg_name) + dev_info(dev, " Topology file: %s/%s\n", profile->tplg_path, + profile->tplg_name); + else + dev_info(dev, " Topology path: %s\n", profile->tplg_path); +} + +int sof_create_ipc_file_profile(struct snd_sof_dev *sdev, + struct sof_loadable_file_profile *base_profile, + struct sof_loadable_file_profile *out_profile) +{ + const struct sof_dev_desc *desc = sdev->pdata->desc; + struct device *dev = sdev->dev; + int ret; + + memset(out_profile, 0, sizeof(*out_profile)); + + ret = sof_file_profile_for_ipc_type(dev, desc, base_profile, out_profile); + if (ret) + return ret; + + sof_print_profile_info(dev, out_profile); + + return 0; +} +EXPORT_SYMBOL(sof_create_ipc_file_profile); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index faa8a19ed737..6d7897bf9607 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -695,6 +695,13 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev); */ extern struct snd_compress_ops sof_compressed_ops; +/* + * Firmware (firmware, libraries, topologies) file location + */ +int sof_create_ipc_file_profile(struct snd_sof_dev *sdev, + struct sof_loadable_file_profile *base_profile, + struct sof_loadable_file_profile *out_profile); + /* * Firmware loading. */ From b2b0bba36f0a81743e7143a8801ca75d2238bdac Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:22 +0200 Subject: [PATCH 129/337] ASoC: SOF: sof-acpi-dev: Rely on core to create the file paths The core is now using the information from ipc_file_profile_base to create the paths for the loadable files, no need to set it in here anymore. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-9-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-acpi-dev.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/sof/sof-acpi-dev.c b/sound/soc/sof/sof-acpi-dev.c index 87c0c2edc4c9..2977f0a63fba 100644 --- a/sound/soc/sof/sof-acpi-dev.c +++ b/sound/soc/sof/sof-acpi-dev.c @@ -74,18 +74,6 @@ int sof_acpi_probe(struct platform_device *pdev, const struct sof_dev_desc *desc sof_pdata->desc = desc; sof_pdata->dev = &pdev->dev; - sof_pdata->fw_filename = desc->default_fw_filename[SOF_IPC_TYPE_3]; - - /* alternate fw and tplg filenames ? */ - if (fw_path) - sof_pdata->fw_filename_prefix = fw_path; - else - sof_pdata->fw_filename_prefix = desc->default_fw_path[SOF_IPC_TYPE_3]; - - if (tplg_path) - sof_pdata->tplg_filename_prefix = tplg_path; - else - sof_pdata->tplg_filename_prefix = desc->default_tplg_path[SOF_IPC_TYPE_3]; sof_pdata->ipc_file_profile_base.ipc_type = desc->ipc_default; sof_pdata->ipc_file_profile_base.fw_path = fw_path; From 8616168928f278723fdc3f4d7cd3d611dcdae8f8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:23 +0200 Subject: [PATCH 130/337] ASoC: SOF: sof-of-dev: Rely on core to create the file paths The core is now using the information from ipc_file_profile_base to create the paths for the loadable files, no need to set it in here anymore. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-10-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-of-dev.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index 7b58f45790f7..fa92da5ee9b3 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -87,17 +87,6 @@ int sof_of_probe(struct platform_device *pdev) sof_pdata->desc = desc; sof_pdata->dev = &pdev->dev; - sof_pdata->fw_filename = desc->default_fw_filename[SOF_IPC_TYPE_3]; - - if (fw_path) - sof_pdata->fw_filename_prefix = fw_path; - else - sof_pdata->fw_filename_prefix = desc->default_fw_path[SOF_IPC_TYPE_3]; - - if (tplg_path) - sof_pdata->tplg_filename_prefix = tplg_path; - else - sof_pdata->tplg_filename_prefix = desc->default_tplg_path[SOF_IPC_TYPE_3]; sof_pdata->ipc_file_profile_base.ipc_type = desc->ipc_default; sof_pdata->ipc_file_profile_base.fw_path = fw_path; From 8a83f180abb5b95f524fc9b5eb2291f0e39bed30 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:24 +0200 Subject: [PATCH 131/337] ASoC: SOF: sof-pci-dev: Rely on core to create the file paths The core is now using the information from ipc_file_profile_base to create the paths for the loadable files, no need to set it in here anymore. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-11-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-pci-dev.c | 115 ++++++------------------------------ 1 file changed, 17 insertions(+), 98 deletions(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index becc85b27d51..aab5c900cecf 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -233,120 +233,39 @@ int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) sof_pdata->desc = desc; sof_pdata->dev = dev; - sof_pdata->ipc_type = desc->ipc_default; + path_override = &sof_pdata->ipc_file_profile_base; if (sof_pci_ipc_type < 0) { - sof_pdata->ipc_type = desc->ipc_default; + path_override->ipc_type = desc->ipc_default; + } else if (sof_pci_ipc_type < SOF_IPC_TYPE_COUNT) { + path_override->ipc_type = sof_pci_ipc_type; } else { - dev_info(dev, "overriding default IPC %d to requested %d\n", - desc->ipc_default, sof_pci_ipc_type); - if (sof_pci_ipc_type >= SOF_IPC_TYPE_COUNT) { - dev_err(dev, "invalid request value %d\n", sof_pci_ipc_type); - ret = -EINVAL; - goto out; - } - if (!(BIT(sof_pci_ipc_type) & desc->ipc_supported_mask)) { - dev_err(dev, "invalid request value %d, supported mask is %#x\n", - sof_pci_ipc_type, desc->ipc_supported_mask); - ret = -EINVAL; - goto out; - } - sof_pdata->ipc_type = sof_pci_ipc_type; + dev_err(dev, "Invalid IPC type requested: %d\n", sof_pci_ipc_type); + ret = -EINVAL; + goto out; } - if (fw_filename) { - sof_pdata->fw_filename = fw_filename; + path_override->fw_path = fw_path; + path_override->fw_name = fw_filename; + path_override->fw_lib_path = lib_path; + path_override->tplg_path = tplg_path; - dev_dbg(dev, "Module parameter used, changed fw filename to %s\n", - sof_pdata->fw_filename); - } else { - sof_pdata->fw_filename = desc->default_fw_filename[sof_pdata->ipc_type]; + if (dmi_check_system(community_key_platforms) && + sof_dmi_use_community_key) { + path_override->fw_path_postfix = "community"; + path_override->fw_lib_path_postfix = "community"; } - /* - * for platforms using the SOF community key, change the - * default path automatically to pick the right files from the - * linux-firmware tree. This can be overridden with the - * fw_path kernel parameter, e.g. for developers. - */ - - /* alternate fw and tplg filenames ? */ - if (fw_path) { - sof_pdata->fw_filename_prefix = fw_path; - - dev_dbg(dev, - "Module parameter used, changed fw path to %s\n", - sof_pdata->fw_filename_prefix); - - } else if (dmi_check_system(community_key_platforms) && sof_dmi_use_community_key) { - sof_pdata->fw_filename_prefix = - devm_kasprintf(dev, GFP_KERNEL, "%s/%s", - sof_pdata->desc->default_fw_path[sof_pdata->ipc_type], - "community"); - - dev_dbg(dev, - "Platform uses community key, changed fw path to %s\n", - sof_pdata->fw_filename_prefix); - } else { - sof_pdata->fw_filename_prefix = - sof_pdata->desc->default_fw_path[sof_pdata->ipc_type]; - } - - if (lib_path) { - sof_pdata->fw_lib_prefix = lib_path; - - dev_dbg(dev, "Module parameter used, changed fw_lib path to %s\n", - sof_pdata->fw_lib_prefix); - - } else if (sof_pdata->desc->default_lib_path[sof_pdata->ipc_type]) { - if (dmi_check_system(community_key_platforms) && sof_dmi_use_community_key) { - sof_pdata->fw_lib_prefix = - devm_kasprintf(dev, GFP_KERNEL, "%s/%s", - sof_pdata->desc->default_lib_path[sof_pdata->ipc_type], - "community"); - - dev_dbg(dev, - "Platform uses community key, changed fw_lib path to %s\n", - sof_pdata->fw_lib_prefix); - } else { - sof_pdata->fw_lib_prefix = - sof_pdata->desc->default_lib_path[sof_pdata->ipc_type]; - } - } - - if (tplg_path) - sof_pdata->tplg_filename_prefix = tplg_path; - else - sof_pdata->tplg_filename_prefix = - sof_pdata->desc->default_tplg_path[sof_pdata->ipc_type]; - /* * the topology filename will be provided in the machine descriptor, unless * it is overridden by a module parameter or DMI quirk. */ if (tplg_filename) { - sof_pdata->tplg_filename = tplg_filename; - - dev_dbg(dev, "Module parameter used, changed tplg filename to %s\n", - sof_pdata->tplg_filename); + path_override->tplg_name = tplg_filename; } else { dmi_check_system(sof_tplg_table); if (sof_dmi_override_tplg_name) - sof_pdata->tplg_filename = sof_dmi_override_tplg_name; - } - - path_override = &sof_pdata->ipc_file_profile_base; - path_override->ipc_type = sof_pdata->ipc_type; - path_override->fw_path = fw_path; - path_override->fw_name = fw_filename; - path_override->fw_lib_path = lib_path; - path_override->tplg_path = tplg_path; - path_override->tplg_name = sof_pdata->tplg_filename; - - if (dmi_check_system(community_key_platforms) && - sof_dmi_use_community_key) { - path_override->fw_path_postfix = "community"; - path_override->fw_lib_path_postfix = "community"; + path_override->tplg_name = sof_dmi_override_tplg_name; } /* set callback to be called on successful device probe to enable runtime_pm */ From a5a65437b02df8b842c4620b0b776bcd91ce200a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:25 +0200 Subject: [PATCH 132/337] ASoC: SOF: core: Add helper for initialization of paths, ops Add sof_init_environment() as a helper function to contain path and ops initialization. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-12-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 56 +++++++++++++++++++++++++++----------------- 1 file changed, 34 insertions(+), 22 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index f1a083de9f9e..a2e9506e0f85 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -265,6 +265,38 @@ static int sof_select_ipc_and_paths(struct snd_sof_dev *sdev) return 0; } +static int sof_init_environment(struct snd_sof_dev *sdev) +{ + int ret; + + ret = sof_select_ipc_and_paths(sdev); + if (ret) + return ret; + + /* init ops, if necessary */ + ret = sof_ops_init(sdev); + if (ret < 0) + return ret; + + /* check all mandatory ops */ + if (!sof_ops(sdev) || !sof_ops(sdev)->probe) { + dev_err(sdev->dev, "missing mandatory ops\n"); + sof_ops_free(sdev); + return -EINVAL; + } + + if (!sdev->dspless_mode_selected && + (!sof_ops(sdev)->run || !sof_ops(sdev)->block_read || + !sof_ops(sdev)->block_write || !sof_ops(sdev)->send_msg || + !sof_ops(sdev)->load_firmware || !sof_ops(sdev)->ipc_msg_data)) { + dev_err(sdev->dev, "missing mandatory DSP ops\n"); + sof_ops_free(sdev); + return -EINVAL; + } + + return 0; +} + /* * FW Boot State Transition Diagram * @@ -503,31 +535,11 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) } } - ret = sof_select_ipc_and_paths(sdev); + /* Initialize loadable file paths and check the environment validity */ + ret = sof_init_environment(sdev); if (ret) return ret; - /* init ops, if necessary */ - ret = sof_ops_init(sdev); - if (ret < 0) - return ret; - - /* check all mandatory ops */ - if (!sof_ops(sdev) || !sof_ops(sdev)->probe) { - sof_ops_free(sdev); - dev_err(dev, "missing mandatory ops\n"); - return -EINVAL; - } - - if (!sdev->dspless_mode_selected && - (!sof_ops(sdev)->run || !sof_ops(sdev)->block_read || - !sof_ops(sdev)->block_write || !sof_ops(sdev)->send_msg || - !sof_ops(sdev)->load_firmware || !sof_ops(sdev)->ipc_msg_data)) { - sof_ops_free(sdev); - dev_err(dev, "missing mandatory DSP ops\n"); - return -EINVAL; - } - INIT_LIST_HEAD(&sdev->pcm_list); INIT_LIST_HEAD(&sdev->kcontrol_list); INIT_LIST_HEAD(&sdev->widget_list); From 9b6896538ea71b7c24da1ecb38738a311176f6a8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:26 +0200 Subject: [PATCH 133/337] ASoC: SOF: Intel: Do not use resource managed allocation for ipc4_data Manage the ipc4_data allocation in code instead of devm since the ops_init might be called more than once due to IPC type fallback. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-13-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/cnl.c | 2 +- sound/soc/sof/intel/hda-dai.c | 3 +++ sound/soc/sof/intel/icl.c | 2 +- sound/soc/sof/intel/lnl.c | 2 +- sound/soc/sof/intel/mtl.c | 2 +- sound/soc/sof/intel/skl.c | 2 +- sound/soc/sof/intel/tgl.c | 2 +- 8 files changed, 10 insertions(+), 7 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 776b66389c34..dee6c7f73e80 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -55,7 +55,7 @@ int sof_apl_ops_init(struct snd_sof_dev *sdev) if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { struct sof_ipc4_fw_data *ipc4_data; - sdev->private = devm_kzalloc(sdev->dev, sizeof(*ipc4_data), GFP_KERNEL); + sdev->private = kzalloc(sizeof(*ipc4_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 598cf50abadb..85e1e4760d0e 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -402,7 +402,7 @@ int sof_cnl_ops_init(struct snd_sof_dev *sdev) if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { struct sof_ipc4_fw_data *ipc4_data; - sdev->private = devm_kzalloc(sdev->dev, sizeof(*ipc4_data), GFP_KERNEL); + sdev->private = kzalloc(sizeof(*ipc4_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index a20deaf3b428..f4cbc0ad5de3 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -621,6 +621,9 @@ void hda_ops_free(struct snd_sof_dev *sdev) if (!hda_use_tplg_nhlt) intel_nhlt_free(ipc4_data->nhlt); + + kfree(sdev->private); + sdev->private = NULL; } } EXPORT_SYMBOL_NS(hda_ops_free, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/icl.c b/sound/soc/sof/intel/icl.c index 8e29d6bb6fe8..040698591992 100644 --- a/sound/soc/sof/intel/icl.c +++ b/sound/soc/sof/intel/icl.c @@ -123,7 +123,7 @@ int sof_icl_ops_init(struct snd_sof_dev *sdev) if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { struct sof_ipc4_fw_data *ipc4_data; - sdev->private = devm_kzalloc(sdev->dev, sizeof(*ipc4_data), GFP_KERNEL); + sdev->private = kzalloc(sizeof(*ipc4_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index db94b45e53af..03308721ebd4 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -120,7 +120,7 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - sdev->private = devm_kzalloc(sdev->dev, sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); + sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 3ef9e5c37028..f941e2c49d78 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -709,7 +709,7 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev) sof_mtl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; - sdev->private = devm_kzalloc(sdev->dev, sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); + sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/skl.c b/sound/soc/sof/intel/skl.c index d24e64e71b58..93824e6ce573 100644 --- a/sound/soc/sof/intel/skl.c +++ b/sound/soc/sof/intel/skl.c @@ -62,7 +62,7 @@ int sof_skl_ops_init(struct snd_sof_dev *sdev) /* probe/remove/shutdown */ sof_skl_ops.shutdown = hda_dsp_shutdown; - sdev->private = devm_kzalloc(sdev->dev, sizeof(*ipc4_data), GFP_KERNEL); + sdev->private = kzalloc(sizeof(*ipc4_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index f7de1f5ba06d..d890cac6cb01 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -82,7 +82,7 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev) if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) { struct sof_ipc4_fw_data *ipc4_data; - sdev->private = devm_kzalloc(sdev->dev, sizeof(*ipc4_data), GFP_KERNEL); + sdev->private = kzalloc(sizeof(*ipc4_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; From 6c393ebbd74ad341bcfb4e2d0091b2655fad45d0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:53:27 +0200 Subject: [PATCH 134/337] ASoC: SOF: core: Implement IPC version fallback if firmware files are missing If a firmware file is missing for the selected IPC type then try to switch to other supported IPC type and check if that one can be used instead. If for example a platform is changed to IPC4 as default version but the given machine does not yet have the needed topology file created then we will fall back to IPC3 which should have all the needed files. Relocate the sof_init_environment() to be done at a later phase, in sof_probe_continue(). This will only have changes in behavior if CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE is enabled (Intel HDA platforms) by not failing the module probe, but it is not going to be different case compared to for example failed firmware booting or topology loading error. Signed-off-by: Peter Ujfalusi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20231129125327.23708-14-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/Kconfig | 11 ++ sound/soc/sof/core.c | 104 ++++++++++----- sound/soc/sof/fw-file-profile.c | 226 +++++++++++++++++++++++++++++--- 3 files changed, 292 insertions(+), 49 deletions(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index a741ed96e789..32ffd345e07f 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -126,6 +126,17 @@ config SND_SOC_SOF_STRICT_ABI_CHECKS If you are not involved in SOF releases and CI development, select "N". +config SND_SOC_SOF_ALLOW_FALLBACK_TO_NEWER_IPC_VERSION + bool "SOF allow fallback to newer IPC version" + help + This option will allow the kernel to try to 'fallback' to a newer IPC + version if there are missing firmware files to satisfy the default IPC + version. + IPC version fallback to older versions is not affected by this option, + it is always available. + Say Y if you are involved in SOF development and need this option. + If not, select N. + config SND_SOC_SOF_DEBUG bool "SOF debugging features" help diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index a2e9506e0f85..a2afec8f5879 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -212,14 +212,6 @@ static int sof_select_ipc_and_paths(struct snd_sof_dev *sdev) struct device *dev = sdev->dev; int ret; - /* check IPC support */ - if (!(BIT(base_profile->ipc_type) & plat_data->desc->ipc_supported_mask)) { - dev_err(dev, - "ipc_type %d is not supported on this platform, mask is %#x\n", - base_profile->ipc_type, plat_data->desc->ipc_supported_mask); - return -EINVAL; - } - if (base_profile->ipc_type != plat_data->desc->ipc_default) dev_info(dev, "Module parameter used, overriding default IPC %d to %d\n", @@ -260,19 +252,14 @@ static int sof_select_ipc_and_paths(struct snd_sof_dev *sdev) plat_data->fw_filename_prefix = out_profile.fw_path; plat_data->fw_lib_prefix = out_profile.fw_lib_path; plat_data->tplg_filename_prefix = out_profile.tplg_path; - plat_data->tplg_filename = out_profile.tplg_name; return 0; } -static int sof_init_environment(struct snd_sof_dev *sdev) +static int validate_sof_ops(struct snd_sof_dev *sdev) { int ret; - ret = sof_select_ipc_and_paths(sdev); - if (ret) - return ret; - /* init ops, if necessary */ ret = sof_ops_init(sdev); if (ret < 0) @@ -297,6 +284,71 @@ static int sof_init_environment(struct snd_sof_dev *sdev) return 0; } +static int sof_init_sof_ops(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + struct sof_loadable_file_profile *base_profile = &plat_data->ipc_file_profile_base; + + /* check IPC support */ + if (!(BIT(base_profile->ipc_type) & plat_data->desc->ipc_supported_mask)) { + dev_err(sdev->dev, + "ipc_type %d is not supported on this platform, mask is %#x\n", + base_profile->ipc_type, plat_data->desc->ipc_supported_mask); + return -EINVAL; + } + + /* + * Save the selected IPC type and a topology name override before + * selecting ops since platform code might need this information + */ + plat_data->ipc_type = base_profile->ipc_type; + plat_data->tplg_filename = base_profile->tplg_name; + + return validate_sof_ops(sdev); +} + +static int sof_init_environment(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + struct sof_loadable_file_profile *base_profile = &plat_data->ipc_file_profile_base; + int ret; + + /* probe the DSP hardware */ + ret = snd_sof_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "failed to probe DSP %d\n", ret); + sof_ops_free(sdev); + return ret; + } + + /* check machine info */ + ret = sof_machine_check(sdev); + if (ret < 0) { + dev_err(sdev->dev, "failed to get machine info %d\n", ret); + goto err_machine_check; + } + + ret = sof_select_ipc_and_paths(sdev); + if (!ret && plat_data->ipc_type != base_profile->ipc_type) { + /* IPC type changed, re-initialize the ops */ + sof_ops_free(sdev); + + ret = validate_sof_ops(sdev); + if (ret < 0) { + snd_sof_remove(sdev); + return ret; + } + } + +err_machine_check: + if (ret) { + snd_sof_remove(sdev); + sof_ops_free(sdev); + } + + return ret; +} + /* * FW Boot State Transition Diagram * @@ -342,23 +394,13 @@ static int sof_probe_continue(struct snd_sof_dev *sdev) struct snd_sof_pdata *plat_data = sdev->pdata; int ret; - /* probe the DSP hardware */ - ret = snd_sof_probe(sdev); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to probe DSP %d\n", ret); - goto probe_err; - } + /* Initialize loadable file paths and check the environment validity */ + ret = sof_init_environment(sdev); + if (ret) + return ret; sof_set_fw_state(sdev, SOF_FW_BOOT_PREPARE); - /* check machine info */ - ret = sof_machine_check(sdev); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to get machine info %d\n", - ret); - goto dsp_err; - } - /* set up platform component driver */ snd_sof_new_platform_drv(sdev); @@ -478,9 +520,7 @@ fw_load_err: ipc_err: dbg_err: snd_sof_free_debug(sdev); -dsp_err: snd_sof_remove(sdev); -probe_err: snd_sof_remove_late(sdev); sof_ops_free(sdev); @@ -535,8 +575,8 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) } } - /* Initialize loadable file paths and check the environment validity */ - ret = sof_init_environment(sdev); + /* Initialize sof_ops based on the initial selected IPC version */ + ret = sof_init_sof_ops(sdev); if (ret) return ret; diff --git a/sound/soc/sof/fw-file-profile.c b/sound/soc/sof/fw-file-profile.c index 58b55516049e..138a1ca2c4a8 100644 --- a/sound/soc/sof/fw-file-profile.c +++ b/sound/soc/sof/fw-file-profile.c @@ -6,17 +6,99 @@ // Copyright(c) 2023 Intel Corporation. All rights reserved. // +#include #include +#include #include "sof-priv.h" +static int sof_test_firmware_file(struct device *dev, + struct sof_loadable_file_profile *profile, + enum sof_ipc_type *ipc_type_to_adjust) +{ + enum sof_ipc_type fw_ipc_type; + const struct firmware *fw; + const char *fw_filename; + const u32 *magic; + int ret; + + fw_filename = kasprintf(GFP_KERNEL, "%s/%s", profile->fw_path, + profile->fw_name); + if (!fw_filename) + return -ENOMEM; + + ret = firmware_request_nowarn(&fw, fw_filename, dev); + if (ret < 0) { + dev_dbg(dev, "Failed to open firmware file: %s\n", fw_filename); + kfree(fw_filename); + return ret; + } + + /* firmware file exists, check the magic number */ + magic = (const u32 *)fw->data; + switch (*magic) { + case SOF_EXT_MAN_MAGIC_NUMBER: + fw_ipc_type = SOF_IPC_TYPE_3; + break; + case SOF_EXT_MAN4_MAGIC_NUMBER: + fw_ipc_type = SOF_IPC_TYPE_4; + break; + default: + dev_err(dev, "Invalid firmware magic: %#x\n", *magic); + ret = -EINVAL; + goto out; + } + + if (ipc_type_to_adjust) { + *ipc_type_to_adjust = fw_ipc_type; + } else if (fw_ipc_type != profile->ipc_type) { + dev_err(dev, + "ipc type mismatch between %s and expected: %d vs %d\n", + fw_filename, fw_ipc_type, profile->ipc_type); + ret = -EINVAL; + } +out: + release_firmware(fw); + kfree(fw_filename); + + return ret; +} + +static int sof_test_topology_file(struct device *dev, + struct sof_loadable_file_profile *profile) +{ + const struct firmware *fw; + const char *tplg_filename; + int ret; + + if (!profile->tplg_path || !profile->tplg_name) + return 0; + + tplg_filename = kasprintf(GFP_KERNEL, "%s/%s", profile->tplg_path, + profile->tplg_name); + if (!tplg_filename) + return -ENOMEM; + + ret = firmware_request_nowarn(&fw, tplg_filename, dev); + if (!ret) + release_firmware(fw); + else + dev_dbg(dev, "Failed to open topology file: %s\n", tplg_filename); + + kfree(tplg_filename); + + return ret; +} + static int -sof_file_profile_for_ipc_type(struct device *dev, +sof_file_profile_for_ipc_type(struct snd_sof_dev *sdev, + enum sof_ipc_type ipc_type, const struct sof_dev_desc *desc, struct sof_loadable_file_profile *base_profile, struct sof_loadable_file_profile *out_profile) { - enum sof_ipc_type ipc_type = base_profile->ipc_type; + struct snd_sof_pdata *plat_data = sdev->pdata; bool fw_lib_path_allocated = false; + struct device *dev = sdev->dev; bool fw_path_allocated = false; int ret = 0; @@ -41,6 +123,25 @@ sof_file_profile_for_ipc_type(struct device *dev, else out_profile->fw_name = desc->default_fw_filename[ipc_type]; + /* + * Check the custom firmware path/filename and adjust the ipc_type to + * match with the existing file for the remaining path configuration. + * + * For default path and firmware name do a verification before + * continuing further. + */ + if (base_profile->fw_path || base_profile->fw_name) { + ret = sof_test_firmware_file(dev, out_profile, &ipc_type); + if (ret) + return ret; + + if (!(desc->ipc_supported_mask & BIT(ipc_type))) { + dev_err(dev, "Unsupported IPC type %d needed by %s/%s\n", + ipc_type, out_profile->fw_path, + out_profile->fw_name); + return -EINVAL; + } + } /* firmware library path */ if (base_profile->fw_lib_path) { @@ -75,11 +176,17 @@ sof_file_profile_for_ipc_type(struct device *dev, out_profile->tplg_path = desc->default_tplg_path[ipc_type]; /* topology name */ - if (base_profile->tplg_name) - out_profile->tplg_name = base_profile->tplg_name; + out_profile->tplg_name = plat_data->tplg_filename; out_profile->ipc_type = ipc_type; + /* Test only default firmware file */ + if (!base_profile->fw_path && !base_profile->fw_name) + ret = sof_test_firmware_file(dev, out_profile, NULL); + + if (!ret) + ret = sof_test_topology_file(dev, out_profile); + out: if (ret) { /* Free up path strings created with devm_kasprintf */ @@ -94,19 +201,76 @@ out: return ret; } -static void sof_print_profile_info(struct device *dev, +static void +sof_print_missing_firmware_info(struct snd_sof_dev *sdev, + enum sof_ipc_type ipc_type, + struct sof_loadable_file_profile *base_profile) +{ + struct snd_sof_pdata *plat_data = sdev->pdata; + const struct sof_dev_desc *desc = plat_data->desc; + struct device *dev = sdev->dev; + int ipc_type_count, i; + char *marker; + + dev_err(dev, "SOF firmware and/or topology file not found.\n"); + dev_info(dev, "Supported default profiles\n"); + + if (IS_ENABLED(CONFIG_SND_SOC_SOF_ALLOW_FALLBACK_TO_NEWER_IPC_VERSION)) + ipc_type_count = SOF_IPC_TYPE_COUNT - 1; + else + ipc_type_count = base_profile->ipc_type; + + for (i = 0; i <= ipc_type_count; i++) { + if (!(desc->ipc_supported_mask & BIT(i))) + continue; + + if (i == ipc_type) + marker = "Requested"; + else + marker = "Fallback"; + + dev_info(dev, "- ipc type %d (%s):\n", i, marker); + if (base_profile->fw_path_postfix) + dev_info(dev, " Firmware file: %s/%s/%s\n", + desc->default_fw_path[i], + base_profile->fw_path_postfix, + desc->default_fw_filename[i]); + else + dev_info(dev, " Firmware file: %s/%s\n", + desc->default_fw_path[i], + desc->default_fw_filename[i]); + + dev_info(dev, " Topology file: %s/%s\n", + desc->default_tplg_path[i], + plat_data->tplg_filename); + } + + if (base_profile->fw_path || base_profile->fw_name || + base_profile->tplg_path || base_profile->tplg_name) + dev_info(dev, "Verify the path/name override module parameters.\n"); + + dev_info(dev, "Check if you have 'sof-firmware' package installed.\n"); + dev_info(dev, "Optionally it can be manually downloaded from:\n"); + dev_info(dev, " https://github.com/thesofproject/sof-bin/\n"); +} + +static void sof_print_profile_info(struct snd_sof_dev *sdev, + enum sof_ipc_type ipc_type, struct sof_loadable_file_profile *profile) { + struct device *dev = sdev->dev; + + if (ipc_type != profile->ipc_type) + dev_info(dev, + "Using fallback IPC type %d (requested type was %d)\n", + profile->ipc_type, ipc_type); + dev_info(dev, "Firmware paths/files for ipc type %d:\n", profile->ipc_type); dev_info(dev, " Firmware file: %s/%s\n", profile->fw_path, profile->fw_name); if (profile->fw_lib_path) dev_info(dev, " Firmware lib path: %s\n", profile->fw_lib_path); - if (profile->tplg_name) - dev_info(dev, " Topology file: %s/%s\n", profile->tplg_path, - profile->tplg_name); - else - dev_info(dev, " Topology path: %s\n", profile->tplg_path); + dev_info(dev, " Topology file: %s/%s\n", profile->tplg_path, profile->tplg_name); } int sof_create_ipc_file_profile(struct snd_sof_dev *sdev, @@ -114,17 +278,45 @@ int sof_create_ipc_file_profile(struct snd_sof_dev *sdev, struct sof_loadable_file_profile *out_profile) { const struct sof_dev_desc *desc = sdev->pdata->desc; - struct device *dev = sdev->dev; - int ret; + int ipc_fallback_start, ret, i; memset(out_profile, 0, sizeof(*out_profile)); - ret = sof_file_profile_for_ipc_type(dev, desc, base_profile, out_profile); + ret = sof_file_profile_for_ipc_type(sdev, base_profile->ipc_type, desc, + base_profile, out_profile); + if (!ret) + goto out; + + /* + * No firmware file was found for the requested IPC type, as fallback + * if SND_SOC_SOF_ALLOW_FALLBACK_TO_NEWER_IPC_VERSION is selected, check + * all IPC versions in a backwards direction (from newer to older) + * if SND_SOC_SOF_ALLOW_FALLBACK_TO_NEWER_IPC_VERSION is not selected, + * check only older IPC versions than the selected/default version + */ + if (IS_ENABLED(CONFIG_SND_SOC_SOF_ALLOW_FALLBACK_TO_NEWER_IPC_VERSION)) + ipc_fallback_start = SOF_IPC_TYPE_COUNT - 1; + else + ipc_fallback_start = (int)base_profile->ipc_type - 1; + + for (i = ipc_fallback_start; i >= 0 ; i--) { + if (i == base_profile->ipc_type || + !(desc->ipc_supported_mask & BIT(i))) + continue; + + ret = sof_file_profile_for_ipc_type(sdev, i, desc, base_profile, + out_profile); + if (!ret) + break; + } + +out: if (ret) - return ret; + sof_print_missing_firmware_info(sdev, base_profile->ipc_type, + base_profile); + else + sof_print_profile_info(sdev, base_profile->ipc_type, out_profile); - sof_print_profile_info(dev, out_profile); - - return 0; + return ret; } EXPORT_SYMBOL(sof_create_ipc_file_profile); From 2bd512626f8ea3957c981cadd2ebf75feff737dd Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 29 Nov 2023 14:20:21 +0200 Subject: [PATCH 135/337] ASoC: SOF: ipc4: check return value of snd_sof_ipc_msg_data MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_sof_ipc_msg_data could return error. Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231129122021.679-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index a9d9800d2fcc..145d319e041f 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -713,7 +713,14 @@ static void sof_ipc4_rx_msg(struct snd_sof_dev *sdev) return; ipc4_msg->data_size = data_size; - snd_sof_ipc_msg_data(sdev, NULL, ipc4_msg->data_ptr, ipc4_msg->data_size); + err = snd_sof_ipc_msg_data(sdev, NULL, ipc4_msg->data_ptr, ipc4_msg->data_size); + if (err < 0) { + dev_err(sdev->dev, "failed to read IPC notification data: %d\n", err); + kfree(ipc4_msg->data_ptr); + ipc4_msg->data_ptr = NULL; + ipc4_msg->data_size = 0; + return; + } } /* Handle notifications with payload */ From 729bb2cd2e0601ac6d52c53535a543b9db196fad Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 Nov 2023 14:28:05 +0200 Subject: [PATCH 136/337] ASoC: SOF: ipc4: Move window offset configuration earlier With the added exception handling support if the firmware fails to boot up we are trying to do a panic dump from the telemetry slot. The slot offsets would have been configured only after receiving the FW_READY message which makes this panic dump unusable for early boot failures. With IPC4 the DSP window offsets are at standard places unlike IPC3 where the offsets needs to be queried from the FW_READY message. Move the offset configuration to sof_ipc4_init from the fw_ready handler. Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231129122805.10635-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4.c | 55 ++++++++++++++++++++++---------------------- 1 file changed, 27 insertions(+), 28 deletions(-) diff --git a/sound/soc/sof/ipc4.c b/sound/soc/sof/ipc4.c index 145d319e041f..ac5c6bc66d2a 100644 --- a/sound/soc/sof/ipc4.c +++ b/sound/soc/sof/ipc4.c @@ -576,40 +576,12 @@ EXPORT_SYMBOL(sof_ipc4_find_debug_slot_offset_by_type); static int ipc4_fw_ready(struct snd_sof_dev *sdev, struct sof_ipc4_msg *ipc4_msg) { - int inbox_offset, inbox_size, outbox_offset, outbox_size; - /* no need to re-check version/ABI for subsequent boots */ if (!sdev->first_boot) return 0; - /* Set up the windows for IPC communication */ - inbox_offset = snd_sof_dsp_get_mailbox_offset(sdev); - if (inbox_offset < 0) { - dev_err(sdev->dev, "%s: No mailbox offset\n", __func__); - return inbox_offset; - } - inbox_size = SOF_IPC4_MSG_MAX_SIZE; - outbox_offset = snd_sof_dsp_get_window_offset(sdev, SOF_IPC4_OUTBOX_WINDOW_IDX); - outbox_size = SOF_IPC4_MSG_MAX_SIZE; - - sdev->fw_info_box.offset = snd_sof_dsp_get_window_offset(sdev, SOF_IPC4_INBOX_WINDOW_IDX); - sdev->fw_info_box.size = sizeof(struct sof_ipc4_fw_registers); - sdev->dsp_box.offset = inbox_offset; - sdev->dsp_box.size = inbox_size; - sdev->host_box.offset = outbox_offset; - sdev->host_box.size = outbox_size; - - sdev->debug_box.offset = snd_sof_dsp_get_window_offset(sdev, - SOF_IPC4_DEBUG_WINDOW_IDX); - sof_ipc4_create_exception_debugfs_node(sdev); - dev_dbg(sdev->dev, "mailbox upstream 0x%x - size 0x%x\n", - inbox_offset, inbox_size); - dev_dbg(sdev->dev, "mailbox downstream 0x%x - size 0x%x\n", - outbox_offset, outbox_size); - dev_dbg(sdev->dev, "debug box 0x%x\n", sdev->debug_box.offset); - return sof_ipc4_init_msg_memory(sdev); } @@ -796,11 +768,38 @@ static const struct sof_ipc_pm_ops ipc4_pm_ops = { static int sof_ipc4_init(struct snd_sof_dev *sdev) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; + int inbox_offset; mutex_init(&ipc4_data->pipeline_state_mutex); xa_init_flags(&ipc4_data->fw_lib_xa, XA_FLAGS_ALLOC); + /* Set up the windows for IPC communication */ + inbox_offset = snd_sof_dsp_get_mailbox_offset(sdev); + if (inbox_offset < 0) { + dev_err(sdev->dev, "%s: No mailbox offset\n", __func__); + return inbox_offset; + } + + sdev->dsp_box.offset = inbox_offset; + sdev->dsp_box.size = SOF_IPC4_MSG_MAX_SIZE; + sdev->host_box.offset = snd_sof_dsp_get_window_offset(sdev, + SOF_IPC4_OUTBOX_WINDOW_IDX); + sdev->host_box.size = SOF_IPC4_MSG_MAX_SIZE; + + sdev->debug_box.offset = snd_sof_dsp_get_window_offset(sdev, + SOF_IPC4_DEBUG_WINDOW_IDX); + + sdev->fw_info_box.offset = snd_sof_dsp_get_window_offset(sdev, + SOF_IPC4_INBOX_WINDOW_IDX); + sdev->fw_info_box.size = sizeof(struct sof_ipc4_fw_registers); + + dev_dbg(sdev->dev, "mailbox upstream %#x - size %#x\n", + sdev->dsp_box.offset, SOF_IPC4_MSG_MAX_SIZE); + dev_dbg(sdev->dev, "mailbox downstream %#x - size %#x\n", + sdev->host_box.offset, SOF_IPC4_MSG_MAX_SIZE); + dev_dbg(sdev->dev, "debug box %#x\n", sdev->debug_box.offset); + return 0; } From e9a92dfc8d4fde6f7adb978fb13d0b0834567cc5 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 29 Nov 2023 09:09:58 +0000 Subject: [PATCH 137/337] ASoC: core: Fix a handful of spelling mistakes. There is a spelling mistake in a dev_err message and several spelling mistakes in comments. Fix them. Signed-off-by: Colin Ian King Link: https://lore.kernel.org/r/20231129090958.815775-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4ca3319a8e19..132946f82a29 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1059,7 +1059,7 @@ static int snd_soc_compensate_channel_connection_map(struct snd_soc_card *card, /* it should have ch_maps if connection was N:M */ if (dai_link->num_cpus > 1 && dai_link->num_codecs > 1 && dai_link->num_cpus != dai_link->num_codecs && !dai_link->ch_maps) { - dev_err(card->dev, "need to have ch_maps when N:M connction (%s)", + dev_err(card->dev, "need to have ch_maps when N:M connection (%s)", dai_link->name); return -EINVAL; } @@ -1299,7 +1299,7 @@ found: * * To avoid such issue, loop from 63 to 0 here. * Small number of SND_SOC_POSSIBLE_xxx will be Hi priority. - * Basic/Default settings of each part and aboves are defined + * Basic/Default settings of each part and above are defined * as Hi priority (= small number) of SND_SOC_POSSIBLE_xxx. */ for (i = 63; i >= 0; i--) { @@ -1845,7 +1845,7 @@ static void append_dmi_string(struct snd_soc_card *card, const char *str) * @flavour: The flavour "differentiator" for the card amongst its peers. * * An Intel machine driver may be used by many different devices but are - * difficult for userspace to differentiate, since machine drivers ususally + * difficult for userspace to differentiate, since machine drivers usually * use their own name as the card short name and leave the card long name * blank. To differentiate such devices and fix bugs due to lack of * device-specific configurations, this function allows DMI info to be used @@ -1866,7 +1866,7 @@ static void append_dmi_string(struct snd_soc_card *card, const char *str) * We only keep number and alphabet characters and a few separator characters * in the card long name since UCM in the user space uses the card long names * as card configuration directory names and AudoConf cannot support special - * charactors like SPACE. + * characters like SPACE. * * Returns 0 on success, otherwise a negative error code. */ @@ -2461,7 +2461,7 @@ EXPORT_SYMBOL_GPL(snd_soc_add_card_controls); /** * snd_soc_add_dai_controls - add an array of controls to a DAI. - * Convienience function to add a list of controls. + * Convenience function to add a list of controls. * * @dai: DAI to add controls to * @controls: array of controls to add From 8f464a410944650adc8d75c7711d7d56c5506da0 Mon Sep 17 00:00:00 2001 From: Baofeng Tian Date: Wed, 29 Nov 2023 14:22:34 +0200 Subject: [PATCH 138/337] ASoC: SOF: ipc4-topology: Add module ID print during module set up MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This module ID will be used for module performance automatic analysis for different modules, module name, module ID and module instance ID will be combined as a new generated ID for current module, this ID will be further used by analysis tools to identify current module. Take below case as example: 0x030006 gain.11.1 3 is module instance ID, 6 is module ID and gain.11.1 is module name. For pipeline widget print, keep as it is. Signed-off-by: Baofeng Tian Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Chao Song Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20231129122234.14515-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 8ab9c42069ee..4e7ab4e562e5 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -2383,6 +2383,8 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget } if (swidget->id != snd_soc_dapm_scheduler) { + int module_id = msg->primary & SOF_IPC4_MOD_ID_MASK; + ret = sof_ipc4_widget_assign_instance_id(sdev, swidget); if (ret < 0) { dev_err(sdev->dev, "failed to assign instance id for %s\n", @@ -2398,9 +2400,15 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget msg->extension &= ~SOF_IPC4_MOD_EXT_PPL_ID_MASK; msg->extension |= SOF_IPC4_MOD_EXT_PPL_ID(pipe_widget->instance_id); + + dev_dbg(sdev->dev, "Create widget %s (pipe %d) - ID %d, instance %d, core %d\n", + swidget->widget->name, swidget->pipeline_id, module_id, + swidget->instance_id, swidget->core); + } else { + dev_dbg(sdev->dev, "Create pipeline %s (pipe %d) - instance %d, core %d\n", + swidget->widget->name, swidget->pipeline_id, + swidget->instance_id, swidget->core); } - dev_dbg(sdev->dev, "Create widget %s instance %d - pipe %d - core %d\n", - swidget->widget->name, swidget->instance_id, swidget->pipeline_id, swidget->core); msg->data_size = ipc_size; msg->data_ptr = ipc_data; From 9cce9c4806a89439ea34aad2e382150d68c7ea95 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 29 Nov 2023 12:31:17 +0100 Subject: [PATCH 139/337] ASoC: fsl_rpmsg: update Kconfig dependencies SND_SOC_IMX_RPMSG gained a new dependency and gets selected by SND_SOC_FSL_RPMSG, which as a result needs to have the same dependency, or produce a build failure based on that: WARNING: unmet direct dependencies detected for SND_SOC_IMX_RPMSG Depends on [n]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=y] && RPMSG [=y] && OF [=y] && I2C [=n] Selected by [y]: - SND_SOC_FSL_RPMSG [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && COMMON_CLK [=y] && RPMSG [=y] && (SND_IMX_SOC [=y] || SND_IMX_SOC [=y]=n) && SND_IMX_SOC [=y]!=n x86_64-linux-ld: sound/soc/fsl/imx-rpmsg.o: in function `imx_rpmsg_late_probe': imx-rpmsg.c:(.text+0x11e): undefined reference to `i2c_find_device_by_fwnode' Fixes: f83d38def6b1 ("ASoC: imx-rpmsg: SND_SOC_IMX_RPMSG should depend on OF and I2C") Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20231129113204.2869356-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index bec2aa561930..13cb6960aae7 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -121,6 +121,7 @@ config SND_SOC_FSL_UTILS config SND_SOC_FSL_RPMSG tristate "NXP Audio Base On RPMSG support" depends on COMMON_CLK + depends on OF && I2C depends on RPMSG depends on SND_IMX_SOC || SND_IMX_SOC = n select SND_SOC_IMX_RPMSG if SND_IMX_SOC != n From d32bac9cb09cce4dc3131ec5d0b6ba3c277502ac Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 28 Nov 2023 17:56:37 +0100 Subject: [PATCH 140/337] ASoC: qcom: Add helper for allocating Soundwire stream runtime Newer Qualcomm SoC soundcards will need to allocate Soundwire stream runtime in their startup op. The code will be exactly the same for all soundcards, so add a helper for that. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231128165638.757665-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sdw.c | 45 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/qcom/sdw.h | 1 + 2 files changed, 45 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c index dd275123d31d..77dbe0c28b29 100644 --- a/sound/soc/qcom/sdw.c +++ b/sound/soc/qcom/sdw.c @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018-2023, Linaro Limited. // Copyright (c) 2018, The Linux Foundation. All rights reserved. #include @@ -7,6 +7,49 @@ #include #include "sdw.h" +/** + * qcom_snd_sdw_startup() - Helper to start Soundwire stream for SoC audio card + * @substream: The PCM substream from audio, as passed to snd_soc_ops->startup() + * + * Helper for the SoC audio card (snd_soc_ops->startup()) to allocate and set + * Soundwire stream runtime to each codec DAI. + * + * The shutdown() callback should call sdw_release_stream() on the same + * sdw_stream_runtime. + * + * Return: 0 or errno + */ +int qcom_snd_sdw_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime; + struct snd_soc_dai *codec_dai; + int ret, i; + + sruntime = sdw_alloc_stream(cpu_dai->name); + if (!sruntime) + return -ENOMEM; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_stream(codec_dai, sruntime, + substream->stream); + if (ret < 0 && ret != -ENOTSUPP) { + dev_err(rtd->dev, "Failed to set sdw stream on %s\n", + codec_dai->name); + goto err_set_stream; + } + } + + return 0; + +err_set_stream: + sdw_release_stream(sruntime); + + return ret; +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_startup); + int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, struct sdw_stream_runtime *sruntime, bool *stream_prepared) diff --git a/sound/soc/qcom/sdw.h b/sound/soc/qcom/sdw.h index d74cbb84da13..392e3455f1b1 100644 --- a/sound/soc/qcom/sdw.h +++ b/sound/soc/qcom/sdw.h @@ -6,6 +6,7 @@ #include +int qcom_snd_sdw_startup(struct snd_pcm_substream *substream); int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, struct sdw_stream_runtime *runtime, bool *stream_prepared); From 15c7fab0e0477d7d7185eac574ca43c15b59b015 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 28 Nov 2023 17:56:38 +0100 Subject: [PATCH 141/337] ASoC: qcom: Move Soundwire runtime stream alloc to soundcards Currently the Qualcomm Soundwire controller in its DAI startup op allocates the Soundwire stream runtime. This works fine for existing designs, but has limitations for stream runtimes with multiple controllers, like upcoming Qualcomm X1E80100 SoC with four WSA8840 speakers on two Soundwire controllers. When two Soundwire controllers are added to sound card codecs, Soundwire startup() is called twice, one for each Soundwire controller, and second execution overwrites what was set before. During shutdown() this causes double free. It is expected to have only one Soundwire stream runtime, thus it should be allocated from SoC soundcard context startup(), not from each Soundwire startup(). Such way will properly handle both cases: one and two Soundwire controllers in the stream runtime. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231128165638.757665-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- drivers/soundwire/qcom.c | 33 +-------------------------------- sound/soc/qcom/sc8280xp.c | 13 +++++++++++++ sound/soc/qcom/sm8250.c | 15 ++++++++++++++- 3 files changed, 28 insertions(+), 33 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index a1e2d6c98186..8076d40407d4 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1265,10 +1265,7 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev); - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct sdw_stream_runtime *sruntime; - struct snd_soc_dai *codec_dai; - int ret, i; + int ret; ret = pm_runtime_get_sync(ctrl->dev); if (ret < 0 && ret != -EACCES) { @@ -1279,33 +1276,7 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, return ret; } - sruntime = sdw_alloc_stream(dai->name); - if (!sruntime) { - ret = -ENOMEM; - goto err_alloc; - } - - ctrl->sruntime[dai->id] = sruntime; - - for_each_rtd_codec_dais(rtd, i, codec_dai) { - ret = snd_soc_dai_set_stream(codec_dai, sruntime, - substream->stream); - if (ret < 0 && ret != -ENOTSUPP) { - dev_err(dai->dev, "Failed to set sdw stream on %s\n", - codec_dai->name); - goto err_set_stream; - } - } - return 0; - -err_set_stream: - sdw_release_stream(sruntime); -err_alloc: - pm_runtime_mark_last_busy(ctrl->dev); - pm_runtime_put_autosuspend(ctrl->dev); - - return ret; } static void qcom_swrm_shutdown(struct snd_pcm_substream *substream, @@ -1314,8 +1285,6 @@ static void qcom_swrm_shutdown(struct snd_pcm_substream *substream, struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev); swrm_wait_for_wr_fifo_done(ctrl); - sdw_release_stream(ctrl->sruntime[dai->id]); - ctrl->sruntime[dai->id] = NULL; pm_runtime_mark_last_busy(ctrl->dev); pm_runtime_put_autosuspend(ctrl->dev); diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index d93b18f07be5..7c20b25ba3de 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -31,6 +31,17 @@ static int sc8280xp_snd_init(struct snd_soc_pcm_runtime *rtd) return qcom_snd_wcd_jack_setup(rtd, &data->jack, &data->jack_setup); } +static void sc8280xp_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sc8280xp_snd_data *pdata = snd_soc_card_get_drvdata(rtd->card); + struct sdw_stream_runtime *sruntime = pdata->sruntime[cpu_dai->id]; + + pdata->sruntime[cpu_dai->id] = NULL; + sdw_release_stream(sruntime); +} + static int sc8280xp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -91,6 +102,8 @@ static int sc8280xp_snd_hw_free(struct snd_pcm_substream *substream) } static const struct snd_soc_ops sc8280xp_be_ops = { + .startup = qcom_snd_sdw_startup, + .shutdown = sc8280xp_snd_shutdown, .hw_params = sc8280xp_snd_hw_params, .hw_free = sc8280xp_snd_hw_free, .prepare = sc8280xp_snd_prepare, diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 9cc869fd70ac..f298167c2a23 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -66,7 +66,19 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream) default: break; } - return 0; + + return qcom_snd_sdw_startup(substream); +} + +static void sm2450_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sm8250_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + + data->sruntime[cpu_dai->id] = NULL; + sdw_release_stream(sruntime); } static int sm8250_snd_hw_params(struct snd_pcm_substream *substream, @@ -103,6 +115,7 @@ static int sm8250_snd_hw_free(struct snd_pcm_substream *substream) static const struct snd_soc_ops sm8250_be_ops = { .startup = sm8250_snd_startup, + .shutdown = sm2450_snd_shutdown, .hw_params = sm8250_snd_hw_params, .hw_free = sm8250_snd_hw_free, .prepare = sm8250_snd_prepare, From f8dd1f89bd5ecbfcc547ee5b222a4b9ddb0a0160 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:30 +0100 Subject: [PATCH 142/337] ASoC: cs35l32: Drop legacy include This driver includes the legacy GPIO API but does not use any symbols from it. Drop the include. Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-1-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l32.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 138040618438..d1350ffbf3bd 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -13,7 +13,6 @@ #include #include #include -#include #include #include #include From 50678d339d670a92658e5538ebee30447c88ccb3 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:31 +0100 Subject: [PATCH 143/337] ASoC: cs35l33: Fix GPIO name and drop legacy include This driver includes the legacy GPIO APIs and but does not use any symbols from any of them. Drop the includes. Further the driver is requesting "reset-gpios" rather than just "reset" from the GPIO framework. This is wrong because the gpiolib core will add "-gpios" before processing the request from e.g. device tree. Drop the suffix. The last problem means that the optional RESET GPIO has never been properly retrieved and used even if it existed, but nobody noticed. Fixes: 3333cb7187b9 ("ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.") Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-2-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l33.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 4010a2d33a33..a19a2bafb37c 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -22,13 +22,11 @@ #include #include #include -#include #include #include #include #include #include -#include #include #include "cs35l33.h" @@ -1165,7 +1163,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client) /* We could issue !RST or skip it based on AMP topology */ cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_HIGH); + "reset", GPIOD_OUT_HIGH); if (IS_ERR(cs35l33->reset_gpio)) { dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", __func__); From a6122b0b4211d132934ef99e7b737910e6d54d2f Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:32 +0100 Subject: [PATCH 144/337] ASoC: cs35l34: Fix GPIO name and drop legacy include This driver includes the legacy GPIO APIs and but does not use any symbols from any of them. Drop the includes. Further the driver is requesting "reset-gpios" rather than just "reset" from the GPIO framework. This is wrong because the gpiolib core will add "-gpios" before processing the request from e.g. device tree. Drop the suffix. The last problem means that the optional RESET GPIO has never been properly retrieved and used even if it existed, but nobody noticed. Fixes: c1124c09e103 ("ASoC: cs35l34: Initial commit of the cs35l34 CODEC driver.") Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-3-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l34.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index e5871736fa29..cca59de66b73 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -20,14 +20,12 @@ #include #include #include -#include #include #include #include #include #include #include -#include #include #include #include @@ -1061,7 +1059,7 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client) dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_LOW); + "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l34->reset_gpio)) { ret = PTR_ERR(cs35l34->reset_gpio); goto err_regulator; From 490d2d9f190a9a6125495ac4d833fa0af41763d0 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:33 +0100 Subject: [PATCH 145/337] ASoC: cs35l35: Drop legacy includes This driver includes the legacy GPIO APIs and but does not use any symbols from any of them. Drop the includes. Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-4-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l35.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 63a538f747d3..ddb7d63213a3 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -18,14 +18,12 @@ #include #include #include -#include #include #include #include #include #include #include -#include #include #include #include From 194ef700d4e255686173a03b65e15cac637cc15a Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:34 +0100 Subject: [PATCH 146/337] ASoC: cs35l36: Drop legacy includes This driver includes the legacy GPIO APIs and but does not use any symbols from any of them. Drop the includes. Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-5-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l36.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index f2fde6e652b9..f5bd32e434a0 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -17,15 +17,14 @@ #include #include #include +#include #include -#include #include #include #include #include #include #include -#include #include #include #include From 42d1178d223ba9498c1ed40a5fc243a43d35ec97 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:35 +0100 Subject: [PATCH 147/337] ASoC: cs4271: Convert to GPIO descriptors This converts the Cirrus CS4271 ASoC codec driver to use GPIO descriptors. It turns out that there are two in-kernel users of the platform data passing mechanism so these are switched over as well. One locally defined GPIO "gpio_disabled" is declared in the state struct but completely unused in the driver, so we delete it. Reviewed-by: Alexander Sverdlin Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-6-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- arch/arm/mach-ep93xx/edb93xx.c | 32 ++++++++++++++++++++--- arch/arm/mach-ep93xx/vision_ep9307.c | 12 ++++++++- include/sound/cs4271.h | 1 - sound/soc/codecs/cs4271.c | 39 ++++++++++------------------ 4 files changed, 52 insertions(+), 32 deletions(-) diff --git a/arch/arm/mach-ep93xx/edb93xx.c b/arch/arm/mach-ep93xx/edb93xx.c index 4b90899a66e9..dbdb822a0100 100644 --- a/arch/arm/mach-ep93xx/edb93xx.c +++ b/arch/arm/mach-ep93xx/edb93xx.c @@ -88,7 +88,7 @@ static void __init edb93xx_register_i2c(void) * EDB93xx SPI peripheral handling *************************************************************************/ static struct cs4271_platform_data edb93xx_cs4271_data = { - .gpio_nreset = -EINVAL, /* filled in later */ + /* Intentionally left blank */ }; static struct spi_board_info edb93xx_spi_board_info[] __initdata = { @@ -114,14 +114,38 @@ static struct ep93xx_spi_info edb93xx_spi_info __initdata = { /* Intentionally left blank */ }; +static struct gpiod_lookup_table edb93xx_cs4272_edb9301_gpio_table = { + .dev_id = "spi0.0", /* CS0 on SPI0 */ + .table = { + GPIO_LOOKUP("A", 1, "reset", GPIO_ACTIVE_LOW), + { }, + }, +}; + +static struct gpiod_lookup_table edb93xx_cs4272_edb9302_gpio_table = { + .dev_id = "spi0.0", /* CS0 on SPI0 */ + .table = { + GPIO_LOOKUP("H", 2, "reset", GPIO_ACTIVE_LOW), + { }, + }, +}; + +static struct gpiod_lookup_table edb93xx_cs4272_edb9315_gpio_table = { + .dev_id = "spi0.0", /* CS0 on SPI0 */ + .table = { + GPIO_LOOKUP("B", 6, "reset", GPIO_ACTIVE_LOW), + { }, + }, +}; + static void __init edb93xx_register_spi(void) { if (machine_is_edb9301() || machine_is_edb9302()) - edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO1; + gpiod_add_lookup_table(&edb93xx_cs4272_edb9301_gpio_table); else if (machine_is_edb9302a() || machine_is_edb9307a()) - edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_H(2); + gpiod_add_lookup_table(&edb93xx_cs4272_edb9302_gpio_table); else if (machine_is_edb9315a()) - edb93xx_cs4271_data.gpio_nreset = EP93XX_GPIO_LINE_EGPIO14; + gpiod_add_lookup_table(&edb93xx_cs4272_edb9315_gpio_table); gpiod_add_lookup_table(&edb93xx_spi_cs_gpio_table); ep93xx_register_spi(&edb93xx_spi_info, edb93xx_spi_board_info, diff --git a/arch/arm/mach-ep93xx/vision_ep9307.c b/arch/arm/mach-ep93xx/vision_ep9307.c index 30d9cf3791eb..9471938df64c 100644 --- a/arch/arm/mach-ep93xx/vision_ep9307.c +++ b/arch/arm/mach-ep93xx/vision_ep9307.c @@ -164,7 +164,7 @@ static struct i2c_board_info vision_i2c_info[] __initdata = { * SPI CS4271 Audio Codec *************************************************************************/ static struct cs4271_platform_data vision_cs4271_data = { - .gpio_nreset = EP93XX_GPIO_LINE_H(2), + /* Intentionally left blank */ }; /************************************************************************* @@ -241,6 +241,15 @@ static struct spi_board_info vision_spi_board_info[] __initdata = { }, }; +static struct gpiod_lookup_table vision_spi_cs4271_gpio_table = { + .dev_id = "spi0.0", /* cs4271 @ CS0 */ + .table = { + /* RESET */ + GPIO_LOOKUP_IDX("H", 2, NULL, 0, GPIO_ACTIVE_LOW), + { }, + }, +}; + static struct gpiod_lookup_table vision_spi_cs_gpio_table = { .dev_id = "spi0", .table = { @@ -292,6 +301,7 @@ static void __init vision_init_machine(void) ep93xx_register_i2c(vision_i2c_info, ARRAY_SIZE(vision_i2c_info)); + gpiod_add_lookup_table(&vision_spi_cs4271_gpio_table); gpiod_add_lookup_table(&vision_spi_mmc_gpio_table); gpiod_add_lookup_table(&vision_spi_cs_gpio_table); ep93xx_register_spi(&vision_spi_master, vision_spi_board_info, diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h index 6ff23ab48894..5a55d135bdea 100644 --- a/include/sound/cs4271.h +++ b/include/sound/cs4271.h @@ -9,7 +9,6 @@ #define __CS4271_H struct cs4271_platform_data { - int gpio_nreset; /* GPIO driving Reset pin, if any */ bool amutec_eq_bmutec; /* flag to enable AMUTEC=BMUTEC */ /* diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 9e6f8a048dd5..74a84832d958 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -13,9 +13,8 @@ #include #include #include -#include +#include #include -#include #include #include #include @@ -160,9 +159,7 @@ struct cs4271_private { /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ - int gpio_nreset; - /* GPIO that disable serial bus, if any */ - int gpio_disable; + struct gpio_desc *reset; /* enable soft reset workaround */ bool enable_soft_reset; struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; @@ -487,12 +484,10 @@ static int cs4271_reset(struct snd_soc_component *component) { struct cs4271_private *cs4271 = snd_soc_component_get_drvdata(component); - if (gpio_is_valid(cs4271->gpio_nreset)) { - gpio_direction_output(cs4271->gpio_nreset, 0); - mdelay(1); - gpio_set_value(cs4271->gpio_nreset, 1); - mdelay(1); - } + gpiod_direction_output(cs4271->reset, 1); + mdelay(1); + gpiod_set_value(cs4271->reset, 0); + mdelay(1); return 0; } @@ -612,9 +607,8 @@ static void cs4271_component_remove(struct snd_soc_component *component) { struct cs4271_private *cs4271 = snd_soc_component_get_drvdata(component); - if (gpio_is_valid(cs4271->gpio_nreset)) - /* Set codec to the reset state */ - gpio_set_value(cs4271->gpio_nreset, 0); + /* Set codec to the reset state */ + gpiod_set_value(cs4271->reset, 1); regcache_mark_dirty(cs4271->regmap); regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); @@ -639,7 +633,6 @@ static const struct snd_soc_component_driver soc_component_dev_cs4271 = { static int cs4271_common_probe(struct device *dev, struct cs4271_private **c) { - struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; int i, ret; @@ -647,17 +640,11 @@ static int cs4271_common_probe(struct device *dev, if (!cs4271) return -ENOMEM; - cs4271->gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0); - - if (cs4271plat) - cs4271->gpio_nreset = cs4271plat->gpio_nreset; - - if (gpio_is_valid(cs4271->gpio_nreset)) { - ret = devm_gpio_request(dev, cs4271->gpio_nreset, - "CS4271 Reset"); - if (ret < 0) - return ret; - } + cs4271->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_ASIS); + if (IS_ERR(cs4271->reset)) + return dev_err_probe(dev, PTR_ERR(cs4271->reset), + "error retrieveing RESET GPIO\n"); + gpiod_set_consumer_name(cs4271->reset, "CS4271 Reset"); for (i = 0; i < ARRAY_SIZE(supply_names); i++) cs4271->supplies[i].supply = supply_names[i]; From b191a524b225a3acd7528c3fa8ebeb15302c979b Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:36 +0100 Subject: [PATCH 148/337] ASoC: cirrus: edb93xx: Drop legacy include This driver includes the legacy GPIO API but does not use any symbols from it. Drop the include. Reviewed-by: Alexander Sverdlin Link: https://patchwork.kernel.org/project/alsa-devel/patch/20231122-ep93xx-v5-38-d59a76d5df29@maquefel.me/ Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-7-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/cirrus/edb93xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 6b6817256331..8bb67d7d2b4b 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -11,7 +11,6 @@ */ #include -#include #include #include #include From 0ec65e8e2219a91987cbb041387d94d40cab38b2 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:37 +0100 Subject: [PATCH 149/337] ASoC: cs42l42: Drop legacy include This driver includes the legacy GPIO API but does not use any symbols from it. Drop the include. Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-8-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l42.h | 2 +- 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 94bcab812629..2d11c5125f73 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -15,7 +15,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 7785125b73ab..3d85ebc59489 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -14,7 +14,7 @@ #include #include -#include +#include #include #include #include From c6324cafd8370c2254f90a33b7327c45e7a58bbd Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:38 +0100 Subject: [PATCH 150/337] ASoC: cs43130: Drop legacy includes This driver includes the legacy GPIO APIs and but does not use any symbols from any of them. Drop the includes. Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-9-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 0f3ead84665f..b6d829bbe3cc 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -11,7 +11,6 @@ #include #include #include -#include #include #include #include @@ -26,7 +25,6 @@ #include #include #include -#include #include #include #include From 9c16cfe42d9fbfd258a3928b4a054d6b3ef932b0 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 1 Dec 2023 14:20:39 +0100 Subject: [PATCH 151/337] ASoC: cs4349: Drop legacy include This driver includes the legacy GPIO API but does not use any symbols from it. Drop the include. Acked-by: Charles Keepax Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231201-descriptors-sound-cirrus-v2-10-ee9f9d4655eb@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/cs4349.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 9083228495d4..ca8f21aa4837 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -13,7 +13,6 @@ #include #include #include -#include #include #include #include From 625ed9457de50d7726ccb3f2bc4e01e543ceb126 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 1 Dec 2023 14:53:31 +0100 Subject: [PATCH 152/337] ASoC: qcom: sc8280xp: set card driver name from match data Sound machine drivers for all newer Qualcomm SoC platforms are the exactly same, therefore it makes sense to use same machine driver for newer platforms as well. Choice of sound topology and user-space Alsa UCM files depends however on card driver name, which must be customized per each board. Allow such customization by using driver match data as sound card driver name. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231201135332.154017-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 7c20b25ba3de..77722d6416b8 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -14,8 +14,6 @@ #include "common.h" #include "sdw.h" -#define DRIVER_NAME "sc8280xp" - struct sc8280xp_snd_data { bool stream_prepared[AFE_PORT_MAX]; struct snd_soc_card *card; @@ -146,13 +144,13 @@ static int sc8280xp_platform_probe(struct platform_device *pdev) if (ret) return ret; - card->driver_name = DRIVER_NAME; + card->driver_name = of_device_get_match_data(dev); sc8280xp_add_be_ops(card); return devm_snd_soc_register_card(dev, card); } static const struct of_device_id snd_sc8280xp_dt_match[] = { - {.compatible = "qcom,sc8280xp-sndcard",}, + {.compatible = "qcom,sc8280xp-sndcard", "sc8280xp"}, {} }; From fdcaecfc71e2f4ab70ce9469f14dd64c23bf401a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 1 Dec 2023 14:53:32 +0100 Subject: [PATCH 153/337] ASoC: qcom: sc8280xp: Add support for SM8450 and SM8550 Add compatibles for sound card on Qualcomm SM8450 and SM8550 boards. The compatibles were already documented. Signed-off-by: Krzysztof Kozlowski Link: https://lore.kernel.org/r/20231201135332.154017-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 77722d6416b8..249a43e1dee3 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -151,6 +151,8 @@ static int sc8280xp_platform_probe(struct platform_device *pdev) static const struct of_device_id snd_sc8280xp_dt_match[] = { {.compatible = "qcom,sc8280xp-sndcard", "sc8280xp"}, + {.compatible = "qcom,sm8450-sndcard", "sm8450"}, + {.compatible = "qcom,sm8550-sndcard", "sm8550"}, {} }; From d0ae9dc48e24f5f704abcbb2dca3e4651bf0ff59 Mon Sep 17 00:00:00 2001 From: Chen-Yu Tsai Date: Mon, 4 Dec 2023 11:35:47 +0800 Subject: [PATCH 154/337] ASoC: SOF: Move sof_of_machine_select() to core.c from sof-of-dev.c Commit 014fdeb0d747 ("ASoC: SOF: Move sof_of_machine_select() to sof-of-dev.c from sof-audio.c") caused a circular dependency between the snd_sof and snd_sof_of modules: depmod: ERROR: Cycle detected: snd_sof -> snd_sof_of -> snd_sof depmod: ERROR: Found 2 modules in dependency cycles! Move the function back with sof_machine_select(). Fixes: 014fdeb0d747 ("ASoC: SOF: Move sof_of_machine_select() to sof-of-dev.c from sof-audio.c") Signed-off-by: Chen-Yu Tsai Acked-by: Peter Ujfalusi Reviewed-by: Daniel Baluta Link: https://lore.kernel.org/r/20231204033549.2020289-1-wenst@chromium.org Signed-off-by: Mark Brown --- sound/soc/sof/core.c | 22 ++++++++++++++++++++++ sound/soc/sof/sof-of-dev.c | 23 ----------------------- sound/soc/sof/sof-of-dev.h | 9 --------- 3 files changed, 22 insertions(+), 32 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index a2afec8f5879..425b023b03b4 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -144,6 +144,28 @@ void sof_set_fw_state(struct snd_sof_dev *sdev, enum sof_fw_state new_state) } EXPORT_SYMBOL(sof_set_fw_state); +static struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) +{ + struct snd_sof_pdata *sof_pdata = sdev->pdata; + const struct sof_dev_desc *desc = sof_pdata->desc; + struct snd_sof_of_mach *mach = desc->of_machines; + + if (!mach) + return NULL; + + for (; mach->compatible; mach++) { + if (of_machine_is_compatible(mach->compatible)) { + sof_pdata->tplg_filename = mach->sof_tplg_filename; + if (mach->fw_filename) + sof_pdata->fw_filename = mach->fw_filename; + + return mach; + } + } + + return NULL; +} + /* SOF Driver enumeration */ static int sof_machine_check(struct snd_sof_dev *sdev) { diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index fa92da5ee9b3..b9a499e92b9a 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -41,29 +41,6 @@ static void sof_of_probe_complete(struct device *dev) pm_runtime_enable(dev); } -struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) -{ - struct snd_sof_pdata *sof_pdata = sdev->pdata; - const struct sof_dev_desc *desc = sof_pdata->desc; - struct snd_sof_of_mach *mach = desc->of_machines; - - if (!mach) - return NULL; - - for (; mach->compatible; mach++) { - if (of_machine_is_compatible(mach->compatible)) { - sof_pdata->tplg_filename = mach->sof_tplg_filename; - if (mach->fw_filename) - sof_pdata->fw_filename = mach->fw_filename; - - return mach; - } - } - - return NULL; -} -EXPORT_SYMBOL(sof_of_machine_select); - int sof_of_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; diff --git a/sound/soc/sof/sof-of-dev.h b/sound/soc/sof/sof-of-dev.h index 547e358a37e3..b6cc70595f3b 100644 --- a/sound/soc/sof/sof-of-dev.h +++ b/sound/soc/sof/sof-of-dev.h @@ -16,15 +16,6 @@ struct snd_sof_of_mach { const char *sof_tplg_filename; }; -#if IS_ENABLED(CONFIG_SND_SOC_SOF_OF_DEV) -struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev); -#else -static inline struct snd_sof_of_mach *sof_of_machine_select(struct snd_sof_dev *sdev) -{ - return NULL; -} -#endif - extern const struct dev_pm_ops sof_of_pm; int sof_of_probe(struct platform_device *pdev); From 9c8bec3b63255ca04e5dad87471a35790aec06dc Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sat, 2 Dec 2023 13:39:43 +0100 Subject: [PATCH 155/337] ASoC: es83xx: add ACPI DSM helper module Most of the ES83xx codec configuration is exposed in the DSDT table and accessible via a _DSM method. Start adding basic definitions and helpers to dump the information. Reviewed-by: Mauro Carvalho Chehab Co-developed-by: David Yang Signed-off-by: David Yang Signed-off-by: Pierre-Louis Bossart Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231202123946.54347-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es83xx-dsm-common.c | 89 ++++++ sound/soc/codecs/es83xx-dsm-common.h | 393 +++++++++++++++++++++++++++ 4 files changed, 488 insertions(+) create mode 100644 sound/soc/codecs/es83xx-dsm-common.c create mode 100644 sound/soc/codecs/es83xx-dsm-common.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3429419ca694..59f9742e9ff4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1076,6 +1076,10 @@ config SND_SOC_ES7134 config SND_SOC_ES7241 tristate "Everest Semi ES7241 CODEC" +config SND_SOC_ES83XX_DSM_COMMON + depends on ACPI + tristate + config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2078bb0d981e..f53baa2b9565 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -116,6 +116,7 @@ snd-soc-da9055-objs := da9055.o snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o +snd-soc-es83xx-dsm-common-objs := es83xx-dsm-common.o snd-soc-es8316-objs := es8316.o snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o @@ -505,6 +506,7 @@ obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o +obj-$(CONFIG_SND_SOC_ES83XX_DSM_COMMON) += snd-soc-es83xx-dsm-common.