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[PATCH v2 0/9] ASoC: codecs: qcom fix validation failures
by Srinivas Kandagatla 23 Feb '22

23 Feb '22

Thanks for pointing out to mixer kselftest and VALIDATION Kconfig. This patchset addresses some of the issues in Qualcomm codecs that are discovered with recent mixer kselftest and validations added to ASoC. Mostly these are under 1. accessing integer value type for enum controls, which is clearly an array out of bounds access. 2. Fix incorrect ranges. 3. Fix return values for put functions. Changes since v1: - removed tlv min max patches as "9bdd10d57a ("ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min")" fixed the issue in core. - added checks in wcd934x_rx_hph_mode_put before returning thanks, srini Srinivas Kandagatla (9): ASoC: codecs: rx-macro: fix accessing compander for aux ASoC: codecs: rx-macro: fix accessing array out of bounds for enum type ASoC: codecs: tx-macro: fix accessing array out of bounds for enum type ASoC: codecs: va-macro: fix accessing array out of bounds for enum type ASoC: codecs: wsa-macro: fix accessing array out of bounds for enum type ASoC: codecs: wc938x: fix accessing array out of bounds for enum type ASoC: codecs: wcd938x: fix kcontrol max values ASoC: codecs: wcd934x: fix kcontrol max values ASoC: codecs: wcd934x: fix return value of wcd934x_rx_hph_mode_put sound/soc/codecs/lpass-rx-macro.c | 12 ++++++++---- sound/soc/codecs/lpass-tx-macro.c | 8 ++++---- sound/soc/codecs/lpass-va-macro.c | 4 ++-- sound/soc/codecs/lpass-wsa-macro.c | 10 +++++----- sound/soc/codecs/wcd934x.c | 11 +++++++---- sound/soc/codecs/wcd938x.c | 10 +++++----- 6 files changed, 31 insertions(+), 24 deletions(-) -- 2.21.0

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[PATCH 00/16] ASoC: codecs: qcom fix validation failures
by Srinivas Kandagatla 23 Feb '22

23 Feb '22

Thanks for pointing out to mixer kselftest and VALIDATION Kconfig. This patchset addresses some of the issues in Qualcomm codecs that are discovered with recent mixer kselftest and validations added to ASoC. Mostly these are under 1. accessing integer value type for enum controls, which is clearly an array out of bounds access. 2. Fix incorrect ranges. 3. Fix incorrect min max for tlv controls. (I can see that other codecs also suffer same issue) 4. Fix return values for put functions. thanks, srini Srinivas Kandagatla (9): ASoC: codecs: rx-macro: fix accessing compander for aux ASoC: codecs: rx-macro: fix accessing array out of bounds for enum type ASoC: codecs: tx-macro: fix accessing array out of bounds for enum type ASoC: codecs: va-macro: fix accessing array out of bounds for enum type ASoC: codecs: wsa-macro: fix accessing array out of bounds for enum type ASoC: codecs: wc938x: fix accessing array out of bounds for enum type ASoC: codecs: wcd938x: fix kcontrol max values ASoC: codecs: wcd934x: fix kcontrol max values ASoC: codecs: wcd934x: fix return value of wcd934x_rx_hph_mode_put ASoC: codecs: rx-macro: fix tlv min max range ASoC: codecs: tx-macro: fix tlv min max range ASoC: codecs: va-macro: fix tlv min max range ASoC: codecs: wsa-macro: fix tlv min max range ASoC: codecs: wcd938x: fix tlv min max range ASoC: codecs: wcd9335: fix tlv min max range ASoC: codecs: msm8916-wcd-digital: fix tlv min max range sound/soc/codecs/lpass-rx-macro.c | 40 ++++++++------- sound/soc/codecs/lpass-tx-macro.c | 24 ++++----- sound/soc/codecs/lpass-va-macro.c | 12 ++--- sound/soc/codecs/lpass-wsa-macro.c | 14 +++--- sound/soc/codecs/msm8916-wcd-digital.c | 14 +++--- sound/soc/codecs/wcd9335.c | 36 ++++++------- sound/soc/codecs/wcd934x.c | 70 +++++++++++++------------- sound/soc/codecs/wcd938x.c | 10 ++-- 8 files changed, 112 insertions(+), 108 deletions(-) -- 2.21.0

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[PATCH AUTOSEL 4.19 05/13] ASoC: rt5682: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4c33de0673ced9c7c37b3bbd9bfe0fda72340b2a ] The current rt5682_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-3-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7a78bb00f874d..5979165ac37cf 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1039,11 +1039,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682->calibrate_mutex); -- 2.34.1

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[PATCH AUTOSEL 4.19 04/13] ASoC: rt5668: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit a6d78661dc903d90a327892bbc34268f3a5f4b9c ] The current rt5668_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-2-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5668.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 3c19d03f2446b..a78503f24aa8a 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1025,11 +1025,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); -- 2.34.1

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[PATCH AUTOSEL 5.4 05/13] ASoC: rt5682: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4c33de0673ced9c7c37b3bbd9bfe0fda72340b2a ] The current rt5682_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-3-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 05e883a65d7a7..a8cf4c7451304 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1052,11 +1052,13 @@ static void rt5682_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682->calibrate_mutex); -- 2.34.1

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[PATCH AUTOSEL 5.4 04/13] ASoC: rt5668: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit a6d78661dc903d90a327892bbc34268f3a5f4b9c ] The current rt5668_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-2-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5668.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 5716cede99cb4..acc2b34ca334a 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); -- 2.34.1

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[PATCH AUTOSEL 5.10 05/18] ASoC: rt5682: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4c33de0673ced9c7c37b3bbd9bfe0fda72340b2a ] The current rt5682_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-3-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index aaef76cc151fa..113ed00ddf1e5 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1081,11 +1081,13 @@ void rt5682_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682->calibrate_mutex); -- 2.34.1

1 0

[PATCH AUTOSEL 5.10 04/18] ASoC: rt5668: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit a6d78661dc903d90a327892bbc34268f3a5f4b9c ] The current rt5668_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-2-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5668.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index bc69adc9c8b70..e625df57c69e5 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); -- 2.34.1

1 0

[PATCH AUTOSEL 5.15 05/28] ASoC: rt5682: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4c33de0673ced9c7c37b3bbd9bfe0fda72340b2a ] The current rt5682_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-3-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 757d1362b5f48..6ad3159eceb98 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1090,11 +1090,13 @@ void rt5682_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682->calibrate_mutex); -- 2.34.1

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[PATCH AUTOSEL 5.15 04/28] ASoC: rt5668: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit a6d78661dc903d90a327892bbc34268f3a5f4b9c ] The current rt5668_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-2-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5668.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 6ab1a8bc37352..1186ceb5a978e 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); -- 2.34.1

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[PATCH AUTOSEL 5.16 07/30] ASoC: rt5682: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 4c33de0673ced9c7c37b3bbd9bfe0fda72340b2a ] The current rt5682_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-3-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7e6e2f26accd0..e3643ae6de66d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1092,11 +1092,13 @@ void rt5682_jack_detect_handler(struct work_struct *work) struct snd_soc_dapm_context *dapm; int val, btn_type; - while (!rt5682->component) - usleep_range(10000, 15000); - - while (!rt5682->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682->component || !rt5682->component->card || + !rt5682->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(15)); + return; + } dapm = snd_soc_component_get_dapm(rt5682->component); -- 2.34.1

1 0

[PATCH AUTOSEL 5.16 06/30] ASoC: rt5668: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit a6d78661dc903d90a327892bbc34268f3a5f4b9c ] The current rt5668_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-2-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5668.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index fb09715bf9328..5b12cbf2ba215 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1022,11 +1022,13 @@ static void rt5668_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5668_priv, jack_detect_work.work); int val, btn_type; - while (!rt5668->component) - usleep_range(10000, 15000); - - while (!rt5668->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5668->component || !rt5668->component->card || + !rt5668->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5668->calibrate_mutex); -- 2.34.1

1 0

[PATCH AUTOSEL 5.16 05/30] ASoC: rt5682s: do not block workqueue if card is unbound
by Sasha Levin 23 Feb '22

23 Feb '22

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit d7b530fdc45e75a54914a194c4becd9672a4e24f ] The current rt5682s_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> Link: https://lore.kernel.org/r/20220207153000.3452802-1-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682s.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d49a4f68566d2..d79b548d23fa4 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -824,11 +824,13 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682s_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682s->component) - usleep_range(10000, 15000); - - while (!rt5682s->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682s->component || !rt5682s->component->card || + !rt5682s->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682s->jdet_mutex); mutex_lock(&rt5682s->calibrate_mutex); -- 2.34.1

1 0

[PATCH 00/10] ASoC: codec: add pm runtime support for Qualcomm codecs
by Srinivas Kandagatla 22 Feb '22

22 Feb '22

This patchset adds support for runtime pm on tx/rx/wsa/wcd lpass macro, wsa881x and wcd938x codecs that are wired up on SoundWire bus. During pm testing it was also found that soundwire clks enabled by lpass macros are not enabling all the required clocks correctly, so last 3 patches corrects them. Tested this on SM8250 MTP along SoundWire In band Headset Button wakeup interrupts. Srinivas Kandagatla (10): ASoC: codecs: va-macro: add runtime pm support ASoC: codecs: wsa-macro: add runtime pm support ASoC: codecs: rx-macro: add runtime pm support ASoC: codecs: tx-macro: add runtime pm support ASoC: codecs: wsa881x: add runtime pm support ASoC: codecs: wcd938x: add simple clk stop support ASoC: codecs: wcd-mbhc: add runtime pm support ASoC: codecs: wsa-macro: setup soundwire clks correctly ASoC: codecs: tx-macro: setup soundwire clks correctly ASoC: codecs: rx-macro: setup soundwire clks correctly sound/soc/codecs/lpass-rx-macro.c | 49 +++++++++++++++++++++++++-- sound/soc/codecs/lpass-tx-macro.c | 45 ++++++++++++++++++++++++- sound/soc/codecs/lpass-va-macro.c | 36 ++++++++++++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 43 +++++++++++++++++++++++- sound/soc/codecs/wcd-mbhc-v2.c | 26 ++++++++++++++ sound/soc/codecs/wcd938x-sdw.c | 1 + sound/soc/codecs/wsa881x.c | 54 ++++++++++++++++++++++++++++++ 7 files changed, 250 insertions(+), 4 deletions(-) -- 2.21.0

