From 5871321fb4558c55bf9567052b618ff0be6b975e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 4 Jun 2022 11:52:46 +0100 Subject: [PATCH 01/43] ASoC: ops: Fix off by one in range control validation We currently report that range controls accept a range of 0..(max-min) but accept writes in the range 0..(max-min+1). Remove that extra +1. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220604105246.4055214-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index e693070f51fe..d867f449d82d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -526,7 +526,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) @@ -547,7 +547,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, return -EINVAL; if (mc->platform_max && tmp > mc->platform_max) return -EINVAL; - if (tmp > mc->max - mc->min + 1) + if (tmp > mc->max - mc->min) return -EINVAL; if (invert) From ae8b1631561a3634cc09d0c62bbdd938eade05ec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 Jun 2022 12:11:32 +0200 Subject: [PATCH 02/43] ALSA: usb-audio: Workarounds for Behringer UMC 204/404 HD Both Behringer UMC 202 HD and 404 HD need explicit quirks to enable the implicit feedback mode and start the playback stream primarily. The former seems fixing the stuttering and the latter is required for a playback-only case. Note that the "clock source 41 is not valid" error message still appears even after this fix, but it should be only once at probe. The reason of the error is still unknown, but this seems to be mostly harmless as it's a one-off error and the driver retires the clock setup and it succeeds afterwards. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215934 Cc: Link: https://lore.kernel.org/r/20220624101132.14528-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e8468f9b007d..12ce69b04f63 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1842,6 +1842,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1395, 0x740a, /* Sennheiser DECT */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x1397, 0x0508, /* Behringer UMC204HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x1397, 0x0509, /* Behringer UMC404HD */ + QUIRK_FLAG_PLAYBACK_FIRST | QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x13e5, 0x0001, /* Serato Phono */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x154e, 0x1002, /* Denon DCD-1500RE */ From ac63716da3070f8cb6baaba3a058a0c7f22aeb5b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:46 -0500 Subject: [PATCH 03/43] ASoC: Realtek/Maxim SoundWire codecs: disable pm_runtime on remove When binding/unbinding codec drivers, the following warnings are thrown: [ 107.266879] rt715-sdca sdw:3:025d:0714:01: Unbalanced pm_runtime_enable! [ 306.879700] rt711-sdca sdw:0:025d:0711:01: Unbalanced pm_runtime_enable! Add a remove callback for all Realtek/Maxim SoundWire codecs and remove this warning. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/max98373-sdw.c | 12 +++++++++++- sound/soc/codecs/rt1308-sdw.c | 11 +++++++++++ sound/soc/codecs/rt1316-sdw.c | 11 +++++++++++ sound/soc/codecs/rt5682-sdw.c | 5 ++++- sound/soc/codecs/rt700-sdw.c | 6 +++++- sound/soc/codecs/rt711-sdca-sdw.c | 6 +++++- sound/soc/codecs/rt711-sdw.c | 6 +++++- sound/soc/codecs/rt715-sdca-sdw.c | 12 ++++++++++++ sound/soc/codecs/rt715-sdw.c | 12 ++++++++++++ 9 files changed, 76 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index f47e956d4f55..97b64477dde6 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -862,6 +862,16 @@ static int max98373_sdw_probe(struct sdw_slave *slave, return max98373_init(slave, regmap); } +static int max98373_sdw_remove(struct sdw_slave *slave) +{ + struct max98373_priv *max98373 = dev_get_drvdata(&slave->dev); + + if (max98373->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + #if defined(CONFIG_OF) static const struct of_device_id max98373_of_match[] = { { .compatible = "maxim,max98373", }, @@ -893,7 +903,7 @@ static struct sdw_driver max98373_sdw_driver = { .pm = &max98373_pm, }, .probe = max98373_sdw_probe, - .remove = NULL, + .remove = max98373_sdw_remove, .ops = &max98373_slave_ops, .id_table = max98373_id, }; diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 1c11b42dd76e..72f673f278ee 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -691,6 +691,16 @@ static int rt1308_sdw_probe(struct sdw_slave *slave, return 0; } +static int rt1308_sdw_remove(struct sdw_slave *slave) +{ + struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); + + if (rt1308->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt1308_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x1308, 0x2, 0, 0), {}, @@ -750,6 +760,7 @@ static struct sdw_driver rt1308_sdw_driver = { .pm = &rt1308_pm, }, .probe = rt1308_sdw_probe, + .remove = rt1308_sdw_remove, .ops = &rt1308_slave_ops, .id_table = rt1308_id, }; diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 60baa9ff1907..2d6b5f9d4d77 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -676,6 +676,16 @@ static int rt1316_sdw_probe(struct sdw_slave *slave, return rt1316_sdw_init(&slave->dev, regmap, slave); } +static int rt1316_sdw_remove(struct sdw_slave *slave) +{ + struct rt1316_sdw_priv *rt1316 = dev_get_drvdata(&slave->dev); + + if (rt1316->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt1316_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x1316, 0x3, 0x1, 0), {}, @@ -735,6 +745,7 @@ static struct sdw_driver rt1316_sdw_driver = { .pm = &rt1316_pm, }, .probe = rt1316_sdw_probe, + .remove = rt1316_sdw_remove, .ops = &rt1316_slave_ops, .id_table = rt1316_id, }; diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 248257a2e4e0..f04e18c32489 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -719,9 +719,12 @@ static int rt5682_sdw_remove(struct sdw_slave *slave) { struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); - if (rt5682 && rt5682->hw_init) + if (rt5682->hw_init) cancel_delayed_work_sync(&rt5682->jack_detect_work); + if (rt5682->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index bda594899664..f7439e40ca8b 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include "rt700.h" @@ -463,11 +464,14 @@ static int rt700_sdw_remove(struct sdw_slave *slave) { struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev); - if (rt700 && rt700->hw_init) { + if (rt700->hw_init) { cancel_delayed_work_sync(&rt700->jack_detect_work); cancel_delayed_work_sync(&rt700->jack_btn_check_work); } + if (rt700->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index aaf5af153d3f..c722a2b0041f 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -11,6 +11,7 @@ #include #include #include +#include #include "rt711-sdca.h" #include "rt711-sdca-sdw.h" @@ -364,11 +365,14 @@ static int rt711_sdca_sdw_remove(struct sdw_slave *slave) { struct rt711_sdca_priv *rt711 = dev_get_drvdata(&slave->dev); - if (rt711 && rt711->hw_init) { + if (rt711->hw_init) { cancel_delayed_work_sync(&rt711->jack_detect_work); cancel_delayed_work_sync(&rt711->jack_btn_check_work); } + if (rt711->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index bda2cc9439c9..f49c94baa37c 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include "rt711.h" @@ -464,12 +465,15 @@ static int rt711_sdw_remove(struct sdw_slave *slave) { struct rt711_priv *rt711 = dev_get_drvdata(&slave->dev); - if (rt711 && rt711->hw_init) { + if (rt711->hw_init) { cancel_delayed_work_sync(&rt711->jack_detect_work); cancel_delayed_work_sync(&rt711->jack_btn_check_work); cancel_work_sync(&rt711->calibration_work); } + if (rt711->first_hw_init) + pm_runtime_disable(&slave->dev); + return 0; } diff --git a/sound/soc/codecs/rt715-sdca-sdw.c b/sound/soc/codecs/rt715-sdca-sdw.c index 0ecd2948f7aa..13e731d16675 100644 --- a/sound/soc/codecs/rt715-sdca-sdw.c +++ b/sound/soc/codecs/rt715-sdca-sdw.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include "rt715-sdca.h" @@ -193,6 +194,16 @@ static int rt715_sdca_sdw_probe(struct sdw_slave *slave, return rt715_sdca_init(&slave->dev, mbq_regmap, regmap, slave); } +static int rt715_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt715_sdca_priv *rt715 = dev_get_drvdata(&slave->dev); + + if (rt715->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt715_sdca_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x3, 0x1, 0), SDW_SLAVE_ENTRY_EXT(0x025d, 0x714, 0x3, 0x1, 0), @@ -267,6 +278,7 @@ static struct sdw_driver rt715_sdw_driver = { .pm = &rt715_pm, }, .probe = rt715_sdca_sdw_probe, + .remove = rt715_sdca_sdw_remove, .ops = &rt715_sdca_slave_ops, .id_table = rt715_sdca_id, }; diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index a7b21b03c08b..b047bf87a100 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -514,6 +515,16 @@ static int rt715_sdw_probe(struct sdw_slave *slave, return 0; } +static int rt715_sdw_remove(struct sdw_slave *slave) +{ + struct rt715_priv *rt715 = dev_get_drvdata(&slave->dev); + + if (rt715->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + static const struct sdw_device_id rt715_id[] = { SDW_SLAVE_ENTRY_EXT(0x025d, 0x714, 0x2, 0, 0), SDW_SLAVE_ENTRY_EXT(0x025d, 0x715, 0x2, 0, 0), @@ -575,6 +586,7 @@ static struct sdw_driver rt715_sdw_driver = { .pm = &rt715_pm, }, .probe = rt715_sdw_probe, + .remove = rt715_sdw_remove, .ops = &rt715_slave_ops, .id_table = rt715_id, }; From ed0a7fb29c9fd4f53eeb37d1fe2354df7a038047 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:47 -0500 Subject: [PATCH 04/43] ASoC: rt711-sdca-sdw: fix calibrate mutex initialization In codec driver bind/unbind test, the following warning is thrown: DEBUG_LOCKS_WARN_ON(lock->magic != lock) ... [ 699.182495] rt711_sdca_jack_init+0x1b/0x1d0 [snd_soc_rt711_sdca] [ 699.182498] rt711_sdca_set_jack_detect+0x3b/0x90 [snd_soc_rt711_sdca] [ 699.182500] snd_soc_component_set_jack+0x24/0x50 [snd_soc_core] A quick check in the code shows that the 'calibrate_mutex' used by this driver are not initialized at probe time. Moving the initialization to the probe removes the issue. BugLink: https://github.com/thesofproject/linux/issues/3644 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca-sdw.c | 3 +++ sound/soc/codecs/rt711-sdca.c | 2 +- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index c722a2b0041f..a085b2f530aa 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -373,6 +373,9 @@ static int rt711_sdca_sdw_remove(struct sdw_slave *slave) if (rt711->first_hw_init) pm_runtime_disable(&slave->dev); + mutex_destroy(&rt711->calibrate_mutex); + mutex_destroy(&rt711->disable_irq_lock); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 57629c18db38..af73bcb4560a 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -1412,6 +1412,7 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, rt711->regmap = regmap; rt711->mbq_regmap = mbq_regmap; + mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); /* @@ -1550,7 +1551,6 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt711_sdca_jack_detect_handler); INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_sdca_btn_check_handler); - mutex_init(&rt711->calibrate_mutex); } /* calibration */ From fe154c4ff376bc31041c6441958a08243df09c99 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:48 -0500 Subject: [PATCH 05/43] ASoC: Intel: sof_sdw: handle errors on card registration If the card registration fails, typically because of deferred probes, the device properties added for headset codecs are not removed, which leads to kernel oopses in driver bind/unbind tests. We already clean-up the device properties when the card is removed, this code can be moved as a helper and called upon card registration errors. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 51 ++++++++++++++++++-------------- 1 file changed, 29 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1f00679b4240..ad826ad82d51 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1398,6 +1398,33 @@ static struct snd_soc_card card_sof_sdw = { .late_probe = sof_sdw_card_late_probe, }; +static void mc_dailink_exit_loop(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *link; + int ret; + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (!codec_info_list[i].exit) + continue; + /* + * We don't need to call .exit function if there is no matched + * dai link found. + */ + for_each_card_prelinks(card, j, link) { + if (!strcmp(link->codecs[0].dai_name, + codec_info_list[i].dai_name)) { + ret = codec_info_list[i].exit(card, link); + if (ret) + dev_warn(card->dev, + "codec exit failed %d\n", + ret); + break; + } + } + } +} + static int mc_probe(struct platform_device *pdev) { struct snd_soc_card *card = &card_sof_sdw; @@ -1462,6 +1489,7 @@ static int mc_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(card->dev, "snd_soc_register_card failed %d\n", ret); + mc_dailink_exit_loop(card); return ret; } @@ -1473,29 +1501,8 @@ static int mc_probe(struct platform_device *pdev) static int mc_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct snd_soc_dai_link *link; - int ret; - int i, j; - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { - if (!codec_info_list[i].exit) - continue; - /* - * We don't need to call .exit function if there is no matched - * dai link found. - */ - for_each_card_prelinks(card, j, link) { - if (!strcmp(link->codecs[0].dai_name, - codec_info_list[i].dai_name)) { - ret = codec_info_list[i].exit(card, link); - if (ret) - dev_warn(&pdev->dev, - "codec exit failed %d\n", - ret); - break; - } - } - } + mc_dailink_exit_loop(card); return 0; } From 08bb5dc6ce02374169213cea772b1c297eaf32d5 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:49 -0500 Subject: [PATCH 06/43] ASoC: rt711: fix calibrate mutex initialization Follow the same flow as rt711-sdca and initialize all mutexes at probe time. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdw.c | 3 +++ sound/soc/codecs/rt711.c | 2 +- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index f49c94baa37c..4fe68bcf2a7c 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -474,6 +474,9 @@ static int rt711_sdw_remove(struct sdw_slave *slave) if (rt711->first_hw_init) pm_runtime_disable(&slave->dev); + mutex_destroy(&rt711->calibrate_mutex); + mutex_destroy(&rt711->disable_irq_lock); + return 0; } diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 9838fb4d5b9c..1e35ba433a7e 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -1204,6 +1204,7 @@ int rt711_init(struct device *dev, struct regmap *sdw_regmap, rt711->sdw_regmap = sdw_regmap; rt711->regmap = regmap; + mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); /* @@ -1318,7 +1319,6 @@ int rt711_io_init(struct device *dev, struct sdw_slave *slave) rt711_jack_detect_handler); INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_btn_check_handler); - mutex_init(&rt711->calibrate_mutex); INIT_WORK(&rt711->calibration_work, rt711_calibration_work); schedule_work(&rt711->calibration_work); } From 0484271ab0ce50649329fa9dc23c50853c5b26a4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:50 -0500 Subject: [PATCH 07/43] ASoC: rt7*-sdw: harden jack_detect_handler Realtek headset codec drivers typically check if the card is instantiated before proceeding with the jack detection. The rt700, rt711 and rt711-sdca are however missing a check on the card pointer, which can lead to NULL dereferences encountered in driver bind/unbind tests. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index af32295fa9b9..4a99d5f4706f 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -162,7 +162,7 @@ static void rt700_jack_detect_handler(struct work_struct *work) if (!rt700->hs_jack) return; - if (!rt700->component->card->instantiated) + if (!rt700->component->card || !rt700->component->card->instantiated) return; reg = RT700_VERB_GET_PIN_SENSE | RT700_HP_OUT; diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index af73bcb4560a..93b36f05cb56 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -294,7 +294,7 @@ static void rt711_sdca_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return; - if (!rt711->component->card->instantiated) + if (!rt711->component->card || !rt711->component->card->instantiated) return; /* SDW_SCP_SDCA_INT_SDCA_0 is used for jack detection */ diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 1e35ba433a7e..2f445b27305a 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -242,7 +242,7 @@ static void rt711_jack_detect_handler(struct work_struct *work) if (!rt711->hs_jack) return; - if (!rt711->component->card->instantiated) + if (!rt711->component->card || !rt711->component->card->instantiated) return; if (pm_runtime_status_suspended(rt711->slave->dev.parent)) { From ba98d7d8b60ba410aa03834f6aa48fd3b2e68478 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:51 -0500 Subject: [PATCH 08/43] ASoC: codecs: rt700/rt711/rt711-sdca: initialize workqueues in probe The workqueues are initialized in the io_init functions, which isn't quite right. In some tests, this leads to warnings throw from __queue_delayed_work() WARN_ON_FUNCTION_MISMATCH(timer->function, delayed_work_timer_fn); Move all the initializations to the probe functions. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-7-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt700.c | 12 +++++------- sound/soc/codecs/rt711-sdca.c | 10 +++------- sound/soc/codecs/rt711.c | 12 +++++------- 3 files changed, 13 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 4a99d5f4706f..7a6cf3434591 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -1115,6 +1115,11 @@ int rt700_init(struct device *dev, struct regmap *sdw_regmap, mutex_init(&rt700->disable_irq_lock); + INIT_DELAYED_WORK(&rt700->jack_detect_work, + rt700_jack_detect_handler); + INIT_DELAYED_WORK(&rt700->jack_btn_check_work, + rt700_btn_check_handler); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1209,13 +1214,6 @@ int rt700_io_init(struct device *dev, struct sdw_slave *slave) /* Finish Initial Settings, set power to D3 */ regmap_write(rt700->regmap, RT700_SET_AUDIO_POWER_STATE, AC_PWRST_D3); - if (!rt700->first_hw_init) { - INIT_DELAYED_WORK(&rt700->jack_detect_work, - rt700_jack_detect_handler); - INIT_DELAYED_WORK(&rt700->jack_btn_check_work, - rt700_btn_check_handler); - } - /* * if set_jack callback occurred early than io_init, * we set up the jack detection function now diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 93b36f05cb56..2b3b77577d1f 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -1415,6 +1415,9 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); + INIT_DELAYED_WORK(&rt711->jack_detect_work, rt711_sdca_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_sdca_btn_check_handler); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1546,13 +1549,6 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, RT711_PUSH_BTN_INT_CTL0, 0x20, 0x00); - if (!rt711->first_hw_init) { - INIT_DELAYED_WORK(&rt711->jack_detect_work, - rt711_sdca_jack_detect_handler); - INIT_DELAYED_WORK(&rt711->jack_btn_check_work, - rt711_sdca_btn_check_handler); - } - /* calibration */ ret = rt711_sdca_calibration(rt711); if (ret < 0) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 2f445b27305a..5709a6bbe8fc 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -1207,6 +1207,10 @@ int rt711_init(struct device *dev, struct regmap *sdw_regmap, mutex_init(&rt711->calibrate_mutex); mutex_init(&rt711->disable_irq_lock); + INIT_DELAYED_WORK(&rt711->jack_detect_work, rt711_jack_detect_handler); + INIT_DELAYED_WORK(&rt711->jack_btn_check_work, rt711_btn_check_handler); + INIT_WORK(&rt711->calibration_work, rt711_calibration_work); + /* * Mark hw_init to false * HW init will be performed when device reports present @@ -1314,14 +1318,8 @@ int rt711_io_init(struct device *dev, struct sdw_slave *slave) if (rt711->first_hw_init) rt711_calibration(rt711); - else { - INIT_DELAYED_WORK(&rt711->jack_detect_work, - rt711_jack_detect_handler); - INIT_DELAYED_WORK(&rt711->jack_btn_check_work, - rt711_btn_check_handler); - INIT_WORK(&rt711->calibration_work, rt711_calibration_work); + else schedule_work(&rt711->calibration_work); - } /* * if set_jack callback occurred early than io_init, From 40737057b48f1b4db67b0d766b95c87ba8fc5e03 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 6 Jun 2022 15:37:52 -0500 Subject: [PATCH 09/43] ASoC: codecs: rt700/rt711/rt711-sdca: resume bus/codec in .set_jack_detect The .set_jack_detect() codec component callback is invoked during card registration, which happens when the machine driver is probed. The issue is that this callback can race with the bus suspend/resume, and IO timeouts can happen. This can be reproduced very easily if the machine driver is 'blacklisted' and manually probed after the bus suspends. The bus and codec need to be re-initialized using pm_runtime helpers. Previous contributions tried to make sure accesses to the bus during the .set_jack_detect() component callback only happen when the bus is active. This was done by changing the regcache status on a component remove. This is however a layering violation, the regcache status should only be modified on device probe, suspend and resume. The component probe/remove should not modify how the device regcache is handled. This solution also didn't handle all the possible race conditions, and the RT700 headset codec was not handled. This patch tries to