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o diff --git a/sound/soc/codecs/es83xx-dsm-common.c b/sound/soc/codecs/es83xx-dsm-common.c new file mode 100644 index 000000000000..94fd7d54c53b --- /dev/null +++ b/sound/soc/codecs/es83xx-dsm-common.c @@ -0,0 +1,89 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright (c) Intel Corporation, 2022 +// Copyright Everest Semiconductor Co.,Ltd + +#include +#include +#include "es83xx-dsm-common.h" + +/* UUID ("a9800c04-e016-343e-41f4-6bcce70f4332") */ +static const guid_t es83xx_dsm_guid = + GUID_INIT(0xa9800c04, 0xe016, 0x343e, + 0x41, 0xf4, 0x6b, 0xcc, 0xe7, 0x0f, 0x43, 0x32); + +#define ES83xx_DSM_REVID 1 + +int es83xx_dsm(struct device *dev, int arg, int *value) +{ + acpi_handle dhandle; + union acpi_object *obj; + int ret = 0; + + dhandle = ACPI_HANDLE(dev); + if (!dhandle) + return -ENOENT; + + obj = acpi_evaluate_dsm(dhandle, &es83xx_dsm_guid, ES83xx_DSM_REVID, + arg, NULL); + if (!obj) { + dev_err(dev, "%s: acpi_evaluate_dsm() failed\n", __func__); + ret = -EINVAL; + goto out; + } + + if (obj->type != ACPI_TYPE_INTEGER) { + dev_err(dev, "%s: object is not ACPI_TYPE_INTEGER\n", __func__); + ret = -EINVAL; + goto err; + } + + *value = obj->integer.value; +err: + ACPI_FREE(obj); +out: + return ret; +} +EXPORT_SYMBOL_GPL(es83xx_dsm); + +int es83xx_dsm_dump(struct device *dev) +{ + int value; + int ret; + + ret = es83xx_dsm(dev, PLATFORM_MAINMIC_TYPE_ARG, &value); + if (ret < 0) + return ret; + dev_info(dev, "PLATFORM_MAINMIC_TYPE %#x\n", value); + + ret = es83xx_dsm(dev, PLATFORM_HPMIC_TYPE_ARG, &value); + if (ret < 0) + return ret; + dev_info(dev, "PLATFORM_HPMIC_TYPE %#x\n", value); + + ret = es83xx_dsm(dev, PLATFORM_SPK_TYPE_ARG, &value); + if (ret < 0) + return ret; + dev_info(dev, "PLATFORM_SPK_TYPE %#x\n", value); + + ret = es83xx_dsm(dev, PLATFORM_HPDET_INV_ARG, &value); + if (ret < 0) + return ret; + dev_info(dev, "PLATFORM_HPDET_INV %#x\n", value); + + ret = es83xx_dsm(dev, PLATFORM_PCM_TYPE_ARG, &value); + if (ret < 0) + return ret; + dev_info(dev, "PLATFORM_PCM_TYPE %#x\n", value); + + ret = es83xx_dsm(dev, PLATFORM_MIC_DE_POP_ARG, &value); + if (ret < 0) + return ret; + dev_info(dev, "PLATFORM_MIC_DE_POP %#x\n", value); + + return 0; +} +EXPORT_SYMBOL_GPL(es83xx_dsm_dump); + +MODULE_DESCRIPTION("Everest Semi ES83xx DSM helpers"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es83xx-dsm-common.h b/sound/soc/codecs/es83xx-dsm-common.h new file mode 100644 index 000000000000..91c9a89e75e9 --- /dev/null +++ b/sound/soc/codecs/es83xx-dsm-common.h @@ -0,0 +1,393 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (c) Intel Corporation, 2022 + * Copyright Everest Semiconductor Co.,Ltd + */ + +/* Definitions extracted from ASL file provided at + * https://github.com/thesofproject/linux/files/9398723/ESSX8326.zip + */ + +#ifndef _ES83XX_DSM_COMMON_H +#define _ES83XX_DSM_COMMON_H + +/*************************************************** + * DSM arguments * + ***************************************************/ + +#define PLATFORM_MAINMIC_TYPE_ARG 0x00 +#define PLATFORM_HPMIC_TYPE_ARG 0x01 +#define PLATFORM_SPK_TYPE_ARG 0x02 +#define PLATFORM_HPDET_INV_ARG 0x03 +#define PLATFORM_PCM_TYPE_ARG 0x04 + +#define PLATFORM_MIC_DE_POP_ARG 0x06 +#define PLATFORM_CODEC_TYPE_ARG 0x0E +#define PLATFORM_BUS_SLOT_ARG 0x0F + +#define HP_CODEC_LINEIN_PGA_GAIN_ARG 0x10 +#define MAIN_CODEC_LINEIN_PGA_GAIN_ARG 0x20 + +#define HP_CODEC_D2SEPGA_GAIN_ARG 0x11 +#define MAIN_CODEC_D2SEPGA_GAIN_ARG 0x21 + +#define HP_CODEC_ADC_VOLUME_ARG 0x12 +#define MAIN_CODEC_ADC_VOLUME_ARG 0x22 + +#define HP_CODEC_ADC_ALC_ENABLE_ARG 0x13 +#define MAIN_CODEC_ADC_ALC_ENABLE_ARG 0x23 + +#define HP_CODEC_ADC_ALC_TARGET_LEVEL_ARG 0x14 +#define MAIN_CODEC_ADC_ALC_TARGET_LEVEL_ARG 0x24 + +#define HP_CODEC_ADC_ALC_MAXGAIN_ARG 0x15 +#define MAIN_CODEC_ADC_ALC_MAXGAIN_ARG 0x25 + +#define HP_CODEC_ADC_ALC_MINGAIN_ARG 0x16 +#define MAIN_CODEC_ADC_ALC_MINGAIN_ARG 0x26 + +#define HP_CODEC_ADC_ALC_HLDTIME_ARG 0x17 +#define MAIN_CODEC_ADC_ALC_HLDTIME_ARG 0x27 + +#define HP_CODEC_ADC_ALC_DCYTIME_ARG 0x18 +#define MAIN_CODEC_ADC_ALC_DCYTIME_ARG 0x28 + +#define HP_CODEC_ADC_ALC_ATKTIME_ARG 0x19 +#define MAIN_CODEC_ADC_ALC_ATKTIME_ARG 0x29 + +#define HP_CODEC_ADC_ALC_NGTYPE_ARG 0x1a +#define MAIN_CODEC_ADC_ALC_NGTYPE_ARG 0x2a + +#define HP_CODEC_ADC_ALC_NGTHLD_ARG 0x1b +#define MAIN_CODEC_ADC_ALC_NGTHLD_ARG 0x2b + +#define MAIN_CODEC_ADC_GUI_STEP_ARG 0x2c +#define MAIN_CODEC_ADC_GUI_GAIN_RANGE_ARG 0x2c + +#define HEADPHONE_DUMMY_REMOVE_ENABLE_ARG 0x2e + +#define HP_CODEC_DAC_HPMIX_HIGAIN_ARG 0x40 +#define SPK_CODEC_DAC_HPMIX_HIGAIN_ARG 0x50 + +#define HP_CODEC_DAC_HPMIX_VOLUME_ARG 0x41 +#define SPK_CODEC_DAC_HPMIX_VOLUME_ARG 0x51 + +#define HP_CODEC_DAC_HPOUT_VOLUME_ARG 0x42 +#define SPK_CODEC_DAC_HPOUT_VOLUME_ARG 0x52 + +#define HP_CODEC_LDAC_VOLUME_ARG 0x44 +#define HP_CODEC_RDAC_VOLUME_ARG 0x54 + +#define SPK_CODEC_LDAC_VOLUME_ARG 0x45 +#define SPK_CODEC_RDAC_VOLUME_ARG 0x55 + +#define HP_CODEC_DAC_AUTOMUTE_ARG 0x46 +#define SPK_CODEC_DAC_AUTOMUTE_ARG 0x56 + +#define HP_CODEC_DAC_MONO_ARG 0x4A +#define SPK_CODEC_DAC_MONO_ARG 0x5A + +#define HP_CTL_IO_LEVEL_ARG 0x4B +#define SPK_CTL_IO_LEVEL_ARG 0x5B + +#define CODEC_GPIO0_FUNC_ARG 0x80 +#define CODEC_GPIO1_FUNC_ARG 0x81 +#define CODEC_GPIO2_FUNC_ARG 0x82 +#define CODEC_GPIO3_FUNC_ARG 0x83 +#define CODEC_GPIO4_FUNC_ARG 0x84 + +#define PLATFORM_MCLK_LRCK_FREQ_ARG 0x85 + +/*************************************************** + * Values for arguments * + ***************************************************/ + +/* Main and HP Mic */ +#define PLATFORM_MIC_DMIC_HIGH_LEVEL 0xAA +#define PLATFORM_MIC_DMIC_LOW_LEVEL 0x55 +#define PLATFORM_MIC_AMIC_LIN1RIN1 0xBB +#define PLATFORM_MIC_AMIC_LIN2RIN2 0xCC + +/* Speaker */ +#define PLATFORM_SPK_NONE 0x00 +#define PLATFORM_SPK_MONO 0x01 +#define PLATFORM_SPK_STEREO 0x02 + +/* Jack Detection */ +#define PLATFORM_HPDET_NORMAL 0x00 +#define PLATFORM_HPDET_INVERTED 0x01 + +/* PCM type (Port number + protocol) */ +/* + * RETURNED VALUE = 0x00, PCM PORT0, I2S + * 0x01, PCM PORT0, LJ + * 0x02, PCM PORT0, RJ + * 0x03, PCM PORT0, DSP-A + * 0x04, PCM PORT0, DSP-B + * 0x10, PCM PORT1, I2S + * 0x11, PCM PORT1, LJ + * 0x12, PCM PORT1, RJ + * 0x13, PCM PORT1, DSP-A + * 0x14, PCM PORT1, DSP-B + * 0xFF, Use default + * + * This is not used in Linux (defined by topology) and in + * Windows it's always DSP-A + */ + +/* Depop */ +#define PLATFORM_MIC_DE_POP_OFF 0x00 +#define PLATFORM_MIC_DE_POP_ON 0x01 + +/* Codec type */ +#define PLATFORM_CODEC_8316 16 +#define PLATFORM_CODEC_8326 26 +#define PLATFORM_CODEC_8336 36 +#define PLATFORM_CODEC_8395 95 +#define PLATFORM_CODEC_8396 96 + +/* Bus slot (on the host) */ +/* BIT[3:0] FOR BUS NUMBER, BIT[7:4] FOR SLOT NUMBER + * BIT[3:0] 0 for I2S0, 1 for IS21, 2 for I2S2. + * + * On Intel platforms this refers to SSP0..2. This information + * is not really useful for Linux, the information is already + * inferred from NHLT but can be used to double-check NHLT + */ + +/* Volume - Gain */ +#define LINEIN_GAIN_0db 0x00 /* gain = 0db */ +#define LINEIN_GAIN_3db 0x01 /* gain = +3db */ +#define LINEIN_GAIN_6db 0x02 /* gain = +6db */ +#define LINEIN_GAIN_9db 0x03 /* gain = +9db */ +#define LINEIN_GAIN_12db 0x04 /* gain = +12db */ +#define LINEIN_GAIN_15db 0x05 /* gain = +15db */ +#define LINEIN_GAIN_18db 0x06 /* gain = +18db */ +#define LINEIN_GAIN_21db 0x07 /* gain = +21db */ +#define LINEIN_GAIN_24db 0x08 /* gain = +24db */ +#define LINEIN_GAIN_27db 0x09 /* gain = +27db */ +#define LINEIN_GAIN_30db 0x0a /* gain = +30db */ + +#define ADC_GUI_STEP_3db 0x03 /* gain = +3db */ +#define ADC_GUI_STEP_6db 0x06 /* gain = +6db */ +#define ADC_GUI_STEP_10db 0x0a /* gain = +10db */ + +#define D2SEPGA_GAIN_0db 0x00 /* gain = 0db */ +#define D2SEPGA_GAIN_15db 0x01 /* gain = +15db */ + +/* ADC volume: base = 0db, -0.5db/setp, 0xc0 <-> -96db */ + +#define ADC_ALC_DISABLE 0x00 +#define ADC_ALC_ENABLE 0x01 + +#define ADC_ALC_TARGET_LEVEL_m16_5db 0x00 /* gain = -16.5db */ +#define ADC_ALC_TARGET_LEVEL_m15db 0x01 /* gain = -15db */ +#define ADC_ALC_TARGET_LEVEL_m13_5db 0x02 /* gain = -13.5db */ +#define ADC_ALC_TARGET_LEVEL_m12db 0x03 /* gain = -12db */ +#define ADC_ALC_TARGET_LEVEL_m10_5db 0x04 /* gain = -10.5db */ +#define ADC_ALC_TARGET_LEVEL_m9db 0x05 /* gain = -9db */ +#define ADC_ALC_TARGET_LEVEL_m7_5db 0x06 /* gain = -7.5db */ +#define ADC_ALC_TARGET_LEVEL_m6db 0x07 /* gain = -6db */ +#define ADC_ALC_TARGET_LEVEL_m4_5db 0x08 /* gain = -4.5db */ +#define ADC_ALC_TARGET_LEVEL_m_3db 0x09 /* gain = -3db */ +#define ADC_ALC_TARGET_LEVEL_m1_5db 0x0a /* gain = -1.5db */ + +#define ADC_ALC_MAXGAIN_m6_5db 0x00 /* gain = -6.5db */ +#define ADC_ALC_MAXGAIN_m5db 0x01 /* gain = -5db */ +#define ADC_ALC_MAXGAIN_m3_5db 0x02 /* gain = -3.5db */ +#define ADC_ALC_MAXGAIN_m2db 0x03 /* gain = -2db */ +#define ADC_ALC_MAXGAIN_m0_5db 0x04 /* gain = -0.5db */ +#define ADC_ALC_MAXGAIN_1db 0x05 /* gain = +1db */ +#define ADC_ALC_MAXGAIN_2_5db 0x06 /* gain = +2.5db */ +#define ADC_ALC_MAXGAIN_4db 0x07 /* gain = +4db */ +#define ADC_ALC_MAXGAIN_5_5db 0x08 /* gain = +5.5db */ +#define ADC_ALC_MAXGAIN_7db 0x09 /* gain = +7db */ +#define ADC_ALC_MAXGAIN_8_5db 0x0a /* gain = +8.5db */ +#define ADC_ALC_MAXGAIN_10db 0x0b /* gain = +10db */ +#define ADC_ALC_MAXGAIN_11_5db 0x0c /* gain = +11.5db */ +#define ADC_ALC_MAXGAIN_13db 0x0d /* gain = +13db */ +#define ADC_ALC_MAXGAIN_14_5db 0x0e /* gain = +14.5db */ +#define ADC_ALC_MAXGAIN_16db 0x0f /* gain = +16db */ +#define ADC_ALC_MAXGAIN_17_5db 0x10 /* gain = +17.5db */ +#define ADC_ALC_MAXGAIN_19db 0x11 /* gain = +19db */ +#define ADC_ALC_MAXGAIN_20_5db 0x12 /* gain = +20.5db */ +#define ADC_ALC_MAXGAIN_22db 0x13 /* gain = +22db */ +#define ADC_ALC_MAXGAIN_23_5db 0x14 /* gain = +23.5db */ +#define ADC_ALC_MAXGAIN_25db 0x15 /* gain = +25db */ +#define ADC_ALC_MAXGAIN_26_5db 0x16 /* gain = +26.5db */ +#define ADC_ALC_MAXGAIN_28db 0x17 /* gain = +28db */ +#define ADC_ALC_MAXGAIN_29_5db 0x18 /* gain = +29.5db */ +#define ADC_ALC_MAXGAIN_31db 0x19 /* gain = +31db */ +#define ADC_ALC_MAXGAIN_32_5db 0x1a /* gain = +32.5db */ +#define ADC_ALC_MAXGAIN_34db 0x1b /* gain = +34db */ +#define ADC_ALC_MAXGAIN_35_5db 0x1c /* gain = +35.5db */ + +#define ADC_ALC_MINGAIN_m12db 0x00 /* gain = -12db */ +#define ADC_ALC_MINGAIN_m10_5db 0x01 /* gain = -10.5db */ +#define ADC_ALC_MINGAIN_m9db 0x02 /* gain = -9db */ +#define ADC_ALC_MINGAIN_m7_5db 0x03 /* gain = -7.5db */ +#define ADC_ALC_MINGAIN_m6db 0x04 /* gain = -6db */ +#define ADC_ALC_MINGAIN_m4_51db 0x05 /* gain = -4.51db */ +#define ADC_ALC_MINGAIN_m3db 0x06 /* gain = -3db */ +#define ADC_ALC_MINGAIN_m1_5db 0x07 /* gain = -1.5db */ +#define ADC_ALC_MINGAIN_0db 0x08 /* gain = 0db */ +#define ADC_ALC_MINGAIN_1_5db 0x09 /* gain = +1.5db */ +#define ADC_ALC_MINGAIN_3db 0x0a /* gain = +3db */ +#define ADC_ALC_MINGAIN_4_5db 0x0b /* gain = +4.5db */ +#define ADC_ALC_MINGAIN_6db 0x0c /* gain = +6db */ +#define ADC_ALC_MINGAIN_7_5db 0x0d /* gain = +7.5db */ +#define ADC_ALC_MINGAIN_9db 0x0e /* gain = +9db */ +#define ADC_ALC_MINGAIN_10_5db 0x0f /* gain = +10.5db */ +#define ADC_ALC_MINGAIN_12db 0x10 /* gain = +12db */ +#define ADC_ALC_MINGAIN_13_5db 0x11 /* gain = +13.5db */ +#define ADC_ALC_MINGAIN_15db 0x12 /* gain = +15db */ +#define ADC_ALC_MINGAIN_16_5db 0x13 /* gain = +16.5db */ +#define ADC_ALC_MINGAIN_18db 0x14 /* gain = +18db */ +#define ADC_ALC_MINGAIN_19_5db 0x15 /* gain = +19.5db */ +#define ADC_ALC_MINGAIN_21db 0x16 /* gain = +21db */ +#define ADC_ALC_MINGAIN_22_5db 0x17 /* gain = +22.5db */ +#define ADC_ALC_MINGAIN_24db 0x18 /* gain = +24db */ +#define ADC_ALC_MINGAIN_25_5db 0x19 /* gain = +25.5db */ +#define ADC_ALC_MINGAIN_27db 0x1a /* gain = +27db */ +#define ADC_ALC_MINGAIN_28_5db 0x1b /* gain = +28.5db */ +#define ADC_ALC_MINGAIN_30db 0x1c /* gain = +30db */ + +/* ADC volume: step 1dB */ + +/* ALC Hold, Decay, Attack */ +#define ADC_ALC_HLDTIME_0_US 0x00 +#define ADC_ALC_HLDTIME_0000266_US 0x01 //time = 2.67ms +#define ADC_ALC_HLDTIME_0000533_US 0x02 //time = 5.33ms +#define ADC_ALC_HLDTIME_0001066_US 0x03 //time = 10.66ms +#define ADC_ALC_HLDTIME_0002132_US 0x04 //time = 21.32ms +#define ADC_ALC_HLDTIME_0004264_US 0x05 //time = 42.64ms +#define ADC_ALC_HLDTIME_0008538_US 0x06 //time = 85.38ms +#define ADC_ALC_HLDTIME_0017076_US 0x07 //time = 170.76ms +#define ADC_ALC_HLDTIME_0034152_US 0x08 //time = 341.52ms +#define ADC_ALC_HLDTIME_0680000_US 0x09 //time = 0.68s +#define ADC_ALC_HLDTIME_1360000_US 0x0a //time = 1.36s + +#define ADC_ALC_DCYTIME_000410_US 0x00 //time = 410us +#define ADC_ALC_DCYTIME_000820_US 0x01 //time = 820us +#define ADC_ALC_DCYTIME_001640_US 0x02 //time = 1.64ms +#define ADC_ALC_DCYTIME_003280_US 0x03 //time = 3.28ms +#define ADC_ALC_DCYTIME_006560_US 0x04 //time = 6.56ms +#define ADC_ALC_DCYTIME_013120_US 0x05 //time = 13.12ms +#define ADC_ALC_DCYTIME_026240_US 0x06 //time = 26.24ms +#define ADC_ALC_DCYTIME_058480_US 0x07 //time = 52.48ms +#define ADC_ALC_DCYTIME_104960_US 0x08 //time = 104.96ms +#define ADC_ALC_DCYTIME_209920_US 0x09 //time = 209.92ms +#define ADC_ALC_DCYTIME_420000_US 0x0a //time = 420ms + +#define ADC_ALC_ATKTIME_000104_US 0x00 //time = 104us +#define ADC_ALC_ATKTIME_000208_US 0x01 //time = 208us +#define ADC_ALC_ATKTIME_000416_US 0x02 //time = 416ms +#define ADC_ALC_ATKTIME_003832_US 0x03 //time = 832ms +#define ADC_ALC_ATKTIME_001664_US 0x04 //time = 1.664ms +#define ADC_ALC_ATKTIME_003328_US 0x05 //time = 3.328ms +#define ADC_ALC_ATKTIME_006656_US 0x06 //time = 6.656ms +#define ADC_ALC_ATKTIME_013312_US 0x07 //time = 13.312ms +#define ADC_ALC_ATKTIME_026624_US 0x08 //time = 26.624ms +#define ADC_ALC_ATKTIME_053248_US 0x09 //time = 53.248ms +#define ADC_ALC_ATKTIME_106496_US 0x0a //time = 106.496ms + +/* ALC Noise Gate */ +#define ADC_ALC_NGTYPE_DISABLE 0x00 //noise gate disable +#define ADC_ALC_NGTYPE_ENABLE_HOLD 0x01 //noise gate enable, hold gain type +#define ADC_ALC_NGTYPE_ENABLE_MUTE 0x03 //noise gate enable, mute type + +#define ADC_ALC_NGTHLD_m76_5db 0x00 /* Threshold = -76.5db */ +#define ADC_ALC_NGTHLD_m75db 0x01 /* Threshold = -75db */ +#define ADC_ALC_NGTHLD_m73_5db 0x02 /* Threshold = -73.5db */ +#define ADC_ALC_NGTHLD_m72db 0x03 /* Threshold = -72db */ +#define ADC_ALC_NGTHLD_m70_5db 0x04 /* Threshold = -70.5db */ +#define ADC_ALC_NGTHLD_m69db 0x05 /* Threshold = -69db */ +#define ADC_ALC_NGTHLD_m67_5db 0x06 /* Threshold = -67.5db */ +#define ADC_ALC_NGTHLD_m66db 0x07 /* Threshold = -66db */ +#define ADC_ALC_NGTHLD_m64_5db 0x08 /* Threshold = -64.5db */ +#define ADC_ALC_NGTHLD_m63db 0x09 /* Threshold = -63db */ +#define ADC_ALC_NGTHLD_m61_5db 0x0a /* Threshold = -61.5db */ +#define ADC_ALC_NGTHLD_m60db 0x0b /* Threshold = -60db */ +#define ADC_ALC_NGTHLD_m58_5db 0x0c /* Threshold = -58.5db */ +#define ADC_ALC_NGTHLD_m57db 0x0d /* Threshold = -57db */ +#define ADC_ALC_NGTHLD_m55_5db 0x0e /* Threshold = -55.5db */ +#define ADC_ALC_NGTHLD_m54db 0x0f /* Threshold = -54db */ +#define ADC_ALC_NGTHLD_m52_5db 0x10 /* Threshold = -52.5db */ +#define ADC_ALC_NGTHLD_m51db 0x11 /* Threshold = -51db */ +#define ADC_ALC_NGTHLD_m49_5db 0x12 /* Threshold = -49.5db */ +#define ADC_ALC_NGTHLD_m48db 0x13 /* Threshold = -48db */ +#define ADC_ALC_NGTHLD_m46_5db 0x14 /* Threshold = -46.5db */ +#define ADC_ALC_NGTHLD_m45db 0x15 /* Threshold = -45db */ +#define ADC_ALC_NGTHLD_m43_5db 0x16 /* Threshold = -43.5db */ +#define ADC_ALC_NGTHLD_m42db 0x17 /* Threshold = -42db */ +#define ADC_ALC_NGTHLD_m40_5db 0x18 /* Threshold = -40.5db */ +#define ADC_ALC_NGTHLD_m39db 0x19 /* Threshold = -39db */ +#define ADC_ALC_NGTHLD_m37_5db 0x1a /* Threshold = -37.5db */ +#define ADC_ALC_NGTHLD_m36db 0x1b /* Threshold = -36db */ +#define ADC_ALC_NGTHLD_m34_5db 0x1c /* Threshold = -34.5db */ +#define ADC_ALC_NGTHLD_m33db 0x1d /* Threshold = -33db */ +#define ADC_ALC_NGTHLD_m31_5db 0x1e /* Threshold = -31.5db */ +#define ADC_ALC_NGTHLD_m30db 0x1f /* Threshold = -30db */ + +/* Headphone dummy - Windows Specific flag, not needed for Linux */ + +/* HPMIX HIGAIN and VOLUME */ +#define DAC_HPMIX_HIGAIN_0db 0x00 /* gain = 0db */ +#define DAC_HPMIX_HIGAIN_m6db 0x88 /* gain = -6db */ + +#define DAC_HPMIX_VOLUME_m12db 0x00 /* volume = -12db */ +#define DAC_HPMIX_VOLUME_m10_5db 0x11 /* volume = -10.5db */ +#define DAC_HPMIX_VOLUME_m9db 0x22 /* volume = -9db */ +#define DAC_HPMIX_VOLUME_m7_5db 0x33 /* volume = -7.5db */ +#define DAC_HPMIX_VOLUME_m6db 0x44 /* volume = -6db */ +#define DAC_HPMIX_VOLUME_m4_5db 0x88 /* volume = -4.5db */ +#define DAC_HPMIX_VOLUME_m3db 0x99 /* volume = -3db */ +#define DAC_HPMIX_VOLUME_m1_5db 0xaa /* volume = -1.5db */ +#define DAC_HPMIX_VOLUME_0db 0xbb /* volume = 0db */ + +/* HPOUT VOLUME */ +#define DAC_HPOUT_VOLUME_0db 0x00 /* volume = 0db */ +#define DAC_HPOUT_VOLUME_m12db 0x11 /* volume = -12db */ +#define DAC_HPOUT_VOLUME_m24db 0x22 /* volume = -24db */ +#define DAC_HPOUT_VOLUME_m48db 0x33 /* volume = -48db */ + +/* LDAC/RDAC volume = 0db, -0.5db/setp, 0xc0 <-> -96db */ + +/* Automute */ +#define DAC_AUTOMUTE_NONE 0x00 /* no automute */ +#define DAC_AUTOMUTE_DIGITAL 0x01 /* digital mute */ +#define DAC_AUTOMUTE_ANALOG 0x02 /* analog mute */ + +/* Mono - Windows specific, on Linux the information comes from DAI/topology */ +#define HEADPHONE_MONO 0x01 /* on channel */ +#define HEADPHONE_STEREO 0x00 /* stereo */ + +/* Speaker and headphone GPIO control */ +#define GPIO_CTL_IO_LEVEL_LOW 0x00 /* low level enable */ +#define GPIO_CTL_IO_LEVEL_HIGH 0x01 /* high level enable */ + +/* GPIO */ +/* FIXME: for ES8396, no need to use */ + +/* Platform clocks */ +/* + * BCLK AND MCLK FREQ + * BIT[7:4] MCLK FREQ + * 0 - 19.2MHz + * 1 - 24MHz + * 2 - 12.288MHz + * F - Default for 19.2MHz + * + * BIT[3:0] BCLK FREQ + * 0 - 4.8MHz + * 1 - 2.4MHz + * 2 - 2.304MHz + * 3 - 3.072MHz + * 4 - 4.096MHz + * F - Default for 4.8MHz + */ + +int es83xx_dsm(struct device *dev, int arg, int *value); +int es83xx_dsm_dump(struct device *dev); + +#endif From b71e1d3789946a20e0e34349f4a874604ac65c3e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Sat, 2 Dec 2023 13:39:44 +0100 Subject: [PATCH 156/337] ASoC: Intel: bytcht_es8316: Dump basic _DSM information Instead of adding DMI quirks for each new tablet model which uses the ESS8316 codec, the plan is to switch to querying the same ACPI Device-Specific-Method (DSM) as Windows uses to determine things like speaker and mic routing. Call the new es83xx_dsm_dump() helper which logs various basic settings which can be queried through the ACPI DSM method on the codec ACPI device, this is intended to help with developing a DSM based solution to replace most DMI quirks. Signed-off-by: Pierre-Louis Bossart [hdegoede@redhat.com: improve commit message] Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231202123946.54347-3-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bytcht_es8316.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 99ebe48216ea..8fd5e7f83054 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -229,6 +229,7 @@ config SND_SOC_INTEL_BYT_CHT_ES8316_MACH depends on GPIOLIB || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_ES8316 + select SND_SOC_ES83XX_DSM_COMMON help This adds support for ASoC machine driver for Intel(R) Baytrail & Cherrytrail platforms with ES8316 audio codec. diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 8a0b0e864fbb..e6ab1ee6ab29 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -27,6 +27,7 @@ #include #include #include +#include "../../codecs/es83xx-dsm-common.h" #include "../atom/sst-atom-controls.h" #include "../common/soc-intel-quirks.h" @@ -520,6 +521,8 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) return ret; } + es83xx_dsm_dump(priv->codec_dev); + /* Check for BYTCR or other platform and setup quirks */ dmi_id = dmi_first_match(byt_cht_es8316_quirk_table); if (dmi_id) { From e8acf91a4013202934313f3c2968e6962daaffff Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 2 Dec 2023 13:39:45 +0100 Subject: [PATCH 157/337] ASoC: Intel: bytcht_es8316: Add is_bytcr helper variable Add a is_bytcr helper variable to probe(). This is a preparation patch for determining the quirks through querying the ACPI Device-Specific-Method (DSM) on the codec-device. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231202123946.54347-4-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index e6ab1ee6ab29..0d6574678317 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -471,10 +471,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) struct byt_cht_es8316_private *priv; const struct dmi_system_id *dmi_id; struct fwnode_handle *fwnode; + bool sof_parent, is_bytcr; const char *platform_name; struct acpi_device *adev; struct device *codec_dev; - bool sof_parent; unsigned int cnt = 0; int dai_index = 0; int i; @@ -524,11 +524,11 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) es83xx_dsm_dump(priv->codec_dev); /* Check for BYTCR or other platform and setup quirks */ + is_bytcr = soc_intel_is_byt() && mach->mach_params.acpi_ipc_irq_index == 0; dmi_id = dmi_first_match(byt_cht_es8316_quirk_table); if (dmi_id) { quirk = (unsigned long)dmi_id->driver_data; - } else if (soc_intel_is_byt() && - mach->mach_params.acpi_ipc_irq_index == 0) { + } else if (is_bytcr) { /* On BYTCR default to SSP0, internal-mic-in2-map, mono-spk */ quirk = BYT_CHT_ES8316_SSP0 | BYT_CHT_ES8316_INTMIC_IN2_MAP | BYT_CHT_ES8316_MONO_SPEAKER; From 7650862f4e72d2533356ec001b8ea8d5839aced0 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sat, 2 Dec 2023 13:39:46 +0100 Subject: [PATCH 158/337] ASoC: Intel: bytcht_es8316: Determine quirks/routing with codec-dev ACPI DSM Add support for querying the same ACPI Device-Specific-Method (DSM) as Windows uses to determine things like speaker and mic routing. This avoids the need to add DMI quirks for each new ESS8316 tablet model. This has been tested on the following devices: 1. Chuwi Hi12 CHT with stereo speakers and IN2-mic-map, this avoids the need to add a DMI quirk for this model. 2. Nanote UMPC-01 CHT with stereo speakers and IN1-mic-map, the existing DMI quirk is still necessary because of a bug in the DSM return values for the speakers (it returns mono). 3. Onda V80 plus CHT with mono speaker and IN1-mic-map, DSM set quirks match the previously used defaults. 4. GP-electronic T701 BYT with mono speaker and IN2-mic-map, DSM set quirks match the previously used defaults. Signed-off-by: Hans de Goede Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231202123946.54347-5-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_es8316.c | 62 ++++++++++++++++++++++++++ 1 file changed, 62 insertions(+) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 0d6574678317..1564a88a885e 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -462,6 +462,66 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { {} }; +static int byt_cht_es8316_get_quirks_from_dsm(struct byt_cht_es8316_private *priv, + bool is_bytcr) +{ + int ret, val1, val2, dsm_quirk = 0; + + if (is_bytcr) + dsm_quirk |= BYT_CHT_ES8316_SSP0; + + ret = es83xx_dsm(priv->codec_dev, PLATFORM_MAINMIC_TYPE_ARG, &val1); + if (ret < 0) + return ret; + + ret = es83xx_dsm(priv->codec_dev, PLATFORM_HPMIC_TYPE_ARG, &val2); + if (ret < 0) + return ret; + + if (val1 == PLATFORM_MIC_AMIC_LIN1RIN1 && val2 == PLATFORM_MIC_AMIC_LIN2RIN2) { + dsm_quirk |= BYT_CHT_ES8316_INTMIC_IN1_MAP; + } else if (val1 == PLATFORM_MIC_AMIC_LIN2RIN2 && val2 == PLATFORM_MIC_AMIC_LIN1RIN1) { + dsm_quirk |= BYT_CHT_ES8316_INTMIC_IN2_MAP; + } else { + dev_warn(priv->codec_dev, "Unknown mic settings mainmic 0x%02x hpmic 0x%02x\n", + val1, val2); + return -EINVAL; + } + + ret = es83xx_dsm(priv->codec_dev, PLATFORM_SPK_TYPE_ARG, &val1); + if (ret < 0) + return ret; + + switch (val1) { + case PLATFORM_SPK_MONO: + dsm_quirk |= BYT_CHT_ES8316_MONO_SPEAKER; + break; + case PLATFORM_SPK_STEREO: + break; + default: + dev_warn(priv->codec_dev, "Unknown speaker setting 0x%02x\n", val1); + return -EINVAL; + } + + ret = es83xx_dsm(priv->codec_dev, PLATFORM_HPDET_INV_ARG, &val1); + if (ret < 0) + return ret; + + switch (val1) { + case PLATFORM_HPDET_NORMAL: + break; + case PLATFORM_HPDET_INVERTED: + dsm_quirk |= BYT_CHT_ES8316_JD_INVERTED; + break; + default: + dev_warn(priv->codec_dev, "Unknown hpdet-inv setting 0x%02x\n", val1); + return -EINVAL; + } + + quirk = dsm_quirk; + return 0; +} + static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; @@ -528,6 +588,8 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) dmi_id = dmi_first_match(byt_cht_es8316_quirk_table); if (dmi_id) { quirk = (unsigned long)dmi_id->driver_data; + } else if (!byt_cht_es8316_get_quirks_from_dsm(priv, is_bytcr)) { + dev_info(dev, "Using ACPI DSM info for quirks\n"); } else if (is_bytcr) { /* On BYTCR default to SSP0, internal-mic-in2-map, mono-spk */ quirk = BYT_CHT_ES8316_SSP0 | BYT_CHT_ES8316_INTMIC_IN2_MAP | From e17999750649c4bd4ba945419b406d1d1a3e92e2 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 4 Dec 2023 13:56:14 +0000 Subject: [PATCH 159/337] ASoC: Intel: soc-acpi-intel-tgl-match: add cs42l43 and cs35l56 support This is a test configuration for UpExtreme with Cirrus Logic CS35L56-EIGHT-C board. The codec layout is configured as: - Link3: CS42L43 Jack - Link0: 2x CS35L56 Speaker (amps 1 and 2) - Link1: 2x CS35L56 Speaker (amps 7 and 8) Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Peter Ujfalusi Signed-off-by: Richard Fitzgerald Link: https://lore.kernel.org/r/20231204135614.2169624-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-tgl-match.c | 78 +++++++++++++++++++ 1 file changed, 78 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 5804926c8b56..e5f721ba5ed4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -41,6 +41,20 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = { .group_id = 1, }; +static const struct snd_soc_acpi_endpoint spk_2_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 2, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_3_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 3, + .group_id = 1, +}; + static const struct snd_soc_acpi_endpoint rt712_endpoints[] = { { .num = 0, @@ -400,6 +414,64 @@ static const struct snd_soc_acpi_link_adr tgl_712_only[] = { {} }; +static const struct snd_soc_acpi_adr_device cs42l43_3_adr[] = { + { + .adr = 0x00033001FA424301ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "cs42l43" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_0_adr[] = { + { + .adr = 0x00003301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00003201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP2" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_1_adr[] = { + { + .adr = 0x00013701FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP8" + }, + { + .adr = 0x00013601FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP7" + } +}; + +static const struct snd_soc_acpi_link_adr tgl_cs42l43_cs35l56[] = { + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs42l43_3_adr), + .adr_d = cs42l43_3_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs35l56_0_adr), + .adr_d = cs35l56_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(cs35l56_1_adr), + .adr_d = cs35l56_1_adr, + }, + {} +}; + static const struct snd_soc_acpi_codecs tgl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} @@ -494,6 +566,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-tgl-rt715-rt711-rt1308-mono.tplg", }, + { + .link_mask = 0xB, + .links = tgl_cs42l43_cs35l56, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-tgl-cs42l43-l3-cs35l56-l01.tplg", + }, { .link_mask = 0xF, /* 4 active links required */ .links = tgl_3_in_1_default, From 528ee84f0fe08788b2e72aad7bc8a13dd2a169ca Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 4 Dec 2023 15:41:56 -0600 Subject: [PATCH 160/337] ASoC: Intel: sof_nau8825: board id cleanup for adl boards Many board configs are duplicated since codec and amplifier type are removed from board quirk. Introduce "adl_nau8825_def" board to reduce the number of adl board configs. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214200.203100-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 28 +------------------ .../intel/common/soc-acpi-intel-adl-match.c | 8 +++--- 2 files changed, 5 insertions(+), 31 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 15ea6732ff94..eb3813400593 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -353,33 +353,7 @@ static const struct platform_device_id board_ids[] = { SOF_NAU8825_NUM_HDMIDEV(4)), }, { - .name = "adl_max98373_8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - /* The limitation of length of char array, shorten the name */ - .name = "adl_mx98360a_8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - - }, - { - .name = "adl_rt1015p_8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "adl_nau8318_8825", + .name = "adl_nau8825_def", .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | SOF_NAU8825_SSP_AMP(1) | SOF_NAU8825_NUM_HDMIDEV(4) | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 6e712ad954c8..d3d913458c60 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -528,14 +528,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10508825", - .drv_name = "adl_max98373_8825", + .drv_name = "adl_nau8825_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98373_amp, .sof_tplg_filename = "sof-adl-max98373-nau8825.tplg", }, { .id = "10508825", - .drv_name = "adl_mx98360a_8825", + .drv_name = "adl_nau8825_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98360a_amp, .sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg", @@ -549,14 +549,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10508825", - .drv_name = "adl_rt1015p_8825", + .drv_name = "adl_nau8825_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_rt1015p_amp, .sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg", }, { .id = "10508825", - .drv_name = "adl_nau8318_8825", + .drv_name = "adl_nau8825_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_nau8318_amp, .sof_tplg_filename = "sof-adl-nau8318-nau8825.tplg", From 996727aad856e55f6e65fdc6093281e51f0534da Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 4 Dec 2023 15:41:57 -0600 Subject: [PATCH 161/337] ASoC: Intel: sof_nau8825: board id cleanup for rpl boards Many board configs are duplicated since codec and amplifier type are removed from board quirk. Introduce "rpl_nau8825_def" board to reduce the number of rpl board configs. Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214200.203100-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_nau8825.c | 18 +----------------- .../intel/common/soc-acpi-intel-rpl-match.c | 6 +++--- 2 files changed, 4 insertions(+), 20 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index eb3813400593..719c2fbaf515 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -361,23 +361,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "rpl_max98373_8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "rpl_mx98360a_8825", - .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | - SOF_NAU8825_SSP_AMP(1) | - SOF_NAU8825_NUM_HDMIDEV(4) | - SOF_BT_OFFLOAD_SSP(2) | - SOF_SSP_BT_OFFLOAD_PRESENT), - }, - { - .name = "rpl_nau8318_8825", + .name = "rpl_nau8825_def", .driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) | SOF_NAU8825_SSP_AMP(1) | SOF_NAU8825_NUM_HDMIDEV(4) | diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c index 5b6f57e3a583..c0a643f4725a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -402,21 +402,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { }, { .id = "10508825", - .drv_name = "rpl_max98373_8825", + .drv_name = "rpl_nau8825_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rpl_max98373_amp, .sof_tplg_filename = "sof-rpl-max98373-nau8825.tplg", }, { .id = "10508825", - .drv_name = "rpl_mx98360a_8825", + .drv_name = "rpl_nau8825_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rpl_max98360a_amp, .sof_tplg_filename = "sof-rpl-max98360a-nau8825.tplg", }, { .id = "10508825", - .drv_name = "rpl_nau8318_8825", + .drv_name = "rpl_nau8825_def", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &rpl_nau8318_amp, .sof_tplg_filename = "sof-rpl-nau8318-nau8825.tplg", From 486ede0df82dd74472c6f5651e38ff48f7f766c1 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Mon, 4 Dec 2023 15:41:58 -0600 Subject: [PATCH 162/337] ASoC: Intel: glk_rt5682_max98357a: fix board id mismatch The drv_name in enumeration table for ALC5682I-VS codec does not match the board id string in machine driver. Modify the entry of "10EC5682" to enumerate "RTL5682" as well and remove invalid entry. Fixes: 88b4d77d6035 ("ASoC: Intel: glk_rt5682_max98357a: support ALC5682I-VS codec") Reported-by: Curtis Malainey Reviewed-by: Curtis Malainey Reviewed-by: Bard Liao Signed-off-by: Brent Lu Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214200.203100-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-glk-match.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-glk-match.c b/sound/soc/intel/common/soc-acpi-intel-glk-match.c index 387e73100884..8911c90bbaf6 100644 --- a/sound/soc/intel/common/soc-acpi-intel-glk-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-glk-match.c @@ -19,6 +19,11 @@ static const struct snd_soc_acpi_codecs glk_codecs = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs glk_rt5682_rt5682s_hp = { + .num_codecs = 2, + .codecs = {"10EC5682", "RTL5682"}, +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { { .id = "INT343A", @@ -35,20 +40,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_glk_machines[] = { .sof_tplg_filename = "sof-glk-da7219.tplg", }, { - .id = "10EC5682", + .comp_ids = &glk_rt5682_rt5682s_hp, .drv_name = "glk_rt5682_mx98357a", .fw_filename = "intel/dsp_fw_glk.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &glk_codecs, .sof_tplg_filename = "sof-glk-rt5682.tplg", }, - { - .id = "RTL5682", - .drv_name = "glk_rt5682_max98357a", - .machine_quirk = snd_soc_acpi_codec_list, - .quirk_data = &glk_codecs, - .sof_tplg_filename = "sof-glk-rt5682.tplg", - }, { .id = "10134242", .drv_name = "glk_cs4242_mx98357a", From e38e252dbceeef7d2f848017132efd68e9ae1416 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 4 Dec 2023 15:41:59 -0600 Subject: [PATCH 163/337] ASoC: Intel: sof_sdw_rt_sdca_jack_common: ctx->headset_codec_dev = NULL MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit sof_sdw_rt_sdca_jack_exit() are used by different codecs, and some of them use the same dai name. For example, rt712 and rt713 both use "rt712-sdca-aif1" and sof_sdw_rt_sdca_jack_exit(). As a result, sof_sdw_rt_sdca_jack_exit() will be called twice by mc_dailink_exit_loop(). Set ctx->headset_codec_dev = NULL; after put_device(ctx->headset_codec_dev); to avoid ctx->headset_codec_dev being put twice. Fixes: 5360c6704638 ("ASoC: Intel: sof_sdw: add rt712 support") Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214200.203100-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index e430be7681d2..1e5725f0ae33 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -176,6 +176,7 @@ int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link device_remove_software_node(ctx->headset_codec_dev); put_device(ctx->headset_codec_dev); + ctx->headset_codec_dev = NULL; return 0; } From 70a6b66d6e8e70966274cab2fc9ee75fd60e36bf Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 4 Dec 2023 15:42:00 -0600 Subject: [PATCH 164/337] ASoC: Intel: sof_sdw_rt_sdca_jack_common: check ctx->headset_codec_dev instead of playback MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit 'if (!playback)' will not work if the dai is only on capture dai link or is on more than one playback dai links. Check 'if (ctx->headset_codec_dev)' instead. Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214200.203100-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 1e5725f0ae33..d9c283829fc7 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -192,10 +192,10 @@ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, int ret; /* - * headset should be initialized once. - * Do it with dai link for playback. + * Jack detection should be only initialized once for headsets since + * the playback/capture is sharing the same jack */ - if (!playback) + if (ctx->headset_codec_dev) return 0; sdw_dev = bus_find_device_by_name(&sdw_bus_type, NULL, dai_links->codecs[0].name); From 2f03970198d6438d95b96f69041254bd39aafed0 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Mon, 4 Dec 2023 15:47:10 -0600 Subject: [PATCH 165/337] ASoC: SOF: topology: Use partial match for disconnecting DAI link and DAI widget We use partial match for connecting DAI link and DAI widget. We need to use partial match for disconnecting, too. Fixes: fe88788779fc ("ASoC: SOF: topology: Use partial match for connecting DAI link and DAI widget") Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214713.208951-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9f717366cddc..c1f66ba0e987 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1135,7 +1135,7 @@ static void sof_disconnect_dai_widget(struct snd_soc_component *scomp, list_for_each_entry(rtd, &card->rtd_list, list) { /* does stream match DAI link ? */ if (!rtd->dai_link->stream_name || - strcmp(sname, rtd->dai_link->stream_name)) + !strstr(rtd->dai_link->stream_name, sname)) continue; for_each_rtd_cpu_dais(rtd, i, cpu_dai) From 6c4df324d78ceb6a3ebb0d42243d131a424c85c3 Mon Sep 17 00:00:00 2001 From: Baofeng Tian Date: Mon, 4 Dec 2023 15:47:11 -0600 Subject: [PATCH 166/337] ASoC: SOF: align topology header file with sof topology header Add missed definition and align variable names with sof topology header file. Signed-off-by: Baofeng Tian Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214713.208951-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/topology.h | 25 ++++++++++++++++--------- 1 file changed, 16 insertions(+), 9 deletions(-) diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h index 906e2f327ad2..466ab8e73d98 100644 --- a/include/sound/sof/topology.h +++ b/include/sound/sof/topology.h @@ -39,6 +39,7 @@ enum sof_comp_type { SOF_COMP_ASRC, /**< Asynchronous sample rate converter */ SOF_COMP_DCBLOCK, SOF_COMP_SMART_AMP, /**< smart amplifier component */ + SOF_COMP_MODULE_ADAPTER, /**< module adapter */ /* keep FILEREAD/FILEWRITE as the last ones */ SOF_COMP_FILEREAD = 10000, /**< host test based file IO */ SOF_COMP_FILEWRITE = 10001, /**< host test based file IO */ @@ -68,14 +69,15 @@ struct sof_ipc_comp { /* * SOF memory capabilities, add new ones at the end */ -#define SOF_MEM_CAPS_RAM (1 << 0) -#define SOF_MEM_CAPS_ROM (1 << 1) -#define SOF_MEM_CAPS_EXT (1 << 2) /**< external */ -#define SOF_MEM_CAPS_LP (1 << 3) /**< low power */ -#define SOF_MEM_CAPS_HP (1 << 4) /**< high performance */ -#define SOF_MEM_CAPS_DMA (1 << 5) /**< DMA'able */ -#define SOF_MEM_CAPS_CACHE (1 << 6) /**< cacheable */ -#define SOF_MEM_CAPS_EXEC (1 << 7) /**< executable */ +#define SOF_MEM_CAPS_RAM BIT(0) +#define SOF_MEM_CAPS_ROM BIT(1) +#define SOF_MEM_CAPS_EXT BIT(2) /**< external */ +#define SOF_MEM_CAPS_LP BIT(3) /**< low power */ +#define SOF_MEM_CAPS_HP BIT(4) /**< high performance */ +#define SOF_MEM_CAPS_DMA BIT(5) /**< DMA'able */ +#define SOF_MEM_CAPS_CACHE BIT(6) /**< cacheable */ +#define SOF_MEM_CAPS_EXEC BIT(7) /**< executable */ +#define SOF_MEM_CAPS_L3 BIT(8) /**< L3 memory */ /* * overrun will cause ring buffer overwrite, instead of XRUN. @@ -87,6 +89,9 @@ struct sof_ipc_comp { */ #define SOF_BUF_UNDERRUN_PERMITTED BIT(1) +/* the UUID size in bytes, shared between FW and host */ +#define SOF_UUID_SIZE 16 + /* create new component buffer - SOF_IPC_TPLG_BUFFER_NEW */ struct sof_ipc_buffer { struct sof_ipc_comp comp; @@ -140,6 +145,8 @@ enum sof_volume_ramp { SOF_VOLUME_LOG, SOF_VOLUME_LINEAR_ZC, SOF_VOLUME_LOG_ZC, + SOF_VOLUME_WINDOWS_FADE, + SOF_VOLUME_WINDOWS_NO_FADE, }; /* generic volume component */ @@ -234,7 +241,7 @@ struct sof_ipc_comp_process { /* reserved for future use */ uint32_t reserved[7]; - uint8_t data[]; + unsigned char data[]; } __packed; /* frees components, buffers and pipelines From 8ec56af3da4dad008ab10115eb4cec34836afc04 Mon Sep 17 00:00:00 2001 From: Baofeng Tian Date: Mon, 4 Dec 2023 15:47:12 -0600 Subject: [PATCH 167/337] ASoC: SOF: add alignment for topology header file struct definition sof header file requires these struct with 4 byte aligned, so add same alignment in sof driver definition. Signed-off-by: Baofeng Tian Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214713.208951-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/topology.h | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h index 466ab8e73d98..b3ca886fa28f 100644 --- a/include/sound/sof/topology.h +++ b/include/sound/sof/topology.h @@ -60,7 +60,7 @@ struct sof_ipc_comp { /* extended data length, 0 if no extended data */ uint32_t ext_data_length; -} __packed; +} __packed __aligned(4); /* * Component Buffers @@ -99,7 +99,7 @@ struct sof_ipc_buffer { uint32_t caps; /**< SOF_MEM_CAPS_ */ uint32_t flags; /**< SOF_BUF_ flags defined above */ uint32_t reserved; /**< reserved for future use */ -} __packed; +} __packed __aligned(4); /* generic component config data - must always be after struct sof_ipc_comp */ struct sof_ipc_comp_config { @@ -112,7 +112,7 @@ struct sof_ipc_comp_config { /* reserved for future use */ uint32_t reserved[2]; -} __packed; +} __packed __aligned(4); /* generic host component */ struct sof_ipc_comp_host { @@ -121,7 +121,7 @@ struct sof_ipc_comp_host { uint32_t direction; /**< SOF_IPC_STREAM_ */ uint32_t no_irq; /**< don't send periodic IRQ to host/DSP */ uint32_t dmac_config; /**< DMA engine specific */ -} __packed; +} __packed __aligned(4); /* generic DAI component */ struct sof_ipc_comp_dai { @@ -131,13 +131,13 @@ struct sof_ipc_comp_dai { uint32_t dai_index; /**< index of this type dai */ uint32_t type; /**< DAI type - SOF_DAI_ */ uint32_t reserved; /**< reserved */ -} __packed; +} __packed __aligned(4); /* generic mixer component */ struct sof_ipc_comp_mixer { struct sof_ipc_comp comp; struct sof_ipc_comp_config config; -} __packed; +} __packed __aligned(4); /* volume ramping types */ enum sof_volume_ramp { @@ -158,7 +158,7 @@ struct sof_ipc_comp_volume { uint32_t max_value; uint32_t ramp; /**< SOF_VOLUME_ */ uint32_t initial_ramp; /**< ramp space in ms */ -} __packed; +} __packed __aligned(4); /* generic SRC component */ struct sof_ipc_comp_src { @@ -168,7 +168,7 @@ struct sof_ipc_comp_src { uint32_t source_rate; /**< source rate or 0 for variable */ uint32_t sink_rate; /**< sink rate or 0 for variable */ uint32_t rate_mask; /**< SOF_RATE_ supported rates */ -} __packed; +} __packed __aligned(4); /* generic ASRC component */ struct sof_ipc_comp_asrc { @@ -194,13 +194,13 @@ struct sof_ipc_comp_asrc { /* reserved for future use */ uint32_t reserved[4]; -} __attribute__((packed)); +} __packed __aligned(4); /* generic MUX component */ struct sof_ipc_comp_mux { struct sof_ipc_comp comp; struct sof_ipc_comp_config config; -} __packed; +} __packed __aligned(4); /* generic tone generator component */ struct sof_ipc_comp_tone { @@ -215,7 +215,7 @@ struct sof_ipc_comp_tone { int32_t period; int32_t repeats; int32_t ramp_step; -} __packed; +} __packed __aligned(4); /** \brief Types of processing components */ enum sof_ipc_process_type { @@ -242,7 +242,7 @@ struct sof_ipc_comp_process { uint32_t reserved[7]; unsigned char data[]; -} __packed; +} __packed __aligned(4); /* frees components, buffers and pipelines * SOF_IPC_TPLG_COMP_FREE, SOF_IPC_TPLG_PIPE_FREE, SOF_IPC_TPLG_BUFFER_FREE @@ -250,13 +250,13 @@ struct sof_ipc_comp_process { struct sof_ipc_free { struct sof_ipc_cmd_hdr hdr; uint32_t id; -} __packed; +} __packed __aligned(4); struct sof_ipc_comp_reply { struct sof_ipc_reply rhdr; uint32_t id; uint32_t offset; -} __packed; +} __packed __aligned(4); /* * Pipeline @@ -281,25 +281,25 @@ struct sof_ipc_pipe_new { uint32_t frames_per_sched;/**< output frames of pipeline, 0 is variable */ uint32_t xrun_limit_usecs; /**< report xruns greater than limit */ uint32_t time_domain; /**< scheduling time domain */ -} __packed; +} __packed __aligned(4); /* pipeline construction complete - SOF_IPC_TPLG_PIPE_COMPLETE */ struct sof_ipc_pipe_ready { struct sof_ipc_cmd_hdr hdr; uint32_t comp_id; -} __packed; +} __packed __aligned(4); struct sof_ipc_pipe_free { struct sof_ipc_cmd_hdr hdr; uint32_t comp_id; -} __packed; +} __packed __aligned(4); /* connect two components in pipeline - SOF_IPC_TPLG_COMP_CONNECT */ struct sof_ipc_pipe_comp_connect { struct sof_ipc_cmd_hdr hdr; uint32_t source_id; uint32_t sink_id; -} __packed; +} __packed __aligned(4); /* external events */ enum sof_event_types { From ebd12b2ca6145550a7e42cd2320870db02dd0f3c Mon Sep 17 00:00:00 2001 From: Curtis Malainey Date: Mon, 4 Dec 2023 15:47:13 -0600 Subject: [PATCH 168/337] ASoC: SOF: Wire up buffer flags Buffer flags have been in firmware for ages but were never fully implemented in the topology/kernel system. This commit finishes off the implementation. Reviewed-by: Ranjani Sridharan Signed-off-by: Curtis Malainey Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214713.208951-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/uapi/sound/sof/tokens.h | 1 + sound/soc/sof/ipc3-topology.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h index 0fb39780f9bd..ee5708934614 100644 --- a/include/uapi/sound/sof/tokens.h +++ b/include/uapi/sound/sof/tokens.h @@ -35,6 +35,7 @@ /* buffers */ #define SOF_TKN_BUF_SIZE 100 #define SOF_TKN_BUF_CAPS 101 +#define SOF_TKN_BUF_FLAGS 102 /* DAI */ /* Token retired with ABI 3.2, do not use for new capabilities diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 7a4932c152a9..a8e0054cb8a6 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -72,6 +72,8 @@ static const struct sof_topology_token buffer_tokens[] = { offsetof(struct sof_ipc_buffer, size)}, {SOF_TKN_BUF_CAPS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc_buffer, caps)}, + {SOF_TKN_BUF_FLAGS, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, + offsetof(struct sof_ipc_buffer, flags)}, }; /* DAI */ From 6da9a662154c8d55fd39820ca882d0646d526e53 Mon Sep 17 00:00:00 2001 From: Chao Song Date: Mon, 4 Dec 2023 15:37:21 -0600 Subject: [PATCH 169/337] ASoC: rt722-sdca: Set lane_control_support for multilane The RT722 SDCA codec supports 3 data lanes, lane_control_support property has to be set to use additional two lanes. Reviewed-by: Jack Yu Reviewed-by: Rander Wang Signed-off-by: Chao Song Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204213721.197785-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt722-sdca-sdw.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index a38ec5862214..e24b9cbdc10c 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -256,6 +256,9 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave) /* wake-up event */ prop->wake_capable = 1; + /* Three data lanes are supported by rt722-sdca codec */ + prop->lane_control_support = true; + return 0; } From 0be9595d8a1170474867b8ee2caf14394db45d8b Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Tue, 5 Dec 2023 10:17:40 +0000 Subject: [PATCH 170/337] ASoC: cs4271: Fix spelling mistake "retrieveing" -> "retrieving" There is a spelling mistake in a dev_err_probe error message. Fix it. Signed-off-by: Colin Ian King Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20231205101740.2820813-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 74a84832d958..e864188ae5eb 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -643,7 +643,7 @@ static int cs4271_common_probe(struct device *dev, cs4271->reset = devm_gpiod_get_optional(dev, "reset", GPIOD_ASIS); if (IS_ERR(cs4271->reset)) return dev_err_probe(dev, PTR_ERR(cs4271->reset), - "error retrieveing RESET GPIO\n"); + "error retrieving RESET GPIO\n"); gpiod_set_consumer_name(cs4271->reset, "CS4271 Reset"); for (i = 0; i < ARRAY_SIZE(supply_names); i++) From f31c166a5027e927e5c032d40ef2e484d9ecd612 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Mon, 4 Dec 2023 15:44:07 -0600 Subject: [PATCH 171/337] ASoC: SOF: Intel: lnl: add core get and set support for dsp core Driver uses get and set ops to change the power state of dsp core. Closes: https://github.com/thesofproject/sof/issues/8478 Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231204214407.208528-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 4 ++++ sound/soc/sof/intel/mtl.c | 4 ++-- sound/soc/sof/intel/mtl.h | 3 +++ 3 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 03308721ebd4..a095f5bcf50d 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -120,6 +120,10 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) sof_lnl_ops.get_stream_position = mtl_dsp_get_stream_hda_link_position; + /* dsp core get/put */ + sof_lnl_ops.core_get = mtl_dsp_core_get; + sof_lnl_ops.core_put = mtl_dsp_core_put; + sdev->private = kzalloc(sizeof(struct sof_ipc4_fw_data), GFP_KERNEL); if (!sdev->private) return -ENOMEM; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index f941e2c49d78..3121a89219dd 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -638,7 +638,7 @@ u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, return ((u64)llp_u << 32) | llp_l; } -static int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) +int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) { const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; @@ -651,7 +651,7 @@ static int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core) return 0; } -static int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core) +int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core) { const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm; int ret; diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index 95696b3d7c4c..cc5a1f46fd09 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -106,3 +106,6 @@ void mtl_ipc_dump(struct snd_sof_dev *sdev); u64 mtl_dsp_get_stream_hda_link_position(struct snd_sof_dev *sdev, struct snd_soc_component *component, struct snd_pcm_substream *substream); + +int mtl_dsp_core_get(struct snd_sof_dev *sdev, int core); +int mtl_dsp_core_put(struct snd_sof_dev *sdev, int core); From 138a4e2a26ec73197e22fe64ee3957b1594eabb3 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 5 Dec 2023 13:50:01 +0000 Subject: [PATCH 172/337] ASoC: Intel: sof_sdw_cs_amp: Connect outputs to a speaker widget Hookup the CS35L56 DAPM_OUTPUT widgets to a DAPM_SPK widget so that there is a complete logical path to a speaker. There is no particular reason to use multiple speaker widgets. The CS35L56 are designed to work together as a set so they have all been connected to a single speaker widget. Instead of a hardcoded list of codec widget names, the code walks through all the codecs on the dailink and for every cs35l56 it uses its name prefix to construct the source end of the route. This adds a small amount of overhead during probe but has the benefit that it isn't dependent on every system using the same prefixes. Signed-off-by: Richard Fitzgerald Acked-by: Pierre-Louis Bossart Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20231205135001.2506070-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_cs_amp.c | 30 +++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index 98f6546f484b..f88c01552a92 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -9,15 +9,24 @@ #include #include #include +#include #include "sof_sdw_common.h" #define CODEC_NAME_SIZE 8 +static const struct snd_soc_dapm_widget sof_widgets[] = { + SND_SOC_DAPM_SPK("Speakers", NULL), +}; + static int cs_spk_init(struct snd_soc_pcm_runtime *rtd) { const char *dai_name = rtd->dai_link->codecs->dai_name; struct snd_soc_card *card = rtd->card; char codec_name[CODEC_NAME_SIZE]; + char widget_name[16]; + struct snd_soc_dapm_route route = { "Speakers", NULL, widget_name }; + struct snd_soc_dai *codec_dai; + int i, ret; snprintf(codec_name, CODEC_NAME_SIZE, "%s", dai_name); card->components = devm_kasprintf(card->dev, GFP_KERNEL, @@ -26,17 +35,34 @@ static int cs_spk_init(struct snd_soc_pcm_runtime *rtd) if (!card->components) return -ENOMEM; + ret = snd_soc_dapm_new_controls(&card->dapm, sof_widgets, + ARRAY_SIZE(sof_widgets)); + if (ret) { + dev_err(card->dev, "widgets addition failed: %d\n", ret); + return ret; + } + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (!strstr(codec_dai->name, "cs35l56")) + continue; + + snprintf(widget_name, sizeof(widget_name), "%s SPK", + codec_dai->component->name_prefix); + ret = snd_soc_dapm_add_routes(&card->dapm, &route, 1); + if (ret) + return ret; + } + return 0; } - int sof_sdw_cs_amp_init(struct snd_soc_card *card, const struct snd_soc_acpi_link_adr *link, struct snd_soc_dai_link *dai_links, struct sof_sdw_codec_info *info, bool playback) { - /* Count amp number and do init on playback link only. */ + /* Do init on playback link only. */ if (!playback) return 0; From a6b5f50fefe93676af8798ecc1f633581d1702f8 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 6 Dec 2023 08:30:47 -0300 Subject: [PATCH 173/337] ASoC: dt-bindings: fsl,xcvr: Adjust the number of interrupts Unlike i.MX8MP, i.MX93 has two XCVR interrupts. Describe the two interrupts for the i.MX93 to fix the following dt-schema warning: imx93-11x11-evk.dtb: xcvr@42680000: interrupts: [[0, 203, 4], [0, 204, 4]] is too long from schema $id: http://devicetree.org/schemas/sound/fsl,xcvr.yaml# Signed-off-by: Fabio Estevam Acked-by: Conor Dooley Link: https://lore.kernel.org/r/20231206113047.2240055-1-festevam@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,xcvr.yaml | 22 ++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml index 799b362ba498..0eb0c1ba8710 100644 --- a/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,xcvr.yaml @@ -38,7 +38,10 @@ properties: - const: txfifo interrupts: - maxItems: 1 + items: + - description: WAKEUPMIX Audio XCVR Interrupt 1 + - description: WAKEUPMIX Audio XCVR Interrupt 2 + minItems: 1 clocks: items: @@ -78,6 +81,23 @@ required: - dma-names - resets +allOf: + - if: + properties: + compatible: + contains: + enum: + - fsl,imx93-xcvr + then: + properties: + interrupts: + minItems: 2 + maxItems: 2 + else: + properties: + interrupts: + maxItems: 1 + additionalProperties: false examples: From c3ab23a10771bbe06300e5374efa809789c65455 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:12 +0530 Subject: [PATCH 174/337] ASoC: amd: Add new dmi entries for acp5x platform Add sys_vendor and product_name dmi entries for acp5x platform. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 20cee7104c2b..19a30f145143 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Advanced Micro Devices, Inc. +// Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. // // Authors: Ajit Kumar Pandey // @@ -47,6 +47,19 @@ static const struct config_entry config_table[] = { {} }, }, + { + .flags = FLAG_AMD_LEGACY, + .device = ACP_PCI_DEV_ID, + .dmi_table = (const struct dmi_system_id []) { + { + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Valve"), + DMI_MATCH(DMI_PRODUCT_NAME, "Jupiter"), + }, + }, + {} + }, + }, { .flags = FLAG_AMD_SOF, .device = ACP_PCI_DEV_ID, From f18818eb0dbe0339c0efd02a34a3f5651749cb84 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:13 +0530 Subject: [PATCH 175/337] ASoC: amd: vangogh: Add condition check for acp config flag Add condition check for acp config flag to load legacy driver only. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-2-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/vangogh/pci-acp5x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index c4634a8a17cd..3826443d77b9 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -2,7 +2,7 @@ // // AMD Vangogh ACP PCI Driver // -// Copyright (C) 2021 Advanced Micro Devices, Inc. All rights reserved. +// Copyright (C) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. #include #include @@ -13,6 +13,7 @@ #include #include "acp5x.h" +#include "../mach-config.h" struct acp5x_dev_data { void __iomem *acp5x_base; @@ -131,7 +132,7 @@ static int snd_acp5x_probe(struct pci_dev *pci, /* Return if acp config flag is defined */ flag = snd_amd_acp_find_config(pci); - if (flag) + if (flag != FLAG_AMD_LEGACY) return -ENODEV; irqflags = IRQF_SHARED; From e12678141835c539fc17a2318ec4017a845935bd Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:14 +0530 Subject: [PATCH 176/337] ASoC: amd: Remove extra dmi parameter Remove extra dmi_product_family entry in amd config entry table. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-3-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 19a30f145143..04b58fb29a35 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -68,7 +68,6 @@ static const struct config_entry config_table[] = { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Valve"), DMI_MATCH(DMI_PRODUCT_NAME, "Galileo"), - DMI_MATCH(DMI_PRODUCT_FAMILY, "Sephiroth"), }, }, {} From 671dd2ffbd8b92e2228fa84ea4274a051b704dec Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:15 +0530 Subject: [PATCH 177/337] ASoC: amd: acp: Add new cpu dai and dailink creation for I2S BT instance Add sof_bt cpu id and create dailink for i2s bt instance in acp common machine driver. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-4-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 25 ++++++++++++++++++++++++- sound/soc/amd/acp/acp-mach.h | 3 +++ 2 files changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 34b14f2611ba..4631af028f15 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Advanced Micro Devices, Inc. +// Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. // // Authors: Ajit Kumar Pandey // Vijendar Mukunda @@ -1290,6 +1290,8 @@ SND_SOC_DAILINK_DEF(sof_hs, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-hs"))); SND_SOC_DAILINK_DEF(sof_hs_virtual, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-hs-virtual"))); +SND_SOC_DAILINK_DEF(sof_bt, + DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-bt"))); SND_SOC_DAILINK_DEF(sof_dmic, DAILINK_COMP_ARRAY(COMP_CPU("acp-sof-dmic"))); SND_SOC_DAILINK_DEF(pdm_dmic, @@ -1348,6 +1350,8 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) if (drv_data->hs_cpu_id) num_links++; + if (drv_data->bt_cpu_id) + num_links++; if (drv_data->amp_cpu_id) num_links++; if (drv_data->dmic_cpu_id) @@ -1497,6 +1501,25 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) i++; } + if (drv_data->bt_cpu_id == I2S_BT) { + links[i].name = "acp-bt-codec"; + links[i].id = BT_BE_ID; + links[i].cpus = sof_bt; + links[i].num_cpus = ARRAY_SIZE(sof_bt); + links[i].platforms = sof_component; + links[i].num_platforms = ARRAY_SIZE(sof_component); + links[i].dpcm_playback = 1; + links[i].dpcm_capture = 1; + links[i].nonatomic = true; + links[i].no_pcm = 1; + if (!drv_data->bt_codec_id) { + /* Use dummy codec if codec id not specified */ + links[i].codecs = &snd_soc_dummy_dlc; + links[i].num_codecs = 1; + } + i++; + } + if (drv_data->dmic_cpu_id == DMIC) { links[i].name = "acp-dmic-codec"; links[i].id = DMIC_BE_ID; diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index cd681101bea7..a48546d8d407 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -28,6 +28,7 @@ enum be_id { HEADSET_BE_ID = 0, AMP_BE_ID, DMIC_BE_ID, + BT_BE_ID, }; enum cpu_endpoints { @@ -68,9 +69,11 @@ struct acp_mach_ops { struct acp_card_drvdata { unsigned int hs_cpu_id; unsigned int amp_cpu_id; + unsigned int bt_cpu_id; unsigned int dmic_cpu_id; unsigned int hs_codec_id; unsigned int amp_codec_id; + unsigned int bt_codec_id; unsigned int dmic_codec_id; unsigned int dai_fmt; unsigned int platform; From e6a382cf7a69cc80e57978bbf0c7a674dfb09621 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:16 +0530 Subject: [PATCH 178/337] ASoC: amd: acp: Add i2s bt support for nau8821-max card Add i2s bt support for sof-nau8821-max sound card. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-5-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sof-mach.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index 5223033a122f..2a9fd3275e42 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Advanced Micro Devices, Inc. +// Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. // // Authors: Ajit Kumar Pandey // @@ -86,9 +86,11 @@ static struct acp_card_drvdata sof_rt5682s_hs_rt1019_data = { static struct acp_card_drvdata sof_nau8821_max98388_data = { .hs_cpu_id = I2S_SP, .amp_cpu_id = I2S_HS, + .bt_cpu_id = I2S_BT, .dmic_cpu_id = NONE, .hs_codec_id = NAU8821, .amp_codec_id = MAX98388, + .bt_codec_id = NONE, .dmic_codec_id = NONE, .soc_mclk = true, .tdm_mode = false, From e249839bf33f3f9727d6220536ed5c7d4f5bc31d Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:17 +0530 Subject: [PATCH 179/337] ASoC: amd: acp: Enable dpcm_capture for MAX98388 codec Enable dpcm_capture for amplifier codec MAX98388 for reference stream capture in smart amplifier case. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-6-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 4631af028f15..b72beb8e9b13 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1483,6 +1483,7 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) links[i].init = acp_card_maxim_init; } if (drv_data->amp_codec_id == MAX98388) { + links[i].dpcm_capture = 1; links[i].codecs = max98388; links[i].num_codecs = ARRAY_SIZE(max98388); links[i].ops = &acp_max98388_ops; From ff5a698c0ffb08eee9c1ce0dfc79c91f273122d5 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:18 +0530 Subject: [PATCH 180/337] ASoC: amd: acp: Set bclk as source to set pll for rt5682s codec Some platforms doesn't have reference mclk pin to codec, so set bclk as a clk source for rt5682s codec pll in tdm mode. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-7-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index b72beb8e9b13..74e83c2dae53 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -464,6 +464,22 @@ static int acp_card_rt5682s_hw_params(struct snd_pcm_substream *substream, return ret; } + if (drvdata->tdm_mode) { + ret = snd_soc_dai_set_pll(codec_dai, RT5682S_PLL1, RT5682S_PLL_S_BCLK1, + 6144000, 49152000); + if (ret < 0) { + dev_err(rtd->dev, "Failed to set codec PLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682S_SCLK_S_PLL1, + 49152000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "Failed to set codec SYSCLK: %d\n", ret); + return ret; + } + } + /* Set tdm/i2s1 master bclk ratio */ ret = snd_soc_dai_set_bclk_ratio(codec_dai, ch * format); if (ret < 0) { From 14b4b5fd3d7aac2f17800b821a0fac30d705a25d Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 6 Dec 2023 16:36:18 +0530 Subject: [PATCH 181/337] ASoC: amd: acp: Set bclk as source to set pll for rt5682s codec Some platforms doesn't have reference mclk pin to codec, so set bclk as a clk source for rt5682s codec pll in tdm mode. Signed-off-by: Venkata Prasad Potturu Link: https://lore.kernel.org/r/20231206110620.1695591-7-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 74e83c2dae53..f7bcf210f0fd 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -282,6 +282,22 @@ static int acp_card_rt5682_hw_params(struct snd_pcm_substream *substream, return ret; } + if (drvdata->tdm_mode) { + ret = snd_soc_dai_set_pll(codec_dai, RT5682S_PLL1, RT5682S_PLL_S_BCLK1, + 6144000, 49152000); + if (ret < 0) { + dev_err(rtd->dev, "Failed to set codec PLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682S_SCLK_S_PLL1, + 49152000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "Failed to set codec SYSCLK: %d\n", ret); + return ret; + } + } + /* Set tdm/i2s1 master bclk ratio */ ret = snd_soc_dai_set_bclk_ratio(codec_dai, ch * format); if (ret < 0) { From 5ec42bf04d72fd6d0a6855810cc779e0ee31dfd7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 4 Dec 2023 15:27:06 -0600 Subject: [PATCH 182/337] PCI: add INTEL_HDA_ARL to pci_ids.h MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The PCI ID insertion follows the increasing order in the table, but this hardware follows MTL (MeteorLake). Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Acked-by: Mark Brown Link: https://lore.kernel.org/r/20231204212710.185976-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- include/linux/pci_ids.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 275799b5f535..97cc0baad0f4 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -3065,6 +3065,7 @@ #define PCI_DEVICE_ID_INTEL_82443GX_0 0x71a0 #define PCI_DEVICE_ID_INTEL_82443GX_2 0x71a2 #define PCI_DEVICE_ID_INTEL_82372FB_1 0x7601 +#define PCI_DEVICE_ID_INTEL_HDA_ARL 0x7728 #define PCI_DEVICE_ID_INTEL_HDA_RPL_S 0x7a50 #define PCI_DEVICE_ID_INTEL_HDA_ADL_S 0x7ad0 #define PCI_DEVICE_ID_INTEL_HDA_MTL 0x7e28 From a31014ebad617868c246d3985ff80d891f03711e Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 4 Dec 2023 15:27:07 -0600 Subject: [PATCH 183/337] ALSA: hda: Intel: add HDA_ARL PCI ID support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Yet another PCI ID. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Acked-by: Mark Brown Link: https://lore.kernel.org/r/20231204212710.185976-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index db90feb49c16..d0ade91ab5ac 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2502,6 +2502,8 @@ static const struct pci_device_id azx_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_LNL_P, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Arrow Lake-S */ { PCI_DEVICE_DATA(INTEL, HDA_ARL_S, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, + /* Arrow Lake */ + { PCI_DEVICE_DATA(INTEL, HDA_ARL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE) }, /* Apollolake (Broxton-P) */ { PCI_DEVICE_DATA(INTEL, HDA_APL, AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON) }, /* Gemini-Lake */ From 7a9d6bbe8a663c817080be55d9fecf19a4a8fd8f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 4 Dec 2023 15:27:08 -0600 Subject: [PATCH 184/337] ALSA: hda: intel-dspcfg: add filters for ARL-S and ARL MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Same usual filters, SOF is required for DMIC and/or SoundWire support. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Acked-by: Mark Brown Link: https://lore.kernel.org/r/20231204212710.185976-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 756fa0aa69bb..6a384b922e4f 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -521,6 +521,16 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = PCI_DEVICE_ID_INTEL_HDA_MTL, }, + /* ArrowLake-S */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = PCI_DEVICE_ID_INTEL_HDA_ARL_S, + }, + /* ArrowLake */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = PCI_DEVICE_ID_INTEL_HDA_ARL, + }, #endif /* Lunar Lake */ From a00be6dc9bb80796244196033aa5eb258b6af47a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 4 Dec 2023 15:27:09 -0600 Subject: [PATCH 185/337] ASoC: SOF: Intel: pci-mtl: fix ARL-S definitions MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The initial copy/paste from MTL was incorrect, the hardware is different and requires different descriptors along with a dedicated firmware binary. Fixes: 3851831f529e ("ASoC: SOF: Intel: pci-mtl: use ARL specific firmware definitions") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Acked-by: Mark Brown Link: https://lore.kernel.org/r/20231204212710.185976-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/mtl.c | 28 ++++++++++++++++++++++++++++ sound/soc/sof/intel/pci-mtl.c | 12 ++++++------ 3 files changed, 35 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index d628d6a3a7e5..1592e27bc14d 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -882,6 +882,7 @@ extern const struct sof_intel_dsp_desc ehl_chip_info; extern const struct sof_intel_dsp_desc jsl_chip_info; extern const struct sof_intel_dsp_desc adls_chip_info; extern const struct sof_intel_dsp_desc mtl_chip_info; +extern const struct sof_intel_dsp_desc arl_s_chip_info; extern const struct sof_intel_dsp_desc lnl_chip_info; /* Probes support */ diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 254dbbeac1d0..7946110e7adf 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -746,3 +746,31 @@ const struct sof_intel_dsp_desc mtl_chip_info = { .hw_ip_version = SOF_INTEL_ACE_1_0, }; EXPORT_SYMBOL_NS(mtl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); + +const struct sof_intel_dsp_desc arl_s_chip_info = { + .cores_num = 2, + .init_core_mask = BIT(0), + .host_managed_cores_mask = BIT(0), + .ipc_req = MTL_DSP_REG_HFIPCXIDR, + .ipc_req_mask = MTL_DSP_REG_HFIPCXIDR_BUSY, + .ipc_ack = MTL_DSP_REG_HFIPCXIDA, + .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, + .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, + .rom_status_reg = MTL_DSP_ROM_STS, + .rom_init_timeout = 300, + .ssp_count = MTL_SSP_COUNT, + .ssp_base_offset = CNL_SSP_BASE_OFFSET, + .sdw_shim_base = SDW_SHIM_BASE_ACE, + .sdw_alh_base = SDW_ALH_BASE_ACE, + .d0i3_offset = MTL_HDA_VS_D0I3C, + .read_sdw_lcount = hda_sdw_check_lcount_common, + .enable_sdw_irq = mtl_enable_sdw_irq, + .check_sdw_irq = mtl_dsp_check_sdw_irq, + .check_sdw_wakeen_irq = hda_sdw_check_wakeen_irq_common, + .check_ipc_irq = mtl_dsp_check_ipc_irq, + .cl_init = mtl_dsp_cl_init, + .power_down_dsp = mtl_power_down_dsp, + .disable_interrupts = mtl_dsp_disable_interrupts, + .hw_ip_version = SOF_INTEL_ACE_1_0, +}; +EXPORT_SYMBOL_NS(arl_s_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index 0f378f45486d..60d5e73cdad2 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -50,7 +50,7 @@ static const struct sof_dev_desc mtl_desc = { .ops_free = hda_ops_free, }; -static const struct sof_dev_desc arl_desc = { +static const struct sof_dev_desc arl_s_desc = { .use_acpi_target_states = true, .machines = snd_soc_acpi_intel_arl_machines, .alt_machines = snd_soc_acpi_intel_arl_sdw_machines, @@ -58,21 +58,21 @@ static const struct sof_dev_desc arl_desc = { .resindex_pcicfg_base = -1, .resindex_imr_base = -1, .irqindex_host_ipc = -1, - .chip_info = &mtl_chip_info, + .chip_info = &arl_s_chip_info, .ipc_supported_mask = BIT(SOF_IPC_TYPE_4), .ipc_default = SOF_IPC_TYPE_4, .dspless_mode_supported = true, /* Only supported for HDaudio */ .default_fw_path = { - [SOF_IPC_TYPE_4] = "intel/sof-ipc4/arl", + [SOF_IPC_TYPE_4] = "intel/sof-ipc4/arl-s", }, .default_lib_path = { - [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/arl", + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/arl-s", }, .default_tplg_path = { [SOF_IPC_TYPE_4] = "intel/sof-ace-tplg", }, .default_fw_filename = { - [SOF_IPC_TYPE_4] = "sof-arl.ri", + [SOF_IPC_TYPE_4] = "sof-arl-s.ri", }, .nocodec_tplg_filename = "sof-arl-nocodec.tplg", .ops = &sof_mtl_ops, @@ -83,7 +83,7 @@ static const struct sof_dev_desc arl_desc = { /* PCI IDs */ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_MTL, &mtl_desc) }, - { PCI_DEVICE_DATA(INTEL, HDA_ARL_S, &arl_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ARL_S, &arl_s_desc) }, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); From 1ccffc2f760a960372093106558411e644b89c57 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 4 Dec 2023 15:27:10 -0600 Subject: [PATCH 186/337] ASoC: SOF: Intel: pci-mtl: add HDA_ARL PCI support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add yet another PCI ID - the hardware shares the same descriptors as MTL but we use a dedicated firmware binary file to allow for different signature keys. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Reviewed-by: Kai Vehmanen Acked-by: Mark Brown Link: https://lore.kernel.org/r/20231204212710.185976-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/soc/sof/intel/pci-mtl.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index 60d5e73cdad2..cacc985d80f4 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -50,6 +50,36 @@ static const struct sof_dev_desc mtl_desc = { .ops_free = hda_ops_free, }; +static const struct sof_dev_desc arl_desc = { + .use_acpi_target_states = true, + .machines = snd_soc_acpi_intel_arl_machines, + .alt_machines = snd_soc_acpi_intel_arl_sdw_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &mtl_chip_info, + .ipc_supported_mask = BIT(SOF_IPC_TYPE_4), + .ipc_default = SOF_IPC_TYPE_4, + .dspless_mode_supported = true, /* Only supported for HDaudio */ + .default_fw_path = { + [SOF_IPC_TYPE_4] = "intel/sof-ipc4/arl", + }, + .default_lib_path = { + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/arl", + }, + .default_tplg_path = { + [SOF_IPC_TYPE_4] = "intel/sof-ace-tplg", + }, + .default_fw_filename = { + [SOF_IPC_TYPE_4] = "sof-arl.ri", + }, + .nocodec_tplg_filename = "sof-arl-nocodec.tplg", + .ops = &sof_mtl_ops, + .ops_init = sof_mtl_ops_init, + .ops_free = hda_ops_free, +}; + static const struct sof_dev_desc arl_s_desc = { .use_acpi_target_states = true, .machines = snd_soc_acpi_intel_arl_machines, @@ -84,6 +114,7 @@ static const struct sof_dev_desc arl_s_desc = { static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE_DATA(INTEL, HDA_MTL, &mtl_desc) }, { PCI_DEVICE_DATA(INTEL, HDA_ARL_S, &arl_s_desc) }, + { PCI_DEVICE_DATA(INTEL, HDA_ARL, &arl_desc) }, { 0, } }; MODULE_DEVICE_TABLE(pci, sof_pci_ids); From ce17aa4cf2db7d6b95a3f854ac52d0d49881dd49 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 7 Dec 2023 11:54:25 +0200 Subject: [PATCH 187/337] ASoC: SOF: Intel: hda-codec: Delay the codec device registration The current code flow is: 1. snd_hdac_device_register() 2. set parameters needed by the hdac driver 3. request_codec_module() the hdac driver is probed at this point During boot the codec drivers are not loaded when the hdac device is registered, it is going to be probed later when loading the codec module, which point the parameters are set. On module remove/insert rmmod snd_sof_pci_intel_tgl modprobe snd_sof_pci_intel_tgl The codec module remains loaded and the driver will be probed when the hdac device is created right away, before the parameters for the driver has been configured: 1. snd_hdac_device_register() the hdac driver is probed at this point 2. set parameters needed by the hdac driver 3. request_codec_module() will be a NOP as the module is already loaded Move the snd_hdac_device_register() later, to be done right before requesting the codec module to make sure that the parameters are all set before the device is created: 1. set parameters needed by the hdac driver 2. snd_hdac_device_register() 3. request_codec_module() This way at the hdac driver probe all parameters will be set in all cases. Link: https://github.com/thesofproject/linux/issues/4731 Fixes: a0575b4add21 ("ASoC: hdac_hda: Conditionally register dais for HDMI and Analog") Signed-off-by: Peter Ujfalusi Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20231207095425.19597-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-codec.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 28ecbebb4b84..9f84b0d287a5 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -54,8 +54,16 @@ static int request_codec_module(struct hda_codec *codec) static int hda_codec_load_module(struct hda_codec *codec) { - int ret = request_codec_module(codec); + int ret; + ret = snd_hdac_device_register(&codec->core); + if (ret) { + dev_err(&codec->core.dev, "failed to register hdac device\n"); + put_device(&codec->core.dev); + return ret; + } + + ret = request_codec_module(codec); if (ret <= 0) { codec->probe_id = HDA_CODEC_ID_GENERIC; ret = request_codec_module(codec); @@ -116,7 +124,6 @@ EXPORT_SYMBOL_NS_GPL(hda_codec_jack_check, SND_SOC_SOF_HDA_AUDIO_CODEC); static struct hda_codec *hda_codec_device_init(struct hdac_bus *bus, int addr, int type) { struct hda_codec *codec; - int ret; codec = snd_hda_codec_device_init(to_hda_bus(bus), addr, "ehdaudio%dD%d", bus->idx, addr); if (IS_ERR(codec)) { @@ -126,13 +133,6 @@ static struct hda_codec *hda_codec_device_init(struct hdac_bus *bus, int addr, i codec->core.type = type; - ret = snd_hdac_device_register(&codec->core); - if (ret) { - dev_err(bus->dev, "failed to register hdac device\n"); - put_device(&codec->core.dev); - return ERR_PTR(ret); - } - return codec; } From 8527ecc6cf25417a1940f91eb91fca0c69a5c553 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 7 Dec 2023 10:25:01 +0530 Subject: [PATCH 188/337] ASoC: amd: acp: modify config flag read logic Modify acp config flag read logic from ACP v7.0 onwards. Instead of reading from DMI table match entry, read the config flag value from BIOS ACPI table. This will remove updating DMI table when new platform support is added. Use FLAG_AMD_LEGACY_ONLY_DMIC flag as default one. Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20231207045505.1519151-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 04b58fb29a35..067b1fdfbc9d 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -19,6 +19,8 @@ #include "../sof/amd/acp.h" #include "mach-config.h" +#define ACP_7_0_REV 0x70 + static int acp_quirk_data; static const struct config_entry config_table[] = { @@ -145,15 +147,33 @@ static const struct config_entry config_table[] = { }, }; +static int snd_amd_acp_acpi_find_config(struct pci_dev *pci) +{ + const union acpi_object *obj; + int acp_flag = FLAG_AMD_LEGACY_ONLY_DMIC; + + if (!acpi_dev_get_property(ACPI_COMPANION(&pci->dev), "acp-audio-config-flag", + ACPI_TYPE_INTEGER, &obj)) + acp_flag = obj->integer.value; + + return acp_flag; +} + int snd_amd_acp_find_config(struct pci_dev *pci) { const struct config_entry *table = config_table; u16 device = pci->device; int i; - /* Do not enable FLAGS on older platforms with Rev id zero */ + /* Do not enable FLAGS on older platforms with Rev Id zero + * For platforms which has ACP 7.0 or higher, read the acp + * config flag from BIOS ACPI table and for older platforms + * read it from DMI tables. + */ if (!pci->revision) return 0; + else if (pci->revision >= ACP_7_0_REV) + return snd_amd_acp_acpi_find_config(pci); for (i = 0; i < ARRAY_SIZE(config_table); i++, table++) { if (table->device != device) From d685aea5e0a89b66679e5266320ab2ba4378c754 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 4 Dec 2023 15:42:07 +0300 Subject: [PATCH 189/337] ASoC: audio-graph-card2: fix off by one in graph_parse_node_multi_nm() The > comparison should be >= to avoid writing one element beyond the end of the dai_link->ch_maps[] array. The dai_link->ch_maps[] array is allocated in graph_parse_node_multi() and it has "nm_max" elements. Fixes: e2de6808df4a ("ASoC: audio-graph-card2: add CPU:Codec = N:M support") Signed-off-by: Dan Carpenter Acked-by: Kuninori Morimoto Link: https://lore.kernel.org/r/1032216f-902f-48f9-aa49-9d5ece8e87f2@moroto.mountain Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index d9e10308a508..78d9679decda 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -557,7 +557,7 @@ static int graph_parse_node_multi_nm(struct snd_soc_dai_link *dai_link, struct device_node *mcodec_port; int codec_idx; - if (*nm_idx > nm_max) + if (*nm_idx >= nm_max) break; mcpu_ep_n = of_get_next_child(mcpu_port, mcpu_ep_n); From b53d47775651aa51bb98cdeb968dedb45699d9a1 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 8 Dec 2023 11:09:25 +0100 Subject: [PATCH 190/337] ASoC: wm0010: Convert to GPIO descriptors This converts the WM0010 codec to use GPIO descriptors. It's a pretty straight-forward conversion also switching over the single in-tree user in the S3C Cragganmore module for S3C 6410. Signed-off-by: Linus Walleij Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20231208-descriptors-sound-wlf-v1-1-c4dab6f521ec@linaro.org Signed-off-by: Mark Brown --- arch/arm/mach-s3c/mach-crag6410-module.c | 16 +++++++-- include/sound/wm0010.h | 6 ---- sound/soc/codecs/wm0010.c | 44 ++++++------------------ 3 files changed, 23 insertions(+), 43 deletions(-) diff --git a/arch/arm/mach-s3c/mach-crag6410-module.c b/arch/arm/mach-s3c/mach-crag6410-module.c index 8fce1e815ee8..a9a713641047 100644 --- a/arch/arm/mach-s3c/mach-crag6410-module.c +++ b/arch/arm/mach-s3c/mach-crag6410-module.c @@ -32,9 +32,18 @@ #include "crag6410.h" +static struct gpiod_lookup_table wm0010_gpiod_table = { + .dev_id = "spi0.0", /* SPI device name */ + .table = { + /* Active high for Glenfarclas Rev 2 */ + GPIO_LOOKUP("GPION", 6, + "reset", GPIO_ACTIVE_HIGH), + { }, + }, +}; + static struct wm0010_pdata wm0010_pdata = { - .gpio_reset = S3C64XX_GPN(6), - .reset_active_high = 1, /* Active high for Glenfarclas Rev 2 */ + /* Intentionally left blank */ }; static struct spi_board_info wm1253_devs[] = { @@ -337,7 +346,8 @@ static const struct { { .id = 0x21, .rev = 0xff, .name = "1275-EV1 Mortlach" }, { .id = 0x25, .rev = 0xff, .name = "1274-EV1 Glencadam" }, { .id = 0x31, .rev = 0xff, .name = "1253-EV1 Tomatin", - .spi_devs = wm1253_devs, .num_spi_devs = ARRAY_SIZE(wm1253_devs) }, + .spi_devs = wm1253_devs, .num_spi_devs = ARRAY_SIZE(wm1253_devs), + .gpiod_table = &wm0010_gpiod_table }, { .id = 0x32, .rev = 0xff, .name = "XXXX-EV1 Caol Illa" }, { .id = 0x33, .rev = 0xff, .name = "XXXX-EV1 Oban" }, { .id = 0x34, .rev = 0xff, .name = "WM0010-6320-CS42 Balblair", diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h index 13b473935ca1..14ff9056c5d0 100644 --- a/include/sound/wm0010.h +++ b/include/sound/wm0010.h @@ -11,12 +11,6 @@ #define WM0010_PDATA_H struct wm0010_pdata { - int gpio_reset; - - /* Set if there is an inverter between the GPIO controlling - * the reset signal and the device. - */ - int reset_active_high; int irq_flags; }; diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 1d4259433f47..8f862729a2ca 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -18,7 +18,7 @@ #include #include #include -#include +#include #include #include #include @@ -94,8 +94,7 @@ struct wm0010_priv { struct wm0010_pdata pdata; - int gpio_reset; - int gpio_reset_value; + struct gpio_desc *reset; struct regulator_bulk_data core_supplies[2]; struct regulator *dbvdd; @@ -174,8 +173,7 @@ static void wm0010_halt(struct snd_soc_component *component) case WM0010_STAGE2: case WM0010_FIRMWARE: /* Remember to put chip back into reset */ - gpio_set_value_cansleep(wm0010->gpio_reset, - wm0010->gpio_reset_value); + gpiod_set_value_cansleep(wm0010->reset, 1); /* Disable the regulators */ regulator_disable(wm0010->dbvdd); regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), @@ -610,7 +608,7 @@ static int wm0010_boot(struct snd_soc_component *component) } /* Release reset */ - gpio_set_value_cansleep(wm0010->gpio_reset, !wm0010->gpio_reset_value); + gpiod_set_value_cansleep(wm0010->reset, 0); spin_lock_irqsave(&wm0010->irq_lock, flags); wm0010->state = WM0010_OUT_OF_RESET; spin_unlock_irqrestore(&wm0010->irq_lock, flags); @@ -863,7 +861,6 @@ static int wm0010_probe(struct snd_soc_component *component) static int wm0010_spi_probe(struct spi_device *spi) { - unsigned long gpio_flags; int ret; int trigger; int irq; @@ -903,31 +900,11 @@ static int wm0010_spi_probe(struct spi_device *spi) return ret; } - if (wm0010->pdata.gpio_reset) { - wm0010->gpio_reset = wm0010->pdata.gpio_reset; - - if (wm0010->pdata.reset_active_high) - wm0010->gpio_reset_value = 1; - else - wm0010->gpio_reset_value = 0; - - if (wm0010->gpio_reset_value) - gpio_flags = GPIOF_OUT_INIT_HIGH; - else - gpio_flags = GPIOF_OUT_INIT_LOW; - - ret = devm_gpio_request_one(wm0010->dev, wm0010->gpio_reset, - gpio_flags, "wm0010 reset"); - if (ret < 0) { - dev_err(wm0010->dev, - "Failed to request GPIO for DSP reset: %d\n", - ret); - return ret; - } - } else { - dev_err(wm0010->dev, "No reset GPIO configured\n"); - return -EINVAL; - } + wm0010->reset = devm_gpiod_get(wm0010->dev, "reset", GPIOD_OUT_HIGH); + if (IS_ERR(wm0010->reset)) + return dev_err_probe(wm0010->dev, PTR_ERR(wm0010->reset), + "could not get RESET GPIO\n"); + gpiod_set_consumer_name(wm0010->reset, "wm0010 reset"); wm0010->state = WM0010_POWER_OFF; @@ -972,8 +949,7 @@ static void wm0010_spi_remove(struct spi_device *spi) { struct wm0010_priv *wm0010 = spi_get_drvdata(spi); - gpio_set_value_cansleep(wm0010->gpio_reset, - wm0010->gpio_reset_value); + gpiod_set_value_cansleep(wm0010->reset, 1); irq_set_irq_wake(wm0010->irq, 0); From 10a366f36e2a2e240d9db6da90e501fa16655282 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 8 Dec 2023 11:09:26 +0100 Subject: [PATCH 191/337] ASoC: wm1250-ev1: Convert to GPIO descriptors This converts the WM1250-EV1 codec to use GPIO descriptors. It turns out that the platform data was only used to pass some global GPIO numbers from a board file, so we get rid of this and also switch over the single in-tree user in the S3C Cragganmore module for S3C 6410. The driver obtains two GPIO lines named OSR and master and just pull them low, we leave this behaviour as it was. Signed-off-by: Linus Walleij Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20231208-descriptors-sound-wlf-v1-2-c4dab6f521ec@linaro.org Signed-off-by: Mark Brown --- arch/arm/mach-s3c/mach-crag6410.c | 24 +++---- include/sound/wm1250-ev1.h | 24 ------- sound/soc/codecs/wm1250-ev1.c | 114 +++++++++++------------------- 3 files changed, 54 insertions(+), 108 deletions(-) delete mode 100644 include/sound/wm1250-ev1.h diff --git a/arch/arm/mach-s3c/mach-crag6410.c b/arch/arm/mach-s3c/mach-crag6410.c index 7c4bed4370a1..e5df2cb51ab2 100644 --- a/arch/arm/mach-s3c/mach-crag6410.c +++ b/arch/arm/mach-s3c/mach-crag6410.c @@ -39,8 +39,6 @@ #include #include -#include - #include #include @@ -713,13 +711,16 @@ static struct wm831x_pdata glenfarclas_pmic_pdata = { .disable_touch = true, }; -static struct wm1250_ev1_pdata wm1250_ev1_pdata = { - .gpios = { - [WM1250_EV1_GPIO_CLK_ENA] = S3C64XX_GPN(12), - [WM1250_EV1_GPIO_CLK_SEL0] = S3C64XX_GPL(12), - [WM1250_EV1_GPIO_CLK_SEL1] = S3C64XX_GPL(13), - [WM1250_EV1_GPIO_OSR] = S3C64XX_GPL(14), - [WM1250_EV1_GPIO_MASTER] = S3C64XX_GPL(8), +static struct gpiod_lookup_table crag_wm1250_ev1_gpiod_table = { + /* The WM1250-EV1 is device 0027 on I2C bus 1 */ + .dev_id = "1-0027", + .table = { + GPIO_LOOKUP("GPION", 12, "clk-ena", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOL", 12, "clk-sel0", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOL", 13, "clk-sel1", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOL", 14, "osr", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOL", 8, "master", GPIO_ACTIVE_HIGH), + { }, }, }; @@ -733,9 +734,7 @@ static struct i2c_board_info i2c_devs1[] = { { I2C_BOARD_INFO("wlf-gf-module", 0x24) }, { I2C_BOARD_INFO("wlf-gf-module", 0x25) }, { I2C_BOARD_INFO("wlf-gf-module", 0x26) }, - - { I2C_BOARD_INFO("wm1250-ev1", 0x27), - .platform_data = &wm1250_ev1_pdata }, + { I2C_BOARD_INFO("wm1250-ev1", 0x27), }, }; static struct s3c2410_platform_i2c i2c1_pdata = { @@ -862,6 +861,7 @@ static void __init crag6410_machine_init(void) gpiod_add_lookup_table(&crag_pmic_gpiod_table); i2c_register_board_info(0, i2c_devs0, ARRAY_SIZE(i2c_devs0)); + gpiod_add_lookup_table(&crag_wm1250_ev1_gpiod_table); i2c_register_board_info(1, i2c_devs1, ARRAY_SIZE(i2c_devs1)); samsung_keypad_set_platdata(&crag6410_keypad_data); diff --git a/include/sound/wm1250-ev1.h b/include/sound/wm1250-ev1.h deleted file mode 100644 index d16614ebecb4..000000000000 --- a/include/sound/wm1250-ev1.h +++ /dev/null @@ -1,24 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * linux/sound/wm1250-ev1.h - Platform data for WM1250-EV1 - * - * Copyright 2011 Wolfson Microelectronics. PLC. - */ - -#ifndef __LINUX_SND_WM1250_EV1_H -#define __LINUX_SND_WM1250_EV1_H - -#define WM1250_EV1_NUM_GPIOS 5 - -#define WM1250_EV1_GPIO_CLK_ENA 0 -#define WM1250_EV1_GPIO_CLK_SEL0 1 -#define WM1250_EV1_GPIO_CLK_SEL1 2 -#define WM1250_EV1_GPIO_OSR 3 -#define WM1250_EV1_GPIO_MASTER 4 - - -struct wm1250_ev1_pdata { - int gpios[WM1250_EV1_NUM_GPIOS]; -}; - -#endif diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index d7eeb41ba60f..a2a8b2a4b19b 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -9,34 +9,23 @@ #include #include #include -#include +#include #include #include -#include - -static const char *wm1250_gpio_names[WM1250_EV1_NUM_GPIOS] = { - "WM1250 CLK_ENA", - "WM1250 CLK_SEL0", - "WM1250 CLK_SEL1", - "WM1250 OSR", - "WM1250 MASTER", -}; struct wm1250_priv { - struct gpio gpios[WM1250_EV1_NUM_GPIOS]; + struct gpio_desc *clk_ena; + struct gpio_desc *clk_sel0; + struct gpio_desc *clk_sel1; + struct gpio_desc *osr; + struct gpio_desc *master; }; static int wm1250_ev1_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { struct wm1250_priv *wm1250 = dev_get_drvdata(component->dev); - int ena; - - if (wm1250) - ena = wm1250->gpios[WM1250_EV1_GPIO_CLK_ENA].gpio; - else - ena = -1; switch (level) { case SND_SOC_BIAS_ON: @@ -46,13 +35,11 @@ static int wm1250_ev1_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_STANDBY: - if (ena >= 0) - gpio_set_value_cansleep(ena, 1); + gpiod_set_value_cansleep(wm1250->clk_ena, 1); break; case SND_SOC_BIAS_OFF: - if (ena >= 0) - gpio_set_value_cansleep(ena, 0); + gpiod_set_value_cansleep(wm1250->clk_ena, 0); break; } @@ -80,28 +67,20 @@ static int wm1250_ev1_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, - 1); - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, - 1); + gpiod_set_value(wm1250->clk_sel0, 1); + gpiod_set_value(wm1250->clk_sel1, 1); break; case 16000: - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, - 0); - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, - 1); + gpiod_set_value(wm1250->clk_sel0, 0); + gpiod_set_value(wm1250->clk_sel1, 1); break; case 32000: - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, - 1); - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, - 0); + gpiod_set_value(wm1250->clk_sel0, 1); + gpiod_set_value(wm1250->clk_sel1, 0); break; case 64000: - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].gpio, - 0); - gpio_set_value(wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].gpio, - 0); + gpiod_set_value(wm1250->clk_sel0, 0); + gpiod_set_value(wm1250->clk_sel1, 0); break; default: return -EINVAL; @@ -150,45 +129,43 @@ static int wm1250_ev1_pdata(struct i2c_client *i2c) { struct wm1250_ev1_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm1250_priv *wm1250; - int i, ret; + int ret; if (!pdata) return 0; wm1250 = devm_kzalloc(&i2c->dev, sizeof(*wm1250), GFP_KERNEL); - if (!wm1250) { - ret = -ENOMEM; - goto err; - } + if (!wm1250) + return -ENOMEM; - for (i = 0; i < ARRAY_SIZE(wm1250->gpios); i++) { - wm1250->gpios[i].gpio = pdata->gpios[i]; - wm1250->gpios[i].label = wm1250_gpio_names[i]; - wm1250->gpios[i].flags = GPIOF_OUT_INIT_LOW; - } - wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].flags = GPIOF_OUT_INIT_HIGH; - wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].flags = GPIOF_OUT_INIT_HIGH; + wm1250->clk_ena = devm_gpiod_get(&i2c->dev, "clk-ena", GPIOD_OUT_LOW); + if (IS_ERR(wm1250->clk_ena)) + return dev_err_probe(&i2c->dev, PTR_ERR(wm1250->clk_ena), + "failed to get clock enable GPIO\n"); - ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); - if (ret != 0) { - dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); - goto err; - } + wm1250->clk_sel0 = devm_gpiod_get(&i2c->dev, "clk-sel0", GPIOD_OUT_HIGH); + if (IS_ERR(wm1250->clk_sel0)) + return dev_err_probe(&i2c->dev, PTR_ERR(wm1250->clk_sel0), + "failed to get clock sel0 GPIO\n"); + + wm1250->clk_sel1 = devm_gpiod_get(&i2c->dev, "clk-sel1", GPIOD_OUT_HIGH); + if (IS_ERR(wm1250->clk_sel1)) + return dev_err_probe(&i2c->dev, PTR_ERR(wm1250->clk_sel1), + "failed to get clock sel1 GPIO\n"); + + wm1250->osr = devm_gpiod_get(&i2c->dev, "osr", GPIOD_OUT_LOW); + if (IS_ERR(wm1250->osr)) + return dev_err_probe(&i2c->dev, PTR_ERR(wm1250->osr), + "failed to get OSR GPIO\n"); + + wm1250->master = devm_gpiod_get(&i2c->dev, "master", GPIOD_OUT_LOW); + if (IS_ERR(wm1250->master)) + return dev_err_probe(&i2c->dev, PTR_ERR(wm1250->master), + "failed to get MASTER GPIO\n"); dev_set_drvdata(&i2c->dev, wm1250); return ret; - -err: - return ret; -} - -static void wm1250_ev1_free(struct i2c_client *i2c) -{ - struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); - - if (wm1250) - gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); } static int wm1250_ev1_probe(struct i2c_client *i2c) @@ -221,18 +198,12 @@ static int wm1250_ev1_probe(struct i2c_client *i2c) &wm1250_ev1_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - wm1250_ev1_free(i2c); return ret; } return 0; } -static void wm1250_ev1_remove(struct i2c_client *i2c) -{ - wm1250_ev1_free(i2c); -} - static const struct i2c_device_id wm1250_ev1_i2c_id[] = { { "wm1250-ev1", 0 }, { } @@ -244,7 +215,6 @@ static struct i2c_driver wm1250_ev1_i2c_driver = { .name = "wm1250-ev1", }, .probe = wm1250_ev1_probe, - .remove = wm1250_ev1_remove, .id_table = wm1250_ev1_i2c_id, }; From 0119b2a24eb592f967a2849b772887c96617ad80 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 8 Dec 2023 11:09:27 +0100 Subject: [PATCH 192/337] ASoC: wm2200: Convert to GPIO descriptors This converts the WM2200 codec to use GPIO descriptors. This is a pretty straight-forward conversion, and it also switches over the single in-tree user in the S3C Cragganmore module for S3C 6410. This coded does not seem to get selected or be selectable through Kconfig, I had to hack another soundcard Kconfig entry to select it for compile tests. Signed-off-by: Linus Walleij Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20231208-descriptors-sound-wlf-v1-3-c4dab6f521ec@linaro.org Signed-off-by: Mark Brown --- arch/arm/mach-s3c/mach-crag6410-module.c | 13 ++++- include/sound/wm2200.h | 2 - sound/soc/codecs/wm2200.c | 67 ++++++++++++------------ 3 files changed, 44 insertions(+), 38 deletions(-) diff --git a/arch/arm/mach-s3c/mach-crag6410-module.c b/arch/arm/mach-s3c/mach-crag6410-module.c index a9a713641047..1df6b7e52c9d 100644 --- a/arch/arm/mach-s3c/mach-crag6410-module.c +++ b/arch/arm/mach-s3c/mach-crag6410-module.c @@ -305,12 +305,20 @@ static const struct i2c_board_info wm6230_i2c_devs[] = { }; static struct wm2200_pdata wm2200_pdata = { - .ldo_ena = S3C64XX_GPN(7), .gpio_defaults = { [2] = 0x0005, /* GPIO3 24.576MHz output clock */ }, }; +static struct gpiod_lookup_table wm2200_gpiod_table = { + .dev_id = "1-003a", /* Device 003a on I2C bus 1 */ + .table = { + GPIO_LOOKUP("GPION", 7, + "wlf,ldo1ena", GPIO_ACTIVE_HIGH), + { }, + }, +}; + static const struct i2c_board_info wm2200_i2c[] = { { I2C_BOARD_INFO("wm2200", 0x3a), .platform_data = &wm2200_pdata, }, @@ -372,7 +380,8 @@ static const struct { .num_spi_devs = ARRAY_SIZE(wm5102_spi_devs), .gpiod_table = &wm5102_gpiod_table }, { .id = 0x3f, .rev = -1, .name = "WM2200-6271-CS90-M-REV1", - .i2c_devs = wm2200_i2c, .num_i2c_devs = ARRAY_SIZE(wm2200_i2c) }, + .i2c_devs = wm2200_i2c, .num_i2c_devs = ARRAY_SIZE(wm2200_i2c), + .gpiod_table = &wm2200_gpiod_table }, }; static int wlf_gf_module_probe(struct i2c_client *i2c) diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h index 9987e6c09bdc..2e4913ee2505 100644 --- a/include/sound/wm2200.h +++ b/include/sound/wm2200.h @@ -42,8 +42,6 @@ struct wm2200_micbias { }; struct wm2200_pdata { - int reset; /** GPIO controlling /RESET, if any */ - int ldo_ena; /** GPIO controlling LODENA, if any */ int irq_flags; int gpio_defaults[4]; diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 9679906c6bd5..69c9c2bd7e7b 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -14,7 +14,7 @@ #include #include #include -#include +#include #include #include #include @@ -79,6 +79,8 @@ struct wm2200_priv { struct snd_soc_component *component; struct wm2200_pdata pdata; struct regulator_bulk_data core_supplies[WM2200_NUM_CORE_SUPPLIES]; + struct gpio_desc *ldo_ena; + struct gpio_desc *reset; struct completion fll_lock; int fll_fout; @@ -975,9 +977,10 @@ static const struct reg_sequence wm2200_reva_patch[] = { static int wm2200_reset(struct wm2200_priv *wm2200) { - if (wm2200->pdata.reset) { - gpio_set_value_cansleep(wm2200->pdata.reset, 0); - gpio_set_value_cansleep(wm2200->pdata.reset, 1); + if (wm2200->reset) { + /* Descriptor flagged active low, so this will be inverted */ + gpiod_set_value_cansleep(wm2200->reset, 1); + gpiod_set_value_cansleep(wm2200->reset, 0); return 0; } else { @@ -2246,28 +2249,28 @@ static int wm2200_i2c_probe(struct i2c_client *i2c) return ret; } - if (wm2200->pdata.ldo_ena) { - ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, - "WM2200 LDOENA"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", - wm2200->pdata.ldo_ena, ret); - goto err_enable; - } + wm2200->ldo_ena = devm_gpiod_get_optional(&i2c->dev, "wlf,ldo1ena", + GPIOD_OUT_HIGH); + if (IS_ERR(wm2200->ldo_ena)) { + ret = PTR_ERR(wm2200->ldo_ena); + dev_err(&i2c->dev, "Failed to request LDOENA GPIO %d\n", + ret); + goto err_enable; + } + if (wm2200->ldo_ena) { + gpiod_set_consumer_name(wm2200->ldo_ena, "WM2200 LDOENA"); msleep(2); } - if (wm2200->pdata.reset) { - ret = devm_gpio_request_one(&i2c->dev, wm2200->pdata.reset, - GPIOF_OUT_INIT_HIGH, - "WM2200 /RESET"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", - wm2200->pdata.reset, ret); - goto err_ldo; - } + wm2200->reset = devm_gpiod_get_optional(&i2c->dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm2200->reset)) { + ret = PTR_ERR(wm2200->reset); + dev_err(&i2c->dev, "Failed to request RESET GPIO %d\n", + ret); + goto err_ldo; } + gpiod_set_consumer_name(wm2200->reset, "WM2200 /RESET"); ret = regmap_read(wm2200->regmap, WM2200_SOFTWARE_RESET, ®); if (ret < 0) { @@ -2403,11 +2406,9 @@ err_pm_runtime: if (i2c->irq) free_irq(i2c->irq, wm2200); err_reset: - if (wm2200->pdata.reset) - gpio_set_value_cansleep(wm2200->pdata.reset, 0); + gpiod_set_value_cansleep(wm2200->reset, 1); err_ldo: - if (wm2200->pdata.ldo_ena) - gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + gpiod_set_value_cansleep(wm2200->ldo_ena, 0); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); @@ -2421,10 +2422,9 @@ static void wm2200_i2c_remove(struct i2c_client *i2c) pm_runtime_disable(&i2c->dev); if (i2c->irq) free_irq(i2c->irq, wm2200); - if (wm2200->pdata.reset) - gpio_set_value_cansleep(wm2200->pdata.reset, 0); - if (wm2200->pdata.ldo_ena) - gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + /* Assert RESET, disable LDO */ + gpiod_set_value_cansleep(wm2200->reset, 1); + gpiod_set_value_cansleep(wm2200->ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); } @@ -2436,8 +2436,7 @@ static int wm2200_runtime_suspend(struct device *dev) regcache_cache_only(wm2200->regmap, true); regcache_mark_dirty(wm2200->regmap); - if (wm2200->pdata.ldo_ena) - gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 0); + gpiod_set_value_cansleep(wm2200->ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); @@ -2457,8 +2456,8 @@ static int wm2200_runtime_resume(struct device *dev) return ret; } - if (wm2200->pdata.ldo_ena) { - gpio_set_value_cansleep(wm2200->pdata.ldo_ena, 1); + if (wm2200->ldo_ena) { + gpiod_set_value_cansleep(wm2200->ldo_ena, 1); msleep(2); } From 8563cfe39ba5d00d974df25e310fb61b5b3687e1 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 8 Dec 2023 11:09:28 +0100 Subject: [PATCH 193/337] ASoC: wm5100: Convert to GPIO descriptors This converts the WM5100 codec to use GPIO descriptors, a pretty straight-forward conversion with the following peculiarities: - The driver is instantiating a GPIO chip named wm5100, and the headphone polarity detection GPIO is lifted from there. We add this to the GPIO descriptor table as well, and we can then get rid of also the base address for the GPIO chip from the platform data and just use dynamic numbering. - Fix up the only in-tree user which is the Cragganmore 6410 module. Signed-off-by: Linus Walleij Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20231208-descriptors-sound-wlf-v1-4-c4dab6f521ec@linaro.org Signed-off-by: Mark Brown --- arch/arm/mach-s3c/mach-crag6410-module.c | 17 +++- include/sound/wm5100.h | 4 - sound/soc/codecs/wm5100.c | 107 +++++++++-------------- 3 files changed, 52 insertions(+), 76 deletions(-) diff --git a/arch/arm/mach-s3c/mach-crag6410-module.c b/arch/arm/mach-s3c/mach-crag6410-module.c index 1df6b7e52c9d..59456fbebf1b 100644 --- a/arch/arm/mach-s3c/mach-crag6410-module.c +++ b/arch/arm/mach-s3c/mach-crag6410-module.c @@ -70,10 +70,19 @@ static struct spi_board_info balblair_devs[] = { }, }; +static struct gpiod_lookup_table wm5100_gpiod_table = { + .dev_id = "1-001a", /* Device 001a on I2C bus 1 */ + .table = { + GPIO_LOOKUP("GPION", 7, + "wlf,ldo1ena", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("wm5100", 3, + "hp-pol", GPIO_ACTIVE_HIGH), + { }, + }, +}; + static struct wm5100_pdata wm5100_pdata = { - .ldo_ena = S3C64XX_GPN(7), .irq_flags = IRQF_TRIGGER_HIGH, - .gpio_base = CODEC_GPIO_BASE, .in_mode = { WM5100_IN_DIFF, @@ -82,7 +91,6 @@ static struct wm5100_pdata wm5100_pdata = { WM5100_IN_SE, }, - .hp_pol = CODEC_GPIO_BASE + 3, .jack_modes = { { WM5100_MICDET_MICBIAS3, 0, 0 }, { WM5100_MICDET_MICBIAS2, 1, 1 }, @@ -366,7 +374,8 @@ static const struct { { .id = 0x3a, .rev = 0xff, .name = "1259-EV1 Tobermory", .i2c_devs = wm1259_devs, .num_i2c_devs = ARRAY_SIZE(wm1259_devs) }, { .id = 0x3b, .rev = 0xff, .name = "1255-EV1 Kilchoman", - .i2c_devs = wm1255_devs, .num_i2c_devs = ARRAY_SIZE(wm1255_devs) }, + .i2c_devs = wm1255_devs, .num_i2c_devs = ARRAY_SIZE(wm1255_devs), + .gpiod_table = &wm5100_gpiod_table }, { .id = 0x3c, .rev = 0xff, .name = "1273-EV1 Longmorn" }, { .id = 0x3d, .rev = 0xff, .name = "1277-EV1 Littlemill", .i2c_devs = wm1277_devs, .num_i2c_devs = ARRAY_SIZE(wm1277_devs), diff --git a/include/sound/wm5100.h b/include/sound/wm5100.h index b94badf72947..1c48090fdb2c 100644 --- a/include/sound/wm5100.h +++ b/include/sound/wm5100.h @@ -36,11 +36,7 @@ struct wm5100_jack_mode { #define WM5100_GPIO_SET 0x10000 struct wm5100_pdata { - int reset; /** GPIO controlling /RESET, if any */ - int ldo_ena; /** GPIO controlling LODENA, if any */ - int hp_pol; /** GPIO controlling headset polarity, if any */ int irq_flags; - int gpio_base; struct wm5100_jack_mode jack_modes[2]; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index ff63723928a1..7ee4b45c0834 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -15,7 +15,7 @@ #include #include #include -#include +#include #include #include #include @@ -55,6 +55,9 @@ struct wm5100_priv { struct snd_soc_component *component; struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; + struct gpio_desc *reset; + struct gpio_desc *ldo_ena; + struct gpio_desc *hp_pol; int rev; @@ -205,9 +208,9 @@ static void wm5100_free_sr(struct snd_soc_component *component, int rate) static int wm5100_reset(struct wm5100_priv *wm5100) { - if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 0); - gpio_set_value_cansleep(wm5100->pdata.reset, 1); + if (wm5100->reset) { + gpiod_set_value_cansleep(wm5100->reset, 1); + gpiod_set_value_cansleep(wm5100->reset, 0); return 0; } else { @@ -1974,7 +1977,7 @@ static void wm5100_set_detect_mode(struct wm5100_priv *wm5100, int the_mode) if (WARN_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes))) return; - gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); + gpiod_set_value_cansleep(wm5100->hp_pol, mode->hp_pol); regmap_update_bits(wm5100->regmap, WM5100_ACCESSORY_DETECT_MODE_1, WM5100_ACCDET_BIAS_SRC_MASK | WM5100_ACCDET_SRC, @@ -2299,11 +2302,7 @@ static void wm5100_init_gpio(struct i2c_client *i2c) wm5100->gpio_chip = wm5100_template_chip; wm5100->gpio_chip.ngpio = 6; wm5100->gpio_chip.parent = &i2c->dev; - - if (wm5100->pdata.gpio_base) - wm5100->gpio_chip.base = wm5100->pdata.gpio_base; - else - wm5100->gpio_chip.base = -1; + wm5100->gpio_chip.base = -1; ret = gpiochip_add_data(&wm5100->gpio_chip, wm5100); if (ret != 0) @@ -2349,35 +2348,20 @@ static int wm5100_probe(struct snd_soc_component *component) snd_soc_dapm_new_controls(dapm, wm5100_dapm_widgets_noirq, ARRAY_SIZE(wm5100_dapm_widgets_noirq)); - if (wm5100->pdata.hp_pol) { - ret = gpio_request_one(wm5100->pdata.hp_pol, - GPIOF_OUT_INIT_HIGH, "WM5100 HP_POL"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request HP_POL %d: %d\n", - wm5100->pdata.hp_pol, ret); - goto err_gpio; - } + wm5100->hp_pol = devm_gpiod_get_optional(&i2c->dev, "hp-pol", + GPIOD_OUT_HIGH); + if (IS_ERR(wm5100->hp_pol)) { + ret = PTR_ERR(wm5100->hp_pol); + dev_err(&i2c->dev, "Failed to request HP_POL GPIO: %d\n", + ret); + return ret; } return 0; - -err_gpio: - - return ret; -} - -static void wm5100_remove(struct snd_soc_component *component) -{ - struct wm5100_priv *wm5100 = snd_soc_component_get_drvdata(component); - - if (wm5100->pdata.hp_pol) { - gpio_free(wm5100->pdata.hp_pol); - } } static const struct snd_soc_component_driver soc_component_dev_wm5100 = { .probe = wm5100_probe, - .remove = wm5100_remove, .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, .seq_notifier = wm5100_seq_notifier, @@ -2460,26 +2444,26 @@ static int wm5100_i2c_probe(struct i2c_client *i2c) goto err; } - if (wm5100->pdata.ldo_ena) { - ret = gpio_request_one(wm5100->pdata.ldo_ena, - GPIOF_OUT_INIT_HIGH, "WM5100 LDOENA"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request LDOENA %d: %d\n", - wm5100->pdata.ldo_ena, ret); - goto err_enable; - } + wm5100->ldo_ena = devm_gpiod_get_optional(&i2c->dev, "wlf,ldo1ena", + GPIOD_OUT_HIGH); + if (IS_ERR(wm5100->ldo_ena)) { + ret = PTR_ERR(wm5100->ldo_ena); + dev_err(&i2c->dev, "Failed to request LDOENA GPIO: %d\n", ret); + goto err_enable; + } + if (wm5100->ldo_ena) { + gpiod_set_consumer_name(wm5100->ldo_ena, "WM5100 LDOENA"); msleep(2); } - if (wm5100->pdata.reset) { - ret = gpio_request_one(wm5100->pdata.reset, - GPIOF_OUT_INIT_HIGH, "WM5100 /RESET"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request /RESET %d: %d\n", - wm5100->pdata.reset, ret); - goto err_ldo; - } + wm5100->reset = devm_gpiod_get_optional(&i2c->dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(wm5100->reset)) { + ret = PTR_ERR(wm5100->reset); + dev_err(&i2c->dev, "Failed to request /RESET GPIO: %d\n", ret); + goto err_ldo; } + gpiod_set_consumer_name(wm5100->reset, "WM5100 /RESET"); ret = regmap_read(wm5100->regmap, WM5100_SOFTWARE_RESET, ®); if (ret < 0) { @@ -2619,15 +2603,9 @@ err_reset: if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); - if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 0); - gpio_free(wm5100->pdata.reset); - } + gpiod_set_value_cansleep(wm5100->reset, 1); err_ldo: - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - gpio_free(wm5100->pdata.ldo_ena); - } + gpiod_set_value_cansleep(wm5100->ldo_ena, 0); err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); @@ -2643,14 +2621,8 @@ static void wm5100_i2c_remove(struct i2c_client *i2c) if (i2c->irq) free_irq(i2c->irq, wm5100); wm5100_free_gpio(i2c); - if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 0); - gpio_free(wm5100->pdata.reset); - } - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); - gpio_free(wm5100->pdata.ldo_ena); - } + gpiod_set_value_cansleep(wm5100->reset, 1); + gpiod_set_value_cansleep(wm5100->ldo_ena, 0); } #ifdef CONFIG_PM @@ -2660,8 +2632,7 @@ static int wm5100_runtime_suspend(struct device *dev) regcache_cache_only(wm5100->regmap, true); regcache_mark_dirty(wm5100->regmap); - if (wm5100->pdata.ldo_ena) - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); + gpiod_set_value_cansleep(wm5100->ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); @@ -2681,8 +2652,8 @@ static int wm5100_runtime_resume(struct device *dev) return ret; } - if (wm5100->pdata.ldo_ena) { - gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 1); + if (wm5100->ldo_ena) { + gpiod_set_value_cansleep(wm5100->ldo_ena, 1); msleep(2); } From 729f02ec02ae12e5d8a53171bd37e9de56f33677 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Fri, 8 Dec 2023 11:09:29 +0100 Subject: [PATCH 194/337] ASoC: wm8996: Convert to GPIO descriptors This converts the WM8996 codec to use GPIO descriptors, an a similar way to WM5100. The driver is instantiating a GPIO chip named wm8996, and we get rid of the base address for the GPIO chip from the platform data and just use dynamic numbering. Move base and ngpio into the static gpio_chip template. Fix up the only in-tree user which is the Cragganmore 6410 module. Signed-off-by: Linus Walleij Reviewed-by: Charles Keepax Link: https://lore.kernel.org/r/20231208-descriptors-sound-wlf-v1-5-c4dab6f521ec@linaro.org Signed-off-by: Mark Brown --- arch/arm/mach-s3c/mach-crag6410-module.c | 14 ++++-- include/sound/wm8996.h | 3 -- sound/soc/codecs/wm8996.c | 58 ++++++++++-------------- 3 files changed, 36 insertions(+), 39 deletions(-) diff --git a/arch/arm/mach-s3c/mach-crag6410-module.c b/arch/arm/mach-s3c/mach-crag6410-module.c index 59456fbebf1b..2de1a89f6e99 100644 --- a/arch/arm/mach-s3c/mach-crag6410-module.c +++ b/arch/arm/mach-s3c/mach-crag6410-module.c @@ -127,9 +127,16 @@ static struct wm8996_retune_mobile_config wm8996_retune[] = { }, }; +static struct gpiod_lookup_table wm8996_gpiod_table = { + .dev_id = "1-001a", /* Device 001a on I2C bus 1 */ + .table = { + GPIO_LOOKUP("GPION", 7, + "wlf,ldo1ena", GPIO_ACTIVE_HIGH), + { }, + }, +}; + static struct wm8996_pdata wm8996_pdata __initdata = { - .ldo_ena = S3C64XX_GPN(7), - .gpio_base = CODEC_GPIO_BASE, .micdet_def = 1, .inl_mode = WM8996_DIFFERRENTIAL_1, .inr_mode = WM8996_DIFFERRENTIAL_1, @@ -370,7 +377,8 @@ static const struct { .spi_devs = balblair_devs, .num_spi_devs = ARRAY_SIZE(balblair_devs) }, { .id = 0x39, .rev = 0xff, .name = "1254-EV1 Dallas Dhu", - .i2c_devs = wm1254_devs, .num_i2c_devs = ARRAY_SIZE(wm1254_devs) }, + .i2c_devs = wm1254_devs, .num_i2c_devs = ARRAY_SIZE(wm1254_devs), + .gpiod_table = &wm8996_gpiod_table }, { .id = 0x3a, .rev = 0xff, .name = "1259-EV1 Tobermory", .i2c_devs = wm1259_devs, .num_i2c_devs = ARRAY_SIZE(wm1259_devs) }, { .id = 0x3b, .rev = 0xff, .name = "1255-EV1 Kilchoman", diff --git a/include/sound/wm8996.h b/include/sound/wm8996.h index 247f9917e33d..342abeef288f 100644 --- a/include/sound/wm8996.h +++ b/include/sound/wm8996.h @@ -33,8 +33,6 @@ struct wm8996_retune_mobile_config { struct wm8996_pdata { int irq_flags; /** Set IRQ trigger flags; default active low */ - int ldo_ena; /** GPIO for LDO1; -1 for none */ - int micdet_def; /** Default MICDET_SRC/HP1FB_SRC/MICD_BIAS */ enum wm8996_inmode inl_mode; @@ -42,7 +40,6 @@ struct wm8996_pdata { u32 spkmute_seq; /** Value for register 0x802 */ - int gpio_base; u32 gpio_default[5]; int num_retune_mobile_cfgs; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index df6195778c57..e738326e33ed 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -14,7 +14,7 @@ #include #include #include -#include +#include #include #include #include @@ -51,7 +51,7 @@ struct wm8996_priv { struct regmap *regmap; struct snd_soc_component *component; - int ldo1ena; + struct gpio_desc *ldo_ena; int sysclk; int sysclk_src; @@ -1596,9 +1596,9 @@ static int wm8996_set_bias_level(struct snd_soc_component *component, return ret; } - if (wm8996->pdata.ldo_ena >= 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, - 1); + if (wm8996->ldo_ena) { + gpiod_set_value_cansleep(wm8996->ldo_ena, + 1); msleep(5); } @@ -1615,8 +1615,8 @@ static int wm8996_set_bias_level(struct snd_soc_component *component, case SND_SOC_BIAS_OFF: regcache_cache_only(wm8996->regmap, true); - if (wm8996->pdata.ldo_ena >= 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + if (wm8996->ldo_ena) { + gpiod_set_value_cansleep(wm8996->ldo_ena, 0); regcache_cache_only(wm8996->regmap, true); } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), @@ -2188,6 +2188,8 @@ static const struct gpio_chip wm8996_template_chip = { .direction_input = wm8996_gpio_direction_in, .get = wm8996_gpio_get, .can_sleep = 1, + .ngpio = 5, + .base = -1, }; static void wm8996_init_gpio(struct wm8996_priv *wm8996) @@ -2195,14 +2197,8 @@ static void wm8996_init_gpio(struct wm8996_priv *wm8996) int ret; wm8996->gpio_chip = wm8996_template_chip; - wm8996->gpio_chip.ngpio = 5; wm8996->gpio_chip.parent = wm8996->dev; - if (wm8996->pdata.gpio_base) - wm8996->gpio_chip.base = wm8996->pdata.gpio_base; - else - wm8996->gpio_chip.base = -1; - ret = gpiochip_add_data(&wm8996->gpio_chip, wm8996); if (ret != 0) dev_err(wm8996->dev, "Failed to add GPIOs: %d\n", ret); @@ -2771,15 +2767,15 @@ static int wm8996_i2c_probe(struct i2c_client *i2c) memcpy(&wm8996->pdata, dev_get_platdata(&i2c->dev), sizeof(wm8996->pdata)); - if (wm8996->pdata.ldo_ena > 0) { - ret = gpio_request_one(wm8996->pdata.ldo_ena, - GPIOF_OUT_INIT_LOW, "WM8996 ENA"); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to request GPIO %d: %d\n", - wm8996->pdata.ldo_ena, ret); - goto err; - } + wm8996->ldo_ena = devm_gpiod_get_optional(&i2c->dev, "wlf,ldo1ena", + GPIOD_OUT_LOW); + if (IS_ERR(wm8996->ldo_ena)) { + ret = PTR_ERR(wm8996->ldo_ena); + dev_err(&i2c->dev, "Failed to request LDO ENA GPIO: %d\n", + ret); + goto err; } + gpiod_set_consumer_name(wm8996->ldo_ena, "WM8996 ENA"); for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) wm8996->supplies[i].supply = wm8996_supply_names[i]; @@ -2814,8 +2810,8 @@ static int wm8996_i2c_probe(struct i2c_client *i2c) goto err_gpio; } - if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); + if (wm8996->ldo_ena) { + gpiod_set_value_cansleep(wm8996->ldo_ena, 1); msleep(5); } @@ -2847,8 +2843,8 @@ static int wm8996_i2c_probe(struct i2c_client *i2c) dev_info(&i2c->dev, "revision %c\n", (reg & WM8996_CHIP_REV_MASK) + 'A'); - if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + if (wm8996->ldo_ena) { + gpiod_set_value_cansleep(wm8996->ldo_ena, 0); regcache_cache_only(wm8996->regmap, true); } else { ret = regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, @@ -3054,12 +3050,10 @@ err_gpiolib: wm8996_free_gpio(wm8996); err_regmap: err_enable: - if (wm8996->pdata.ldo_ena > 0) - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + if (wm8996->ldo_ena) + gpiod_set_value_cansleep(wm8996->ldo_ena, 0); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); err_gpio: - if (wm8996->pdata.ldo_ena > 0) - gpio_free(wm8996->pdata.ldo_ena); err: return ret; @@ -3070,10 +3064,8 @@ static void wm8996_i2c_remove(struct i2c_client *client) struct wm8996_priv *wm8996 = i2c_get_clientdata(client); wm8996_free_gpio(wm8996); - if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - gpio_free(wm8996->pdata.ldo_ena); - } + if (wm8996->ldo_ena) + gpiod_set_value_cansleep(wm8996->ldo_ena, 0); } static const struct i2c_device_id wm8996_i2c_id[] = { From 5012f9d8acd411bc86229d75c0bb9517f84a60ec Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Sat, 9 Dec 2023 23:01:27 +0100 Subject: [PATCH 195/337] ASoC: wm1250-ev1: Fix uninitialized ret The GPIO descriptor conversion patch left an unitialized ret behind by mistake, fix it by getting rid of the variable altogether. Fixes: 10a366f36e2a ("ASoC: wm1250-ev1: Convert to GPIO descriptors") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202312090953.DiUo3mue-lkp@intel.com/ Signed-off-by: Linus Walleij Link: https://lore.kernel.org/r/20231209-descriptors-sound-wlf-v1-1-5b885ce43ae1@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wm1250-ev1.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index a2a8b2a4b19b..9fa6df48799b 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -129,7 +129,6 @@ static int wm1250_ev1_pdata(struct i2c_client *i2c) { struct wm1250_ev1_pdata *pdata = dev_get_platdata(&i2c->dev); struct wm1250_priv *wm1250; - int ret; if (!pdata) return 0; @@ -165,7 +164,7 @@ static int wm1250_ev1_pdata(struct i2c_client *i2c) dev_set_drvdata(&i2c->dev, wm1250); - return ret; + return 0; } static int wm1250_ev1_probe(struct i2c_client *i2c) From ef14f40a3613ddd43a8dd736b5df6f865dcbb817 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 30 Nov 2023 19:07:56 +0100 Subject: [PATCH 196/337] ASoC: qcom: audioreach: Commonize setting channel mappings Move code assigning channel mapping values to a common helper function. This simplifies three out of four cases, with the last case using incompatible type (uint16_t array instead of uint8_t array). Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231130180758.212172-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/audioreach.c | 35 ++++++++++++++----------------- 1 file changed, 16 insertions(+), 19 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5974c7929dd3..3db5ff367a29 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -267,6 +267,16 @@ void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t token } EXPORT_SYMBOL_GPL(audioreach_alloc_apm_cmd_pkt); +static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) +{ + if (num_channels == 1) { + ch_map[0] = PCM_CHANNEL_L; + } else if (num_channels == 2) { + ch_map[0] = PCM_CHANNEL_L; + ch_map[1] = PCM_CHANNEL_R; + } +} + static void apm_populate_container_config(struct apm_container_obj *cfg, struct audioreach_container *cont) { @@ -864,12 +874,8 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr, mp3_cfg->endianness = PCM_LITTLE_ENDIAN; mp3_cfg->num_channels = mcfg->num_channels; - if (mcfg->num_channels == 1) { - mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L; - } else if (mcfg->num_channels == 2) { - mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L; - mp3_cfg->channel_mapping[1] = PCM_CHANNEL_R; - } + audioreach_set_channel_mapping(mp3_cfg->channel_mapping, + mcfg->num_channels); break; case SND_AUDIOCODEC_AAC: media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED; @@ -1089,13 +1095,8 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph, media_cfg->q_factor = mcfg->bit_width - 1; media_cfg->bits_per_sample = mcfg->bit_width; - if (num_channels == 1) { - media_cfg->channel_mapping[0] = PCM_CHANNEL_L; - } else if (num_channels == 2) { - media_cfg->channel_mapping[0] = PCM_CHANNEL_L; - media_cfg->channel_mapping[1] = PCM_CHANNEL_R; - - } + audioreach_set_channel_mapping(media_cfg->channel_mapping, + num_channels); rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); @@ -1153,12 +1154,8 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, cfg->endianness = PCM_LITTLE_ENDIAN; cfg->num_channels = mcfg->num_channels; - if (mcfg->num_channels == 1) - cfg->channel_mapping[0] = PCM_CHANNEL_L; - else if (num_channels == 2) { - cfg->channel_mapping[0] = PCM_CHANNEL_L; - cfg->channel_mapping[1] = PCM_CHANNEL_R; - } + audioreach_set_channel_mapping(cfg->channel_mapping, + num_channels); } else { rc = audioreach_set_compr_media_format(header, p, mcfg); if (rc) { From bcd684eae5aefc0688fcf43fd555155dd57f27c9 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 30 Nov 2023 19:07:57 +0100 Subject: [PATCH 197/337] ASoC: qcom: audioreach: drop duplicate channel defines q6apm.h header already defines channel mapping values, so drop duplicated devices from audioreach.h. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231130180758.212172-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/audioreach.c | 12 ++++++------ sound/soc/qcom/qdsp6/audioreach.h | 2 -- 2 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 3db5ff367a29..5c7113d46b6f 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -270,10 +270,10 @@ EXPORT_SYMBOL_GPL(audioreach_alloc_apm_cmd_pkt); static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) { if (num_channels == 1) { - ch_map[0] = PCM_CHANNEL_L; + ch_map[0] = PCM_CHANNEL_FL; } else if (num_channels == 2) { - ch_map[0] = PCM_CHANNEL_L; - ch_map[1] = PCM_CHANNEL_R; + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; } } @@ -839,10 +839,10 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph, media_format->num_channels = cfg->num_channels; if (num_channels == 1) { - media_format->channel_mapping[0] = PCM_CHANNEL_L; + media_format->channel_mapping[0] = PCM_CHANNEL_FL; } else if (num_channels == 2) { - media_format->channel_mapping[0] = PCM_CHANNEL_L; - media_format->channel_mapping[1] = PCM_CHANNEL_R; + media_format->channel_mapping[0] = PCM_CHANNEL_FL; + media_format->channel_mapping[1] = PCM_CHANNEL_FR; } rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h index e38111ffd7b9..2c82917b7162 100644 --- a/sound/soc/qcom/qdsp6/audioreach.h +++ b/sound/soc/qcom/qdsp6/audioreach.h @@ -158,8 +158,6 @@ struct param_id_enc_bitrate_param { #define MEDIA_FMT_ID_PCM 0x09001000 #define MEDIA_FMT_ID_MP3 0x09001009 -#define PCM_CHANNEL_L 1 -#define PCM_CHANNEL_R 2 #define SAMPLE_RATE_48K 48000 #define BIT_WIDTH_16 16 From 3c5fcb20e07e3681a645fc3a8d890478ca320825 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 30 Nov 2023 19:07:58 +0100 Subject: [PATCH 198/337] ASoC: qcom: audioreach: Add 4 channel support Add support four channel streams. Map channel 3 and 4 to left/right surround ("quad(side)" from ffmpeg standard channel list) to match what is in qdsp6/q6dsp-common.c driver. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231130180758.212172-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/audioreach.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c index 5c7113d46b6f..5291deac0a0b 100644 --- a/sound/soc/qcom/qdsp6/audioreach.c +++ b/sound/soc/qcom/qdsp6/audioreach.c @@ -274,6 +274,11 @@ static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels) } else if (num_channels == 2) { ch_map[0] = PCM_CHANNEL_FL; ch_map[1] = PCM_CHANNEL_FR; + } else if (num_channels == 4) { + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_LS; + ch_map[3] = PCM_CHANNEL_RS; } } @@ -843,6 +848,11 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph, } else if (num_channels == 2) { media_format->channel_mapping[0] = PCM_CHANNEL_FL; media_format->channel_mapping[1] = PCM_CHANNEL_FR; + } else if (num_channels == 4) { + media_format->channel_mapping[0] = PCM_CHANNEL_FL; + media_format->channel_mapping[1] = PCM_CHANNEL_FR; + media_format->channel_mapping[2] = PCM_CHANNEL_LS; + media_format->channel_mapping[3] = PCM_CHANNEL_RS; } rc = q6apm_send_cmd_sync(graph->apm, pkt, 0); @@ -1063,7 +1073,7 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph, int rc, payload_size; struct gpr_pkt *pkt; - if (num_channels > 2) { + if (num_channels > 4) { dev_err(graph->dev, "Error: Invalid channels (%d)!\n", num_channels); return -EINVAL; } @@ -1117,7 +1127,7 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph, struct gpr_pkt *pkt; void *p; - if (num_channels > 2) { + if (num_channels > 4) { dev_err(graph->dev, "Error: Invalid channels (%d)!\n", num_channels); return -EINVAL; } From 0bfa20b18acbcdd133d41e04e07a2d78bcc04bc5 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 29 Nov 2023 12:30:11 +0100 Subject: [PATCH 199/337] ASoC: dt-bindings: qcom,lpass-rx-macro: Add SM8650 LPASS RX Add bindings for Qualcomm SM8650 Low Power Audio SubSystem (LPASS) RX macro codec, which looks like compatible with earlier SM8550. Cc: Neil Armstrong Signed-off-by: Krzysztof Kozlowski Acked-by: Rob Herring Link: https://msgid.link/r/20231129113014.38837-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../bindings/sound/qcom,lpass-rx-macro.yaml | 21 ++++++++++++------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml index ec4b0ac8ad68..cbc36646100f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -11,12 +11,16 @@ maintainers: properties: compatible: - enum: - - qcom,sc7280-lpass-rx-macro - - qcom,sm8250-lpass-rx-macro - - qcom,sm8450-lpass-rx-macro - - qcom,sm8550-lpass-rx-macro - - qcom,sc8280xp-lpass-rx-macro + oneOf: + - enum: + - qcom,sc7280-lpass-rx-macro + - qcom,sm8250-lpass-rx-macro + - qcom,sm8450-lpass-rx-macro + - qcom,sm8550-lpass-rx-macro + - qcom,sc8280xp-lpass-rx-macro + - items: + - const: qcom,sm8650-lpass-rx-macro + - const: qcom,sm8550-lpass-rx-macro reg: maxItems: 1 @@ -96,8 +100,9 @@ allOf: - if: properties: compatible: - enum: - - qcom,sm8550-lpass-rx-macro + contains: + enum: + - qcom,sm8550-lpass-rx-macro then: properties: clocks: From 5a5085c9ce381f92399c755be6deaf1d76ad57e8 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 29 Nov 2023 12:30:12 +0100 Subject: [PATCH 200/337] ASoC: dt-bindings: qcom,lpass-tx-macro: Add SM8650 LPASS TX Add bindings for Qualcomm SM8650 Low Power Audio SubSystem (LPASS) TX macro codec, which looks like compatible with earlier SM8550. Cc: Neil Armstrong Signed-off-by: Krzysztof Kozlowski Acked-by: Rob Herring Link: https://msgid.link/r/20231129113014.38837-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../bindings/sound/qcom,lpass-tx-macro.yaml | 23 +++++++++++-------- 1 file changed, 14 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml index 962701e9eb42..cee79ac42a33 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -11,13 +11,17 @@ maintainers: properties: compatible: - enum: - - qcom,sc7280-lpass-tx-macro - - qcom,sm6115-lpass-tx-macro - - qcom,sm8250-lpass-tx-macro - - qcom,sm8450-lpass-tx-macro - - qcom,sm8550-lpass-tx-macro - - qcom,sc8280xp-lpass-tx-macro + oneOf: + - enum: + - qcom,sc7280-lpass-tx-macro + - qcom,sm6115-lpass-tx-macro + - qcom,sm8250-lpass-tx-macro + - qcom,sm8450-lpass-tx-macro + - qcom,sm8550-lpass-tx-macro + - qcom,sc8280xp-lpass-tx-macro + - items: + - const: qcom,sm8650-lpass-tx-macro + - const: qcom,sm8550-lpass-tx-macro reg: maxItems: 1 @@ -118,8 +122,9 @@ allOf: - if: properties: compatible: - enum: - - qcom,sm8550-lpass-tx-macro + contains: + enum: + - qcom,sm8550-lpass-tx-macro then: properties: clocks: From f243ef746d0ace20fe092fc1ee9987ecf003f7a4 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 29 Nov 2023 12:30:13 +0100 Subject: [PATCH 201/337] ASoC: dt-bindings: qcom,lpass-va-macro: Add SM8650 LPASS VA Add bindings for Qualcomm SM8650 Low Power Audio SubSystem (LPASS) VA macro codec, which looks like compatible with earlier SM8550. Cc: Neil Armstrong Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231129113014.38837-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../bindings/sound/qcom,lpass-va-macro.yaml | 21 ++++++++++++------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml index 4a56108c444b..ca6b07d5826d 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -11,12 +11,16 @@ maintainers: properties: compatible: - enum: - - qcom,sc7280-lpass-va-macro - - qcom,sm8250-lpass-va-macro - - qcom,sm8450-lpass-va-macro - - qcom,sm8550-lpass-va-macro - - qcom,sc8280xp-lpass-va-macro + oneOf: + - enum: + - qcom,sc7280-lpass-va-macro + - qcom,sm8250-lpass-va-macro + - qcom,sm8450-lpass-va-macro + - qcom,sm8550-lpass-va-macro + - qcom,sc8280xp-lpass-va-macro + - items: + - const: qcom,sm8650-lpass-va-macro + - const: qcom,sm8550-lpass-va-macro reg: maxItems: 1 @@ -115,8 +119,9 @@ allOf: properties: compatible: contains: - enum: - - qcom,sm8550-lpass-va-macro + contains: + enum: + - qcom,sm8550-lpass-va-macro then: properties: clocks: From ab8921e1da8fdca14192c44775151f50c1cdb763 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 29 Nov 2023 12:30:14 +0100 Subject: [PATCH 202/337] ASoC: dt-bindings: qcom,lpass-wsa-macro: Add SM8650 LPASS WSA Add bindings for Qualcomm SM8650 Low Power Audio SubSystem (LPASS) WSA macro codec, which looks like compatible with earlier SM8550. Cc: Neil Armstrong Signed-off-by: Krzysztof Kozlowski Acked-by: Rob Herring Link: https://msgid.link/r/20231129113014.38837-4-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 21 ++++++++++++------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index eea7609d1b33..5fb39d35c8ec 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -11,12 +11,16 @@ maintainers: properties: compatible: - enum: - - qcom,sc7280-lpass-wsa-macro - - qcom,sm8250-lpass-wsa-macro - - qcom,sm8450-lpass-wsa-macro - - qcom,sm8550-lpass-wsa-macro - - qcom,sc8280xp-lpass-wsa-macro + oneOf: + - enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro + - qcom,sm8450-lpass-wsa-macro + - qcom,sm8550-lpass-wsa-macro + - qcom,sc8280xp-lpass-wsa-macro + - items: + - const: qcom,sm8650-lpass-wsa-macro + - const: qcom,sm8550-lpass-wsa-macro reg: maxItems: 1 @@ -94,8 +98,9 @@ allOf: - if: properties: compatible: - enum: - - qcom,sm8550-lpass-wsa-macro + contains: + enum: + - qcom,sm8550-lpass-wsa-macro then: properties: clocks: From 28b0b18d53469833bf513a87732c638775073ba4 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Mon, 11 Dec 2023 12:40:57 +0100 Subject: [PATCH 203/337] ASoC: codec: wsa884x: make use of new mute_unmute_on_trigger flag This fix is based on commit [1] fixing click and pop sounds during SoundWire port start because PA is left unmuted. making use of new mute_unmute_on_trigger flag and removing unmute at PA setup, removes the Click/Pop issue at SoundWire enable. [1] 805ce81826c8 ("ASoC: codecs: wsa883x: make use of new mute_unmute_on_trigger flag") Cc: Srinivas Kandagatla Signed-off-by: Neil Armstrong Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231211-topic-sm8x50-upstream-wsa884x-fix-plop-v1-1-0dc630a19172@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa884x.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 993d76b18b53..f2653df84e4a 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -1654,15 +1654,9 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x1); - snd_soc_component_write_field(component, WSA884X_PA_FSM_EN, - WSA884X_PA_FSM_EN_GLOBAL_PA_EN_MASK, - 0x1); break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_component_write_field(component, WSA884X_PA_FSM_EN, - WSA884X_PA_FSM_EN_GLOBAL_PA_EN_MASK, - 0x0); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); @@ -1786,6 +1780,7 @@ static const struct snd_soc_dai_ops wsa884x_dai_ops = { .hw_free = wsa884x_hw_free, .mute_stream = wsa884x_mute_stream, .set_stream = wsa884x_set_stream, + .mute_unmute_on_trigger = true, }; static struct snd_soc_dai_driver wsa884x_dais[] = { From bbbc18d8c27cc9d40cc9a3b03b61e4df85311146 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 11 Dec 2023 16:00:19 +0000 Subject: [PATCH 204/337] ASoC: cs42l43: Allow HP amp to cool off after current limit Whilst occasional current limiting is fine, constant current limiting should be avoided. Add a back off system that will disable the headphone amp, if a lot of current limiting is seen in a short window of time. Signed-off-by: Charles Keepax Link: https://msgid.link/r/20231211160019.2034442-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43-jack.c | 4 +- sound/soc/codecs/cs42l43.c | 88 +++++++++++++++++++++++++++++++-- sound/soc/codecs/cs42l43.h | 9 ++++ 3 files changed, 96 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c index 73454de068cf..929497edb83d 100644 --- a/sound/soc/codecs/cs42l43-jack.c +++ b/sound/soc/codecs/cs42l43-jack.c @@ -506,7 +506,7 @@ static void cs42l43_start_load_detect(struct cs42l43_codec *priv) priv->load_detect_running = true; - if (priv->hp_ena) { + if (priv->hp_ena && !priv->hp_ilimited) { unsigned long time_left; reinit_completion(&priv->hp_shutdown); @@ -571,7 +571,7 @@ static void cs42l43_stop_load_detect(struct cs42l43_codec *priv) CS42L43_ADC1_EN_MASK | CS42L43_ADC2_EN_MASK, priv->adc_ena); - if (priv->hp_ena) { + if (priv->hp_ena && !priv->hp_ilimited) { unsigned long time_left; reinit_completion(&priv->hp_startup); diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 5c98343ebf71..d2412dab3599 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -138,7 +138,87 @@ CS42L43_IRQ_ERROR(spkr_therm_warm) CS42L43_IRQ_ERROR(spkl_therm_warm) CS42L43_IRQ_ERROR(spkr_sc_detect) CS42L43_IRQ_ERROR(spkl_sc_detect) -CS42L43_IRQ_ERROR(hp_ilimit) + +void cs42l43_hp_ilimit_clear_work(struct work_struct *work) +{ + struct cs42l43_codec *priv = container_of(work, struct cs42l43_codec, + hp_ilimit_clear_work.work); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(priv->component); + + snd_soc_dapm_mutex_lock(dapm); + + priv->hp_ilimit_count--; + + if (priv->hp_ilimit_count) + queue_delayed_work(system_wq, &priv->hp_ilimit_clear_work, + msecs_to_jiffies(CS42L43_HP_ILIMIT_DECAY_MS)); + + snd_soc_dapm_mutex_unlock(dapm); +} + +void cs42l43_hp_ilimit_work(struct work_struct *work) +{ + struct cs42l43_codec *priv = container_of(work, struct cs42l43_codec, + hp_ilimit_work); + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(priv->component); + struct cs42l43 *cs42l43 = priv->core; + + snd_soc_dapm_mutex_lock(dapm); + + if (priv->hp_ilimit_count < CS42L43_HP_ILIMIT_MAX_COUNT) { + if (!priv->hp_ilimit_count) + queue_delayed_work(system_wq, &priv->hp_ilimit_clear_work, + msecs_to_jiffies(CS42L43_HP_ILIMIT_DECAY_MS)); + + priv->hp_ilimit_count++; + snd_soc_dapm_mutex_unlock(dapm); + return; + } + + dev_err(priv->dev, "Disabling headphone for %dmS, due to frequent current limit\n", + CS42L43_HP_ILIMIT_BACKOFF_MS); + + priv->hp_ilimited = true; + + // No need to wait for disable, as just disabling for a period of time + regmap_update_bits(cs42l43->regmap, CS42L43_BLOCK_EN8, + CS42L43_HP_EN_MASK, 0); + + snd_soc_dapm_mutex_unlock(dapm); + + msleep(CS42L43_HP_ILIMIT_BACKOFF_MS); + + snd_soc_dapm_mutex_lock(dapm); + + if (priv->hp_ena && !priv->load_detect_running) { + unsigned long time_left; + + reinit_completion(&priv->hp_startup); + + regmap_update_bits(cs42l43->regmap, CS42L43_BLOCK_EN8, + CS42L43_HP_EN_MASK, priv->hp_ena); + + time_left = wait_for_completion_timeout(&priv->hp_startup, + msecs_to_jiffies(CS42L43_HP_TIMEOUT_MS)); + if (!time_left) + dev_err(priv->dev, "ilimit HP restore timed out\n"); + } + + priv->hp_ilimited = false; + + snd_soc_dapm_mutex_unlock(dapm); +} + +static irqreturn_t cs42l43_hp_ilimit(int irq, void *data) +{ + struct cs42l43_codec *priv = data; + + dev_dbg(priv->dev, "headphone ilimit IRQ\n"); + + queue_work(system_long_wq, &priv->hp_ilimit_work); + + return IRQ_HANDLED; +} #define CS42L43_IRQ_COMPLETE(name) \ static irqreturn_t cs42l43_##name(int irq, void *data) \ @@ -1452,13 +1532,13 @@ static int cs42l43_hp_ev(struct snd_soc_dapm_widget *w, if (ret) return ret; - if (!priv->load_detect_running) + if (!priv->load_detect_running && !priv->hp_ilimited) regmap_update_bits(cs42l43->regmap, CS42L43_BLOCK_EN8, mask, val); break; case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_POST_PMD: - if (priv->load_detect_running) + if (priv->load_detect_running || priv->hp_ilimited) break; ret = cs42l43_dapm_wait_completion(&priv->hp_startup, @@ -2169,7 +2249,9 @@ static int cs42l43_codec_probe(struct platform_device *pdev) INIT_DELAYED_WORK(&priv->tip_sense_work, cs42l43_tip_sense_work); INIT_DELAYED_WORK(&priv->bias_sense_timeout, cs42l43_bias_sense_timeout); INIT_DELAYED_WORK(&priv->button_press_work, cs42l43_button_press_work); + INIT_DELAYED_WORK(&priv->hp_ilimit_clear_work, cs42l43_hp_ilimit_clear_work); INIT_WORK(&priv->button_release_work, cs42l43_button_release_work); + INIT_WORK(&priv->hp_ilimit_work, cs42l43_hp_ilimit_work); pm_runtime_set_autosuspend_delay(priv->dev, 100); pm_runtime_use_autosuspend(priv->dev); diff --git a/sound/soc/codecs/cs42l43.h b/sound/soc/codecs/cs42l43.h index bf4f728eea3e..125e36861d5d 100644 --- a/sound/soc/codecs/cs42l43.h +++ b/sound/soc/codecs/cs42l43.h @@ -28,6 +28,10 @@ #define CS42L43_HP_TIMEOUT_MS 2000 #define CS42L43_LOAD_TIMEOUT_MS 1000 +#define CS42L43_HP_ILIMIT_BACKOFF_MS 1000 +#define CS42L43_HP_ILIMIT_DECAY_MS 300 +#define CS42L43_HP_ILIMIT_MAX_COUNT 4 + #define CS42L43_ASP_MAX_CHANNELS 6 #define CS42L43_N_EQ_COEFFS 15 @@ -88,6 +92,11 @@ struct cs42l43_codec { bool button_detect_running; bool jack_present; int jack_override; + + struct work_struct hp_ilimit_work; + struct delayed_work hp_ilimit_clear_work; + bool hp_ilimited; + int hp_ilimit_count; }; #if IS_REACHABLE(CONFIG_SND_SOC_CS42L43_SDW) From 5ed06e489d20dca141047bdb025d885306ea66da Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 12 Dec 2023 10:41:49 +0000 Subject: [PATCH 205/337] ASoC: cs42l43: Add missing statics for hp_ilimit functions Fixes: bbbc18d8c27c ("ASoC: cs42l43: Allow HP amp to cool off after current limit") Reported-by: Stephen Rothwell Signed-off-by: Charles Keepax Link: https://msgid.link/r/20231212104149.2388753-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index d2412dab3599..6a64681767de 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -139,7 +139,7 @@ CS42L43_IRQ_ERROR(spkl_therm_warm) CS42L43_IRQ_ERROR(spkr_sc_detect) CS42L43_IRQ_ERROR(spkl_sc_detect) -void cs42l43_hp_ilimit_clear_work(struct work_struct *work) +static void cs42l43_hp_ilimit_clear_work(struct work_struct *work) { struct cs42l43_codec *priv = container_of(work, struct cs42l43_codec, hp_ilimit_clear_work.work); @@ -156,7 +156,7 @@ void cs42l43_hp_ilimit_clear_work(struct work_struct *work) snd_soc_dapm_mutex_unlock(dapm); } -void cs42l43_hp_ilimit_work(struct work_struct *work) +static void cs42l43_hp_ilimit_work(struct work_struct *work) { struct cs42l43_codec *priv = container_of(work, struct cs42l43_codec, hp_ilimit_work); From 6475b8e1821c9d14e60592a74c10d75431500c7c Mon Sep 17 00:00:00 2001 From: Daniel Golle Date: Tue, 12 Dec 2023 23:10:07 +0000 Subject: [PATCH 206/337] ASoC: mediatek: mt7986: silence error in case of -EPROBE_DEFER If probe is defered no error should be printed. Use dev_err_probe() to have it muted. Signed-off-by: Daniel Golle Reviewed-by: Maso Huang Reviewed-by: AngeloGioacchino Del Regno Link: https://msgid.link/r/b941a404d97c01ef3e30c49925927b9a7dafeb19.1702422544.git.daniel@makrotopia.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt7986/mt7986-wm8960.c b/sound/soc/mediatek/mt7986/mt7986-wm8960.c index c1390b373410..6982e833421d 100644 --- a/sound/soc/mediatek/mt7986/mt7986-wm8960.c +++ b/sound/soc/mediatek/mt7986/mt7986-wm8960.c @@ -144,7 +144,7 @@ static int mt7986_wm8960_machine_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "%s snd_soc_register_card fail: %d\n", __func__, ret); + dev_err_probe(&pdev->dev, ret, "%s snd_soc_register_card fail\n", __func__); goto err_of_node_put; } From d29351e8c20d61a852bbdfcab7bb7166bd916558 Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 28 Nov 2023 10:11:18 +0200 Subject: [PATCH 207/337] ASoC: audio-graph-card2: Introduce playback-only/capture-only DAI link flags We need this to support MICFIL PDM found on i.MX8MP where the DAI link supports only capture direction. Signed-off-by: Daniel Baluta Link: https://msgid.link/r/20231128081119.106360-2-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- include/sound/simple_card_utils.h | 3 +++ sound/soc/generic/audio-graph-card2.c | 6 ++++++ sound/soc/generic/simple-card-utils.c | 19 +++++++++++++++++++ 3 files changed, 28 insertions(+) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index e5da10b4c43b..ad67957b7b48 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -195,6 +195,9 @@ int graph_util_is_ports0(struct device_node *port); int graph_util_parse_dai(struct device *dev, struct device_node *ep, struct snd_soc_dai_link_component *dlc, int *is_single_link); +int graph_util_parse_link_direction(struct device_node *np, + bool *is_playback_only, bool *is_capture_only); + #ifdef DEBUG static inline void simple_util_debug_dai(struct simple_util_priv *priv, char *name, diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 78d9679decda..f880a7f73522 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -760,6 +760,7 @@ static void graph_link_init(struct simple_util_priv *priv, struct device_node *ep; struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; + bool playback_only = 0, capture_only = 0; unsigned int bit_frame = 0; if (graph_lnk_is_multi(port)) { @@ -798,6 +799,11 @@ static void graph_link_init(struct simple_util_priv *priv, if (is_cpu_node) daiclk = snd_soc_daifmt_clock_provider_flipped(daiclk); + graph_util_parse_link_direction(port, &playback_only, &capture_only); + + dai_link->playback_only = playback_only; + dai_link->capture_only = capture_only; + dai_link->dai_fmt = daifmt | daiclk; dai_link->init = simple_util_dai_init; dai_link->ops = &graph_ops; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index cfa70a56ff0f..9006ef5e95f5 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -1129,6 +1129,25 @@ parse_dai_end: } EXPORT_SYMBOL_GPL(graph_util_parse_dai); +int graph_util_parse_link_direction(struct device_node *np, + bool *playback_only, bool *capture_only) +{ + bool is_playback_only = false; + bool is_capture_only = false; + + is_playback_only = of_property_read_bool(np, "playback-only"); + is_capture_only = of_property_read_bool(np, "capture-only"); + + if (is_playback_only && is_capture_only) + return -EINVAL; + + *playback_only = is_playback_only; + *capture_only = is_capture_only; + + return 0; +} +EXPORT_SYMBOL_GPL(graph_util_parse_link_direction); + /* Module information */ MODULE_AUTHOR("Kuninori Morimoto "); MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); From af29e51bee8223d8b26e574489d2433b88cdeb2f Mon Sep 17 00:00:00 2001 From: Daniel Baluta Date: Tue, 28 Nov 2023 10:11:19 +0200 Subject: [PATCH 208/337] ASoC: dt-bindings: audio-graph-port: Document new DAI link flags playback-only/capture-only Document new playback-only and capture-only flags which can be used when dai link can only support just one direction: playback or capture but not both. Signed-off-by: Daniel Baluta Link: https://msgid.link/r/20231128081119.106360-3-daniel.baluta@oss.nxp.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/audio-graph-port.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml index 60b5e3fd1115..b13c08de505e 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-port.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph-port.yaml @@ -19,6 +19,12 @@ definitions: properties: mclk-fs: $ref: simple-card.yaml#/definitions/mclk-fs + playback-only: + description: port connection used only for playback + $ref: /schemas/types.yaml#/definitions/flag + capture-only: + description: port connection used only for capture + $ref: /schemas/types.yaml#/definitions/flag endpoint-base: allOf: From c18852cf16dc0ee98d661f48edfa815ee240ee57 Mon Sep 17 00:00:00 2001 From: Ghanshyam Agrawal Date: Fri, 15 Dec 2023 08:41:44 +0530 Subject: [PATCH 209/337] ALSA: au88x0: fixed spelling mistakes in au88x0_core.c Multiple spelling mistakes were reported by codespell. They were fixed. Signed-off-by: Ghanshyam Agrawal Link: https://lore.kernel.org/r/20231215031144.521359-1-ghanshyam1898@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_core.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index f217c02dfdfa..e5d867637336 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1195,7 +1195,7 @@ static int vortex_adbdma_bufshift(vortex_t * vortex, int adbdma) VORTEX_ADBDMA_BUFBASE + (((adbdma << 2) + pp) << 2), snd_pcm_sgbuf_get_addr(dma->substream, dma->period_bytes * p)); - /* Force write thru cache. */ + /* Force write through cache. */ hwread(vortex->mmio, VORTEX_ADBDMA_BUFBASE + (((adbdma << 2) + pp) << 2)); } @@ -1237,7 +1237,7 @@ static void vortex_adbdma_resetup(vortex_t *vortex, int adbdma) { VORTEX_ADBDMA_BUFBASE + (((adbdma << 2) + pp) << 2), snd_pcm_sgbuf_get_addr(dma->substream, dma->period_bytes * p)); - /* Force write thru cache. */ + /* Force write through cache. */ hwread(vortex->mmio, VORTEX_ADBDMA_BUFBASE + (((adbdma << 2)+pp) << 2)); } } @@ -1466,7 +1466,7 @@ static int vortex_wtdma_bufshift(vortex_t * vortex, int wtdma) (((wtdma << 2) + pp) << 2), snd_pcm_sgbuf_get_addr(dma->substream, dma->period_bytes * p)); - /* Force write thru cache. */ + /* Force write through cache. */ hwread(vortex->mmio, VORTEX_WTDMA_BUFBASE + (((wtdma << 2) + pp) << 2)); } @@ -1854,7 +1854,7 @@ vortex_connection_mixin_mix(vortex_t * vortex, int en, unsigned char mixin, vortex_mix_disableinput(vortex, mix, mixin, a); } -// Connect absolut address to mixin. +// Connect absolute address to mixin. static void vortex_connection_adb_mixin(vortex_t * vortex, int en, unsigned char channel, unsigned char source, @@ -1880,7 +1880,7 @@ vortex_connection_src_src_adbdma(vortex_t * vortex, int en, ADB_DMA(adbdma)); } -// mix to absolut address. +// mix to absolute address. static void vortex_connection_mix_adb(vortex_t * vortex, int en, unsigned char ch, unsigned char mix, unsigned char dest) From 57cd29a82574b0e9d99ed5789801c96f765e8fcb Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Fri, 15 Dec 2023 16:31:00 +0800 Subject: [PATCH 210/337] ASoC: SOF: IPC4: synchronize fw_config_params with fw definitions MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Update fw_config_params in driver. Signed-off-by: Rander Wang Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://msgid.link/r/20231215083102.3064200-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/sound/sof/ipc4/header.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/sound/sof/ipc4/header.h b/include/sound/sof/ipc4/header.h index 2c81d6dde577..1eb538e18236 100644 --- a/include/sound/sof/ipc4/header.h +++ b/include/sound/sof/ipc4/header.h @@ -423,6 +423,12 @@ enum sof_ipc4_fw_config_params { SOF_IPC4_FW_CFG_RESERVED, SOF_IPC4_FW_CFG_POWER_GATING_POLICY, SOF_IPC4_FW_CFG_ASSERT_MODE, + SOF_IPC4_FW_RESERVED1, + SOF_IPC4_FW_RESERVED2, + SOF_IPC4_FW_RESERVED3, + SOF_IPC4_FW_RESERVED4, + SOF_IPC4_FW_RESERVED5, + SOF_IPC4_FW_CONTEXT_SAVE }; struct sof_ipc4_fw_version { From 855a4772be9dc777cbcd580c8a07d9c54908219b Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Fri, 15 Dec 2023 16:31:01 +0800 Subject: [PATCH 211/337] ASoC: SOF: IPC4: query fw_context_save feature from fw MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Driver queries fw_context_save feature when fw is ready and can skip library reload with this feature since library is saved in persistent memory. The default value of fw_context_save is true unless fw reports false. Signed-off-by: Rander Wang Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://msgid.link/r/20231215083102.3064200-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/lnl.c | 2 ++ sound/soc/sof/intel/mtl.c | 2 ++ sound/soc/sof/intel/tgl.c | 2 ++ sound/soc/sof/ipc4-loader.c | 3 +++ sound/soc/sof/ipc4-priv.h | 1 + 5 files changed, 10 insertions(+) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index a095f5bcf50d..30712ea05a7a 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -133,6 +133,8 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) ipc4_data->mtrace_type = SOF_IPC4_MTRACE_INTEL_CAVS_2; + ipc4_data->fw_context_save = true; + /* External library loading support */ ipc4_data->load_library = hda_dsp_ipc4_load_library; diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 3121a89219dd..c14a80dd90fd 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -718,6 +718,8 @@ int sof_mtl_ops_init(struct snd_sof_dev *sdev) ipc4_data->mtrace_type = SOF_IPC4_MTRACE_INTEL_CAVS_2; + ipc4_data->fw_context_save = true; + /* External library loading support */ ipc4_data->load_library = hda_dsp_ipc4_load_library; diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index d890cac6cb01..c2bb04c89b9d 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -91,6 +91,8 @@ int sof_tgl_ops_init(struct snd_sof_dev *sdev) ipc4_data->mtrace_type = SOF_IPC4_MTRACE_INTEL_CAVS_2; + ipc4_data->fw_context_save = true; + /* External library loading support */ ipc4_data->load_library = hda_dsp_ipc4_load_library; diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index eaa04762eb11..3539b0a66e1b 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -391,6 +391,9 @@ int sof_ipc4_query_fw_configuration(struct snd_sof_dev *sdev) goto out; } break; + case SOF_IPC4_FW_CONTEXT_SAVE: + ipc4_data->fw_context_save = *tuple->value; + break; default: break; } diff --git a/sound/soc/sof/ipc4-priv.h b/sound/soc/sof/ipc4-priv.h index fea93b693f4d..1d39836d5efa 100644 --- a/sound/soc/sof/ipc4-priv.h +++ b/sound/soc/sof/ipc4-priv.h @@ -81,6 +81,7 @@ struct sof_ipc4_fw_data { u32 mtrace_log_bytes; int max_num_pipelines; u32 max_libs_count; + bool fw_context_save; int (*load_library)(struct snd_sof_dev *sdev, struct sof_ipc4_fw_library *fw_lib, bool reload); From 3a0e7bb86f8728d94d55c56fb73e86be7976c163 Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Fri, 15 Dec 2023 16:31:02 +0800 Subject: [PATCH 212/337] ASoC: SOF: Intel: check fw_context_save for library reload MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If fw_context_save is defined by fw, driver can skip library reload on d3 exit or reload library. Signed-off-by: Rander Wang Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Link: https://msgid.link/r/20231215083102.3064200-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 1805cf754beb..b81f231abee3 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -519,14 +519,15 @@ int hda_dsp_ipc4_load_library(struct snd_sof_dev *sdev, struct sof_ipc4_fw_library *fw_lib, bool reload) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + struct sof_ipc4_fw_data *ipc4_data = sdev->private; struct hdac_ext_stream *hext_stream; struct firmware stripped_firmware; struct sof_ipc4_msg msg = {}; struct snd_dma_buffer dmab; int ret, ret1; - /* IMR booting will restore the libraries as well, skip the loading */ - if (reload && hda->booted_from_imr) + /* if IMR booting is enabled and fw context is saved for D3 state, skip the loading */ + if (reload && hda->booted_from_imr && ipc4_data->fw_context_save) return 0; /* the fw_lib has been verified during loading, we can trust the validity here */ From 02842209fc29bddb2b673ceb8f681267857c4bc1 Mon Sep 17 00:00:00 2001 From: Wang Jinchao Date: Fri, 15 Dec 2023 17:20:00 +0800 Subject: [PATCH 213/337] ASoC: SOF: amd: remove duplicated including remove the second \#include "../sof-audio.h" Signed-off-by: Wang Jinchao Link: https://msgid.link/r/202312151719+0800-wangjinchao@xfusion.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-common.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index 3a0c7688dcfe..2d72c6d55dc8 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -13,7 +13,6 @@ #include "../sof-priv.h" #include "../sof-audio.h" #include "../ops.h" -#include "../sof-audio.h" #include "acp.h" #include "acp-dsp-offset.h" #include From e7a4a2fd9a4116286a1523ea1a5cbabd2c36f5b9 Mon Sep 17 00:00:00 2001 From: Wang Jinchao Date: Fri, 15 Dec 2023 17:13:51 +0800 Subject: [PATCH 214/337] ASoC: fsl_mqs: remove duplicated including rm the second \#include Signed-off-by: Wang Jinchao Link: https://msgid.link/r/202312151713+0800-wangjinchao@xfusion.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_mqs.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index 86704ba5f6f0..60929c36a0e3 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -11,7 +11,6 @@ #include #include #include -#include #include #include #include From c95a2a0be0b1bba2e051faa105c2e0401fc2de33 Mon Sep 17 00:00:00 2001 From: Syed Saba Kareem Date: Fri, 15 Dec 2023 18:32:42 +0530 Subject: [PATCH 215/337] ASoC: amd: acp: add pm ops support for renoir platform. Add pm ops for renoir platform. Signed-off-by: Syed Saba Kareem Reviewed-by: Mario Limonciello Link: https://msgid.link/r/20231215130300.1247475-1-Syed.SabaKareem@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-renoir.c | 37 ++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index a591482a0726..b0e181c9a733 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -20,6 +20,7 @@ #include #include #include +#include #include "amd.h" #include "acp-mach.h" @@ -196,6 +197,11 @@ static int renoir_audio_probe(struct platform_device *pdev) acp_enable_interrupts(adata); acp_platform_register(dev); + pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); + pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); + pm_runtime_enable(&pdev->dev); return 0; } @@ -208,11 +214,42 @@ static void renoir_audio_remove(struct platform_device *pdev) acp_platform_unregister(dev); } +static int __maybe_unused rn_pcm_resume(struct device *dev) +{ + struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_stream *stream; + struct snd_pcm_substream *substream; + snd_pcm_uframes_t buf_in_frames; + u64 buf_size; + + spin_lock(&adata->acp_lock); + list_for_each_entry(stream, &adata->stream_list, list) { + substream = stream->substream; + if (substream && substream->runtime) { + buf_in_frames = (substream->runtime->buffer_size); + buf_size = frames_to_bytes(substream->runtime, buf_in_frames); + config_pte_for_stream(adata, stream); + config_acp_dma(adata, stream, buf_size); + if (stream->dai_id) + restore_acp_i2s_params(substream, adata, stream); + else + restore_acp_pdm_params(substream, adata); + } + } + spin_unlock(&adata->acp_lock); + return 0; +} + +static const struct dev_pm_ops rn_dma_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(NULL, rn_pcm_resume) +}; + static struct platform_driver renoir_driver = { .probe = renoir_audio_probe, .remove_new = renoir_audio_remove, .driver = { .name = "acp_asoc_renoir", + .pm = &rn_dma_pm_ops, }, }; From c7e37b07cc7564a07125ae48c11fd1ca2bcbeae2 Mon Sep 17 00:00:00 2001 From: Ghanshyam Agrawal Date: Mon, 18 Dec 2023 12:24:42 +0530 Subject: [PATCH 216/337] ALSA: au88x0: fixed a typo Fixed typo in the word communicate Signed-off-by: Ghanshyam Agrawal Link: https://lore.kernel.org/r/20231218065442.43523-1-ghanshyam1898@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index eb234153691b..62b10b0e07b1 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -10,7 +10,7 @@ * Thanks to the ALSA developers, they helped a lot working out * the ALSA part. * Thanks also to Sourceforge for maintaining the old binary drivers, - * and the forum, where developers could comunicate. + * and the forum, where developers could communicate. * * Now at least i can play Legacy DOOM with MIDI music :-) */ From 487b467206fb2f3a21c93759d3b0ffe7044ed197 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Thu, 14 Dec 2023 14:15:42 +0100 Subject: [PATCH 217/337] ASoC: hisilicon: Drop GPIO include This driver is including the legacy GPIO header but not using any symbols from it. Drop the include. Signed-off-by: Linus Walleij Link: https://msgid.link/r/20231214-gpio-descriptors-sound-misc-v1-1-e3004176bd8b@linaro.org Signed-off-by: Mark Brown --- sound/soc/hisilicon/hi6210-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index dd7d2a077248..250ae3781d14 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -15,7 +15,6 @@ #include #include #include -#include #include #include #include From 809fc84b371a0364160254037d2bc34a8f5ce372 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Thu, 14 Dec 2023 14:15:43 +0100 Subject: [PATCH 218/337] ASoC: qcom: sc7180: Drop GPIO include This driver is including the legacy GPIO header but not using any symbols from it. Drop the include. Signed-off-by: Linus Walleij Link: https://msgid.link/r/20231214-gpio-descriptors-sound-misc-v1-2-e3004176bd8b@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sc7180.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index b0320a74d508..4ab34a8842ce 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -6,7 +6,6 @@ #include #include -#include #include #include #include From 4504f63321e1a581a3c0cbc8de91bd0175d94783 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Thu, 14 Dec 2023 14:15:44 +0100 Subject: [PATCH 219/337] ASoC: simple-card-utils: Drop GPIO include The generic card utilities are including the legacy GPIO header but not using any symbols from it. Drop the include from all files. Signed-off-by: Linus Walleij Link: https://msgid.link/r/20231214-gpio-descriptors-sound-misc-v1-3-e3004176bd8b@linaro.org Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 1 - sound/soc/generic/audio-graph-card2.c | 1 - sound/soc/generic/simple-card-utils.c | 1 - 3 files changed, 3 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 76a9f1e8cdd5..83e3ba773fbd 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -9,7 +9,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index f880a7f73522..9c94677f681a 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -8,7 +8,6 @@ // based on ${LINUX}/sound/soc/generic/audio-graph-card.c #include #include -#include #include #include #include diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 9006ef5e95f5..81077d16d22f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -5,7 +5,6 @@ // Copyright (c) 2016 Kuninori Morimoto #include -#include #include #include #include From 26e91f61d6b91ccfb0bbb15cbc81845dd1d223af Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Thu, 14 Dec 2023 14:15:45 +0100 Subject: [PATCH 220/337] ASoC: tegra: tegra20_ac97: Convert to use GPIO descriptors The Tegra20 AC97 driver is using the legacy GPIO APIs in and to obtain GPIOs for reset and sync. Convert it over and fix the polarity error on the RESET line in the process: this reset line is clearly active low. Just fix the one in-tree device tree site using it at the same time. Signed-off-by: Linus Walleij Link: https://msgid.link/r/20231214-gpio-descriptors-sound-misc-v1-4-e3004176bd8b@linaro.org Signed-off-by: Mark Brown --- arch/arm/boot/dts/nvidia/tegra20-colibri.dtsi | 2 +- sound/soc/tegra/tegra20_ac97.c | 53 +++++++++---------- sound/soc/tegra/tegra20_ac97.h | 4 +- 3 files changed, 28 insertions(+), 31 deletions(-) diff --git a/arch/arm/boot/dts/nvidia/tegra20-colibri.dtsi b/arch/arm/boot/dts/nvidia/tegra20-colibri.dtsi index 16b374e6482f..8c1d5c9fa483 100644 --- a/arch/arm/boot/dts/nvidia/tegra20-colibri.dtsi +++ b/arch/arm/boot/dts/nvidia/tegra20-colibri.dtsi @@ -446,7 +446,7 @@ tegra_ac97: ac97@70002000 { status = "okay"; nvidia,codec-reset-gpio = - <&gpio TEGRA_GPIO(V, 0) GPIO_ACTIVE_HIGH>; + <&gpio TEGRA_GPIO(V, 0) GPIO_ACTIVE_LOW>; nvidia,codec-sync-gpio = <&gpio TEGRA_GPIO(P, 0) GPIO_ACTIVE_HIGH>; }; diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e713feca25fa..8011afe93c96 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -12,12 +12,11 @@ #include #include #include -#include +#include #include #include #include #include -#include #include #include #include @@ -39,11 +38,15 @@ static void tegra20_ac97_codec_reset(struct snd_ac97 *ac97) u32 readback; unsigned long timeout; - /* reset line is not driven by DAC pad group, have to toggle GPIO */ - gpio_set_value(workdata->reset_gpio, 0); + /* + * The reset line is not driven by DAC pad group, have to toggle GPIO. + * The RESET line is active low but this is abstracted by the GPIO + * library. + */ + gpiod_set_value(workdata->reset_gpio, 1); udelay(2); - gpio_set_value(workdata->reset_gpio, 1); + gpiod_set_value(workdata->reset_gpio, 0); udelay(2); timeout = jiffies + msecs_to_jiffies(100); @@ -66,14 +69,10 @@ static void tegra20_ac97_codec_warm_reset(struct snd_ac97 *ac97) * the controller cmd is not working, have to toggle sync line * manually. */ - gpio_request(workdata->sync_gpio, "codec-sync"); - - gpio_direction_output(workdata->sync_gpio, 1); - + gpiod_direction_output(workdata->sync_gpio, 1); udelay(2); - gpio_set_value(workdata->sync_gpio, 0); + gpiod_set_value(workdata->sync_gpio, 0); udelay(2); - gpio_free(workdata->sync_gpio); timeout = jiffies + msecs_to_jiffies(100); @@ -342,28 +341,26 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) goto err_clk_put; } - ac97->reset_gpio = of_get_named_gpio(pdev->dev.of_node, - "nvidia,codec-reset-gpio", 0); - if (gpio_is_valid(ac97->reset_gpio)) { - ret = devm_gpio_request_one(&pdev->dev, ac97->reset_gpio, - GPIOF_OUT_INIT_HIGH, "codec-reset"); - if (ret) { - dev_err(&pdev->dev, "could not get codec-reset GPIO\n"); - goto err_clk_put; - } - } else { - dev_err(&pdev->dev, "no codec-reset GPIO supplied\n"); - ret = -EINVAL; + /* Obtain RESET de-asserted */ + ac97->reset_gpio = devm_gpiod_get(&pdev->dev, + "nvidia,codec-reset", + GPIOD_OUT_LOW); + if (IS_ERR(ac97->reset_gpio)) { + ret = PTR_ERR(ac97->reset_gpio); + dev_err(&pdev->dev, "no RESET GPIO supplied: %d\n", ret); goto err_clk_put; } + gpiod_set_consumer_name(ac97->reset_gpio, "codec-reset"); - ac97->sync_gpio = of_get_named_gpio(pdev->dev.of_node, - "nvidia,codec-sync-gpio", 0); - if (!gpio_is_valid(ac97->sync_gpio)) { - dev_err(&pdev->dev, "no codec-sync GPIO supplied\n"); - ret = -EINVAL; + ac97->sync_gpio = devm_gpiod_get(&pdev->dev, + "nvidia,codec-sync", + GPIOD_OUT_LOW); + if (IS_ERR(ac97->sync_gpio)) { + ret = PTR_ERR(ac97->sync_gpio); + dev_err(&pdev->dev, "no codec-sync GPIO supplied: %d\n", ret); goto err_clk_put; } + gpiod_set_consumer_name(ac97->sync_gpio, "codec-sync"); ac97->capture_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_RX1; ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; diff --git a/sound/soc/tegra/tegra20_ac97.h b/sound/soc/tegra/tegra20_ac97.h index 870ea09ff301..116d7b2db27e 100644 --- a/sound/soc/tegra/tegra20_ac97.h +++ b/sound/soc/tegra/tegra20_ac97.h @@ -80,7 +80,7 @@ struct tegra20_ac97 { struct snd_dmaengine_dai_dma_data playback_dma_data; struct reset_control *reset; struct regmap *regmap; - int reset_gpio; - int sync_gpio; + struct gpio_desc *reset_gpio; + struct gpio_desc *sync_gpio; }; #endif /* __TEGRA20_AC97_H__ */ From 773df207fdd6e17d7a43abf83ea155ade9a95f79 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Tue, 12 Dec 2023 09:08:19 +0100 Subject: [PATCH 221/337] ASoC: dt-bindings: qcom,sm8250: document SM8650 sound card Add sound card for SM8650, which as of now looks fully compatible with SM8450. Signed-off-by: Neil Armstrong Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231212-topic-sm8650-upstream-snd-card-v1-1-fbfc38471204@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index ec641fa2cd4b..ce6b1242b06b 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -24,6 +24,7 @@ properties: - items: - enum: - qcom,sm8550-sndcard + - qcom,sm8650-sndcard - const: qcom,sm8450-sndcard - enum: - qcom,apq8016-sbc-sndcard From 7211094dd065908747a143f9adeff41cfdcf37c0 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Tue, 12 Dec 2023 09:08:20 +0100 Subject: [PATCH 222/337] ASoC: qcom: sc8280xp: Add support for SM8650 Add compatibles for sound card on Qualcomm SM8650 boards. Signed-off-by: Neil Armstrong Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231212-topic-sm8650-upstream-snd-card-v1-2-fbfc38471204@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/sc8280xp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 249a43e1dee3..abef39b5c8ca 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -153,6 +153,7 @@ static const struct of_device_id snd_sc8280xp_dt_match[] = { {.compatible = "qcom,sc8280xp-sndcard", "sc8280xp"}, {.compatible = "qcom,sm8450-sndcard", "sm8450"}, {.compatible = "qcom,sm8550-sndcard", "sm8550"}, + {.compatible = "qcom,sm8650-sndcard", "sm8650"}, {} }; From ddd1ee12a8fb6e4d6f86eddeba64c135eee56623 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Sat, 9 Dec 2023 22:32:19 +0200 Subject: [PATCH 223/337] ASoC: amd: vangogh: Drop conflicting ACPI-based probing The Vangogh machine driver variant based on the MAX98388 amplifier, as found on Valve's Steam Deck OLED, relies on probing via an ACPI match table. This worked fine until commit 197b1f7f0df1 ("ASoC: amd: Add new dmi entries to config entry") enabled SOF support for the target machine (i.e. Galileo product), causing the sound card to enter the deferred probe state indefinitely: $ cat /sys/kernel/debug/devices_deferred AMDI8821:00 acp5x_mach: Register card (acp5x-max98388) failed The issue is related to commit e89f45edb747 ("ASoC: amd: vangogh: Add check for acp config flags in vangogh platform"), which tries to mitigate potential conflicts between SOF and generic ACP Vangogh drivers, due to sharing the PCI device IDs. However, the solution is effective only if the machine driver is directly probed by pci-acp5x through platform_device_register_full(). Hence, remove the conflicting ACPI based probing and rely exclusively on DMI quirks for sound card setup. Fixes: dba22efd0d17 ("ASoC: amd: vangogh: Add support for NAU8821/MAX98388 variant") Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231209203229.878730-2-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/vangogh/acp5x-mach.c | 35 +++++++++++------------------- 1 file changed, 13 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index de4b478a983d..7878e061ecb9 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -439,7 +439,15 @@ static const struct dmi_system_id acp5x_vg_quirk_table[] = { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Valve"), DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Jupiter"), - } + }, + .driver_data = (void *)&acp5x_8821_35l41_card, + }, + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "Valve"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "Galileo"), + }, + .driver_data = (void *)&acp5x_8821_98388_card, }, {} }; @@ -452,25 +460,15 @@ static int acp5x_probe(struct platform_device *pdev) struct snd_soc_card *card; int ret; - card = (struct snd_soc_card *)device_get_match_data(dev); - if (!card) { - /* - * This is normally the result of directly probing the driver - * in pci-acp5x through platform_device_register_full(), which - * is necessary for the CS35L41 variant, as it doesn't support - * ACPI probing and relies on DMI quirks. - */ - dmi_id = dmi_first_match(acp5x_vg_quirk_table); - if (!dmi_id) - return -ENODEV; - - card = &acp5x_8821_35l41_card; - } + dmi_id = dmi_first_match(acp5x_vg_quirk_table); + if (!dmi_id || !dmi_id->driver_data) + return -ENODEV; machine = devm_kzalloc(dev, sizeof(*machine), GFP_KERNEL); if (!machine) return -ENOMEM; + card = dmi_id->driver_data; card->dev = dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); @@ -482,17 +480,10 @@ static int acp5x_probe(struct platform_device *pdev) return 0; } -static const struct acpi_device_id acp5x_acpi_match[] = { - { "AMDI8821", (kernel_ulong_t)&acp5x_8821_98388_card }, - {}, -}; -MODULE_DEVICE_TABLE(acpi, acp5x_acpi_match); - static struct platform_driver acp5x_mach_driver = { .