3 17

[PATCH v8 0/7] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 22 Feb '22

22 Feb '22

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V7: -- Update optional clock voting with conditional check. -- Add const to lpi_pinctrl_variant_data structure. -- Update required headers and remove redundant. -- Change EXPORT_SYMBOL to EXPORT_SYMBOL_GPL -- Fix typo errors. Changes Since V6: -- Update conditional clock voting to optional clock voting. -- Update Kconfig depends on field with select. -- Fix typo errors. Changes Since V5: -- Create new patch by updating macro name to lpi specific. -- Create new patch by updating lpi pin group structure with core group_desc structure. -- Fix typo errors. -- Sort macros in the make file and configuration file. Changes Since V4: -- Update commit message and description of the chip specific extraction patch. -- Sort macros in kconfig and makefile. -- Update optional clock voting to conditional clock voting. -- Fix typo errors. -- Move to quicinc domain email id's. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (7): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Update macro name to LPI specific pinctrl: qcom: Update lpi pin group structure pinctrl: qcom: Extract chip specific LPASS LPI code pinctrl: qcom: Add SC7280 lpass pin configuration pinctrl: qcom: Update clock voting as optional Tested this on SM8250 MTP with WSA and WCD codecs. Tested-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 133 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 133 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 255 ++------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 86 +++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 169 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 166 ++++++++++++++ 9 files changed, 705 insertions(+), 370 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- 2.7.4

2 8

Asking for information about ALSA's kselftest
by Giacomo Guiduzzi 22 Feb '22

22 Feb '22

[RESENDING BECAUSE BOUNCED BY SPAMASSASSIN] Greetings Mr. Brown and users of the alsa-devel mailing list, I’m a student at the University of Modena and Reggio Emilia. I am working on the Kernel Hacking course’s exam, a course kept by Paolo Valente which introduces students to the world of the Linux kernel and its magics. I was taking a look at the kernel’s kselftests from Linus Torvald’s repository to find some test to fix for the exam and I stumbled upon the ALSA’s selftest, which I’ve passed the last few days on, debugging and exploring. During the execution the test goes as it should except for two or three tests. Until now I’ve tried understanding test 71, which sometimes succeeds and sometimes fails, and test 72. I have no previous knowledge of how ALSA works and I’m completely new to the kselftests and the kernel in general. Test 71 gives the following output: # Surround Playback Volume.0 expected 0 but read 1, is_volatile 0 # Surround Playback Volume.1 expected 1 but read 0, is_volatile 0 not ok 71 write_default.0.22 I wanted to ask you if it is normal or not that this test fails on my hardware, and if not, what I could do to debug the problem and give you useful information to help understand what the issue is. I am currently testing on an Ubuntu Desktop VirtualBox VM on kernel 5.17.0-rc3. The audio hardware (as the output of the lshw command, multimedia section) is: description: Multimedia audio controller product: 82801AA AC'97 Audio Controller vendor: Intel Corporation physical id: 5 bus info: pci@0000:00:05.0 version: 01 width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=snd_intel8x0 latency=64 resources: irq:21 ioport:d100(size=256) ioport:d200(size=64) From what I have seen until now it doesn’t look like a bug from the test nor an issue from the kernel, so I’m not really sure where to look at: I have compiled alsalib’s source code from the Git repo with debug symbols and I’ve followed the execution of the test until I reached a call to the ioctl syscall, which doesn’t return an error code so it doesn’t look like the problem relies in the kernel; I’m no expert though, I’ll let you confirm this or not. I am at your complete disposal to give you any further useful information as I can’t do much more from my side alone. Thank you in advance, looking forward to your kind response. Best Regards, Giacomo Guiduzzi

2 1

[PATCH] ASoC: samsung: i2s: check the return value of kstrdup()
by xkernel.wang＠foxmail.com 22 Feb '22

22 Feb '22

From: Xiaoke Wang <xkernel.wang(a)foxmail.com> kstrdup() is a memory allocation function which can return NULL when some internal memory errors happen. It is better to check the return value of it to catch the error in time. Signed-off-by: Xiaoke Wang <xkernel.wang(a)foxmail.com> --- sound/soc/samsung/i2s.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 309badc..70c8271 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1349,6 +1349,10 @@ static int i2s_create_secondary_device(struct samsung_i2s_priv *priv) return -ENOMEM; pdev_sec->driver_override = kstrdup("samsung-i2s", GFP_KERNEL); + if (!pdev_sec->driver_override) { + platform_device_put(pdev_sec); + return -ENOMEM; + } ret = platform_device_add(pdev_sec); if (ret < 0) { --

2 1

[PATCH 0/3] ASoC: amd: acp: Add new machines and minor tweaks
by Ajit Kumar Pandey 22 Feb '22

22 Feb '22

This patch series will add sound card support for newer machines and done some minor tweaks related to legacy machine driver. Ajit Kumar Pandey (3): ASoC: amd: acp: Change card name for Guybrush Machine ASoC: amd: acp-legacy: Add legacy card support for new machines ASoC: amd: acp: Add DMIC machine driver ops sound/soc/amd/acp/acp-legacy-mach.c | 34 +++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach-common.c | 27 +++++++++++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 19 +++++++++++++++- 3 files changed, 77 insertions(+), 3 deletions(-) -- 2.25.1

3 6

jack: implement pcm operation sw_params
by GitHub pull_request - opened 21 Feb '22

21 Feb '22

alsa-project/alsa-plugins pull request #43 was opened from aditpape: to reflect an updated avail_min value during runtime Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-plugins/pull/43 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/43.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ASoC: mediatek: mt8195: enable apll tuner
by Trevor Wu 21 Feb '22