resume the codec device before handling the jack initializations. In case the codec has not yet been initialized, pm_runtime may not be enabled yet, so we don't squelch the -EACCES error code and only stop the jack information. When the codec reports as attached, the jack initialization will proceed as usual. BugLink: https://github.com/thesofproject/linux/issues/3643 Fixes: 7ad4d237e7c4a ('ASoC: rt711-sdca: Add RT711 SDCA vendor-specific driver') Fixes: 899b12542b089 ('ASoC: rt711: add snd_soc_component remove callback') Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220606203752.144159-8-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt700.c | 16 +++++++++++++--- sound/soc/codecs/rt711-sdca.c | 26 ++++++++++++++------------ sound/soc/codecs/rt711.c | 24 +++++++++++++----------- 3 files changed, 40 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/rt700.c b/sound/soc/codecs/rt700.c index 7a6cf3434591..9bceeeb830b1 100644 --- a/sound/soc/codecs/rt700.c +++ b/sound/soc/codecs/rt700.c @@ -315,17 +315,27 @@ static int rt700_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt700_priv *rt700 = snd_soc_component_get_drvdata(component); + int ret; rt700->hs_jack = hs_jack; - if (!rt700->hw_init) { - dev_dbg(&rt700->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt700_jack_init(rt700); + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 2b3b77577d1f..dfe3c9299ebd 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -487,16 +487,27 @@ static int rt711_sdca_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711->hs_jack = hs_jack; - if (!rt711->hw_init) { - dev_dbg(&rt711->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt711_sdca_jack_init(rt711); + + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } @@ -1190,14 +1201,6 @@ static int rt711_sdca_probe(struct snd_soc_component *component) return 0; } -static void rt711_sdca_remove(struct snd_soc_component *component) -{ - struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - - regcache_cache_only(rt711->regmap, true); - regcache_cache_only(rt711->mbq_regmap, true); -} - static const struct snd_soc_component_driver soc_sdca_dev_rt711 = { .probe = rt711_sdca_probe, .controls = rt711_sdca_snd_controls, @@ -1207,7 +1210,6 @@ static const struct snd_soc_component_driver soc_sdca_dev_rt711 = { .dapm_routes = rt711_sdca_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_sdca_audio_map), .set_jack = rt711_sdca_set_jack_detect, - .remove = rt711_sdca_remove, .endianness = 1, }; diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5709a6bbe8fc..9df800abfc2d 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -457,17 +457,27 @@ static int rt711_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) { struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); + int ret; rt711->hs_jack = hs_jack; - if (!rt711->hw_init) { - dev_dbg(&rt711->slave->dev, - "%s hw_init not ready yet\n", __func__); + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); return 0; } rt711_jack_init(rt711); + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + return 0; } @@ -932,13 +942,6 @@ static int rt711_probe(struct snd_soc_component *component) return 0; } -static void rt711_remove(struct snd_soc_component *component) -{ - struct rt711_priv *rt711 = snd_soc_component_get_drvdata(component); - - regcache_cache_only(rt711->regmap, true); -} - static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .probe = rt711_probe, .set_bias_level = rt711_set_bias_level, @@ -949,7 +952,6 @@ static const struct snd_soc_component_driver soc_codec_dev_rt711 = { .dapm_routes = rt711_audio_map, .num_dapm_routes = ARRAY_SIZE(rt711_audio_map), .set_jack = rt711_set_jack_detect, - .remove = rt711_remove, .endianness = 1, }; From ed0073bd0fccec459b526918be70bf9dc551581a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 8 Jun 2022 02:09:16 +0000 Subject: [PATCH 10/43] ASoC: ak4613: cares Simple-Audio-Card case for TDM Renesas is the only user of ak4613 on upstream for now, and commit f28dbaa958fbd8 ("ASoC: ak4613: add TDM256 support") added TDM256 support. Renesas tested part of it, because of board connection. It was assuming ak4613 is probed via Audio-Graph-Card, but it might be probed via Simple-Audio-Card either. It will indicates WARNING in such case. This patch fixup it. Reported-by: Geert Uytterhoeven Signed-off-by: Kuninori Morimoto Tested-by: Geert Uytterhoeven Link: https://lore.kernel.org/r/87h74v29f7.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 55e773f92122..93606e5afd8f 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -868,10 +868,12 @@ static void ak4613_parse_of(struct ak4613_priv *priv, /* * connected STDI + * TDM support is assuming it is probed via Audio-Graph-Card style here. + * Default is SDTIx1 if it was probed via Simple-Audio-Card for now. */ sdti_num = of_graph_get_endpoint_count(np); - if (WARN_ON((sdti_num > 3) || (sdti_num < 1))) - return; + if ((sdti_num >= SDTx_MAX) || (sdti_num < 1)) + sdti_num = 1; AK4613_CONFIG_SDTI_set(priv, sdti_num); } From 08f8a93198e300dff9649bbae424cd805d313326 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Jun 2022 11:59:47 +0300 Subject: [PATCH 11/43] ASoC: SOF: Intel: hda-dsp: Expose hda_dsp_core_power_up() The hda_dsp_core_power_up() needs to be exposed so that it can be used in hda-loader.c to correct the boot flow. The first step must not unstall the core, it should only power up the core(s). Add sanity check for the core_mask while exposing it to be safe. Complements: 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609085949.29062-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-dsp.c | 10 +++++++++- sound/soc/sof/intel/hda.h | 1 + 2 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 000ea906670c..e24eea725acb 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -181,12 +181,20 @@ int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask) * Power Management. */ -static int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask) { + struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; + const struct sof_intel_dsp_desc *chip = hda->desc; unsigned int cpa; u32 adspcs; int ret; + /* restrict core_mask to host managed cores mask */ + core_mask &= chip->host_managed_cores_mask; + /* return if core_mask is not valid */ + if (!core_mask) + return 0; + /* update bits */ snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPCS, HDA_DSP_ADSPCS_SPA_MASK(core_mask), diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3e0f7b0c586a..0f57ef5d9b8e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -497,6 +497,7 @@ struct sof_intel_hda_stream { */ int hda_dsp_probe(struct snd_sof_dev *sdev); int hda_dsp_remove(struct snd_sof_dev *sdev); +int hda_dsp_core_power_up(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_core_run(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_enable_core(struct snd_sof_dev *sdev, unsigned int core_mask); int hda_dsp_core_reset_power_down(struct snd_sof_dev *sdev, From c31691e0d126ec5d60d2b6b03f699c11b613b219 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Jun 2022 11:59:48 +0300 Subject: [PATCH 12/43] ASoC: SOF: Intel: hda-loader: Make sure that the fw load sequence is followed The hda_dsp_enable_core() is powering up _and_ unstall the core in one call while the first step of the firmware loading must not unstall the core. The core can be unstalled only after the set cpb_cfp and the configuration of the IPC register for the ROM_CONTROL message. Complements: 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609085949.29062-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 64290125d7cd..103e62bcfa82 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -110,7 +110,7 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) int ret; /* step 1: power up corex */ - ret = hda_dsp_enable_core(sdev, chip->host_managed_cores_mask); + ret = hda_dsp_core_power_up(sdev, chip->host_managed_cores_mask); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, "error: dsp core 0/1 power up failed\n"); From bbfef046c6613404c01aeb9e9928bebb78dd327a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 9 Jun 2022 11:59:49 +0300 Subject: [PATCH 13/43] ASoC: SOF: Intel: hda-loader: Clarify the cl_dsp_init() flow Update the comment for the cl_dsp_init() to clarify what is done by the function and use the chip->init_core_mask instead of BIT(0) when unstalling/running the init core. Complements: 2a68ff846164 ("ASoC: SOF: Intel: hda: Revisit IMR boot sequence") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220609085949.29062-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 103e62bcfa82..d3ec5996a9a3 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -95,9 +95,9 @@ out_put: } /* - * first boot sequence has some extra steps. core 0 waits for power - * status on core 1, so power up core 1 also momentarily, keep it in - * reset/stall and then turn it off + * first boot sequence has some extra steps. + * power on all host managed cores and only unstall/run the boot core to boot the + * DSP then turn off all non boot cores (if any) is powered on. */ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) { @@ -127,7 +127,7 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag, bool imr_boot) snd_sof_dsp_write(sdev, HDA_DSP_BAR, chip->ipc_req, ipc_hdr); /* step 3: unset core 0 reset state & unstall/run core 0 */ - ret = hda_dsp_core_run(sdev, BIT(0)); + ret = hda_dsp_core_run(sdev, chip->init_core_mask); if (ret < 0) { if (hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS) dev_err(sdev->dev, From c2d1aec3f5da2475aa13a487d329823b7d27d499 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 10 Jun 2022 11:47:35 +0300 Subject: [PATCH 14/43] ASoC: SOF: ipc3-topology: Move and correct size checks in sof_ipc3_control_load_bytes() Move the size checks prior to allocating memory as these checks do not need the data to be allocated and in case of an error we would not need to free the allocation. The max size must not be less than the size of struct sof_ipc_ctrl_data + struct sof_abi_hdr as the ABI header needs to be present under all circumstances. The check was incorrectly used or between the two size checks. Fixes: b5cee8feb1d4 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220610084735.19397-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 27 +++++++++++++-------------- 1 file changed, 13 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 043554d7cb4a..10740c55294d 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1577,24 +1577,23 @@ static int sof_ipc3_control_load_bytes(struct snd_sof_dev *sdev, struct snd_sof_ struct sof_ipc_ctrl_data *cdata; int ret; + if (scontrol->max_size < (sizeof(*cdata) + sizeof(struct sof_abi_hdr))) { + dev_err(sdev->dev, "%s: insufficient size for a bytes control: %zu.