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, - .acpi_match_table = acp5x_acpi_match, }, .probe = acp5x_probe, }; From 2cef11ec3dfd5f14d8ddef917682408ed01e5805 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Sat, 9 Dec 2023 22:32:20 +0200 Subject: [PATCH 224/337] ASoC: amd: vangogh: Allow probing ACP PCI when SOF is disabled Since commit e89f45edb747 ("ASoC: amd: vangogh: Add check for acp config flags in vangogh platform"), the Vangogh ACP PCI driver could not be used anymore for boards which happen to have a matching entry in acp-config list. Commit f18818eb0dbe ("ASoC: amd: vangogh: Add condition check for acp config flag") slightly changed the behaviour to permit loading the driver if AMD_LEGACY flag is set. However, for AMD_SOF flag the probing is still denied, even if SOF support is disabled in kernel configuration. While this helps preventing conflicts between SOF and generic ACP drivers, there are cases where a fallback to the generic non-SOF support would still be needed or useful, e.g. SOF firmware is not available or doesn't work properly, SOF driver is broken or doesn't provide full support for a particular hardware, or simply for testing/debugging the alternative solution. A real-life example is Steam Deck OLED, which works with both drivers. Prevent returning from probe() when ACP config indicates SOF support for the current board *and* the Vangogh SOF driver is not enabled in kernel configuration. Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231209203229.878730-3-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/vangogh/pci-acp5x.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index 3826443d77b9..10755c07949c 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -130,9 +130,13 @@ static int snd_acp5x_probe(struct pci_dev *pci, int ret, i; u32 addr, val; - /* Return if acp config flag is defined */ + /* + * Return if ACP config flag is defined, except when board + * supports SOF while it is not being enabled in kernel config. + */ flag = snd_amd_acp_find_config(pci); - if (flag != FLAG_AMD_LEGACY) + if (flag != FLAG_AMD_LEGACY && + (flag != FLAG_AMD_SOF || IS_ENABLED(CONFIG_SND_SOC_SOF_AMD_VANGOGH))) return -ENODEV; irqflags = IRQF_SHARED; From 78d3924675d4e076faa5600b48b8565fcb135ee0 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Sat, 9 Dec 2023 22:32:21 +0200 Subject: [PATCH 225/337] ASoC: amd: vangogh: Switch to {RUNTIME,SYSTEM_SLEEP}_PM_OPS Replace the old SET_{RUNTIME,SYSTEM_SLEEP}_PM_OPS() helpers with their modern alternatives and drop the now unnecessary __maybe_unused qualifier in the suspend and resume functions. Additionally, make use of pm_ptr() to ensure the PM ops are dropped when building with CONFIG_PM disabled. Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231209203229.878730-4-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/vangogh/pci-acp5x.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index 10755c07949c..af56ff09f02a 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -264,7 +264,7 @@ disable_pci: return ret; } -static int __maybe_unused snd_acp5x_suspend(struct device *dev) +static int snd_acp5x_suspend(struct device *dev) { int ret; struct acp5x_dev_data *adata; @@ -279,7 +279,7 @@ static int __maybe_unused snd_acp5x_suspend(struct device *dev) return ret; } -static int __maybe_unused snd_acp5x_resume(struct device *dev) +static int snd_acp5x_resume(struct device *dev) { int ret; struct acp5x_dev_data *adata; @@ -294,9 +294,8 @@ static int __maybe_unused snd_acp5x_resume(struct device *dev) } static const struct dev_pm_ops acp5x_pm = { - SET_RUNTIME_PM_OPS(snd_acp5x_suspend, - snd_acp5x_resume, NULL) - SET_SYSTEM_SLEEP_PM_OPS(snd_acp5x_suspend, snd_acp5x_resume) + RUNTIME_PM_OPS(snd_acp5x_suspend, snd_acp5x_resume, NULL) + SYSTEM_SLEEP_PM_OPS(snd_acp5x_suspend, snd_acp5x_resume) }; static void snd_acp5x_remove(struct pci_dev *pci) @@ -332,7 +331,7 @@ static struct pci_driver acp5x_driver = { .probe = snd_acp5x_probe, .remove = snd_acp5x_remove, .driver = { - .pm = &acp5x_pm, + .pm = pm_ptr(&acp5x_pm), } }; From 6e202e758b4b8d85dfb909c8eb710db8c6160303 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Sat, 9 Dec 2023 22:32:22 +0200 Subject: [PATCH 226/337] ASoC: amd: acp-config: Add missing MODULE_DESCRIPTION Add the missing MODULE_DESCRIPTION() to avoid the following warning when building with W=1: WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/snd-acp-config.o Fixes: f1bdd8d385a8 ("ASoC: amd: Add module to determine ACP configuration") Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231209203229.878730-5-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 067b1fdfbc9d..65420ccc7623 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -307,4 +307,5 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sof_machines[] = { }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp63_sof_machines); +MODULE_DESCRIPTION("AMD ACP Machine Configuration Module"); MODULE_LICENSE("Dual BSD/GPL"); From 576f3aef47f42f368db08fd5e3f49880c4493bf5 Mon Sep 17 00:00:00 2001 From: Cristian Ciocaltea Date: Sat, 9 Dec 2023 22:32:23 +0200 Subject: [PATCH 227/337] ASoC: amd: acp: Add missing MODULE_DESCRIPTION in mach-common Add a MODULE_DESCRIPTION() in the generic ACP machine driver to avoid the following warning when building with W=1: WARNING: modpost: missing MODULE_DESCRIPTION() in sound/soc/amd/acp/snd-acp-mach.o Fixes: d4c750f2c7d4 ("ASoC: amd: acp: Add generic machine driver support for ACP cards") Signed-off-by: Cristian Ciocaltea Reviewed-by: Emil Velikov Link: https://msgid.link/r/20231209203229.878730-6-cristian.ciocaltea@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-mach-common.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index f7bcf210f0fd..c90ec3419247 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1773,4 +1773,5 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) } EXPORT_SYMBOL_NS_GPL(acp_legacy_dai_links_create, SND_SOC_AMD_MACH); +MODULE_DESCRIPTION("AMD ACP Common Machine driver"); MODULE_LICENSE("GPL v2"); From ea244b35a4da60a92d0e3be528f82ebcbcf10753 Mon Sep 17 00:00:00 2001 From: Rui Zhou Date: Tue, 12 Dec 2023 20:30:47 +0800 Subject: [PATCH 228/337] ASoC: dt-bindings: mt8188-mt6359: add es8326 support Add compatible string "mediatek,mt8188-es8326" to support new board with es8326 codec. Acked-by: Conor Dooley Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Rui Zhou Link: https://msgid.link/r/20231212123050.4080083-2-zhourui@huaqin.corp-partner.google.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml index 4c8c95057ef7..f94ad0715e32 100644 --- a/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8188-mt6359.yaml @@ -15,6 +15,7 @@ allOf: properties: compatible: enum: + - mediatek,mt8188-es8326 - mediatek,mt8188-mt6359-evb - mediatek,mt8188-nau8825 - mediatek,mt8188-rt5682s From 1a268000b03a162bd5feb7fce1c130f1b31602b5 Mon Sep 17 00:00:00 2001 From: Rui Zhou Date: Tue, 12 Dec 2023 20:30:48 +0800 Subject: [PATCH 229/337] ASoC: mediatek: mt8188-mt6359: commonize headset codec init/exit api Reduce code duplication, unify the headset codec init/exit api. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Rui Zhou Link: https://msgid.link/r/20231212123050.4080083-3-zhourui@huaqin.corp-partner.google.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 67 ++--------------------- 1 file changed, 6 insertions(+), 61 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index 33d477cc2e54..b4606a28794c 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -726,7 +726,7 @@ static int mt8188_max98390_codec_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int mt8188_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) +static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); @@ -775,68 +775,13 @@ static int mt8188_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) return 0; }; -static int mt8188_rt5682s_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(card); - struct mt8188_mt6359_priv *priv = soc_card_data->mach_priv; - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_jack *jack = &priv->headset_jack; - int ret; - - ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_nau8825_widgets, - ARRAY_SIZE(mt8188_nau8825_widgets)); - if (ret) { - dev_err(rtd->dev, "unable to add rt5682s card widget, ret %d\n", ret); - return ret; - } - - ret = snd_soc_add_card_controls(card, mt8188_nau8825_controls, - ARRAY_SIZE(mt8188_nau8825_controls)); - if (ret) { - dev_err(rtd->dev, "unable to add rt5682s card controls, ret %d\n", ret); - return ret; - } - - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | - SND_JACK_BTN_1 | SND_JACK_BTN_2 | - SND_JACK_BTN_3, - jack, - nau8825_jack_pins, - ARRAY_SIZE(nau8825_jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); - return ret; - } - - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); - - if (ret) { - dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); - return ret; - } - - return 0; -}; - -static void mt8188_nau8825_codec_exit(struct snd_soc_pcm_runtime *rtd) +static void mt8188_headset_codec_exit(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; snd_soc_component_set_jack(component, NULL, NULL); } -static void mt8188_rt5682s_codec_exit(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - - snd_soc_component_set_jack(component, NULL, NULL); -} static int mt8188_nau8825_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -1407,15 +1352,15 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev) } else if (!strcmp(dai_link->codecs->dai_name, NAU8825_CODEC_DAI)) { dai_link->ops = &mt8188_nau8825_ops; if (!init_nau8825) { - dai_link->init = mt8188_nau8825_codec_init; - dai_link->exit = mt8188_nau8825_codec_exit; + dai_link->init = mt8188_headset_codec_init; + dai_link->exit = mt8188_headset_codec_exit; init_nau8825 = true; } } else if (!strcmp(dai_link->codecs->dai_name, RT5682S_CODEC_DAI)) { dai_link->ops = &mt8188_rt5682s_i2s_ops; if (!init_rt5682s) { - dai_link->init = mt8188_rt5682s_codec_init; - dai_link->exit = mt8188_rt5682s_codec_exit; + dai_link->init = mt8188_headset_codec_init; + dai_link->exit = mt8188_headset_codec_exit; init_rt5682s = true; } } else { From e794a894427b1d64f2ebff24f003c60373d68c2c Mon Sep 17 00:00:00 2001 From: Rui Zhou Date: Tue, 12 Dec 2023 20:30:49 +0800 Subject: [PATCH 230/337] ASoC: mediatek: mt8188-mt6359: add es8326 support To use ES8326 as the codec, add a new sound card named mt8186_es8326. Reviewed-by: Trevor Wu Signed-off-by: Rui Zhou Reviewed-by: AngeloGioacchino Del Regno Link: https://msgid.link/r/20231212123050.4080083-4-zhourui@huaqin.corp-partner.google.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 56 ++++++++++++++++++++++- 2 files changed, 55 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index b93d455744ab..296b434caf81 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -252,6 +252,7 @@ config SND_SOC_MT8188_MT6359 select SND_SOC_NAU8315 select SND_SOC_NAU8825 select SND_SOC_RT5682S + select SND_SOC_ES8326 help This adds support for ASoC machine driver for MediaTek MT8188 boards with the MT6359 and other I2S audio codecs. diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index b4606a28794c..d1884f23a1a7 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -34,6 +34,8 @@ #define NAU8825_HS_PRESENT BIT(0) #define RT5682S_HS_PRESENT BIT(1) +#define ES8326_HS_PRESENT BIT(2) +#define MAX98390_TWO_AMP BIT(3) /* * Maxim MAX98390 */ @@ -48,6 +50,11 @@ */ #define NAU8825_CODEC_DAI "nau8825-hifi" +/* + * ES8326 + */ +#define ES8326_CODEC_DAI "ES8326 HiFi" + #define SOF_DMA_DL2 "SOF_DMA_DL2" #define SOF_DMA_DL3 "SOF_DMA_DL3" #define SOF_DMA_UL4 "SOF_DMA_UL4" @@ -888,6 +895,30 @@ static const struct snd_soc_ops mt8188_sof_be_ops = { .hw_params = mt8188_sof_be_hw_params, }; +static int mt8188_es8326_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); + unsigned int rate = params_rate(params); + int ret; + + /* Configure MCLK for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, rate * 256, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set MCLK %d\n", ret); + return ret; + } + + /* Configure MCLK for cpu */ + return snd_soc_dai_set_sysclk(cpu_dai, 0, rate * 256, SND_SOC_CLOCK_OUT); +} + +static const struct snd_soc_ops mt8188_es8326_ops = { + .hw_params = mt8188_es8326_hw_params, +}; + static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { /* FE */ [DAI_LINK_DL2_FE] = { @@ -1197,7 +1228,7 @@ static void mt8188_fixup_controls(struct snd_soc_card *card) struct mt8188_card_data *card_data = (struct mt8188_card_data *)priv->private_data; struct snd_kcontrol *kctl; - if (card_data->quirk & (NAU8825_HS_PRESENT | RT5682S_HS_PRESENT)) { + if (card_data->quirk & (NAU8825_HS_PRESENT | RT5682S_HS_PRESENT | ES8326_HS_PRESENT)) { struct snd_soc_dapm_widget *w, *next_w; for_each_card_widgets_safe(card, w, next_w) { @@ -1238,6 +1269,7 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev) struct mt8188_card_data *card_data; struct snd_soc_dai_link *dai_link; bool init_mt6359 = false; + bool init_es8326 = false; bool init_nau8825 = false; bool init_rt5682s = false; bool init_max98390 = false; @@ -1344,7 +1376,14 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev) strcmp(dai_link->name, "ETDM1_IN_BE") == 0 || strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { if (!strcmp(dai_link->codecs->dai_name, MAX98390_CODEC_DAI)) { - dai_link->ops = &mt8188_max98390_ops; + /* + * The TDM protocol settings with fixed 4 slots are defined in + * mt8188_max98390_ops. Two amps is I2S mode, + * SOC and codec don't require TDM settings. + */ + if (!(card_data->quirk & MAX98390_TWO_AMP)) { + dai_link->ops = &mt8188_max98390_ops; + } if (!init_max98390) { dai_link->init = mt8188_max98390_codec_init; init_max98390 = true; @@ -1363,6 +1402,13 @@ static int mt8188_mt6359_dev_probe(struct platform_device *pdev) dai_link->exit = mt8188_headset_codec_exit; init_rt5682s = true; } + } else if (!strcmp(dai_link->codecs->dai_name, ES8326_CODEC_DAI)) { + dai_link->ops = &mt8188_es8326_ops; + if (!init_es8326) { + dai_link->init = mt8188_headset_codec_init; + dai_link->exit = mt8188_headset_codec_exit; + init_es8326 = true; + } } else { if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) { if (!init_dumb) { @@ -1405,10 +1451,16 @@ static struct mt8188_card_data mt8188_rt5682s_card = { .quirk = RT5682S_HS_PRESENT, }; +static struct mt8188_card_data mt8188_es8326_card = { + .name = "mt8188_es8326", + .quirk = ES8326_HS_PRESENT | MAX98390_TWO_AMP, +}; + static const struct of_device_id mt8188_mt6359_dt_match[] = { { .compatible = "mediatek,mt8188-mt6359-evb", .data = &mt8188_evb_card, }, { .compatible = "mediatek,mt8188-nau8825", .data = &mt8188_nau8825_card, }, { .compatible = "mediatek,mt8188-rt5682s", .data = &mt8188_rt5682s_card, }, + { .compatible = "mediatek,mt8188-es8326", .data = &mt8188_es8326_card, }, { /* sentinel */ }, }; MODULE_DEVICE_TABLE(of, mt8188_mt6359_dt_match); From 3423c3db22e9213acd279ff3800bb1e91aa2ac89 Mon Sep 17 00:00:00 2001 From: Rui Zhou Date: Tue, 12 Dec 2023 20:30:50 +0800 Subject: [PATCH 231/337] ASoC: mediatek: mt8188-mt6359: Enable dual amp for mt8188-rt5682s Enable support for dual MAX98390 amplifiers on the mt8188-rt5682s board. Reviewed-by: Trevor Wu Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Rui Zhou Link: https://msgid.link/r/20231212123050.4080083-5-zhourui@huaqin.corp-partner.google.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index d1884f23a1a7..a391066ab204 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -1448,7 +1448,7 @@ static struct mt8188_card_data mt8188_nau8825_card = { static struct mt8188_card_data mt8188_rt5682s_card = { .name = "mt8188_rt5682s", - .quirk = RT5682S_HS_PRESENT, + .quirk = RT5682S_HS_PRESENT | MAX98390_TWO_AMP, }; static struct mt8188_card_data mt8188_es8326_card = { From 6b9dc2da66578acff36401ba87fee93c2abf2a6e Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 4 Dec 2023 11:01:15 +0100 Subject: [PATCH 232/337] ASoC: qcom: Add x1e80100 sound machine driver Add sound machine driver for the soundcards on Qualcomm X1E80100 SoC, supporting up to four channel audio playback over Soundwire bus. The driver is based on existing sc8280xp.c driver. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231204100116.211898-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/Kconfig | 12 +++ sound/soc/qcom/Makefile | 2 + sound/soc/qcom/x1e80100.c | 168 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 182 insertions(+) create mode 100644 sound/soc/qcom/x1e80100.c diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index e7b00d1d9e99..762491d6f2f2 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -221,4 +221,16 @@ config SND_SOC_SC7280 SC7280 SoC-based systems. Say Y or M if you want to use audio device on this SoCs. +config SND_SOC_X1E80100 + tristate "SoC Machine driver for X1E80100 boards" + depends on QCOM_APR && SOUNDWIRE + depends on COMMON_CLK + select SND_SOC_QDSP6 + select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW + help + Add support for audio on Qualcomm Technologies Inc. + X1E80100 SoC-based systems. + Say Y or M if you want to use audio device on this SoCs. + endif #SND_SOC_QCOM diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 254350d9dc06..34f3fcb8ee9a 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -29,6 +29,7 @@ snd-soc-sm8250-objs := sm8250.o snd-soc-sc8280xp-objs := sc8280xp.o snd-soc-qcom-common-objs := common.o snd-soc-qcom-sdw-objs := sdw.o +snd-soc-x1e80100-objs := x1e80100.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o obj-$(CONFIG_SND_SOC_SM8250) += snd-soc-sm8250.o obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o obj-$(CONFIG_SND_SOC_QCOM_SDW) += snd-soc-qcom-sdw.o +obj-$(CONFIG_SND_SOC_X1E80100) += snd-soc-x1e80100.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c new file mode 100644 index 000000000000..c3c8bf7ffb5b --- /dev/null +++ b/sound/soc/qcom/x1e80100.c @@ -0,0 +1,168 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2023, Linaro Limited + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "common.h" +#include "qdsp6/q6afe.h" +#include "sdw.h" + +struct x1e80100_snd_data { + bool stream_prepared[AFE_PORT_MAX]; + struct snd_soc_card *card; + struct sdw_stream_runtime *sruntime[AFE_PORT_MAX]; + struct snd_soc_jack jack; + bool jack_setup; +}; + +static int x1e80100_snd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + + return qcom_snd_wcd_jack_setup(rtd, &data->jack, &data->jack_setup); +} + +static void x1e80100_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + + data->sruntime[cpu_dai->id] = NULL; + sdw_release_stream(sruntime); +} + +static int x1e80100_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = 48000; + switch (cpu_dai->id) { + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + channels->min = 1; + break; + default: + break; + } + + return 0; +} + +static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + + return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); +} + +static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + + return qcom_snd_sdw_prepare(substream, sruntime, + &data->stream_prepared[cpu_dai->id]); +} + +static int x1e80100_snd_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; + + return qcom_snd_sdw_hw_free(substream, sruntime, + &data->stream_prepared[cpu_dai->id]); +} + +static const struct snd_soc_ops x1e80100_be_ops = { + .startup = qcom_snd_sdw_startup, + .shutdown = x1e80100_snd_shutdown, + .hw_params = x1e80100_snd_hw_params, + .hw_free = x1e80100_snd_hw_free, + .prepare = x1e80100_snd_prepare, +}; + +static void x1e80100_add_be_ops(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int i; + + for_each_card_prelinks(card, i, link) { + if (link->no_pcm == 1) { + link->init = x1e80100_snd_init; + link->be_hw_params_fixup = x1e80100_be_hw_params_fixup; + link->ops = &x1e80100_be_ops; + } + } +} + +static int x1e80100_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct x1e80100_snd_data *data; + struct device *dev = &pdev->dev; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + /* Allocate the private data */ + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + card->owner = THIS_MODULE; + card->dev = dev; + dev_set_drvdata(dev, card); + snd_soc_card_set_drvdata(card, data); + + ret = qcom_snd_parse_of(card); + if (ret) + return ret; + + card->driver_name = "x1e80100"; + x1e80100_add_be_ops(card); + + return devm_snd_soc_register_card(dev, card); +} + +static const struct of_device_id snd_x1e80100_dt_match[] = { + { .compatible = "qcom,x1e80100-sndcard", }, + {} +}; +MODULE_DEVICE_TABLE(of, snd_x1e80100_dt_match); + +static struct platform_driver snd_x1e80100_driver = { + .probe = x1e80100_platform_probe, + .driver = { + .name = "snd-x1e80100", + .of_match_table = snd_x1e80100_dt_match, + }, +}; +module_platform_driver(snd_x1e80100_driver); +MODULE_AUTHOR("Srinivas Kandagatla "); +MODULE_DESCRIPTION("Qualcomm X1E80100 ASoC Machine Driver"); +MODULE_LICENSE("GPL"); From 337d93b4285a92280edd7d0a910c3b7cbc70d717 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 4 Dec 2023 11:01:16 +0100 Subject: [PATCH 233/337] ASoC: dt-bindings: qcom,sm8250: Add X1E80100 sound card Document bindings for the Qualcomm X1E80100 SoC sound card. The bindings are the same as for other newer Qualcomm ADSP sound cards, thus keep them in existing qcom,sm8250.yaml file, even though Linux driver is separate. Signed-off-by: Krzysztof Kozlowski Acked-by: Conor Dooley Link: https://msgid.link/r/20231204100116.211898-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index ec641fa2cd4b..4673fdffe312 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -34,6 +34,7 @@ properties: - qcom,sdm845-sndcard - qcom,sm8250-sndcard - qcom,sm8450-sndcard + - qcom,x1e80100-sndcard audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array From 8f039360897bdd2f1f455b46a7f504b677405913 Mon Sep 17 00:00:00 2001 From: Chancel Liu Date: Mon, 4 Dec 2023 19:15:32 +0800 Subject: [PATCH 234/337] ASoC: soc-pcm.c: Complete the active count for components without DAIs Some components like platforms don't have DAIs. If the active count of these components is ignored pinctrl may be wrongly selected between default and sleep state. So need to increment or decrement the active count for components without DAIs to avoid it. Signed-off-by: Chancel Liu Link: https://msgid.link/r/20231204111532.3165-1-chancel.liu@nxp.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c20573aeb756..16a4644eff3a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -292,6 +292,7 @@ static void dpcm_set_be_update_state(struct snd_soc_pcm_runtime *be, void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { + struct snd_soc_component *component; struct snd_soc_dai *dai; int i; @@ -299,6 +300,13 @@ void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, for_each_rtd_dais(rtd, i, dai) snd_soc_dai_action(dai, stream, action); + + /* Increments/Decrements the active count for components without DAIs */ + for_each_rtd_components(rtd, i, component) { + if (component->num_dai) + continue; + component->active += action; + } } EXPORT_SYMBOL_GPL(snd_soc_runtime_action); From bb3392453d3ba44e60b85381e3bfa3c551a44e5d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 4 Dec 2023 11:00:48 +0100 Subject: [PATCH 235/337] ASoC: qcom: Fix trivial code style issues Fix few trivial code style issues, pointed out by checkpatch, so they do not get copied to new code (when old code is used as template): WARNING: Prefer "GPL" over "GPL v2" - see commit bf7fbeeae6db ("module: Cure the MODULE_LICENSE "GPL" vs. "GPL v2" bogosity") WARNING: function definition argument 'struct platform_device *' should also have an identifier name ERROR: code indent should use tabs where possible WARNING: please, no spaces at the start of a line WARNING: Missing a blank line after declarations WARNING: unnecessary whitespace before a quoted newline Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231204100048.211800-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/apq8096.c | 2 +- sound/soc/qcom/common.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/lpass-hdmi.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass.h | 2 +- sound/soc/qcom/qdsp6/q6afe.c | 8 ++++---- sound/soc/qcom/qdsp6/q6apm-dai.c | 4 ++-- sound/soc/qcom/qdsp6/q6asm.h | 20 ++++++++++---------- sound/soc/qcom/qdsp6/topology.c | 3 ++- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sc8280xp.c | 2 +- sound/soc/qcom/sdm845.c | 2 +- sound/soc/qcom/sdw.c | 2 +- sound/soc/qcom/sm8250.c | 2 +- sound/soc/qcom/storm.c | 2 +- 20 files changed, 34 insertions(+), 33 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index efbdbb4dd753..4834a56eaa88 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -344,4 +344,4 @@ module_platform_driver(apq8016_sbc_platform_driver); MODULE_AUTHOR("Srinivas Kandagatla state = Q6APM_STREAM_STOPPED; break; case APM_CLIENT_EVENT_DATA_WRITE_DONE: - spin_lock_irqsave(&prtd->lock, flags); + spin_lock_irqsave(&prtd->lock, flags); prtd->pos += prtd->pcm_count; spin_unlock_irqrestore(&prtd->lock, flags); snd_pcm_period_elapsed(substream); @@ -143,7 +143,7 @@ static void event_handler(uint32_t opcode, uint32_t token, uint32_t *payload, vo break; case APM_CLIENT_EVENT_DATA_READ_DONE: - spin_lock_irqsave(&prtd->lock, flags); + spin_lock_irqsave(&prtd->lock, flags); prtd->pos += prtd->pcm_count; spin_unlock_irqrestore(&prtd->lock, flags); snd_pcm_period_elapsed(substream); diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 0103d8dae5da..519e1b3a3f7c 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -35,16 +35,16 @@ enum { #define ASM_LAST_BUFFER_FLAG BIT(30) struct q6asm_flac_cfg { - u32 sample_rate; - u32 ext_sample_rate; - u32 min_frame_size; - u32 max_frame_size; - u16 stream_info_present; - u16 min_blk_size; - u16 max_blk_size; - u16 ch_cfg; - u16 sample_size; - u16 md5_sum; + u32 sample_rate; + u32 ext_sample_rate; + u32 min_frame_size; + u32 max_frame_size; + u16 stream_info_present; + u16 min_blk_size; + u16 max_blk_size; + u16 ch_cfg; + u16 sample_size; + u16 md5_sum; }; struct q6asm_wma_cfg { diff --git a/sound/soc/qcom/qdsp6/topology.c b/sound/soc/qcom/qdsp6/topology.c index 130b22a34fb3..70572c83e101 100644 --- a/sound/soc/qcom/qdsp6/topology.c +++ b/sound/soc/qcom/qdsp6/topology.c @@ -545,6 +545,7 @@ static struct audioreach_module *audioreach_parse_common_tokens(struct q6apm *ap if (mod) { int pn, id = 0; + mod->module_id = module_id; mod->max_ip_port = max_ip_port; mod->max_op_port = max_op_port; @@ -1271,7 +1272,7 @@ int audioreach_tplg_init(struct snd_soc_component *component) ret = request_firmware(&fw, tplg_fw_name, dev); if (ret < 0) { - dev_err(dev, "tplg firmware loading %s failed %d \n", tplg_fw_name, ret); + dev_err(dev, "tplg firmware loading %s failed %d\n", tplg_fw_name, ret); goto err; } diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index b0320a74d508..a15f385ede45 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -578,4 +578,4 @@ static struct platform_driver sc7180_snd_driver = { module_platform_driver(sc7180_snd_driver); MODULE_DESCRIPTION("sc7180 ASoC Machine Driver"); -MODULE_LICENSE("GPL v2"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index 249a43e1dee3..ebbbb8e6b68c 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -168,4 +168,4 @@ static struct platform_driver snd_sc8280xp_driver = { module_platform_driver(snd_sc8280xp_driver); MODULE_AUTHOR("Srinivas Kandagatla Date: Mon, 11 Dec 2023 13:31:01 +0100 Subject: [PATCH 236/337] ASoC: dt-bindings: qcom,lpass-rx-macro: Add X1E80100 LPASS RX Add bindings for Qualcomm X1E80100 SoC Low Power Audio SubSystem (LPASS) RX macro codec, which looks like compatible with earlier SM8550. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231211123104.72963-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,lpass-rx-macro.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml index cbc36646100f..b8540b30741e 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-rx-macro.yaml @@ -19,7 +19,9 @@ properties: - qcom,sm8550-lpass-rx-macro - qcom,sc8280xp-lpass-rx-macro - items: - - const: qcom,sm8650-lpass-rx-macro + - enum: + - qcom,sm8650-lpass-rx-macro + - qcom,x1e80100-lpass-rx-macro - const: qcom,sm8550-lpass-rx-macro reg: From 7de2109ce1619951e957eaafab83545a4cab8609 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 11 Dec 2023 13:31:02 +0100 Subject: [PATCH 237/337] ASoC: dt-bindings: qcom,lpass-rx-macro: Add X1E80100 LPASS TX Add bindings for Qualcomm X1E80100 SoC Low Power Audio SubSystem (LPASS) TX macro codec, which looks like compatible with earlier SM8550. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231211123104.72963-2-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,lpass-tx-macro.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml index cee79ac42a33..3e2ae16c6aba 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml @@ -20,7 +20,9 @@ properties: - qcom,sm8550-lpass-tx-macro - qcom,sc8280xp-lpass-tx-macro - items: - - const: qcom,sm8650-lpass-tx-macro + - enum: + - qcom,sm8650-lpass-tx-macro + - qcom,x1e80100-lpass-tx-macro - const: qcom,sm8550-lpass-tx-macro reg: From f990306adf2715cc2bdb85470065f7326a42081b Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 11 Dec 2023 13:31:03 +0100 Subject: [PATCH 238/337] ASoC: dt-bindings: qcom,lpass-rx-macro: Add X1E80100 LPASS VA Add bindings for Qualcomm X1E80100 SoC Low Power Audio SubSystem (LPASS) VA macro codec, which looks like compatible with earlier SM8550. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231211123104.72963-3-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,lpass-va-macro.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml index ca6b07d5826d..c03ff9472a85 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -19,7 +19,9 @@ properties: - qcom,sm8550-lpass-va-macro - qcom,sc8280xp-lpass-va-macro - items: - - const: qcom,sm8650-lpass-va-macro + - enum: + - qcom,sm8650-lpass-va-macro + - qcom,x1e80100-lpass-va-macro - const: qcom,sm8550-lpass-va-macro reg: From 173a3b20a4980265bab52dbc60b616e739664b0d Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Mon, 11 Dec 2023 13:31:04 +0100 Subject: [PATCH 239/337] ASoC: dt-bindings: qcom,lpass-rx-macro: Add X1E80100 LPASS WSA Add bindings for Qualcomm X1E80100 SoC Low Power Audio SubSystem (LPASS) WSA macro codec, which looks like compatible with earlier SM8550. Signed-off-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231211123104.72963-4-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 5fb39d35c8ec..06b5f7be3608 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -19,7 +19,9 @@ properties: - qcom,sm8550-lpass-wsa-macro - qcom,sc8280xp-lpass-wsa-macro - items: - - const: qcom,sm8650-lpass-wsa-macro + - enum: + - qcom,sm8650-lpass-wsa-macro + - qcom,x1e80100-lpass-wsa-macro - const: qcom,sm8550-lpass-wsa-macro reg: From ee00330a5b78e2acf4b3aac32913da43e2c12a26 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Thu, 14 Dec 2023 01:25:39 +0100 Subject: [PATCH 240/337] ASoC: tas2781: add support for FW version 0x0503 Layout of FW version 0x0503 is compatible with 0x0502. Already supported by TI's tas2781-linux-driver tree. https://git.ti.com/cgit/tas2781-linux-drivers/tas2781-linux-driver/ Fixes: 915f5eadebd2 ("ASoC: tas2781: firmware lib") Signed-off-by: Gergo Koteles Link: https://msgid.link/r/98d4ee4e01e834af72a1a0bea6736facf43582e0.1702513517.git.soyer@irl.hu Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 4efe95b60aaa..9b01e6d4ae85 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1982,6 +1982,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, case 0x301: case 0x302: case 0x502: + case 0x503: tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = From 8b69dba103650b8247a336945c5fedc64ab5ddea Mon Sep 17 00:00:00 2001 From: Himanshu Bhavani Date: Mon, 18 Dec 2023 20:02:08 +0530 Subject: [PATCH 241/337] ASoC: amd: acp: Remove redundant ret variable Removed Unneeded variable: "ret" Signed-off-by: Himanshu Bhavani Link: https://msgid.link/r/20231218143214.939885-1-himanshu.bhavani@siliconsignals.io Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index 6cd3352dc38d..f85b85ea4be9 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -222,7 +222,6 @@ static int acp3x_es83xx_resume_post(struct snd_soc_card *card) static int acp3x_es83xx_configure_gpios(struct acp3x_es83xx_private *priv) { - int ret = 0; priv->enable_spk_gpio.crs_entry_index = 0; priv->enable_hp_gpio.crs_entry_index = 1; @@ -245,7 +244,7 @@ static int acp3x_es83xx_configure_gpios(struct acp3x_es83xx_private *priv) priv->enable_spk_gpio.active_low ? "low" : "high", priv->enable_hp_gpio.crs_entry_index, priv->enable_hp_gpio.active_low ? "low" : "high"); - return ret; + return 0; } static int acp3x_es83xx_configure_mics(struct acp3x_es83xx_private *priv) From 7465582e0b18859d3681192ec2ccf22a81370040 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Dec 2023 05:09:53 +0000 Subject: [PATCH 242/337] ASoC: fsl: fsl-asoc-card: don't need DUMMY Platform We can use SND_SOC_DAILINK_REG() with 2 parameter. DUMMY Platform is not needed. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/877cla93ry.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl-asoc-card.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 7518ab9d768e..bc07f26ba303 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -305,8 +305,7 @@ SND_SOC_DAILINK_DEFS(hifi_fe, SND_SOC_DAILINK_DEFS(hifi_be, DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_DUMMY())); + DAILINK_COMP_ARRAY(COMP_EMPTY())); static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ From 56558d6ab8c09c416bdb6d72b7e02894539a882a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Dec 2023 05:10:02 +0000 Subject: [PATCH 243/337] ASoC: samsung: odroid: don't need DUMMY Platform We can use SND_SOC_DAILINK_REG() with 2 parameter. DUMMY Platform is not needed. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/875y0u93rq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/samsung/odroid.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index e95f3d3f0401..110ae14dd7ea 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -157,8 +157,7 @@ SND_SOC_DAILINK_DEFS(primary, SND_SOC_DAILINK_DEFS(mixer, DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_DUMMY())); + DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(secondary, DAILINK_COMP_ARRAY(COMP_EMPTY()), From c2dfe29f30d8850af324449f416491b171af19aa Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Dec 2023 05:10:09 +0000 Subject: [PATCH 244/337] ASoC: intel: hdaudio.c: use snd_soc_dummy_dlc We already have snd_soc_dummy_dlc. Let's use it. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/874jge93ri.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/boards/hdaudio.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c index 844a918f9a81..79b4aca41333 100644 --- a/sound/soc/intel/avs/boards/hdaudio.c +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -155,8 +155,6 @@ static int avs_probing_link_init(struct snd_soc_pcm_runtime *rtm) return 0; } -SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); - static struct snd_soc_dai_link probing_link = { .name = "probing-LINK", .id = -1, @@ -164,8 +162,8 @@ static struct snd_soc_dai_link probing_link = { .no_pcm = 1, .dpcm_playback = 1, .dpcm_capture = 1, - .cpus = dummy, - .num_cpus = ARRAY_SIZE(dummy), + .cpus = &snd_soc_dummy_dlc, + .num_cpus = 1, .init = avs_probing_link_init, }; From e8776ff9ce9f5a8a9d8294101fd2924cebdd2da1 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Dec 2023 05:10:19 +0000 Subject: [PATCH 245/337] ASoC: sof: use snd_soc_dummy_dlc We already have snd_soc_dummy_dlc. Let's use it. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/8734vy93r8.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-client-probes.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 7cc9e8f18de7..30f771ac7bbf 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -381,8 +381,6 @@ static const struct snd_soc_component_driver sof_probes_component = { .legacy_dai_naming = 1, }; -SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); - static int sof_probes_client_probe(struct auxiliary_device *auxdev, const struct auxiliary_device_id *id) { @@ -475,7 +473,7 @@ static int sof_probes_client_probe(struct auxiliary_device *auxdev, links[0].cpus = &cpus[0]; links[0].num_cpus = 1; links[0].cpus->dai_name = "Probe Extraction CPU DAI"; - links[0].codecs = dummy; + links[0].codecs = &snd_soc_dummy_dlc; links[0].num_codecs = 1; links[0].platforms = platform_component; links[0].num_platforms = ARRAY_SIZE(platform_component); From 13f58267cda3d6946c8f4de368ad5d4a003baa61 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 19 Dec 2023 05:10:33 +0000 Subject: [PATCH 246/337] ASoC: soc.h: don't create dummy Component via COMP_DUMMY() Many ASoC drivers define CPU/Codec/Platform dai_link by below macro. SND_SOC_DAILINK_DEFS(link, (A) DAILINK_COMP_ARRAY(COMP_CPU("cpu_dai")), (B) DAILINK_COMP_ARRAY(COMP_CODEC("codec", "dai1"), (B) COMP_CODEC("codec", "dai2")), (C) DAILINK_COMP_ARRAY(COMP_EMPTY())); In this case, this macro will be converted to like below [o] = static struct snd_soc_dai_link_component (A) [o] link_cpus[] = {{ .dai_name = "cpu_dai" }}; (B) [o] link_codecs[] = {{ .dai_name = "dai1", .name = "codec" }, { .dai_name = "dai2", .name = "codec" }} (C) [o] link_platforms[] = {{ }}; CPU and Codec info will be filled by COMP_CPU() / COMP_CODEC (= A,B), and Platform will have empty data by COMP_EMPTY() (= C) in this case. Platform empty info will be filled when driver probe() (most of case, CPU info will be copied to use soc-generic-dmaengine-pcm). For example in case of DPCM FE/BE, it will be like below. Codec will be dummy Component / DAI in this case (X). SND_SOC_DAILINK_DEFS(link, DAILINK_COMP_ARRAY(COMP_CPU(...)), (X) DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); (X) part will converted like below [o] link_codecs[] = {{ .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", }} Even though we already have common asoc_dummy_dlc for dummy Component / DAI, this macro will re-create new dummy dlc. Some drivers defines many dai_link info via SND_SOC_DAILINK_DEFS(), this means many dummy dlc also will be re-created. This is waste of memory. If we can use existing common asoc_dummy_dlc at (X), we can avoid to re-creating dummy dlc, then, we can save the memory. At that time, we want to keep existing code as much as possible, because too many drivers are using this macro. But because of its original style, using common asoc_dummy_dlc from it is very difficult or impossible. So let's change the mind. The macro is used like below SND_SOC_DAILINK_DEFS(link, DAILINK_COMP_ARRAY(COMP_CPU(...)), (x) DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); static struct snd_soc_dai_link dai_links[] = { { .name = ..., .stream_name = ..., (y) SND_SOC_DAILINK_REG(link), }, (y) part will be like below static struct snd_soc_dai_link dai_links[] = { { .name = ..., .stream_name = ..., ^ ... | .codecs = link_codecs, (y) .num_codecs = ARRAY_SIZE(link_codecs), v ... } This patch try to use trick on COMP_DUMMY() - #define COMP_DUMMY() { .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", } + #define COMP_DUMMY() By this tric, (x) part will be like below. before [o] link_codecs[] = {{ .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", }} after [o] link_codecs[] = { }; This is same as below [o] link_codecs[0]; This means it has pointer (link_codecs), but the array size is 0. (y) part will be like below. static struct snd_soc_dai_link dai_links[] = { { ... .codecs = link_codecs, .num_codecs = 0, ... }, This is very special settings that normal use usually not do, but new macro do. We can find this special settings on soc-core.c and fill it as "dummy DAI" (= asoc_dummy_dlc). By this tric, we can avoid to re-create dummy dlc and save the memory. This patch add tric at COMP_DUMMY() and add snd_soc_fill_dummy_dai() to fill dummy DAI. Signed-off-by: Kuninori Morimoto Link: https://msgid.link/r/871qbi93qu.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 24 ++++++++++++++++++++++++ 2 files changed, 25 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f3803c2dc349..7cbe85ca040d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -938,7 +938,7 @@ snd_soc_link_to_platform(struct snd_soc_dai_link *link, int n) { #define COMP_PLATFORM(_name) { .name = _name } #define COMP_AUX(_name) { .name = _name } #define COMP_CODEC_CONF(_name) { .name = _name } -#define COMP_DUMMY() { .name = "snd-soc-dummy", .dai_name = "snd-soc-dummy-dai", } +#define COMP_DUMMY() /* see snd_soc_fill_dummy_dai() */ extern struct snd_soc_dai_link_component null_dailink_component[0]; extern struct snd_soc_dai_link_component snd_soc_dummy_dlc; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 132946f82a29..f8524b5bfb33 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -576,6 +576,28 @@ free_rtd: return NULL; } +static void snd_soc_fill_dummy_dai(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + int i; + + /* + * COMP_DUMMY() creates size 0 array on dai_link. + * Fill it as dummy DAI in case of CPU/Codec here. + * Do nothing for Platform. + */ + for_each_card_prelinks(card, i, dai_link) { + if (dai_link->num_cpus == 0 && dai_link->cpus) { + dai_link->num_cpus = 1; + dai_link->cpus = &snd_soc_dummy_dlc; + } + if (dai_link->num_codecs == 0 && dai_link->codecs) { + dai_link->num_codecs = 1; + dai_link->codecs = &snd_soc_dummy_dlc; + } + } +} + static void snd_soc_flush_all_delayed_work(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -2131,6 +2153,8 @@ static int snd_soc_bind_card(struct snd_soc_card *card) mutex_lock(&client_mutex); snd_soc_card_mutex_lock_root(card); + snd_soc_fill_dummy_dai(card); + snd_soc_dapm_init(&card->dapm, card, NULL); /* check whether any platform is ignore machine FE and using topology */ From 802134c8c2c8889f7cc504ab1ba6ada9816ca969 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 19 Dec 2023 16:54:09 +0530 Subject: [PATCH 247/337] ASoC: SOF: amd: Refactor spinlock_irq(&sdev->ipc_lock) sequence in irq_handler Refactor spinlock_irq(&sdev->ipc_lock) sequence in irq_handler to avoid race conditions for acquiring hw_semaphore. Signed-off-by: Venkata Prasad Potturu Link: https://msgid.link/r/20231219112416.3334928-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-ipc.c | 4 +--- sound/soc/sof/amd/acp.c | 3 +++ 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c index fcb54f545fea..2743f07a5e08 100644 --- a/sound/soc/sof/amd/acp-ipc.c +++ b/sound/soc/sof/amd/acp-ipc.c @@ -3,7 +3,7 @@ // This file is provided under a dual BSD/GPLv2 license. When using or // redistributing this file, you may do so under either license. // -// Copyright(c) 2021 Advanced Micro Devices, Inc. +// Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. // // Authors: Balakishore Pati // Ajit Kumar Pandey @@ -188,13 +188,11 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context) dsp_ack = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SCRATCH_REG_0 + dsp_ack_write); if (dsp_ack) { - spin_lock_irq(&sdev->ipc_lock); /* handle immediate reply from DSP core */ acp_dsp_ipc_get_reply(sdev); snd_sof_ipc_reply(sdev, 0); /* set the done bit */ acp_dsp_ipc_dsp_done(sdev); - spin_unlock_irq(&sdev->ipc_lock); ipc_irq = true; } diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 603ea5fc0d0d..7860724c4d2d 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -343,11 +343,13 @@ static irqreturn_t acp_irq_thread(int irq, void *context) const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int count = ACP_HW_SEM_RETRY_COUNT; + spin_lock_irq(&sdev->ipc_lock); while (snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset)) { /* Wait until acquired HW Semaphore lock or timeout */ count--; if (!count) { dev_err(sdev->dev, "%s: Failed to acquire HW lock\n", __func__); + spin_unlock_irq(&sdev->ipc_lock); return IRQ_NONE; } } @@ -356,6 +358,7 @@ static irqreturn_t acp_irq_thread(int irq, void *context) /* Unlock or Release HW Semaphore */ snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->hw_semaphore_offset, 0x0); + spin_unlock_irq(&sdev->ipc_lock); return IRQ_HANDLED; }; From 3953de2dbdcd0592aa7f877b67135a51e18f006a Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 19 Dec 2023 16:54:10 +0530 Subject: [PATCH 248/337] ASoC: SOF: Refactor sof_i2s_tokens reading to update acpbt dai Refactor sof_i2s_tokens reading to update config->acpbt. Signed-off-by: Venkata Prasad Potturu Link: https://msgid.link/r/20231219112416.3334928-2-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index a8e0054cb8a6..914eb187c5ac 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1219,6 +1219,7 @@ static int sof_link_acp_bt_load(struct snd_soc_component *scomp, struct snd_sof_ struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); + int ret; /* handle master/slave and inverted clocks */ sof_dai_set_format(hw_config, config); @@ -1227,12 +1228,14 @@ static int sof_link_acp_bt_load(struct snd_soc_component *scomp, struct snd_sof_ memset(&config->acpbt, 0, sizeof(config->acpbt)); config->hdr.size = size; - config->acpbt.fsync_rate = le32_to_cpu(hw_config->fsync_rate); - config->acpbt.tdm_slots = le32_to_cpu(hw_config->tdm_slots); + ret = sof_update_ipc_object(scomp, &config->acpbt, SOF_ACPI2S_TOKENS, slink->tuples, + slink->num_tuples, size, slink->num_hw_configs); + if (ret < 0) + return ret; - dev_info(scomp->dev, "ACP_BT config ACP%d channel %d rate %d\n", + dev_info(scomp->dev, "ACP_BT config ACP%d channel %d rate %d tdm_mode %d\n", config->dai_index, config->acpbt.tdm_slots, - config->acpbt.fsync_rate); + config->acpbt.fsync_rate, config->acpbt.tdm_mode); dai->number_configs = 1; dai->current_config = 0; From de111c9b521ddea4c7609155a617b5a0e93ad833 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 19 Dec 2023 16:54:11 +0530 Subject: [PATCH 249/337] ASoC: SOF: Add i2s bt dai configuration support for AMD platforms Add support for i2s bt dai configuration from topology. Signed-off-by: Venkata Prasad Potturu Link: https://msgid.link/r/20231219112416.3334928-3-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index c1f66ba0e987..22467c17f2b2 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1954,6 +1954,7 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ token_id = SOF_ACPDMIC_TOKENS; num_tuples += token_list[SOF_ACPDMIC_TOKENS].count; break; + case SOF_DAI_AMD_BT: case SOF_DAI_AMD_SP: case SOF_DAI_AMD_HS: case SOF_DAI_AMD_SP_VIRTUAL: From ced7151b9b0c74af1bc05ac4ad93648900709bb0 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 19 Dec 2023 16:54:12 +0530 Subject: [PATCH 250/337] ASoC: SOF: Rename amd_bt sof_dai_type Rename amd_bt sof_dai_type from ACP to ACP_BT. Signed-off-by: Venkata Prasad Potturu Link: https://msgid.link/r/20231219112416.3334928-4-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 22467c17f2b2..dd5a903a0515 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -289,7 +289,7 @@ static const struct sof_dai_types sof_dais[] = { {"ALH", SOF_DAI_INTEL_ALH}, {"SAI", SOF_DAI_IMX_SAI}, {"ESAI", SOF_DAI_IMX_ESAI}, - {"ACP", SOF_DAI_AMD_BT}, + {"ACPBT", SOF_DAI_AMD_BT}, {"ACPSP", SOF_DAI_AMD_SP}, {"ACPDMIC", SOF_DAI_AMD_DMIC}, {"ACPHS", SOF_DAI_AMD_HS}, From 55d7bbe433467a64ac82c41b4efd425aa24acdce Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 19 Dec 2023 16:54:13 +0530 Subject: [PATCH 251/337] ASoC: SOF: amd: Add acp-psp mailbox interface for iram-dram fence register modification Add acp-psp mailbox communication interface for iram-dram size modification to notify psp. Signed-off-by: Venkata Prasad Potturu Link: https://msgid.link/r/20231219112416.3334928-5-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 11 +++++++++++ sound/soc/sof/amd/acp.h | 5 +++++ 2 files changed, 16 insertions(+) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 7860724c4d2d..32a741fcb84f 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -278,6 +278,17 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } + /* psp_send_cmd only required for vangogh platform (rev - 5) */ + if (desc->rev == 5) { + /* Modify IRAM and DRAM size */ + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); + if (ret) + return ret; + ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | MBOX_ISREADY_FLAG); + if (ret) + return ret; + } + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_SHA_DSP_FW_QUALIFIER, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index c536cfde0e44..c645aee216fd 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -74,9 +74,14 @@ #define MP0_C2PMSG_114_REG 0x3810AC8 #define MP0_C2PMSG_73_REG 0x3810A24 #define MBOX_ACP_SHA_DMA_COMMAND 0x70000 +#define MBOX_ACP_IRAM_DRAM_FENCE_COMMAND 0x80000 #define MBOX_DELAY_US 1000 #define MBOX_READY_MASK 0x80000000 #define MBOX_STATUS_MASK 0xFFFF +#define MBOX_ISREADY_FLAG 0x40000000 +#define IRAM_DRAM_FENCE_0 0X0 +#define IRAM_DRAM_FENCE_1 0X01 +#define IRAM_DRAM_FENCE_2 0X02 #define BOX_SIZE_512 0x200 #define BOX_SIZE_1024 0x400 From 1b08e7697f1eef88de902820d181d5c4291f074c Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Tue, 19 Dec 2023 05:41:19 +0100 Subject: [PATCH 252/337] ASoC: sprd: Simplify memory allocation in sprd_platform_compr_dma_config() 'sg' is freed at the end sprd_platform_compr_dma_config() both in the normal and in the error handling path. There is no need to use the devm_kcalloc()/devm_kfree(), kcalloc()/kfree() is enough. Signed-off-by: Christophe JAILLET Link: https://msgid.link/r/d16f22ae0627249a9fc658927832590cd88c544e.1702960856.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/sprd/sprd-pcm-compress.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sprd/sprd-pcm-compress.c b/sound/soc/sprd/sprd-pcm-compress.c index 6cfab8844d0f..57bd1a0728ac 100644 --- a/sound/soc/sprd/sprd-pcm-compress.c +++ b/sound/soc/sprd/sprd-pcm-compress.c @@ -160,7 +160,7 @@ static int sprd_platform_compr_dma_config(struct snd_soc_component *component, return -ENODEV; } - sgt = sg = devm_kcalloc(dev, sg_num, sizeof(*sg), GFP_KERNEL); + sgt = sg = kcalloc(sg_num, sizeof(*sg), GFP_KERNEL); if (!sg) { ret = -ENOMEM; goto sg_err; @@ -250,12 +250,12 @@ static int sprd_platform_compr_dma_config(struct snd_soc_component *component, dma->desc->callback_param = cstream; } - devm_kfree(dev, sg); + kfree(sg); return 0; config_err: - devm_kfree(dev, sg); + kfree(sg); sg_err: dma_release_channel(dma->chan); return ret; From c13cf1991f4231d38f1c43fcf51ec1cf29c8c82d Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Tue, 19 Dec 2023 14:23:37 +0100 Subject: [PATCH 253/337] ASoC: dt-bindings: qcom,lpass-va-macro: remove spurious contains in if statement Remove this spurious "contains" which causes the bindings check of qcom,sm8450-lpass-va-macro compatible to fail with: codec@33f0000: clocks: [[156, 57, 1], [156, 102, 1], [156, 103, 1], [156, 70, 1]] is too long from schema $id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml# codec@33f0000: clock-names: ['mclk', 'macro', 'dcodec', 'npl'] is too long from schema $id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml# Seems the double "contains" was considered as valid by the tool but broke the entire if statements. Fixes: f243ef746d0a ("ASoC: dt-bindings: qcom,lpass-va-macro: Add SM8650 LPASS VA") Signed-off-by: Neil Armstrong Reviewed-by: Krzysztof Kozlowski Link: https://msgid.link/r/20231219-topic-sm8x50-upstream-va-macro-bindings-fix-v1-1-ae133886f70e@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/qcom,lpass-va-macro.yaml | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml index c03ff9472a85..6b483fa3c428 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml @@ -121,9 +121,8 @@ allOf: properties: compatible: contains: - contains: - enum: - - qcom,sm8550-lpass-va-macro + enum: + - qcom,sm8550-lpass-va-macro then: properties: clocks: From 51add1687f39292af626ac3c2046f49241713273 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 26 Nov 2023 22:40:18 +0100 Subject: [PATCH 254/337] ASoC: rt5645: Drop double EF20 entry from dmi_platform_data[] dmi_platform_data[] first contains a DMI entry matching: DMI_MATCH(DMI_PRODUCT_NAME, "EF20"), and then contains an identical entry except for the match being: DMI_MATCH(DMI_PRODUCT_NAME, "EF20EA"), Since these are partial (non exact) DMI matches the first match will also match any board with "EF20EA" in their DMI product-name, drop the second, redundant, entry. Fixes: a4dae468cfdd ("ASoC: rt5645: Add ACPI-defined GPIO for ECS EF20 series") Cc: Chris Chiu Signed-off-by: Hans de Goede Link: https://msgid.link/r/20231126214024.300505-2-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7938b52d741d..c7089c2f7c5c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3847,14 +3847,6 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&ecs_ef20_platform_data, }, - { - .ident = "EF20EA", - .callback = cht_rt5645_ef20_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_NAME, "EF20EA"), - }, - .driver_data = (void *)&ecs_ef20_platform_data, - }, { } }; From 8f28e1996a786a7538d65e5258d3eecb92943673 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 26 Nov 2023 22:40:19 +0100 Subject: [PATCH 255/337] ASoC: rt5645: Add platform-data for Acer Switch V 10 The Acer Switch V 10 uses the default jack-detect mode 3, but instead of using an analog microphone it is using a DMIC on dmic-data-pin 1, like other models following Intel's Braswell's reference design. Add a DMI quirk pointing to the intel_braswell_platform_data for this. Signed-off-by: Hans de Goede Link: https://msgid.link/r/20231126214024.300505-3-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c7089c2f7c5c..20bbdf76ffd7 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3847,6 +3847,14 @@ static const struct dmi_system_id dmi_platform_data[] = { }, .driver_data = (void *)&ecs_ef20_platform_data, }, + { + .ident = "Acer Switch V 10 (SW5-017)", + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Acer"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "SW5-017"), + }, + .driver_data = (void *)&intel_braswell_platform_data, + }, { } }; From f72a9c2b8f1487181302d69fb82d5c76226be3fb Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 26 Nov 2023 22:40:20 +0100 Subject: [PATCH 256/337] ASoC: rt5645: Refactor rt5645_parse_dt() Refactor rt5645_parse_dt(), make it take a pointer to struct rt5645_platform_data as argument instead of passing in the complete rt5645_priv struct. While at it also make it void since it always succeeds. This is a preparation patch for factoring the code to get the platform-data out into a separate helper function. Signed-off-by: Hans de Goede Link: https://msgid.link/r/20231126214024.300505-4-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 20bbdf76ffd7..7d0ad5650b44 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3869,18 +3869,12 @@ static bool rt5645_check_dp(struct device *dev) return false; } -static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) +static void rt5645_parse_dt(struct device *dev, struct rt5645_platform_data *pdata) { - rt5645->pdata.in2_diff = device_property_read_bool(dev, - "realtek,in2-differential"); - device_property_read_u32(dev, - "realtek,dmic1-data-pin", &rt5645->pdata.dmic1_data_pin); - device_property_read_u32(dev, - "realtek,dmic2-data-pin", &rt5645->pdata.dmic2_data_pin); - device_property_read_u32(dev, - "realtek,jd-mode", &rt5645->pdata.jd_mode); - - return 0; + pdata->in2_diff = device_property_read_bool(dev, "realtek,in2-differential"); + device_property_read_u32(dev, "realtek,dmic1-data-pin", &pdata->dmic1_data_pin); + device_property_read_u32(dev, "realtek,dmic2-data-pin", &pdata->dmic2_data_pin); + device_property_read_u32(dev, "realtek,jd-mode", &pdata->jd_mode); } static int rt5645_i2c_probe(struct i2c_client *i2c) @@ -3909,7 +3903,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) if (pdata) rt5645->pdata = *pdata; else if (rt5645_check_dp(&i2c->dev)) - rt5645_parse_dt(rt5645, &i2c->dev); + rt5645_parse_dt(&i2c->dev, &rt5645->pdata); else rt5645->pdata = jd_mode3_platform_data; From b4635b9cd9ae48050d72b645cc53175788bebf52 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 26 Nov 2023 22:40:21 +0100 Subject: [PATCH 257/337] ASoC: rt5645: Add rt5645_get_pdata() helper Add a rt5645_get_pdata() helper function which retreives the platform-data and overrides it with the quirks module parameter if that is set. This is a preparation patch for adding the rt5645_components() function. Signed-off-by: Hans de Goede Link: https://msgid.link/r/20231126214024.300505-5-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 51 ++++++++++++++++++++------------------- 1 file changed, 26 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7d0ad5650b44..f79a447eaffe 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3877,10 +3877,33 @@ static void rt5645_parse_dt(struct device *dev, struct rt5645_platform_data *pda device_property_read_u32(dev, "realtek,jd-mode", &pdata->jd_mode); } +static void rt5645_get_pdata(struct device *codec_dev, struct rt5645_platform_data *pdata) +{ + const struct dmi_system_id *dmi_data; + + dmi_data = dmi_first_match(dmi_platform_data); + if (dmi_data) { + dev_info(codec_dev, "Detected %s platform\n", dmi_data->ident); + *pdata = *((struct rt5645_platform_data *)dmi_data->driver_data); + } else if (rt5645_check_dp(codec_dev)) { + rt5645_parse_dt(codec_dev, pdata); + } else { + *pdata = jd_mode3_platform_data; + } + + if (quirk != -1) { + pdata->in2_diff = QUIRK_IN2_DIFF(quirk); + pdata->level_trigger_irq = QUIRK_LEVEL_IRQ(quirk); + pdata->inv_jd1_1 = QUIRK_INV_JD1_1(quirk); + pdata->inv_hp_pol = QUIRK_INV_HP_POL(quirk); + pdata->jd_mode = QUIRK_JD_MODE(quirk); + pdata->dmic1_data_pin = QUIRK_DMIC1_DATA_PIN(quirk); + pdata->dmic2_data_pin = QUIRK_DMIC2_DATA_PIN(quirk); + } +} + static int rt5645_i2c_probe(struct i2c_client *i2c) { - struct rt5645_platform_data *pdata = NULL; - const struct dmi_system_id *dmi_data; struct rt5645_priv *rt5645; int ret, i; unsigned int val; @@ -3893,29 +3916,7 @@ static int rt5645_i2c_probe(struct i2c_client *i2c) rt5645->i2c = i2c; i2c_set_clientdata(i2c, rt5645); - - dmi_data = dmi_first_match(dmi_platform_data); - if (dmi_data) { - dev_info(&i2c->dev, "Detected %s platform\n", dmi_data->ident); - pdata = dmi_data->driver_data; - } - - if (pdata) - rt5645->pdata = *pdata; - else if (rt5645_check_dp(&i2c->dev)) - rt5645_parse_dt(&i2c->dev, &rt5645->pdata); - else - rt5645->pdata = jd_mode3_platform_data; - - if (quirk != -1) { - rt5645->pdata.in2_diff = QUIRK_IN2_DIFF(quirk); - rt5645->pdata.level_trigger_irq = QUIRK_LEVEL_IRQ(quirk); - rt5645->pdata.inv_jd1_1 = QUIRK_INV_JD1_1(quirk); - rt5645->pdata.inv_hp_pol = QUIRK_INV_HP_POL(quirk); - rt5645->pdata.jd_mode = QUIRK_JD_MODE(quirk); - rt5645->pdata.dmic1_data_pin = QUIRK_DMIC1_DATA_PIN(quirk); - rt5645->pdata.dmic2_data_pin = QUIRK_DMIC2_DATA_PIN(quirk); - } + rt5645_get_pdata(&i2c->dev, &rt5645->pdata); if (has_acpi_companion(&i2c->dev)) { if (cht_rt5645_gpios) { From 4cd7654553b3cf1ce5a7560f0492f0431785dfae Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 26 Nov 2023 22:40:22 +0100 Subject: [PATCH 258/337] ASoC: rt5645: Add a rt5645_components() helper The rt5645 codec driver uses DMI quirks to configure the DMIC data-pins, which means that it knows which DMIC interface is used on a specific device. ATM we duplicate this DMI matching inside the UCM profiles to select the right DMIC interface. Add a rt5645_components() helper which the machine-driver can use to set the components string of the card so that UCM can get the info from the components string. This way we only need to add new DMI quirks in one place. Signed-off-by: Hans de Goede Link: https://msgid.link/r/20231126214024.300505-6-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 24 ++++++++++++++++++++++++ sound/soc/codecs/rt5645.h | 3 +++ 2 files changed, 27 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index f79a447eaffe..4f3ef004f555 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3902,6 +3902,30 @@ static void rt5645_get_pdata(struct device *codec_dev, struct rt5645_platform_da } } +const char *rt5645_components(struct device *codec_dev) +{ + struct rt5645_platform_data pdata = { }; + static char buf[32]; + const char *mic; + int spk = 2; + + rt5645_get_pdata(codec_dev, &pdata); + + if (pdata.dmic1_data_pin && pdata.dmic2_data_pin) + mic = "dmics12"; + else if (pdata.dmic1_data_pin) + mic = "dmic1"; + else if (pdata.dmic2_data_pin) + mic = "dmic2"; + else + mic = "in2"; + + snprintf(buf, sizeof(buf), "cfg-spk:%d cfg-mic:%s", spk, mic); + + return buf; +} +EXPORT_SYMBOL_GPL(rt5645_components); + static int rt5645_i2c_probe(struct i2c_client *i2c) { struct rt5645_priv *rt5645; diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index ac3de6f3bc2f..90816b2c5489 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -2201,4 +2201,7 @@ int rt5645_sel_asrc_clk_src(struct snd_soc_component *component, int rt5645_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack, struct snd_soc_jack *btn_jack); + +const char *rt5645_components(struct device *codec_dev); + #endif /* __RT5645_H__ */ From 8184e1db699befc5101bcfe401283becfc228a0b Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 26 Nov 2023 22:40:23 +0100 Subject: [PATCH 259/337] ASoC: rt5645: Add mono speaker information to the components string The GPD Win and Teclast X80 Pro both only have 1 speaker add information about this to the components string. Signed-off-by: Hans de Goede Link: https://msgid.link/r/20231126214024.300505-7-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5645.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4f3ef004f555..e109ee2e321c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -428,6 +428,9 @@ struct rt5645_platform_data { /* Invert HP detect status polarity */ bool inv_hp_pol; + /* Only 1 speaker connected */ + bool mono_speaker; + /* Value to assign to snd_soc_card.long_name */ const char *long_name; @@ -3653,6 +3656,7 @@ static const struct rt5645_platform_data buddy_platform_data = { static const struct rt5645_platform_data gpd_win_platform_data = { .jd_mode = 3, .inv_jd1_1 = true, + .mono_speaker = true, .long_name = "gpd-win-pocket-rt5645", /* The GPD pocket has a diff. mic, for the win this does not matter. */ .in2_diff = true, @@ -3676,6 +3680,11 @@ static const struct rt5645_platform_data lenovo_ideapad_miix_310_pdata = { .in2_diff = true, }; +static const struct rt5645_platform_data jd_mode3_monospk_platform_data = { + .jd_mode = 3, + .mono_speaker = true, +}; + static const struct rt5645_platform_data jd_mode3_platform_data = { .jd_mode = 3, }; @@ -3795,7 +3804,7 @@ static const struct dmi_system_id dmi_platform_data[] = { DMI_MATCH(DMI_SYS_VENDOR, "TECLAST"), DMI_MATCH(DMI_PRODUCT_NAME, "X80 Pro"), }, - .driver_data = (void *)&jd_mode3_platform_data, + .driver_data = (void *)&jd_mode3_monospk_platform_data, }, { .ident = "Lenovo Ideapad Miix 310", @@ -3911,6 +3920,9 @@ const char *rt5645_components(struct device *codec_dev) rt5645_get_pdata(codec_dev, &pdata); + if (pdata.mono_speaker) + spk = 1; + if (pdata.dmic1_data_pin && pdata.dmic2_data_pin) mic = "dmics12"; else if (pdata.dmic1_data_pin) From f87b4402163be352601f7a012ab0d8dba7ecc64d Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 26 Nov 2023 22:40:24 +0100 Subject: [PATCH 260/337] ASoC: Intel: cht_bsw_rt5645: Set card.components string Set the card.components string using the new rt5645_components() helper which returns a components string based on the DMI quirks inside the rt5645 codec driver. Signed-off-by: Hans de Goede Link: https://msgid.link/r/20231126214024.300505-8-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/cht_bsw_rt5645.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index df23a581c10e..c952a96cde7e 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -534,6 +534,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) const char *platform_name; struct cht_mc_private *drv; struct acpi_device *adev; + struct device *codec_dev; bool sof_parent; bool found = false; bool is_bytcr = false; @@ -583,7 +584,14 @@ static int snd_cht_mc_probe(struct platform_device *pdev) "i2c-%s", acpi_dev_name(adev)); cht_dailink[dai_index].codecs->name = cht_rt5645_codec_name; } + /* acpi_get_first_physical_node() returns a borrowed ref, no need to deref */ + codec_dev = acpi_get_first_physical_node(adev); acpi_dev_put(adev); + if (!codec_dev) + return -EPROBE_DEFER; + + snd_soc_card_chtrt5645.components = rt5645_components(codec_dev); + snd_soc_card_chtrt5650.components = rt5645_components(codec_dev); /* * swap SSP0 if bytcr is detected From b6190c452a2264ccd88c849b91990fe854a7ec72 Mon Sep 17 00:00:00 2001 From: Shengjiu Wang Date: Wed, 27 Dec 2023 17:27:43 +0800 Subject: [PATCH 261/337] ASoC: SOF: imx: Add SNDRV_PCM_INFO_BATCH flag The sof imx pcm device is a device which should support double buffering. Found this issue with pipewire. When there is no SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will set headroom to be zero, and because sof pcm device don't support residue report, when the latency setting is small, the "delay" always larger than "target" in alsa-pcm.c, that reading next period data is not scheduled on time. With SNDRV_PCM_INFO_BATCH flag in driver, the pipewire will select a smaller period size for device, then the task of reading next period data will be scheduled on time. Signed-off-by: Shengjiu Wang Link: https://lore.kernel.org/r/1703669263-13832-1-git-send-email-shengjiu.wang@nxp.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx8.c | 1 + sound/soc/sof/imx/imx8m.c | 1 + sound/soc/sof/imx/imx8ulp.c | 1 + 3 files changed, 3 insertions(+) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 170740bce839..d777e70250ef 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -603,6 +603,7 @@ static struct snd_sof_dsp_ops sof_imx8x_ops = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP }; diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index f088fd1a672b..1b976fa500aa 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -472,6 +472,7 @@ static struct snd_sof_dsp_ops sof_imx8m_ops = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, }; diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index ca6edb85ff71..2badca75782b 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -463,6 +463,7 @@ static struct snd_sof_dsp_ops sof_imx8ulp_ops = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, /* PM */ From 126c18a4bb6460c3d82b57c56941104ed34b7ba6 Mon Sep 17 00:00:00 2001 From: Dmitry Antipov Date: Thu, 21 Dec 2023 12:16:00 +0300 Subject: [PATCH 262/337] ALSA: seq: fix kvmalloc_array() arguments order When compiling with gcc version 14.0.0 20231220 (experimental) and W=1, I've noticed the following warning: sound/core/seq/seq_memory.c: In function 'snd_seq_pool_init': sound/core/seq/seq_memory.c:445:41: warning: 'kvmalloc_array' sizes specified with 'sizeof' in the earlier argument and not in the later argument [-Wcalloc-transposed-args] 445 | cellptr = kvmalloc_array(sizeof(struct snd_seq_event_cell), pool->size, | ^~~~~~ Since 'n' and 'size' arguments of 'kvmalloc_array()' are multiplied to calculate the final size, their actual order doesn't affect the result and so this is not a bug. But it's still worth to fix it. Signed-off-by: Dmitry Antipov Link: https://lore.kernel.org/r/20231221091605.14660-1-dmantipov@yandex.ru Signed-off-by: Takashi Iwai --- sound/core/seq/seq_memory.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index b603bb93f896..e705e7538118 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -442,7 +442,8 @@ int snd_seq_pool_init(struct snd_seq_pool *pool) if (snd_BUG_ON(!pool)) return -EINVAL; - cellptr = kvmalloc_array(sizeof(struct snd_seq_event_cell), pool->size, + cellptr = kvmalloc_array(pool->size, + sizeof(struct snd_seq_event_cell), GFP_KERNEL); if (!cellptr) return -ENOMEM; From ee694e7db47e1af00ffb29f569904a9ed576868f Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 21 Dec 2023 13:25:16 +0000 Subject: [PATCH 263/337] ALSA: hda: cs35l41: Support additional Dell models without _DSD Add new model entries into configuration table. Signed-off-by: Stefan Binding Cc: # v6.7+ Link: https://lore.kernel.org/r/20231221132518.3213-2-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 194e1179a253..5eab2de0d4bb 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -35,6 +35,10 @@ struct cs35l41_config { }; static const struct cs35l41_config cs35l41_config_table[] = { + { "10280B27", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280B28", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280BEB", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, + { "10280C4D", I2C, 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT }, 0, 1, -1, 1000, 4500, 24 }, /* * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. * We can override the _DSD to correct the boost type here. @@ -347,6 +351,10 @@ struct cs35l41_prop_model { static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CLSA0100", NULL, lenovo_legion_no_acpi }, { "CLSA0101", NULL, lenovo_legion_no_acpi }, + { "CSC3551", "10280B27", generic_dsd_config }, + { "CSC3551", "10280B28", generic_dsd_config }, + { "CSC3551", "10280BEB", generic_dsd_config }, + { "CSC3551", "10280C4D", generic_dsd_config }, { "CSC3551", "103C89C6", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, From d110858a6925827609d11db8513d76750483ec06 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 21 Dec 2023 13:25:17 +0000 Subject: [PATCH 264/337] ALSA: hda: cs35l41: Prevent firmware load if SPI speed too low Some laptops without _DSD have the SPI speed set very low in the BIOS. Since the SPI controller uses this speed as its max speed, the SPI transactions are very slow. Firmware download writes to many registers, and if the SPI speed is too slow, it can take a long time to download. For this reason, disable firmware loading if the maximum SPI speed is too low. Without Firmware, audio playback will work, but the volume will be low to ensure safe operation of the CS35L41. Signed-off-by: Stefan Binding Cc: # v6.7+ Link: https://lore.kernel.org/r/20231221132518.3213-3-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda.c | 25 ++++++++-- sound/pci/hda/cs35l41_hda.h | 12 ++++- sound/pci/hda/cs35l41_hda_i2c.c | 2 +- sound/pci/hda/cs35l41_hda_property.c | 74 +++++++++++++--------------- sound/pci/hda/cs35l41_hda_spi.c | 2 +- 5 files changed, 70 insertions(+), 45 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 92ca2b3b6c92..d3fa6e136744 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -12,6 +12,7 @@ #include #include #include +#include #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" @@ -996,6 +997,11 @@ static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) __be32 halo_sts; int ret; + if (cs35l41->bypass_fw) { + dev_warn(cs35l41->dev, "Bypassing Firmware.\n"); + return 0; + } + ret = cs35l41_init_dsp(cs35l41); if (ret) { dev_warn(cs35l41->dev, "Cannot Initialize Firmware. Error: %d\n", ret); @@ -1588,6 +1594,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i u32 values[HDA_MAX_COMPONENTS]; struct acpi_device *adev; struct device *physdev; + struct spi_device *spi; const char *sub; char *property; size_t nval; @@ -1610,7 +1617,7 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); if (!ret) { dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); - goto put_physdev; + goto out; } property = "cirrus,dev-index"; @@ -1701,8 +1708,20 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i hw_cfg->bst_type = CS35L41_EXT_BOOST; hw_cfg->valid = true; +out: put_device(physdev); + cs35l41->bypass_fw = false; + if (cs35l41->control_bus == SPI) { + spi = to_spi_device(cs35l41->dev); + if (spi->max_speed_hz < CS35L41_MAX_ACCEPTABLE_SPI_SPEED_HZ) { + dev_warn(cs35l41->dev, + "SPI speed is too slow to support firmware download: %d Hz.