21 Feb '22

Normally, the clock source of audio module is either 26M or APLL1/APLL2, but APLL1/APLL2 are not the multiple of 26M. In the patch, APLL1 and APLL2 tuners are enabled to handle sample rate mismatch when the data path crosses two different clock domains. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 282 ++++++++++++++++++++- sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 11 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3 - 3 files changed, 292 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c index c2543f4cffb7..efd5cc364a35 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c @@ -40,6 +40,8 @@ static const char *aud_clks[MT8195_CLK_NUM] = { [MT8195_CLK_SCP_ADSP_AUDIODSP] = "scp_adsp_audiodsp", /* afe clock gate */ [MT8195_CLK_AUD_AFE] = "aud_afe", + [MT8195_CLK_AUD_APLL1_TUNER] = "aud_apll1_tuner", + [MT8195_CLK_AUD_APLL2_TUNER] = "aud_apll2_tuner", [MT8195_CLK_AUD_APLL] = "aud_apll", [MT8195_CLK_AUD_APLL2] = "aud_apll2", [MT8195_CLK_AUD_DAC] = "aud_dac", @@ -77,6 +79,268 @@ static const char *aud_clks[MT8195_CLK_NUM] = { [MT8195_CLK_AUD_MEMIF_DL11] = "aud_memif_dl11", }; +struct mt8195_afe_tuner_cfg { + unsigned int id; + int apll_div_reg; + unsigned int apll_div_shift; + unsigned int apll_div_maskbit; + unsigned int apll_div_default; + int ref_ck_sel_reg; + unsigned int ref_ck_sel_shift; + unsigned int ref_ck_sel_maskbit; + unsigned int ref_ck_sel_default; + int tuner_en_reg; + unsigned int tuner_en_shift; + unsigned int tuner_en_maskbit; + int upper_bound_reg; + unsigned int upper_bound_shift; + unsigned int upper_bound_maskbit; + unsigned int upper_bound_default; + spinlock_t ctrl_lock; /* lock for apll tuner ctrl*/ + int ref_cnt; +}; + +static struct mt8195_afe_tuner_cfg mt8195_afe_tuner_cfgs[MT8195_AUD_PLL_NUM] = { + [MT8195_AUD_PLL1] = { + .id = MT8195_AUD_PLL1, + .apll_div_reg = AFE_APLL_TUNER_CFG, + .apll_div_shift = 4, + .apll_div_maskbit = 0xf, + .apll_div_default = 0x7, + .ref_ck_sel_reg = AFE_APLL_TUNER_CFG, + .ref_ck_sel_shift = 1, + .ref_ck_sel_maskbit = 0x3, + .ref_ck_sel_default = 0x2, + .tuner_en_reg = AFE_APLL_TUNER_CFG, + .tuner_en_shift = 0, + .tuner_en_maskbit = 0x1, + .upper_bound_reg = AFE_APLL_TUNER_CFG, + .upper_bound_shift = 8, + .upper_bound_maskbit = 0xff, + .upper_bound_default = 0x2, + }, + [MT8195_AUD_PLL2] = { + .id = MT8195_AUD_PLL2, + .apll_div_reg = AFE_APLL_TUNER_CFG1, + .apll_div_shift = 4, + .apll_div_maskbit = 0xf, + .apll_div_default = 0x7, + .ref_ck_sel_reg = AFE_APLL_TUNER_CFG1, + .ref_ck_sel_shift = 1, + .ref_ck_sel_maskbit = 0x3, + .ref_ck_sel_default = 0x1, + .tuner_en_reg = AFE_APLL_TUNER_CFG1, + .tuner_en_shift = 0, + .tuner_en_maskbit = 0x1, + .upper_bound_reg = AFE_APLL_TUNER_CFG1, + .upper_bound_shift = 8, + .upper_bound_maskbit = 0xff, + .upper_bound_default = 0x2, + }, + [MT8195_AUD_PLL3] = { + .id = MT8195_AUD_PLL3, + .apll_div_reg = AFE_EARC_APLL_TUNER_CFG, + .apll_div_shift = 4, + .apll_div_maskbit = 0x3f, + .apll_div_default = 0x3, + .ref_ck_sel_reg = AFE_EARC_APLL_TUNER_CFG, + .ref_ck_sel_shift = 24, + .ref_ck_sel_maskbit = 0x3, + .ref_ck_sel_default = 0x0, + .tuner_en_reg = AFE_EARC_APLL_TUNER_CFG, + .tuner_en_shift = 0, + .tuner_en_maskbit = 0x1, + .upper_bound_reg = AFE_EARC_APLL_TUNER_CFG, + .upper_bound_shift = 12, + .upper_bound_maskbit = 0xff, + .upper_bound_default = 0x4, + }, + [MT8195_AUD_PLL4] = { + .id = MT8195_AUD_PLL4, + .apll_div_reg = AFE_SPDIFIN_APLL_TUNER_CFG, + .apll_div_shift = 4, + .apll_div_maskbit = 0x3f, + .apll_div_default = 0x7, + .ref_ck_sel_reg = AFE_SPDIFIN_APLL_TUNER_CFG1, + .ref_ck_sel_shift = 8, + .ref_ck_sel_maskbit = 0x1, + .ref_ck_sel_default = 0, + .tuner_en_reg = AFE_SPDIFIN_APLL_TUNER_CFG, + .tuner_en_shift = 0, + .tuner_en_maskbit = 0x1, + .upper_bound_reg = AFE_SPDIFIN_APLL_TUNER_CFG, + .upper_bound_shift = 12, + .upper_bound_maskbit = 0xff, + .upper_bound_default = 0x4, + }, + [MT8195_AUD_PLL5] = { + .id = MT8195_AUD_PLL5, + .apll_div_reg = AFE_LINEIN_APLL_TUNER_CFG, + .apll_div_shift = 4, + .apll_div_maskbit = 0x3f, + .apll_div_default = 0x3, + .ref_ck_sel_reg = AFE_LINEIN_APLL_TUNER_CFG, + .ref_ck_sel_shift = 24, + .ref_ck_sel_maskbit = 0x1, + .ref_ck_sel_default = 0, + .tuner_en_reg = AFE_LINEIN_APLL_TUNER_CFG, + .tuner_en_shift = 0, + .tuner_en_maskbit = 0x1, + .upper_bound_reg = AFE_LINEIN_APLL_TUNER_CFG, + .upper_bound_shift = 12, + .upper_bound_maskbit = 0xff, + .upper_bound_default = 0x4, + }, +}; + +static struct mt8195_afe_tuner_cfg *mt8195_afe_found_apll_tuner(unsigned int id) +{ + if (id >= MT8195_AUD_PLL_NUM) + return NULL; + + return &mt8195_afe_tuner_cfgs[id]; +} + +static int mt8195_afe_init_apll_tuner(unsigned int id) +{ + struct mt8195_afe_tuner_cfg *cfg = mt8195_afe_found_apll_tuner(id); + + if (!cfg) + return -EINVAL; + + cfg->ref_cnt = 0; + spin_lock_init(&cfg->ctrl_lock); + + return 0; +} + +static int mt8195_afe_setup_apll_tuner(struct mtk_base_afe *afe, + unsigned int id) +{ + const struct mt8195_afe_tuner_cfg *cfg = mt8195_afe_found_apll_tuner(id); + + if (!cfg) + return -EINVAL; + + regmap_update_bits(afe->regmap, cfg->apll_div_reg, + cfg->apll_div_maskbit << cfg->apll_div_shift, + cfg->apll_div_default << cfg->apll_div_shift); + + regmap_update_bits(afe->regmap, cfg->ref_ck_sel_reg, + cfg->ref_ck_sel_maskbit << cfg->ref_ck_sel_shift, + cfg->ref_ck_sel_default << cfg->ref_ck_sel_shift); + + regmap_update_bits(afe->regmap, cfg->upper_bound_reg, + cfg->upper_bound_maskbit << cfg->upper_bound_shift, + cfg->upper_bound_default << cfg->upper_bound_shift); + + return 0; +} + +static int mt8195_afe_enable_tuner_clk(struct mtk_base_afe *afe, + unsigned int id) +{ + struct mt8195_afe_private *afe_priv = afe->platform_priv; + + switch (id) { + case MT8195_AUD_PLL1: + mt8195_afe_enable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL]); + mt8195_afe_enable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL1_TUNER]); + break; + case MT8195_AUD_PLL2: + mt8195_afe_enable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL2]); + mt8195_afe_enable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL2_TUNER]); + break; + default: + break; + } + + return 0; +} + +static int mt8195_afe_disable_tuner_clk(struct mtk_base_afe *afe, + unsigned int id) +{ + struct mt8195_afe_private *afe_priv = afe->platform_priv; + + switch (id) { + case MT8195_AUD_PLL1: + mt8195_afe_disable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL1_TUNER]); + mt8195_afe_disable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL]); + break; + case MT8195_AUD_PLL2: + mt8195_afe_disable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL2_TUNER]); + mt8195_afe_disable_clk(afe, afe_priv->clk[MT8195_CLK_AUD_APLL2]); + break; + default: + break; + } + + return 0; +} + +static int mt8195_afe_enable_apll_tuner(struct mtk_base_afe *afe, + unsigned int id) +{ + struct mt8195_afe_tuner_cfg *cfg = mt8195_afe_found_apll_tuner(id); + unsigned long flags; + int ret = 0; + + if (!cfg) + return -EINVAL; + + ret = mt8195_afe_setup_apll_tuner(afe, id); + if (ret) + return ret; + + ret = mt8195_afe_enable_tuner_clk(afe, id); + if (ret) + return ret; + + spin_lock_irqsave(&cfg->ctrl_lock, flags); + + cfg->ref_cnt++; + if (cfg->ref_cnt == 1) + regmap_update_bits(afe->regmap, + cfg->tuner_en_reg, + cfg->tuner_en_maskbit << cfg->tuner_en_shift, + 1 << cfg->tuner_en_shift); + + spin_unlock_irqrestore(&cfg->ctrl_lock, flags); + + return ret; +} + +static int mt8195_afe_disable_apll_tuner(struct mtk_base_afe *afe, + unsigned int id) +{ + struct mt8195_afe_tuner_cfg *cfg = mt8195_afe_found_apll_tuner(id); + unsigned long flags; + int ret = 0; + + if (!cfg) + return -EINVAL; + + spin_lock_irqsave(&cfg->ctrl_lock, flags); + + cfg->ref_cnt--; + if (cfg->ref_cnt == 0) + regmap_update_bits(afe->regmap, + cfg->tuner_en_reg, + cfg->tuner_en_maskbit << cfg->tuner_en_shift, + 0 << cfg->tuner_en_shift); + else if (cfg->ref_cnt < 0) + cfg->ref_cnt = 0; + + spin_unlock_irqrestore(&cfg->ctrl_lock, flags); + + ret = mt8195_afe_disable_tuner_clk(afe, id); + if (ret) + return ret; + + return ret; +} + int mt8195_afe_get_mclk_source_clk_id(int sel) { switch (sel) { @@ -113,7 +377,7 @@ int mt8195_afe_get_default_mclk_source_by_rate(int rate) int mt8195_afe_init_clock(struct mtk_base_afe *afe) { struct mt8195_afe_private *afe_priv = afe->platform_priv; - int i; + int i, ret; mt8195_audsys_clk_register(afe); @@ -133,6 +397,16 @@ int mt8195_afe_init_clock(struct mtk_base_afe *afe) } } + /* initial tuner */ + for (i = 0; i < MT8195_AUD_PLL_NUM; i++) { + ret = mt8195_afe_init_apll_tuner(i); + if (ret) { + dev_dbg(afe->dev, "%s(), init apll_tuner%d failed", + __func__, (i + 1)); + return -EINVAL; + } + } + return 0; } @@ -428,11 +702,17 @@ int mt8195_afe_enable_main_clock(struct mtk_base_afe *afe) mt8195_afe_enable_afe_on(afe); + mt8195_afe_enable_apll_tuner(afe, MT8195_AUD_PLL1); + mt8195_afe_enable_apll_tuner(afe, MT8195_AUD_PLL2); + return 0; } int mt8195_afe_disable_main_clock(struct mtk_base_afe *afe) { + mt8195_afe_disable_apll_tuner(afe, MT8195_AUD_PLL2); + mt8195_afe_disable_apll_tuner(afe, MT8195_AUD_PLL1); + mt8195_afe_disable_afe_on(afe); mt8195_afe_disable_timing_sys(afe); diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-clk.h b/sound/soc/mediatek/mt8195/mt8195-afe-clk.h index f8e6eeb29a89..40663e31becd 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-clk.h +++ b/sound/soc/mediatek/mt8195/mt8195-afe-clk.h @@ -35,6 +35,8 @@ enum { MT8195_CLK_INFRA_AO_AUDIO_26M_B, MT8195_CLK_SCP_ADSP_AUDIODSP, MT8195_CLK_AUD_AFE, + MT8195_CLK_AUD_APLL1_TUNER, + MT8195_CLK_AUD_APLL2_TUNER, MT8195_CLK_AUD_APLL, MT8195_CLK_AUD_APLL2, MT8195_CLK_AUD_DAC, @@ -84,6 +86,15 @@ enum { MT8195_MCK_SEL_NUM, }; +enum { + MT8195_AUD_PLL1, + MT8195_AUD_PLL2, + MT8195_AUD_PLL3, + MT8195_AUD_PLL4, + MT8195_AUD_PLL5, + MT8195_AUD_PLL_NUM, +}; + struct mtk_base_afe; int mt8195_afe_get_mclk_source_clk_id(int sel); diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index e425f868476a..b77c2ba5a629 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -2583,8 +2583,6 @@ static bool mt8195_is_volatile_reg(struct device *dev, unsigned int reg) case AFE_IRQ3_CON_MON: case AFE_IRQ_MCU_MON2: case ADSP_IRQ_STATUS: - case AFE_APLL_TUNER_CFG: - case AFE_APLL_TUNER_CFG1: case AUDIO_TOP_STA0: case AUDIO_TOP_STA1: case AFE_GAIN1_CUR: @@ -2623,7 +2621,6 @@ static bool mt8195_is_volatile_reg(struct device *dev, unsigned int reg) case SPDIFIN_USERCODE10: case SPDIFIN_USERCODE11: case SPDIFIN_USERCODE12: - case AFE_SPDIFIN_APLL_TUNER_CFG: case AFE_LINEIN_APLL_TUNER_MON: case AFE_EARC_APLL_TUNER_MON: case AFE_CM0_MON: -- 2.18.0

2 1

[PATCH v2] ASoC: SOF: Intel: Add topology overwrite for Felwinter
by Brent Lu 21 Feb '22

21 Feb '22

From: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> The Felwinter uses four max98360a amplifiers on corresponding CH0~CH3. There are four amps on the board connecting to headphone to SSP0 port, amp to SSP1,and the DAI format would be DSP_A,8-slots, 32 bit slot-width. CH0: L(Woofer), CH1:R(Woofer), CH2:L(Tweeter), CH3:R(Tweeter) Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Reviewed-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 20c6ca37dbc4..61f2afd54c3e 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -67,6 +67,14 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO_AMP-MAX98360_ALC5682VS_I2S_2WAY"), + }, + .driver_data = "sof-adl-max98360a-rt5682-2way.tplg", + }, {} }; -- 2.25.1

2 1

[PATCH] ASoC: SOF: Intel: Add topology overwrite for Felwinter
by Brent Lu 21 Feb '22

21 Feb '22

From: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> The Felwinter uses four max98360a amplifiers on corresponding CH0~CH3. There are four amps on the board connecting to headphone to SSP0 port, amp to SSP1,and the DAI format would be DSP_A,8-slots, 32 bit slot-width. CH0: L(Woofer), CH1:R(Woofer), CH2:L(Tweeter), CH3:R(Tweeter) Signed-off-by: Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/sof/sof-pci-dev.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 20c6ca37dbc4..61f2afd54c3e 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -67,6 +67,14 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO_AMP-MAX98360_ALC5682VS_I2S_2WAY"), + }, + .driver_data = "sof-adl-max98360a-rt5682-2way.tplg", + }, {} }; -- 2.25.1

4 4

Codec without controls in device tree
by Sascha Hauer 21 Feb '22

21 Feb '22

Hi, I have several i.MX8 boards here which have codecs connected to them which are really just plane D/A converters without any control interface like a pcm1754 or a pcm4104 in hardware mode. Currently we use the simple-audio-card OF driver with custom codec code in sound/soc/codecs/. The codec does nothing except specifying desired rates, channels and formats. Is this the way to go or is there some device tree solution to get rid of the codec? Sascha -- Pengutronix e.K. | | Steuerwalder Str. 21 | http://www.pengutronix.de/ | 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