\n", + __func__, scontrol->max_size); + return -EINVAL; + } + + if (scontrol->priv_size > scontrol->max_size - sizeof(*cdata)) { + dev_err(sdev->dev, + "%s: bytes data size %zu exceeds max %zu.\n", __func__, + scontrol->priv_size, scontrol->max_size - sizeof(*cdata)); + return -EINVAL; + } + scontrol->ipc_control_data = kzalloc(scontrol->max_size, GFP_KERNEL); if (!scontrol->ipc_control_data) return -ENOMEM; - if (scontrol->max_size < sizeof(*cdata) || - scontrol->max_size < sizeof(struct sof_abi_hdr)) { - ret = -EINVAL; - goto err; - } - - /* init the get/put bytes data */ - if (scontrol->priv_size > scontrol->max_size - sizeof(*cdata)) { - dev_err(sdev->dev, "err: bytes data size %zu exceeds max %zu.\n", - scontrol->priv_size, scontrol->max_size - sizeof(*cdata)); - ret = -EINVAL; - goto err; - } - scontrol->size = sizeof(struct sof_ipc_ctrl_data) + scontrol->priv_size; cdata = scontrol->ipc_control_data; From af2d146a8041c67efdd620c9463973ce0650b7b7 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 10 Jun 2022 14:42:57 +0200 Subject: [PATCH 15/43] ASoC: Intel: avs: Fix parsing UUIDs in topology MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Use correct type for parsing UUIDs, this eliminates warning present, when compiling with W=1. Fixes: 34ae2cd53673 ("ASoC: Intel: avs: Add topology parsing infrastructure") Reported-by: Pierre-Louis Bossart Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220610124257.4160658-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index 0d11cc8aab0b..6a06fe387d13 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -128,10 +128,10 @@ struct avs_tplg_token_parser { static int avs_parse_uuid_token(struct snd_soc_component *comp, void *elem, void *object, u32 offset) { - struct snd_soc_tplg_vendor_value_elem *tuple = elem; + struct snd_soc_tplg_vendor_uuid_elem *tuple = elem; guid_t *val = (guid_t *)((u8 *)object + offset); - guid_copy((guid_t *)val, (const guid_t *)&tuple->value); + guid_copy((guid_t *)val, (const guid_t *)&tuple->uuid); return 0; } From 12abc4d10d5502e4f3d8f1c6f9e8245747a44708 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Amadeusz=20S=C5=82awi=C5=84ski?= Date: Fri, 10 Jun 2022 14:44:20 +0200 Subject: [PATCH 16/43] ASoC: Remove unused hw_write_t type MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Commit 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") removed use of hw_write_t in struct snd_soc_codec, but it left type definition. Fully clean it up. Fixes: 81da8a0b7975 ("ASoC: remove codec hw_write/control_data") Signed-off-by: Amadeusz Sławiński Reviewed-by: Cezary Rojewski Link: https://lore.kernel.org/r/20220610124420.4160986-1-amadeuszx.slawinski@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f20f5f890794..b276dcb5d4e8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -408,8 +408,6 @@ struct snd_soc_jack_pin; struct snd_soc_jack_gpio; -typedef int (*hw_write_t)(void *,const char* ,int); - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, From 58136d93d4e2c1207a5e4f3044815cd40b1d95fd Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 10 Jun 2022 15:48:18 +0100 Subject: [PATCH 17/43] ASoC: qdsp6: q6apm-dai: unprepare stream if its already prepared prepare callback can be called multiple times, so unprepare the stream if its already prepared. Without this DSP is not happy to setting the params on a already prepared graph. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Reported-by: Srinivasa Rao Mandadapu Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220610144818.511797-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/qcom/qdsp6/q6apm-dai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 19c4a90ec1ea..ee59ef36b85a 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -147,6 +147,12 @@ static int q6apm_dai_prepare(struct snd_soc_component *component, cfg.num_channels = runtime->channels; cfg.bit_width = prtd->bits_per_sample; + if (prtd->state) { + /* clear the previous setup if any */ + q6apm_graph_stop(prtd->graph); + q6apm_unmap_memory_regions(prtd->graph, substream->stream); + } + prtd->pcm_count = snd_pcm_lib_period_bytes(substream); prtd->pos = 0; /* rate and channels are sent to audio driver */ From be6dd72edb216f20fc80e426ece9fe9b8aabf033 Mon Sep 17 00:00:00 2001 From: Yassine Oudjana Date: Mon, 6 Jun 2022 19:22:26 +0400 Subject: [PATCH 18/43] ASoC: wcd9335: Remove RX channel from old list before adding it to a new one Currently in slim_rx_mux_put, an RX channel gets added to a new list even if it is already in one. This can mess up links and make either it, the new list head, or both, get linked to the wrong entries. This can cause an entry to link to itself which in turn ends up making list_for_each_entry in other functions loop infinitely. To avoid issues, always remove the RX channel from any list it's in before adding it to a new list. Signed-off-by: Yassine Oudjana Link: https://lore.kernel.org/r/20220606152226.149164-1-y.oudjana@protonmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 617a36a89dfe..597420679505 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1289,9 +1289,12 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; + /* Remove channel from any list it's in before adding it to a new one */ + list_del_init(&wcd->rx_chs[port_id].list); + switch (wcd->rx_port_value[port_id]) { case 0: - list_del_init(&wcd->rx_chs[port_id].list); + /* Channel already removed from lists. Nothing to do here */ break; case 1: list_add_tail(&wcd->rx_chs[port_id].list, From a7786cbae4b2732815da98efa39df96746b5bd0d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2022 14:46:09 +0200 Subject: [PATCH 19/43] ASoC: wcd9335: Fix spurious event generation The slimbus mux put operation unconditionally reports a change in value which means that spurious events are generated. Fix this by exiting early in that case. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220603124609.4024666-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd9335.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 597420679505..d9f135200688 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -1287,6 +1287,9 @@ static int slim_rx_mux_put(struct snd_kcontrol *kc, struct snd_soc_dapm_update *update = NULL; u32 port_id = w->shift; + if (wcd->rx_port_value[port_id] == ucontrol->value.enumerated.item[0]) + return 0; + wcd->rx_port_value[port_id] = ucontrol->value.enumerated.item[0]; /* Remove channel from any list it's in before adding it to a new one */ From 10e7ff0047921e32b919ecee7be706dd33c107f8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2022 14:25:26 +0200 Subject: [PATCH 20/43] ASoC: wcd938x: Fix event generation for some controls Currently wcd938x_*_put() unconditionally report that the value of the control changed, resulting in spurious events being generated. Return 0 in that case instead as we should. There is still an issue in the compander control which is a bit more complex. Signed-off-by: Mark Brown Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220603122526.3914942-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index c1b61b997f69..781ae569be29 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -2519,6 +2519,9 @@ static int wcd938x_tx_mode_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int path = e->shift_l; + if (wcd938x->tx_mode[path] == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->tx_mode[path] = ucontrol->value.enumerated.item[0]; return 1; @@ -2541,6 +2544,9 @@ static int wcd938x_rx_hph_mode_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->hph_mode == ucontrol->value.enumerated.item[0]) + return 0; + wcd938x->hph_mode = ucontrol->value.enumerated.item[0]; return 1; @@ -2632,6 +2638,9 @@ static int wcd938x_ldoh_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->ldoh == ucontrol->value.integer.value[0]) + return 0; + wcd938x->ldoh = ucontrol->value.integer.value[0]; return 1; @@ -2654,6 +2663,9 @@ static int wcd938x_bcs_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + if (wcd938x->bcs_dis == ucontrol->value.integer.value[0]) + return 0; + wcd938x->bcs_dis = ucontrol->value.integer.value[0]; return 1; From a7d9391dc3d570aed87ed764db95b16760c898e4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 10 Jun 2022 16:43:13 -0500 Subject: [PATCH 21/43] MAINTAINERS: update ASoC/Intel/SOF maintainers MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Keyon Jie was a key contributor to the Intel ASoC and SOF Intel drivers, but he's moved on to a different role within Intel. We wish him all the best in his new endeavors. Bard Liao, Kai Vehmanen, Ranjani Sridharan and Peter Ujfalusi have been involved in the Intel multi-maintainer team, it's time to update the MAINTAINERS entry to reflect their contributions and clarify their role. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Reviewed-by: Bard Liao Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220610214313.42903-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- MAINTAINERS | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a6d3bd9d2a8d..440f3d7c93b9 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -9803,7 +9803,10 @@ INTEL ASoC DRIVERS M: Cezary Rojewski M: Pierre-Louis Bossart M: Liam Girdwood -M: Jie Yang +M: Peter Ujfalusi +M: Bard Liao +M: Ranjani Sridharan +M: Kai Vehmanen L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/intel/ @@ -18670,8 +18673,10 @@ F: sound/soc/ SOUND - SOUND OPEN FIRMWARE (SOF) DRIVERS M: Pierre-Louis Bossart M: Liam Girdwood +M: Peter Ujfalusi +M: Bard Liao M: Ranjani Sridharan -M: Kai Vehmanen +R: Kai Vehmanen M: Daniel Baluta L: sound-open-firmware@alsa-project.org (moderated for non-subscribers) S: Supported From 4e07479eab8a044cc9542414ccb4aeb8eb033bde Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 12 Jun 2022 17:56:52 +0200 Subject: [PATCH 22/43] ASoC: Intel: bytcr_wm5102: Fix GPIO related probe-ordering problem The "wlf,spkvdd-ena" GPIO needed by the bytcr_wm5102 driver is made available through a gpio-lookup table. This gpio-lookup table is registered by drivers/mfd/arizona-spi.c, which may get probed after the bytcr_wm5102 driver. If the gpio-lookup table has not registered yet then the gpiod_get() will return -ENOENT. Treat -ENOENT as -EPROBE_DEFER to still keep things working in this case. Signed-off-by: Hans de Goede Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220612155652.107310-1-hdegoede@redhat.