\n", + spi->max_speed_hz); + cs35l41->bypass_fw = true; + } + } + return 0; err: @@ -1711,14 +1730,13 @@ err: hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; acpi_dev_put(cs35l41->dacpi); -put_physdev: put_device(physdev); return ret; } int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, - struct regmap *regmap) + struct regmap *regmap, enum control_bus control_bus) { unsigned int regid, reg_revid; struct cs35l41_hda *cs35l41; @@ -1737,6 +1755,7 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i cs35l41->dev = dev; cs35l41->irq = irq; cs35l41->regmap = regmap; + cs35l41->control_bus = control_bus; dev_set_drvdata(dev, cs35l41); ret = cs35l41_hda_read_acpi(cs35l41, device_name, id); diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index 3d925d677213..43d55292b327 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -20,6 +20,8 @@ #include #include +#define CS35L41_MAX_ACCEPTABLE_SPI_SPEED_HZ 1000000 + struct cs35l41_amp_cal_data { u32 calTarget[2]; u32 calTime[2]; @@ -46,6 +48,11 @@ enum cs35l41_hda_gpio_function { CS35l41_SYNC, }; +enum control_bus { + I2C, + SPI +}; + struct cs35l41_hda { struct device *dev; struct regmap *regmap; @@ -74,6 +81,9 @@ struct cs35l41_hda { struct cs_dsp cs_dsp; struct acpi_device *dacpi; bool mute_override; + enum control_bus control_bus; + bool bypass_fw; + }; enum halo_state { @@ -85,7 +95,7 @@ enum halo_state { extern const struct dev_pm_ops cs35l41_hda_pm_ops; int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, - struct regmap *regmap); + struct regmap *regmap, enum control_bus control_bus); void cs35l41_hda_remove(struct device *dev); int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id); diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index b44536fbba17..603e9bff3a71 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -30,7 +30,7 @@ static int cs35l41_hda_i2c_probe(struct i2c_client *clt) return -ENODEV; return cs35l41_hda_probe(&clt->dev, device_name, clt->addr, clt->irq, - devm_regmap_init_i2c(clt, &cs35l41_regmap_i2c)); + devm_regmap_init_i2c(clt, &cs35l41_regmap_i2c), I2C); } static void cs35l41_hda_i2c_remove(struct i2c_client *clt) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 5eab2de0d4bb..be2b01b596c2 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -16,10 +16,6 @@ struct cs35l41_config { const char *ssid; - enum { - SPI, - I2C - } bus; int num_amps; enum { INTERNAL, @@ -35,46 +31,46 @@ struct cs35l41_config { }; static const struct cs35l41_config cs35l41_config_table[] = { - { "10280B27", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10280B28", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10280BEB", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, - { "10280C4D", I2C, 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT }, 0, 1, -1, 1000, 4500, 24 }, + { "10280B27", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280B28", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10280BEB", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, + { "10280C4D", 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, CS35L41_LEFT, CS35L41_RIGHT }, 0, 1, -1, 1000, 4500, 24 }, /* * Device 103C89C6 does have _DSD, however it is setup to use the wrong boost type. * We can override the _DSD to correct the boost type here. * Since this laptop has valid ACPI, we do not need to handle cs-gpios, since that already exists * in the ACPI. The Reset GPIO is also valid, so we can use the Reset defined in _DSD. */ - { "103C89C6", SPI, 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, - { "104312AF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431433", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431463", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431473", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, - { "10431483", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, - { "10431493", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "104314D3", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "104314E3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431503", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431533", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431573", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431663", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, - { "104316D3", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "104316F3", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "104317F3", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431863", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "104318D3", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, - { "10431C9F", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CAF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CCF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CDF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431CEF", SPI, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, - { "10431D1F", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431DA2", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "10431E02", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, - { "10431EE2", I2C, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, - { "10431F12", I2C, 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, - { "10431F1F", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, - { "10431F62", SPI, 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "103C89C6", 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, + { "104312AF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431433", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431463", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431473", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431483", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "10431493", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104314D3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104314E3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431503", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431533", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431573", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431663", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 1000, 4500, 24 }, + { "104316D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "104316F3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "104317F3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431863", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "104318D3", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "10431C9F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CAF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CCF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CDF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431CEF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, + { "10431D1F", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431DA2", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10431E02", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "10431EE2", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, -1, -1, 0, 0, 0 }, + { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, + { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, + { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, {} }; @@ -212,7 +208,7 @@ static int generic_dsd_config(struct cs35l41_hda *cs35l41, struct device *physde "_DSD already exists.\n"); } - if (cfg->bus == SPI) { + if (cs35l41->control_bus == SPI) { cs35l41->index = id; #if IS_ENABLED(CONFIG_SPI) diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index eb287aa5f782..b76c0dfd5fef 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -26,7 +26,7 @@ static int cs35l41_hda_spi_probe(struct spi_device *spi) return -ENODEV; return cs35l41_hda_probe(&spi->dev, device_name, spi_get_chipselect(spi, 0), spi->irq, - devm_regmap_init_spi(spi, &cs35l41_regmap_spi)); + devm_regmap_init_spi(spi, &cs35l41_regmap_spi), SPI); } static void cs35l41_hda_spi_remove(struct spi_device *spi) From 423206604b28174698d77bf5ea81365cdd6c0f77 Mon Sep 17 00:00:00 2001 From: Stefan Binding Date: Thu, 21 Dec 2023 13:25:18 +0000 Subject: [PATCH 265/337] ALSA: hda/realtek: Add quirks for Dell models These models use 2 or 4 CS35L41 amps with HDA using SPI and I2C. Models use internal and external boost. All models require DSD support to be added inside cs35l41_hda_property.c Signed-off-by: Stefan Binding Cc: # v6.7+ Link: https://lore.kernel.org/r/20231221132518.3213-4-sbinding@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c3a756528886..19040887ff67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6956,6 +6956,11 @@ static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); } +static void cs35l41_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 4); +} + static void cs35l41_fixup_spi_two(struct hda_codec *codec, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(codec, action, "spi", "CSC3551", 2); @@ -7441,6 +7446,7 @@ enum { ALC287_FIXUP_LEGION_16ACHG6, ALC287_FIXUP_CS35L41_I2C_2, ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED, + ALC287_FIXUP_CS35L41_I2C_4, ALC245_FIXUP_CS35L41_SPI_2, ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_4, @@ -9427,6 +9433,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_MUTE_LED, }, + [ALC287_FIXUP_CS35L41_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_four, + }, [ALC245_FIXUP_CS35L41_SPI_2] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_two, @@ -9703,6 +9713,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0b27, "Dell", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1028, 0x0b28, "Dell", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2), @@ -9713,6 +9725,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0c1c, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1d, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), + SND_PCI_QUIRK(0x1028, 0x0c4d, "Dell", ALC287_FIXUP_CS35L41_I2C_4), SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC289_FIXUP_DELL_CS35L41_SPI_2), From 649cc9e543a745620a2cdaa934efea2b123e594a Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:06:42 +1030 Subject: [PATCH 266/337] ALSA: scarlett2: Update maintainer info Update MAINTAINERS and "enabled" message with GitHub repository links. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/62f32404eaa8663cc304648354b85bcb5914ce72.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- MAINTAINERS | 6 ++++-- sound/usb/mixer_scarlett2.c | 3 ++- 2 files changed, 6 insertions(+), 3 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index ea790149af79..ae3f72f57854 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8272,11 +8272,13 @@ L: linux-input@vger.kernel.org S: Maintained F: drivers/input/joystick/fsia6b.c -FOCUSRITE SCARLETT GEN 2/3 MIXER DRIVER +FOCUSRITE SCARLETT2 MIXER DRIVER (Scarlett Gen 2+ and Clarett) M: Geoffrey D. Bennett L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained -T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git +W: https://github.com/geoffreybennett/scarlett-gen2 +B: https://github.com/geoffreybennett/scarlett-gen2/issues +T: git https://github.com/geoffreybennett/scarlett-gen2.git F: sound/usb/mixer_scarlett2.c FORCEDETH GIGABIT ETHERNET DRIVER diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 33a3d1161885..51f5471d3fda 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -4534,7 +4534,8 @@ int snd_scarlett2_init(struct usb_mixer_interface *mixer) usb_audio_info(chip, "Focusrite %s Mixer Driver enabled (pid=0x%04x); " - "report any issues to g@b4.vu", + "report any issues to " + "https://github.com/geoffreybennett/scarlett-gen2/issues", entry->series_name, USB_ID_PRODUCT(chip->usb_id)); From 5f6ff6931a1c0065a55448108940371e1ac8075f Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:07:00 +1030 Subject: [PATCH 267/337] ALSA: scarlett2: Add missing error check to scarlett2_config_save() scarlett2_config_save() was ignoring the return value from scarlett2_usb(). As this function is not called from user-space we can't return the error, so call usb_audio_err() instead. Signed-off-by: Geoffrey D. Bennett Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface") Link: https://lore.kernel.org/r/bf0a15332d852d7825fa6da87d2a0d9c0b702053.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 51f5471d3fda..0f38301dd3d5 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1524,9 +1524,11 @@ static void scarlett2_config_save(struct usb_mixer_interface *mixer) { __le32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); - scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, - &req, sizeof(u32), - NULL, 0); + int err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(u32), + NULL, 0); + if (err < 0) + usb_audio_err(mixer->chip, "config save failed: %d\n", err); } /* Delayed work to save config */ From ca459dfa7d4ed9098fcf13e410963be6ae9b6bf3 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:07:21 +1030 Subject: [PATCH 268/337] ALSA: scarlett2: Add missing error check to scarlett2_usb_set_config() scarlett2_usb_set_config() calls scarlett2_usb_get() but was not checking the result. Return the error if it fails rather than continuing with an invalid value. Signed-off-by: Geoffrey D. Bennett Fixes: 9e15fae6c51a ("ALSA: usb-audio: scarlett2: Allow bit-level access to config") Link: https://lore.kernel.org/r/def110c5c31dbdf0a7414d258838a0a31c0fab67.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 0f38301dd3d5..c030368efdae 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1577,7 +1577,10 @@ static int scarlett2_usb_set_config( size = 1; offset = config_item->offset; - scarlett2_usb_get(mixer, offset, &tmp, 1); + err = scarlett2_usb_get(mixer, offset, &tmp, 1); + if (err < 0) + return err; + if (value) tmp |= (1 << index); else From 50603a67daef161c78c814580d57f7f0be57167e Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:07:37 +1030 Subject: [PATCH 269/337] ALSA: scarlett2: Add missing error checks to *_ctl_get() The *_ctl_get() functions which call scarlett2_update_*() were not checking the return value. Fix to check the return value and pass to the caller. Signed-off-by: Geoffrey D. Bennett Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface") Link: https://lore.kernel.org/r/32a5fdc83b05fa74e0fcdd672fbf71d75c5f0a6d.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 196 +++++++++++++++++++++++++----------- 1 file changed, 137 insertions(+), 59 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index c030368efdae..30751bcba468 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2100,14 +2100,20 @@ static int scarlett2_sync_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->sync_updated) - scarlett2_update_sync(mixer); - ucontrol->value.enumerated.item[0] = private->sync; - mutex_unlock(&private->data_mutex); - return 0; + if (private->sync_updated) { + err = scarlett2_update_sync(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->sync; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static const struct snd_kcontrol_new scarlett2_sync_ctl = { @@ -2190,14 +2196,20 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->vol_updated) - scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); + if (private->vol_updated) { + err = scarlett2_update_volumes(mixer); + if (err < 0) + goto unlock; + } ucontrol->value.integer.value[0] = private->master_vol; - return 0; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int line_out_remap(struct scarlett2_data *private, int index) @@ -2223,14 +2235,20 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; int index = line_out_remap(private, elem->control); + int err = 0; mutex_lock(&private->data_mutex); - if (private->vol_updated) - scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); + if (private->vol_updated) { + err = scarlett2_update_volumes(mixer); + if (err < 0) + goto unlock; + } ucontrol->value.integer.value[0] = private->vol[index]; - return 0; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, @@ -2297,14 +2315,20 @@ static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; int index = line_out_remap(private, elem->control); + int err = 0; mutex_lock(&private->data_mutex); - if (private->vol_updated) - scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); + if (private->vol_updated) { + err = scarlett2_update_volumes(mixer); + if (err < 0) + goto unlock; + } ucontrol->value.integer.value[0] = private->mute_switch[index]; - return 0; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_mute_ctl_put(struct snd_kcontrol *kctl, @@ -2550,14 +2574,20 @@ static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, const struct scarlett2_device_info *info = private->info; int index = elem->control + info->level_input_first; + int err = 0; mutex_lock(&private->data_mutex); - if (private->input_other_updated) - scarlett2_update_input_other(mixer); - ucontrol->value.enumerated.item[0] = private->level_switch[index]; - mutex_unlock(&private->data_mutex); - return 0; + if (private->input_other_updated) { + err = scarlett2_update_input_other(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->level_switch[index]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, @@ -2608,15 +2638,21 @@ static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->input_other_updated) - scarlett2_update_input_other(mixer); + + if (private->input_other_updated) { + err = scarlett2_update_input_other(mixer); + if (err < 0) + goto unlock; + } ucontrol->value.integer.value[0] = private->pad_switch[elem->control]; - mutex_unlock(&private->data_mutex); - return 0; +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, @@ -2666,14 +2702,20 @@ static int scarlett2_air_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->input_other_updated) - scarlett2_update_input_other(mixer); - ucontrol->value.integer.value[0] = private->air_switch[elem->control]; - mutex_unlock(&private->data_mutex); - return 0; + if (private->input_other_updated) { + err = scarlett2_update_input_other(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = private->air_switch[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_air_ctl_put(struct snd_kcontrol *kctl, @@ -2723,15 +2765,21 @@ static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->input_other_updated) - scarlett2_update_input_other(mixer); + + if (private->input_other_updated) { + err = scarlett2_update_input_other(mixer); + if (err < 0) + goto unlock; + } ucontrol->value.integer.value[0] = private->phantom_switch[elem->control]; - mutex_unlock(&private->data_mutex); - return 0; +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, @@ -2903,14 +2951,20 @@ static int scarlett2_direct_monitor_ctl_get( struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = elem->head.mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->monitor_other_updated) - scarlett2_update_monitor_other(mixer); - ucontrol->value.enumerated.item[0] = private->direct_monitor_switch; - mutex_unlock(&private->data_mutex); - return 0; + if (private->monitor_other_updated) { + err = scarlett2_update_monitor_other(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->direct_monitor_switch; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_direct_monitor_ctl_put( @@ -3010,14 +3064,20 @@ static int scarlett2_speaker_switch_enum_ctl_get( struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->monitor_other_updated) - scarlett2_update_monitor_other(mixer); - ucontrol->value.enumerated.item[0] = private->speaker_switching_switch; - mutex_unlock(&private->data_mutex); - return 0; + if (private->monitor_other_updated) { + err = scarlett2_update_monitor_other(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->speaker_switching_switch; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } /* when speaker switching gets enabled, switch the main/alt speakers @@ -3165,14 +3225,20 @@ static int scarlett2_talkback_enum_ctl_get( struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->monitor_other_updated) - scarlett2_update_monitor_other(mixer); - ucontrol->value.enumerated.item[0] = private->talkback_switch; - mutex_unlock(&private->data_mutex); - return 0; + if (private->monitor_other_updated) { + err = scarlett2_update_monitor_other(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->talkback_switch; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_talkback_enum_ctl_put( @@ -3320,14 +3386,20 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + int err = 0; mutex_lock(&private->data_mutex); - if (private->vol_updated) - scarlett2_update_volumes(mixer); - mutex_unlock(&private->data_mutex); + if (private->vol_updated) { + err = scarlett2_update_volumes(mixer); + if (err < 0) + goto unlock; + } ucontrol->value.integer.value[0] = private->dim_mute[elem->control]; - return 0; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, @@ -3698,14 +3770,20 @@ static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; int index = line_out_remap(private, elem->control); + int err = 0; mutex_lock(&private->data_mutex); - if (private->mux_updated) - scarlett2_usb_get_mux(mixer); - ucontrol->value.enumerated.item[0] = private->mux[index]; - mutex_unlock(&private->data_mutex); - return 0; + if (private->mux_updated) { + err = scarlett2_usb_get_mux(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->mux[index]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, From 04f8f053252b86c7583895c962d66747ecdc61b7 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:07:52 +1030 Subject: [PATCH 270/337] ALSA: scarlett2: Add clamp() in scarlett2_mixer_ctl_put() Ensure the value passed to scarlett2_mixer_ctl_put() is between 0 and SCARLETT2_MIXER_MAX_VALUE so we don't attempt to access outside scarlett2_mixer_values[]. Signed-off-by: Geoffrey D. Bennett Fixes: 9e4d5c1be21f ("ALSA: usb-audio: Scarlett Gen 2 mixer interface") Link: https://lore.kernel.org/r/3b19fb3da641b587749b85fe1daa1b4e696c0c1b.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 30751bcba468..8a60c2169259 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -3663,7 +3663,8 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); oval = private->mix[index]; - val = ucontrol->value.integer.value[0]; + val = clamp(ucontrol->value.integer.value[0], + 0L, (long)SCARLETT2_MIXER_MAX_VALUE); num_mixer_in = port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT]; mix_num = index / num_mixer_in; From 993f7b42fa066b055e3a19b7f76ad8157c0927a0 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:08:09 +1030 Subject: [PATCH 271/337] ALSA: scarlett2: Add missing mutex lock around get meter levels As scarlett2_meter_ctl_get() uses meter_level_map[], the data_mutex should be locked while accessing it. Signed-off-by: Geoffrey D. Bennett Fixes: 3473185f31df ("ALSA: scarlett2: Remap Level Meter values") Link: https://lore.kernel.org/r/77e093c27402c83d0730681448fa4f57583349dd.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 8a60c2169259..a3cbeba7bd24 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -3880,10 +3880,12 @@ static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, u16 meter_levels[SCARLETT2_MAX_METERS]; int i, err; + mutex_lock(&private->data_mutex); + err = scarlett2_usb_get_meter_levels(elem->head.mixer, elem->channels, meter_levels); if (err < 0) - return err; + goto unlock; /* copy & translate from meter_levels[] using meter_level_map[] */ for (i = 0; i < elem->channels; i++) { @@ -3898,7 +3900,10 @@ static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, ucontrol->value.integer.value[i] = value; } - return 0; +unlock: + mutex_unlock(&private->data_mutex); + + return err; } static const struct snd_kcontrol_new scarlett2_meter_ctl = { From 103c23ccacaab14e5f8a63d60bdc4e5c81e166fc Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:08:42 +1030 Subject: [PATCH 272/337] ALSA: scarlett2: Add #defines for firmware upgrade Add #defines for SCARLETT2_USB_* needed for firmware upgrade: reboot, info-flash, info-segment, erase-segment, get-erase, and write-segment. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/3077651c21bc8d4f046c68b79ec387aa16fcc5e4.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 28 +++++++++++++++++----------- 1 file changed, 17 insertions(+), 11 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a3cbeba7bd24..a2d6d99bc13d 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1137,17 +1137,23 @@ static int scarlett2_get_port_start_num( #define SCARLETT2_USB_CMD_REQ 2 #define SCARLETT2_USB_CMD_RESP 3 -#define SCARLETT2_USB_INIT_1 0x00000000 -#define SCARLETT2_USB_INIT_2 0x00000002 -#define SCARLETT2_USB_GET_METER 0x00001001 -#define SCARLETT2_USB_GET_MIX 0x00002001 -#define SCARLETT2_USB_SET_MIX 0x00002002 -#define SCARLETT2_USB_GET_MUX 0x00003001 -#define SCARLETT2_USB_SET_MUX 0x00003002 -#define SCARLETT2_USB_GET_SYNC 0x00006004 -#define SCARLETT2_USB_GET_DATA 0x00800000 -#define SCARLETT2_USB_SET_DATA 0x00800001 -#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_INIT_1 0x00000000 +#define SCARLETT2_USB_INIT_2 0x00000002 +#define SCARLETT2_USB_REBOOT 0x00000003 +#define SCARLETT2_USB_GET_METER 0x00001001 +#define SCARLETT2_USB_GET_MIX 0x00002001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_GET_MUX 0x00003001 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_INFO_FLASH 0x00004000 +#define SCARLETT2_USB_INFO_SEGMENT 0x00004001 +#define SCARLETT2_USB_ERASE_SEGMENT 0x00004002 +#define SCARLETT2_USB_GET_ERASE 0x00004003 +#define SCARLETT2_USB_WRITE_SEGMENT 0x00004004 +#define SCARLETT2_USB_GET_SYNC 0x00006004 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 #define SCARLETT2_USB_CONFIG_SAVE 6 From 34101a0fb1d47d8def145e50b4221201f5348cd0 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:08:58 +1030 Subject: [PATCH 273/337] ALSA: scarlett2: Retrieve useful flash segment numbers Call SCARLETT2_USB_INFO_FLASH and SCARLETT2_USB_INFO_SEGMENT to find the App_Settings and App_Upgrade flash segment numbers, and store them in the scarlett2_data struct. These will be used later to implement reset to factory defaults and firmware upgrade functions. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/70f0108a9cf99b69f7aa920c4bcdb0cf4bf3da98.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 103 ++++++++++++++++++++++++++++++++++++ 1 file changed, 103 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a2d6d99bc13d..b62fc0038671 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -190,6 +190,11 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { 16345 }; +/* Flash segments that we may manipulate */ +#define SCARLETT2_SEGMENT_ID_SETTINGS 0 +#define SCARLETT2_SEGMENT_ID_FIRMWARE 1 +#define SCARLETT2_SEGMENT_ID_COUNT 2 + /* Maximum number of analogue outputs */ #define SCARLETT2_ANALOGUE_MAX 10 @@ -429,6 +434,8 @@ struct scarlett2_data { int num_mux_srcs; int num_mux_dsts; u32 firmware_version; + u8 flash_segment_nums[SCARLETT2_SEGMENT_ID_COUNT]; + u8 flash_segment_blocks[SCARLETT2_SEGMENT_ID_COUNT]; u16 scarlett2_seq; u8 sync_updated; u8 vol_updated; @@ -1160,6 +1167,13 @@ static int scarlett2_get_port_start_num( #define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 #define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 +#define SCARLETT2_FLASH_BLOCK_SIZE 4096 +#define SCARLETT2_SEGMENT_NUM_MIN 1 +#define SCARLETT2_SEGMENT_NUM_MAX 4 + +#define SCARLETT2_SEGMENT_SETTINGS_NAME "App_Settings" +#define SCARLETT2_SEGMENT_FIRMWARE_NAME "App_Upgrade" + /* volume status is read together (matches scarlett2_config_items[1]) */ struct scarlett2_usb_volume_status { /* dim/mute buttons */ @@ -4202,6 +4216,90 @@ static int scarlett2_usb_init(struct usb_mixer_interface *mixer) return 0; } +/* Get the flash segment numbers for the App_Settings and App_Upgrade + * segments and put them in the private data + */ +static int scarlett2_get_flash_segment_nums(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err, count, i; + + struct { + __le32 size; + __le32 count; + u8 unknown[8]; + } __packed flash_info; + + struct { + __le32 size; + __le32 flags; + char name[16]; + } __packed segment_info; + + err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_FLASH, + NULL, 0, + &flash_info, sizeof(flash_info)); + if (err < 0) + return err; + + count = le32_to_cpu(flash_info.count); + + /* sanity check count */ + if (count < SCARLETT2_SEGMENT_NUM_MIN || + count > SCARLETT2_SEGMENT_NUM_MAX + 1) { + usb_audio_err(mixer->chip, + "invalid flash segment count: %d\n", count); + return -EINVAL; + } + + for (i = 0; i < count; i++) { + __le32 segment_num_req = cpu_to_le32(i); + int flash_segment_id; + + err = scarlett2_usb(mixer, SCARLETT2_USB_INFO_SEGMENT, + &segment_num_req, sizeof(segment_num_req), + &segment_info, sizeof(segment_info)); + if (err < 0) { + usb_audio_err(mixer->chip, + "failed to get flash segment info %d: %d\n", + i, err); + return err; + } + + if (!strncmp(segment_info.name, + SCARLETT2_SEGMENT_SETTINGS_NAME, 16)) + flash_segment_id = SCARLETT2_SEGMENT_ID_SETTINGS; + else if (!strncmp(segment_info.name, + SCARLETT2_SEGMENT_FIRMWARE_NAME, 16)) + flash_segment_id = SCARLETT2_SEGMENT_ID_FIRMWARE; + else + continue; + + private->flash_segment_nums[flash_segment_id] = i; + private->flash_segment_blocks[flash_segment_id] = + le32_to_cpu(segment_info.size) / + SCARLETT2_FLASH_BLOCK_SIZE; + } + + /* segment 0 is App_Gold and we never want to touch that, so + * use 0 as the "not-found" value + */ + if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_SETTINGS]) { + usb_audio_err(mixer->chip, + "failed to find flash segment %s\n", + SCARLETT2_SEGMENT_SETTINGS_NAME); + return -EINVAL; + } + if (!private->flash_segment_nums[SCARLETT2_SEGMENT_ID_FIRMWARE]) { + usb_audio_err(mixer->chip, + "failed to find flash segment %s\n", + SCARLETT2_SEGMENT_FIRMWARE_NAME); + return -EINVAL; + } + + return 0; +} + /* Read configuration from the interface on start */ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { @@ -4517,6 +4615,11 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; + /* Get the upgrade & settings flash segment numbers */ + err = scarlett2_get_flash_segment_nums(mixer); + if (err < 0) + return err; + /* Add firmware version control */ err = scarlett2_add_firmware_version_ctl(mixer); if (err < 0) From 337b2f0e778f78f61972497994b70a05e8f6447d Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:09:23 +1030 Subject: [PATCH 274/337] ALSA: scarlett2: Add skeleton hwdep/ioctl interface Add skeleton hwdep/ioctl interface, beginning with SCARLETT2_IOCTL_PVERSION and SCARLETT2_IOCTL_REBOOT. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/24ffcd47a8a02ebad3c8b2438104af8f0169164e.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- MAINTAINERS | 1 + include/uapi/sound/scarlett2.h | 34 +++++++++++++++++ sound/usb/mixer_scarlett2.c | 67 +++++++++++++++++++++++++++++++++- 3 files changed, 101 insertions(+), 1 deletion(-) create mode 100644 include/uapi/sound/scarlett2.h diff --git a/MAINTAINERS b/MAINTAINERS index ae3f72f57854..80c65096538c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -8279,6 +8279,7 @@ S: Maintained W: https://github.com/geoffreybennett/scarlett-gen2 B: https://github.com/geoffreybennett/scarlett-gen2/issues T: git https://github.com/geoffreybennett/scarlett-gen2.git +F: include/uapi/sound/scarlett2.h F: sound/usb/mixer_scarlett2.c FORCEDETH GIGABIT ETHERNET DRIVER diff --git a/include/uapi/sound/scarlett2.h b/include/uapi/sound/scarlett2.h new file mode 100644 index 000000000000..ec0b7da335ff --- /dev/null +++ b/include/uapi/sound/scarlett2.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ +/* + * Focusrite Scarlett 2 Protocol Driver for ALSA + * (including Scarlett 2nd Gen, 3rd Gen, Clarett USB, and Clarett+ + * series products) + * + * Copyright (c) 2023 by Geoffrey D. Bennett + */ +#ifndef __UAPI_SOUND_SCARLETT2_H +#define __UAPI_SOUND_SCARLETT2_H + +#include +#include + +#define SCARLETT2_HWDEP_MAJOR 1 +#define SCARLETT2_HWDEP_MINOR 0 +#define SCARLETT2_HWDEP_SUBMINOR 0 + +#define SCARLETT2_HWDEP_VERSION \ + ((SCARLETT2_HWDEP_MAJOR << 16) | \ + (SCARLETT2_HWDEP_MINOR << 8) | \ + SCARLETT2_HWDEP_SUBMINOR) + +#define SCARLETT2_HWDEP_VERSION_MAJOR(v) (((v) >> 16) & 0xFF) +#define SCARLETT2_HWDEP_VERSION_MINOR(v) (((v) >> 8) & 0xFF) +#define SCARLETT2_HWDEP_VERSION_SUBMINOR(v) ((v) & 0xFF) + +/* Get protocol version */ +#define SCARLETT2_IOCTL_PVERSION _IOR('S', 0x60, int) + +/* Reboot */ +#define SCARLETT2_IOCTL_REBOOT _IO('S', 0x61) + +#endif /* __UAPI_SOUND_SCARLETT2_H */ diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index b62fc0038671..d27628e4bda8 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -146,6 +146,9 @@ #include #include +#include + +#include #include "usbaudio.h" #include "mixer.h" @@ -1439,6 +1442,16 @@ static int scarlett2_usb( /* validate the response */ if (err != resp_buf_size) { + + /* ESHUTDOWN and EPROTO are valid responses to a + * reboot request + */ + if (cmd == SCARLETT2_USB_REBOOT && + (err == -ESHUTDOWN || err == -EPROTO)) { + err = 0; + goto unlock; + } + usb_audio_err( mixer->chip, "%s USB response result cmd %x was %d expected %zu\n", @@ -4697,6 +4710,49 @@ static int snd_scarlett2_controls_create( return 0; } +/*** hwdep interface ***/ + +/* Reboot the device. */ +static int scarlett2_reboot(struct usb_mixer_interface *mixer) +{ + return scarlett2_usb(mixer, SCARLETT2_USB_REBOOT, NULL, 0, NULL, 0); +} + +static int scarlett2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, + unsigned int cmd, unsigned long arg) +{ + struct usb_mixer_interface *mixer = hw->private_data; + + switch (cmd) { + + case SCARLETT2_IOCTL_PVERSION: + return put_user(SCARLETT2_HWDEP_VERSION, + (int __user *)arg) ? -EFAULT : 0; + + case SCARLETT2_IOCTL_REBOOT: + return scarlett2_reboot(mixer); + + default: + return -ENOIOCTLCMD; + } +} + +static int scarlett2_hwdep_init(struct usb_mixer_interface *mixer) +{ + struct snd_hwdep *hw; + int err; + + err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); + if (err < 0) + return err; + + hw->private_data = mixer; + hw->exclusive = 1; + hw->ops.ioctl = scarlett2_hwdep_ioctl; + + return 0; +} + int snd_scarlett2_init(struct usb_mixer_interface *mixer) { struct snd_usb_audio *chip = mixer->chip; @@ -4738,11 +4794,20 @@ int snd_scarlett2_init(struct usb_mixer_interface *mixer) USB_ID_PRODUCT(chip->usb_id)); err = snd_scarlett2_controls_create(mixer, entry); - if (err < 0) + if (err < 0) { usb_audio_err(mixer->chip, "Error initialising %s Mixer Driver: %d", entry->series_name, err); + return err; + } + + err = scarlett2_hwdep_init(mixer); + if (err < 0) + usb_audio_err(mixer->chip, + "Error creating %s hwdep device: %d", + entry->series_name, + err); return err; } From 6a7508e64ee3e8320c886020bcdcd70f7fcbff2c Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 20 Dec 2023 04:20:29 +1030 Subject: [PATCH 275/337] ALSA: scarlett2: Add ioctl commands to erase flash segments Add ioctls: - SCARLETT2_IOCTL_SELECT_FLASH_SEGMENT - SCARLETT2_IOCTL_ERASE_FLASH_SEGMENT - SCARLETT2_IOCTL_GET_ERASE_PROGRESS The settings or the firmware flash segment can be selected and then erased (asynchronous operation), and the erase progress can be monitored. If the erase progress is not monitored, then subsequent hwdep operations will block until the erase is complete. Once the erase is started, ALSA controls that communicate with the device will all return -EBUSY, and the device must be rebooted. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/227409adb672f174bf3db211e9bda016fb4646ea.1703001053.git.g@b4.vu Signed-off-by: Takashi Iwai --- include/uapi/sound/scarlett2.h | 20 ++ sound/usb/mixer_scarlett2.c | 428 ++++++++++++++++++++++++++++++++- 2 files changed, 442 insertions(+), 6 deletions(-) diff --git a/include/uapi/sound/scarlett2.h b/include/uapi/sound/scarlett2.h index ec0b7da335ff..d0ff38ffa154 100644 --- a/include/uapi/sound/scarlett2.h +++ b/include/uapi/sound/scarlett2.h @@ -31,4 +31,24 @@ /* Reboot */ #define SCARLETT2_IOCTL_REBOOT _IO('S', 0x61) +/* Select flash segment */ +#define SCARLETT2_SEGMENT_ID_SETTINGS 0 +#define SCARLETT2_SEGMENT_ID_FIRMWARE 1 +#define SCARLETT2_SEGMENT_ID_COUNT 2 + +#define SCARLETT2_IOCTL_SELECT_FLASH_SEGMENT _IOW('S', 0x62, int) + +/* Erase selected flash segment */ +#define SCARLETT2_IOCTL_ERASE_FLASH_SEGMENT _IO('S', 0x63) + +/* Get selected flash segment erase progress + * 1 through to num_blocks, or 255 for complete + */ +struct scarlett2_flash_segment_erase_progress { + unsigned char progress; + unsigned char num_blocks; +}; +#define SCARLETT2_IOCTL_GET_ERASE_PROGRESS \ + _IOR('S', 0x64, struct scarlett2_flash_segment_erase_progress) + #endif /* __UAPI_SOUND_SCARLETT2_H */ diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index d27628e4bda8..2d60fa607bd8 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -193,11 +193,6 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { 16345 }; -/* Flash segments that we may manipulate */ -#define SCARLETT2_SEGMENT_ID_SETTINGS 0 -#define SCARLETT2_SEGMENT_ID_FIRMWARE 1 -#define SCARLETT2_SEGMENT_ID_COUNT 2 - /* Maximum number of analogue outputs */ #define SCARLETT2_ANALOGUE_MAX 10 @@ -267,6 +262,13 @@ enum { SCARLETT2_DIM_MUTE_COUNT = 2, }; +/* Flash Write State */ +enum { + SCARLETT2_FLASH_WRITE_STATE_IDLE = 0, + SCARLETT2_FLASH_WRITE_STATE_SELECTED = 1, + SCARLETT2_FLASH_WRITE_STATE_ERASING = 2 +}; + static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { "Mute Playback Switch", "Dim Playback Switch" }; @@ -427,6 +429,9 @@ struct scarlett2_data { struct usb_mixer_interface *mixer; struct mutex usb_mutex; /* prevent sending concurrent USB requests */ struct mutex data_mutex; /* lock access to this data */ + u8 hwdep_in_use; + u8 selected_flash_segment_id; + u8 flash_write_state; struct delayed_work work; const struct scarlett2_device_info *info; const char *series_name; @@ -2137,6 +2142,11 @@ static int scarlett2_sync_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->sync_updated) { err = scarlett2_update_sync(mixer); if (err < 0) @@ -2233,6 +2243,11 @@ static int scarlett2_master_volume_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->vol_updated) { err = scarlett2_update_volumes(mixer); if (err < 0) @@ -2272,6 +2287,11 @@ static int scarlett2_volume_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->vol_updated) { err = scarlett2_update_volumes(mixer); if (err < 0) @@ -2295,6 +2315,11 @@ static int scarlett2_volume_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->vol[index]; val = ucontrol->value.integer.value[0]; @@ -2352,6 +2377,11 @@ static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->vol_updated) { err = scarlett2_update_volumes(mixer); if (err < 0) @@ -2375,6 +2405,11 @@ static int scarlett2_mute_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->mute_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -2514,6 +2549,11 @@ static int scarlett2_sw_hw_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->vol_sw_hw_switch[index]; val = !!ucontrol->value.enumerated.item[0]; @@ -2611,6 +2651,11 @@ static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->input_other_updated) { err = scarlett2_update_input_other(mixer); if (err < 0) @@ -2636,6 +2681,11 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->level_switch[index]; val = !!ucontrol->value.enumerated.item[0]; @@ -2675,6 +2725,11 @@ static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->input_other_updated) { err = scarlett2_update_input_other(mixer); if (err < 0) @@ -2700,6 +2755,11 @@ static int scarlett2_pad_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->pad_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -2739,6 +2799,11 @@ static int scarlett2_air_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->input_other_updated) { err = scarlett2_update_input_other(mixer); if (err < 0) @@ -2763,6 +2828,11 @@ static int scarlett2_air_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->air_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -2802,6 +2872,11 @@ static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->input_other_updated) { err = scarlett2_update_input_other(mixer); if (err < 0) @@ -2827,6 +2902,11 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->phantom_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -2878,6 +2958,11 @@ static int scarlett2_phantom_persistence_ctl_put( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->phantom_persistence; val = !!ucontrol->value.integer.value[0]; @@ -2988,6 +3073,11 @@ static int scarlett2_direct_monitor_ctl_get( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->monitor_other_updated) { err = scarlett2_update_monitor_other(mixer); if (err < 0) @@ -3012,6 +3102,11 @@ static int scarlett2_direct_monitor_ctl_put( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->direct_monitor_switch; val = min(ucontrol->value.enumerated.item[0], 2U); @@ -3101,6 +3196,11 @@ static int scarlett2_speaker_switch_enum_ctl_get( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->monitor_other_updated) { err = scarlett2_update_monitor_other(mixer); if (err < 0) @@ -3181,6 +3281,11 @@ static int scarlett2_speaker_switch_enum_ctl_put( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->speaker_switching_switch; val = min(ucontrol->value.enumerated.item[0], 2U); @@ -3262,6 +3367,11 @@ static int scarlett2_talkback_enum_ctl_get( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->monitor_other_updated) { err = scarlett2_update_monitor_other(mixer); if (err < 0) @@ -3285,6 +3395,11 @@ static int scarlett2_talkback_enum_ctl_put( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->talkback_switch; val = min(ucontrol->value.enumerated.item[0], 2U); @@ -3349,6 +3464,11 @@ static int scarlett2_talkback_map_ctl_put( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->talkback_map[index]; val = !!ucontrol->value.integer.value[0]; @@ -3423,6 +3543,11 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->vol_updated) { err = scarlett2_update_volumes(mixer); if (err < 0) @@ -3451,6 +3576,11 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->dim_mute[index]; val = !!ucontrol->value.integer.value[0]; @@ -3695,6 +3825,11 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->mix[index]; val = clamp(ucontrol->value.integer.value[0], 0L, (long)SCARLETT2_MIXER_MAX_VALUE); @@ -3808,6 +3943,11 @@ static int scarlett2_mux_src_enum_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + if (private->mux_updated) { err = scarlett2_usb_get_mux(mixer); if (err < 0) @@ -3831,6 +3971,11 @@ static int scarlett2_mux_src_enum_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->mux[index]; val = min(ucontrol->value.enumerated.item[0], private->num_mux_srcs - 1U); @@ -3915,6 +4060,11 @@ static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + err = scarlett2_usb_get_meter_levels(elem->head.mixer, elem->channels, meter_levels); if (err < 0) @@ -3983,6 +4133,11 @@ static int scarlett2_msd_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->msd_switch; val = !!ucontrol->value.integer.value[0]; @@ -4050,6 +4205,11 @@ static int scarlett2_standalone_ctl_put(struct snd_kcontrol *kctl, mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + oval = private->standalone_switch; val = !!ucontrol->value.integer.value[0]; @@ -4712,12 +4872,241 @@ static int snd_scarlett2_controls_create( /*** hwdep interface ***/ -/* Reboot the device. */ +/* Set private->hwdep_in_use; prevents access to the ALSA controls + * while doing a config erase/firmware upgrade. + */ +static void scarlett2_lock(struct scarlett2_data *private) +{ + mutex_lock(&private->data_mutex); + private->hwdep_in_use = 1; + mutex_unlock(&private->data_mutex); +} + +/* Call SCARLETT2_USB_GET_ERASE to get the erase progress */ +static int scarlett2_get_erase_progress(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int segment_id, segment_num, err; + u8 erase_resp; + + struct { + __le32 segment_num; + __le32 pad; + } __packed erase_req; + + segment_id = private->selected_flash_segment_id; + segment_num = private->flash_segment_nums[segment_id]; + + if (segment_num < SCARLETT2_SEGMENT_NUM_MIN || + segment_num > SCARLETT2_SEGMENT_NUM_MAX) + return -EFAULT; + + /* Send the erase progress request */ + erase_req.segment_num = cpu_to_le32(segment_num); + erase_req.pad = 0; + + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_ERASE, + &erase_req, sizeof(erase_req), + &erase_resp, sizeof(erase_resp)); + if (err < 0) + return err; + + return erase_resp; +} + +/* Repeatedly call scarlett2_get_erase_progress() until it returns + * 0xff (erase complete) or we've waited 10 seconds (it usually takes + * <3 seconds). + */ +static int scarlett2_wait_for_erase(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i = 0; i < 100; i++) { + err = scarlett2_get_erase_progress(mixer); + if (err < 0) + return err; + + if (err == 0xff) + return 0; + + msleep(100); + } + + return -ETIMEDOUT; +} + +/* Reboot the device; wait for the erase to complete if one is in + * progress. + */ static int scarlett2_reboot(struct usb_mixer_interface *mixer) { + struct scarlett2_data *private = mixer->private_data; + + if (private->flash_write_state == + SCARLETT2_FLASH_WRITE_STATE_ERASING) { + int err = scarlett2_wait_for_erase(mixer); + + if (err < 0) + return err; + } + return scarlett2_usb(mixer, SCARLETT2_USB_REBOOT, NULL, 0, NULL, 0); } +/* Select a flash segment for erasing (and possibly writing to) */ +static int scarlett2_ioctl_select_flash_segment( + struct usb_mixer_interface *mixer, + unsigned long arg) +{ + struct scarlett2_data *private = mixer->private_data; + int segment_id, segment_num; + + if (get_user(segment_id, (int __user *)arg)) + return -EFAULT; + + /* Check the segment ID and segment number */ + if (segment_id < 0 || segment_id >= SCARLETT2_SEGMENT_ID_COUNT) + return -EINVAL; + + segment_num = private->flash_segment_nums[segment_id]; + if (segment_num < SCARLETT2_SEGMENT_NUM_MIN || + segment_num > SCARLETT2_SEGMENT_NUM_MAX) { + usb_audio_err(mixer->chip, + "%s: invalid segment number %d\n", + __func__, segment_id); + return -EFAULT; + } + + /* If erasing, wait for it to complete */ + if (private->flash_write_state == SCARLETT2_FLASH_WRITE_STATE_ERASING) { + int err = scarlett2_wait_for_erase(mixer); + + if (err < 0) + return err; + } + + /* Save the selected segment ID and set the state to SELECTED */ + private->selected_flash_segment_id = segment_id; + private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_SELECTED; + + return 0; +} + +/* Erase the previously-selected flash segment */ +static int scarlett2_ioctl_erase_flash_segment( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int segment_id, segment_num, err; + + struct { + __le32 segment_num; + __le32 pad; + } __packed erase_req; + + if (private->flash_write_state != SCARLETT2_FLASH_WRITE_STATE_SELECTED) + return -EINVAL; + + segment_id = private->selected_flash_segment_id; + segment_num = private->flash_segment_nums[segment_id]; + + if (segment_num < SCARLETT2_SEGMENT_NUM_MIN || + segment_num > SCARLETT2_SEGMENT_NUM_MAX) + return -EFAULT; + + /* Prevent access to ALSA controls that access the device from + * here on + */ + scarlett2_lock(private); + + /* Send the erase request */ + erase_req.segment_num = cpu_to_le32(segment_num); + erase_req.pad = 0; + + err = scarlett2_usb(mixer, SCARLETT2_USB_ERASE_SEGMENT, + &erase_req, sizeof(erase_req), + NULL, 0); + if (err < 0) + return err; + + /* On success, change the state from SELECTED to ERASING */ + private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_ERASING; + + return 0; +} + +/* Get the erase progress from the device */ +static int scarlett2_ioctl_get_erase_progress( + struct usb_mixer_interface *mixer, + unsigned long arg) +{ + struct scarlett2_data *private = mixer->private_data; + struct scarlett2_flash_segment_erase_progress progress; + int segment_id, segment_num, err; + u8 erase_resp; + + struct { + __le32 segment_num; + __le32 pad; + } __packed erase_req; + + /* Check that we're erasing */ + if (private->flash_write_state != SCARLETT2_FLASH_WRITE_STATE_ERASING) + return -EINVAL; + + segment_id = private->selected_flash_segment_id; + segment_num = private->flash_segment_nums[segment_id]; + + if (segment_num < SCARLETT2_SEGMENT_NUM_MIN || + segment_num > SCARLETT2_SEGMENT_NUM_MAX) + return -EFAULT; + + /* Send the erase progress request */ + erase_req.segment_num = cpu_to_le32(segment_num); + erase_req.pad = 0; + + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_ERASE, + &erase_req, sizeof(erase_req), + &erase_resp, sizeof(erase_resp)); + if (err < 0) + return err; + + progress.progress = erase_resp; + progress.num_blocks = private->flash_segment_blocks[segment_id]; + + if (copy_to_user((void __user *)arg, &progress, sizeof(progress))) + return -EFAULT; + + /* If the erase is complete, change the state from ERASING to + * IDLE. + */ + if (progress.progress == 0xff) + private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_IDLE; + + return 0; +} + +static int scarlett2_hwdep_open(struct snd_hwdep *hw, struct file *file) +{ + struct usb_mixer_interface *mixer = hw->private_data; + struct scarlett2_data *private = mixer->private_data; + + /* If erasing, wait for it to complete */ + if (private->flash_write_state == + SCARLETT2_FLASH_WRITE_STATE_ERASING) { + int err = scarlett2_wait_for_erase(mixer); + + if (err < 0) + return err; + } + + /* Set the state to IDLE */ + private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_IDLE; + + return 0; +} + static int scarlett2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { @@ -4732,11 +5121,36 @@ static int scarlett2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, case SCARLETT2_IOCTL_REBOOT: return scarlett2_reboot(mixer); + case SCARLETT2_IOCTL_SELECT_FLASH_SEGMENT: + return scarlett2_ioctl_select_flash_segment(mixer, arg); + + case SCARLETT2_IOCTL_ERASE_FLASH_SEGMENT: + return scarlett2_ioctl_erase_flash_segment(mixer); + + case SCARLETT2_IOCTL_GET_ERASE_PROGRESS: + return scarlett2_ioctl_get_erase_progress(mixer, arg); + default: return -ENOIOCTLCMD; } } +static int scarlett2_hwdep_release(struct snd_hwdep *hw, struct file *file) +{ + struct usb_mixer_interface *mixer = hw->private_data; + struct scarlett2_data *private = mixer->private_data; + + /* Return from the SELECTED or WRITE state to IDLE. + * The ERASING state is left as-is, and checked on next open. + */ + if (private && + private->hwdep_in_use && + private->flash_write_state != SCARLETT2_FLASH_WRITE_STATE_ERASING) + private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_IDLE; + + return 0; +} + static int scarlett2_hwdep_init(struct usb_mixer_interface *mixer) { struct snd_hwdep *hw; @@ -4748,7 +5162,9 @@ static int scarlett2_hwdep_init(struct usb_mixer_interface *mixer) hw->private_data = mixer; hw->exclusive = 1; + hw->ops.open = scarlett2_hwdep_open; hw->ops.ioctl = scarlett2_hwdep_ioctl; + hw->ops.release = scarlett2_hwdep_release; return 0; } From 1abfbd3c952781881dc17d8e3624cac9df6a70b5 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Fri, 29 Dec 2023 22:49:23 +1030 Subject: [PATCH 276/337] ALSA: scarlett2: Add support for uploading new firmware Add ops.write to the hwdep interface. Once the upgrade firmware flash segment has been erased, writes to the hwdep fd are permitted, and translated to SCARLETT2_USB_WRITE_SEGMENT commands to the device. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/ZY65S0ojShSNSeRQ@m.b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 96 +++++++++++++++++++++++++++++++++++-- 1 file changed, 93 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 2d60fa607bd8..6a9e8e2e246f 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -266,7 +266,8 @@ enum { enum { SCARLETT2_FLASH_WRITE_STATE_IDLE = 0, SCARLETT2_FLASH_WRITE_STATE_SELECTED = 1, - SCARLETT2_FLASH_WRITE_STATE_ERASING = 2 + SCARLETT2_FLASH_WRITE_STATE_ERASING = 2, + SCARLETT2_FLASH_WRITE_STATE_WRITE = 3 }; static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { @@ -1176,6 +1177,7 @@ static int scarlett2_get_port_start_num( #define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 #define SCARLETT2_FLASH_BLOCK_SIZE 4096 +#define SCARLETT2_FLASH_WRITE_MAX 1024 #define SCARLETT2_SEGMENT_NUM_MIN 1 #define SCARLETT2_SEGMENT_NUM_MAX 4 @@ -5079,10 +5081,10 @@ static int scarlett2_ioctl_get_erase_progress( return -EFAULT; /* If the erase is complete, change the state from ERASING to - * IDLE. + * WRITE. */ if (progress.progress == 0xff) - private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_IDLE; + private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_WRITE; return 0; } @@ -5135,6 +5137,93 @@ static int scarlett2_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, } } +static long scarlett2_hwdep_write(struct snd_hwdep *hw, + const char __user *buf, + long count, loff_t *offset) +{ + struct usb_mixer_interface *mixer = hw->private_data; + struct scarlett2_data *private = mixer->private_data; + int segment_id, segment_num, err, len; + int flash_size; + + /* SCARLETT2_USB_WRITE_SEGMENT request data */ + struct { + __le32 segment_num; + __le32 offset; + __le32 pad; + u8 data[]; + } __packed *req; + + /* Calculate the maximum permitted in data[] */ + const size_t max_data_size = SCARLETT2_FLASH_WRITE_MAX - + offsetof(typeof(*req), data); + + /* If erasing, wait for it to complete */ + if (private->flash_write_state == + SCARLETT2_FLASH_WRITE_STATE_ERASING) { + err = scarlett2_wait_for_erase(mixer); + if (err < 0) + return err; + private->flash_write_state = SCARLETT2_FLASH_WRITE_STATE_WRITE; + + /* Check that an erase has been done & completed */ + } else if (private->flash_write_state != + SCARLETT2_FLASH_WRITE_STATE_WRITE) { + return -EINVAL; + } + + /* Check that we're writing to the upgrade firmware */ + segment_id = private->selected_flash_segment_id; + if (segment_id != SCARLETT2_SEGMENT_ID_FIRMWARE) + return -EINVAL; + + segment_num = private->flash_segment_nums[segment_id]; + if (segment_num < SCARLETT2_SEGMENT_NUM_MIN || + segment_num > SCARLETT2_SEGMENT_NUM_MAX) + return -EFAULT; + + /* Validate the offset and count */ + flash_size = private->flash_segment_blocks[segment_id] * + SCARLETT2_FLASH_BLOCK_SIZE; + + if (count < 0 || *offset < 0 || *offset + count >= flash_size) + return -EINVAL; + + if (!count) + return 0; + + /* Limit the *req size to SCARLETT2_FLASH_WRITE_MAX */ + if (count > max_data_size) + count = max_data_size; + + /* Create and send the request */ + len = struct_size(req, data, count); + req = kzalloc(len, GFP_KERNEL); + if (!req) + return -ENOMEM; + + req->segment_num = cpu_to_le32(segment_num); + req->offset = cpu_to_le32(*offset); + req->pad = 0; + + if (copy_from_user(req->data, buf, count)) { + err = -EFAULT; + goto error; + } + + err = scarlett2_usb(mixer, SCARLETT2_USB_WRITE_SEGMENT, + req, len, NULL, 0); + if (err < 0) + goto error; + + *offset += count; + err = count; + +error: + kfree(req); + return err; +} + static int scarlett2_hwdep_release(struct snd_hwdep *hw, struct file *file) { struct usb_mixer_interface *mixer = hw->private_data; @@ -5164,6 +5253,7 @@ static int scarlett2_hwdep_init(struct usb_mixer_interface *mixer) hw->exclusive = 1; hw->ops.open = scarlett2_hwdep_open; hw->ops.ioctl = scarlett2_hwdep_ioctl; + hw->ops.write = scarlett2_hwdep_write; hw->ops.release = scarlett2_hwdep_release; return 0; From a2bb6c7d80575a67edb7d8a37dae95b76a86be3f Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:50:09 +1030 Subject: [PATCH 277/337] ALSA: scarlett2: Simplify enums by removing explicit values This commit removes the explicit integer assignments from the enums. The actual values matter little, and not assigning explicit values makes it easier to modify the longer lists in the future. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/67f0f1bb8b90d7c76dfe7062d22d33bbde19cf93.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 76 ++++++++++++++++++------------------- 1 file changed, 38 insertions(+), 38 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6a9e8e2e246f..6e044b63ec91 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -223,11 +223,11 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { * devices, dependent on series and model. */ enum { - SCARLETT2_CONFIG_SET_GEN_2 = 0, - SCARLETT2_CONFIG_SET_GEN_3A = 1, - SCARLETT2_CONFIG_SET_GEN_3B = 2, - SCARLETT2_CONFIG_SET_CLARETT = 3, - SCARLETT2_CONFIG_SET_COUNT = 4 + SCARLETT2_CONFIG_SET_GEN_2, + SCARLETT2_CONFIG_SET_GEN_3A, + SCARLETT2_CONFIG_SET_GEN_3B, + SCARLETT2_CONFIG_SET_CLARETT, + SCARLETT2_CONFIG_SET_COUNT }; /* Hardware port types: @@ -239,35 +239,35 @@ enum { * - PCM I/O */ enum { - SCARLETT2_PORT_TYPE_NONE = 0, - SCARLETT2_PORT_TYPE_ANALOGUE = 1, - SCARLETT2_PORT_TYPE_SPDIF = 2, - SCARLETT2_PORT_TYPE_ADAT = 3, - SCARLETT2_PORT_TYPE_MIX = 4, - SCARLETT2_PORT_TYPE_PCM = 5, - SCARLETT2_PORT_TYPE_COUNT = 6, + SCARLETT2_PORT_TYPE_NONE, + SCARLETT2_PORT_TYPE_ANALOGUE, + SCARLETT2_PORT_TYPE_SPDIF, + SCARLETT2_PORT_TYPE_ADAT, + SCARLETT2_PORT_TYPE_MIX, + SCARLETT2_PORT_TYPE_PCM, + SCARLETT2_PORT_TYPE_COUNT }; /* I/O count of each port type kept in struct scarlett2_ports */ enum { - SCARLETT2_PORT_IN = 0, - SCARLETT2_PORT_OUT = 1, - SCARLETT2_PORT_DIRNS = 2, + SCARLETT2_PORT_IN, + SCARLETT2_PORT_OUT, + SCARLETT2_PORT_DIRNS }; /* Dim/Mute buttons on the 18i20 */ enum { - SCARLETT2_BUTTON_MUTE = 0, - SCARLETT2_BUTTON_DIM = 1, - SCARLETT2_DIM_MUTE_COUNT = 2, + SCARLETT2_BUTTON_MUTE, + SCARLETT2_BUTTON_DIM, + SCARLETT2_DIM_MUTE_COUNT }; /* Flash Write State */ enum { - SCARLETT2_FLASH_WRITE_STATE_IDLE = 0, - SCARLETT2_FLASH_WRITE_STATE_SELECTED = 1, - SCARLETT2_FLASH_WRITE_STATE_ERASING = 2, - SCARLETT2_FLASH_WRITE_STATE_WRITE = 3 + SCARLETT2_FLASH_WRITE_STATE_IDLE, + SCARLETT2_FLASH_WRITE_STATE_SELECTED, + SCARLETT2_FLASH_WRITE_STATE_ERASING, + SCARLETT2_FLASH_WRITE_STATE_WRITE }; static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { @@ -1211,22 +1211,22 @@ struct scarlett2_usb_volume_status { /* Configuration parameters that can be read and written */ enum { - SCARLETT2_CONFIG_DIM_MUTE = 0, - SCARLETT2_CONFIG_LINE_OUT_VOLUME = 1, - SCARLETT2_CONFIG_MUTE_SWITCH = 2, - SCARLETT2_CONFIG_SW_HW_SWITCH = 3, - SCARLETT2_CONFIG_LEVEL_SWITCH = 4, - SCARLETT2_CONFIG_PAD_SWITCH = 5, - SCARLETT2_CONFIG_MSD_SWITCH = 6, - SCARLETT2_CONFIG_AIR_SWITCH = 7, - SCARLETT2_CONFIG_STANDALONE_SWITCH = 8, - SCARLETT2_CONFIG_PHANTOM_SWITCH = 9, - SCARLETT2_CONFIG_PHANTOM_PERSISTENCE = 10, - SCARLETT2_CONFIG_DIRECT_MONITOR = 11, - SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH = 12, - SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE = 13, - SCARLETT2_CONFIG_TALKBACK_MAP = 14, - SCARLETT2_CONFIG_COUNT = 15 + SCARLETT2_CONFIG_DIM_MUTE, + SCARLETT2_CONFIG_LINE_OUT_VOLUME, + SCARLETT2_CONFIG_MUTE_SWITCH, + SCARLETT2_CONFIG_SW_HW_SWITCH, + SCARLETT2_CONFIG_LEVEL_SWITCH, + SCARLETT2_CONFIG_PAD_SWITCH, + SCARLETT2_CONFIG_MSD_SWITCH, + SCARLETT2_CONFIG_AIR_SWITCH, + SCARLETT2_CONFIG_STANDALONE_SWITCH, + SCARLETT2_CONFIG_PHANTOM_SWITCH, + SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, + SCARLETT2_CONFIG_DIRECT_MONITOR, + SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH, + SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE, + SCARLETT2_CONFIG_TALKBACK_MAP, + SCARLETT2_CONFIG_COUNT }; /* Location, size, and activation command number for the configuration From 3a4e1afe7d984a3de8cc3ef8447ab085800f6b54 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:50:52 +1030 Subject: [PATCH 278/337] ALSA: scarlett2: Infer has_msd_mode from config items Rather than storing has_msd_mode in the per-device structure, infer this from the presence of the SCARLETT2_CONFIG_MSD_SWITCH entry in the device's configuration set. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/ecbf3740e6b30a245333528ae4c504f37a9bc6bf.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6e044b63ec91..703cd65e42f4 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -355,12 +355,6 @@ struct scarlett2_meter_entry { }; struct scarlett2_device_info { - /* Gen 3 devices have an internal MSD mode switch that needs - * to be disabled in order to access the full functionality of - * the device. - */ - u8 has_msd_mode; - /* which set of configuration parameters the device uses */ u8 config_set; @@ -652,7 +646,6 @@ static const struct scarlett2_device_info s18i20_gen2_info = { }; static const struct scarlett2_device_info solo_gen3_info = { - .has_msd_mode = 1, .config_set = SCARLETT2_CONFIG_SET_GEN_3A, .level_input_count = 1, .level_input_first = 1, @@ -663,7 +656,6 @@ static const struct scarlett2_device_info solo_gen3_info = { }; static const struct scarlett2_device_info s2i2_gen3_info = { - .has_msd_mode = 1, .config_set = SCARLETT2_CONFIG_SET_GEN_3A, .level_input_count = 2, .air_input_count = 2, @@ -673,7 +665,6 @@ static const struct scarlett2_device_info s2i2_gen3_info = { }; static const struct scarlett2_device_info s4i4_gen3_info = { - .has_msd_mode = 1, .config_set = SCARLETT2_CONFIG_SET_GEN_3B, .level_input_count = 2, .pad_input_count = 2, @@ -723,7 +714,6 @@ static const struct scarlett2_device_info s4i4_gen3_info = { }; static const struct scarlett2_device_info s8i6_gen3_info = { - .has_msd_mode = 1, .config_set = SCARLETT2_CONFIG_SET_GEN_3B, .level_input_count = 2, .pad_input_count = 2, @@ -782,7 +772,6 @@ static const struct scarlett2_device_info s8i6_gen3_info = { }; static const struct scarlett2_device_info s18i8_gen3_info = { - .has_msd_mode = 1, .config_set = SCARLETT2_CONFIG_SET_GEN_3B, .line_out_hw_vol = 1, .has_speaker_switching = 1, @@ -863,7 +852,6 @@ static const struct scarlett2_device_info s18i8_gen3_info = { }; static const struct scarlett2_device_info s18i20_gen3_info = { - .has_msd_mode = 1, .config_set = SCARLETT2_CONFIG_SET_GEN_3B, .line_out_hw_vol = 1, .has_speaker_switching = 1, @@ -1518,6 +1506,19 @@ static int scarlett2_usb_get( &req, sizeof(req), buf, size); } +/* Return true if the given configuration item is present in the + * configuration set used by this device. + */ +static int scarlett2_has_config_item( + struct scarlett2_data *private, int config_item_num) +{ + const struct scarlett2_device_info *info = private->info; + const struct scarlett2_config *config_item = + &scarlett2_config_items[info->config_set][config_item_num]; + + return !!config_item->offset; +} + /* Send a USB message to get configuration parameters; result placed in *buf */ static int scarlett2_usb_get_config( struct usb_mixer_interface *mixer, @@ -4170,9 +4171,8 @@ static const struct snd_kcontrol_new scarlett2_msd_ctl = { static int scarlett2_add_msd_ctl(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - if (!info->has_msd_mode) + if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) return 0; /* If MSD mode is off, hide the switch by default */ @@ -4488,7 +4488,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) struct scarlett2_usb_volume_status volume_status; int err, i; - if (info->has_msd_mode) { + if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) { err = scarlett2_usb_get_config( mixer, SCARLETT2_CONFIG_MSD_SWITCH, 1, &private->msd_switch); From 3978fefdf416d4c14ba43365ef912dcc4e2ee5b6 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:51:40 +1030 Subject: [PATCH 279/337] ALSA: scarlett2: Infer standalone switch from config items Rather than assuming the standalone switch is present for all devices with a mixer, instead check for the presence of the SCARLETT2_CONFIG_STANDALONE_SWITCH config item. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/59c30885b02d65feaab2c338cf46889d72d01813.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 703cd65e42f4..064d6d34a8ab 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -4244,7 +4244,8 @@ static int scarlett2_add_standalone_ctl(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; - if (private->info->config_set == SCARLETT2_CONFIG_SET_GEN_3A) + if (!scarlett2_has_config_item(private, + SCARLETT2_CONFIG_STANDALONE_SWITCH)) return 0; /* Add standalone control */ @@ -4512,11 +4513,14 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (info->config_set == SCARLETT2_CONFIG_SET_GEN_3A) return 0; - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_STANDALONE_SWITCH, - 1, &private->standalone_switch); - if (err < 0) - return err; + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_STANDALONE_SWITCH)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_STANDALONE_SWITCH, + 1, &private->standalone_switch); + if (err < 0) + return err; + } err = scarlett2_update_sync(mixer); if (err < 0) From 2edc76dddee8a2e785f6566d2baddb49ad62fb5b Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:51:56 +1030 Subject: [PATCH 280/337] ALSA: scarlett2: Check for phantom persistence config item Allow for the phantom persistence config item to not exist. This is needed for the Scarlett Gen 4 series. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/3ccaf8069280827bd6c44f103fcb770bd50b7e2e.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 064d6d34a8ab..84dd9c43afde 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2621,11 +2621,15 @@ static int scarlett2_update_input_other(struct usb_mixer_interface *mixer) if (err < 0) return err; - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, - 1, &private->phantom_persistence); - if (err < 0) - return err; + if (scarlett2_has_config_item( + private, + SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, + 1, &private->phantom_persistence); + if (err < 0) + return err; + } } return 0; @@ -3779,7 +3783,9 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) return err; } } - if (info->phantom_count) { + if (info->phantom_count && + scarlett2_has_config_item(private, + SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) { err = scarlett2_add_new_ctl( mixer, &scarlett2_phantom_persistence_ctl, 0, 1, "Phantom Power Persistence Capture Switch", NULL); From c13d43a8582aa9b003b40fc6eecb66bd4ee8b5d6 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:52:24 +1030 Subject: [PATCH 281/337] ALSA: scarlett2: Check presence of mixer using mux_assignment Currently the presence of a mixer is determined by checking if the device uses the GEN_3A config set. Add scarlett2_has_mixer() function which checks for the presence of mux_assignment entries instead. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/ef6f4d360c2fe682ab65f83cccbe5be66ccc6296.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 84dd9c43afde..26bdd1beae2e 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1680,6 +1680,12 @@ static int scarlett2_usb_get_volume_status( buf, sizeof(*buf)); } +/* Return true if the device has a mixer that we can control */ +static int scarlett2_has_mixer(struct scarlett2_data *private) +{ + return !!private->info->mux_assignment[0][0].count; +} + /* Send a USB message to get the volumes for all inputs of one mix * and put the values into private->mix[] */ @@ -2175,7 +2181,7 @@ static int scarlett2_add_sync_ctl(struct usb_mixer_interface *mixer) struct scarlett2_data *private = mixer->private_data; /* devices without a mixer also don't support reporting sync status */ - if (private->info->config_set == SCARLETT2_CONFIG_SET_GEN_3A) + if (!scarlett2_has_mixer(private)) return 0; return scarlett2_add_new_ctl(mixer, &scarlett2_sync_ctl, @@ -4111,7 +4117,7 @@ static int scarlett2_add_meter_ctl(struct usb_mixer_interface *mixer) struct scarlett2_data *private = mixer->private_data; /* devices without a mixer also don't support reporting levels */ - if (private->info->config_set == SCARLETT2_CONFIG_SET_GEN_3A) + if (!scarlett2_has_mixer(private)) return 0; return scarlett2_add_new_ctl(mixer, &scarlett2_meter_ctl, @@ -4516,7 +4522,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) return err; /* the rest of the configuration is for devices with a mixer */ - if (info->config_set == SCARLETT2_CONFIG_SET_GEN_3A) + if (!scarlett2_has_mixer(private)) return 0; if (scarlett2_has_config_item(private, From c0a7e1d859e76b121afe177f1ca04e121ecbb27d Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:52:47 +1030 Subject: [PATCH 282/337] ALSA: scarlett2: Add config set struct Add struct scarlett2_config_set so that data which is common to all devices in a config set can be stored there rather than in the model-specific data. Accordingly, rename scarlett2_config_items[] to scarlett2_config_sets[]. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/bfdb04cd6239af9a8c26a52da0537980f77c0437.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 167 +++++++++++++++++++----------------- 1 file changed, 89 insertions(+), 78 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 26bdd1beae2e..8be62413d17e 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1226,119 +1226,130 @@ struct scarlett2_config { u8 activate; }; -static const struct scarlett2_config - scarlett2_config_items[SCARLETT2_CONFIG_SET_COUNT] - [SCARLETT2_CONFIG_COUNT] = +struct scarlett2_config_set { + const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; +}; + +static const struct scarlett2_config_set + scarlett2_config_sets[SCARLETT2_CONFIG_SET_COUNT] = /* Gen 2 devices: 6i6, 18i8, 18i20 */ -{ { - [SCARLETT2_CONFIG_DIM_MUTE] = { - .offset = 0x31, .size = 8, .activate = 2 }, +{ [SCARLETT2_CONFIG_SET_GEN_2] = { + .items = { + [SCARLETT2_CONFIG_DIM_MUTE] = { + .offset = 0x31, .size = 8, .activate = 2 }, - [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { - .offset = 0x34, .size = 16, .activate = 1 }, + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, - [SCARLETT2_CONFIG_MUTE_SWITCH] = { - .offset = 0x5c, .size = 8, .activate = 1 }, + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, - [SCARLETT2_CONFIG_SW_HW_SWITCH] = { - .offset = 0x66, .size = 8, .activate = 3 }, + [SCARLETT2_CONFIG_SW_HW_SWITCH] = { + .offset = 0x66, .size = 8, .activate = 3 }, - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x7c, .size = 8, .activate = 7 }, + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, - [SCARLETT2_CONFIG_PAD_SWITCH] = { - .offset = 0x84, .size = 8, .activate = 8 }, + [SCARLETT2_CONFIG_PAD_SWITCH] = { + .offset = 0x84, .size = 8, .activate = 8 }, - [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { - .offset = 0x8d, .size = 8, .activate = 6 }, + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x8d, .size = 8, .activate = 6 }, + }, /* Gen 3 devices without a mixer (Solo and 2i2) */ -}, { - [SCARLETT2_CONFIG_MSD_SWITCH] = { - .offset = 0x04, .size = 8, .activate = 6 }, +}, [SCARLETT2_CONFIG_SET_GEN_3A] = { + .items = { + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x04, .size = 8, .activate = 6 }, - [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { - .offset = 0x05, .size = 8, .activate = 6 }, + [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { + .offset = 0x05, .size = 8, .activate = 6 }, - [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { - .offset = 0x06, .size = 8, .activate = 3 }, + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x06, .size = 8, .activate = 3 }, - [SCARLETT2_CONFIG_DIRECT_MONITOR] = { - .offset = 0x07, .size = 8, .activate = 4 }, + [SCARLETT2_CONFIG_DIRECT_MONITOR] = { + .offset = 0x07, .size = 8, .activate = 4 }, - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x08, .size = 1, .activate = 7 }, + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x08, .size = 1, .activate = 7 }, - [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x09, .size = 1, .activate = 8 }, + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x09, .size = 1, .activate = 8 }, + }, /* Gen 3 devices: 4i4, 8i6, 18i8, 18i20 */ -}, { - [SCARLETT2_CONFIG_DIM_MUTE] = { - .offset = 0x31, .size = 8, .activate = 2 }, +}, [SCARLETT2_CONFIG_SET_GEN_3B] = { + .items = { + [SCARLETT2_CONFIG_DIM_MUTE] = { + .offset = 0x31, .size = 8, .activate = 2 }, - [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { - .offset = 0x34, .size = 16, .activate = 1 }, + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, - [SCARLETT2_CONFIG_MUTE_SWITCH] = { - .offset = 0x5c, .size = 8, .activate = 1 }, + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, - [SCARLETT2_CONFIG_SW_HW_SWITCH] = { - .offset = 0x66, .size = 8, .activate = 3 }, + [SCARLETT2_CONFIG_SW_HW_SWITCH] = { + .offset = 0x66, .size = 8, .activate = 3 }, - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x7c, .size = 8, .activate = 7 }, + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, - [SCARLETT2_CONFIG_PAD_SWITCH] = { - .offset = 0x84, .size = 8, .activate = 8 }, + [SCARLETT2_CONFIG_PAD_SWITCH] = { + .offset = 0x84, .size = 8, .activate = 8 }, - [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x8c, .size = 8, .activate = 8 }, + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x8c, .size = 8, .activate = 8 }, - [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { - .offset = 0x95, .size = 8, .activate = 6 }, + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x95, .size = 8, .activate = 6 }, - [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { - .offset = 0x9c, .size = 1, .activate = 8 }, + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x9c, .size = 1, .activate = 8 }, - [SCARLETT2_CONFIG_MSD_SWITCH] = { - .offset = 0x9d, .size = 8, .activate = 6 }, + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x9d, .size = 8, .activate = 6 }, - [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { - .offset = 0x9e, .size = 8, .activate = 6 }, + [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { + .offset = 0x9e, .size = 8, .activate = 6 }, - [SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH] = { - .offset = 0x9f, .size = 1, .activate = 10 }, + [SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH] = { + .offset = 0x9f, .size = 1, .activate = 10 }, - [SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE] = { - .offset = 0xa0, .size = 1, .activate = 10 }, + [SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE] = { + .offset = 0xa0, .size = 1, .activate = 10 }, - [SCARLETT2_CONFIG_TALKBACK_MAP] = { - .offset = 0xb0, .size = 16, .activate = 10 }, + [SCARLETT2_CONFIG_TALKBACK_MAP] = { + .offset = 0xb0, .size = 16, .activate = 10 }, + }, /* Clarett USB and Clarett+ devices: 2Pre, 4Pre, 8Pre */ -}, { - [SCARLETT2_CONFIG_DIM_MUTE] = { - .offset = 0x31, .size = 8, .activate = 2 }, +}, [SCARLETT2_CONFIG_SET_CLARETT] = { + .items = { + [SCARLETT2_CONFIG_DIM_MUTE] = { + .offset = 0x31, .size = 8, .activate = 2 }, - [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { - .offset = 0x34, .size = 16, .activate = 1 }, + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, - [SCARLETT2_CONFIG_MUTE_SWITCH] = { - .offset = 0x5c, .size = 8, .activate = 1 }, + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, - [SCARLETT2_CONFIG_SW_HW_SWITCH] = { - .offset = 0x66, .size = 8, .activate = 3 }, + [SCARLETT2_CONFIG_SW_HW_SWITCH] = { + .offset = 0x66, .size = 8, .activate = 3 }, - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x7c, .size = 8, .activate = 7 }, + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, - [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x95, .size = 8, .activate = 8 }, + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x95, .size = 8, .activate = 8 }, - [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { - .offset = 0x8d, .size = 8, .activate = 6 }, + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x8d, .size = 8, .activate = 6 }, + } } }; /* proprietary request/response format */ @@ -1514,7 +1525,7 @@ static int scarlett2_has_config_item( { const struct scarlett2_device_info *info = private->info; const struct scarlett2_config *config_item = - &scarlett2_config_items[info->config_set][config_item_num]; + &scarlett2_config_sets[info->config_set].items[config_item_num]; return !!config_item->offset; } @@ -1527,7 +1538,7 @@ static int scarlett2_usb_get_config( struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_config *config_item = - &scarlett2_config_items[info->config_set][config_item_num]; + &scarlett2_config_sets[info->config_set].items[config_item_num]; int size, err, i; u8 *buf_8; u8 value; @@ -1589,7 +1600,7 @@ static int scarlett2_usb_set_config( struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const struct scarlett2_config *config_item = - &scarlett2_config_items[info->config_set][config_item_num]; + &scarlett2_config_sets[info->config_set].items[config_item_num]; struct { __le32 offset; __le32 bytes; From cbd6f148aa555c1383bef27dd003e1a2f1670e82 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:53:40 +1030 Subject: [PATCH 283/337] ALSA: scarlett2: Remove scarlett2_config_sets array Replace array index into config sets with a pointer to a config set. Copy the config_set pointer to the scarlett2_data struct. This simplifies both the definition and use of the config sets. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/61f69519fb6fbb677e066891a3a6771aeeec106d.