2 1

Asking for information about ALSA's kselftest
by Giacomo Guiduzzi 21 Feb '22

21 Feb '22

Good morning Mr. Broonie and users of the alsa-devel mailing list, I’m a student at the University of Modena and Reggio Emilia. I am working at the Kernel Hacking course’s exam, a course kept by Paolo Valente which introduces students to the world of the Linux kernel and its magic. I was taking a look at the kernel’s kselftests from Linus Torvald’s repository to find some test to fix for the exam and I stumbled upon the ALSA’s selftest, which I’ve passed the last few days on, debugging and exploring. During the execution the test goes as it should except for two or three tests. Until now I’ve tried understanding test 71, which sometimes succeeds and sometimes fails, and test 72. I have no previous knowledge of how ALSA works as I’m completely new to the kselftests and the kernel in general. Test 71 gives the following output: # Surround Playback Volume.0 expected 0 but read 1, is_volatile 0 # Surround Playback Volume.1 expected 1 but read 0, is_volatile 0 not ok 71 write_default.0.22 I wanted to ask you if it is normal or not that this test fails on my hardware, and if not, what I could do to debug the problem and give you useful information to help understand what the issue is. I am currently testing on an Ubuntu Desktop VirtualBox VM which audio hardware (as the output of the lshw command, multimedia section) is: description: Multimedia audio controller product: 82801AA AC'97 Audio Controller vendor: Intel Corporation physical id: 5 bus info: pci@0000:00:05.0 version: 01 width: 32 bits clock: 33MHz capabilities: bus_master configuration: driver=snd_intel8x0 latency=64 resources: irq:21 ioport:d100(size=256) ioport:d200(size=64) From what I have seen until now it doesn’t look like a bug from the test nor an issue from the kernel, so I’m not really sure where to look at: I have compiled alsalib’s source code from the Git repo with debug symbols and I’ve followed the execution of the test until I reached a call to the ioctl syscall, which doesn’t return an error code so it doesn’t look like the problem relies in the kernel; I’m no expert though, I’ll let you confirm this or not. I am at your complete disposal to give you any further useful information as I can’t do much more from my side alone. Thank you in advance, looking forward to your kind response. Best Regards, Giacomo Guiduzzi

1 0

[PATCH v2 0/4] ALSA: hda: Expose codec organization functions
by Cezary Rojewski 21 Feb '22

21 Feb '22

Changes expose several function that are currently unavailable for HDA-DSP drivers for use. Those functions are: snd_hda_codec_cleanup_for_unbind() snd_hda_codec_set_power_save() snd_hda_codec_register() snd_hda_codec_unregister() snd_hda_codec_device_init() This allows upcoming AVS driver [1] to re-use even mode of HDA related code that is currently available in sound/pci/hda and sound/hda and prevent any code duplication within avs-driver that would otherwise had to happen. Last patch in the series provides snd_hdac_ext_bus_link_at() - a helper function which allows for retrieval of HDA segment (link) based on codec address directly. This is simpler than parsing codec-name first to extract the address what is the case for snd_hdac_ext_bus_get_link(). The latter function is updated to re-use newly added one so core logic is not duplicated after the addition. Changes v1 -> v2: - snd_hda_codec_new() now calls snd_hda_codec_device_new() with 'snddev_managed' param set to 'true', reported by Kai [1]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (4): ALSA: hda: Add snd_hdac_ext_bus_link_at() helper ALSA: hda: Update and expose snd_hda_codec_device_init() ALSA: hda: Update and expose codec register procedures ALSA: hda: Expose codec cleanup and power-save functions include/sound/hda_codec.h | 11 +++- include/sound/hdaudio_ext.h | 1 + sound/hda/ext/hdac_ext_controller.c | 31 +++++++--- sound/pci/hda/hda_codec.c | 94 ++++++++++++++++++++--------- sound/pci/hda/hda_local.h | 2 - sound/soc/codecs/hdac_hda.c | 2 +- 6 files changed, 100 insertions(+), 41 deletions(-) -- 2.25.1

2 6

[RFC PATCH 1/3] ASoC: simple-card-utils: add support for componants provideing jack events via set_jack
by Carl Philipp Klemm 20 Feb '22

20 Feb '22

This allows componants that want a jack to report state on to do so by calling set_jack on components implementing this function. Im not entirely sure this is the right way to do this so RFC Signed-off-by: Carl Philipp Klemm <philipp(a)uvos.xyz> --- include/sound/simple_card_utils.h | 6 ++-- sound/soc/generic/simple-card-utils.c | 47 +++++++++++++++++++-------- 2 files changed, 36 insertions(+), 17 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 51b3b485a92e..547ad537613d 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -57,8 +57,8 @@ struct asoc_simple_priv { struct prop_nums num; unsigned int mclk_fs; } *dai_props; - struct asoc_simple_jack hp_jack; - struct asoc_simple_jack mic_jack; + struct asoc_simple_jack *hp_jack; + struct asoc_simple_jack *mic_jack; struct snd_soc_dai_link *dai_link; struct asoc_simple_dai *dais; struct snd_soc_dai_link_component *dlcs; @@ -173,7 +173,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card, char *prefix); int asoc_simple_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, + struct asoc_simple_jack **sjack, int is_hp, char *prefix, char *pin); int asoc_simple_init_priv(struct asoc_simple_priv *priv, struct link_info *li); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 10c63b73900c..1899feba16cc 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -395,6 +395,7 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); struct asoc_simple_dai *dai; + struct snd_soc_component *component; int i, ret; for_each_prop_dai_codec(props, i, dai) { @@ -412,6 +413,21 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + for_each_rtd_components(rtd, i, component) { + if (component->driver->set_jack) { + if (!priv->hp_jack) { + priv->hp_jack = devm_kzalloc(priv->snd_card.dev, + sizeof(*priv->hp_jack), GFP_KERNEL); + snd_soc_card_jack_new(&priv->snd_card, + "Headphones", + SND_JACK_HEADPHONE, + &priv->hp_jack->jack, + NULL, 0); + } + snd_soc_component_set_jack(component, &priv->hp_jack->jack, NULL); + } + } + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_dai_init); @@ -554,7 +570,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card, EXPORT_SYMBOL_GPL(asoc_simple_parse_pin_switches); int asoc_simple_init_jack(struct snd_soc_card *card, - struct asoc_simple_jack *sjack, + struct asoc_simple_jack **sjack, int is_hp, char *prefix, char *pin) { @@ -569,8 +585,6 @@ int asoc_simple_init_jack(struct snd_soc_card *card, if (!prefix) prefix = ""; - sjack->gpio.gpio = -ENOENT; - if (is_hp) { snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); pin_name = pin ? pin : "Headphones"; @@ -588,21 +602,26 @@ int asoc_simple_init_jack(struct snd_soc_card *card, return -EPROBE_DEFER; if (gpio_is_valid(det)) { - sjack->pin.pin = pin_name; - sjack->pin.mask = mask; + struct asoc_simple_jack *sjack_d; + + sjack = devm_kzalloc(dev, sizeof(*(*sjack)), GFP_KERNEL); + sjack_d = *sjack; + + sjack_d->pin.pin = pin_name; + sjack_d->pin.mask = mask; - sjack->gpio.name = gpio_name; - sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); - sjack->gpio.debounce_time = 150; + sjack_d->gpio.name = gpio_name; + sjack_d->gpio.report = mask; + sjack_d->gpio.gpio = det; + sjack_d->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack_d->gpio.debounce_time = 150; snd_soc_card_jack_new(card, pin_name, mask, - &sjack->jack, - &sjack->pin, 1); + &sjack_d->jack, + &sjack_d->pin, 1); - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); + snd_soc_jack_add_gpios(&sjack_d->jack, 1, + &sjack_d->gpio); } return 0; -- 2.34.1

3 14

[PATCH 6/6] Documentation: sound: audio-graph-card: update dts
by Carl Philipp Klemm 20 Feb '22

20 Feb '22

bindings to add has-hp-jack Message-Id: <20220220011601.4424ad416d65b836e3165cfa(a)uvos.xyz> In-Reply-To: <OS3PR01MB8426FD3FD1B45BB31C971A56D4509(a)OS3PR01MB8426.jpnprd01.prod.outlook.com> References: <20211228190931.df5d518220080a734532ebfd(a)uvos.xyz> <87a6gawxpe.wl-kuninori.morimoto.gx(a)gmail.com> <20220105101028.620b7f2cb7727eedfccd933e(a)uvos.xyz> <87r19ld48v.wl-kuninori.morimoto.gx(a)gmail.com> <20220108145749.f55500222b049f344415a5c3(a)uvos.xyz> <OS3PR01MB8426FD3FD1B45BB31C971A56D4509(a)OS3PR01MB8426.jpnprd01.prod.outlook.com> X-Mailer: Sylpheed 3.7.0 (GTK+ 2.24.33; x86_64-unknown-linux-gnu) Mime-Version: 1.0 Content-Type: text/plain; charset=US-ASCII Content-Transfer-Encoding: 7bit Signed-off-by: Carl Philipp Klemm <philipp(a)uvos.xyz> --- Documentation/devicetree/bindings/sound/audio-graph.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/audio-graph.yaml b/Documentation/devicetree/bindings/sound/audio-graph.yaml index 4b46794e5153..5844b900789a 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph.yaml +++ b/Documentation/devicetree/bindings/sound/audio-graph.yaml @@ -38,6 +38,8 @@ properties: maxItems: 1 mic-det-gpio: maxItems: 1 + hp-det-gpio: + description: Set this properties if the device has a headphone jack to be reported on by a snd_soc_component required: - dais -- 2.35.1

1 0

[RESEND v13 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 19 Feb '22

19 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V12: -- Fix arguments type mismatch. Changes Since V11: -- Fix kernel robot issue on arguments type mismatch. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 304 ++++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 447 +++++++++++++++ sound/soc/qcom/lpass.h | 145 +++++ 9 files changed, 1887 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

5 28

[PATCH v14 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 19 Feb '22

19 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V13: -- Change bulk cdc clock voting to individual clock voting. -- Remove redundant code, conditional check and prints. -- Fix typo errors. Changes Since V12: -- Fix arguments type mismatch. Changes Since V11: -- Fix kernel robot issue on arguments type mismatch. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 305 +++++++++++ sound/soc/qcom/lpass-cpu.c | 252 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 601 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 447 +++++++++++++++ sound/soc/qcom/lpass.h | 143 +++++ 9 files changed, 1881 insertions(+), 84 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

2 11

[PATCH 0/1] sound: add quirk for Huawei D15
by Mauro Carvalho Chehab 19 Feb '22

19 Feb '22

Hi Pierre-Louis, Based on my tests, Huawei D15 (Intel) uses SSP0 on es8336. Add a quirk for it. Please notice that, currently, only the internal speaker is working. The topology for the internal microphone and for the headphones is wrong. Enabling/disabling the other two quirks (GPIO and/or DMIC) doesn't cause any audible results, nor change the devices listed on pavucontrol (tested with pipewire-pulse). Mauro Carvalho Chehab (1): ASoC: Intel: sof_es8336: add quirk for Huawei D15 2021 sound/soc/intel/boards/sof_es8336.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.33.1

3 4

[bug report] ASoC: codec: wcd938x: Add switch control for selecting CTIA/OMTP Headset
by Dan Carpenter 18 Feb '22