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcr_wm5102.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index 00384c6fbcaa..330c0ace1638 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -421,8 +421,17 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) priv->spkvdd_en_gpio = gpiod_get(codec_dev, "wlf,spkvdd-ena", GPIOD_OUT_LOW); put_device(codec_dev); - if (IS_ERR(priv->spkvdd_en_gpio)) - return dev_err_probe(dev, PTR_ERR(priv->spkvdd_en_gpio), "getting spkvdd-GPIO\n"); + if (IS_ERR(priv->spkvdd_en_gpio)) { + ret = PTR_ERR(priv->spkvdd_en_gpio); + /* + * The spkvdd gpio-lookup is registered by: drivers/mfd/arizona-spi.c, + * so -ENOENT means that arizona-spi hasn't probed yet. + */ + if (ret == -ENOENT) + ret = -EPROBE_DEFER; + + return dev_err_probe(dev, ret, "getting spkvdd-GPIO\n"); + } /* override platform name, if required */ byt_wm5102_card.dev = dev; From 427eb3e1ed530792231bc5bbee6dec99fc57aeb7 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 15 Jun 2022 11:19:44 +0300 Subject: [PATCH 23/43] ASoC: SOF: mediatek: Fix error code in probe This should return PTR_ERR() instead of IS_ERR(). Fixes: e0100bfd383c ("ASoC: SOF: mediatek: Add mt8186 ipc support") Signed-off-by: Dan Carpenter Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/YqmWIK8sTj578OJP@kili Signed-off-by: Mark Brown --- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 3333a0634e29..e006532caf2f 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -392,7 +392,7 @@ static int mt8186_dsp_probe(struct snd_sof_dev *sdev) PLATFORM_DEVID_NONE, pdev, sizeof(*pdev)); if (IS_ERR(priv->ipc_dev)) { - ret = IS_ERR(priv->ipc_dev); + ret = PTR_ERR(priv->ipc_dev); dev_err(sdev->dev, "failed to create mtk-adsp-ipc device\n"); goto err_adsp_off; } From ca7ab1dcf58dfce5bc851bf7e50fd94822c24665 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 16 Jun 2022 15:19:53 -0500 Subject: [PATCH 24/43] ASoC: SOF: Intel: hda: Fix compressed stream position tracking Commit 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") modified the PCM path only, but left the compressed data patch using an obsolete option. Move the functionality in a helper that can be called for both PCM and compressed data. Reviewed-by: Ranjani Sridharan Fixes: 288fad2f71fa ("ASoC: SOF: Intel: hda: add quirks for HDAudio DMA position information") Signed-off-by: Peter Ujfalusi Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220616201953.130876-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-pcm.c | 74 +------------------------ sound/soc/sof/intel/hda-stream.c | 94 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda.h | 3 + 3 files changed, 94 insertions(+), 77 deletions(-) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index dc1f743730c0..6888e0a4665d 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -192,79 +192,7 @@ snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, goto found; } - switch (sof_hda_position_quirk) { - case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: - /* - * This legacy code, inherited from the Skylake driver, - * mixes DPIB registers and DPIB DDR updates and - * does not seem to follow any known hardware recommendations. - * It's not clear e.g. why there is a different flow - * for capture and playback, the only information that matters is - * what traffic class is used, and on all SOF-enabled platforms - * only VC0 is supported so the work-around was likely not necessary - * and quite possibly wrong. - */ - - /* DPIB/posbuf position mode: - * For Playback, Use DPIB register from HDA space which - * reflects the actual data transferred. - * For Capture, Use the position buffer for pointer, as DPIB - * is not accurate enough, its update may be completed - * earlier than the data written to DDR. - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - } else { - /* - * For capture stream, we need more workaround to fix the - * position incorrect issue: - * - * 1. Wait at least 20us before reading position buffer after - * the interrupt generated(IOC), to make sure position update - * happens on frame boundary i.e. 20.833uSec for 48KHz. - * 2. Perform a dummy Read to DPIB register to flush DMA - * position value. - * 3. Read the DMA Position from posbuf. Now the readback - * value should be >= period boundary. - */ - usleep_range(20, 21); - snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - pos = snd_hdac_stream_get_pos_posbuf(hstream); - } - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: - /* - * In case VC1 traffic is disabled this is the recommended option - */ - pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - break; - case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: - /* - * This is the recommended option when VC1 is enabled. - * While this isn't needed for SOF platforms it's added for - * consistency and debug. - */ - pos = snd_hdac_stream_get_pos_posbuf(hstream); - break; - default: - dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", - sof_hda_position_quirk); - pos = 0; - break; - } - - if (pos >= hstream->bufsize) - pos = 0; - + pos = hda_dsp_stream_get_position(hstream, substream->stream, true); found: pos = bytes_to_frames(substream->runtime, pos); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index daeb64c495e4..d95ae17e81cc 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -707,12 +707,13 @@ bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) } static void -hda_dsp_set_bytes_transferred(struct hdac_stream *hstream, u64 buffer_size) +hda_dsp_compr_bytes_transferred(struct hdac_stream *hstream, int direction) { + u64 buffer_size = hstream->bufsize; u64 prev_pos, pos, num_bytes; div64_u64_rem(hstream->curr_pos, buffer_size, &prev_pos); - pos = snd_hdac_stream_get_pos_posbuf(hstream); + pos = hda_dsp_stream_get_position(hstream, direction, false); if (pos < prev_pos) num_bytes = (buffer_size - prev_pos) + pos; @@ -748,8 +749,7 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) if (s->substream && sof_hda->no_ipc_position) { snd_sof_pcm_period_elapsed(s->substream); } else if (s->cstream) { - hda_dsp_set_bytes_transferred(s, - s->cstream->runtime->buffer_size); + hda_dsp_compr_bytes_transferred(s, s->cstream->direction); snd_compr_fragment_elapsed(s->cstream); } } @@ -1009,3 +1009,89 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) devm_kfree(sdev->dev, hda_stream); } } + +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep) +{ + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); + struct sof_intel_hda_stream *hda_stream = hstream_to_sof_hda_stream(hext_stream); + struct snd_sof_dev *sdev = hda_stream->sdev; + snd_pcm_uframes_t pos; + + switch (sof_hda_position_quirk) { + case SOF_HDA_POSITION_QUIRK_USE_SKYLAKE_LEGACY: + /* + * This legacy code, inherited from the Skylake driver, + * mixes DPIB registers and DPIB DDR updates and + * does not seem to follow any known hardware recommendations. + * It's not clear e.g. why there is a different flow + * for capture and playback, the only information that matters is + * what traffic class is used, and on all SOF-enabled platforms + * only VC0 is supported so the work-around was likely not necessary + * and quite possibly wrong. + */ + + /* DPIB/posbuf position mode: + * For Playback, Use DPIB register from HDA space which + * reflects the actual data transferred. + * For Capture, Use the position buffer for pointer, as DPIB + * is not accurate enough, its update may be completed + * earlier than the data written to DDR. + */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + } else { + /* + * For capture stream, we need more workaround to fix the + * position incorrect issue: + * + * 1. Wait at least 20us before reading position buffer after + * the interrupt generated(IOC), to make sure position update + * happens on frame boundary i.e. 20.833uSec for 48KHz. + * 2. Perform a dummy Read to DPIB register to flush DMA + * position value. + * 3. Read the DMA Position from posbuf. Now the readback + * value should be >= period boundary. + */ + if (can_sleep) + usleep_range(20, 21); + + snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + pos = snd_hdac_stream_get_pos_posbuf(hstream); + } + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_REGISTERS: + /* + * In case VC1 traffic is disabled this is the recommended option + */ + pos = snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, + AZX_REG_VS_SDXDPIB_XBASE + + (AZX_REG_VS_SDXDPIB_XINTERVAL * + hstream->index)); + break; + case SOF_HDA_POSITION_QUIRK_USE_DPIB_DDR_UPDATE: + /* + * This is the recommended option when VC1 is enabled. + * While this isn't needed for SOF platforms it's added for + * consistency and debug. + */ + pos = snd_hdac_stream_get_pos_posbuf(hstream); + break; + default: + dev_err_once(sdev->dev, "hda_position_quirk value %d not supported\n", + sof_hda_position_quirk); + pos = 0; + break; + } + + if (pos >= hstream->bufsize) + pos = 0; + + return pos; +} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 0f57ef5d9b8e..06476ffe96d7 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -565,6 +565,9 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); +snd_pcm_uframes_t hda_dsp_stream_get_position(struct hdac_stream *hstream, + int direction, bool can_sleep); + struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags); int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag); From a933084558c61cac8c902d2474b39444d87fba46 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 15:18:16 -0500 Subject: [PATCH 25/43] ASoC: SOF: pm: add explicit behavior for ACPI S1 and S2 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The existing code only deals with S0 and S3, let's start adding S1 and S2. No functional change. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616201818.130802-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 18eb327a57f0..239f39a5166a 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -335,8 +335,18 @@ int snd_sof_prepare(struct device *dev) return 0; #if defined(CONFIG_ACPI) - if (acpi_target_system_state() == ACPI_STATE_S0) + switch (acpi_target_system_state()) { + case ACPI_STATE_S0: sdev->system_suspend_target = SOF_SUSPEND_S0IX; + break; + case ACPI_STATE_S1: + case ACPI_STATE_S2: + case ACPI_STATE_S3: + sdev->system_suspend_target = SOF_SUSPEND_S3; + break; + default: + break; + } #endif return 0; From 9d2d462713384538477703e68577b05131c7d97d Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 15:18:17 -0500 Subject: [PATCH 26/43] ASoC: SOF: pm: add definitions for S4 and S5 states MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We currently don't have a means to differentiate between S3, S4 and S5. Add definitions so that we have select different code paths depending on the target state in follow-up patches. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616201818.130802-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/pm.c | 9 +++++++++ sound/soc/sof/sof-priv.h | 2 ++ 2 files changed, 11 insertions(+) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 239f39a5166a..df740be645e8 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -23,6 +23,9 @@ static u32 snd_sof_dsp_power_target(struct snd_sof_dev *sdev) u32 target_dsp_state; switch (sdev->system_suspend_target) { + case SOF_SUSPEND_S5: + case SOF_SUSPEND_S4: + /* DSP should be in D3 if the system is suspending to S3+ */ case SOF_SUSPEND_S3: /* DSP should be in D3 if the system is suspending to S3 */ target_dsp_state = SOF_DSP_PM_D3; @@ -344,6 +347,12 @@ int snd_sof_prepare(struct device *dev) case ACPI_STATE_S3: sdev->system_suspend_target = SOF_SUSPEND_S3; break; + case ACPI_STATE_S4: + sdev->system_suspend_target = SOF_SUSPEND_S4; + break; + case ACPI_STATE_S5: + sdev->system_suspend_target = SOF_SUSPEND_S5; + break; default: break; } diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 9d7f53ff9c70..f0f3d72c0da7 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -85,6 +85,8 @@ enum sof_system_suspend_state { SOF_SUSPEND_NONE = 0, SOF_SUSPEND_S0IX, SOF_SUSPEND_S3, + SOF_SUSPEND_S4, + SOF_SUSPEND_S5, }; enum sof_dfsentry_type { From 391153522d186f19a008d824bb3a05950351ce6c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 16 Jun 2022 15:18:18 -0500 Subject: [PATCH 27/43] ASoC: SOF: Intel: disable IMR boot when resuming from ACPI S4 and S5 states MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The IMR was assumed to be preserved when suspending to S4 and S5 states, but community reports invalidate that assumption, the hardware seems to be powered off and the IMR memory content cleared. Make sure regular boot with firmware download is used for S4 and S5. BugLink: https://github.com/thesofproject/sof/issues/5892 Fixes: 5fb5f51185126 ("ASoC: SOF: Intel: hda-loader: add IMR restore support") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220616201818.130802-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-loader.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index d3ec5996a9a3..145d483bd129 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -389,7 +389,8 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) struct snd_dma_buffer dmab; int ret, ret1, i; - if (hda->imrboot_supported && !sdev->first_boot) { + if (sdev->system_suspend_target < SOF_SUSPEND_S4 && + hda->imrboot_supported && !sdev->first_boot) { dev_dbg(sdev->dev, "IMR restore supported, booting from IMR directly\n"); hda->boot_iteration = 0; ret = hda_dsp_boot_imr(sdev); From a5450aba737dae3ee1a64b282e609d8375d6700c Mon Sep 17 00:00:00 2001 From: Judy Hsiao Date: Wed, 15 Jun 2022 04:56:43 +0000 Subject: [PATCH 28/43] ASoC: rockchip: i2s: switch BCLK to GPIO We discoverd that the state of BCLK on, LRCLK off and SD_MODE on may cause the speaker melting issue. Removing LRCLK while BCLK is present can cause unexpected output behavior including a large DC output voltage as described in the Max98357a datasheet. In order to: 1. prevent BCLK from turning on by other component. 2. keep BCLK and LRCLK being present at the same time This patch switches BCLK to GPIO func before LRCLK output, and configures BCLK func back during LRCLK is output. Without this fix, BCLK is turned on 11 ms earlier than LRCK by the da7219. With this fix, BCLK is turned on only 0.4 ms earlier than LRCK by the rockchip codec. Signed-off-by: Judy Hsiao Link: https://lore.kernel.org/r/20220615045643.3137287-1-judyhsiao@chromium.org Signed-off-by: Mark Brown --- sound/soc/rockchip/rockchip_i2s.c | 160 ++++++++++++++++++++++++------ 1 file changed, 129 insertions(+), 31 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 4ce5d2579387..99a128a666fb 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -54,8 +55,40 @@ struct rk_i2s_dev { const struct rk_i2s_pins *pins; unsigned int bclk_ratio; spinlock_t lock; /* tx/rx lock */ + struct pinctrl *pinctrl; + struct pinctrl_state *bclk_on; + struct pinctrl_state *bclk_off; }; +static int i2s_pinctrl_select_bclk_on(struct rk_i2s_dev *i2s) +{ + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_on)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_on); + + if (ret) + dev_err(i2s->dev, "bclk enable failed %d\n", ret); + + return ret; +} + +static int i2s_pinctrl_select_bclk_off(struct rk_i2s_dev *i2s) +{ + + int ret = 0; + + if (!IS_ERR(i2s->pinctrl) && !IS_ERR_OR_NULL(i2s->bclk_off)) + ret = pinctrl_select_state(i2s->pinctrl, + i2s->bclk_off); + + if (ret) + dev_err(i2s->dev, "bclk disable failed %d\n", ret); + + return ret; +} + static int i2s_runtime_suspend(struct device *dev) { struct rk_i2s_dev *i2s = dev_get_drvdata(dev); @@ -92,38 +125,49 @@ static inline struct rk_i2s_dev *to_info(struct snd_soc_dai *dai) return snd_soc_dai_get_drvdata(dai); } -static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->tx_start = true; } else { i2s->tx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, - I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, + I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + ret = regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + ret = regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); @@ -138,44 +182,57 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } -static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) +static int rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) { unsigned int val = 0; int retry = 10; + int ret = 0; spin_lock(&i2s->lock); if (on) { - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); + if (ret < 0) + goto end; - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_START | I2S_XFER_RXS_START); + if (ret < 0) + goto end; i2s->rx_start = true; } else { i2s->rx_start = false; - regmap_update_bits(i2s->regmap, I2S_DMACR, + ret = regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); + if (ret < 0) + goto end; if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, + ret = regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START | I2S_XFER_RXS_START, I2S_XFER_TXS_STOP | I2S_XFER_RXS_STOP); - + if (ret < 0) + goto end; udelay(150); - regmap_update_bits(i2s->regmap, I2S_CLR, + ret = regmap_update_bits(i2s->regmap, I2S_CLR, I2S_CLR_TXC | I2S_CLR_RXC, I2S_CLR_TXC | I2S_CLR_RXC); - + if (ret < 0) + goto end; regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ while (val) { regmap_read(i2s->regmap, I2S_CLR, &val); @@ -187,7 +244,12 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) } } } +end: spin_unlock(&i2s->lock); + if (ret < 0) + dev_err(i2s->dev, "lrclk update failed\n"); + + return ret; } static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, @@ -425,17 +487,26 @@ static int rockchip_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 1); + ret = rockchip_snd_rxctrl(i2s, 1); else - rockchip_snd_txctrl(i2s, 1); + ret = rockchip_snd_txctrl(i2s, 1); + /* Do not turn on bclk if lrclk open fails. */ + if (ret < 0) + return ret; + i2s_pinctrl_select_bclk_on(i2s); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - rockchip_snd_rxctrl(i2s, 0); - else - rockchip_snd_txctrl(i2s, 0); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!i2s->tx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_rxctrl(i2s, 0); + } else { + if (!i2s->rx_start) + i2s_pinctrl_select_bclk_off(i2s); + ret = rockchip_snd_txctrl(i2s, 0); + } break; default: ret = -EINVAL; @@ -736,6 +807,33 @@ static int rockchip_i2s_probe(struct platform_device *pdev) } i2s->bclk_ratio = 64; + i2s->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(i2s->pinctrl)) + dev_err(&pdev->dev, "failed to find i2s pinctrl\n"); + + i2s->bclk_on = pinctrl_lookup_state(i2s->pinctrl, + "bclk_on"); + if (IS_ERR_OR_NULL(i2s->bclk_on)) + dev_err(&pdev->dev, "failed to find i2s default state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk state\n"); + + i2s->bclk_off = pinctrl_lookup_state(i2s->pinctrl, + "bclk_off"); + if (IS_ERR_OR_NULL(i2s->bclk_off)) + dev_err(&pdev->dev, "failed to find i2s gpio state\n"); + else + dev_dbg(&pdev->dev, "find i2s bclk_off state\n"); + + i2s_pinctrl_select_bclk_off(i2s); + + i2s->playback_dma_data.addr = res->start + I2S_TXDR; + i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->playback_dma_data.maxburst = 4; + + i2s->capture_dma_data.addr = res->start + I2S_RXDR; + i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + i2s->capture_dma_data.maxburst = 4; dev_set_drvdata(&pdev->dev, i2s); From f2c2f31f00ce48d96dec28b9f8d70f73213ed4af Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 17 Jun 2022 14:02:30 -0700 Subject: [PATCH 29/43] MAINTAINERS: update ASoC Qualcomm maintainer email-id Update Banajit's email address from codeaurora.org to quicinc.com, as codeaurora.org is not in use anymore. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20220617210230.7685-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 440f3d7c93b9..171fa3160696 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -16247,7 +16247,7 @@ F: drivers/crypto/qat/ QCOM AUDIO (ASoC) DRIVERS M: Srinivas Kandagatla -M: Banajit Goswami +M: Banajit Goswami L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/codecs/lpass-va-macro.c From 9896c029f0df628c6cb108253d09b1d61f1d4a88 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jun 2022 11:20:38 +0100 Subject: [PATCH 30/43] ASoC: wm_adsp: Fix event for preloader The preloader controls on ADSP should return a value of 1 if the preloader value was changed, update to correct this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220621102041.1713504-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6d7fd88243aa..a7784ac15dde 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -997,7 +997,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, snd_soc_dapm_sync(dapm); } - return 0; + return 1; } EXPORT_SYMBOL_GPL(wm_adsp2_preloader_put); From 0bc0ae9a5938d512fd5d44f11c9c04892dcf4961 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jun 2022 11:20:39 +0100 Subject: [PATCH 31/43] ASoC: wm5110: Fix DRE control The DRE controls on wm5110 should return a value of 1 if the DRE state is actually changed, update to fix this. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220621102041.1713504-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4973ba1ed779..4ab7a672f8de 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -413,6 +413,7 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, unsigned int rnew = (!!ucontrol->value.integer.value[1]) << mc->rshift; unsigned int lold, rold; unsigned int lena, rena; + bool change = false; int ret; snd_soc_dapm_mutex_lock(dapm); @@ -440,8 +441,8 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, goto err; } - ret = regmap_update_bits(arizona->regmap, ARIZONA_DRE_ENABLE, - mask, lnew | rnew); + ret = regmap_update_bits_check(arizona->regmap, ARIZONA_DRE_ENABLE, + mask, lnew | rnew, &change); if (ret) { dev_err(arizona->dev, "Failed to set DRE: %d\n", ret); goto err; @@ -454,6 +455,9 @@ static int wm5110_put_dre(struct snd_kcontrol *kcontrol, if (!rnew && rold) wm5110_clear_pga_volume(arizona, mc->rshift); + if (change) + ret = 1; + err: snd_soc_dapm_mutex_unlock(dapm); From c6a5f22f9b4fd5f21414be690ce34046d9712f05 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jun 2022 11:20:40 +0100 Subject: [PATCH 32/43] ASoC: cs35l41: Correct some control names Various boolean controls on cs35l41 are missing the required "Switch" in the name, add these. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220621102041.1713504-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 3e68a07a3c8e..71ab2a5d1c55 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -333,7 +333,7 @@ static const struct snd_kcontrol_new cs35l41_aud_controls[] = { SOC_SINGLE("HW Noise Gate Enable", CS35L41_NG_CFG, 8, 63, 0), SOC_SINGLE("HW Noise Gate Delay", CS35L41_NG_CFG, 4, 7, 0), SOC_SINGLE("HW Noise Gate Threshold", CS35L41_NG_CFG, 0, 7, 0), - SOC_SINGLE("Aux Noise Gate CH1 Enable", + SOC_SINGLE("Aux Noise Gate CH1 Switch", CS35L41_MIXER_NGATE_CH1_CFG, 16, 1, 0), SOC_SINGLE("Aux Noise Gate CH1 Entry Delay", CS35L41_MIXER_NGATE_CH1_CFG, 8, 15, 0), @@ -341,15 +341,15 @@ static const struct snd_kcontrol_new cs35l41_aud_controls[] = { CS35L41_MIXER_NGATE_CH1_CFG, 0, 7, 0), SOC_SINGLE("Aux Noise Gate CH2 Entry Delay", CS35L41_MIXER_NGATE_CH2_CFG, 8, 15, 0), - SOC_SINGLE("Aux Noise Gate CH2 Enable", + SOC_SINGLE("Aux Noise Gate CH2 Switch", CS35L41_MIXER_NGATE_CH2_CFG, 16, 1, 0), SOC_SINGLE("Aux Noise Gate CH2 Threshold", CS35L41_MIXER_NGATE_CH2_CFG, 0, 7, 0), - SOC_SINGLE("SCLK Force", CS35L41_SP_FORMAT, CS35L41_SCLK_FRC_SHIFT, 1, 0), - SOC_SINGLE("LRCLK Force", CS35L41_SP_FORMAT, CS35L41_LRCLK_FRC_SHIFT, 1, 0), - SOC_SINGLE("Invert Class D", CS35L41_AMP_DIG_VOL_CTRL, + SOC_SINGLE("SCLK Force Switch", CS35L41_SP_FORMAT, CS35L41_SCLK_FRC_SHIFT, 1, 0), + SOC_SINGLE("LRCLK Force Switch", CS35L41_SP_FORMAT, CS35L41_LRCLK_FRC_SHIFT, 1, 0), + SOC_SINGLE("Invert Class D Switch", CS35L41_AMP_DIG_VOL_CTRL, CS35L41_AMP_INV_PCM_SHIFT, 1, 0), - SOC_SINGLE("Amp Gain ZC", CS35L41_AMP_GAIN_CTRL, + SOC_SINGLE("Amp Gain ZC Switch", CS35L41_AMP_GAIN_CTRL, CS35L41_AMP_GAIN_ZC_SHIFT, 1, 0), WM_ADSP2_PRELOAD_SWITCH("DSP1", 1), WM_ADSP_FW_CONTROL("DSP1", 0), From 1df793d479bef546569fc2e409ff8bb3f0fb8e99 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Tue, 21 Jun 2022 17:07:19 +0800 Subject: [PATCH 33/43] ASoC: rt711-sdca: fix kernel NULL pointer dereference when IO error The initial settings will be written before the codec probe function. But, the rt711->component doesn't be assigned yet. If IO error happened during initial settings operations, it will cause the kernel panic. This patch changed component->dev to slave->dev to fix this issue. Signed-off-by: Shuming Fan Link: https://lore.kernel.org/r/20220621090719.30558-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt711-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index dfe3c9299ebd..5ad53bbc8528 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -34,7 +34,7 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, ret = regmap_write(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to set private value: %06x <= %04x ret=%d\n", addr, value, ret); @@ -50,7 +50,7 @@ static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, ret = regmap_read(regmap, addr, value); if (ret < 0) - dev_err(rt711->component->dev, + dev_err(&rt711->slave->dev, "Failed to get private value: %06x => %04x ret=%d\n", addr, *value, ret); From 11d7a12f7f50baa5af9090b131c9b03af59503e7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 11:51:15 +0100 Subject: [PATCH 34/43] ASoC: dapm: Initialise kcontrol data for mux/demux controls DAPM keeps a copy of the current value of mux/demux controls, however this value is only initialised in the case of autodisable controls. This leads to false notification events when first modifying a DAPM kcontrol that has a non-zero default. Autodisable controls are left as they are, since they already initialise the value, and there would be more work required to support autodisable muxes where the first option isn't disabled and/or that isn't the default. Technically this issue could affect mixer/switch elements as well, although not on any of the devices I am currently running. There is also a little more work to do to address the issue there due to that side supporting stereo controls, so that has not been tackled in this patch. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623105120.1981154-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 869c76506b66..a8e842e02cdc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -62,6 +62,8 @@ struct snd_soc_dapm_widget * snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); +static unsigned int soc_dapm_read(struct snd_soc_dapm_context *dapm, int reg); + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 1, @@ -442,6 +444,9 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, snd_soc_dapm_add_path(widget->dapm, data->widget, widget, NULL, NULL); + } else if (e->reg != SND_SOC_NOPM) { + data->value = soc_dapm_read(widget->dapm, e->reg) & + (e->mask << e->shift_l); } break; default: From 46b0d050c8c7df6dfb2c376aaa149bf2cfc5ca3e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 11:51:16 +0100 Subject: [PATCH 35/43] ASoC: cs35l41: Add ASP TX3/4 source to register patch The mixer controls for ASP TX3/4 are set to values that are not included in their enumeration control. This will cause spurious event notifications when the controls are first changed, as the register value changes whilst the actual visible enumeration value does not. Use the register patch to set them to a known value, zero, which equates to zero fill, thereby avoiding the spurious notifications. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623105120.1981154-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-lib.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index 6d3070ea9e06..198cfe54a46f 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -37,8 +37,8 @@ static const struct reg_default cs35l41_reg[] = { { CS35L41_DAC_PCM1_SRC, 0x00000008 }, { CS35L41_ASP_TX1_SRC, 0x00000018 }, { CS35L41_ASP_TX2_SRC, 0x00000019 }, - { CS35L41_ASP_TX3_SRC, 0x00000020 }, - { CS35L41_ASP_TX4_SRC, 0x00000021 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, { CS35L41_DSP1_RX1_SRC, 0x00000008 }, { CS35L41_DSP1_RX2_SRC, 0x00000009 }, { CS35L41_DSP1_RX3_SRC, 0x00000018 }, @@ -644,6 +644,8 @@ static const struct reg_sequence cs35l41_reva0_errata_patch[] = { { CS35L41_DSP1_XM_ACCEL_PL0_PRI, 0x00000000 }, { CS35L41_PWR_CTRL2, 0x00000000 }, { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, }; static const struct reg_sequence cs35l41_revb0_errata_patch[] = { @@ -655,6 +657,8 @@ static const struct reg_sequence cs35l41_revb0_errata_patch[] = { { CS35L41_DSP1_XM_ACCEL_PL0_PRI, 0x00000000 }, { CS35L41_PWR_CTRL2, 0x00000000 }, { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, }; static const struct reg_sequence cs35l41_revb2_errata_patch[] = { @@ -666,6 +670,8 @@ static const struct reg_sequence cs35l41_revb2_errata_patch[] = { { CS35L41_DSP1_XM_ACCEL_PL0_PRI, 0x00000000 }, { CS35L41_PWR_CTRL2, 0x00000000 }, { CS35L41_AMP_GAIN_CTRL, 0x00000000 }, + { CS35L41_ASP_TX3_SRC, 0x00000000 }, + { CS35L41_ASP_TX4_SRC, 0x00000000 }, }; static const struct reg_sequence cs35l41_fs_errata_patch[] = { From 7f103af4a10f375b9b346b4d0b730f6a66b8c451 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 11:51:17 +0100 Subject: [PATCH 36/43] ASoC: cs47l15: Fix event generation for low power mux control cs47l15_in1_adc_put always returns zero regardless of if the control value was updated. This results in missing notifications to user-space of the control change. Update the handling to return 1 when the value is changed. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623105120.1981154-3-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs47l15.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index 391fd7da331f..1c7d52bef893 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -122,6 +122,9 @@ static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, snd_soc_kcontrol_component(kcontrol); struct cs47l15 *cs47l15 = snd_soc_component_get_drvdata(component); + if (!!ucontrol->value.integer.value[0] == cs47l15->in1_lp_mode) + return 0; + switch (ucontrol->value.integer.value[0]) { case 0: /* Set IN1 to normal mode */ @@ -150,7 +153,7 @@ static int cs47l15_in1_adc_put(struct snd_kcontrol *kcontrol, break; } - return 0; + return 1; } static const struct snd_kcontrol_new cs47l15_snd_controls[] = { From e3cabbef3db8269207a6b8808f510137669f8deb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 11:51:18 +0100 Subject: [PATCH 37/43] ASoC: madera: Fix event generation for OUT1 demux madera_out1_demux_put returns the value of snd_soc_dapm_mux_update_power, which returns a 1 if a path was found for the kcontrol. This is obviously different to the expected return a 1 if the control was updated value. This results in spurious notifications to user-space. Update the handling to only return a 1 when the value is changed. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623105120.1981154-4-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/madera.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 272041c6236a..8095a87117cf 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -618,7 +618,13 @@ int madera_out1_demux_put(struct snd_kcontrol *kcontrol, end: snd_soc_dapm_mutex_unlock(dapm); - return snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + ret = snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + if (ret < 0) { + dev_err(madera->dev, "Failed to update demux power state: %d\n", ret); + return ret; + } + + return change; } EXPORT_SYMBOL_GPL(madera_out1_demux_put); From 980555e95f7cabdc9c80a07107622b097ba23703 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jun 2022 11:51:19 +0100 Subject: [PATCH 38/43] ASoC: madera: Fix event generation for rate controls madera_adsp_rate_put always returns zero regardless of if the control value was updated. This results in missing notifications to user-space of the control change. Update the handling to return 1 when the value is changed. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20220623105120.1981154-5-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/madera.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c index 8095a87117cf..b9f19fbd2911 100644 --- a/sound/soc/codecs/madera.c +++ b/sound/soc/codecs/madera.c @@ -899,7 +899,7 @@ static int madera_adsp_rate_put(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; const int adsp_num = e->shift_l; const unsigned int item = ucontrol->value.enumerated.item[0]; - int ret; + int ret = 0; if (item >= e->items) return -EINVAL; @@ -916,10 +916,10 @@ static int madera_adsp_rate_put(struct snd_kcontrol *kcontrol, "Cannot change '%s' while in use by active audio paths\n", kcontrol->id.name); ret = -EBUSY; - } else { + } else if (priv->adsp_rate_cache[adsp_num] != e->values[item]) { /* Volatile register so defer until the codec is powered up */ priv->adsp_rate_cache[adsp_num] = e->values[item]; - ret = 0; + ret = 1; } mutex_unlock(&priv->rate_lock); From 6e2c9105e0b743c92a157389d40f00b81bdd09fe Mon Sep 17 00:00:00 2001 From: John Veness Date: Fri, 24 Jun 2022 15:07:57 +0100 Subject: [PATCH 39/43] ALSA: usb-audio: Add quirks for MacroSilicon MS2100/MS2106 devices Treat the claimed 96kHz 1ch in the descriptors as 48kHz 2ch, so that the audio stream doesn't sound mono. Also fix initial stream alignment, so that left and right channels are in the correct order. Signed-off-by: John Veness Link: https://lore.kernel.org/r/20220624140757.28758-1-john-linux@pelago.org.uk Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 48 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 3 +++ 2 files changed, 51 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4f56e1784932..853da162fd18 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3802,6 +3802,54 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_FLAG_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + USB_AUDIO_DEVICE(0x534d, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2109 based HDMI capture cards * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 12ce69b04f63..a7bcae0a2c75 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1478,6 +1478,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; @@ -1908,6 +1909,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x534d, 0x0021, /* MacroSilicon MS2100/MS2106 */ + QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ From 11bea26929a1a3a9dd1a287b60c2f471701bf706 Mon Sep 17 00:00:00 2001 From: Tim Crawford Date: Fri, 24 Jun 2022 08:41:09 -0600 Subject: [PATCH 40/43] ALSA: hda/realtek: Add quirk for Clevo L140PU Fixes headset detection on Clevo L140PU. Signed-off-by: Tim Crawford Cc: Link: https://lore.kernel.org/r/20220624144109.3957-1-tcrawford@system76.com 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 cee69fa7e246..007dd8b5e1f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9212,6 +9212,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), From 4fb7c24f69c48fdc02ea7858dbd5a60ff08bf7e5 Mon Sep 17 00:00:00 2001 From: Egor Vorontsov Date: Mon, 27 Jun 2022 13:00:34 +0300 Subject: [PATCH 41/43] ALSA: usb-audio: Add quirk for Fiero SC-01 Fiero SC-01 is a USB sound card with two mono inputs and a single stereo output. The inputs are composed into a single stereo stream. The device uses a vendor-provided driver on Windows and does not work at all without it. The driver mostly provides ASIO functionality, but also alters the way the sound card is queried for sample rates and clocks. ALSA queries those failing with an EPIPE (same as Windows 10 does). Presumably, the vendor-provided driver does not query it at all, simply matching by VID:PID. Thus, I consider this a buggy firmware and adhere to a set of fixed endpoint quirks instead. The soundcard has an internal clock. Implicit feedback mode is required for the playback. I have updated my device to v1.1.0 from a Windows 10 VM using a vendor- provided binary prior to the development, hoping for it to just begin working. The device provides no obvious way to downgrade the firmware, and regardless, there's no binary available for v1.0.0 anyway. Thus, I will be getting another unit to extend the patch with support for that. Expected to be a simple copy-paste of the existing one, though. There were no previous reports of that device in context of Linux anywhere. Other issues have been reported though, but that's out of the scope. Signed-off-by: Egor Vorontsov Link: https://lore.kernel.org/r/20220627100041.2861494-1-sdoregor@sdore.me Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 68 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 2 ++ 2 files changed, 70 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 853da162fd18..7067d314fecd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -4167,6 +4167,74 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Fiero SC-01 (firmware v1.1.0) + */ + USB_DEVICE(0x2b53, 0x0031), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Fiero", + .product_name = "SC-01", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .rate_min = 48000, + .rate_max = 96000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { 48000, 96000 }, + .clock = 0x29 + } + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .rate_min = 48000, + .rate_max = 96000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { 48000, 96000 }, + .clock = 0x29 + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a7bcae0a2c75..51138350f03c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1915,6 +1915,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), /* Vendor matches */ VENDOR_FLG(0x045e, /* MS Lifecam */ From 2307a0e1ca0b5c1337b37ac6302f96e017ebac3c Mon Sep 17 00:00:00 2001 From: Egor Vorontsov Date: Mon, 27 Jun 2022 13:00:35 +0300 Subject: [PATCH 42/43] ALSA: usb-audio: Add quirk for Fiero SC-01 (fw v1.0.0) The patch applies the same quirks used for SC-01 at firmware v1.1.0 to the ones running v1.0.0, with respect to hard-coded sample rates. I got two more units and successfully tested the patch series with both firmwares. The support is now complete (not accounting ASIO). Signed-off-by: Egor Vorontsov Link: https://lore.kernel.org/r/20220627100041.2861494-2-sdoregor@sdore.me Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 132 +++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 4 ++ 2 files changed, 136 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 7067d314fecd..f93201a830b5 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -4167,6 +4167,138 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Fiero SC-01 (firmware v1.0.0 @ 48 kHz) + */ + USB_DEVICE(0x2b53, 0x0023), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Fiero", + .product_name = "SC-01", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 }, + .clock = 0x29 + } + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 }, + .clock = 0x29 + } + }, + { + .ifnum = -1 + } + } + } +}, +{ + /* + * Fiero SC-01 (firmware v1.0.0 @ 96 kHz) + */ + USB_DEVICE(0x2b53, 0x0024), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Fiero", + .product_name = "SC-01", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_96000, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 96000 }, + .clock = 0x29 + } + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 2, + .fmt_bits = 24, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC | + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_96000, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 96000 }, + .clock = 0x29 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Fiero SC-01 (firmware v1.1.0) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 51138350f03c..968d90caeefa 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1915,6 +1915,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x2b53, 0x0023, /* Fiero SC-01 (firmware v1.0.0 @ 48 kHz) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), + DEVICE_FLG(0x2b53, 0x0024, /* Fiero SC-01 (firmware v1.0.0 @ 96 kHz) */ + QUIRK_FLAG_GENERIC_IMPLICIT_FB), DEVICE_FLG(0x2b53, 0x0031, /* Fiero SC-01 (firmware v1.1.0) */ QUIRK_FLAG_GENERIC_IMPLICIT_FB), From c5e58c4545a69677d078b4c813b5d10d3481be9c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Jul 2022 17:23:36 +0200 Subject: [PATCH 43/43] ALSA: cs46xx: Fix missing snd_card_free() call at probe error The previous cleanup with devres may lead to the incorrect release orders at the probe error handling due to the devres's nature. Until we register the card, snd_card_free() has to be called at first for releasing the stuff properly when the driver tries to manage and release the stuff via card->private_free(). This patch fixes it by calling snd_card_free() manually on the error from the probe callback. Fixes: 5bff69b3645d ("ALSA: cs46xx: Allocate resources with device-managed APIs") Cc: Reported-and-tested-by: Jan Engelhardt Link: https://lore.kernel.org/r/p2p1s96o-746-74p4-s95-61qo1p7782pn@vanv.qr Link: https://lore.kernel.org/r/20220705152336.350-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx.c | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index bd60308769ff..8634004a606b 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -74,36 +74,36 @@ static int snd_card_cs46xx_probe(struct pci_dev *pci, err = snd_cs46xx_create(card, pci, external_amp[dev], thinkpad[dev]); if (err < 0) - return err; + goto error; card->private_data = chip; chip->accept_valid = mmap_valid[dev]; err = snd_cs46xx_pcm(chip, 0); if (err < 0) - return err; + goto error; #ifdef CONFIG_SND_CS46XX_NEW_DSP err = snd_cs46xx_pcm_rear(chip, 1); if (err < 0) - return err; + goto error; err = snd_cs46xx_pcm_iec958(chip, 2); if (err < 0) - return err; + goto error; #endif err = snd_cs46xx_mixer(chip, 2); if (err < 0) - return err; + goto error; #ifdef CONFIG_SND_CS46XX_NEW_DSP if (chip->nr_ac97_codecs ==2) { err = snd_cs46xx_pcm_center_lfe(chip, 3); if (err < 0) - return err; + goto error; } #endif err = snd_cs46xx_midi(chip, 0); if (err < 0) - return err; + goto error; err = snd_cs46xx_start_dsp(chip); if (err < 0) - return err; + goto error; snd_cs46xx_gameport(chip); @@ -117,11 +117,15 @@ static int snd_card_cs46xx_probe(struct pci_dev *pci, err = snd_card_register(card); if (err < 0) - return err; + goto error; pci_set_drvdata(pci, card); dev++; return 0; + + error: + snd_card_free(card); + return err; } static struct pci_driver cs46xx_driver = {