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 361 +++++++++++++++++------------------- 1 file changed, 173 insertions(+), 188 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 8be62413d17e..74bcecbd6923 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -219,17 +219,6 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { /* Maximum number of meters (sum of output port counts) */ #define SCARLETT2_MAX_METERS 65 -/* There are different sets of configuration parameters across the - * devices, dependent on series and model. - */ -enum { - SCARLETT2_CONFIG_SET_GEN_2, - SCARLETT2_CONFIG_SET_GEN_3A, - SCARLETT2_CONFIG_SET_GEN_3B, - SCARLETT2_CONFIG_SET_CLARETT, - SCARLETT2_CONFIG_SET_COUNT -}; - /* Hardware port types: * - None (no input to mux) * - Analogue I/O @@ -274,6 +263,161 @@ static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { "Mute Playback Switch", "Dim Playback Switch" }; +/* Configuration parameters that can be read and written */ +enum { + SCARLETT2_CONFIG_DIM_MUTE, + SCARLETT2_CONFIG_LINE_OUT_VOLUME, + SCARLETT2_CONFIG_MUTE_SWITCH, + SCARLETT2_CONFIG_SW_HW_SWITCH, + SCARLETT2_CONFIG_LEVEL_SWITCH, + SCARLETT2_CONFIG_PAD_SWITCH, + SCARLETT2_CONFIG_MSD_SWITCH, + SCARLETT2_CONFIG_AIR_SWITCH, + SCARLETT2_CONFIG_STANDALONE_SWITCH, + SCARLETT2_CONFIG_PHANTOM_SWITCH, + SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, + SCARLETT2_CONFIG_DIRECT_MONITOR, + SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH, + SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE, + SCARLETT2_CONFIG_TALKBACK_MAP, + SCARLETT2_CONFIG_COUNT +}; + +/* Location, size, and activation command number for the configuration + * parameters. Size is in bits and may be 1, 8, or 16. + */ +struct scarlett2_config { + u8 offset; + u8 size; + u8 activate; +}; + +struct scarlett2_config_set { + const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; +}; + +/* Gen 2 devices: 6i6, 18i8, 18i20 */ +static const struct scarlett2_config_set scarlett2_config_set_gen2 = { + .items = { + [SCARLETT2_CONFIG_DIM_MUTE] = { + .offset = 0x31, .size = 8, .activate = 2 }, + + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, + + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, + + [SCARLETT2_CONFIG_SW_HW_SWITCH] = { + .offset = 0x66, .size = 8, .activate = 3 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, + + [SCARLETT2_CONFIG_PAD_SWITCH] = { + .offset = 0x84, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x8d, .size = 8, .activate = 6 }, + } +}; + +/* Gen 3 devices without a mixer (Solo and 2i2) */ +static const struct scarlett2_config_set scarlett2_config_set_gen3a = { + .items = { + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x04, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { + .offset = 0x05, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x06, .size = 8, .activate = 3 }, + + [SCARLETT2_CONFIG_DIRECT_MONITOR] = { + .offset = 0x07, .size = 8, .activate = 4 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x08, .size = 1, .activate = 7 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x09, .size = 1, .activate = 8 }, + } +}; + +/* Gen 3 devices: 4i4, 8i6, 18i8, 18i20 */ +static const struct scarlett2_config_set scarlett2_config_set_gen3b = { + .items = { + [SCARLETT2_CONFIG_DIM_MUTE] = { + .offset = 0x31, .size = 8, .activate = 2 }, + + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, + + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, + + [SCARLETT2_CONFIG_SW_HW_SWITCH] = { + .offset = 0x66, .size = 8, .activate = 3 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, + + [SCARLETT2_CONFIG_PAD_SWITCH] = { + .offset = 0x84, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x8c, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x95, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x9c, .size = 1, .activate = 8 }, + + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x9d, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { + .offset = 0x9e, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH] = { + .offset = 0x9f, .size = 1, .activate = 10 }, + + [SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE] = { + .offset = 0xa0, .size = 1, .activate = 10 }, + + [SCARLETT2_CONFIG_TALKBACK_MAP] = { + .offset = 0xb0, .size = 16, .activate = 10 }, + } +}; + +/* Clarett USB and Clarett+ devices: 2Pre, 4Pre, 8Pre */ +static const struct scarlett2_config_set scarlett2_config_set_clarett = { + .items = { + [SCARLETT2_CONFIG_DIM_MUTE] = { + .offset = 0x31, .size = 8, .activate = 2 }, + + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, + + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, + + [SCARLETT2_CONFIG_SW_HW_SWITCH] = { + .offset = 0x66, .size = 8, .activate = 3 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x95, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x8d, .size = 8, .activate = 6 }, + } +}; + /* Description of each hardware port type: * - id: hardware ID of this port type * - src_descr: printf format string for mux input selections @@ -356,7 +500,7 @@ struct scarlett2_meter_entry { struct scarlett2_device_info { /* which set of configuration parameters the device uses */ - u8 config_set; + const struct scarlett2_config_set *config_set; /* line out hw volume is sw controlled */ u8 line_out_hw_vol; @@ -429,6 +573,7 @@ struct scarlett2_data { u8 flash_write_state; struct delayed_work work; const struct scarlett2_device_info *info; + const struct scarlett2_config_set *config_set; const char *series_name; __u8 bInterfaceNumber; __u8 bEndpointAddress; @@ -485,7 +630,7 @@ struct scarlett2_data { /*** Model-specific data ***/ static const struct scarlett2_device_info s6i6_gen2_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_2, + .config_set = &scarlett2_config_set_gen2, .level_input_count = 2, .pad_input_count = 2, @@ -535,7 +680,7 @@ static const struct scarlett2_device_info s6i6_gen2_info = { }; static const struct scarlett2_device_info s18i8_gen2_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_2, + .config_set = &scarlett2_config_set_gen2, .level_input_count = 2, .pad_input_count = 4, @@ -588,7 +733,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }; static const struct scarlett2_device_info s18i20_gen2_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_2, + .config_set = &scarlett2_config_set_gen2, .line_out_hw_vol = 1, .line_out_descrs = { @@ -646,7 +791,7 @@ static const struct scarlett2_device_info s18i20_gen2_info = { }; static const struct scarlett2_device_info solo_gen3_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_3A, + .config_set = &scarlett2_config_set_gen3a, .level_input_count = 1, .level_input_first = 1, .air_input_count = 1, @@ -656,7 +801,7 @@ static const struct scarlett2_device_info solo_gen3_info = { }; static const struct scarlett2_device_info s2i2_gen3_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_3A, + .config_set = &scarlett2_config_set_gen3a, .level_input_count = 2, .air_input_count = 2, .phantom_count = 1, @@ -665,7 +810,7 @@ static const struct scarlett2_device_info s2i2_gen3_info = { }; static const struct scarlett2_device_info s4i4_gen3_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_3B, + .config_set = &scarlett2_config_set_gen3b, .level_input_count = 2, .pad_input_count = 2, .air_input_count = 2, @@ -714,7 +859,7 @@ static const struct scarlett2_device_info s4i4_gen3_info = { }; static const struct scarlett2_device_info s8i6_gen3_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_3B, + .config_set = &scarlett2_config_set_gen3b, .level_input_count = 2, .pad_input_count = 2, .air_input_count = 2, @@ -772,7 +917,7 @@ static const struct scarlett2_device_info s8i6_gen3_info = { }; static const struct scarlett2_device_info s18i8_gen3_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_3B, + .config_set = &scarlett2_config_set_gen3b, .line_out_hw_vol = 1, .has_speaker_switching = 1, .level_input_count = 2, @@ -852,7 +997,7 @@ static const struct scarlett2_device_info s18i8_gen3_info = { }; static const struct scarlett2_device_info s18i20_gen3_info = { - .config_set = SCARLETT2_CONFIG_SET_GEN_3B, + .config_set = &scarlett2_config_set_gen3b, .line_out_hw_vol = 1, .has_speaker_switching = 1, .has_talkback = 1, @@ -923,7 +1068,7 @@ static const struct scarlett2_device_info s18i20_gen3_info = { }; static const struct scarlett2_device_info clarett_2pre_info = { - .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .config_set = &scarlett2_config_set_clarett, .line_out_hw_vol = 1, .level_input_count = 2, .air_input_count = 2, @@ -971,7 +1116,7 @@ static const struct scarlett2_device_info clarett_2pre_info = { }; static const struct scarlett2_device_info clarett_4pre_info = { - .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .config_set = &scarlett2_config_set_clarett, .line_out_hw_vol = 1, .level_input_count = 2, .air_input_count = 4, @@ -1024,7 +1169,7 @@ static const struct scarlett2_device_info clarett_4pre_info = { }; static const struct scarlett2_device_info clarett_8pre_info = { - .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .config_set = &scarlett2_config_set_clarett, .line_out_hw_vol = 1, .level_input_count = 2, .air_input_count = 8, @@ -1197,161 +1342,6 @@ struct scarlett2_usb_volume_status { s16 master_vol; } __packed; -/* Configuration parameters that can be read and written */ -enum { - SCARLETT2_CONFIG_DIM_MUTE, - SCARLETT2_CONFIG_LINE_OUT_VOLUME, - SCARLETT2_CONFIG_MUTE_SWITCH, - SCARLETT2_CONFIG_SW_HW_SWITCH, - SCARLETT2_CONFIG_LEVEL_SWITCH, - SCARLETT2_CONFIG_PAD_SWITCH, - SCARLETT2_CONFIG_MSD_SWITCH, - SCARLETT2_CONFIG_AIR_SWITCH, - SCARLETT2_CONFIG_STANDALONE_SWITCH, - SCARLETT2_CONFIG_PHANTOM_SWITCH, - SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, - SCARLETT2_CONFIG_DIRECT_MONITOR, - SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH, - SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE, - SCARLETT2_CONFIG_TALKBACK_MAP, - SCARLETT2_CONFIG_COUNT -}; - -/* Location, size, and activation command number for the configuration - * parameters. Size is in bits and may be 1, 8, or 16. - */ -struct scarlett2_config { - u8 offset; - u8 size; - u8 activate; -}; - -struct scarlett2_config_set { - const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; -}; - -static const struct scarlett2_config_set - scarlett2_config_sets[SCARLETT2_CONFIG_SET_COUNT] = - -/* Gen 2 devices: 6i6, 18i8, 18i20 */ -{ [SCARLETT2_CONFIG_SET_GEN_2] = { - .items = { - [SCARLETT2_CONFIG_DIM_MUTE] = { - .offset = 0x31, .size = 8, .activate = 2 }, - - [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { - .offset = 0x34, .size = 16, .activate = 1 }, - - [SCARLETT2_CONFIG_MUTE_SWITCH] = { - .offset = 0x5c, .size = 8, .activate = 1 }, - - [SCARLETT2_CONFIG_SW_HW_SWITCH] = { - .offset = 0x66, .size = 8, .activate = 3 }, - - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x7c, .size = 8, .activate = 7 }, - - [SCARLETT2_CONFIG_PAD_SWITCH] = { - .offset = 0x84, .size = 8, .activate = 8 }, - - [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { - .offset = 0x8d, .size = 8, .activate = 6 }, - }, - -/* Gen 3 devices without a mixer (Solo and 2i2) */ -}, [SCARLETT2_CONFIG_SET_GEN_3A] = { - .items = { - [SCARLETT2_CONFIG_MSD_SWITCH] = { - .offset = 0x04, .size = 8, .activate = 6 }, - - [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { - .offset = 0x05, .size = 8, .activate = 6 }, - - [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { - .offset = 0x06, .size = 8, .activate = 3 }, - - [SCARLETT2_CONFIG_DIRECT_MONITOR] = { - .offset = 0x07, .size = 8, .activate = 4 }, - - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x08, .size = 1, .activate = 7 }, - - [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x09, .size = 1, .activate = 8 }, - }, - -/* Gen 3 devices: 4i4, 8i6, 18i8, 18i20 */ -}, [SCARLETT2_CONFIG_SET_GEN_3B] = { - .items = { - [SCARLETT2_CONFIG_DIM_MUTE] = { - .offset = 0x31, .size = 8, .activate = 2 }, - - [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { - .offset = 0x34, .size = 16, .activate = 1 }, - - [SCARLETT2_CONFIG_MUTE_SWITCH] = { - .offset = 0x5c, .size = 8, .activate = 1 }, - - [SCARLETT2_CONFIG_SW_HW_SWITCH] = { - .offset = 0x66, .size = 8, .activate = 3 }, - - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x7c, .size = 8, .activate = 7 }, - - [SCARLETT2_CONFIG_PAD_SWITCH] = { - .offset = 0x84, .size = 8, .activate = 8 }, - - [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x8c, .size = 8, .activate = 8 }, - - [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { - .offset = 0x95, .size = 8, .activate = 6 }, - - [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { - .offset = 0x9c, .size = 1, .activate = 8 }, - - [SCARLETT2_CONFIG_MSD_SWITCH] = { - .offset = 0x9d, .size = 8, .activate = 6 }, - - [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { - .offset = 0x9e, .size = 8, .activate = 6 }, - - [SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH] = { - .offset = 0x9f, .size = 1, .activate = 10 }, - - [SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE] = { - .offset = 0xa0, .size = 1, .activate = 10 }, - - [SCARLETT2_CONFIG_TALKBACK_MAP] = { - .offset = 0xb0, .size = 16, .activate = 10 }, - }, - -/* Clarett USB and Clarett+ devices: 2Pre, 4Pre, 8Pre */ -}, [SCARLETT2_CONFIG_SET_CLARETT] = { - .items = { - [SCARLETT2_CONFIG_DIM_MUTE] = { - .offset = 0x31, .size = 8, .activate = 2 }, - - [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { - .offset = 0x34, .size = 16, .activate = 1 }, - - [SCARLETT2_CONFIG_MUTE_SWITCH] = { - .offset = 0x5c, .size = 8, .activate = 1 }, - - [SCARLETT2_CONFIG_SW_HW_SWITCH] = { - .offset = 0x66, .size = 8, .activate = 3 }, - - [SCARLETT2_CONFIG_LEVEL_SWITCH] = { - .offset = 0x7c, .size = 8, .activate = 7 }, - - [SCARLETT2_CONFIG_AIR_SWITCH] = { - .offset = 0x95, .size = 8, .activate = 8 }, - - [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { - .offset = 0x8d, .size = 8, .activate = 6 }, - } -} }; - /* proprietary request/response format */ struct scarlett2_usb_packet { __le32 cmd; @@ -1523,11 +1513,7 @@ static int scarlett2_usb_get( static int scarlett2_has_config_item( struct scarlett2_data *private, int config_item_num) { - const struct scarlett2_device_info *info = private->info; - const struct scarlett2_config *config_item = - &scarlett2_config_sets[info->config_set].items[config_item_num]; - - return !!config_item->offset; + return !!private->config_set->items[config_item_num].offset; } /* Send a USB message to get configuration parameters; result placed in *buf */ @@ -1536,9 +1522,8 @@ static int scarlett2_usb_get_config( int config_item_num, int count, void *buf) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; const struct scarlett2_config *config_item = - &scarlett2_config_sets[info->config_set].items[config_item_num]; + &private->config_set->items[config_item_num]; int size, err, i; u8 *buf_8; u8 value; @@ -1598,9 +1583,8 @@ static int scarlett2_usb_set_config( int config_item_num, int index, int value) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; const struct scarlett2_config *config_item = - &scarlett2_config_sets[info->config_set].items[config_item_num]; + &private->config_set->items[config_item_num]; struct { __le32 offset; __le32 bytes; @@ -4365,6 +4349,7 @@ static int scarlett2_init_private(struct usb_mixer_interface *mixer, mixer->private_suspend = scarlett2_private_suspend; private->info = entry->info; + private->config_set = entry->info->config_set; private->series_name = entry->series_name; scarlett2_count_mux_io(private); private->scarlett2_seq = 0; From 43222a612374facb3408300ea1a789cc1b33fb9b Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:55:10 +1030 Subject: [PATCH 284/337] ALSA: scarlett2: Add check for config_item presence Update scarlett2_usb_get_config() and scarlett2_usb_set_config() to make sure that the config_item_num is valid for the device. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/b0572b23291ffd1b208f21d298adaf4d9f1fe4bc.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 74bcecbd6923..ad92c3d1f8f5 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1528,6 +1528,12 @@ static int scarlett2_usb_get_config( u8 *buf_8; u8 value; + /* Check that the configuration item is present in the + * configuration set used by this device + */ + if (!config_item->offset) + return -EFAULT; + /* For byte-sized parameters, retrieve directly into buf */ if (config_item->size >= 8) { size = config_item->size / 8 * count; @@ -1594,6 +1600,12 @@ static int scarlett2_usb_set_config( int offset, size; int err; + /* Check that the configuration item is present in the + * configuration set used by this device + */ + if (!config_item->offset) + return -EFAULT; + /* Cancel any pending NVRAM save */ cancel_delayed_work_sync(&private->work); From 7f4d8dbea2156802cc4cef74faa26eba8a7aa7d6 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:56:43 +1030 Subject: [PATCH 285/337] ALSA: scarlett2: Refactor scarlett2_usb_set_config() Pull out common code from scarlett2_usb_set_config() and create scarlett2_usb_set_data() and scarlett2_usb_activate_config(). Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/257eca0b07708339133f916930e388057d116eb8.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 57 ++++++++++++++++++++++++++----------- 1 file changed, 41 insertions(+), 16 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index ad92c3d1f8f5..9284c6edd4a4 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1583,7 +1583,45 @@ static void scarlett2_config_save_work(struct work_struct *work) scarlett2_config_save(private->mixer); } -/* Send a USB message to set a SCARLETT2_CONFIG_* parameter */ +/* Send a SCARLETT2_USB_SET_DATA command. + * offset: location in the device's data space + * size: size in bytes of the value (1, 2, 4) + */ +static int scarlett2_usb_set_data( + struct usb_mixer_interface *mixer, + int offset, int size, int value) +{ + struct scarlett2_data *private = mixer->private_data; + struct { + __le32 offset; + __le32 size; + __le32 value; + } __packed req; + + req.offset = cpu_to_le32(offset); + req.size = cpu_to_le32(size); + req.value = cpu_to_le32(value); + return scarlett2_usb(private->mixer, SCARLETT2_USB_SET_DATA, + &req, sizeof(u32) * 2 + size, NULL, 0); +} + +/* Send a SCARLETT2_USB_DATA_CMD command. + * Configuration changes require activation with this after they have + * been uploaded by a previous SCARLETT2_USB_SET_DATA. + * The value for activate needed is determined by the configuration + * item. + */ +static int scarlett2_usb_activate_config( + struct usb_mixer_interface *mixer, int activate) +{ + __le32 req; + + req = cpu_to_le32(activate); + return scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, + &req, sizeof(req), NULL, 0); +} + +/* Send USB messages to set a SCARLETT2_CONFIG_* parameter */ static int scarlett2_usb_set_config( struct usb_mixer_interface *mixer, int config_item_num, int index, int value) @@ -1591,12 +1629,6 @@ static int scarlett2_usb_set_config( struct scarlett2_data *private = mixer->private_data; const struct scarlett2_config *config_item = &private->config_set->items[config_item_num]; - struct { - __le32 offset; - __le32 bytes; - __le32 value; - } __packed req; - __le32 req2; int offset, size; int err; @@ -1638,19 +1670,12 @@ static int scarlett2_usb_set_config( } /* Send the configuration parameter data */ - req.offset = cpu_to_le32(offset); - req.bytes = cpu_to_le32(size); - req.value = cpu_to_le32(value); - err = scarlett2_usb(mixer, SCARLETT2_USB_SET_DATA, - &req, sizeof(u32) * 2 + size, - NULL, 0); + err = scarlett2_usb_set_data(mixer, offset, size, value); if (err < 0) return err; /* Activate the change */ - req2 = cpu_to_le32(config_item->activate); - err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, - &req2, sizeof(req2), NULL, 0); + err = scarlett2_usb_activate_config(mixer, config_item->activate); if (err < 0) return err; From 9c2ea88e9e3b00705f519b112683a75e9b6eba1d Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:57:16 +1030 Subject: [PATCH 286/337] ALSA: scarlett2: Refactor scarlett2_config_save() Use the new scarlett2_usb_activate_config() helper function rather than preparing the request manually and calling scarlett2_usb(). Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/bbc733dc081f311fb3167e81b15cd76324aa6307.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 40 ++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 21 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 9284c6edd4a4..2b2e6baeac20 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1562,27 +1562,6 @@ static int scarlett2_usb_get_config( return 0; } -/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ -static void scarlett2_config_save(struct usb_mixer_interface *mixer) -{ - __le32 req = cpu_to_le32(SCARLETT2_USB_CONFIG_SAVE); - - int err = scarlett2_usb(mixer, SCARLETT2_USB_DATA_CMD, - &req, sizeof(u32), - NULL, 0); - if (err < 0) - usb_audio_err(mixer->chip, "config save failed: %d\n", err); -} - -/* Delayed work to save config */ -static void scarlett2_config_save_work(struct work_struct *work) -{ - struct scarlett2_data *private = - container_of(work, struct scarlett2_data, work.work); - - scarlett2_config_save(private->mixer); -} - /* Send a SCARLETT2_USB_SET_DATA command. * offset: location in the device's data space * size: size in bytes of the value (1, 2, 4) @@ -1686,6 +1665,25 @@ static int scarlett2_usb_set_config( return 0; } +/* Send SCARLETT2_USB_DATA_CMD SCARLETT2_USB_CONFIG_SAVE */ +static void scarlett2_config_save(struct usb_mixer_interface *mixer) +{ + int err; + + err = scarlett2_usb_activate_config(mixer, SCARLETT2_USB_CONFIG_SAVE); + if (err < 0) + usb_audio_err(mixer->chip, "config save failed: %d\n", err); +} + +/* Delayed work to save config */ +static void scarlett2_config_save_work(struct work_struct *work) +{ + struct scarlett2_data *private = + container_of(work, struct scarlett2_data, work.work); + + scarlett2_config_save(private->mixer); +} + /* Send a USB message to get sync status; result placed in *sync */ static int scarlett2_usb_get_sync_status( struct usb_mixer_interface *mixer, From b5fe6c47a55f82f16e60a90528ada8bcc3558984 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:57:34 +1030 Subject: [PATCH 287/337] ALSA: scarlett2: Formatting fixes Add missing blank line before comment. For consistency with other functions that have few parameters, move the parameters onto the same line as the function name. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/72be568b02eea12621b0c4a96f8e8cc65b0c13c0.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 22 ++++++++-------------- 1 file changed, 8 insertions(+), 14 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 2b2e6baeac20..5bc60cded5f6 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2149,6 +2149,7 @@ static int scarlett2_add_firmware_version_ctl( return scarlett2_add_new_ctl(mixer, &scarlett2_firmware_version_ctl, 0, 0, "Firmware Version", NULL); } + /*** Sync Control ***/ /* Update sync control after receiving notification that the status @@ -3373,8 +3374,7 @@ static const struct snd_kcontrol_new scarlett2_speaker_switch_enum_ctl = { .put = scarlett2_speaker_switch_enum_ctl_put, }; -static int scarlett2_add_speaker_switch_ctl( - struct usb_mixer_interface *mixer) +static int scarlett2_add_speaker_switch_ctl(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; @@ -3542,8 +3542,7 @@ static const struct snd_kcontrol_new scarlett2_talkback_map_ctl = { .put = scarlett2_talkback_map_ctl_put, }; -static int scarlett2_add_talkback_ctls( - struct usb_mixer_interface *mixer) +static int scarlett2_add_talkback_ctls(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; @@ -4611,8 +4610,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) } /* Notify on sync change */ -static void scarlett2_notify_sync( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_sync(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; @@ -4623,8 +4621,7 @@ static void scarlett2_notify_sync( } /* Notify on monitor change */ -static void scarlett2_notify_monitor( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -4650,8 +4647,7 @@ static void scarlett2_notify_monitor( } /* Notify on dim/mute change */ -static void scarlett2_notify_dim_mute( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -4677,8 +4673,7 @@ static void scarlett2_notify_dim_mute( } /* Notify on "input other" change (level/pad/air) */ -static void scarlett2_notify_input_other( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -4704,8 +4699,7 @@ static void scarlett2_notify_input_other( /* Notify on "monitor other" change (direct monitor, speaker * switching, talkback) */ -static void scarlett2_notify_monitor_other( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; From 648bd468b28c37a8c84050953627011eacad84f5 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:57:52 +1030 Subject: [PATCH 288/337] ALSA: scarlett2: Parameterise notifications The notification values were previously #define'd, and checked with a series of if() statements calling functions. Replace with an array of masks/callback function pointers, and a pointer to that array in the scarlett2_config_set definitions. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/0ee2a3786f9d30c89eeae59d7e933424e8f39162.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 62 +++++++++++++++++++++++++++---------- 1 file changed, 45 insertions(+), 17 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 5bc60cded5f6..10f383b5f808 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -263,6 +263,29 @@ static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { "Mute Playback Switch", "Dim Playback Switch" }; +/* Notification callback functions */ +struct scarlett2_notification { + u32 mask; + void (*func)(struct usb_mixer_interface *mixer); +}; + +static void scarlett2_notify_sync(struct usb_mixer_interface *mixer); +static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer); +static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer); +static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer); + +/* Array of notification callback functions */ +static const struct scarlett2_notification scarlett2_notifications[] = { + { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000008, scarlett2_notify_sync }, + { 0x00200000, scarlett2_notify_dim_mute }, + { 0x00400000, scarlett2_notify_monitor }, + { 0x00800000, scarlett2_notify_input_other }, + { 0x01000000, scarlett2_notify_monitor_other }, + { 0, NULL } +}; + /* Configuration parameters that can be read and written */ enum { SCARLETT2_CONFIG_DIM_MUTE, @@ -293,11 +316,13 @@ struct scarlett2_config { }; struct scarlett2_config_set { + const struct scarlett2_notification *notifications; const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; }; /* Gen 2 devices: 6i6, 18i8, 18i20 */ static const struct scarlett2_config_set scarlett2_config_set_gen2 = { + .notifications = scarlett2_notifications, .items = { [SCARLETT2_CONFIG_DIM_MUTE] = { .offset = 0x31, .size = 8, .activate = 2 }, @@ -324,6 +349,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen2 = { /* Gen 3 devices without a mixer (Solo and 2i2) */ static const struct scarlett2_config_set scarlett2_config_set_gen3a = { + .notifications = scarlett2_notifications, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x04, .size = 8, .activate = 6 }, @@ -347,6 +373,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3a = { /* Gen 3 devices: 4i4, 8i6, 18i8, 18i20 */ static const struct scarlett2_config_set scarlett2_config_set_gen3b = { + .notifications = scarlett2_notifications, .items = { [SCARLETT2_CONFIG_DIM_MUTE] = { .offset = 0x31, .size = 8, .activate = 2 }, @@ -394,6 +421,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3b = { /* Clarett USB and Clarett+ devices: 2Pre, 4Pre, 8Pre */ static const struct scarlett2_config_set scarlett2_config_set_clarett = { + .notifications = scarlett2_notifications, .items = { [SCARLETT2_CONFIG_DIM_MUTE] = { .offset = 0x31, .size = 8, .activate = 2 }, @@ -1274,13 +1302,6 @@ static int scarlett2_get_port_start_num( /*** USB Interactions ***/ -/* Notifications from the interface */ -#define SCARLETT2_USB_NOTIFY_SYNC 0x00000008 -#define SCARLETT2_USB_NOTIFY_DIM_MUTE 0x00200000 -#define SCARLETT2_USB_NOTIFY_MONITOR 0x00400000 -#define SCARLETT2_USB_NOTIFY_INPUT_OTHER 0x00800000 -#define SCARLETT2_USB_NOTIFY_MONITOR_OTHER 0x01000000 - /* Commands for sending/receiving requests/responses */ #define SCARLETT2_USB_CMD_INIT 0 #define SCARLETT2_USB_CMD_REQ 2 @@ -4745,21 +4766,28 @@ static void scarlett2_notify(struct urb *urb) int len = urb->actual_length; int ustatus = urb->status; u32 data; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_notification *notifications = + private->config_set->notifications; if (ustatus != 0 || len != 8) goto requeue; data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); - if (data & SCARLETT2_USB_NOTIFY_SYNC) - scarlett2_notify_sync(mixer); - if (data & SCARLETT2_USB_NOTIFY_MONITOR) - scarlett2_notify_monitor(mixer); - if (data & SCARLETT2_USB_NOTIFY_DIM_MUTE) - scarlett2_notify_dim_mute(mixer); - if (data & SCARLETT2_USB_NOTIFY_INPUT_OTHER) - scarlett2_notify_input_other(mixer); - if (data & SCARLETT2_USB_NOTIFY_MONITOR_OTHER) - scarlett2_notify_monitor_other(mixer); + + while (data && notifications->mask) { + if (data & notifications->mask) { + data &= ~notifications->mask; + if (notifications->func) + notifications->func(mixer); + } + notifications++; + } + + if (data) + usb_audio_warn(mixer->chip, + "%s: Unhandled notification: 0x%08x\n", + __func__, data); requeue: if (ustatus != -ENOENT && From e5fab78cd8e867b2201eadabe4871100881e2a64 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:58:07 +1030 Subject: [PATCH 289/337] ALSA: scarlett2: Change num_mux_* from int to u8 num_mux_srcs and num_mux_dsts will fit into a u8, so change the type. More similar counts are coming soon. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/886fbd9ce7f06b13c6dbf36f64e6b2d107d16a83.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 10f383b5f808..e34b57e1acf0 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -607,8 +607,8 @@ struct scarlett2_data { __u8 bEndpointAddress; __u16 wMaxPacketSize; __u8 bInterval; - int num_mux_srcs; - int num_mux_dsts; + u8 num_mux_srcs; + u8 num_mux_dsts; u32 firmware_version; u8 flash_segment_nums[SCARLETT2_SEGMENT_ID_COUNT]; u8 flash_segment_blocks[SCARLETT2_SEGMENT_ID_COUNT]; From 42caae0e20323af7c5b41ac089798f5590b54845 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:58:38 +1030 Subject: [PATCH 290/337] ALSA: scarlett2: Refactor common port_count lookups Rather than looking up the analogue and mixer I/O counts repeatedly in info->port_count[SCARLETT2_PORT_TYPE_*][SCARLETT2_PORT_*], save those numbers in private variables. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/db0a5b56bdff476e2e31ad8e5ee15008314412b7.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 107 ++++++++++++------------------------ 1 file changed, 35 insertions(+), 72 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index e34b57e1acf0..713a2ff7ccf2 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -609,6 +609,9 @@ struct scarlett2_data { __u8 bInterval; u8 num_mux_srcs; u8 num_mux_dsts; + u8 num_mix_in; + u8 num_mix_out; + u8 num_line_out; u32 firmware_version; u8 flash_segment_nums[SCARLETT2_SEGMENT_ID_COUNT]; u8 flash_segment_blocks[SCARLETT2_SEGMENT_ID_COUNT]; @@ -1744,10 +1747,8 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, int mix_num) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - int num_mixer_in = - info->port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT]; + int num_mixer_in = private->num_mix_in; int err, i, j, k; struct { @@ -1787,7 +1788,6 @@ static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, int mix_num) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; struct { __le16 mix_num; @@ -1795,8 +1795,7 @@ static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, } __packed req; int i, j; - int num_mixer_in = - info->port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT]; + int num_mixer_in = private->num_mix_in; req.mix_num = cpu_to_le16(mix_num); @@ -1907,9 +1906,6 @@ static void scarlett2_usb_populate_mux(struct scarlett2_data *private, static void scarlett2_update_meter_level_map(struct scarlett2_data *private) { const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int line_out_count = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; const struct scarlett2_meter_entry *entry; /* sources already assigned to a destination @@ -1938,7 +1934,7 @@ static void scarlett2_update_meter_level_map(struct scarlett2_data *private) /* convert mux_idx using line_out_unmap[] */ int map_mux_idx = ( info->line_out_remap_enable && - mux_idx < line_out_count + mux_idx < private->num_line_out ) ? info->line_out_unmap[mux_idx] : mux_idx; @@ -2249,10 +2245,7 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; struct scarlett2_usb_volume_status volume_status; - int num_line_out = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; int err, i; int mute; @@ -2272,7 +2265,7 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) mute = private->dim_mute[SCARLETT2_BUTTON_MUTE]; - for (i = 0; i < num_line_out; i++) + for (i = 0; i < private->num_line_out; i++) if (private->vol_sw_hw_switch[i]) { private->vol[i] = private->master_vol; private->mute_switch[i] = mute; @@ -2324,14 +2317,11 @@ unlock: static int line_out_remap(struct scarlett2_data *private, int index) { const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int line_out_count = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; if (!info->line_out_remap_enable) return index; - if (index >= line_out_count) + if (index >= private->num_line_out) return index; return info->line_out_remap[index]; @@ -3104,10 +3094,6 @@ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer) private->speaker_switching_switch = monitor_other_switch[0] + 1; if (info->has_talkback) { - const int (*port_count)[SCARLETT2_PORT_DIRNS] = - info->port_count; - int num_mixes = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN]; u16 bitmap; int i; @@ -3121,7 +3107,7 @@ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer) 1, &bitmap); if (err < 0) return err; - for (i = 0; i < num_mixes; i++, bitmap >>= 1) + for (i = 0; i < private->num_mix_out; i++, bitmap >>= 1) private->talkback_map[i] = bitmap & 1; } @@ -3518,10 +3504,6 @@ static int scarlett2_talkback_map_ctl_put( struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = - private->info->port_count; - int num_mixes = port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN]; - int index = elem->control; int oval, val, err = 0, i; u16 bitmap = 0; @@ -3541,7 +3523,7 @@ static int scarlett2_talkback_map_ctl_put( private->talkback_map[index] = val; - for (i = 0; i < num_mixes; i++) + for (i = 0; i < private->num_mix_out; i++) bitmap |= private->talkback_map[i] << i; /* Send updated bitmap to the device */ @@ -3567,8 +3549,6 @@ static int scarlett2_add_talkback_ctls(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int num_mixes = port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN]; int err, i; char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -3582,7 +3562,7 @@ static int scarlett2_add_talkback_ctls(struct usb_mixer_interface *mixer) if (err < 0) return err; - for (i = 0; i < num_mixes; i++) { + for (i = 0; i < private->num_mix_out; i++) { snprintf(s, sizeof(s), "Talkback Mix %c Playback Switch", i + 'A'); err = scarlett2_add_new_ctl(mixer, &scarlett2_talkback_map_ctl, @@ -3629,11 +3609,6 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int num_line_out = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; - int index = elem->control; int oval, val, err = 0, i; @@ -3659,7 +3634,7 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, err = 1; if (index == SCARLETT2_BUTTON_MUTE) - for (i = 0; i < num_line_out; i++) { + for (i = 0; i < private->num_line_out; i++) { int line_index = line_out_remap(private, i); if (private->vol_sw_hw_switch[line_index]) { @@ -3689,9 +3664,6 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int num_line_out = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; int err, i; char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -3706,7 +3678,7 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) } /* Add volume controls */ - for (i = 0; i < num_line_out; i++) { + for (i = 0; i < private->num_line_out; i++) { int index = line_out_remap(private, i); /* Fader */ @@ -3883,9 +3855,7 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int oval, val, num_mixer_in, mix_num, err = 0; + int oval, val, mix_num, err = 0; int index = elem->control; mutex_lock(&private->data_mutex); @@ -3898,8 +3868,7 @@ static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, oval = private->mix[index]; val = clamp(ucontrol->value.integer.value[0], 0L, (long)SCARLETT2_MIXER_MAX_VALUE); - num_mixer_in = port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT]; - mix_num = index / num_mixer_in; + mix_num = index / private->num_mix_in; if (oval == val) goto unlock; @@ -3935,19 +3904,12 @@ static const struct snd_kcontrol_new scarlett2_mixer_ctl = { static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; int err, i, j; int index; char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - int num_inputs = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT]; - int num_outputs = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN]; - - for (i = 0, index = 0; i < num_outputs; i++) - for (j = 0; j < num_inputs; j++, index++) { + for (i = 0, index = 0; i < private->num_mix_out; i++) + for (j = 0; j < private->num_mix_in; j++, index++) { snprintf(s, sizeof(s), "Mix %c Input %02d Playback Volume", 'A' + i, j + 1); @@ -4336,12 +4298,13 @@ static void scarlett2_private_suspend(struct usb_mixer_interface *mixer) /*** Initialisation ***/ -static void scarlett2_count_mux_io(struct scarlett2_data *private) +static void scarlett2_count_io(struct scarlett2_data *private) { const struct scarlett2_device_info *info = private->info; const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; int port_type, srcs = 0, dsts = 0; + /* Count the number of mux sources and destinations */ for (port_type = 0; port_type < SCARLETT2_PORT_TYPE_COUNT; port_type++) { @@ -4351,6 +4314,17 @@ static void scarlett2_count_mux_io(struct scarlett2_data *private) private->num_mux_srcs = srcs; private->num_mux_dsts = dsts; + + /* Mixer inputs are mux outputs and vice versa */ + private->num_mix_in = + port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT]; + + private->num_mix_out = + port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN]; + + /* Number of analogue line outputs */ + private->num_line_out = + port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; } /* Look through the interface descriptors for the Focusrite Control @@ -4406,7 +4380,7 @@ static int scarlett2_init_private(struct usb_mixer_interface *mixer, private->info = entry->info; private->config_set = entry->info->config_set; private->series_name = entry->series_name; - scarlett2_count_mux_io(private); + scarlett2_count_io(private); private->scarlett2_seq = 0; private->mixer = mixer; @@ -4544,11 +4518,6 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int num_line_out = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; - int num_mixer_out = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN]; struct scarlett2_usb_volume_status volume_status; int err, i; @@ -4601,7 +4570,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) volume_status.master_vol + SCARLETT2_VOLUME_BIAS, 0, SCARLETT2_VOLUME_BIAS); - for (i = 0; i < num_line_out; i++) { + for (i = 0; i < private->num_line_out; i++) { int volume, mute; private->vol_sw_hw_switch[i] = @@ -4621,7 +4590,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) private->mute_switch[i] = mute; } - for (i = 0; i < num_mixer_out; i++) { + for (i = 0; i < private->num_mix_out; i++) { err = scarlett2_usb_get_mix(mixer, i); if (err < 0) return err; @@ -4647,9 +4616,6 @@ static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int num_line_out = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; int i; /* if line_out_hw_vol is 0, there are no controls to update */ @@ -4661,7 +4627,7 @@ static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &private->master_vol_ctl->id); - for (i = 0; i < num_line_out; i++) + for (i = 0; i < private->num_line_out; i++) if (private->vol_sw_hw_switch[line_out_remap(private, i)]) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->vol_ctls[i]->id); @@ -4673,9 +4639,6 @@ static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - const int (*port_count)[SCARLETT2_PORT_DIRNS] = info->port_count; - int num_line_out = - port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; int i; private->vol_updated = 1; @@ -4687,7 +4650,7 @@ static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->dim_mute_ctls[i]->id); - for (i = 0; i < num_line_out; i++) + for (i = 0; i < private->num_line_out; i++) if (private->vol_sw_hw_switch[line_out_remap(private, i)]) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->mute_ctls[i]->id); From 80c7933e74c3d7dfbaf994a340317a199a6a640c Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:58:49 +1030 Subject: [PATCH 291/337] ALSA: scarlett2: Remove struct scarlett2_usb_volume_status The struct scarlett2_usb_volume_status matched the config space layout of a few volume controls that could be read together and were in fixed locations between Gen 2 and Gen 3 devices. Gen 4 devices have removed, moved, and new related controls, so this needs to be cleaned up. By adding SCARLETT2_CONFIG_MASTER_VOLUME (the only config item that didn't already have its own entry, because it is read-only), we can remove: - struct scarlett2_usb_volume_state, - #define SCARLETT2_USB_VOLUME_STATUS_OFFSET, and - scarlett2_usb_get_volume_status() and replace with calls to scarlett2_usb_get_config(). Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/2ee88994857246bf89fab8e62ac279f3bcf96192.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 126 +++++++++++++++++------------------- 1 file changed, 59 insertions(+), 67 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 713a2ff7ccf2..d2e63c944433 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -292,6 +292,7 @@ enum { SCARLETT2_CONFIG_LINE_OUT_VOLUME, SCARLETT2_CONFIG_MUTE_SWITCH, SCARLETT2_CONFIG_SW_HW_SWITCH, + SCARLETT2_CONFIG_MASTER_VOLUME, SCARLETT2_CONFIG_LEVEL_SWITCH, SCARLETT2_CONFIG_PAD_SWITCH, SCARLETT2_CONFIG_MSD_SWITCH, @@ -336,6 +337,9 @@ static const struct scarlett2_config_set scarlett2_config_set_gen2 = { [SCARLETT2_CONFIG_SW_HW_SWITCH] = { .offset = 0x66, .size = 8, .activate = 3 }, + [SCARLETT2_CONFIG_MASTER_VOLUME] = { + .offset = 0x76, .size = 16 }, + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { .offset = 0x7c, .size = 8, .activate = 7 }, @@ -387,6 +391,9 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3b = { [SCARLETT2_CONFIG_SW_HW_SWITCH] = { .offset = 0x66, .size = 8, .activate = 3 }, + [SCARLETT2_CONFIG_MASTER_VOLUME] = { + .offset = 0x76, .size = 16 }, + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { .offset = 0x7c, .size = 8, .activate = 7 }, @@ -435,6 +442,9 @@ static const struct scarlett2_config_set scarlett2_config_set_clarett = { [SCARLETT2_CONFIG_SW_HW_SWITCH] = { .offset = 0x66, .size = 8, .activate = 3 }, + [SCARLETT2_CONFIG_MASTER_VOLUME] = { + .offset = 0x76, .size = 16 }, + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { .offset = 0x7c, .size = 8, .activate = 7 }, @@ -1330,7 +1340,6 @@ static int scarlett2_get_port_start_num( #define SCARLETT2_USB_CONFIG_SAVE 6 -#define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 #define SCARLETT2_USB_METER_LEVELS_GET_MAGIC 1 #define SCARLETT2_FLASH_BLOCK_SIZE 4096 @@ -1341,31 +1350,6 @@ static int scarlett2_get_port_start_num( #define SCARLETT2_SEGMENT_SETTINGS_NAME "App_Settings" #define SCARLETT2_SEGMENT_FIRMWARE_NAME "App_Upgrade" -/* volume status is read together (matches scarlett2_config_items[1]) */ -struct scarlett2_usb_volume_status { - /* dim/mute buttons */ - u8 dim_mute[SCARLETT2_DIM_MUTE_COUNT]; - - u8 pad1; - - /* software volume setting */ - s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; - - /* actual volume of output inc. dim (-18dB) */ - s16 hw_vol[SCARLETT2_ANALOGUE_MAX]; - - /* internal mute buttons */ - u8 mute_switch[SCARLETT2_ANALOGUE_MAX]; - - /* sw (0) or hw (1) controlled */ - u8 sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; - - u8 pad3[6]; - - /* front panel volume knob */ - s16 master_vol; -} __packed; - /* proprietary request/response format */ struct scarlett2_usb_packet { __le32 cmd; @@ -1725,15 +1709,6 @@ static int scarlett2_usb_get_sync_status( return 0; } -/* Send a USB message to get volume status; result placed in *buf */ -static int scarlett2_usb_get_volume_status( - struct usb_mixer_interface *mixer, - struct scarlett2_usb_volume_status *buf) -{ - return scarlett2_usb_get(mixer, SCARLETT2_USB_VOLUME_STATUS_OFFSET, - buf, sizeof(*buf)); -} - /* Return true if the device has a mixer that we can control */ static int scarlett2_has_mixer(struct scarlett2_data *private) { @@ -2245,23 +2220,32 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - struct scarlett2_usb_volume_status volume_status; + s16 vol; int err, i; int mute; private->vol_updated = 0; - err = scarlett2_usb_get_volume_status(mixer, &volume_status); + if (!info->line_out_hw_vol) + return 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_MASTER_VOLUME, + 1, &vol); if (err < 0) return err; - private->master_vol = clamp( - volume_status.master_vol + SCARLETT2_VOLUME_BIAS, - 0, SCARLETT2_VOLUME_BIAS); + private->master_vol = clamp(vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); - if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) - private->dim_mute[i] = !!volume_status.dim_mute[i]; + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_DIM_MUTE, + SCARLETT2_DIM_MUTE_COUNT, private->dim_mute); + if (err < 0) + return err; + + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) + private->dim_mute[i] = !!private->dim_mute[i]; mute = private->dim_mute[SCARLETT2_BUTTON_MUTE]; @@ -4518,8 +4502,8 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - struct scarlett2_usb_volume_status volume_status; int err, i; + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) { err = scarlett2_usb_get_config( @@ -4558,38 +4542,46 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; - err = scarlett2_usb_get_volume_status(mixer, &volume_status); + /* read SW line out volume */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + private->num_line_out, &sw_vol); if (err < 0) return err; - if (info->line_out_hw_vol) - for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) - private->dim_mute[i] = !!volume_status.dim_mute[i]; + for (i = 0; i < private->num_line_out; i++) + private->vol[i] = clamp( + sw_vol[i] + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); - private->master_vol = clamp( - volume_status.master_vol + SCARLETT2_VOLUME_BIAS, - 0, SCARLETT2_VOLUME_BIAS); + /* read SW mute */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_MUTE_SWITCH, + private->num_line_out, &private->mute_switch); + if (err < 0) + return err; - for (i = 0; i < private->num_line_out; i++) { - int volume, mute; + for (i = 0; i < private->num_line_out; i++) + private->mute_switch[i] = + !!private->mute_switch[i]; - private->vol_sw_hw_switch[i] = - info->line_out_hw_vol - && volume_status.sw_hw_switch[i]; + /* read SW/HW switches */ + if (info->line_out_hw_vol) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + private->num_line_out, &private->vol_sw_hw_switch); + if (err < 0) + return err; - volume = private->vol_sw_hw_switch[i] - ? volume_status.master_vol - : volume_status.sw_vol[i]; - volume = clamp(volume + SCARLETT2_VOLUME_BIAS, - 0, SCARLETT2_VOLUME_BIAS); - private->vol[i] = volume; - - mute = private->vol_sw_hw_switch[i] - ? private->dim_mute[SCARLETT2_BUTTON_MUTE] - : volume_status.mute_switch[i]; - private->mute_switch[i] = mute; + for (i = 0; i < private->num_line_out; i++) + private->vol_sw_hw_switch[i] = + !!private->vol_sw_hw_switch[i]; } + err = scarlett2_update_volumes(mixer); + if (err < 0) + return err; + for (i = 0; i < private->num_mix_out; i++) { err = scarlett2_usb_get_mix(mixer, i); if (err < 0) From e79aea579a19ebdc703868c5955136abd80bb1a9 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:59:14 +1030 Subject: [PATCH 292/337] ALSA: scarlett2: Split dim_mute_update from vol_updated Scarlett Gen 2 and Gen 3 devices combine volume and dim/mute notifications. The Scarlett 4i4 Gen 4 has volume change notification but no dim/mute control so split dim_mute_update out from vol_update. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/bf63f48bcc68ae739bd9948c8ee2f87ee7af22a2.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 63 ++++++++++++++++++++++++------------- 1 file changed, 42 insertions(+), 21 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index d2e63c944433..a72eb4bacaeb 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -628,6 +628,7 @@ struct scarlett2_data { u16 scarlett2_seq; u8 sync_updated; u8 vol_updated; + u8 dim_mute_updated; u8 input_other_updated; u8 monitor_other_updated; u8 mux_updated; @@ -2222,7 +2223,6 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) const struct scarlett2_device_info *info = private->info; s16 vol; int err, i; - int mute; private->vol_updated = 0; @@ -2238,22 +2238,9 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) private->master_vol = clamp(vol + SCARLETT2_VOLUME_BIAS, 0, SCARLETT2_VOLUME_BIAS); - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_DIM_MUTE, - SCARLETT2_DIM_MUTE_COUNT, private->dim_mute); - if (err < 0) - return err; - - for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) - private->dim_mute[i] = !!private->dim_mute[i]; - - mute = private->dim_mute[SCARLETT2_BUTTON_MUTE]; - for (i = 0; i < private->num_line_out; i++) - if (private->vol_sw_hw_switch[i]) { + if (private->vol_sw_hw_switch[i]) private->vol[i] = private->master_vol; - private->mute_switch[i] = mute; - } return 0; } @@ -2401,6 +2388,36 @@ static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { /*** Mute Switch Controls ***/ +static int scarlett2_update_dim_mute(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + u8 mute; + + private->dim_mute_updated = 0; + + if (!info->line_out_hw_vol) + return 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_DIM_MUTE, + SCARLETT2_DIM_MUTE_COUNT, private->dim_mute); + if (err < 0) + return err; + + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) + private->dim_mute[i] = !!private->dim_mute[i]; + + mute = private->dim_mute[SCARLETT2_BUTTON_MUTE]; + + for (i = 0; i < private->num_line_out; i++) + if (private->vol_sw_hw_switch[i]) + private->mute_switch[i] = mute; + + return 0; +} + static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -2417,8 +2434,8 @@ static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->vol_updated) { - err = scarlett2_update_volumes(mixer); + if (private->dim_mute_updated) { + err = scarlett2_update_dim_mute(mixer); if (err < 0) goto unlock; } @@ -3575,8 +3592,8 @@ static int scarlett2_dim_mute_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->vol_updated) { - err = scarlett2_update_volumes(mixer); + if (private->dim_mute_updated) { + err = scarlett2_update_dim_mute(mixer); if (err < 0) goto unlock; } @@ -4582,6 +4599,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; + err = scarlett2_update_dim_mute(mixer); + if (err < 0) + return err; + for (i = 0; i < private->num_mix_out; i++) { err = scarlett2_usb_get_mix(mixer, i); if (err < 0) @@ -4633,11 +4654,11 @@ static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) const struct scarlett2_device_info *info = private->info; int i; - private->vol_updated = 1; - if (!info->line_out_hw_vol) return; + private->dim_mute_updated = 1; + for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->dim_mute_ctls[i]->id); From c6b3e71e2c0899426dcdc46a778c6bd0c35925d1 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 05:59:26 +1030 Subject: [PATCH 293/337] ALSA: scarlett2: Remove line_out_hw_vol device info entry By splitting config set gen2 into gen2a/b (for 6i6/18i8 vs 18i20), and gen3b into gen3b/c (for 4i4/8i6 vs 18i8/18i20), we can use scarlett2_has_config_item() instead of the per-device line_out_hw_vol. As Gen 4 has a master volume control but no SW/HW switches, check for both SCARLETT2_CONFIG_MASTER_VOLUME and SCARLETT2_CONFIG_SW_HW_SWITCH as needed, even though for Gen 2 and Gen 3 the former implies the latter. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/307c4f8d6d2e034f3e386b51d72a39d77c8a9fce.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 145 ++++++++++++++++++++++++------------ 1 file changed, 96 insertions(+), 49 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a72eb4bacaeb..0d86c41432fc 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -321,8 +321,31 @@ struct scarlett2_config_set { const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; }; -/* Gen 2 devices: 6i6, 18i8, 18i20 */ -static const struct scarlett2_config_set scarlett2_config_set_gen2 = { +/* Gen 2 devices without SW/HW volume switch: 6i6, 18i8 */ + +static const struct scarlett2_config_set scarlett2_config_set_gen2a = { + .notifications = scarlett2_notifications, + .items = { + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, + + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, + + [SCARLETT2_CONFIG_PAD_SWITCH] = { + .offset = 0x84, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x8d, .size = 8, .activate = 6 }, + } +}; + +/* Gen 2 devices with SW/HW volume switch: 18i20 */ + +static const struct scarlett2_config_set scarlett2_config_set_gen2b = { .notifications = scarlett2_notifications, .items = { [SCARLETT2_CONFIG_DIM_MUTE] = { @@ -375,8 +398,41 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3a = { } }; -/* Gen 3 devices: 4i4, 8i6, 18i8, 18i20 */ +/* Gen 3 devices without SW/HW volume switch: 4i4, 8i6 */ static const struct scarlett2_config_set scarlett2_config_set_gen3b = { + .notifications = scarlett2_notifications, + .items = { + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, + + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, + + [SCARLETT2_CONFIG_PAD_SWITCH] = { + .offset = 0x84, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x8c, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x95, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x9c, .size = 1, .activate = 8 }, + + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x9d, .size = 8, .activate = 6 }, + + [SCARLETT2_CONFIG_PHANTOM_PERSISTENCE] = { + .offset = 0x9e, .size = 8, .activate = 6 }, + } +}; + +/* Gen 3 devices with SW/HW volume switch: 18i8, 18i20 */ +static const struct scarlett2_config_set scarlett2_config_set_gen3c = { .notifications = scarlett2_notifications, .items = { [SCARLETT2_CONFIG_DIM_MUTE] = { @@ -540,9 +596,6 @@ struct scarlett2_device_info { /* which set of configuration parameters the device uses */ const struct scarlett2_config_set *config_set; - /* line out hw volume is sw controlled */ - u8 line_out_hw_vol; - /* support for main/alt speaker switching */ u8 has_speaker_switching; @@ -672,7 +725,7 @@ struct scarlett2_data { /*** Model-specific data ***/ static const struct scarlett2_device_info s6i6_gen2_info = { - .config_set = &scarlett2_config_set_gen2, + .config_set = &scarlett2_config_set_gen2a, .level_input_count = 2, .pad_input_count = 2, @@ -722,7 +775,7 @@ static const struct scarlett2_device_info s6i6_gen2_info = { }; static const struct scarlett2_device_info s18i8_gen2_info = { - .config_set = &scarlett2_config_set_gen2, + .config_set = &scarlett2_config_set_gen2a, .level_input_count = 2, .pad_input_count = 4, @@ -775,8 +828,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }; static const struct scarlett2_device_info s18i20_gen2_info = { - .config_set = &scarlett2_config_set_gen2, - .line_out_hw_vol = 1, + .config_set = &scarlett2_config_set_gen2b, .line_out_descrs = { "Monitor L", @@ -959,8 +1011,7 @@ static const struct scarlett2_device_info s8i6_gen3_info = { }; static const struct scarlett2_device_info s18i8_gen3_info = { - .config_set = &scarlett2_config_set_gen3b, - .line_out_hw_vol = 1, + .config_set = &scarlett2_config_set_gen3c, .has_speaker_switching = 1, .level_input_count = 2, .pad_input_count = 4, @@ -1039,8 +1090,7 @@ static const struct scarlett2_device_info s18i8_gen3_info = { }; static const struct scarlett2_device_info s18i20_gen3_info = { - .config_set = &scarlett2_config_set_gen3b, - .line_out_hw_vol = 1, + .config_set = &scarlett2_config_set_gen3c, .has_speaker_switching = 1, .has_talkback = 1, .level_input_count = 2, @@ -1111,7 +1161,6 @@ static const struct scarlett2_device_info s18i20_gen3_info = { static const struct scarlett2_device_info clarett_2pre_info = { .config_set = &scarlett2_config_set_clarett, - .line_out_hw_vol = 1, .level_input_count = 2, .air_input_count = 2, @@ -1159,7 +1208,6 @@ static const struct scarlett2_device_info clarett_2pre_info = { static const struct scarlett2_device_info clarett_4pre_info = { .config_set = &scarlett2_config_set_clarett, - .line_out_hw_vol = 1, .level_input_count = 2, .air_input_count = 4, @@ -1212,7 +1260,6 @@ static const struct scarlett2_device_info clarett_4pre_info = { static const struct scarlett2_device_info clarett_8pre_info = { .config_set = &scarlett2_config_set_clarett, - .line_out_hw_vol = 1, .level_input_count = 2, .air_input_count = 8, @@ -2220,27 +2267,28 @@ static int scarlett2_add_sync_ctl(struct usb_mixer_interface *mixer) static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; s16 vol; int err, i; private->vol_updated = 0; - if (!info->line_out_hw_vol) - return 0; + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_MASTER_VOLUME)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_MASTER_VOLUME, + 1, &vol); + if (err < 0) + return err; - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_MASTER_VOLUME, - 1, &vol); - if (err < 0) - return err; + private->master_vol = clamp(vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); - private->master_vol = clamp(vol + SCARLETT2_VOLUME_BIAS, - 0, SCARLETT2_VOLUME_BIAS); - - for (i = 0; i < private->num_line_out; i++) - if (private->vol_sw_hw_switch[i]) - private->vol[i] = private->master_vol; + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_SW_HW_SWITCH)) + for (i = 0; i < private->num_line_out; i++) + if (private->vol_sw_hw_switch[i]) + private->vol[i] = private->master_vol; + } return 0; } @@ -2391,13 +2439,12 @@ static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { static int scarlett2_update_dim_mute(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; int err, i; u8 mute; private->dim_mute_updated = 0; - if (!info->line_out_hw_vol) + if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH)) return 0; err = scarlett2_usb_get_config( @@ -3669,7 +3716,8 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* Add R/O HW volume control */ - if (info->line_out_hw_vol) { + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_MASTER_VOLUME)) { snprintf(s, sizeof(s), "Master HW Playback Volume"); err = scarlett2_add_new_ctl(mixer, &scarlett2_master_volume_ctl, @@ -3708,14 +3756,16 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) if (err < 0) return err; - /* Make the fader and mute controls read-only if the - * SW/HW switch is set to HW - */ - if (private->vol_sw_hw_switch[index]) - scarlett2_vol_ctl_set_writable(mixer, i, 0); - /* SW/HW Switch */ - if (info->line_out_hw_vol) { + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_SW_HW_SWITCH)) { + + /* Make the fader and mute controls read-only if the + * SW/HW switch is set to HW + */ + if (private->vol_sw_hw_switch[index]) + scarlett2_vol_ctl_set_writable(mixer, i, 0); + snprintf(s, sizeof(s), "Line Out %02d Volume Control Playback Enum", i + 1); @@ -3735,7 +3785,7 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) } /* Add dim/mute controls */ - if (info->line_out_hw_vol) + if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_DIM_MUTE)) for (i = 0; i < SCARLETT2_DIM_MUTE_COUNT; i++) { err = scarlett2_add_new_ctl( mixer, &scarlett2_dim_mute_ctl, @@ -4518,7 +4568,6 @@ static int scarlett2_get_flash_segment_nums(struct usb_mixer_interface *mixer) static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; int err, i; s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; @@ -4583,7 +4632,8 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) !!private->mute_switch[i]; /* read SW/HW switches */ - if (info->line_out_hw_vol) { + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_SW_HW_SWITCH)) { err = scarlett2_usb_get_config( mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, private->num_line_out, &private->vol_sw_hw_switch); @@ -4628,11 +4678,9 @@ static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; int i; - /* if line_out_hw_vol is 0, there are no controls to update */ - if (!info->line_out_hw_vol) + if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH)) return; private->vol_updated = 1; @@ -4651,10 +4699,9 @@ static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; int i; - if (!info->line_out_hw_vol) + if (!scarlett2_has_config_item(private, SCARLETT2_CONFIG_SW_HW_SWITCH)) return; private->dim_mute_updated = 1; From 90d8fef837a53e6b5c2c3c14c6063ee8d78277a9 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 06:00:05 +1030 Subject: [PATCH 294/337] ALSA: scarlett2: Allow for interfaces without per-channel volume Currently-supported interfaces with a mixer have per-channel volume controls, but this changes in Gen 4. Add a check so that the Playback Volume and associated controls don't get created unless the SCARLETT2_CONFIG_LINE_OUT_VOLUME config item is present. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/30f68cb311e27f2cc1351cb846218f7248a90263.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 72 +++++++++++++++++++++---------------- 1 file changed, 42 insertions(+), 30 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 0d86c41432fc..95f6f1454fbf 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -3726,6 +3726,13 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) return err; } + /* Remaining controls are only applicable if the device + * has per-channel line-out volume controls. + */ + if (!scarlett2_has_config_item(private, + SCARLETT2_CONFIG_LINE_OUT_VOLUME)) + return 0; + /* Add volume controls */ for (i = 0; i < private->num_line_out; i++) { int index = line_out_remap(private, i); @@ -4569,7 +4576,6 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; int err, i; - s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) { err = scarlett2_usb_get_config( @@ -4608,41 +4614,47 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; - /* read SW line out volume */ - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, - private->num_line_out, &sw_vol); - if (err < 0) - return err; - - for (i = 0; i < private->num_line_out; i++) - private->vol[i] = clamp( - sw_vol[i] + SCARLETT2_VOLUME_BIAS, - 0, SCARLETT2_VOLUME_BIAS); - - /* read SW mute */ - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_MUTE_SWITCH, - private->num_line_out, &private->mute_switch); - if (err < 0) - return err; - - for (i = 0; i < private->num_line_out; i++) - private->mute_switch[i] = - !!private->mute_switch[i]; - - /* read SW/HW switches */ if (scarlett2_has_config_item(private, - SCARLETT2_CONFIG_SW_HW_SWITCH)) { + SCARLETT2_CONFIG_LINE_OUT_VOLUME)) { + s16 sw_vol[SCARLETT2_ANALOGUE_MAX]; + + /* read SW line out volume */ err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, - private->num_line_out, &private->vol_sw_hw_switch); + mixer, SCARLETT2_CONFIG_LINE_OUT_VOLUME, + private->num_line_out, &sw_vol); if (err < 0) return err; for (i = 0; i < private->num_line_out; i++) - private->vol_sw_hw_switch[i] = - !!private->vol_sw_hw_switch[i]; + private->vol[i] = clamp( + sw_vol[i] + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + + /* read SW mute */ + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_MUTE_SWITCH, + private->num_line_out, &private->mute_switch); + if (err < 0) + return err; + + for (i = 0; i < private->num_line_out; i++) + private->mute_switch[i] = + !!private->mute_switch[i]; + + /* read SW/HW switches */ + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_SW_HW_SWITCH)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_SW_HW_SWITCH, + private->num_line_out, + &private->vol_sw_hw_switch); + if (err < 0) + return err; + + for (i = 0; i < private->num_line_out; i++) + private->vol_sw_hw_switch[i] = + !!private->vol_sw_hw_switch[i]; + } } err = scarlett2_update_volumes(mixer); From 56275126aca20e0a0bfa51e3307576aee76b7264 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 06:00:15 +1030 Subject: [PATCH 295/337] ALSA: scarlett2: Add scarlett2_mixer_value_to_db() Refactor scarlett2_usb_get_mix(), moving the scarlett2_mixer_values[] lookup into scarlett2_mixer_value_to_db(). Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/7adf869852aba2819fddb850b0ea8df5f7d73931.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 28 +++++++++++++++++----------- 1 file changed, 17 insertions(+), 11 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 95f6f1454fbf..6b026648ac3c 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1763,6 +1763,19 @@ static int scarlett2_has_mixer(struct scarlett2_data *private) return !!private->info->mux_assignment[0][0].count; } +/* Map from mixer value to (db + 80) * 2 + * (reverse of scarlett2_mixer_values[]) + */ +static int scarlett2_mixer_value_to_db(int value) +{ + int i; + + for (i = 0; i < SCARLETT2_MIXER_VALUE_COUNT; i++) + if (scarlett2_mixer_values[i] >= value) + return i; + return SCARLETT2_MIXER_MAX_VALUE; +} + /* Send a USB message to get the volumes for all inputs of one mix * and put the values into private->mix[] */ @@ -1772,7 +1785,7 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, struct scarlett2_data *private = mixer->private_data; int num_mixer_in = private->num_mix_in; - int err, i, j, k; + int err, i, j; struct { __le16 mix_num; @@ -1790,16 +1803,9 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, if (err < 0) return err; - for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) { - u16 mixer_value = le16_to_cpu(data[i]); - - for (k = 0; k < SCARLETT2_MIXER_VALUE_COUNT; k++) - if (scarlett2_mixer_values[k] >= mixer_value) - break; - if (k == SCARLETT2_MIXER_VALUE_COUNT) - k = SCARLETT2_MIXER_MAX_VALUE; - private->mix[j] = k; - } + for (i = 0, j = mix_num * num_mixer_in; i < num_mixer_in; i++, j++) + private->mix[j] = scarlett2_mixer_value_to_db( + le16_to_cpu(data[i])); return 0; } From a1faecfcfe35ae774e75145470998d4418a78f0a Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 06:00:29 +1030 Subject: [PATCH 296/337] ALSA: scarlett2: Add #define for SCARLETT2_MIX_MAX Add a #define for SCARLETT2_MIX_MAX (max of mixer inputs * outputs) as that will be used again soon. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/83cec5ccd75f0db2bd061a76d31a7023d26300c1.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6b026648ac3c..71d09c03b1f4 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -208,6 +208,9 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { /* Maximum number of outputs from the mixer */ #define SCARLETT2_OUTPUT_MIX_MAX 12 +/* Maximum number of mixer gain controls */ +#define SCARLETT2_MIX_MAX (SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX) + /* Maximum size of the data in the USB mux assignment message: * 20 inputs, 20 outputs, 25 matrix inputs, 12 spare */ @@ -719,7 +722,7 @@ struct scarlett2_data { struct snd_kcontrol *speaker_switching_ctl; struct snd_kcontrol *talkback_ctl; u8 mux[SCARLETT2_MUX_MAX]; - u8 mix[SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX]; + u8 mix[SCARLETT2_MIX_MAX]; }; /*** Model-specific data ***/ From ad5174608eca55bd3fe6dc16057248fa03e6673d Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 06:00:38 +1030 Subject: [PATCH 297/337] ALSA: scarlett2: Rename db_scale_scarlett2_gain to volume db_scale_scarlett2_gain is the TLV for the output volume controls. Gen 4 has software-controllable input gain controls, so rename this to db_scale_scarlett2_volume so we can use that name for the inputs. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/d544ec7cc5d5a849da104a5a78b17f61f50657c1.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 71d09c03b1f4..2aa523f05dae 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2417,7 +2417,7 @@ unlock: } static const DECLARE_TLV_DB_MINMAX( - db_scale_scarlett2_gain, -SCARLETT2_VOLUME_BIAS * 100, 0 + db_scale_scarlett2_volume, -SCARLETT2_VOLUME_BIAS * 100, 0 ); static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { @@ -2428,7 +2428,7 @@ static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { .info = scarlett2_volume_ctl_info, .get = scarlett2_master_volume_ctl_get, .private_value = 0, /* max value */ - .tlv = { .p = db_scale_scarlett2_gain } + .tlv = { .p = db_scale_scarlett2_volume } }; static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { @@ -2440,7 +2440,7 @@ static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { .get = scarlett2_volume_ctl_get, .put = scarlett2_volume_ctl_put, .private_value = 0, /* max value */ - .tlv = { .p = db_scale_scarlett2_gain } + .tlv = { .p = db_scale_scarlett2_volume } }; /*** Mute Switch Controls ***/ From d9b63123fbb0a9cf9938dddbe7e623c731491d7d Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 06:00:53 +1030 Subject: [PATCH 298/337] ALSA: scarlett2: Split input_other into level/pad/air/phantom Gen 2/3 devices have a single notification value for "input other" changes. Gen 4 has separate notification values for level, pad, air, and phantom power changes. Therefore, split the input_other_updated field and the scarlett2_update_input_other() function into the four components so that they can be handled separately later. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/a1a1d190659d56689792aa20ceeb53a6175171ad.