18 Feb '22

Hello Srinivasa Rao Mandadapu, This is a semi-automatic email about new static checker warnings. The patch 013cc2aea0f6: "ASoC: codec: wcd938x: Add switch control for selecting CTIA/OMTP Headset" from Feb 12, 2022, leads to the following Smatch complaint: sound/soc/codecs/wcd938x.c:4210 wcd938x_swap_gnd_mic() error: we previously assumed 'component' could be null (see line 4209) sound/soc/codecs/wcd938x.c 4203 static bool wcd938x_swap_gnd_mic(struct snd_soc_component *component, bool active) 4204 { 4205 int value; 4206 4207 struct wcd938x_priv *wcd938x; 4208 4209 if (!component) { ^^^^^^^^^ Check for NULL. 4210 dev_err(component->dev, "%s component is NULL\n", __func__); ^^^^^^^^^^^^^^ NULL Dereference. But can it really be NULL? 4211 return false; 4212 } 4213 regards, dan carpenter

1 0

Re: [PATCH 5/5] spi: make remove callback a void function
by Hans Verkuil 18 Feb '22

18 Feb '22

On 23/01/2022 18:52, Uwe Kleine-König wrote: > The value returned by an spi driver's remove function is mostly ignored. > (Only an error message is printed if the value is non-zero that the > error is ignored.) > > So change the prototype of the remove function to return no value. This > way driver authors are not tempted to assume that passing an error to > the upper layer is a good idea. All drivers are adapted accordingly. > There is no intended change of behaviour, all callbacks were prepared to > return 0 before. > > Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> > --- > drivers/bus/moxtet.c | 4 +--- > drivers/char/tpm/st33zp24/spi.c | 4 +--- > drivers/char/tpm/tpm_tis_spi_main.c | 3 +-- > drivers/clk/clk-lmk04832.c | 4 +--- > drivers/gpio/gpio-74x164.c | 4 +--- > drivers/gpio/gpio-max3191x.c | 4 +--- > drivers/gpio/gpio-max7301.c | 4 +--- > drivers/gpio/gpio-mc33880.c | 4 +--- > drivers/gpio/gpio-pisosr.c | 4 +--- > drivers/gpu/drm/panel/panel-abt-y030xx067a.c | 4 +--- > drivers/gpu/drm/panel/panel-ilitek-ili9322.c | 4 +--- > drivers/gpu/drm/panel/panel-ilitek-ili9341.c | 3 +-- > drivers/gpu/drm/panel/panel-innolux-ej030na.c | 4 +--- > drivers/gpu/drm/panel/panel-lg-lb035q02.c | 4 +--- > drivers/gpu/drm/panel/panel-lg-lg4573.c | 4 +--- > drivers/gpu/drm/panel/panel-nec-nl8048hl11.c | 4 +--- > drivers/gpu/drm/panel/panel-novatek-nt39016.c | 4 +--- > drivers/gpu/drm/panel/panel-samsung-db7430.c | 3 +-- > drivers/gpu/drm/panel/panel-samsung-ld9040.c | 4 +--- > drivers/gpu/drm/panel/panel-samsung-s6d27a1.c | 3 +-- > drivers/gpu/drm/panel/panel-samsung-s6e63m0-spi.c | 3 +-- > drivers/gpu/drm/panel/panel-sitronix-st7789v.c | 4 +--- > drivers/gpu/drm/panel/panel-sony-acx565akm.c | 4 +--- > drivers/gpu/drm/panel/panel-tpo-td028ttec1.c | 4 +--- > drivers/gpu/drm/panel/panel-tpo-td043mtea1.c | 4 +--- > drivers/gpu/drm/panel/panel-tpo-tpg110.c | 3 +-- > drivers/gpu/drm/panel/panel-widechips-ws2401.c | 3 +-- > drivers/gpu/drm/tiny/hx8357d.c | 4 +--- > drivers/gpu/drm/tiny/ili9163.c | 4 +--- > drivers/gpu/drm/tiny/ili9225.c | 4 +--- > drivers/gpu/drm/tiny/ili9341.c | 4 +--- > drivers/gpu/drm/tiny/ili9486.c | 4 +--- > drivers/gpu/drm/tiny/mi0283qt.c | 4 +--- > drivers/gpu/drm/tiny/repaper.c | 4 +--- > drivers/gpu/drm/tiny/st7586.c | 4 +--- > drivers/gpu/drm/tiny/st7735r.c | 4 +--- > drivers/hwmon/adcxx.c | 4 +--- > drivers/hwmon/adt7310.c | 3 +-- > drivers/hwmon/max1111.c | 3 +-- > drivers/hwmon/max31722.c | 4 +--- > drivers/iio/accel/bma400_spi.c | 4 +--- > drivers/iio/accel/bmc150-accel-spi.c | 4 +--- > drivers/iio/accel/bmi088-accel-spi.c | 4 +--- > drivers/iio/accel/kxsd9-spi.c | 4 +--- > drivers/iio/accel/mma7455_spi.c | 4 +--- > drivers/iio/accel/sca3000.c | 4 +--- > drivers/iio/adc/ad7266.c | 4 +--- > drivers/iio/adc/ltc2496.c | 4 +--- > drivers/iio/adc/mcp320x.c | 4 +--- > drivers/iio/adc/mcp3911.c | 4 +--- > drivers/iio/adc/ti-adc12138.c | 4 +--- > drivers/iio/adc/ti-ads7950.c | 4 +--- > drivers/iio/adc/ti-ads8688.c | 4 +--- > drivers/iio/adc/ti-tlc4541.c | 4 +--- > drivers/iio/amplifiers/ad8366.c | 4 +--- > drivers/iio/common/ssp_sensors/ssp_dev.c | 4 +--- > drivers/iio/dac/ad5360.c | 4 +--- > drivers/iio/dac/ad5380.c | 4 +--- > drivers/iio/dac/ad5446.c | 4 +--- > drivers/iio/dac/ad5449.c | 4 +--- > drivers/iio/dac/ad5504.c | 4 +--- > drivers/iio/dac/ad5592r.c | 4 +--- > drivers/iio/dac/ad5624r_spi.c | 4 +--- > drivers/iio/dac/ad5686-spi.c | 4 +--- > drivers/iio/dac/ad5761.c | 4 +--- > drivers/iio/dac/ad5764.c | 4 +--- > drivers/iio/dac/ad5791.c | 4 +--- > drivers/iio/dac/ad8801.c | 4 +--- > drivers/iio/dac/ltc1660.c | 4 +--- > drivers/iio/dac/ltc2632.c | 4 +--- > drivers/iio/dac/mcp4922.c | 4 +--- > drivers/iio/dac/ti-dac082s085.c | 4 +--- > drivers/iio/dac/ti-dac7311.c | 3 +-- > drivers/iio/frequency/adf4350.c | 4 +--- > drivers/iio/gyro/bmg160_spi.c | 4 +--- > drivers/iio/gyro/fxas21002c_spi.c | 4 +--- > drivers/iio/health/afe4403.c | 4 +--- > drivers/iio/magnetometer/bmc150_magn_spi.c | 4 +--- > drivers/iio/magnetometer/hmc5843_spi.c | 4 +--- > drivers/iio/potentiometer/max5487.c | 4 +--- > drivers/iio/pressure/ms5611_spi.c | 4 +--- > drivers/iio/pressure/zpa2326_spi.c | 4 +--- > drivers/input/keyboard/applespi.c | 4 +--- > drivers/input/misc/adxl34x-spi.c | 4 +--- > drivers/input/touchscreen/ads7846.c | 4 +--- > drivers/input/touchscreen/cyttsp4_spi.c | 4 +--- > drivers/input/touchscreen/tsc2005.c | 4 +--- > drivers/leds/leds-cr0014114.c | 4 +--- > drivers/leds/leds-dac124s085.c | 4 +--- > drivers/leds/leds-el15203000.c | 4 +--- > drivers/leds/leds-spi-byte.c | 4 +--- > drivers/media/spi/cxd2880-spi.c | 4 +--- > drivers/media/spi/gs1662.c | 4 +--- > drivers/media/tuners/msi001.c | 3 +-- A bit late, but for drivers/media: Acked-by: Hans Verkuil <hverkuil-cisco(a)xs4all.nl> Thanks! Hans

1 0

Multiple codecs of the same type on the same card
by Ricard Wanderlof 18 Feb '22

18 Feb '22

When adding codecs to a card, all the controls of each individual codec are visible as controls for the complete card. But how does one handle the case of multiple codecs of the same type, for instance two identical codecs being used in TDM mode to create a multi channel card? Is there some method for resolving the fact that the control names naturally conflict? Or is this expected to be resolved in the machine driver? Currently I've been using simple-card, but perhaps this type of configuration is too advanced to be resolved for that, and a specialized machine driver is needed? /Ricard -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 1

[PATCH] ASoC: codecs: Add power domains support in digital macro codecs
by Srinivasa Rao Mandadapu 17 Feb '22

17 Feb '22

Add support for enabling required power domains in digital macro codecs. macro and dcodec power domains are being requested as clocks by HLOS in ADSP based architectures and ADSP internally handling as powerdomains. In ADSP bypass case need to handle them as power domains explicitly. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lpass-macro-common.c | 72 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 18 +++++++++ sound/soc/codecs/lpass-rx-macro.c | 13 ++++++- sound/soc/codecs/lpass-tx-macro.c | 10 +++++ sound/soc/codecs/lpass-va-macro.c | 11 +++++- sound/soc/qcom/Kconfig | 1 + 8 files changed, 129 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2627f7..d3a850f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -244,6 +244,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_WCD9335 imply SND_SOC_WCD934X imply SND_SOC_WCD938X_SDW + imply SND_SOC_LPASS_MACRO_COMMON imply SND_SOC_LPASS_RX_MACRO imply SND_SOC_LPASS_TX_MACRO imply SND_SOC_WL1273 @@ -2008,6 +2009,9 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_LPASS_MACRO_COMMON + tristate + config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK select REGMAP_MMIO diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b4e11c3..c3c6059 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-lochnagar-sc-objs := lochnagar-sc.o +snd-soc-lpass-macro-common-objs := lpass-macro-common.o snd-soc-lpass-rx-macro-objs := lpass-rx-macro.o snd-soc-lpass-tx-macro-objs := lpass-tx-macro.o snd-soc-lpass-wsa-macro-objs := lpass-wsa-macro.o @@ -676,6 +677,7 @@ obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_LPASS_MACRO_COMMON) += snd-soc-lpass-macro-common.o obj-$(CONFIG_SND_SOC_LPASS_WSA_MACRO) += snd-soc-lpass-wsa-macro.o obj-$(CONFIG_SND_SOC_LPASS_VA_MACRO) += snd-soc-lpass-va-macro.o obj-$(CONFIG_SND_SOC_LPASS_RX_MACRO) += snd-soc-lpass-rx-macro.o diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c new file mode 100644 index 0000000..b8e50e6 --- /dev/null +++ b/sound/soc/codecs/lpass-macro-common.c @@ -0,0 +1,72 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2022, The Linux Foundation. All rights reserved. + +#include <linux/export.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/pm_domain.h> +#include <linux/pm_runtime.h> + +#include "lpass-macro-common.h" + +int lpass_macro_pds_init(struct platform_device *pdev, struct lpass_macro **pds) +{ + struct device *dev = &pdev->dev; + struct lpass_macro *l_pds; + int ret; + + const struct property *prop = of_find_property(dev->of_node, "power-domains", NULL); + + if (!prop) + return 0; + + l_pds = devm_kzalloc(dev, sizeof(*l_pds), GFP_KERNEL); + if (!l_pds) + return -ENOMEM; + + l_pds->macro_pd = dev_pm_domain_attach_by_name(dev, "macro"); + if (IS_ERR_OR_NULL(l_pds->macro_pd)) { + ret = PTR_ERR(l_pds->macro_pd) ? : -ENODATA; + return ret; + } + ret = pm_runtime_get_sync(l_pds->macro_pd); + if (ret < 0) { + dev_err(dev, "%s failed for macro_pd, ret %d\n", __func__, ret); + dev_pm_domain_detach(l_pds->macro_pd, false); + pm_runtime_put_noidle(l_pds->macro_pd); + return ret; + } + + l_pds->dcodec_pd = dev_pm_domain_attach_by_name(dev, "dcodec"); + if (IS_ERR_OR_NULL(l_pds->dcodec_pd)) { + ret = PTR_ERR(l_pds->dcodec_pd) ? : -ENODATA; + dev_pm_domain_detach(l_pds->macro_pd, false); + return ret; + } + + ret = pm_runtime_get_sync(l_pds->dcodec_pd); + if (ret < 0) { + dev_err(dev, "%s failed for dcodec_pd, ret %d\n", __func__, ret); + + dev_pm_domain_detach(l_pds->dcodec_pd, false); + pm_runtime_put_noidle(l_pds->dcodec_pd); + return ret; + } + *pds = l_pds; + return ret; +} +EXPORT_SYMBOL_GPL(lpass_macro_pds_init); + +void lpass_macro_pds_exit(struct platform_device *pdev, struct lpass_macro *pds) +{ + pm_runtime_put(pds->macro_pd); + pm_runtime_put(pds->dcodec_pd); + dev_pm_domain_detach(pds->macro_pd, false); + dev_pm_domain_detach(pds->dcodec_pd, false); +} +EXPORT_SYMBOL_GPL(lpass_macro_pds_exit); + +MODULE_DESCRIPTION("QTI SC7280 LPI GPIO pin control driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h new file mode 100644 index 0000000..c343f0e --- /dev/null +++ b/sound/soc/codecs/lpass-macro-common.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (c) 2022, The Linux Foundation. All rights reserved. + */ + +#ifndef __LPASS_MACRO_COMMON_H__ +#define __LPASS_MACRO_COMMON_H__ + + +struct lpass_macro { + struct device *macro_pd; + struct device *dcodec_pd; +}; + +int lpass_macro_pds_init(struct platform_device *pdev, struct lpass_macro **pds); +void lpass_macro_pds_exit(struct platform_device *pdev, struct lpass_macro *pds); + +#endif /* __LPASS_MACRO_COMMON_H__ */ diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 29d214f..db32090 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -14,6 +14,8 @@ #include <linux/of_clk.h> #include <linux/clk-provider.h> +#include "lpass-macro-common.h" + #define CDC_RX_TOP_TOP_CFG0 (0x0000) #define CDC_RX_TOP_SWR_CTRL (0x0008) #define CDC_RX_TOP_DEBUG (0x000C) @@ -606,7 +608,7 @@ struct rx_macro { int is_softclip_on; int is_aux_hpf_on; int softclip_clk_users; - + struct lpass_macro *pds; struct regmap *regmap; struct clk_bulk_data clks[RX_NUM_CLKS_MAX]; struct clk_hw hw; @@ -3537,6 +3539,12 @@ static int rx_macro_probe(struct platform_device *pdev) return ret; } + ret = lpass_macro_pds_init(pdev, &rx->pds); + if (ret < 0) { + dev_err(dev, "Enabling power domains failed in %s\n", __func__); + return ret; + } + base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); @@ -3575,6 +3583,9 @@ static int rx_macro_remove(struct platform_device *pdev) of_clk_del_provider(pdev->dev.of_node); clk_bulk_disable_unprepare(RX_NUM_CLKS_MAX, rx->clks); + + lpass_macro_pds_exit(pdev, rx->pds); + return 0; } diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 9c96ab1..4d1e5ab 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -13,6 +13,8 @@ #include <linux/of_clk.h> #include <linux/clk-provider.h> +#include "lpass-macro-common.h" + #define CDC_TX_CLK_RST_CTRL_MCLK_CONTROL (0x0000) #define CDC_TX_MCLK_EN_MASK BIT(0) #define CDC_TX_MCLK_ENABLE BIT(0) @@ -266,6 +268,7 @@ struct tx_macro { u16 dmic_clk_div; bool bcs_enable; int dec_mode[NUM_DECIMATORS]; + struct lpass_macro *pds; bool bcs_clk_en; }; #define to_tx_macro(_hw) container_of(_hw, struct tx_macro, hw) @@ -1802,6 +1805,11 @@ static int tx_macro_probe(struct platform_device *pdev) return ret; } + ret = lpass_macro_pds_init(pdev, &tx->pds); + if (ret < 0) { + dev_err(dev, "Enabling power domains failed in %s\n", __func__); + return ret; + } base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); @@ -1859,6 +1867,8 @@ static int tx_macro_remove(struct platform_device *pdev) clk_bulk_disable_unprepare(TX_NUM_CLKS_MAX, tx->clks); + lpass_macro_pds_exit(pdev, tx->pds); + return 0; } diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 11147e3..b29b9a1 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -15,6 +15,8 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include "lpass-macro-common.h" + /* VA macro registers */ #define CDC_VA_CLK_RST_CTRL_MCLK_CONTROL (0x0000) #define CDC_VA_MCLK_CONTROL_EN BIT(0) @@ -195,6 +197,7 @@ struct va_macro { struct regmap *regmap; struct clk_bulk_data clks[VA_NUM_CLKS_MAX]; struct clk_hw hw; + struct lpass_macro *pds; s32 dmic_0_1_clk_cnt; s32 dmic_2_3_clk_cnt; @@ -1413,7 +1416,11 @@ static int va_macro_probe(struct platform_device *pdev) dev_err(dev, "Error getting VA Clocks (%d)\n", ret); return ret; } - + ret = lpass_macro_pds_init(pdev, &va->pds); + if (ret < 0) { + dev_err(dev, "Enabling power domains failed %s\n", __func__); + return ret; + } ret = of_property_read_u32(dev->of_node, "qcom,dmic-sample-rate", &sample_rate); if (ret) { @@ -1468,6 +1475,8 @@ static int va_macro_remove(struct platform_device *pdev) clk_bulk_disable_unprepare(VA_NUM_CLKS_MAX, va->clks); + lpass_macro_pds_exit(pdev, va->pds); + return 0; } diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 52db003..6ffd51a 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -194,6 +194,7 @@ config SND_SOC_SC7280 select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X + select SND_SOC_LPASS_MACRO_COMMON select SND_SOC_LPASS_RX_MACRO select SND_SOC_LPASS_TX_MACRO help -- 2.7.4

2 3

[PATCH] ASoC: SOF: Replace zero-length array with flexible-array member
by Stephen Kitt 17 Feb '22

17 Feb '22

There is a regular need in the kernel to provide a way to declare having a dynamically sized set of trailing elements in a structure. Kernel code should always use "flexible array members"[1] for these cases. The older style of one-element or zero-length arrays should no longer be used[2]. This helps with the ongoing efforts to globally enable -Warray-bounds and get us closer to being able to tighten the FORTIFY_SOURCE routines on memcpy(). [1] https://en.wikipedia.org/wiki/Flexible_array_member [2] https://www.kernel.org/doc/html/latest/process/deprecated.html#zero-length-… Link: https://github.com/KSPP/linux/issues/78 Link: https://github.com/KSPP/linux/issues/180 Suggested-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> Signed-off-by: Stephen Kitt <steve(a)sk2.org> --- include/sound/sof/topology.h | 2 +- sound/soc/sof/topology.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/topology.h b/include/sound/sof/topology.h index d12736e14b69..adee6afd1490 100644 --- a/include/sound/sof/topology.h +++ b/include/sound/sof/topology.h @@ -237,7 +237,7 @@ struct sof_ipc_comp_process { /* reserved for future use */ uint32_t reserved[7]; - uint8_t data[0]; + uint8_t data[]; } __packed; /* frees components, buffers and pipelines diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index e72dcae5e7ee..1d119d1dd69d 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2164,7 +2164,7 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, */ if (ipc_data_size) { for (i = 0; i < widget->num_kcontrols; i++) { - memcpy(&process->data + offset, + memcpy(&process->data[offset], wdata[i].pdata->data, wdata[i].pdata->size); offset += wdata[i].pdata->size; base-commit: f71077a4d84bbe8c7b91b7db7c4ef815755ac5e3 -- 2.27.0

4 3

ucm2: sof-essx8336: initial support
by GitHub pull_request - opened 17 Feb '22

17 Feb '22

alsa-project/alsa-ucm-conf pull request #139 was opened from plbossart: Loosely based on the sof-hda-dsp model for DMIC conditional handling. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/139 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/139.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: SOF: core: unregister clients and machine drivers in .shutdown
by Kai Vehmanen 17 Feb '22

17 Feb '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> On a platform shutdown, the expectation for most drivers is that userspace tasks will release all resources. When those sequences do not complete, it can be the case that PCM devices exposed by ALSA cards are used *after* the DSP shutdown completes, leading to a platform hang. When the clients and machine drivers provide an _unregister callback, let's invoke it in the shutdown sequence. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/core.c | 9 +++++++++ 1 file changed, 9 insertions(+) V1->V2: - correct patch title to have "ASoC: SOF: " prefix (reported by Amadeusz) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index d99ecbb4282d..2a35d8ddf43e 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -463,10 +463,19 @@ EXPORT_SYMBOL(snd_sof_device_remove); int snd_sof_device_shutdown(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_sof_pdata *pdata = sdev->pdata; if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); + /* + * make sure clients and machine driver(s) are unregistered to force + * all userspace devices to be closed prior to the DSP shutdown sequence + */ + sof_unregister_clients(sdev); + + snd_sof_machine_unregister(sdev, pdata); + if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) return snd_sof_shutdown(sdev); base-commit: f7d344a2bd5ec81fbd1ce76928fd059e57ec9bea -- 2.35.1

3 2

[PATCH] ASoC: codec: wcd938x: Update CTIA/OMTP switch control
by Srinivasa Rao Mandadapu 17 Feb '22