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 198 +++++++++++++++++++++++++----------- 1 file changed, 140 insertions(+), 58 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 2aa523f05dae..5e8777866cb6 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -685,7 +685,10 @@ struct scarlett2_data { u8 sync_updated; u8 vol_updated; u8 dim_mute_updated; - u8 input_other_updated; + u8 input_level_updated; + u8 input_pad_updated; + u8 input_air_updated; + u8 input_phantom_updated; u8 monitor_other_updated; u8 mux_updated; u8 speaker_switching_switched; @@ -2687,57 +2690,20 @@ static const struct snd_kcontrol_new scarlett2_sw_hw_enum_ctl = { /*** Line Level/Instrument Level Switch Controls ***/ -static int scarlett2_update_input_other(struct usb_mixer_interface *mixer) +static int scarlett2_update_input_level(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; - private->input_other_updated = 0; + private->input_level_updated = 0; - if (info->level_input_count) { - int err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, - info->level_input_count + info->level_input_first, - private->level_switch); - if (err < 0) - return err; - } + if (!info->level_input_count) + return 0; - if (info->pad_input_count) { - int err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_PAD_SWITCH, - info->pad_input_count, private->pad_switch); - if (err < 0) - return err; - } - - if (info->air_input_count) { - int err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_AIR_SWITCH, - info->air_input_count, private->air_switch); - if (err < 0) - return err; - } - - if (info->phantom_count) { - int err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH, - info->phantom_count, private->phantom_switch); - if (err < 0) - return err; - - if (scarlett2_has_config_item( - private, - SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) { - err = scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, - 1, &private->phantom_persistence); - if (err < 0) - return err; - } - } - - return 0; + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, + info->level_input_count + info->level_input_first, + private->level_switch); } static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, @@ -2768,8 +2734,8 @@ static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->input_other_updated) { - err = scarlett2_update_input_other(mixer); + if (private->input_level_updated) { + err = scarlett2_update_input_level(mixer); if (err < 0) goto unlock; } @@ -2827,6 +2793,21 @@ static const struct snd_kcontrol_new scarlett2_level_enum_ctl = { /*** Pad Switch Controls ***/ +static int scarlett2_update_input_pad(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + private->input_pad_updated = 0; + + if (!info->pad_input_count) + return 0; + + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PAD_SWITCH, + info->pad_input_count, private->pad_switch); +} + static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -2842,8 +2823,8 @@ static int scarlett2_pad_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->input_other_updated) { - err = scarlett2_update_input_other(mixer); + if (private->input_pad_updated) { + err = scarlett2_update_input_pad(mixer); if (err < 0) goto unlock; } @@ -2901,6 +2882,21 @@ static const struct snd_kcontrol_new scarlett2_pad_ctl = { /*** Air Switch Controls ***/ +static int scarlett2_update_input_air(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + private->input_air_updated = 0; + + if (!info->air_input_count) + return 0; + + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_AIR_SWITCH, + info->air_input_count, private->air_switch); +} + static int scarlett2_air_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -2916,8 +2912,8 @@ static int scarlett2_air_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->input_other_updated) { - err = scarlett2_update_input_other(mixer); + if (private->input_air_updated) { + err = scarlett2_update_input_air(mixer); if (err < 0) goto unlock; } @@ -2974,6 +2970,35 @@ static const struct snd_kcontrol_new scarlett2_air_ctl = { /*** Phantom Switch Controls ***/ +static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err; + + private->input_phantom_updated = 0; + + if (!info->phantom_count) + return 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH, + info->phantom_count, private->phantom_switch); + if (err < 0) + return err; + + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_PHANTOM_PERSISTENCE)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PHANTOM_PERSISTENCE, + 1, &private->phantom_persistence); + if (err < 0) + return err; + } + + return 0; +} + static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -2989,8 +3014,8 @@ static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->input_other_updated) { - err = scarlett2_update_input_other(mixer); + if (private->input_phantom_updated) { + err = scarlett2_update_input_phantom(mixer); if (err < 0) goto unlock; } @@ -4598,7 +4623,19 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) return 0; } - err = scarlett2_update_input_other(mixer); + err = scarlett2_update_input_level(mixer); + if (err < 0) + return err; + + err = scarlett2_update_input_pad(mixer); + if (err < 0) + return err; + + err = scarlett2_update_input_air(mixer); + if (err < 0) + return err; + + err = scarlett2_update_input_phantom(mixer); if (err < 0) return err; @@ -4737,30 +4774,75 @@ static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) &private->mute_ctls[i]->id); } -/* Notify on "input other" change (level/pad/air) */ -static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) +/* Notify on input level switch change */ +static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; int i; - private->input_other_updated = 1; + private->input_level_updated = 1; for (i = 0; i < info->level_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->level_ctls[i]->id); +} + +/* Notify on input pad switch change */ +static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + private->input_pad_updated = 1; + for (i = 0; i < info->pad_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->pad_ctls[i]->id); +} + +/* Notify on input air switch change */ +static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + private->input_air_updated = 1; + for (i = 0; i < info->air_input_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->air_ctls[i]->id); +} + +/* Notify on input phantom switch change */ +static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + private->input_phantom_updated = 1; + for (i = 0; i < info->phantom_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->phantom_ctls[i]->id); } +/* Notify on "input other" change (level/pad/air/phantom) */ +static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) +{ + scarlett2_notify_input_level(mixer); + scarlett2_notify_input_pad(mixer); + scarlett2_notify_input_air(mixer); + scarlett2_notify_input_phantom(mixer); +} + /* Notify on "monitor other" change (direct monitor, speaker * switching, talkback) */ From d3cf557b26a77f6a285fe6b9b87c434ffb340ceb Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Mon, 25 Dec 2023 06:01:14 +1030 Subject: [PATCH 299/337] ALSA: scarlett2: Split direct_monitor out from monitor_other The notification value for monitor_other on the large interfaces is the same as the notification value for direct_monitor on the 3rd Gen small interfaces. Add a separate scarlett3a_notifications array and split out the direct_monitor handling. Signed-off-by: Geoffrey D. Bennett Link: https://lore.kernel.org/r/9b56a483e3e9c1447684f18239a88652c1f01445.1703444932.git.g@b4.vu Signed-off-by: Takashi Iwai --- sound/usb/mixer_scarlett2.c | 289 ++++++++++++++++++++---------------- 1 file changed, 158 insertions(+), 131 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 5e8777866cb6..6dd758cfb5cb 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -277,8 +277,10 @@ static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer); static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer); static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer); +static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer); + +/* Arrays of notification callback functions */ -/* Array of notification callback functions */ static const struct scarlett2_notification scarlett2_notifications[] = { { 0x00000001, NULL }, /* ack, gets ignored */ { 0x00000008, scarlett2_notify_sync }, @@ -289,6 +291,13 @@ static const struct scarlett2_notification scarlett2_notifications[] = { { 0, NULL } }; +static const struct scarlett2_notification scarlett3a_notifications[] = { + { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00800000, scarlett2_notify_input_other }, + { 0x01000000, scarlett2_notify_direct_monitor }, + { 0, NULL } +}; + /* Configuration parameters that can be read and written */ enum { SCARLETT2_CONFIG_DIM_MUTE, @@ -379,7 +388,7 @@ static const struct scarlett2_config_set scarlett2_config_set_gen2b = { /* Gen 3 devices without a mixer (Solo and 2i2) */ static const struct scarlett2_config_set scarlett2_config_set_gen3a = { - .notifications = scarlett2_notifications, + .notifications = scarlett3a_notifications, .items = { [SCARLETT2_CONFIG_MSD_SWITCH] = { .offset = 0x04, .size = 8, .activate = 6 }, @@ -690,6 +699,7 @@ struct scarlett2_data { u8 input_air_updated; u8 input_phantom_updated; u8 monitor_other_updated; + u8 direct_monitor_updated; u8 mux_updated; u8 speaker_switching_switched; u8 sync; @@ -3127,7 +3137,7 @@ static const struct snd_kcontrol_new scarlett2_phantom_persistence_ctl = { .put = scarlett2_phantom_persistence_ctl_put, }; -/*** Direct Monitor Control ***/ +/*** Speaker Switching Control ***/ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer) { @@ -3147,11 +3157,6 @@ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer) private->monitor_other_updated = 0; - if (info->direct_monitor) - return scarlett2_usb_get_config( - mixer, SCARLETT2_CONFIG_DIRECT_MONITOR, - 1, &private->direct_monitor_switch); - /* if it doesn't do speaker switching then it also doesn't do * talkback */ @@ -3196,119 +3201,6 @@ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer) return 0; } -static int scarlett2_direct_monitor_ctl_get( - struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) -{ - struct usb_mixer_elem_info *elem = kctl->private_data; - struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_data *private = elem->head.mixer->private_data; - int err = 0; - - mutex_lock(&private->data_mutex); - - if (private->hwdep_in_use) { - err = -EBUSY; - goto unlock; - } - - if (private->monitor_other_updated) { - err = scarlett2_update_monitor_other(mixer); - if (err < 0) - goto unlock; - } - ucontrol->value.enumerated.item[0] = private->direct_monitor_switch; - -unlock: - mutex_unlock(&private->data_mutex); - return err; -} - -static int scarlett2_direct_monitor_ctl_put( - struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) -{ - struct usb_mixer_elem_info *elem = kctl->private_data; - struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_data *private = mixer->private_data; - - int index = elem->control; - int oval, val, err = 0; - - mutex_lock(&private->data_mutex); - - if (private->hwdep_in_use) { - err = -EBUSY; - goto unlock; - } - - oval = private->direct_monitor_switch; - val = min(ucontrol->value.enumerated.item[0], 2U); - - if (oval == val) - goto unlock; - - private->direct_monitor_switch = val; - - /* Send switch change to the device */ - err = scarlett2_usb_set_config( - mixer, SCARLETT2_CONFIG_DIRECT_MONITOR, index, val); - if (err == 0) - err = 1; - -unlock: - mutex_unlock(&private->data_mutex); - return err; -} - -static int scarlett2_direct_monitor_stereo_enum_ctl_info( - struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) -{ - static const char *const values[3] = { - "Off", "Mono", "Stereo" - }; - - return snd_ctl_enum_info(uinfo, 1, 3, values); -} - -/* Direct Monitor for Solo is mono-only and only needs a boolean control - * Direct Monitor for 2i2 is selectable between Off/Mono/Stereo - */ -static const struct snd_kcontrol_new scarlett2_direct_monitor_ctl[2] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "", - .info = snd_ctl_boolean_mono_info, - .get = scarlett2_direct_monitor_ctl_get, - .put = scarlett2_direct_monitor_ctl_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "", - .info = scarlett2_direct_monitor_stereo_enum_ctl_info, - .get = scarlett2_direct_monitor_ctl_get, - .put = scarlett2_direct_monitor_ctl_put, - } -}; - -static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer) -{ - struct scarlett2_data *private = mixer->private_data; - const struct scarlett2_device_info *info = private->info; - const char *s; - - if (!info->direct_monitor) - return 0; - - s = info->direct_monitor == 1 - ? "Direct Monitor Playback Switch" - : "Direct Monitor Playback Enum"; - - return scarlett2_add_new_ctl( - mixer, &scarlett2_direct_monitor_ctl[info->direct_monitor - 1], - 0, 1, s, &private->direct_monitor_ctl); -} - -/*** Speaker Switching Control ***/ - static int scarlett2_speaker_switch_enum_ctl_info( struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { @@ -4014,6 +3906,133 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) return 0; } +/*** Direct Monitor Control ***/ + +static int scarlett2_update_direct_monitor(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + + private->direct_monitor_updated = 0; + + if (!private->info->direct_monitor) + return 0; + + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_DIRECT_MONITOR, + 1, &private->direct_monitor_switch); +} + +static int scarlett2_direct_monitor_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = elem->head.mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->direct_monitor_updated) { + err = scarlett2_update_direct_monitor(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->direct_monitor_switch; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_direct_monitor_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->direct_monitor_switch; + val = min(ucontrol->value.enumerated.item[0], 2U); + + if (oval == val) + goto unlock; + + private->direct_monitor_switch = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_DIRECT_MONITOR, index, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_direct_monitor_stereo_enum_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[3] = { + "Off", "Mono", "Stereo" + }; + + return snd_ctl_enum_info(uinfo, 1, 3, values); +} + +/* Direct Monitor for Solo is mono-only and only needs a boolean control + * Direct Monitor for 2i2 is selectable between Off/Mono/Stereo + */ +static const struct snd_kcontrol_new scarlett2_direct_monitor_ctl[2] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_direct_monitor_ctl_get, + .put = scarlett2_direct_monitor_ctl_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_direct_monitor_stereo_enum_ctl_info, + .get = scarlett2_direct_monitor_ctl_get, + .put = scarlett2_direct_monitor_ctl_put, + } +}; + +static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + const char *s; + + if (!info->direct_monitor) + return 0; + + s = info->direct_monitor == 1 + ? "Direct Monitor Playback Switch" + : "Direct Monitor Playback Enum"; + + return scarlett2_add_new_ctl( + mixer, &scarlett2_direct_monitor_ctl[info->direct_monitor - 1], + 0, 1, s, &private->direct_monitor_ctl); +} + /*** Mux Source Selection Controls ***/ static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, @@ -4639,7 +4658,7 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; - err = scarlett2_update_monitor_other(mixer); + err = scarlett2_update_direct_monitor(mixer); if (err < 0) return err; @@ -4647,6 +4666,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (!scarlett2_has_mixer(private)) return 0; + err = scarlett2_update_monitor_other(mixer); + if (err < 0) + return err; + if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_STANDALONE_SWITCH)) { err = scarlett2_usb_get_config( @@ -4843,9 +4866,7 @@ static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) scarlett2_notify_input_phantom(mixer); } -/* Notify on "monitor other" change (direct monitor, speaker - * switching, talkback) - */ +/* Notify on "monitor other" change (speaker switching, talkback) */ static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; @@ -4854,12 +4875,6 @@ static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer) private->monitor_other_updated = 1; - if (info->direct_monitor) { - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, - &private->direct_monitor_ctl->id); - return; - } - if (info->has_speaker_switching) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->speaker_switching_ctl->id); @@ -4885,6 +4900,18 @@ static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer) } } +/* Notify on direct monitor switch change */ +static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + + private->direct_monitor_updated = 1; + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->direct_monitor_ctl->id); +} + /* Interrupt callback */ static void scarlett2_notify(struct urb *urb) { From 4dedf7ca929a3f29fbed973e85372056d69a1848 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:36:13 +1030 Subject: [PATCH 300/337] ALSA: scarlett2: Remove repeated elem->head.mixer references Use a local variable *mixer rather than repeating elem->header.mixer in scarlett2_direct_monitor_ctl_get() and scarlett2_meter_ctl_get(). Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/b21bacf4056366e10e01077e224d2b4970fdfe31.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6dd758cfb5cb..8f466ad82d5a 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -3927,7 +3927,7 @@ static int scarlett2_direct_monitor_ctl_get( { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; - struct scarlett2_data *private = elem->head.mixer->private_data; + struct scarlett2_data *private = mixer->private_data; int err = 0; mutex_lock(&private->data_mutex); @@ -4191,7 +4191,8 @@ static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_data *private = elem->head.mixer->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; u8 *meter_level_map = private->meter_level_map; u16 meter_levels[SCARLETT2_MAX_METERS]; int i, err; @@ -4203,7 +4204,7 @@ static int scarlett2_meter_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - err = scarlett2_usb_get_meter_levels(elem->head.mixer, elem->channels, + err = scarlett2_usb_get_meter_levels(mixer, elem->channels, meter_levels); if (err < 0) goto unlock; From dd57b1213ab60ec9ad603507bd25ed069f9a6b61 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:36:21 +1030 Subject: [PATCH 301/337] ALSA: scarlett2: Add support for air/phantom control on input 2 The Focusrite Scarlett Gen 4 Solo has Air and Phantom Power controls on analogue input #2 (the Gen 3 Solo had these controls on analogue input #1). Add air_input_first and phantom_first device info options to cater for this. These options are similar to the level_input_first option that was added for the Gen 3 Solo, but these new options do not require adjusting the index of the control. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/19511f18895b8c094985a4a5691fbc1dc028c108.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 8f466ad82d5a..b881881124a6 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -632,9 +632,15 @@ struct scarlett2_device_info { */ u8 air_input_count; + /* the first input with an air control (0-based) */ + u8 air_input_first; + /* the number of phantom (48V) software switchable controls */ u8 phantom_count; + /* the first input with phantom power control (0-based) */ + u8 phantom_first; + /* the number of inputs each phantom switch controls */ u8 inputs_per_phantom; @@ -3043,6 +3049,7 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; int index = elem->control; int oval, val, err = 0; @@ -3064,7 +3071,7 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH, - index, val); + index + info->phantom_first, val); if (err == 0) err = 1; @@ -3763,7 +3770,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input air controls */ for (i = 0; i < info->air_input_count; i++) { - snprintf(s, sizeof(s), fmt, i + 1, "Air", "Switch"); + snprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first, + "Air", "Switch"); err = scarlett2_add_new_ctl(mixer, &scarlett2_air_ctl, i, 1, s, &private->air_ctls[i]); if (err < 0) @@ -3773,7 +3781,8 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input phantom controls */ if (info->inputs_per_phantom == 1) { for (i = 0; i < info->phantom_count; i++) { - scnprintf(s, sizeof(s), fmt, i + 1, + scnprintf(s, sizeof(s), fmt, + i + 1 + info->phantom_first, "Phantom Power", "Switch"); err = scarlett2_add_new_ctl( mixer, &scarlett2_phantom_ctl, From 4fa07ff7b625e5578dd974d5c43d701cb3624d84 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:36:29 +1030 Subject: [PATCH 302/337] ALSA: scarlett2: Add support for Gen 4 style parameters Writing Scarlett Gen 4 parameters differs from Gen 2 and Gen 3: - the values are written into a shared write location - the values are only byte-sized - the read locations now extend beyond 0xFF - a separate NVRAM save step is no longer required This patch implements that alternate write style, triggered by setting the config item size field to zero. The write address is specified through a new config set field gen4_write_addr. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/1624e6d8a0c629c3bdfe53825b16e8b589724fc4.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 53 +++++++++++++++++++++++++++++++++---- 1 file changed, 48 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index b881881124a6..b2555236d113 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -320,16 +320,21 @@ enum { }; /* Location, size, and activation command number for the configuration - * parameters. Size is in bits and may be 1, 8, or 16. + * parameters. Size is in bits and may be 0, 1, 8, or 16. + * + * A size of 0 indicates that the parameter is a byte-sized Scarlett + * Gen 4 configuration which is written through the gen4_write_addr + * location (but still read through the given offset location). */ struct scarlett2_config { - u8 offset; + u16 offset; u8 size; u8 activate; }; struct scarlett2_config_set { const struct scarlett2_notification *notifications; + u16 gen4_write_addr; const struct scarlett2_config items[SCARLETT2_CONFIG_COUNT]; }; @@ -1612,9 +1617,12 @@ static int scarlett2_usb_get_config( if (!config_item->offset) return -EFAULT; + /* Gen 4 style parameters are always 1 byte */ + size = config_item->size ? config_item->size : 8; + /* For byte-sized parameters, retrieve directly into buf */ - if (config_item->size >= 8) { - size = config_item->size / 8 * count; + if (size >= 8) { + size = size / 8 * count; err = scarlett2_usb_get(mixer, config_item->offset, buf, size); if (err < 0) return err; @@ -1684,8 +1692,9 @@ static int scarlett2_usb_set_config( int config_item_num, int index, int value) { struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_config_set *config_set = private->config_set; const struct scarlett2_config *config_item = - &private->config_set->items[config_item_num]; + &config_set->items[config_item_num]; int offset, size; int err; @@ -1695,6 +1704,36 @@ static int scarlett2_usb_set_config( if (!config_item->offset) return -EFAULT; + /* Gen 4 style writes are selected with size = 0; + * these are only byte-sized values written through a shared + * location, different to the read address + */ + if (!config_item->size) { + if (!config_set->gen4_write_addr) + return -EFAULT; + + /* Place index in gen4_write_addr + 1 */ + err = scarlett2_usb_set_data( + mixer, config_set->gen4_write_addr + 1, 1, index); + if (err < 0) + return err; + + /* Place value in gen4_write_addr */ + err = scarlett2_usb_set_data( + mixer, config_set->gen4_write_addr, 1, value); + if (err < 0) + return err; + + /* Request the interface do the write */ + return scarlett2_usb_activate_config( + mixer, config_item->activate); + } + + /* Not-Gen 4 style needs NVRAM save, supports + * bit-modification, and writing is done to the same place + * that the value can be read from + */ + /* Cancel any pending NVRAM save */ cancel_delayed_work_sync(&private->work); @@ -1736,6 +1775,10 @@ static int scarlett2_usb_set_config( if (err < 0) return err; + /* Gen 2 style writes to Gen 4 devices don't need saving */ + if (config_set->gen4_write_addr) + return 0; + /* Schedule the change to be written to NVRAM */ if (config_item->activate != SCARLETT2_USB_CONFIG_SAVE) schedule_delayed_work(&private->work, msecs_to_jiffies(2000)); From 1b53c116232e082db06ab5310c1be1d4bf75dfe1 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:36:38 +1030 Subject: [PATCH 303/337] ALSA: scarlett2: Allow for controls with a "mute mode" Gen 2/3 interfaces would only use 0/1 values for input level and phantom power switch controls. Gen 4 interfaces use the second bit to indicate that the state should be changed (or is changing), and the input is to be muted (or is muted) while that happens. Add a "mute" flag to config items to enable this behaviour for the level/phantom controls. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/f603cd16079c97fad910087e0302828a289d1c15.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 31 ++++++++++++++++++++++++++++--- 1 file changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index b2555236d113..a0f1f47fb817 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -325,11 +325,18 @@ enum { * A size of 0 indicates that the parameter is a byte-sized Scarlett * Gen 4 configuration which is written through the gen4_write_addr * location (but still read through the given offset location). + * + * Some Gen 4 configuration parameters are written with 0x02 for a + * desired value of 0x01, and 0x03 for 0x00. These are indicated with + * mute set to 1. 0x02 and 0x03 are temporary values while the device + * makes the change and the channel and/or corresponding DSP channel + * output is muted. */ struct scarlett2_config { u16 offset; u8 size; u8 activate; + u8 mute; }; struct scarlett2_config_set { @@ -2177,6 +2184,15 @@ static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, return 0; } +/* For config items with mute=1, xor bits 0 & 1 together to get the + * current/next state. This won't have any effect on values which are + * only ever 0/1. + */ +static uint8_t scarlett2_decode_muteable(uint8_t v) +{ + return (v ^ (v >> 1)) & 1; +} + /*** Control Functions ***/ /* helper function to create a new control */ @@ -2798,7 +2814,8 @@ static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, if (err < 0) goto unlock; } - ucontrol->value.enumerated.item[0] = private->level_switch[index]; + ucontrol->value.enumerated.item[0] = scarlett2_decode_muteable( + private->level_switch[index]); unlock: mutex_unlock(&private->data_mutex); @@ -2831,6 +2848,10 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, private->level_switch[index] = val; + /* To set the Gen 4 muteable controls, bit 1 gets set instead */ + if (private->config_set->items[SCARLETT2_CONFIG_LEVEL_SWITCH].mute) + val = (!val) | 0x02; + /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, index, val); @@ -3078,8 +3099,8 @@ static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, if (err < 0) goto unlock; } - ucontrol->value.integer.value[0] = - private->phantom_switch[elem->control]; + ucontrol->value.integer.value[0] = scarlett2_decode_muteable( + private->phantom_switch[elem->control]); unlock: mutex_unlock(&private->data_mutex); @@ -3112,6 +3133,10 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, private->phantom_switch[index] = val; + /* To set the Gen 4 muteable controls, bit 1 gets set */ + if (private->config_set->items[SCARLETT2_CONFIG_PHANTOM_SWITCH].mute) + val = (!val) | 0x02; + /* Send switch change to the device */ err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_PHANTOM_SWITCH, index + info->phantom_first, val); From 038216f2bc85167449359cb4eec5fc5bfffbe4ef Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:36:54 +1030 Subject: [PATCH 304/337] ALSA: scarlett2: Add support for Air Presence + Drive option Extend the existing "air" option support from Scarlett Gen 3, which had two states (off/on), to accommodate Scarlett Gen 4's new state: Presence + Drive. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/a9ccda7222842a72e4ce7aa258614ff45248bb16.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 46 +++++++++++++++++++++++++++++-------- 1 file changed, 36 insertions(+), 10 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a0f1f47fb817..1a955a6decf3 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -647,6 +647,12 @@ struct scarlett2_device_info { /* the first input with an air control (0-based) */ u8 air_input_first; + /* number of additional air options + * 0 for air presence only (Gen 3) + * 1 for air presence+drive (Gen 4) + */ + u8 air_option; + /* the number of phantom (48V) software switchable controls */ u8 phantom_count; @@ -3022,7 +3028,7 @@ static int scarlett2_air_ctl_put(struct snd_kcontrol *kctl, } oval = private->air_switch[index]; - val = !!ucontrol->value.integer.value[0]; + val = ucontrol->value.integer.value[0]; if (oval == val) goto unlock; @@ -3040,12 +3046,31 @@ unlock: return err; } -static const struct snd_kcontrol_new scarlett2_air_ctl = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "", - .info = snd_ctl_boolean_mono_info, - .get = scarlett2_air_ctl_get, - .put = scarlett2_air_ctl_put, +static int scarlett2_air_with_drive_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[3] = { + "Off", "Presence", "Presence + Drive" + }; + + return snd_ctl_enum_info(uinfo, 1, 3, values); +} + +static const struct snd_kcontrol_new scarlett2_air_ctl[2] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_air_ctl_get, + .put = scarlett2_air_ctl_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_air_with_drive_ctl_info, + .get = scarlett2_air_ctl_get, + .put = scarlett2_air_ctl_put, + } }; /*** Phantom Switch Controls ***/ @@ -3839,9 +3864,10 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /* Add input air controls */ for (i = 0; i < info->air_input_count; i++) { snprintf(s, sizeof(s), fmt, i + 1 + info->air_input_first, - "Air", "Switch"); - err = scarlett2_add_new_ctl(mixer, &scarlett2_air_ctl, - i, 1, s, &private->air_ctls[i]); + "Air", info->air_option ? "Enum" : "Switch"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_air_ctl[info->air_option], + i, 1, s, &private->air_ctls[i]); if (err < 0) return err; } From 0a995e38dc440fdfe658a5ebf19bf6eb647a0f5f Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:37:12 +1030 Subject: [PATCH 305/337] ALSA: scarlett2: Add support for software-controllable input gain Some devices in the Scarlett Gen 4 series have support for software-controllable input gain. Along with this comes a channel select option, an auto-gain feature, "safe" mode, and linking two channels into a stereo pair. Mark the new scarlett2_notify_input_*() functions with __always_unused until they get used when the Gen 4 notification callback function arrays are added. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/fc39e80bb39863dd1579d342097203942b4f3034.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 796 +++++++++++++++++++++++++++++++++++- 1 file changed, 795 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 1a955a6decf3..ab42d8f90e46 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -164,6 +164,7 @@ /* some gui mixers can't handle negative ctl values */ #define SCARLETT2_VOLUME_BIAS 127 +#define SCARLETT2_GAIN_BIAS 70 /* mixer range from -80dB to +6dB in 0.5dB steps */ #define SCARLETT2_MIXER_MIN_DB -80 @@ -196,11 +197,12 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { /* Maximum number of analogue outputs */ #define SCARLETT2_ANALOGUE_MAX 10 -/* Maximum number of level and pad switches */ +/* Maximum number of various input controls */ #define SCARLETT2_LEVEL_SWITCH_MAX 2 #define SCARLETT2_PAD_SWITCH_MAX 8 #define SCARLETT2_AIR_SWITCH_MAX 8 #define SCARLETT2_PHANTOM_SWITCH_MAX 2 +#define SCARLETT2_INPUT_GAIN_MAX 2 /* Maximum number of inputs to the mixer */ #define SCARLETT2_INPUT_MIX_MAX 25 @@ -266,6 +268,16 @@ static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { "Mute Playback Switch", "Dim Playback Switch" }; +/* Autogain Status Values */ +enum { + SCARLETT2_AUTOGAIN_STATUS_STOPPED, + SCARLETT2_AUTOGAIN_STATUS_RUNNING, + SCARLETT2_AUTOGAIN_STATUS_FAILED, + SCARLETT2_AUTOGAIN_STATUS_CANCELLED, + SCARLETT2_AUTOGAIN_STATUS_UNKNOWN, + SCARLETT2_AUTOGAIN_STATUS_COUNT +}; + /* Notification callback functions */ struct scarlett2_notification { u32 mask; @@ -316,6 +328,12 @@ enum { SCARLETT2_CONFIG_MONITOR_OTHER_SWITCH, SCARLETT2_CONFIG_MONITOR_OTHER_ENABLE, SCARLETT2_CONFIG_TALKBACK_MAP, + SCARLETT2_CONFIG_AUTOGAIN_SWITCH, + SCARLETT2_CONFIG_AUTOGAIN_STATUS, + SCARLETT2_CONFIG_INPUT_GAIN, + SCARLETT2_CONFIG_SAFE_SWITCH, + SCARLETT2_CONFIG_INPUT_SELECT_SWITCH, + SCARLETT2_CONFIG_INPUT_LINK_SWITCH, SCARLETT2_CONFIG_COUNT }; @@ -662,6 +680,9 @@ struct scarlett2_device_info { /* the number of inputs each phantom switch controls */ u8 inputs_per_phantom; + /* the number of inputs with software-controllable gain */ + u8 gain_input_count; + /* the number of direct monitor options * (0 = none, 1 = mono only, 2 = mono/stereo) */ @@ -722,6 +743,10 @@ struct scarlett2_data { u8 input_pad_updated; u8 input_air_updated; u8 input_phantom_updated; + u8 input_select_updated; + u8 input_gain_updated; + u8 autogain_updated; + u8 input_safe_updated; u8 monitor_other_updated; u8 direct_monitor_updated; u8 mux_updated; @@ -737,6 +762,12 @@ struct scarlett2_data { u8 air_switch[SCARLETT2_AIR_SWITCH_MAX]; u8 phantom_switch[SCARLETT2_PHANTOM_SWITCH_MAX]; u8 phantom_persistence; + u8 input_select_switch; + u8 input_link_switch[SCARLETT2_INPUT_GAIN_MAX / 2]; + u8 gain[SCARLETT2_INPUT_GAIN_MAX]; + u8 autogain_switch[SCARLETT2_INPUT_GAIN_MAX]; + u8 autogain_status[SCARLETT2_INPUT_GAIN_MAX]; + u8 safe_switch[SCARLETT2_INPUT_GAIN_MAX]; u8 direct_monitor_switch; u8 speaker_switching_switch; u8 talkback_switch; @@ -754,6 +785,12 @@ struct scarlett2_data { struct snd_kcontrol *pad_ctls[SCARLETT2_PAD_SWITCH_MAX]; struct snd_kcontrol *air_ctls[SCARLETT2_AIR_SWITCH_MAX]; struct snd_kcontrol *phantom_ctls[SCARLETT2_PHANTOM_SWITCH_MAX]; + struct snd_kcontrol *input_select_ctl; + struct snd_kcontrol *input_link_ctls[SCARLETT2_INPUT_GAIN_MAX / 2]; + struct snd_kcontrol *input_gain_ctls[SCARLETT2_INPUT_GAIN_MAX]; + struct snd_kcontrol *autogain_ctls[SCARLETT2_INPUT_GAIN_MAX]; + struct snd_kcontrol *autogain_status_ctls[SCARLETT2_INPUT_GAIN_MAX]; + struct snd_kcontrol *safe_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX]; struct snd_kcontrol *direct_monitor_ctl; struct snd_kcontrol *speaker_switching_ctl; @@ -2352,6 +2389,610 @@ static int scarlett2_add_sync_ctl(struct usb_mixer_interface *mixer) 0, 1, "Sync Status", &private->sync_ctl); } +/*** Autogain Switch and Status Controls ***/ + +static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int err, i; + u8 raw_autogain_status[SCARLETT2_INPUT_GAIN_MAX]; + + private->autogain_updated = 0; + + if (!info->gain_input_count) + return 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_AUTOGAIN_SWITCH, + info->gain_input_count, private->autogain_switch); + if (err < 0) + return err; + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_AUTOGAIN_STATUS, + info->gain_input_count, raw_autogain_status); + if (err < 0) + return err; + + /* Translate autogain_switch and raw_autogain_status into + * autogain_status + */ + for (i = 0; i < info->gain_input_count; i++) + if (private->autogain_switch[i]) + private->autogain_status[i] = + SCARLETT2_AUTOGAIN_STATUS_RUNNING; + else if (raw_autogain_status[i] == 0) + private->autogain_status[i] = + SCARLETT2_AUTOGAIN_STATUS_STOPPED; + else if (raw_autogain_status[i] >= 2 && + raw_autogain_status[i] <= 5) + private->autogain_status[i] = + SCARLETT2_AUTOGAIN_STATUS_FAILED; + else if (raw_autogain_status[i] == 6) + private->autogain_status[i] = + SCARLETT2_AUTOGAIN_STATUS_CANCELLED; + else + private->autogain_status[i] = + SCARLETT2_AUTOGAIN_STATUS_UNKNOWN; + + return 0; +} + +static int scarlett2_autogain_switch_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->autogain_updated) { + err = scarlett2_update_autogain(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = + private->autogain_switch[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_autogain_status_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->autogain_updated) { + err = scarlett2_update_autogain(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = + private->autogain_status[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_autogain_switch_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->autogain_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->autogain_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_AUTOGAIN_SWITCH, index, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_autogain_status_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[SCARLETT2_AUTOGAIN_STATUS_COUNT] = { + "Stopped", "Running", "Failed", "Cancelled", "Unknown" + }; + + return snd_ctl_enum_info( + uinfo, 1, SCARLETT2_AUTOGAIN_STATUS_COUNT, values); +} + +static const struct snd_kcontrol_new scarlett2_autogain_switch_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_autogain_switch_ctl_get, + .put = scarlett2_autogain_switch_ctl_put +}; + +static const struct snd_kcontrol_new scarlett2_autogain_status_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .name = "", + .info = scarlett2_autogain_status_ctl_info, + .get = scarlett2_autogain_status_ctl_get, +}; + +/*** Input Select Control ***/ + +static int scarlett2_update_input_select(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int link_count = info->gain_input_count / 2; + int err; + + private->input_select_updated = 0; + + if (!link_count) + return 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_INPUT_SELECT_SWITCH, + 1, &private->input_select_switch); + if (err < 0) + return err; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_INPUT_LINK_SWITCH, + link_count, private->input_link_switch); + if (err < 0) + return err; + + /* simplified because no model yet has link_count > 1 */ + if (private->input_link_switch[0]) + private->input_select_switch = 0; + + return 0; +} + +static int scarlett2_input_select_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->input_select_updated) { + err = scarlett2_update_input_select(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->input_select_switch; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_input_select_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int oval, val, err = 0; + int max_val = private->input_link_switch[0] ? 0 : 1; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->input_select_switch; + val = ucontrol->value.integer.value[0]; + + if (val < 0) + val = 0; + else if (val > max_val) + val = max_val; + + if (oval == val) + goto unlock; + + private->input_select_switch = val; + + /* Send switch change to the device if inputs not linked */ + if (!private->input_link_switch[0]) + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_INPUT_SELECT_SWITCH, + 1, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_input_select_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int inputs = private->info->gain_input_count; + int i, j; + int err; + char **values = kcalloc(inputs, sizeof(char *), GFP_KERNEL); + + if (!values) + return -ENOMEM; + + mutex_lock(&private->data_mutex); + + /* Loop through each input + * Linked inputs have one value for the pair + */ + for (i = 0, j = 0; i < inputs; i++) { + if (private->input_link_switch[i / 2]) { + values[j++] = kasprintf( + GFP_KERNEL, "Input %d-%d", i + 1, i + 2); + i++; + } else { + values[j++] = kasprintf( + GFP_KERNEL, "Input %d", i + 1); + } + } + + err = snd_ctl_enum_info(uinfo, 1, j, + (const char * const *)values); + + mutex_unlock(&private->data_mutex); + + for (i = 0; i < inputs; i++) + kfree(values[i]); + kfree(values); + + return err; +} + +static const struct snd_kcontrol_new scarlett2_input_select_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_input_select_ctl_info, + .get = scarlett2_input_select_ctl_get, + .put = scarlett2_input_select_ctl_put, +}; + +/*** Input Link Switch Controls ***/ + +static int scarlett2_input_link_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->input_select_updated) { + err = scarlett2_update_input_select(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = + private->input_link_switch[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_input_link_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->input_link_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->input_link_switch[index] = val; + + /* Notify of change in input select options available */ + snd_ctl_notify(mixer->chip->card, + SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, + &private->input_select_ctl->id); + private->input_select_updated = 1; + + /* Send switch change to the device + * Link for channels 1-2 is at index 1 + * No device yet has more than 2 channels linked + */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_INPUT_LINK_SWITCH, index + 1, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_input_link_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_input_link_ctl_get, + .put = scarlett2_input_link_ctl_put +}; + +/*** Input Gain Controls ***/ + +static int scarlett2_update_input_gain(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + private->input_gain_updated = 0; + + if (!info->gain_input_count) + return 0; + + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_INPUT_GAIN, + info->gain_input_count, private->gain); +} + +static int scarlett2_input_gain_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = elem->channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = SCARLETT2_GAIN_BIAS; + uinfo->value.integer.step = 1; + return 0; +} + +static int scarlett2_input_gain_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->input_gain_updated) { + err = scarlett2_update_input_gain(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = + private->gain[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_input_gain_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->gain[index]; + val = ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->gain[index] = val; + + /* Send gain change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_INPUT_GAIN, + index, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const DECLARE_TLV_DB_MINMAX( + db_scale_scarlett2_gain, -SCARLETT2_GAIN_BIAS * 100, 0 +); + +static const struct snd_kcontrol_new scarlett2_input_gain_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_input_gain_ctl_info, + .get = scarlett2_input_gain_ctl_get, + .put = scarlett2_input_gain_ctl_put, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_gain } +}; + +/*** Safe Controls ***/ + +static int scarlett2_update_input_safe(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + + private->input_safe_updated = 0; + + if (!info->gain_input_count) + return 0; + + return scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_SAFE_SWITCH, + info->gain_input_count, private->safe_switch); +} + +static int scarlett2_safe_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->input_safe_updated) { + err = scarlett2_update_input_safe(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = + private->safe_switch[elem->control]; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_safe_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int index = elem->control; + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->safe_switch[index]; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->safe_switch[index] = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config(mixer, SCARLETT2_CONFIG_SAFE_SWITCH, + index, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_safe_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = snd_ctl_boolean_mono_info, + .get = scarlett2_safe_ctl_get, + .put = scarlett2_safe_ctl_put, +}; + /*** Analogue Line Out Volume Controls ***/ /* Update hardware volume controls after receiving notification that @@ -3908,6 +4549,60 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) return err; } + /* Add software-controllable input gain controls */ + if (info->gain_input_count) { + err = scarlett2_add_new_ctl( + mixer, &scarlett2_input_select_ctl, 0, 1, + "Input Select Capture Enum", + &private->input_select_ctl); + if (err < 0) + return err; + + for (i = 0; i < info->gain_input_count; i++) { + if (i % 2) { + snprintf(s, sizeof(s), + "Line In %d-%d Link Capture Switch", + i, i + 1); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_input_link_ctl, + i / 2, 1, s, + &private->input_link_ctls[i / 2]); + if (err < 0) + return err; + } + + snprintf(s, sizeof(s), fmt, i + 1, + "Gain", "Volume"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_input_gain_ctl, + i, 1, s, &private->input_gain_ctls[i]); + if (err < 0) + return err; + + snprintf(s, sizeof(s), fmt, i + 1, + "Autogain", "Switch"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_autogain_switch_ctl, + i, 1, s, &private->autogain_ctls[i]); + if (err < 0) + return err; + + snprintf(s, sizeof(s), fmt, i + 1, + "Autogain Status", "Enum"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_autogain_status_ctl, + i, 1, s, &private->autogain_status_ctls[i]); + + snprintf(s, sizeof(s), fmt, i + 1, + "Safe", "Switch"); + err = scarlett2_add_new_ctl( + mixer, &scarlett2_safe_ctl, + i, 1, s, &private->safe_ctls[i]); + if (err < 0) + return err; + } + } + return 0; } @@ -4838,6 +5533,22 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; + err = scarlett2_update_input_select(mixer); + if (err < 0) + return err; + + err = scarlett2_update_input_gain(mixer); + if (err < 0) + return err; + + err = scarlett2_update_autogain(mixer); + if (err < 0) + return err; + + err = scarlett2_update_input_safe(mixer); + if (err < 0) + return err; + for (i = 0; i < private->num_mix_out; i++) { err = scarlett2_usb_get_mix(mixer, i); if (err < 0) @@ -4970,6 +5681,89 @@ static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) scarlett2_notify_input_phantom(mixer); } +/* Notify on input select change */ +static __always_unused void scarlett2_notify_input_select( + struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + if (!info->gain_input_count) + return; + + private->input_select_updated = 1; + + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, + &private->input_select_ctl->id); + + for (i = 0; i < info->gain_input_count / 2; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->input_link_ctls[i]->id); +} + +/* Notify on input gain change */ +static __always_unused void scarlett2_notify_input_gain( + struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + if (!info->gain_input_count) + return; + + private->input_gain_updated = 1; + + for (i = 0; i < info->gain_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->input_gain_ctls[i]->id); +} + +/* Notify on autogain change */ +static __always_unused void scarlett2_notify_autogain( + struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + if (!info->gain_input_count) + return; + + private->autogain_updated = 1; + + for (i = 0; i < info->gain_input_count; i++) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->autogain_ctls[i]->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->autogain_status_ctls[i]->id); + } +} + +/* Notify on input safe switch change */ +static __always_unused void scarlett2_notify_input_safe( + struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + if (!info->gain_input_count) + return; + + private->input_safe_updated = 1; + + for (i = 0; i < info->gain_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->safe_ctls[i]->id); +} + /* Notify on "monitor other" change (speaker switching, talkback) */ static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer) { From a1ed1d6caf680e22b933d1fa89f3d4a4c313d91e Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:37:18 +1030 Subject: [PATCH 306/337] ALSA: scarlett2: Minor refactor MSD mode check Create local variable for storing private data pointer in snd_scarlett2_controls_create(). It's currently only used for checking msd_switch, but it will be used again. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/ab50939dca0fdc5fa3493fc8eee3a2fdefffa62d.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index ab42d8f90e46..3627ffa52265 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -5898,6 +5898,7 @@ static int snd_scarlett2_controls_create( struct usb_mixer_interface *mixer, const struct scarlett2_device_entry *entry) { + struct scarlett2_data *private; int err; /* Initialise private data */ @@ -5905,6 +5906,8 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; + private = mixer->private_data; + /* Send proprietary USB initialisation sequence */ err = scarlett2_usb_init(mixer); if (err < 0) @@ -5931,7 +5934,7 @@ static int snd_scarlett2_controls_create( return err; /* If MSD mode is enabled, don't create any other controls */ - if (((struct scarlett2_data *)mixer->private_data)->msd_switch) + if (private->msd_switch) return 0; /* Create the analogue output controls */ From 0d2e791db4c81d295334ca09ad30cc2083f94b04 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:37:36 +1030 Subject: [PATCH 307/337] ALSA: scarlett2: Disable input controls while autogain is running While the autogain function is running, the other input controls (select, link, gain, safe, level, air, and phantom) can't be modified. Update those controls to be read-only during this time. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/ad13bacae77860de8c2d7c89f6ec2a1ee104e65f.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 298 +++++++++++++++++++++++++++++++++--- 1 file changed, 273 insertions(+), 25 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 3627ffa52265..e1b7398fae33 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2391,6 +2391,30 @@ static int scarlett2_add_sync_ctl(struct usb_mixer_interface *mixer) /*** Autogain Switch and Status Controls ***/ +/* Set the access mode of a control to read-only (val = 0) or + * read-write (val = 1). + */ +static void scarlett2_set_ctl_access(struct snd_kcontrol *kctl, int val) +{ + if (val) + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_WRITE; + else + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_WRITE; +} + +/* Check if autogain is running on any input */ +static int scarlett2_autogain_is_running(struct scarlett2_data *private) +{ + int i; + + for (i = 0; i < private->info->gain_input_count; i++) + if (private->autogain_status[i] == + SCARLETT2_AUTOGAIN_STATUS_RUNNING) + return 1; + + return 0; +} + static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; @@ -2438,13 +2462,108 @@ static int scarlett2_update_autogain(struct usb_mixer_interface *mixer) return 0; } +/* Update access mode for controls affected by autogain */ +static void scarlett2_autogain_update_access(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int val = !scarlett2_autogain_is_running(private); + int i; + + scarlett2_set_ctl_access(private->input_select_ctl, val); + for (i = 0; i < info->gain_input_count / 2; i++) + scarlett2_set_ctl_access(private->input_link_ctls[i], val); + for (i = 0; i < info->gain_input_count; i++) { + scarlett2_set_ctl_access(private->input_gain_ctls[i], val); + scarlett2_set_ctl_access(private->safe_ctls[i], val); + } + for (i = 0; i < info->level_input_count; i++) + scarlett2_set_ctl_access(private->level_ctls[i], val); + for (i = 0; i < info->air_input_count; i++) + scarlett2_set_ctl_access(private->air_ctls[i], val); + for (i = 0; i < info->phantom_count; i++) + scarlett2_set_ctl_access(private->phantom_ctls[i], val); +} + +/* Notify of access mode change for all controls read-only while + * autogain runs. + */ +static void scarlett2_autogain_notify_access(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->input_select_ctl->id); + for (i = 0; i < info->gain_input_count / 2; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->input_link_ctls[i]->id); + for (i = 0; i < info->gain_input_count; i++) { + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->input_gain_ctls[i]->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->safe_ctls[i]->id); + } + for (i = 0; i < info->level_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->level_ctls[i]->id); + for (i = 0; i < info->air_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->air_ctls[i]->id); + for (i = 0; i < info->phantom_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->phantom_ctls[i]->id); +} + +/* Call scarlett2_update_autogain() and + * scarlett2_autogain_update_access() if autogain_updated is set. + */ +static int scarlett2_check_autogain_updated( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err; + + if (!private->autogain_updated) + return 0; + + err = scarlett2_update_autogain(mixer); + if (err < 0) + return err; + + scarlett2_autogain_update_access(mixer); + + return 0; +} + +/* If autogain_updated is set when a *_ctl_put() function for a + * control that is meant to be read-only while autogain is running, + * update the autogain status and access mode of affected controls. + * Return -EPERM if autogain is running. + */ +static int scarlett2_check_put_during_autogain( + struct usb_mixer_interface *mixer) +{ + int err = scarlett2_check_autogain_updated(mixer); + + if (err < 0) + return err; + + if (scarlett2_autogain_is_running(mixer->private_data)) + return -EPERM; + + return 0; +} + static int scarlett2_autogain_switch_ctl_get( struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - int err = 0; + int err; mutex_lock(&private->data_mutex); @@ -2453,11 +2572,10 @@ static int scarlett2_autogain_switch_ctl_get( goto unlock; } - if (private->autogain_updated) { - err = scarlett2_update_autogain(mixer); - if (err < 0) - goto unlock; - } + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; + ucontrol->value.enumerated.item[0] = private->autogain_switch[elem->control]; @@ -2472,7 +2590,7 @@ static int scarlett2_autogain_status_ctl_get( struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - int err = 0; + int err; mutex_lock(&private->data_mutex); @@ -2481,11 +2599,10 @@ static int scarlett2_autogain_status_ctl_get( goto unlock; } - if (private->autogain_updated) { - err = scarlett2_update_autogain(mixer); - if (err < 0) - goto unlock; - } + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; + ucontrol->value.enumerated.item[0] = private->autogain_status[elem->control]; @@ -2525,6 +2642,9 @@ static int scarlett2_autogain_switch_ctl_put( if (err == 0) err = 1; + scarlett2_autogain_update_access(mixer); + scarlett2_autogain_notify_access(mixer); + unlock: mutex_unlock(&private->data_mutex); return err; @@ -2624,7 +2744,7 @@ static int scarlett2_input_select_ctl_put( struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - int oval, val, err = 0; + int oval, val, err; int max_val = private->input_link_switch[0] ? 0 : 1; mutex_lock(&private->data_mutex); @@ -2634,6 +2754,10 @@ static int scarlett2_input_select_ctl_put( goto unlock; } + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + oval = private->input_select_switch; val = ucontrol->value.integer.value[0]; @@ -2677,6 +2801,15 @@ static int scarlett2_input_select_ctl_info( mutex_lock(&private->data_mutex); + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; + /* Loop through each input * Linked inputs have one value for the pair */ @@ -2694,6 +2827,7 @@ static int scarlett2_input_select_ctl_info( err = snd_ctl_enum_info(uinfo, 1, j, (const char * const *)values); +unlock: mutex_unlock(&private->data_mutex); for (i = 0; i < inputs; i++) @@ -2713,6 +2847,35 @@ static const struct snd_kcontrol_new scarlett2_input_select_ctl = { /*** Input Link Switch Controls ***/ +/* snd_ctl_boolean_mono_info() with autogain-updated check + * (for controls that are read-only while autogain is running) + */ +static int scarlett2_autogain_disables_ctl_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; + + err = snd_ctl_boolean_mono_info(kctl, uinfo); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + static int scarlett2_input_link_ctl_get( struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -2749,7 +2912,7 @@ static int scarlett2_input_link_ctl_put( struct scarlett2_data *private = mixer->private_data; int index = elem->control; - int oval, val, err = 0; + int oval, val, err; mutex_lock(&private->data_mutex); @@ -2758,6 +2921,10 @@ static int scarlett2_input_link_ctl_put( goto unlock; } + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + oval = private->input_link_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -2789,7 +2956,7 @@ unlock: static const struct snd_kcontrol_new scarlett2_input_link_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", - .info = snd_ctl_boolean_mono_info, + .info = scarlett2_autogain_disables_ctl_info, .get = scarlett2_input_link_ctl_get, .put = scarlett2_input_link_ctl_put }; @@ -2815,13 +2982,30 @@ static int scarlett2_input_gain_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = elem->channels; uinfo->value.integer.min = 0; uinfo->value.integer.max = SCARLETT2_GAIN_BIAS; uinfo->value.integer.step = 1; - return 0; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_input_gain_ctl_get(struct snd_kcontrol *kctl, @@ -2860,7 +3044,7 @@ static int scarlett2_input_gain_ctl_put(struct snd_kcontrol *kctl, struct scarlett2_data *private = mixer->private_data; int index = elem->control; - int oval, val, err = 0; + int oval, val, err; mutex_lock(&private->data_mutex); @@ -2869,6 +3053,10 @@ static int scarlett2_input_gain_ctl_put(struct snd_kcontrol *kctl, goto unlock; } + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + oval = private->gain[index]; val = ucontrol->value.integer.value[0]; @@ -2957,7 +3145,7 @@ static int scarlett2_safe_ctl_put(struct snd_kcontrol *kctl, struct scarlett2_data *private = mixer->private_data; int index = elem->control; - int oval, val, err = 0; + int oval, val, err; mutex_lock(&private->data_mutex); @@ -2966,6 +3154,10 @@ static int scarlett2_safe_ctl_put(struct snd_kcontrol *kctl, goto unlock; } + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + oval = private->safe_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -2988,7 +3180,7 @@ unlock: static const struct snd_kcontrol_new scarlett2_safe_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", - .info = snd_ctl_boolean_mono_info, + .info = scarlett2_autogain_disables_ctl_info, .get = scarlett2_safe_ctl_get, .put = scarlett2_safe_ctl_put, }; @@ -3434,8 +3626,27 @@ static int scarlett2_level_enum_ctl_info(struct snd_kcontrol *kctl, static const char *const values[2] = { "Line", "Inst" }; + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; - return snd_ctl_enum_info(uinfo, 1, 2, values); + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; + + err = snd_ctl_enum_info(uinfo, 1, 2, values); + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_level_enum_ctl_get(struct snd_kcontrol *kctl, @@ -3478,7 +3689,7 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, const struct scarlett2_device_info *info = private->info; int index = elem->control + info->level_input_first; - int oval, val, err = 0; + int oval, val, err; mutex_lock(&private->data_mutex); @@ -3487,6 +3698,10 @@ static int scarlett2_level_enum_ctl_put(struct snd_kcontrol *kctl, goto unlock; } + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + oval = private->level_switch[index]; val = !!ucontrol->value.enumerated.item[0]; @@ -3659,7 +3874,7 @@ static int scarlett2_air_ctl_put(struct snd_kcontrol *kctl, struct scarlett2_data *private = mixer->private_data; int index = elem->control; - int oval, val, err = 0; + int oval, val, err; mutex_lock(&private->data_mutex); @@ -3668,6 +3883,10 @@ static int scarlett2_air_ctl_put(struct snd_kcontrol *kctl, goto unlock; } + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + oval = private->air_switch[index]; val = ucontrol->value.integer.value[0]; @@ -3693,8 +3912,27 @@ static int scarlett2_air_with_drive_ctl_info( static const char *const values[3] = { "Off", "Presence", "Presence + Drive" }; + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; - return snd_ctl_enum_info(uinfo, 1, 3, values); + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + err = scarlett2_check_autogain_updated(mixer); + if (err < 0) + goto unlock; + + err = snd_ctl_enum_info(uinfo, 1, 3, values); + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static const struct snd_kcontrol_new scarlett2_air_ctl[2] = { @@ -3782,7 +4020,7 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, const struct scarlett2_device_info *info = private->info; int index = elem->control; - int oval, val, err = 0; + int oval, val, err; mutex_lock(&private->data_mutex); @@ -3791,6 +4029,10 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, goto unlock; } + err = scarlett2_check_put_during_autogain(mixer); + if (err < 0) + goto unlock; + oval = private->phantom_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -3817,7 +4059,7 @@ unlock: static const struct snd_kcontrol_new scarlett2_phantom_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", - .info = snd_ctl_boolean_mono_info, + .info = scarlett2_autogain_disables_ctl_info, .get = scarlett2_phantom_ctl_get, .put = scarlett2_phantom_ctl_put, }; @@ -5743,6 +5985,8 @@ static __always_unused void scarlett2_notify_autogain( snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->autogain_status_ctls[i]->id); } + + scarlett2_autogain_notify_access(mixer); } /* Notify on input safe switch change */ @@ -5987,6 +6231,10 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; + /* Set the access mode of controls disabled during autogain */ + if (private->info->gain_input_count) + scarlett2_autogain_update_access(mixer); + /* Set up the interrupt polling */ err = scarlett2_init_notify(mixer); if (err < 0) From 882a2a36c41df96d0449c3751dceb86415b44a56 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:37:56 +1030 Subject: [PATCH 308/337] ALSA: scarlett2: Disable autogain during phantom power state change When phantom power is enabled or disabled, the autogain control cannot be enabled until the interface has signalled that the change is complete and the input is unmuted. Update those controls to be read-only during this time. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/f49f7bf9358e1f20713d95d407d8d6a436859877.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 123 +++++++++++++++++++++++++++++++++--- 1 file changed, 113 insertions(+), 10 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index e1b7398fae33..6266f5d3fab8 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -2391,6 +2391,10 @@ static int scarlett2_add_sync_ctl(struct usb_mixer_interface *mixer) /*** Autogain Switch and Status Controls ***/ +/* Forward declarations as phantom power and autogain can disable each other */ +static int scarlett2_check_input_phantom_updated(struct usb_mixer_interface *); +static int scarlett2_phantom_is_switching(struct scarlett2_data *, int); + /* Set the access mode of a control to read-only (val = 0) or * read-write (val = 1). */ @@ -2557,6 +2561,27 @@ static int scarlett2_check_put_during_autogain( return 0; } +static int scarlett2_autogain_switch_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err; + + mutex_lock(&private->data_mutex); + + err = scarlett2_check_input_phantom_updated(mixer); + if (err < 0) + goto unlock; + + err = snd_ctl_boolean_mono_info(kctl, uinfo); + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + static int scarlett2_autogain_switch_ctl_get( struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -2619,7 +2644,7 @@ static int scarlett2_autogain_switch_ctl_put( struct scarlett2_data *private = mixer->private_data; int index = elem->control; - int oval, val, err = 0; + int oval, val, err; mutex_lock(&private->data_mutex); @@ -2628,6 +2653,15 @@ static int scarlett2_autogain_switch_ctl_put( goto unlock; } + err = scarlett2_check_input_phantom_updated(mixer); + if (err < 0) + goto unlock; + + if (scarlett2_phantom_is_switching(private, index)) { + err = -EPERM; + goto unlock; + } + oval = private->autogain_switch[index]; val = !!ucontrol->value.integer.value[0]; @@ -2664,7 +2698,7 @@ static int scarlett2_autogain_status_ctl_info( static const struct snd_kcontrol_new scarlett2_autogain_switch_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "", - .info = snd_ctl_boolean_mono_info, + .info = scarlett2_autogain_switch_ctl_info, .get = scarlett2_autogain_switch_ctl_get, .put = scarlett2_autogain_switch_ctl_put }; @@ -3983,13 +4017,74 @@ static int scarlett2_update_input_phantom(struct usb_mixer_interface *mixer) return 0; } +/* Check if phantom power on the given input is currently changing state */ +static int scarlett2_phantom_is_switching( + struct scarlett2_data *private, int line_num) +{ + const struct scarlett2_device_info *info = private->info; + int index = line_num / info->inputs_per_phantom; + + return !!(private->phantom_switch[index] & 0x02); +} + +/* Update autogain controls' access mode when phantom power changes state */ +static void scarlett2_phantom_update_access(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + /* Disable autogain controls if phantom power is changing state */ + for (i = 0; i < info->gain_input_count; i++) { + int val = !scarlett2_phantom_is_switching(private, i); + + scarlett2_set_ctl_access(private->autogain_ctls[i], val); + } +} + +/* Notify of access mode change for autogain which can't be enabled + * while phantom power is changing. + */ +static void scarlett2_phantom_notify_access(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; + int i; + + for (i = 0; i < info->gain_input_count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + &private->autogain_ctls[i]->id); +} + +/* Call scarlett2_update_input_phantom() and + * scarlett2_phantom_update_access() if input_phantom_updated is set. + */ +static int scarlett2_check_input_phantom_updated( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err; + + if (!private->input_phantom_updated) + return 0; + + err = scarlett2_update_input_phantom(mixer); + if (err < 0) + return err; + + scarlett2_phantom_update_access(mixer); + + return 0; +} + static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; struct usb_mixer_interface *mixer = elem->head.mixer; struct scarlett2_data *private = mixer->private_data; - int err = 0; + int err; mutex_lock(&private->data_mutex); @@ -3998,11 +4093,10 @@ static int scarlett2_phantom_ctl_get(struct snd_kcontrol *kctl, goto unlock; } - if (private->input_phantom_updated) { - err = scarlett2_update_input_phantom(mixer); - if (err < 0) - goto unlock; - } + err = scarlett2_check_input_phantom_updated(mixer); + if (err < 0) + goto unlock; + ucontrol->value.integer.value[0] = scarlett2_decode_muteable( private->phantom_switch[elem->control]); @@ -4051,6 +4145,9 @@ static int scarlett2_phantom_ctl_put(struct snd_kcontrol *kctl, if (err == 0) err = 1; + scarlett2_phantom_update_access(mixer); + scarlett2_phantom_notify_access(mixer); + unlock: mutex_unlock(&private->data_mutex); return err; @@ -5912,6 +6009,8 @@ static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer) for (i = 0; i < info->phantom_count; i++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->phantom_ctls[i]->id); + + scarlett2_phantom_notify_access(mixer); } /* Notify on "input other" change (level/pad/air/phantom) */ @@ -6231,9 +6330,13 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; - /* Set the access mode of controls disabled during autogain */ - if (private->info->gain_input_count) + /* Set the access mode of controls disabled during + * autogain/phantom power switching. + */ + if (private->info->gain_input_count) { scarlett2_autogain_update_access(mixer); + scarlett2_phantom_update_access(mixer); + } /* Set up the interrupt polling */ err = scarlett2_init_notify(mixer); From d7cfa2fdfc8a0ba9bf7d5ebd5cf25e6d24501632 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:02 +1030 Subject: [PATCH 309/337] ALSA: scarlett2: Add power status control Add a control to retrieve the power status from the interface (bus-powered, external-powered, or insufficient power). Mark the new scarlett2_notify_power_status() function with __always_unused until it gets used when the Gen 4 notification callback function arrays are added. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/d9806bc41adc45b1c19749562fec7765ba24351d.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 123 ++++++++++++++++++++++++++++++++++++ 1 file changed, 123 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 6266f5d3fab8..ce842c29be5e 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -278,6 +278,14 @@ enum { SCARLETT2_AUTOGAIN_STATUS_COUNT }; +/* Power Status Values */ +enum { + SCARLETT2_POWER_STATUS_EXT, + SCARLETT2_POWER_STATUS_BUS, + SCARLETT2_POWER_STATUS_FAIL, + SCARLETT2_POWER_STATUS_COUNT +}; + /* Notification callback functions */ struct scarlett2_notification { u32 mask; @@ -334,6 +342,8 @@ enum { SCARLETT2_CONFIG_SAFE_SWITCH, SCARLETT2_CONFIG_INPUT_SELECT_SWITCH, SCARLETT2_CONFIG_INPUT_LINK_SWITCH, + SCARLETT2_CONFIG_POWER_EXT, + SCARLETT2_CONFIG_POWER_STATUS, SCARLETT2_CONFIG_COUNT }; @@ -751,6 +761,7 @@ struct scarlett2_data { u8 direct_monitor_updated; u8 mux_updated; u8 speaker_switching_switched; + u8 power_status_updated; u8 sync; u8 master_vol; u8 vol[SCARLETT2_ANALOGUE_MAX]; @@ -774,6 +785,7 @@ struct scarlett2_data { u8 talkback_map[SCARLETT2_OUTPUT_MIX_MAX]; u8 msd_switch; u8 standalone_switch; + u8 power_status; u8 meter_level_map[SCARLETT2_MAX_METERS]; struct snd_kcontrol *sync_ctl; struct snd_kcontrol *master_vol_ctl; @@ -795,6 +807,7 @@ struct scarlett2_data { struct snd_kcontrol *direct_monitor_ctl; struct snd_kcontrol *speaker_switching_ctl; struct snd_kcontrol *talkback_ctl; + struct snd_kcontrol *power_status_ctl; u8 mux[SCARLETT2_MUX_MAX]; u8 mix[SCARLETT2_MIX_MAX]; }; @@ -5526,6 +5539,91 @@ static int scarlett2_add_standalone_ctl(struct usb_mixer_interface *mixer) 0, 1, "Standalone Switch", NULL); } +/*** Power Status ***/ + +static int scarlett2_update_power_status(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err; + u8 power_ext; + u8 power_status; + + private->power_status_updated = 0; + + err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_POWER_EXT, + 1, &power_ext); + if (err < 0) + return err; + + err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_POWER_STATUS, + 1, &power_status); + if (err < 0) + return err; + + if (power_status > 1) + private->power_status = SCARLETT2_POWER_STATUS_FAIL; + else if (power_ext) + private->power_status = SCARLETT2_POWER_STATUS_EXT; + else + private->power_status = SCARLETT2_POWER_STATUS_BUS; + + return 0; +} + +static int scarlett2_power_status_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->power_status_updated) { + err = scarlett2_update_power_status(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = private->power_status; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_power_status_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[3] = { + "External", "Bus", "Fail" + }; + + return snd_ctl_enum_info(uinfo, 1, 3, values); +} + +static const struct snd_kcontrol_new scarlett2_power_status_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .name = "", + .info = scarlett2_power_status_ctl_info, + .get = scarlett2_power_status_ctl_get, +}; + +static int scarlett2_add_power_status_ctl(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + + if (!scarlett2_has_config_item(private, + SCARLETT2_CONFIG_POWER_EXT)) + return 0; + + /* Add power status control */ + return scarlett2_add_new_ctl(mixer, &scarlett2_power_status_ctl, + 0, 1, "Power Status Card Enum", + &private->power_status_ctl); +} + /*** Cleanup/Suspend Callbacks ***/ static void scarlett2_private_free(struct usb_mixer_interface *mixer) @@ -5817,6 +5915,13 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) return err; } + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_POWER_EXT)) { + err = scarlett2_update_power_status(mixer); + if (err < 0) + return err; + } + err = scarlett2_update_sync(mixer); if (err < 0) return err; @@ -6153,6 +6258,19 @@ static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer) &private->direct_monitor_ctl->id); } +/* Notify on power change */ +static __always_unused void scarlett2_notify_power_status( + struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + + private->power_status_updated = 1; + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->power_status_ctl->id); +} + /* Interrupt callback */ static void scarlett2_notify(struct urb *urb) { @@ -6330,6 +6448,11 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; + /* Create the power status control */ + err = scarlett2_add_power_status_ctl(mixer); + if (err < 0) + return err; + /* Set the access mode of controls disabled during * autogain/phantom power switching. */ From ac19be067aac38479b1c4d1477d4012a24d5730d Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:13 +1030 Subject: [PATCH 310/337] ALSA: scarlett2: Store mix_ctls for Gen 4 Direct Monitor The Scarlett 4th Gen small interfaces have a software-controllable mixer like the large 2nd and 3rd Gen interfaces do. Pressing the "Direct" button on the interface updates the mixer controls, which this driver hasn't needed to deal with previously. This commit stores the ALSA mixer controls, and adds a mix_updated flag so that the controls can be updated when a notification is received. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/3ba27c60230511b80b0fa75727551ea70f17d829.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 50 +++++++++++++++++++++++++++++++------ 1 file changed, 42 insertions(+), 8 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index ce842c29be5e..d6e2f69c32b9 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -760,6 +760,7 @@ struct scarlett2_data { u8 monitor_other_updated; u8 direct_monitor_updated; u8 mux_updated; + u8 mix_updated; u8 speaker_switching_switched; u8 power_status_updated; u8 sync; @@ -804,6 +805,7 @@ struct scarlett2_data { struct snd_kcontrol *autogain_status_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *safe_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX]; + struct snd_kcontrol *mix_ctls[SCARLETT2_MIX_MAX]; struct snd_kcontrol *direct_monitor_ctl; struct snd_kcontrol *speaker_switching_ctl; struct snd_kcontrol *talkback_ctl; @@ -4960,6 +4962,22 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) /*** Mixer Volume Controls ***/ +static int scarlett2_update_mix(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int i, err; + + private->mix_updated = 0; + + for (i = 0; i < private->num_mix_out; i++) { + err = scarlett2_usb_get_mix(mixer, i); + if (err < 0) + return err; + } + + return 1; +} + static int scarlett2_mixer_ctl_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { @@ -4977,10 +4995,27 @@ static int scarlett2_mixer_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_data *private = elem->head.mixer->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->mix_updated) { + err = scarlett2_update_mix(mixer); + if (err < 0) + goto unlock; + } ucontrol->value.integer.value[0] = private->mix[elem->control]; - return 0; + +unlock: + mutex_unlock(&private->data_mutex); + return err; } static int scarlett2_mixer_ctl_put(struct snd_kcontrol *kctl, @@ -5048,7 +5083,8 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) "Mix %c Input %02d Playback Volume", 'A' + i, j + 1); err = scarlett2_add_new_ctl(mixer, &scarlett2_mixer_ctl, - index, 1, s, NULL); + index, 1, s, + &private->mix_ctls[index]); if (err < 0) return err; } @@ -5993,11 +6029,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; - for (i = 0; i < private->num_mix_out; i++) { - err = scarlett2_usb_get_mix(mixer, i); - if (err < 0) - return err; - } + err = scarlett2_update_mix(mixer); + if (err < 0) + return err; return scarlett2_usb_get_mux(mixer); } From e8e14270d8d0d5c9bd8e04e5940d511b56a981e9 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:20 +1030 Subject: [PATCH 311/337] ALSA: scarlett2: Handle Gen 4 Direct Monitor mix updates When the Direct Monitor feature on the Scarlett 4th Gen Solo and 2i2 interfaces is used, the Mix A and B gains are updated by the interface. This patch calls snd_ctl_notify() for the ALSA mix controls when a Direct Monitor notification is received. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/713d032e343e0547212368919bef17d6fa1c9d29.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index d6e2f69c32b9..501444cf0f15 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -6285,11 +6285,23 @@ static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; + int count = private->num_mix_in * private->num_mix_out; + int i; private->direct_monitor_updated = 1; snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &private->direct_monitor_ctl->id); + + if (!scarlett2_has_mixer(private)) + return; + + private->mix_updated = 1; + + /* Notify of change to the mix controls */ + for (i = 0; i < count; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->mix_ctls[i]->id); } /* Notify on power change */ From c6c9f0cf9dba7e86f1f6cca658bb6b86c5d02530 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:28 +1030 Subject: [PATCH 312/337] ALSA: scarlett2: Add support for custom Gen 4 Direct Monitor mixes The mixes used by Direct Monitor feature on the Scarlett 4th Gen Solo and 2i2 interfaces are configurable. This patch adds ALSA controls for the gains which are copied to the mixer when Direct Monitor is enabled. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/96282a805b45f04560e5923d170745363906b7f3.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 145 +++++++++++++++++++++++++++++++++++- 1 file changed, 141 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 501444cf0f15..13f65d963778 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -213,6 +213,13 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { /* Maximum number of mixer gain controls */ #define SCARLETT2_MIX_MAX (SCARLETT2_INPUT_MIX_MAX * SCARLETT2_OUTPUT_MIX_MAX) +/* Maximum number of direct monitor mixer gain controls + * 1 (Solo) or 2 (2i2) direct monitor selections (Mono & Stereo) + * 2 Mix outputs (A/Left & B/Right) + * 4 Mix inputs + */ +#define SCARLETT2_MONITOR_MIX_MAX (2 * 2 * 4) + /* Maximum size of the data in the USB mux assignment message: * 20 inputs, 20 outputs, 25 matrix inputs, 12 spare */ @@ -344,6 +351,7 @@ enum { SCARLETT2_CONFIG_INPUT_LINK_SWITCH, SCARLETT2_CONFIG_POWER_EXT, SCARLETT2_CONFIG_POWER_STATUS, + SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN, SCARLETT2_CONFIG_COUNT }; @@ -742,6 +750,7 @@ struct scarlett2_data { u8 num_mix_in; u8 num_mix_out; u8 num_line_out; + u8 num_monitor_mix_ctls; u32 firmware_version; u8 flash_segment_nums[SCARLETT2_SEGMENT_ID_COUNT]; u8 flash_segment_blocks[SCARLETT2_SEGMENT_ID_COUNT]; @@ -812,6 +821,7 @@ struct scarlett2_data { struct snd_kcontrol *power_status_ctl; u8 mux[SCARLETT2_MUX_MAX]; u8 mix[SCARLETT2_MIX_MAX]; + u8 monitor_mix[SCARLETT2_MONITOR_MIX_MAX]; }; /*** Model-specific data ***/ @@ -5108,6 +5118,28 @@ static int scarlett2_update_direct_monitor(struct usb_mixer_interface *mixer) 1, &private->direct_monitor_switch); } +static int scarlett2_update_monitor_mix(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err, i; + u16 mix_values[SCARLETT2_MONITOR_MIX_MAX]; + + if (!private->num_monitor_mix_ctls) + return 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN, + private->num_monitor_mix_ctls, mix_values); + if (err < 0) + return err; + + for (i = 0; i < private->num_monitor_mix_ctls; i++) + private->monitor_mix[i] = scarlett2_mixer_value_to_db( + mix_values[i]); + + return 0; +} + static int scarlett2_direct_monitor_ctl_get( struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -5201,11 +5233,70 @@ static const struct snd_kcontrol_new scarlett2_direct_monitor_ctl[2] = { } }; -static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer) +static int scarlett2_monitor_mix_ctl_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->monitor_mix[elem->control]; + + return 0; +} + +static int scarlett2_monitor_mix_ctl_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int oval, val, err = 0; + int index = elem->control; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->monitor_mix[index]; + val = clamp(ucontrol->value.integer.value[0], + 0L, (long)SCARLETT2_MIXER_MAX_VALUE); + + if (oval == val) + goto unlock; + + private->monitor_mix[index] = val; + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN, + index, scarlett2_mixer_values[val]); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static const struct snd_kcontrol_new scarlett2_monitor_mix_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_mixer_ctl_info, + .get = scarlett2_monitor_mix_ctl_get, + .put = scarlett2_monitor_mix_ctl_put, + .private_value = SCARLETT2_MIXER_MAX_DB, /* max value */ + .tlv = { .p = db_scale_scarlett2_mixer } +}; + +static int scarlett2_add_direct_monitor_ctls(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; const char *s; + int err, i, j, k, index; if (!info->direct_monitor) return 0; @@ -5214,9 +5305,47 @@ static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer) ? "Direct Monitor Playback Switch" : "Direct Monitor Playback Enum"; - return scarlett2_add_new_ctl( + err = scarlett2_add_new_ctl( mixer, &scarlett2_direct_monitor_ctl[info->direct_monitor - 1], 0, 1, s, &private->direct_monitor_ctl); + if (err < 0) + return err; + + if (!private->num_monitor_mix_ctls) + return 0; + + /* 1 or 2 direct monitor selections (Mono & Stereo) */ + for (i = 0, index = 0; i < info->direct_monitor; i++) { + const char * const format = + "Monitor %sMix %c Input %02d Playback Volume"; + const char *mix_type; + + if (info->direct_monitor == 1) + mix_type = ""; + else if (i == 0) + mix_type = "1 "; + else + mix_type = "2 "; + + /* 2 Mix outputs, A/Left & B/Right */ + for (j = 0; j < 2; j++) + + /* Mix inputs */ + for (k = 0; k < private->num_mix_in; k++, index++) { + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + + snprintf(name, sizeof(name), format, + mix_type, 'A' + j, k + 1); + + err = scarlett2_add_new_ctl( + mixer, &scarlett2_monitor_mix_ctl, + index, 1, name, NULL); + if (err < 0) + return err; + } + } + + return 0; } /*** Mux Source Selection Controls ***/ @@ -5708,6 +5837,10 @@ static void scarlett2_count_io(struct scarlett2_data *private) /* Number of analogue line outputs */ private->num_line_out = port_count[SCARLETT2_PORT_TYPE_ANALOGUE][SCARLETT2_PORT_OUT]; + + /* Number of monitor mix controls */ + private->num_monitor_mix_ctls = + info->direct_monitor * 2 * private->num_mix_in; } /* Look through the interface descriptors for the Focusrite Control @@ -5938,6 +6071,10 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (!scarlett2_has_mixer(private)) return 0; + err = scarlett2_update_monitor_mix(mixer); + if (err < 0) + return err; + err = scarlett2_update_monitor_other(mixer); if (err < 0) return err; @@ -6474,8 +6611,8 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; - /* Create the direct monitor control */ - err = scarlett2_add_direct_monitor_ctl(mixer); + /* Create the direct monitor control(s) */ + err = scarlett2_add_direct_monitor_ctls(mixer); if (err < 0) return err; From 166e1dfb752665226ab482b62a94c274ec2dea17 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:34 +1030 Subject: [PATCH 313/337] ALSA: scarlett2: Add support for DSP mux channels The DSP mux channels in the Scarlett 4th Gen appear as SCARLETT2_PORT_TYPE_MIX ports but do not have corresponding mixer controls. Add a dsp_count option to the device info struct to exclude those DSP channels from the num_mix_in/num_mix_out counts. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/b78bdb1a7624d55783f5bf0e1ffbfa47a9e9a800.