17 Feb '22

Convert gpio api's to gpio descriptor api's in CTIA/OMTP switch control. Remove redundant NULL checks in swap_gnd_mic function. Fixes: 013cc2aea0f6 ("ASoC: codec: wcd938x: Add switch control for selecting CTIA/OMTP Headset") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- sound/soc/codecs/wcd938x.c | 33 +++++++++++---------------------- 1 file changed, 11 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 08d16a9..88a39e1 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -6,6 +6,7 @@ #include <linux/platform_device.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/gpio/consumer.h> #include <linux/kernel.h> #include <linux/pm_runtime.h> #include <linux/component.h> @@ -194,7 +195,7 @@ struct wcd938x_priv { int ear_rx_path; int variant; int reset_gpio; - int us_euro_gpio; + struct gpio_desc *us_euro_gpio; u32 micb1_mv; u32 micb2_mv; u32 micb3_mv; @@ -4203,22 +4204,11 @@ static bool wcd938x_swap_gnd_mic(struct snd_soc_component *component, bool activ struct wcd938x_priv *wcd938x; - if (!component) { - dev_err(component->dev, "%s component is NULL\n", __func__); - return false; - } - wcd938x = snd_soc_component_get_drvdata(component); - if (!wcd938x) { - dev_err(component->dev, "%s private data is NULL\n", __func__); - return false; - } - value = gpio_get_value(wcd938x->us_euro_gpio); + value = gpiod_get_value(wcd938x->us_euro_gpio); - gpio_set_value(wcd938x->us_euro_gpio, !value); - /* 20us sleep required after changing the gpio state*/ - usleep_range(20, 30); + gpiod_set_value(wcd938x->us_euro_gpio, !value); return true; } @@ -4236,16 +4226,15 @@ static int wcd938x_populate_dt_data(struct wcd938x_priv *wcd938x, struct device return wcd938x->reset_gpio; } - wcd938x->us_euro_gpio = of_get_named_gpio(dev->of_node, "us-euro-gpios", 0); - if (wcd938x->us_euro_gpio < 0) { - dev_err(dev, "Failed to get us-euro-gpios gpio: err = %d\n", wcd938x->us_euro_gpio); - } else { - cfg->swap_gnd_mic = wcd938x_swap_gnd_mic; - gpio_direction_output(wcd938x->us_euro_gpio, 0); - /* 20us sleep required after pulling the reset gpio to LOW */ - usleep_range(20, 30); + wcd938x->us_euro_gpio = devm_gpiod_get_optional(dev, "us-euro", + GPIOD_OUT_LOW); + if (IS_ERR(wcd938x->us_euro_gpio)) { + dev_err(dev, "us-euro swap Control GPIO not found\n"); + return PTR_ERR(wcd938x->us_euro_gpio); } + cfg->swap_gnd_mic = wcd938x_swap_gnd_mic; + wcd938x->supplies[0].supply = "vdd-rxtx"; wcd938x->supplies[1].supply = "vdd-io"; wcd938x->supplies[2].supply = "vdd-buck"; -- 2.7.4

2 1

[PATCH 00/15] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 17 Feb '22

17 Feb '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Jiaxin Yu (15): ASoC: mediatek: mt6366: add codec driver ASoC: mediatek: mt8186: support audsys clock control ASoC: mediatek: mt8186: support adda in platform driver ASoC: mediatek: mt8186: support hostless in platform driver ASoC: mediatek: mt8186: support hw gain in platform driver ASoC: mediatek: mt8186: support i2s in platform driver ASoC: mediatek: mt8186: support pcm in platform driver ASoC: mediatek: mt8186: support src in platform driver ASoC: mediatek: mt8186: support tdm in platform driver ASoC: mediatek: mt8186: add platform driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682 dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682 document .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 47 + .../sound/mt8186-mt6366-rt1019-rt5682.yaml | 47 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 21 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 719 ++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 210 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 245 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 262 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 211 + sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3030 +++++++++++++++ sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 151 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 + .../soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 + sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 897 +++++ .../soc/mediatek/mt8186/mt8186-dai-hostless.c | 296 ++ .../soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 246 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1352 +++++++ sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 433 +++ sound/soc/mediatek/mt8186/mt8186-dai-src.c | 758 ++++ sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 723 ++++ .../mediatek/mt8186/mt8186-interconnection.h | 69 + .../soc/mediatek/mt8186/mt8186-misc-control.c | 1729 +++++++++ .../mt8186/mt8186-mt6366-da7219-max98357.c | 903 +++++ .../mt8186/mt8186-mt6366-rt1019-rt5682.c | 887 +++++ sound/soc/mediatek/mt8186/mt8186-reg.h | 3433 +++++++++++++++++ 30 files changed, 16978 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h -- 2.18.0

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Re: [PATCH 10/15] ASoC: mediatek: mt8186: add platform driver
by Mark Brown 17 Feb '22

17 Feb '22

On Fri, Feb 11, 2022 at 06:38:13PM +0800, Jiaxin Yu wrote: > sound/soc/mediatek/Kconfig | 44 + > sound/soc/mediatek/Makefile | 1 + > sound/soc/mediatek/mt8186/Makefile | 21 + > sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 719 ++++ > sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 210 + > sound/soc/mediatek/mt8186/mt8186-afe-common.h | 245 ++ > .../soc/mediatek/mt8186/mt8186-afe-control.c | 262 ++ > sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 211 + > sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + > sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3030 +++++++++++++++ > .../mediatek/mt8186/mt8186-interconnection.h | 69 + > .../soc/mediatek/mt8186/mt8186-misc-control.c | 1729 +++++++++ > sound/soc/mediatek/mt8186/mt8186-reg.h | 3433 +++++++++++++++++ I know it's already a long series but perhaps the clock and GPIO bits could be split out into separate patches? This one patch is over 300K which is a bit much in one go, especially when it's not just all big tables.

2 1

[PATCH 0/2] ASoC: audio_graph_card2: Support variable slot widths
by Richard Fitzgerald 17 Feb '22

17 Feb '22

This adds support for I2S/TDM links where the slot width varies depending on the sample width, in a way that cannot be guessed by component hw_params(). A typical example is: - 16-bit samples use 16-bit slots - 24-bit samples use 32-bit slots There is no way for a component hw_params() to deduce from the information it is passed that 24-bit samples will be in 32-bit slots. Some audio hardware cannot support a fixed slot width or a slot width equal to the sample width in all cases. This is usually due either to limitations of the audio serial port or system clocking restrictions. These two patches add support for defining a mapping between sample widths and sample slots. This allows audio_graph_card2 to be used in these situations instead of having to write a custom machine driver. Richard Fitzgerald (2): ASoC: dt-bindings: audio-graph-port: Add dai-tdm-slot-width-map ASoC: audio_graph_card2: Add support for variable slot widths .../bindings/sound/audio-graph-port.yaml | 7 ++ include/sound/simple_card_utils.h | 11 +++ sound/soc/generic/audio-graph-card2.c | 4 + sound/soc/generic/simple-card-utils.c | 95 +++++++++++++++++++ 4 files changed, 117 insertions(+) -- 2.30.2

2 4

[PATCH 0/3] ALSA: Set max DMA segment size
by Takashi Iwai 17 Feb '22

17 Feb '22

Hi, this is a oneliner patches to address the spurious kernel WARNING from DMA debug code for the missing setup of DMA segment size for SG pages. Takashi === Takashi Iwai (3): ALSA: hda: Set max DMA segment size ASoC: SOF: hda: Set max DMA segment size ASoC: intel: skylake: Set max DMA segment size sound/pci/hda/hda_intel.c | 1 + sound/soc/intel/skylake/skl.c | 1 + sound/soc/sof/intel/hda.c | 1 + 3 files changed, 3 insertions(+) -- 2.31.1

3 7

[PATCH v2] ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2
by Kai Vehmanen 17 Feb '22