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 13f65d963778..59f178dc8050 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -706,6 +706,9 @@ struct scarlett2_device_info { */ u8 direct_monitor; + /* the number of DSP channels */ + u8 dsp_count; + /* remap analogue outputs; 18i8 Gen 3 has "line 3/4" connected * internally to the analogue 7/8 outputs */ @@ -5827,12 +5830,17 @@ static void scarlett2_count_io(struct scarlett2_data *private) private->num_mux_srcs = srcs; private->num_mux_dsts = dsts; - /* Mixer inputs are mux outputs and vice versa */ + /* Mixer inputs are mux outputs and vice versa. + * Scarlett Gen 4 DSP I/O uses SCARLETT2_PORT_TYPE_MIX but + * doesn't have mixer controls. + */ private->num_mix_in = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT]; + port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_OUT] - + info->dsp_count; private->num_mix_out = - port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN]; + port_count[SCARLETT2_PORT_TYPE_MIX][SCARLETT2_PORT_IN] - + info->dsp_count; /* Number of analogue line outputs */ private->num_line_out = From 66946398a4be9fd41eafcc844a306273f5150e69 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:38 +1030 Subject: [PATCH 314/337] ALSA: scarlett2: Rename DSP mux channels The DSP mux channels are of type SCARLETT2_PORT_TYPE_MIX so the ALSA controls would refer to them "Mix X" and "Mixer Input X". This patch fixes them to be called "DSP X" and "DSP Input X". Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/2d91d0a74d5c7f6179e950bed2c80a4498d16649.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 32 ++++++++++++++++++++++++++------ 1 file changed, 26 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 59f178dc8050..a24fd6e867ed 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -583,6 +583,8 @@ struct scarlett2_port { const char * const src_descr; int src_num_offset; const char * const dst_descr; + const char * const dsp_src_descr; + const char * const dsp_dst_descr; }; static const struct scarlett2_port scarlett2_ports[SCARLETT2_PORT_TYPE_COUNT] = { @@ -612,7 +614,9 @@ static const struct scarlett2_port scarlett2_ports[SCARLETT2_PORT_TYPE_COUNT] = .id = 0x300, .src_descr = "Mix %c", .src_num_offset = 'A', - .dst_descr = "Mixer Input %02d Capture" + .dst_descr = "Mixer Input %02d Capture", + .dsp_src_descr = "DSP %d", + .dsp_dst_descr = "DSP Input %d Capture" }, [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, @@ -5378,8 +5382,16 @@ static int scarlett2_mux_src_enum_ctl_info(struct snd_kcontrol *kctl, const struct scarlett2_port *port = &scarlett2_ports[port_type]; - sprintf(uinfo->value.enumerated.name, - port->src_descr, item + port->src_num_offset); + if (port_type == SCARLETT2_PORT_TYPE_MIX && + item >= private->num_mix_out) + sprintf(uinfo->value.enumerated.name, + port->dsp_src_descr, + item - private->num_mix_out + 1); + else + sprintf(uinfo->value.enumerated.name, + port->src_descr, + item + port->src_num_offset); + return 0; } item -= port_count[port_type][SCARLETT2_PORT_IN]; @@ -5472,10 +5484,18 @@ static int scarlett2_add_mux_enums(struct usb_mixer_interface *mixer) channel++, i++) { int err; char s[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - const char *const descr = - scarlett2_ports[port_type].dst_descr; + int channel_num = channel + 1; + const struct scarlett2_port *port = + &scarlett2_ports[port_type]; + const char *descr = port->dst_descr; - snprintf(s, sizeof(s) - 5, descr, channel + 1); + if (port_type == SCARLETT2_PORT_TYPE_MIX && + channel >= private->num_mix_in) { + channel_num -= private->num_mix_in; + descr = port->dsp_dst_descr; + } + + snprintf(s, sizeof(s) - 5, descr, channel_num); strcat(s, " Enum"); err = scarlett2_add_new_ctl(mixer, From f6a817e6795a4f16c106ddf5f4009a7697515868 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:46 +1030 Subject: [PATCH 315/337] ALSA: scarlett2: Add minimum firmware version check Early firmware for the Scarlett Gen 4 devices has sufficient differences that it is better to enforce a minimum firmware version than to try and work around those differences. Add a minimum firmware version field to the device info struct, and display a message if the firmware version is too old. Only create the Firmware Version and MSD (optional) controls in this case. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/5455a7e54bda81556066abd7f761b10e9c5f8a16.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 73 ++++++++++++++++++++++++++++++++++--- 1 file changed, 67 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index a24fd6e867ed..f1c9f6b022ce 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -660,6 +660,9 @@ struct scarlett2_device_info { /* which set of configuration parameters the device uses */ const struct scarlett2_config_set *config_set; + /* minimum firmware version required */ + u16 min_firmware_version; + /* support for main/alt speaker switching */ u8 has_speaker_switching; @@ -2352,6 +2355,45 @@ static int scarlett2_add_firmware_version_ctl( 0, 0, "Firmware Version", NULL); } +/*** Minimum Firmware Version Control ***/ + +static int scarlett2_min_firmware_version_ctl_get( + struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct scarlett2_data *private = elem->head.mixer->private_data; + + ucontrol->value.integer.value[0] = private->info->min_firmware_version; + + return 0; +} + +static int scarlett2_min_firmware_version_ctl_info( + struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + + return 0; +} + +static const struct snd_kcontrol_new scarlett2_min_firmware_version_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .name = "", + .info = scarlett2_min_firmware_version_ctl_info, + .get = scarlett2_min_firmware_version_ctl_get +}; + +static int scarlett2_add_min_firmware_version_ctl( + struct usb_mixer_interface *mixer) +{ + return scarlett2_add_new_ctl(mixer, &scarlett2_min_firmware_version_ctl, + 0, 0, "Minimum Firmware Version", NULL); +} + /*** Sync Control ***/ /* Update sync control after receiving notification that the status @@ -6061,6 +6103,7 @@ static int scarlett2_get_flash_segment_nums(struct usb_mixer_interface *mixer) static int scarlett2_read_configs(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; + const struct scarlett2_device_info *info = private->info; int err, i; if (scarlett2_has_config_item(private, SCARLETT2_CONFIG_MSD_SWITCH)) { @@ -6069,12 +6112,22 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) 1, &private->msd_switch); if (err < 0) return err; - - /* no other controls are created if MSD mode is on */ - if (private->msd_switch) - return 0; } + if (private->firmware_version < info->min_firmware_version) { + usb_audio_err(mixer->chip, + "Focusrite %s firmware version %d is too old; " + "need %d", + private->series_name, + private->firmware_version, + info->min_firmware_version); + return 0; + } + + /* no other controls are created if MSD mode is on */ + if (private->msd_switch) + return 0; + err = scarlett2_update_input_level(mixer); if (err < 0) return err; @@ -6595,6 +6648,11 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; + /* Add minimum firmware version control */ + err = scarlett2_add_min_firmware_version_ctl(mixer); + if (err < 0) + return err; + /* Read volume levels and controls from the interface */ err = scarlett2_read_configs(mixer); if (err < 0) @@ -6605,8 +6663,11 @@ static int snd_scarlett2_controls_create( if (err < 0) return err; - /* If MSD mode is enabled, don't create any other controls */ - if (private->msd_switch) + /* If MSD mode is enabled, or if the firmware version is too + * old, don't create any other controls + */ + if (private->msd_switch || + private->firmware_version < private->info->min_firmware_version) return 0; /* Create the analogue output controls */ From 2ecca0df90cbd082ab61530bb4e758a0fb970d21 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:52 +1030 Subject: [PATCH 316/337] ALSA: scarlett2: Add R/O headphone volume control The Scarlett 4i4 Gen 4 adds a R/O headphone volume control in addition to a R/O master volume control (which is already supported). Mark the new scarlett2_notify_volume() function with __always_unused until it gets used when the Gen 4 notification callback function arrays are added. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/bd4a76da157f8cc3fbfa02eba96d02bdb86817c5.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 82 ++++++++++++++++++++++++++++++++++++- 1 file changed, 81 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index f1c9f6b022ce..94413fca2b6c 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -332,6 +332,7 @@ enum { SCARLETT2_CONFIG_MUTE_SWITCH, SCARLETT2_CONFIG_SW_HW_SWITCH, SCARLETT2_CONFIG_MASTER_VOLUME, + SCARLETT2_CONFIG_HEADPHONE_VOLUME, SCARLETT2_CONFIG_LEVEL_SWITCH, SCARLETT2_CONFIG_PAD_SWITCH, SCARLETT2_CONFIG_MSD_SWITCH, @@ -784,6 +785,7 @@ struct scarlett2_data { u8 power_status_updated; u8 sync; u8 master_vol; + u8 headphone_vol; u8 vol[SCARLETT2_ANALOGUE_MAX]; u8 vol_sw_hw_switch[SCARLETT2_ANALOGUE_MAX]; u8 mute_switch[SCARLETT2_ANALOGUE_MAX]; @@ -809,6 +811,7 @@ struct scarlett2_data { u8 meter_level_map[SCARLETT2_MAX_METERS]; struct snd_kcontrol *sync_ctl; struct snd_kcontrol *master_vol_ctl; + struct snd_kcontrol *headphone_vol_ctl; struct snd_kcontrol *vol_ctls[SCARLETT2_ANALOGUE_MAX]; struct snd_kcontrol *sw_hw_ctls[SCARLETT2_ANALOGUE_MAX]; struct snd_kcontrol *mute_ctls[SCARLETT2_ANALOGUE_MAX]; @@ -3324,6 +3327,18 @@ static int scarlett2_update_volumes(struct usb_mixer_interface *mixer) private->vol[i] = private->master_vol; } + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_HEADPHONE_VOLUME)) { + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_HEADPHONE_VOLUME, + 1, &vol); + if (err < 0) + return err; + + private->headphone_vol = clamp(vol + SCARLETT2_VOLUME_BIAS, + 0, SCARLETT2_VOLUME_BIAS); + } + return 0; } @@ -3367,6 +3382,34 @@ unlock: return err; } +static int scarlett2_headphone_volume_ctl_get( + struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + if (private->vol_updated) { + err = scarlett2_update_volumes(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.integer.value[0] = private->headphone_vol; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + static int line_out_remap(struct scarlett2_data *private, int index) { const struct scarlett2_device_info *info = private->info; @@ -3456,6 +3499,17 @@ static const struct snd_kcontrol_new scarlett2_master_volume_ctl = { .tlv = { .p = db_scale_scarlett2_volume } }; +static const struct snd_kcontrol_new scarlett2_headphone_volume_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_TLV_READ, + .name = "", + .info = scarlett2_volume_ctl_info, + .get = scarlett2_headphone_volume_ctl_get, + .private_value = 0, /* max value */ + .tlv = { .p = db_scale_scarlett2_volume } +}; + static const struct snd_kcontrol_new scarlett2_line_out_volume_ctl = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | @@ -4806,6 +4860,18 @@ static int scarlett2_add_line_out_ctls(struct usb_mixer_interface *mixer) return err; } + /* Add R/O headphone volume control */ + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_HEADPHONE_VOLUME)) { + snprintf(s, sizeof(s), "Headphone Playback Volume"); + err = scarlett2_add_new_ctl(mixer, + &scarlett2_headphone_volume_ctl, + 0, 1, s, + &private->headphone_vol_ctl); + if (err < 0) + return err; + } + /* Remaining controls are only applicable if the device * has per-channel line-out volume controls. */ @@ -6265,7 +6331,7 @@ static void scarlett2_notify_sync(struct usb_mixer_interface *mixer) &private->sync_ctl->id); } -/* Notify on monitor change */ +/* Notify on monitor change (Gen 2/3) */ static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; @@ -6286,6 +6352,20 @@ static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) &private->vol_ctls[i]->id); } +/* Notify on volume change (Gen 4) */ +static __always_unused void scarlett2_notify_volume( + struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + + private->vol_updated = 1; + + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->master_vol_ctl->id); + snd_ctl_notify(mixer->chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->headphone_vol_ctl->id); +} + /* Notify on dim/mute change */ static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer) { From 4e809a299677b8a9a239574a787620cb4f6c086a Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:38:58 +1030 Subject: [PATCH 317/337] ALSA: scarlett2: Add support for Solo, 2i2, and 4i4 Gen 4 Add new Focusrite Scarlett Gen 4 USB IDs, notification arrays, config sets, and device info data. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/b33526d3b7a56bb2c86aa4eb2137a415bd23f1ce.1703612638.git.g@b4.vu --- include/uapi/sound/scarlett2.h | 4 +- sound/usb/mixer_quirks.c | 3 + sound/usb/mixer_scarlett2.c | 361 +++++++++++++++++++++++++++++++-- 3 files changed, 351 insertions(+), 17 deletions(-) diff --git a/include/uapi/sound/scarlett2.h b/include/uapi/sound/scarlett2.h index d0ff38ffa154..91467ab0ed70 100644 --- a/include/uapi/sound/scarlett2.h +++ b/include/uapi/sound/scarlett2.h @@ -1,8 +1,8 @@ /* SPDX-License-Identifier: GPL-2.0 WITH Linux-syscall-note */ /* * Focusrite Scarlett 2 Protocol Driver for ALSA - * (including Scarlett 2nd Gen, 3rd Gen, Clarett USB, and Clarett+ - * series products) + * (including Scarlett 2nd Gen, 3rd Gen, 4th Gen, Clarett USB, and + * Clarett+ series products) * * Copyright (c) 2023 by Geoffrey D. Bennett */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index c8d48566e175..065a4be0d771 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3447,6 +3447,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x8213): /* Focusrite Scarlett 8i6 3rd Gen */ case USB_ID(0x1235, 0x8214): /* Focusrite Scarlett 18i8 3rd Gen */ case USB_ID(0x1235, 0x8215): /* Focusrite Scarlett 18i20 3rd Gen */ + case USB_ID(0x1235, 0x8218): /* Focusrite Scarlett Solo 4th Gen */ + case USB_ID(0x1235, 0x8219): /* Focusrite Scarlett 2i2 4th Gen */ + case USB_ID(0x1235, 0x821a): /* Focusrite Scarlett 4i4 4th Gen */ case USB_ID(0x1235, 0x8206): /* Focusrite Clarett 2Pre USB */ case USB_ID(0x1235, 0x8207): /* Focusrite Clarett 4Pre USB */ case USB_ID(0x1235, 0x8208): /* Focusrite Clarett 8Pre USB */ diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 94413fca2b6c..743054472184 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -1,12 +1,13 @@ // SPDX-License-Identifier: GPL-2.0 /* * Focusrite Scarlett 2 Protocol Driver for ALSA - * (including Scarlett 2nd Gen, 3rd Gen, Clarett USB, and Clarett+ - * series products) + * (including Scarlett 2nd Gen, 3rd Gen, 4th Gen, Clarett USB, and + * Clarett+ series products) * * Supported models: * - 6i6/18i8/18i20 Gen 2 * - Solo/2i2/4i4/8i6/18i8/18i20 Gen 3 + * - Solo/2i2/4i4 Gen 4 * - Clarett 2Pre/4Pre/8Pre USB * - Clarett+ 2Pre/4Pre/8Pre * @@ -68,6 +69,12 @@ * * Support for Clarett 2Pre and 4Pre USB added in Oct 2023. * + * Support for firmware updates added in Dec 2023. + * + * Support for Scarlett Solo/2i2/4i4 Gen 4 added in Dec 2023 (thanks + * to many LinuxMusicians people and to Focusrite for hardware + * donations). + * * This ALSA mixer gives access to (model-dependent): * - input, output, mixer-matrix muxes * - mixer-matrix gain stages @@ -78,6 +85,8 @@ * controls * - disable/enable MSD mode * - disable/enable standalone mode + * - input gain, autogain, safe mode + * - direct monitor mixes * * * /--------------\ 18chn 20chn /--------------\ @@ -130,7 +139,7 @@ * \--------------/ * * - * Gen 3 devices have a Mass Storage Device (MSD) mode where a small + * Gen 3/4 devices have a Mass Storage Device (MSD) mode where a small * disk with registration and driver download information is presented * to the host. To access the full functionality of the device without * proprietary software, MSD mode can be disabled by: @@ -302,9 +311,19 @@ struct scarlett2_notification { static void scarlett2_notify_sync(struct usb_mixer_interface *mixer); static void scarlett2_notify_dim_mute(struct usb_mixer_interface *mixer); static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer); +static void scarlett2_notify_volume(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_level(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_pad(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_air(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_phantom(struct usb_mixer_interface *mixer); static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_gain(struct usb_mixer_interface *mixer); +static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer); +static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer); static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer); static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer); +static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer); /* Arrays of notification callback functions */ @@ -325,6 +344,48 @@ static const struct scarlett2_notification scarlett3a_notifications[] = { { 0, NULL } }; +static const struct scarlett2_notification scarlett4_solo_notifications[] = { + { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000008, scarlett2_notify_sync }, + { 0x00400000, scarlett2_notify_input_air }, + { 0x00800000, scarlett2_notify_direct_monitor }, + { 0x01000000, scarlett2_notify_input_level }, + { 0x02000000, scarlett2_notify_input_phantom }, + { 0, NULL } +}; + +static const struct scarlett2_notification scarlett4_2i2_notifications[] = { + { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000008, scarlett2_notify_sync }, + { 0x00200000, scarlett2_notify_input_safe }, + { 0x00400000, scarlett2_notify_autogain }, + { 0x00800000, scarlett2_notify_input_air }, + { 0x01000000, scarlett2_notify_direct_monitor }, + { 0x02000000, scarlett2_notify_input_select }, + { 0x04000000, scarlett2_notify_input_level }, + { 0x08000000, scarlett2_notify_input_phantom }, + { 0x10000000, NULL }, /* power status, ignored */ + { 0x40000000, scarlett2_notify_input_gain }, + { 0x80000000, NULL }, /* power status, ignored */ + { 0, NULL } +}; + +static const struct scarlett2_notification scarlett4_4i4_notifications[] = { + { 0x00000001, NULL }, /* ack, gets ignored */ + { 0x00000008, scarlett2_notify_sync }, + { 0x00200000, scarlett2_notify_input_safe }, + { 0x00400000, scarlett2_notify_autogain }, + { 0x00800000, scarlett2_notify_input_air }, + { 0x01000000, scarlett2_notify_input_select }, + { 0x02000000, scarlett2_notify_input_level }, + { 0x04000000, scarlett2_notify_input_phantom }, + { 0x08000000, scarlett2_notify_power_status }, /* power external */ + { 0x20000000, scarlett2_notify_input_gain }, + { 0x40000000, scarlett2_notify_power_status }, /* power status */ + { 0x80000000, scarlett2_notify_volume }, + { 0, NULL } +}; + /* Configuration parameters that can be read and written */ enum { SCARLETT2_CONFIG_DIM_MUTE, @@ -543,6 +604,123 @@ static const struct scarlett2_config_set scarlett2_config_set_gen3c = { } }; +/* Solo Gen 4 */ +static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = { + .notifications = scarlett4_solo_notifications, + .gen4_write_addr = 0xd8, + .items = { + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x47, .size = 8, .activate = 4 }, + + [SCARLETT2_CONFIG_DIRECT_MONITOR] = { + .offset = 0x108, .activate = 12 }, + + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x46, .activate = 9, .mute = 1 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x3d, .activate = 10, .mute = 1 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x3e, .activate = 11 }, + + [SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN] = { + .offset = 0x232, .size = 16, .activate = 26 } + } +}; + +/* 2i2 Gen 4 */ +static const struct scarlett2_config_set scarlett2_config_set_gen4_2i2 = { + .notifications = scarlett4_2i2_notifications, + .gen4_write_addr = 0xfc, + .items = { + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x49, .size = 8, .activate = 4 }, // 0x41 ?? + + [SCARLETT2_CONFIG_DIRECT_MONITOR] = { + .offset = 0x14a, .activate = 16 }, + + [SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = { + .offset = 0x135, .activate = 10 }, + + [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { + .offset = 0x137 }, + + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x48, .activate = 11, .mute = 1 }, + + [SCARLETT2_CONFIG_INPUT_GAIN] = { + .offset = 0x4b, .activate = 12 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x3c, .activate = 13, .mute = 1 }, + + [SCARLETT2_CONFIG_SAFE_SWITCH] = { + .offset = 0x147, .activate = 14 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x3e, .activate = 15 }, + + [SCARLETT2_CONFIG_INPUT_SELECT_SWITCH] = { + .offset = 0x14b, .activate = 17 }, + + [SCARLETT2_CONFIG_INPUT_LINK_SWITCH] = { + .offset = 0x14e, .activate = 18 }, + + [SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN] = { + .offset = 0x2a0, .size = 16, .activate = 36 } + } +}; + +/* 4i4 Gen 4 */ +static const struct scarlett2_config_set scarlett2_config_set_gen4_4i4 = { + .notifications = scarlett4_4i4_notifications, + .gen4_write_addr = 0x130, + .items = { + [SCARLETT2_CONFIG_MSD_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 4 }, + + [SCARLETT2_CONFIG_AUTOGAIN_SWITCH] = { + .offset = 0x13e, .activate = 10 }, + + [SCARLETT2_CONFIG_AUTOGAIN_STATUS] = { + .offset = 0x140 }, + + [SCARLETT2_CONFIG_PHANTOM_SWITCH] = { + .offset = 0x5a, .activate = 11, .mute = 1 }, + + [SCARLETT2_CONFIG_INPUT_GAIN] = { + .offset = 0x5e, .activate = 12 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x4e, .activate = 13, .mute = 1 }, + + [SCARLETT2_CONFIG_SAFE_SWITCH] = { + .offset = 0x150, .activate = 14 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x50, .activate = 15 }, + + [SCARLETT2_CONFIG_INPUT_SELECT_SWITCH] = { + .offset = 0x153, .activate = 16 }, + + [SCARLETT2_CONFIG_INPUT_LINK_SWITCH] = { + .offset = 0x156, .activate = 17 }, + + [SCARLETT2_CONFIG_MASTER_VOLUME] = { + .offset = 0x32, .size = 16 }, + + [SCARLETT2_CONFIG_HEADPHONE_VOLUME] = { + .offset = 0x3a, .size = 16 }, + + [SCARLETT2_CONFIG_POWER_EXT] = { + .offset = 0x168 }, + + [SCARLETT2_CONFIG_POWER_STATUS] = { + .offset = 0x66 } + } +}; + /* Clarett USB and Clarett+ devices: 2Pre, 4Pre, 8Pre */ static const struct scarlett2_config_set scarlett2_config_set_clarett = { .notifications = scarlett2_notifications, @@ -1274,6 +1452,160 @@ static const struct scarlett2_device_info s18i20_gen3_info = { } }; +static const struct scarlett2_device_info solo_gen4_info = { + .config_set = &scarlett2_config_set_gen4_solo, + .min_firmware_version = 2115, + + .level_input_count = 1, + .air_input_count = 1, + .air_input_first = 1, + .air_option = 1, + .phantom_count = 1, + .phantom_first = 1, + .inputs_per_phantom = 1, + .direct_monitor = 1, + .dsp_count = 2, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 2, 2 }, + [SCARLETT2_PORT_TYPE_MIX] = { 8, 6 }, + [SCARLETT2_PORT_TYPE_PCM] = { 2, 4 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_MIX, 4, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 2, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 2 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_MIX, 4, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 2, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 2 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_MIX, 4, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 2, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 2 }, + { 0, 0, 0 }, + } }, + + .meter_map = { + { 6, 2 }, + { 4, 2 }, + { 8, 4 }, + { 2, 2 }, + { 0, 2 }, + { 0, 0 } + } +}; + +static const struct scarlett2_device_info s2i2_gen4_info = { + .config_set = &scarlett2_config_set_gen4_2i2, + .min_firmware_version = 2115, + + .level_input_count = 2, + .air_input_count = 2, + .air_option = 1, + .phantom_count = 1, + .inputs_per_phantom = 2, + .gain_input_count = 2, + .direct_monitor = 2, + .dsp_count = 2, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 2, 2 }, + [SCARLETT2_PORT_TYPE_MIX] = { 6, 6 }, + [SCARLETT2_PORT_TYPE_PCM] = { 2, 4 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_MIX, 4, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 2, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 2 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_MIX, 4, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 2, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 2 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_MIX, 4, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 2, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 2 }, + { 0, 0, 0 }, + } }, + + .meter_map = { + { 6, 2 }, + { 4, 2 }, + { 8, 4 }, + { 2, 2 }, + { 0, 2 }, + { 0, 0 } + } +}; + +static const struct scarlett2_device_info s4i4_gen4_info = { + .config_set = &scarlett2_config_set_gen4_4i4, + .min_firmware_version = 2089, + + .level_input_count = 2, + .air_input_count = 2, + .air_option = 1, + .phantom_count = 2, + .inputs_per_phantom = 1, + .gain_input_count = 2, + .dsp_count = 2, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 4, 6 }, + [SCARLETT2_PORT_TYPE_MIX] = { 8, 12 }, + [SCARLETT2_PORT_TYPE_PCM] = { 6, 6 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_MIX, 10, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 6 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 10 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_MIX, 10, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 6 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 10 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_MIX, 10, 2 }, + { SCARLETT2_PORT_TYPE_PCM, 0, 6 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 10 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { 0, 0, 0 }, + } }, + + .meter_map = { + { 16, 8 }, + { 6, 10 }, + { 0, 6 }, + { 0, 0 } + } +}; + static const struct scarlett2_device_info clarett_2pre_info = { .config_set = &scarlett2_config_set_clarett, .level_input_count = 2, @@ -1451,6 +1783,11 @@ static const struct scarlett2_device_entry scarlett2_devices[] = { { USB_ID(0x1235, 0x8214), &s18i8_gen3_info, "Scarlett Gen 3" }, { USB_ID(0x1235, 0x8215), &s18i20_gen3_info, "Scarlett Gen 3" }, + /* Supported Gen 4 devices */ + { USB_ID(0x1235, 0x8218), &solo_gen4_info, "Scarlett Gen 4" }, + { USB_ID(0x1235, 0x8219), &s2i2_gen4_info, "Scarlett Gen 4" }, + { USB_ID(0x1235, 0x821a), &s4i4_gen4_info, "Scarlett Gen 4" }, + /* Supported Clarett USB/Clarett+ devices */ { USB_ID(0x1235, 0x8206), &clarett_2pre_info, "Clarett USB" }, { USB_ID(0x1235, 0x8207), &clarett_4pre_info, "Clarett USB" }, @@ -6353,8 +6690,7 @@ static void scarlett2_notify_monitor(struct usb_mixer_interface *mixer) } /* Notify on volume change (Gen 4) */ -static __always_unused void scarlett2_notify_volume( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_volume(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; @@ -6460,8 +6796,7 @@ static void scarlett2_notify_input_other(struct usb_mixer_interface *mixer) } /* Notify on input select change */ -static __always_unused void scarlett2_notify_input_select( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_input_select(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -6483,8 +6818,7 @@ static __always_unused void scarlett2_notify_input_select( } /* Notify on input gain change */ -static __always_unused void scarlett2_notify_input_gain( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_input_gain(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -6502,8 +6836,7 @@ static __always_unused void scarlett2_notify_input_gain( } /* Notify on autogain change */ -static __always_unused void scarlett2_notify_autogain( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_autogain(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -6526,8 +6859,7 @@ static __always_unused void scarlett2_notify_autogain( } /* Notify on input safe switch change */ -static __always_unused void scarlett2_notify_input_safe( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; @@ -6603,8 +6935,7 @@ static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer) } /* Notify on power change */ -static __always_unused void scarlett2_notify_power_status( - struct usb_mixer_interface *mixer) +static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer) { struct snd_card *card = mixer->chip->card; struct scarlett2_data *private = mixer->private_data; From 4a2c8cc1644738191d9d6c0bca24516cfe390d41 Mon Sep 17 00:00:00 2001 From: "Geoffrey D. Bennett" Date: Wed, 27 Dec 2023 04:39:04 +1030 Subject: [PATCH 318/337] ALSA: scarlett2: Add PCM Input Switch for Solo Gen 4 When the Direct button on the Solo Gen 4 is held for 3 seconds, the PCM 1 and 2 inputs are toggled between DSP Outputs 1 and 2, and Mixer Outputs E and F. This patch adds the corresponding ALSA control. Signed-off-by: Geoffrey D. Bennett Signed-off-by: Takashi Iwai Link: https://lore.kernel.org/r/8c67c6131c459588ac4edab11e1fbc40a8297328.1703612638.git.g@b4.vu --- sound/usb/mixer_scarlett2.c | 151 ++++++++++++++++++++++++++++++++++++ 1 file changed, 151 insertions(+) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 743054472184..1de3ddc50eb6 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -324,6 +324,8 @@ static void scarlett2_notify_input_safe(struct usb_mixer_interface *mixer); static void scarlett2_notify_monitor_other(struct usb_mixer_interface *mixer); static void scarlett2_notify_direct_monitor(struct usb_mixer_interface *mixer); static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer); +static void scarlett2_notify_pcm_input_switch( + struct usb_mixer_interface *mixer); /* Arrays of notification callback functions */ @@ -351,6 +353,7 @@ static const struct scarlett2_notification scarlett4_solo_notifications[] = { { 0x00800000, scarlett2_notify_direct_monitor }, { 0x01000000, scarlett2_notify_input_level }, { 0x02000000, scarlett2_notify_input_phantom }, + { 0x04000000, scarlett2_notify_pcm_input_switch }, { 0, NULL } }; @@ -413,6 +416,7 @@ enum { SCARLETT2_CONFIG_INPUT_LINK_SWITCH, SCARLETT2_CONFIG_POWER_EXT, SCARLETT2_CONFIG_POWER_STATUS, + SCARLETT2_CONFIG_PCM_INPUT_SWITCH, SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN, SCARLETT2_CONFIG_COUNT }; @@ -624,6 +628,9 @@ static const struct scarlett2_config_set scarlett2_config_set_gen4_solo = { [SCARLETT2_CONFIG_AIR_SWITCH] = { .offset = 0x3e, .activate = 11 }, + [SCARLETT2_CONFIG_PCM_INPUT_SWITCH] = { + .offset = 0x206, .activate = 25 }, + [SCARLETT2_CONFIG_DIRECT_MONITOR_GAIN] = { .offset = 0x232, .size = 16, .activate = 26 } } @@ -955,6 +962,7 @@ struct scarlett2_data { u8 input_gain_updated; u8 autogain_updated; u8 input_safe_updated; + u8 pcm_input_switch_updated; u8 monitor_other_updated; u8 direct_monitor_updated; u8 mux_updated; @@ -979,6 +987,7 @@ struct scarlett2_data { u8 autogain_switch[SCARLETT2_INPUT_GAIN_MAX]; u8 autogain_status[SCARLETT2_INPUT_GAIN_MAX]; u8 safe_switch[SCARLETT2_INPUT_GAIN_MAX]; + u8 pcm_input_switch; u8 direct_monitor_switch; u8 speaker_switching_switch; u8 talkback_switch; @@ -1004,6 +1013,7 @@ struct scarlett2_data { struct snd_kcontrol *autogain_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *autogain_status_ctls[SCARLETT2_INPUT_GAIN_MAX]; struct snd_kcontrol *safe_ctls[SCARLETT2_INPUT_GAIN_MAX]; + struct snd_kcontrol *pcm_input_switch_ctl; struct snd_kcontrol *mux_ctls[SCARLETT2_MUX_MAX]; struct snd_kcontrol *mix_ctls[SCARLETT2_MIX_MAX]; struct snd_kcontrol *direct_monitor_ctl; @@ -3633,6 +3643,101 @@ static const struct snd_kcontrol_new scarlett2_safe_ctl = { .put = scarlett2_safe_ctl_put, }; +/*** PCM Input Control ***/ + +static int scarlett2_update_pcm_input_switch(struct usb_mixer_interface *mixer) +{ + struct scarlett2_data *private = mixer->private_data; + int err; + + private->pcm_input_switch_updated = 0; + + err = scarlett2_usb_get_config( + mixer, SCARLETT2_CONFIG_PCM_INPUT_SWITCH, + 1, &private->pcm_input_switch); + if (err < 0) + return err; + + return 0; +} + +static int scarlett2_pcm_input_switch_ctl_get( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = elem->head.mixer->private_data; + int err = 0; + + mutex_lock(&private->data_mutex); + + if (private->pcm_input_switch_updated) { + err = scarlett2_update_pcm_input_switch(mixer); + if (err < 0) + goto unlock; + } + ucontrol->value.enumerated.item[0] = private->pcm_input_switch; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_pcm_input_switch_ctl_put( + struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_info *elem = kctl->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; + + int oval, val, err = 0; + + mutex_lock(&private->data_mutex); + + if (private->hwdep_in_use) { + err = -EBUSY; + goto unlock; + } + + oval = private->pcm_input_switch; + val = !!ucontrol->value.integer.value[0]; + + if (oval == val) + goto unlock; + + private->pcm_input_switch = val; + + /* Send switch change to the device */ + err = scarlett2_usb_set_config( + mixer, SCARLETT2_CONFIG_PCM_INPUT_SWITCH, + 0, val); + if (err == 0) + err = 1; + +unlock: + mutex_unlock(&private->data_mutex); + return err; +} + +static int scarlett2_pcm_input_switch_ctl_info( + struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) +{ + static const char *const values[2] = { + "Direct", "Mixer" + }; + + return snd_ctl_enum_info( + uinfo, 1, 2, values); +} + +static const struct snd_kcontrol_new scarlett2_pcm_input_switch_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "", + .info = scarlett2_pcm_input_switch_ctl_info, + .get = scarlett2_pcm_input_switch_ctl_get, + .put = scarlett2_pcm_input_switch_ctl_put +}; + /*** Analogue Line Out Volume Controls ***/ /* Update hardware volume controls after receiving notification that @@ -5419,6 +5524,17 @@ static int scarlett2_add_line_in_ctls(struct usb_mixer_interface *mixer) } } + /* Add PCM Input Switch control */ + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_PCM_INPUT_SWITCH)) { + err = scarlett2_add_new_ctl( + mixer, &scarlett2_pcm_input_switch_ctl, 0, 1, + "PCM Input Capture Switch", + &private->pcm_input_switch_ctl); + if (err < 0) + return err; + } + return 0; } @@ -6650,6 +6766,13 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) if (err < 0) return err; + if (scarlett2_has_config_item(private, + SCARLETT2_CONFIG_PCM_INPUT_SWITCH)) { + err = scarlett2_update_pcm_input_switch(mixer); + if (err < 0) + return err; + } + err = scarlett2_update_mix(mixer); if (err < 0) return err; @@ -6946,6 +7069,34 @@ static void scarlett2_notify_power_status(struct usb_mixer_interface *mixer) &private->power_status_ctl->id); } +/* Notify on mux change */ +static void scarlett2_notify_mux(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + int i; + + private->mux_updated = 1; + + for (i = 0; i < private->num_mux_dsts; i++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->mux_ctls[i]->id); +} + +/* Notify on PCM input switch change */ +static void scarlett2_notify_pcm_input_switch(struct usb_mixer_interface *mixer) +{ + struct snd_card *card = mixer->chip->card; + struct scarlett2_data *private = mixer->private_data; + + private->pcm_input_switch_updated = 1; + + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &private->pcm_input_switch_ctl->id); + + scarlett2_notify_mux(mixer); +} + /* Interrupt callback */ static void scarlett2_notify(struct urb *urb) { From 5fa3bbb8eba7eaddd5c57952fca1f28bc3520d51 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Fri, 29 Dec 2023 17:29:22 +0800 Subject: [PATCH 319/337] ASoC: rt5663: cancel the work when system suspends This patch makes sure that the workqueue is completed before the system suspends. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20231229092922.853-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5663.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 77246f84de29..9550492605ac 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -74,6 +74,7 @@ struct rt5663_priv { int pll_out; int jack_type; + unsigned int irq; }; static const struct reg_sequence rt5663_patch_list[] = { @@ -3186,6 +3187,12 @@ static int rt5663_suspend(struct snd_soc_component *component) { struct rt5663_priv *rt5663 = snd_soc_component_get_drvdata(component); + if (rt5663->irq) + disable_irq(rt5663->irq); + + cancel_delayed_work_sync(&rt5663->jack_detect_work); + cancel_delayed_work_sync(&rt5663->jd_unplug_work); + regcache_cache_only(rt5663->regmap, true); regcache_mark_dirty(rt5663->regmap); @@ -3201,6 +3208,9 @@ static int rt5663_resume(struct snd_soc_component *component) rt5663_irq(0, rt5663); + if (rt5663->irq) + enable_irq(rt5663->irq); + return 0; } #else @@ -3686,6 +3696,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c) __func__, ret); goto err_enable; } + rt5663->irq = i2c->irq; } ret = devm_snd_soc_register_component(&i2c->dev, From 67508b874844b80ac49f70b78d67036c28b9fe7e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Duje=20Mihanovi=C4=87?= Date: Tue, 26 Dec 2023 21:00:24 +0100 Subject: [PATCH 320/337] ASoC: pxa: sspa: Don't select SND_ARM MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On ARM64 platforms, SND_ARM shouldn't be selectable, but enabling SND_SOC_MMP_SSPA will enable SND_ARM and cause build errors if SND_ARMAACI is enabled (which it is by default). Since the SSPA driver doesn't depend on AACI nor PXA2XX_LIB, remove this false dependency. Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202310230518.zs9Qpg3j-lkp@intel.com/ Signed-off-by: Duje Mihanović Link: https://lore.kernel.org/r/20231226200025.30870-1-duje.mihanovic@skole.hr Signed-off-by: Mark Brown --- sound/soc/pxa/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e6bca9070953..f03c74809324 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -35,7 +35,6 @@ config SND_MMP_SOC_SSPA tristate "SoC Audio via MMP SSPA ports" depends on ARCH_MMP select SND_SOC_GENERIC_DMAENGINE_PCM - select SND_ARM help Say Y if you want to add support for codecs attached to the MMP SSPA interface. From 66e82d219924f6112509f7e8f8e687fcc81a16e3 Mon Sep 17 00:00:00 2001 From: Greg Kroah-Hartman Date: Tue, 19 Dec 2023 14:34:46 +0100 Subject: [PATCH 321/337] ALSA: mark all struct bus_type as const Now that the driver core can properly handle constant struct bus_type, move all of the sound subsystem struct bus_type structures as const, placing them into read-only memory which can not be modified at runtime. Note, this fixes a duplicate definition of ac97_bus_type, which somehow was declared extern in a .h file, and then static as a prototype in a .c file, and then properly later on in the same .c file. Amazing that no compiler warning ever showed up for this. Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Dawei Li Cc: Yu Liao Cc: "Rafael J. Wysocki" Cc: Hans de Goede Cc: linux-sound@vger.kernel.org Signed-off-by: Greg Kroah-Hartman Link: https://lore.kernel.org/r/2023121945-immersion-budget-d0aa@gregkh Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 2 +- include/sound/hdaudio.h | 2 +- sound/ac97/bus.c | 2 -- sound/ac97_bus.c | 2 +- sound/hda/hda_bus_type.c | 2 +- 5 files changed, 4 insertions(+), 6 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index c495c6d5fbe0..4bd3be3a3192 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -410,7 +410,7 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm); int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime); /* ad hoc AC97 device driver access */ -extern struct bus_type ac97_bus_type; +extern const struct bus_type ac97_bus_type; /* AC97 platform_data adding function */ static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 62a3cd154ff2..a73d7f34f4e5 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -33,7 +33,7 @@ struct hda_device_id; /* * exported bus type */ -extern struct bus_type snd_hda_bus_type; +extern const struct bus_type snd_hda_bus_type; /* * generic arrays diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 3173e9d98927..1dc7965eb14b 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -28,8 +28,6 @@ static DEFINE_MUTEX(ac97_controllers_mutex); static DEFINE_IDR(ac97_adapter_idr); static LIST_HEAD(ac97_controllers); -static struct bus_type ac97_bus_type; - static inline struct ac97_controller* to_ac97_controller(struct device *ac97_adapter) { diff --git a/sound/ac97_bus.c b/sound/ac97_bus.c index c7aee8c42c55..7ea274c55900 100644 --- a/sound/ac97_bus.c +++ b/sound/ac97_bus.c @@ -75,7 +75,7 @@ int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, } EXPORT_SYMBOL_GPL(snd_ac97_reset); -struct bus_type ac97_bus_type = { +const struct bus_type ac97_bus_type = { .name = "ac97", }; diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c index 4cd94178df9f..cce2c30511a2 100644 --- a/sound/hda/hda_bus_type.c +++ b/sound/hda/hda_bus_type.c @@ -76,7 +76,7 @@ static int hda_uevent(const struct device *dev, struct kobj_uevent_env *env) return 0; } -struct bus_type snd_hda_bus_type = { +const struct bus_type snd_hda_bus_type = { .name = "hdaudio", .match = hda_bus_match, .uevent = hda_uevent, From 68f7f3ff6c2a0998be9dc07622bd0d16fd1fda20 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Sat, 30 Dec 2023 01:13:41 +0100 Subject: [PATCH 322/337] ALSA: hda/tas2781: configure the amp after firmware load Make the amp available immediately after a module load to avoid having to wait for a PCM hook action. (eg. unloading & loading the module while listening music) Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/7f2f65d9212aa16edd4db8725489ae59dbe74c66.1703895108.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index dfe281b57aa6..c8523df4105f 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -584,6 +584,8 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) */ tas2781_save_calibration(tas_priv); + tasdevice_tuning_switch(tas_hda->priv, 0); + out: mutex_unlock(&tas_hda->priv->codec_lock); if (fmw) From 99af5b11c57d33c32d761797f6308b40936c22ed Mon Sep 17 00:00:00 2001 From: Dorian Cruveiller Date: Sat, 30 Dec 2023 12:40:01 +0100 Subject: [PATCH 323/337] ALSA: hda/realtek: enable SND_PCI_QUIRK for Lenovo Legion Slim 7 Gen 8 (2023) serie Link up the realtek audio chip to the cirrus cs35l41 sound amplifier chip on 4 models of the Lenovo legion slim 7 gen 8 (2023). These models are 16IRH8 (2 differents subsystem id) and 16APH8 (2 differents subsystem ids). Subsystem ids list: - 17AA38B4 - 17AA38B5 - 17AA38B6 - 17AA38B7 Signed-off-by: Dorian Cruveiller Cc: # v6.7 Link: https://lore.kernel.org/r/20231230114001.19855-1-doriancruveiller@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 19040887ff67..bc4e3a85137c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10239,6 +10239,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), From ba7053b4b4a4ddcf530fa2b897e697004715d086 Mon Sep 17 00:00:00 2001 From: Dorian Cruveiller Date: Sat, 30 Dec 2023 12:43:12 +0100 Subject: [PATCH 324/337] ALSA: hda: Add driver properties for cs35l41 for Lenovo Legion Slim 7 Gen 8 serie Add driver properties on 4 models of this laptop serie since they don't have _DSD in the ACPI table Signed-off-by: Dorian Cruveiller Cc: # v6.7 Link: https://lore.kernel.org/r/20231230114312.22118-1-doriancruveiller@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index be2b01b596c2..52820ca9c603 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -71,6 +71,10 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "10431F12", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431F1F", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, -1, 0, 0, 0, 0 }, { "10431F62", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 0, 0, 0 }, + { "17AA38B4", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA38B5", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA38B6", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA38B7", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, {} }; @@ -381,6 +385,10 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10431F12", generic_dsd_config }, { "CSC3551", "10431F1F", generic_dsd_config }, { "CSC3551", "10431F62", generic_dsd_config }, + { "CSC3551", "17AA38B4", generic_dsd_config }, + { "CSC3551", "17AA38B5", generic_dsd_config }, + { "CSC3551", "17AA38B6", generic_dsd_config }, + { "CSC3551", "17AA38B7", generic_dsd_config }, {} }; From 76f5f55c45b906710c9565a7e68c8d782c46b394 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Sat, 30 Dec 2023 01:09:42 +0100 Subject: [PATCH 325/337] ALSA: hda/tas2781: add ptrs to calibration functions Make calibration functions configurable to support different calibration data storage modes. Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/5859c77ffef752b8a9784713b412d815d7e2688c.1703891777.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- include/sound/tas2781.h | 5 +++++ sound/pci/hda/tas2781_hda_i2c.c | 25 +++++++++++-------------- sound/soc/codecs/tas2781-comlib.c | 15 +++++++++++++++ 3 files changed, 31 insertions(+), 14 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index a6c808b22318..e17ceab4fead 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -131,6 +131,9 @@ struct tasdevice_priv { const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); + + int (*save_calibration)(struct tasdevice_priv *tas_priv); + void (*apply_calibration)(struct tasdevice_priv *tas_priv); }; void tas2781_reset(struct tasdevice_priv *tas_dev); @@ -139,6 +142,8 @@ int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); +void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index c8523df4105f..4b639120c981 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -479,7 +479,7 @@ static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", tm->tm_year, tm->tm_mon, tm->tm_mday, tm->tm_hour, tm->tm_min, tm->tm_sec); - tas2781_apply_calib(tas_priv); + tasdevice_apply_calibration(tas_priv); } else tas_priv->cali_data.total_sz = 0; @@ -582,7 +582,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ - tas2781_save_calibration(tas_priv); + tasdevice_save_calibration(tas_priv); tasdevice_tuning_switch(tas_hda->priv, 0); @@ -685,10 +685,6 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) const char *device_name; int ret; - if (strstr(dev_name(&clt->dev), "TIAS2781")) - device_name = "TIAS2781"; - else - return -ENODEV; tas_hda = devm_kzalloc(&clt->dev, sizeof(*tas_hda), GFP_KERNEL); if (!tas_hda) @@ -701,6 +697,13 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) if (!tas_hda->priv) return -ENOMEM; + if (strstr(dev_name(&clt->dev), "TIAS2781")) { + device_name = "TIAS2781"; + tas_hda->priv->save_calibration = tas2781_save_calibration; + tas_hda->priv->apply_calibration = tas2781_apply_calib; + } else + return -ENODEV; + tas_hda->priv->irq_info.irq = clt->irq; ret = tas2781_read_acpi(tas_hda->priv, device_name); if (ret) @@ -767,8 +770,6 @@ static int tas2781_runtime_suspend(struct device *dev) static int tas2781_runtime_resume(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - unsigned long calib_data_sz = - tas_hda->priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; dev_dbg(tas_hda->dev, "Runtime Resume\n"); @@ -779,8 +780,7 @@ static int tas2781_runtime_resume(struct device *dev) /* If calibrated data occurs error, dsp will still works with default * calibrated data inside algo. */ - if (tas_hda->priv->cali_data.total_sz > calib_data_sz) - tas2781_apply_calib(tas_hda->priv); + tasdevice_apply_calibration(tas_hda->priv); mutex_unlock(&tas_hda->priv->codec_lock); @@ -811,8 +811,6 @@ static int tas2781_system_suspend(struct device *dev) static int tas2781_system_resume(struct device *dev) { struct tas2781_hda *tas_hda = dev_get_drvdata(dev); - unsigned long calib_data_sz = - tas_hda->priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; int i, ret; dev_info(tas_hda->priv->dev, "System Resume\n"); @@ -834,8 +832,7 @@ static int tas2781_system_resume(struct device *dev) /* If calibrated data occurs error, dsp will still work with default * calibrated data inside algo. */ - if (tas_hda->priv->cali_data.total_sz > calib_data_sz) - tas2781_apply_calib(tas_hda->priv); + tasdevice_apply_calibration(tas_hda->priv); mutex_unlock(&tas_hda->priv->codec_lock); return 0; diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 00e35169ae49..b7e56ceb1acf 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -412,6 +412,21 @@ void tasdevice_remove(struct tasdevice_priv *tas_priv) } EXPORT_SYMBOL_GPL(tasdevice_remove); +int tasdevice_save_calibration(struct tasdevice_priv *tas_priv) +{ + if (tas_priv->save_calibration) + return tas_priv->save_calibration(tas_priv); + return -EINVAL; +} +EXPORT_SYMBOL_GPL(tasdevice_save_calibration); + +void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv) +{ + if (tas_priv->apply_calibration && tas_priv->cali_data.total_sz) + tas_priv->apply_calibration(tas_priv); +} +EXPORT_SYMBOL_GPL(tasdevice_apply_calibration); + static int tasdevice_clamp(int val, int max, unsigned int invert) { if (val > max) From c021ca729fe870d25a4798491c383c89b3091650 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Sat, 30 Dec 2023 01:09:43 +0100 Subject: [PATCH 326/337] ALSA: hda/tas2781: add configurable global i2c address Make the global i2c address configurable to support compatible amplifiers with different global i2c address. Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/a252f1efeed5049f027f01e699c9e10e1e05bf9e.1703891777.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- include/sound/tas2781.h | 2 ++ sound/pci/hda/tas2781_hda_i2c.c | 3 ++- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index e17ceab4fead..dde9f8120d4c 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -121,6 +121,8 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + unsigned int global_addr; + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset); int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 4b639120c981..503be3fbce56 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -81,7 +81,7 @@ static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) if (i2c_acpi_get_i2c_resource(ares, &sb)) { if (tas_priv->ndev < TASDEVICE_MAX_CHANNELS && - sb->slave_address != TAS2781_GLOBAL_ADDR) { + sb->slave_address != tas_priv->global_addr) { tas_priv->tasdevice[tas_priv->ndev].dev_addr = (unsigned int)sb->slave_address; tas_priv->ndev++; @@ -701,6 +701,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) device_name = "TIAS2781"; tas_hda->priv->save_calibration = tas2781_save_calibration; tas_hda->priv->apply_calibration = tas2781_apply_calib; + tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; } else return -ENODEV; From c3ca4458cc2f719b1652abaa8d2fbf7f3d4311cb Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Sat, 30 Dec 2023 01:09:44 +0100 Subject: [PATCH 327/337] ALSA: hda/tas2781: add TAS2563 support for 14ARB7 The INT8866 belongs to the Lenovo Yoga 7 Gen 7 AMD 14ARB7 laptop. It has two TAS2563 amplifier. Add the PNP ID and calibration functions to handle them. ACPI excerpt: Scope (_SB.I2CD) { Device (TAS) { Name (_HID, "INT8866") // _HID: Hardware ID Name (_UID, Zero) // _UID: Unique ID Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (RBUF, ResourceTemplate () { I2cSerialBusV2 (0x004C, ControllerInitiated, 0x00061A80, AddressingMode7Bit, "\\_SB.I2CD", 0x00, ResourceConsumer, , Exclusive, ) I2cSerialBusV2 (0x004D, ControllerInitiated, 0x00061A80, AddressingMode7Bit, "\\_SB.I2CD", 0x00, ResourceConsumer, , Exclusive, ) GpioInt (Edge, ActiveLow, SharedAndWake, PullNone, 0x0000, "\\_SB.GPIO", 0x00, ResourceConsumer, , ) { // Pin list 0x0020 } }) Return (RBUF) /* \_SB_.I2CD.TAS_._CRS.RBUF */ } Method (_STA, 0, NotSerialized) // _STA: Status { Return (0x0F) } } } Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/3b8d4c602e1a46922f53bc9afc8b705d55aa4872.1703891777.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- include/sound/tas2781.h | 1 + sound/pci/hda/tas2781_hda_i2c.c | 87 +++++++++++++++++++++++++++++++++ 2 files changed, 88 insertions(+) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index dde9f8120d4c..0a86ab8d47b9 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -22,6 +22,7 @@ #define TAS2781_DRV_VER 1 #define SMARTAMP_MODULE_NAME "tas2781" #define TAS2781_GLOBAL_ADDR 0x40 +#define TAS2563_GLOBAL_ADDR 0x48 #define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ SNDRV_PCM_RATE_88200) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 503be3fbce56..4805cf0b6480 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -65,6 +65,24 @@ enum calib_data { CALIB_MAX }; +#define TAS2563_MAX_CHANNELS 4 + +#define TAS2563_CAL_POWER TASDEVICE_REG(0, 0x0d, 0x3c) +#define TAS2563_CAL_R0 TASDEVICE_REG(0, 0x0f, 0x34) +#define TAS2563_CAL_INVR0 TASDEVICE_REG(0, 0x0f, 0x40) +#define TAS2563_CAL_R0_LOW TASDEVICE_REG(0, 0x0f, 0x48) +#define TAS2563_CAL_TLIM TASDEVICE_REG(0, 0x10, 0x14) +#define TAS2563_CAL_N 5 +#define TAS2563_CAL_DATA_SIZE 4 +#define TAS2563_CAL_CH_SIZE 20 +#define TAS2563_CAL_ARRAY_SIZE 80 + +static unsigned int cal_regs[TAS2563_CAL_N] = { + TAS2563_CAL_POWER, TAS2563_CAL_R0, TAS2563_CAL_INVR0, + TAS2563_CAL_R0_LOW, TAS2563_CAL_TLIM, +}; + + struct tas2781_hda { struct device *dev; struct tasdevice_priv *priv; @@ -404,6 +422,69 @@ static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { .put = tasdevice_config_put, }; +static void tas2563_apply_calib(struct tasdevice_priv *tas_priv) +{ + unsigned int data; + int offset = 0; + int ret; + + for (int i = 0; i < tas_priv->ndev; i++) { + for (int j = 0; j < TAS2563_CAL_N; ++j) { + data = cpu_to_be32( + *(uint32_t *)&tas_priv->cali_data.data[offset]); + ret = tasdevice_dev_bulk_write(tas_priv, i, cal_regs[j], + (unsigned char *)&data, TAS2563_CAL_DATA_SIZE); + if (ret) + dev_err(tas_priv->dev, + "Error writing calib regs\n"); + offset += TAS2563_CAL_DATA_SIZE; + } + } +} + +static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) +{ + static efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + + static efi_char16_t *efi_vars[TAS2563_MAX_CHANNELS][TAS2563_CAL_N] = { + { L"Power_1", L"R0_1", L"InvR0_1", L"R0_Low_1", L"TLim_1" }, + { L"Power_2", L"R0_2", L"InvR0_2", L"R0_Low_2", L"TLim_2" }, + { L"Power_3", L"R0_3", L"InvR0_3", L"R0_Low_3", L"TLim_3" }, + { L"Power_4", L"R0_4", L"InvR0_4", L"R0_Low_4", L"TLim_4" }, + }; + + unsigned long max_size = TAS2563_CAL_DATA_SIZE; + unsigned int offset = 0; + efi_status_t status; + unsigned int attr; + + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + TAS2563_CAL_ARRAY_SIZE, GFP_KERNEL); + if (!tas_priv->cali_data.data) + return -ENOMEM; + + for (int i = 0; i < tas_priv->ndev; ++i) { + for (int j = 0; j < TAS2563_CAL_N; ++j) { + status = efi.get_variable(efi_vars[i][j], + &efi_guid, &attr, &max_size, + &tas_priv->cali_data.data[offset]); + if (status != EFI_SUCCESS || + max_size != TAS2563_CAL_DATA_SIZE) { + dev_warn(tas_priv->dev, + "Calibration data read failed %ld\n", status); + return -EINVAL; + } + offset += TAS2563_CAL_DATA_SIZE; + } + } + + tas_priv->cali_data.total_sz = offset; + tasdevice_apply_calibration(tas_priv); + + return 0; +} + static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { static const unsigned char page_array[CALIB_MAX] = { @@ -702,6 +783,11 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) tas_hda->priv->save_calibration = tas2781_save_calibration; tas_hda->priv->apply_calibration = tas2781_apply_calib; tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; + } else if (strstr(dev_name(&clt->dev), "INT8866")) { + device_name = "INT8866"; + tas_hda->priv->save_calibration = tas2563_save_calibration; + tas_hda->priv->apply_calibration = tas2563_apply_calib; + tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; } else return -ENODEV; @@ -851,6 +937,7 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"TIAS2781", 0 }, + {"INT8866", 0 }, {} }; MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); From b5cb53fd32779f3a971c45bcd997ae2608aa1086 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Sat, 30 Dec 2023 01:09:45 +0100 Subject: [PATCH 328/337] ALSA: hda/tas2781: add fixup for Lenovo 14ARB7 The 14ARB7 has two tas2563 amplifier on i2c. Connect it to the tas2781 driver. Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/abce9ee55689523562feb72383377171a489ddc7.1703891777.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc4e3a85137c..664729989a7f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6989,6 +6989,12 @@ static void tas2781_fixup_i2c(struct hda_codec *cdc, tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); } +static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "INT8866"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7460,6 +7466,7 @@ enum { ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, + ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_X360_MUTE_LEDS, ALC287_FIXUP_THINKPAD_I2S_SPK, @@ -9578,6 +9585,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = yoga7_14arb7_fixup_i2c, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, + }, [ALC245_FIXUP_HP_MUTE_LED_COEFBIT] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_mute_led_coefbit, @@ -10231,6 +10244,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), From 7d65d70161ef75a3991480c91668ac11acedf211 Mon Sep 17 00:00:00 2001 From: Lorenz Brun Date: Tue, 2 Jan 2024 22:48:20 +0100 Subject: [PATCH 329/337] ALSA: hda: cs35l41: Support more HP models without _DSD This adds overrides for a series of notebooks using a common config taken from HP's proprietary Windows driver. This has been tested on a HP 15-ey0xxxx device (subsystem 103C8A31) together with another Realtek quirk and the calibration files from the proprietary driver. Signed-off-by: Lorenz Brun Cc: # v6.7 Link: https://lore.kernel.org/r/20240102214821.3394810-1-lorenz@brun.one Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_property.c | 44 ++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 52820ca9c603..fcc605be51cf 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -42,6 +42,28 @@ static const struct cs35l41_config cs35l41_config_table[] = { * in the ACPI. The Reset GPIO is also valid, so we can use the Reset defined in _DSD. */ { "103C89C6", 2, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, -1, -1, -1, 1000, 4500, 24 }, + { "103C8A28", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A29", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2A", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2B", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2C", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2D", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A2E", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A30", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8A31", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BB3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BB4", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE0", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE1", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE2", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE9", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDD", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BDE", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE3", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE5", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8BE6", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8B3A", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "104312AF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431433", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431463", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -356,6 +378,28 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "10280BEB", generic_dsd_config }, { "CSC3551", "10280C4D", generic_dsd_config }, { "CSC3551", "103C89C6", generic_dsd_config }, + { "CSC3551", "103C8A28", generic_dsd_config }, + { "CSC3551", "103C8A29", generic_dsd_config }, + { "CSC3551", "103C8A2A", generic_dsd_config }, + { "CSC3551", "103C8A2B", generic_dsd_config }, + { "CSC3551", "103C8A2C", generic_dsd_config }, + { "CSC3551", "103C8A2D", generic_dsd_config }, + { "CSC3551", "103C8A2E", generic_dsd_config }, + { "CSC3551", "103C8A30", generic_dsd_config }, + { "CSC3551", "103C8A31", generic_dsd_config }, + { "CSC3551", "103C8BB3", generic_dsd_config }, + { "CSC3551", "103C8BB4", generic_dsd_config }, + { "CSC3551", "103C8BDF", generic_dsd_config }, + { "CSC3551", "103C8BE0", generic_dsd_config }, + { "CSC3551", "103C8BE1", generic_dsd_config }, + { "CSC3551", "103C8BE2", generic_dsd_config }, + { "CSC3551", "103C8BE9", generic_dsd_config }, + { "CSC3551", "103C8BDD", generic_dsd_config }, + { "CSC3551", "103C8BDE", generic_dsd_config }, + { "CSC3551", "103C8BE3", generic_dsd_config }, + { "CSC3551", "103C8BE5", generic_dsd_config }, + { "CSC3551", "103C8BE6", generic_dsd_config }, + { "CSC3551", "103C8B3A", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, { "CSC3551", "10431463", generic_dsd_config }, From f90dffdce70ff724f9ee8b0bcc711e86e6663896 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Wed, 3 Jan 2024 11:25:38 +0100 Subject: [PATCH 330/337] ALSA: ac97: fix build regression The ac97_bus_type structure is no longer declared in this file: sound/ac97/bus.c: In function 'ac97_codec_add': sound/ac97/bus.c:112:27: error: 'ac97_bus_type' undeclared (first use in this function); did you mean 'bus_type'? 112 | codec->dev.bus = &ac97_bus_type; | ^~~~~~~~~~~~~ | bus_type sound/ac97/bus.c:112:27: note: each undeclared identifier is reported only once for each function it appears in sound/ac97/bus.c: In function 'snd_ac97_codec_driver_register': sound/ac97/bus.c:191:28: error: 'ac97_bus_type' undeclared (first use in this function); did you mean 'ac97_bus_reset'? 191 | drv->driver.bus = &ac97_bus_type; Include the header that contains the declaration and make sure the definition is const but not static. Fixes: 66e82d219924 ("ALSA: mark all struct bus_type as const") Reviewed-by: Greg Kroah-Hartman Signed-off-by: Arnd Bergmann Link: https://lore.kernel.org/r/20240103102544.3715055-1-arnd@kernel.org Signed-off-by: Takashi Iwai --- sound/ac97/bus.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 1dc7965eb14b..5e46b972a3da 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include @@ -529,7 +530,7 @@ static void ac97_bus_remove(struct device *dev) pm_runtime_disable(dev); } -static struct bus_type ac97_bus_type = { +const struct bus_type ac97_bus_type = { .name = "ac97bus", .dev_groups = ac97_dev_groups, .match = ac97_bus_match, From 7aeb259086487417f0fecf66e325bee133e8813a Mon Sep 17 00:00:00 2001 From: bo liu Date: Mon, 8 Jan 2024 19:02:35 +0800 Subject: [PATCH 331/337] ALSA: hda/conexant: Fix headset auto detect fail in cx8070 and SN6140 When OMTP headset plugin the headset jack of CX8070 and SN6160 sound cards, the headset type detection circuit will recognize the headset type as CTIA. At this point, plugout and plugin the headset will get the correct headset type as OMTP. The reason for the failure of headset type recognition is that the sound card creation will enable the VREF voltage of the headset mic, which interferes with the headset type automatic detection circuit. Plugout and plugin the headset will restart the headset detection and get the correct headset type. The patch is disable the VREF voltage when the headset is not present, and will enable the VREF voltage when the headset is present. Signed-off-by: bo liu Link: https://lore.kernel.org/r/20240108110235.3867-1-bo.liu@senarytech.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 115 ++++++++++++++++++++++++++++++++- 1 file changed, 113 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a889cccdd607..e8819e8a9876 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -21,6 +21,12 @@ #include "hda_jack.h" #include "hda_generic.h" +enum { + CX_HEADSET_NOPRESENT = 0, + CX_HEADSET_PARTPRESENT, + CX_HEADSET_ALLPRESENT, +}; + struct conexant_spec { struct hda_gen_spec gen; @@ -42,7 +48,8 @@ struct conexant_spec { unsigned int gpio_led; unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; - + unsigned int headset_present_flag; + bool is_cx8070_sn6140; }; @@ -164,6 +171,27 @@ static void cxt_init_gpio_led(struct hda_codec *codec) } } +static void cx_fixup_headset_recog(struct hda_codec *codec) +{ + unsigned int mic_persent; + + /* fix some headset type recognize fail issue, such as EDIFIER headset */ + /* set micbiasd output current comparator threshold from 66% to 55%. */ + snd_hda_codec_write(codec, 0x1c, 0, 0x320, 0x010); + /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias registor + * value adjustment trim from 2.2K ohms to 2.0K ohms. + */ + snd_hda_codec_write(codec, 0x1c, 0, 0x3b0, 0xe10); + /* fix reboot headset type recognize fail issue */ + mic_persent = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); + if (mic_persent & AC_PINSENSE_PRESENCE) + /* enable headset mic VREF */ + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24); + else + /* disable headset mic VREF */ + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20); +} + static int cx_auto_init(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -174,6 +202,9 @@ static int cx_auto_init(struct hda_codec *codec) cxt_init_gpio_led(codec); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + if (spec->is_cx8070_sn6140) + cx_fixup_headset_recog(codec); + return 0; } @@ -192,6 +223,77 @@ static void cx_auto_free(struct hda_codec *codec) snd_hda_gen_free(codec); } +static void cx_process_headset_plugin(struct hda_codec *codec) +{ + unsigned int val; + unsigned int count = 0; + + /* Wait headset detect done. */ + do { + val = snd_hda_codec_read(codec, 0x1c, 0, 0xca0, 0x0); + if (val & 0x080) { + codec_dbg(codec, "headset type detect done!\n"); + break; + } + msleep(20); + count++; + } while (count < 3); + val = snd_hda_codec_read(codec, 0x1c, 0, 0xcb0, 0x0); + if (val & 0x800) { + codec_dbg(codec, "headset plugin, type is CTIA\n"); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24); + } else if (val & 0x400) { + codec_dbg(codec, "headset plugin, type is OMTP\n"); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24); + } else { + codec_dbg(codec, "headphone plugin\n"); + } +} + +static void cx_update_headset_mic_vref(struct hda_codec *codec, unsigned int res) +{ + unsigned int phone_present, mic_persent, phone_tag, mic_tag; + struct conexant_spec *spec = codec->spec; + + /* In cx8070 and sn6140, the node 16 can only be config to headphone or disabled, + * the node 19 can only be config to microphone or disabled. + * Check hp&mic tag to process headset pulgin&plugout. + */ + phone_tag = snd_hda_codec_read(codec, 0x16, 0, AC_VERB_GET_UNSOLICITED_RESPONSE, 0x0); + mic_tag = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_UNSOLICITED_RESPONSE, 0x0); + if ((phone_tag & (res >> AC_UNSOL_RES_TAG_SHIFT)) || + (mic_tag & (res >> AC_UNSOL_RES_TAG_SHIFT))) { + phone_present = snd_hda_codec_read(codec, 0x16, 0, AC_VERB_GET_PIN_SENSE, 0x0); + if (!(phone_present & AC_PINSENSE_PRESENCE)) {/* headphone plugout */ + spec->headset_present_flag = CX_HEADSET_NOPRESENT; + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20); + return; + } + if (spec->headset_present_flag == CX_HEADSET_NOPRESENT) { + spec->headset_present_flag = CX_HEADSET_PARTPRESENT; + } else if (spec->headset_present_flag == CX_HEADSET_PARTPRESENT) { + mic_persent = snd_hda_codec_read(codec, 0x19, 0, + AC_VERB_GET_PIN_SENSE, 0x0); + /* headset is present */ + if ((phone_present & AC_PINSENSE_PRESENCE) && + (mic_persent & AC_PINSENSE_PRESENCE)) { + cx_process_headset_plugin(codec); + spec->headset_present_flag = CX_HEADSET_ALLPRESENT; + } + } + } +} + +static void cx_jack_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct conexant_spec *spec = codec->spec; + + if (spec->is_cx8070_sn6140) + cx_update_headset_mic_vref(codec, res); + + snd_hda_jack_unsol_event(codec, res); +} + #ifdef CONFIG_PM static int cx_auto_suspend(struct hda_codec *codec) { @@ -205,7 +307,7 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .build_pcms = snd_hda_gen_build_pcms, .init = cx_auto_init, .free = cx_auto_free, - .unsol_event = snd_hda_jack_unsol_event, + .unsol_event = cx_jack_unsol_event, #ifdef CONFIG_PM .suspend = cx_auto_suspend, .check_power_status = snd_hda_gen_check_power_status, @@ -1042,6 +1144,15 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = cx_auto_patch_ops; + /* init cx8070/sn6140 flag and reset headset_present_flag */ + switch (codec->core.vendor_id) { + case 0x14f11f86: + case 0x14f11f87: + spec->is_cx8070_sn6140 = true; + spec->headset_present_flag = CX_HEADSET_NOPRESENT; + break; + } + cx_auto_parse_eapd(codec); spec->gen.own_eapd_ctl = 1; From 6b3d14b7f9b1acaf7303d8499836bf78ee9c470c Mon Sep 17 00:00:00 2001 From: Tom Jason Schwanke Date: Mon, 8 Jan 2024 16:15:21 +0100 Subject: [PATCH 332/337] ALSA: hda/realtek: Fix mute and mic-mute LEDs for HP Envy X360 13-ay0xxx This enables the mute and mic-mute LEDs on the HP Envy X360 13-ay0xxx convertibles. The quirk 'ALC245_FIXUP_HP_X360_MUTE_LEDS' already exists and is now enabled for this device. Link: https://bugzilla.kernel.org/show_bug.cgi?id=216197 Signed-off-by: Tom Jason Schwanke Cc: Link: https://lore.kernel.org/r/651b26e9-e86b-45dd-aa90-3e43d6b99823@catboys.cloud Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1dcfba27e075..b68c94757051 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9842,6 +9842,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8735, "HP ProBook 435 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x876e, "HP ENVY x360 Convertible 13-ay0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8780, "HP ZBook Fury 17 G7 Mobile Workstation", From dcaca1b5f0d47f4ed59f606795df531cc8a2d7d2 Mon Sep 17 00:00:00 2001 From: Gergo Koteles Date: Mon, 8 Jan 2024 22:16:46 +0100 Subject: [PATCH 333/337] ALSA: hda/tas2781: annotate calibration data endianness Sparse reports an endian mismatch. The amplifier expects the calibration data as big-endian. Use the __be32 type to express endianness better. Fixes: c3ca4458cc2f ("ALSA: hda/tas2781: add TAS2563 support for 14ARB7") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202401072137.Oc7pQgRW-lkp@intel.com/ Signed-off-by: Gergo Koteles Link: https://lore.kernel.org/r/3852ff28ea7d5d8f2086d8dd78aeff8d1d984991.1704748435.git.soyer@irl.hu Signed-off-by: Takashi Iwai --- sound/pci/hda/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 4805cf0b6480..2dd809de62e5 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -424,8 +424,8 @@ static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { static void tas2563_apply_calib(struct tasdevice_priv *tas_priv) { - unsigned int data; int offset = 0; + __be32 data; int ret; for (int i = 0; i < tas_priv->ndev; i++) { From 8c51c13dc63d46e754c44215eabc0890a8bd9bfb Mon Sep 17 00:00:00 2001 From: Mirsad Todorovac Date: Sun, 7 Jan 2024 18:37:02 +0100 Subject: [PATCH 334/337] kselftest/alsa - mixer-test: fix the number of parameters to ksft_exit_fail_msg() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Minor fix in the number of arguments to error reporting function in the test program as reported by GCC 13.2.0 warning. mixer-test.c: In function ‘find_controls’: mixer-test.c:169:44: warning: too many arguments for format [-Wformat-extra-args] 169 | ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n", | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ The number of arguments in call to ksft_exit_fail_msg() doesn't correspond to the format specifiers, so this is adjusted resembling the sibling calls to the error function. Fixes: b1446bda56456 ("kselftest: alsa: Check for event generation when we write to controls") Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Shuah Khan Cc: linux-sound@vger.kernel.org Cc: linux-kselftest@vger.kernel.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mirsad Todorovac Acked-by: Mark Brown Link: https://lore.kernel.org/r/20240107173704.937824-2-mirsad.todorovac@alu.unizg.hr Signed-off-by: Takashi Iwai --- tools/testing/selftests/alsa/mixer-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 23df154fcdd7..208c2170c074 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -166,7 +166,7 @@ static void find_controls(void) err = snd_ctl_poll_descriptors(card_data->handle, &card_data->pollfd, 1); if (err != 1) { - ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n", + ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for card %d: %d\n", card, err); } From 3f47c1ebe5ca9c5883e596c7888dec4bec0176d8 Mon Sep 17 00:00:00 2001 From: Mirsad Todorovac Date: Sun, 7 Jan 2024 18:37:04 +0100 Subject: [PATCH 335/337] kselftest/alsa - mixer-test: Fix the print format specifier warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The GCC 13.2.0 compiler issued the following warning: mixer-test.c: In function ‘ctl_value_index_valid’: mixer-test.c:322:79: warning: format ‘%lld’ expects argument of type ‘long long int’, \ but argument 5 has type ‘long int’ [-Wformat=] 322 | ksft_print_msg("%s.%d value %lld more than maximum %lld\n", | ~~~^ | | | long long int | %ld 323 | ctl->name, index, int64_val, 324 | snd_ctl_elem_info_get_max(ctl->info)); | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ | | | long int Fixing the format specifier as advised by the compiler suggestion removes the warning. Fixes: 3f48b137d88e7 ("kselftest: alsa: Factor out check that values meet constraints") Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Shuah Khan Cc: linux-sound@vger.kernel.org Cc: linux-kselftest@vger.kernel.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mirsad Todorovac Acked-by: Mark Brown Link: https://lore.kernel.org/r/20240107173704.937824-3-mirsad.todorovac@alu.unizg.hr Signed-off-by: Takashi Iwai --- tools/testing/selftests/alsa/mixer-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 208c2170c074..df942149c6f6 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -319,7 +319,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, } if (int64_val > snd_ctl_elem_info_get_max64(ctl->info)) { - ksft_print_msg("%s.%d value %lld more than maximum %lld\n", + ksft_print_msg("%s.%d value %lld more than maximum %ld\n", ctl->name, index, int64_val, snd_ctl_elem_info_get_max(ctl->info)); return false; From f77a255e74c348a158d21a471e993e70ff710737 Mon Sep 17 00:00:00 2001 From: Mirsad Todorovac Date: Sun, 7 Jan 2024 18:37:06 +0100 Subject: [PATCH 336/337] kselftest/alsa - mixer-test: Fix the print format specifier warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit GCC 13.2.0 compiler issued the following warning: mixer-test.c:350:80: warning: format ‘%ld’ expects argument of type ‘long int’, \ but argument 5 has type ‘unsigned int’ [-Wformat=] 350 | ksft_print_msg("%s.%d value %ld more than item count %ld\n", | ~~^ | | | long int | %d 351 | ctl->name, index, int_val, 352 | snd_ctl_elem_info_get_items(ctl->info)); | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ | | | unsigned int Fixing the format specifier in call to ksft_print_msg() according to the compiler suggestion silences the warning. Fixes: 10f2f194663af ("kselftest: alsa: Validate values read from enumerations") Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Shuah Khan Cc: linux-sound@vger.kernel.org Cc: linux-kselftest@vger.kernel.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mirsad Todorovac Acked-by: Mark Brown Link: https://lore.kernel.org/r/20240107173704.937824-4-mirsad.todorovac@alu.unizg.hr Signed-off-by: Takashi Iwai --- tools/testing/selftests/alsa/mixer-test.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index df942149c6f6..1c04e5f638a0 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -347,7 +347,7 @@ static bool ctl_value_index_valid(struct ctl_data *ctl, } if (int_val >= snd_ctl_elem_info_get_items(ctl->info)) { - ksft_print_msg("%s.%d value %ld more than item count %ld\n", + ksft_print_msg("%s.%d value %ld more than item count %u\n", ctl->name, index, int_val, snd_ctl_elem_info_get_items(ctl->info)); return false; From fd38dd6abda589a8771e7872e4dea28c99c6a6ef Mon Sep 17 00:00:00 2001 From: Mirsad Todorovac Date: Sun, 7 Jan 2024 18:37:08 +0100 Subject: [PATCH 337/337] kselftest/alsa - conf: Stringify the printed errno in sysfs_get() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit GCC 13.2.0 reported the warning of the print format specifier: conf.c: In function ‘sysfs_get’: conf.c:181:72: warning: format ‘%s’ expects argument of type ‘char *’, \ but argument 3 has type ‘int’ [-Wformat=] 181 | ksft_exit_fail_msg("sysfs: unable to read value '%s': %s\n", | ~^ | | | char * | %d The fix passes strerror(errno) as it was intended, like in the sibling error exit message. Fixes: aba51cd0949ae ("selftests: alsa - add PCM test") Cc: Mark Brown Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Shuah Khan Cc: linux-sound@vger.kernel.org Cc: linux-kselftest@vger.kernel.org Cc: linux-kernel@vger.kernel.org Signed-off-by: Mirsad Todorovac Acked-by: Mark Brown Link: https://lore.kernel.org/r/20240107173704.937824-5-mirsad.todorovac@alu.unizg.hr Signed-off-by: Takashi Iwai --- tools/testing/selftests/alsa/conf.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c index 00925eb8d9f4..89e3656a042d 100644 --- a/tools/testing/selftests/alsa/conf.c +++ b/tools/testing/selftests/alsa/conf.c @@ -179,7 +179,7 @@ static char *sysfs_get(const char *sysfs_root, const char *id) close(fd); if (len < 0) ksft_exit_fail_msg("sysfs: unable to read value '%s': %s\n", - path, errno); + path, strerror(errno)); while (len > 0 && path[len-1] == '\n') len--; path[len] = '\0';