17 Feb '22

Implement HDA keep alive (KAE) support for Intel display codecs. When no audio stream is active, the display codec will provide a continuous clock and a valid but silent audio stream to any connected HDMI/DP receiver. Without this, upon starting a new playback stream, initial samples may be lost as many receivers require time to initialize for new clock. This is a new feature in Intel AlderLake-P display codec implementation and replaces the Intel i915 silent-stream extension that has been used on older hardware. Main benefit of the new method is that codec no longer needs to be kept in D0 power state. This patch depends on commit 112a87c48e83 ("drm/i915/display: program audio CDCLK-TS for keepalives"). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jyri Sarha <jyri.sarha(a)intel.com> --- sound/pci/hda/Kconfig | 19 ++--- sound/pci/hda/patch_hdmi.c | 157 +++++++++++++++++++++++++++---------- 2 files changed, 127 insertions(+), 49 deletions(-) V1->V2: - fix unbalance in snd_hda_power_up_pm()/snd_hda_power_down_pm() diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index febe1c2b7d9a..9f6c99c1d87b 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -285,15 +285,16 @@ config SND_HDA_INTEL_HDMI_SILENT_STREAM bool "Enable Silent Stream always for HDMI" depends on SND_HDA_INTEL help - Intel hardware has a feature called 'silent stream', that - keeps external HDMI receiver's analog circuitry powered on - avoiding 2-3 sec silence during playback start. This mechanism - relies on setting channel_id as 0xf, sending info packet and - preventing codec D3 entry (increasing platform static power - consumption when HDMI receiver is plugged-in). 2-3 sec silence - at the playback start is expected whenever there is format change. - (default is 2 channel format). - Say Y to enable Silent Stream feature. + Say Y to enable HD-Audio Keep Alive (KAE) aka Silent Stream + for HDMI on hardware that supports the feature. + + When enabled, the HDMI/DisplayPort codec will continue to provide + a continuous clock and a valid but silent data stream to + any connected external receiver. This allows to avoid gaps + at start of playback. Many receivers require multiple seconds + to start playing audio after the clock has been stopped. + This feature can impact power consumption as resources + are kept reserved both at transmitter and receiver. endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 64fe025fda86..a7016d51b2ff 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -120,6 +120,12 @@ struct hdmi_pcm { struct snd_kcontrol *eld_ctl; }; +enum { + SILENT_STREAM_OFF = 0, + SILENT_STREAM_KAE, /* use standard HDA Keep-Alive */ + SILENT_STREAM_I915, /* Intel i915 extension */ +}; + struct hdmi_spec { struct hda_codec *codec; int num_cvts; @@ -179,7 +185,7 @@ struct hdmi_spec { hda_nid_t vendor_nid; const int *port_map; int port_num; - bool send_silent_stream; /* Flag to enable silent stream feature */ + int silent_stream_type; }; #ifdef CONFIG_SND_HDA_COMPONENT @@ -1665,18 +1671,71 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, #define I915_SILENT_FORMAT_BITS 16 #define I915_SILENT_FMT_MASK 0xf +static void silent_stream_enable_i915(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + unsigned int format; + + snd_hdac_sync_audio_rate(&codec->core, per_pin->pin_nid, + per_pin->dev_id, I915_SILENT_RATE); + + /* trigger silent stream generation in hw */ + format = snd_hdac_calc_stream_format(I915_SILENT_RATE, I915_SILENT_CHANNELS, + I915_SILENT_FORMAT, I915_SILENT_FORMAT_BITS, 0); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, + I915_SILENT_FMT_MASK, I915_SILENT_FMT_MASK, format); + usleep_range(100, 200); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, I915_SILENT_FMT_MASK, 0, format); + + per_pin->channels = I915_SILENT_CHANNELS; + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); +} + +static void silent_stream_set_kae(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + bool enable) +{ + unsigned int param; + + codec_dbg(codec, "HDMI: KAE %d cvt-NID=0x%x\n", enable, per_pin->cvt_nid); + + param = snd_hda_codec_read(codec, per_pin->cvt_nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + param = (param >> 16) & 0xff; + + if (enable) + param |= AC_DIG3_KAE; + else + param &= ~AC_DIG3_KAE; + + snd_hda_codec_write(codec, per_pin->cvt_nid, 0, AC_VERB_SET_DIGI_CONVERT_3, param); +} + static void silent_stream_enable(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_cvt *per_cvt; int cvt_idx, pin_idx, err; - unsigned int format; + int keep_power = 0; + + /* + * Power-up will call hdmi_present_sense, so the PM calls + * have to be done without mutex held. + */ + + err = snd_hda_power_up_pm(codec); + if (err < 0 && err != -EACCES) { + codec_err(codec, + "Failed to power up codec for silent stream enable ret=[%d]\n", err); + snd_hda_power_down_pm(codec); + return; + } mutex_lock(&per_pin->lock); if (per_pin->setup) { codec_dbg(codec, "hdmi: PCM already open, no silent stream\n"); + err = -EBUSY; goto unlock_out; } @@ -1703,22 +1762,23 @@ static void silent_stream_enable(struct hda_codec *codec, /* configure unused pins to choose other converters */ pin_cvt_fixup(codec, per_pin, 0); - snd_hdac_sync_audio_rate(&codec->core, per_pin->pin_nid, - per_pin->dev_id, I915_SILENT_RATE); - - /* trigger silent stream generation in hw */ - format = snd_hdac_calc_stream_format(I915_SILENT_RATE, I915_SILENT_CHANNELS, - I915_SILENT_FORMAT, I915_SILENT_FORMAT_BITS, 0); - snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, - I915_SILENT_FMT_MASK, I915_SILENT_FMT_MASK, format); - usleep_range(100, 200); - snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, I915_SILENT_FMT_MASK, 0, format); - - per_pin->channels = I915_SILENT_CHANNELS; - hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + switch (spec->silent_stream_type) { + case SILENT_STREAM_KAE: + silent_stream_set_kae(codec, per_pin, true); + break; + case SILENT_STREAM_I915: + silent_stream_enable_i915(codec, per_pin); + keep_power = 1; + break; + default: + break; + } unlock_out: mutex_unlock(&per_pin->lock); + + if (err || !keep_power) + snd_hda_power_down_pm(codec); } static void silent_stream_disable(struct hda_codec *codec, @@ -1726,7 +1786,16 @@ static void silent_stream_disable(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_cvt *per_cvt; - int cvt_idx; + int cvt_idx, err; + + err = snd_hda_power_up_pm(codec); + if (err < 0 && err != -EACCES) { + codec_err(codec, + "Failed to power up codec for silent stream disable ret=[%d]\n", + err); + snd_hda_power_down_pm(codec); + return; + } mutex_lock(&per_pin->lock); if (!per_pin->silent_stream) @@ -1741,11 +1810,21 @@ static void silent_stream_disable(struct hda_codec *codec, per_cvt->assigned = 0; } + if (spec->silent_stream_type == SILENT_STREAM_I915) { + /* release ref taken in silent_stream_enable() */ + snd_hda_power_down_pm(codec); + } + else if (spec->silent_stream_type == SILENT_STREAM_KAE) { + silent_stream_set_kae(codec, per_pin, false); + } + per_pin->cvt_nid = 0; per_pin->silent_stream = false; unlock_out: mutex_unlock(&per_pin->lock); + + snd_hda_power_down_pm(codec); } /* update ELD and jack state via audio component */ @@ -1767,29 +1846,11 @@ static void sync_eld_via_acomp(struct hda_codec *codec, monitor_next = per_pin->sink_eld.monitor_present; mutex_unlock(&per_pin->lock); - /* - * Power-up will call hdmi_present_sense, so the PM calls - * have to be done without mutex held. - */ - - if (spec->send_silent_stream) { - int pm_ret; - - if (!monitor_prev && monitor_next) { - pm_ret = snd_hda_power_up_pm(codec); - if (pm_ret < 0) - codec_err(codec, - "Monitor plugged-in, Failed to power up codec ret=[%d]\n", - pm_ret); + if (spec->silent_stream_type) { + if (!monitor_prev && monitor_next) silent_stream_enable(codec, per_pin); - } else if (monitor_prev && !monitor_next) { + else if (monitor_prev && !monitor_next) silent_stream_disable(codec, per_pin); - pm_ret = snd_hda_power_down_pm(codec); - if (pm_ret < 0) - codec_err(codec, - "Monitor plugged-out, Failed to power down codec ret=[%d]\n", - pm_ret); - } } } @@ -2982,7 +3043,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, * module param or Kconfig option */ if (send_silent_stream) - spec->send_silent_stream = true; + spec->silent_stream_type = SILENT_STREAM_I915; return parse_intel_hdmi(codec); } @@ -3035,6 +3096,22 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) return ret; } +static int patch_i915_adlp_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int res; + + res = patch_i915_tgl_hdmi(codec); + if (!res) { + spec = codec->spec; + + if (spec->silent_stream_type) + spec->silent_stream_type = SILENT_STREAM_KAE; + } + + return res; +} + /* Intel Baytrail and Braswell; with eld notifier */ static int patch_i915_byt_hdmi(struct hda_codec *codec) { @@ -4391,10 +4468,10 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862818, "Raptorlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), -HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), base-commit: ffef8d871491c18f64cf05f095740383785cd09a -- 2.35.1

2 1

[PATCH v3 0/5] Tegra234 APE support
by Sameer Pujar 17 Feb '22

17 Feb '22

This adds APE support on Tegra234 and enables sound card on Jetson AGX Orin platform. Changelog ========= v2->v3 ------ * Fixed DT build error related to power-domain and iommu reference. Reported-by: kernel test robot <lkp(a)intel.com> v1->v2 ------ * Rebased patches on latest linux-next Mohan Kumar (1): ASoC: tegra: Update AHUB driver for Tegra234 Sameer Pujar (4): dt-bindings: Add Tegra234 APE support dt-bindings: Document Tegra234 APE support arm64: tegra: Add audio devices on Tegra234 arm64: tegra: APE sound card for Jetson AGX Orin .../bindings/bus/nvidia,tegra210-aconnect.yaml | 1 + .../bindings/dma/nvidia,tegra210-adma.yaml | 4 +- .../bindings/interrupt-controller/arm,gic.yaml | 1 + .../bindings/sound/nvidia,tegra186-dspk.yaml | 4 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 4 +- .../bindings/sound/nvidia,tegra210-adx.yaml | 1 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 1 + .../bindings/sound/nvidia,tegra210-amx.yaml | 3 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 1 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 1 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 1 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 1 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 1 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 1783 ++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra234.dtsi | 432 +++++ include/dt-bindings/clock/tegra234-clock.h | 74 +- include/dt-bindings/memory/tegra234-mc.h | 7 + include/dt-bindings/power/tegra234-powergate.h | 9 + sound/soc/tegra/tegra210_ahub.c | 146 +- sound/soc/tegra/tegra210_ahub.h | 4 +- 20 files changed, 2473 insertions(+), 6 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.7.4

3 8

[RESEND v7 0/7] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 17 Feb '22

17 Feb '22

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V6: -- Update conditional clock voting to optional clock voting. -- Update Kconfig depends on field with select. -- Fix typo errors. Changes Since V5: -- Create new patch by updating macro name to lpi specific. -- Create new patch by updating lpi pin group structure with core group_desc structure. -- Fix typo errors. -- Sort macros in the make file and configuration file. Changes Since V4: -- Update commit message and description of the chip specific extraction patch. -- Sort macros in kconfig and makefile. -- Update optional clock voting to conditional clock voting. -- Fix typo errors. -- Move to quicinc domain email id's. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (7): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Update macro name to LPI specific pinctrl: qcom: Update lpi pin group structure pinctrl: qcom: Extract chip specific LPASS LPI code pinctrl: qcom: Add SC7280 lpass pin configuration pinctrl: qcom: Update clock voting as optional Tested this on SM8250 MTP with WSA and WCD codecs. Tested-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 133 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 133 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 245 +-------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 86 ++++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 168 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 166 ++++++++++++++ 9 files changed, 696 insertions(+), 368 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- 2.7.4

1 7

[PATCH v7 7/7] pinctrl: qcom: Update clock voting as optional
by Srinivasa Rao Mandadapu 17 Feb '22

17 Feb '22

Update bulk clock voting to optional voting as ADSP bypass platform doesn't need macro and decodec clocks, these are maintained as power domains and operated from lpass audio core cc. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c index 8a82fd9..30a86c3 100644 --- a/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c +++ b/drivers/pinctrl/qcom/pinctrl-lpass-lpi.c @@ -407,7 +407,7 @@ int lpi_pinctrl_probe(struct platform_device *pdev) return dev_err_probe(dev, PTR_ERR(pctrl->slew_base), "Slew resource not provided\n"); - ret = devm_clk_bulk_get(dev, MAX_LPI_NUM_CLKS, pctrl->clks); + ret = devm_clk_bulk_get_optional(dev, MAX_LPI_NUM_CLKS, pctrl->clks); if (ret) return dev_err_probe(dev, ret, "Can't get clocks\n"); -- 2.7.4

1 0

[PATCH -next] ASoC: codec: wcd938x: Fix NULL but dereferenced coccicheck error
by Yang Li 17 Feb '22

17 Feb '22

Eliminate the following coccicheck warning: ./sound/soc/codecs/wcd938x.c:4210:21-24: ERROR: component is NULL but dereferenced. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Fixes: 013cc2aea0f6 ("ASoC: codec: wcd938x: Add switch control for selecting CTIA/OMTP Headset") Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/wcd938x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 2663fe0bf64f..c46c4bfd84f2 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -4207,7 +4207,7 @@ static bool wcd938x_swap_gnd_mic(struct snd_soc_component *component, bool activ struct wcd938x_priv *wcd938x; if (!component) { - dev_err(component->dev, "%s component is NULL\n", __func__); + pr_err("The snd_soc_component is NULL\n"); return false; } -- 2.20.1.7.g153144c

3 2

[PATCH][next] treewide: Replace zero-length arrays with flexible-array members
by Gustavo A. R. Silva 16 Feb '22

16 Feb '22

There is a regular need in the kernel to provide a way to declare having a dynamically sized set of trailing elements in a structure. Kernel code should always use “flexible array members”[1] for these cases. The older style of one-element or zero-length arrays should no longer be used[2]. This code was transformed with the help of Coccinelle: (next-20220214$ spatch --jobs $(getconf _NPROCESSORS_ONLN) --sp-file script.cocci --include-headers --dir . > output.patch) @@ identifier S, member, array; type T1, T2; @@ struct S { ... T1 member; T2 array[ - 0 ]; }; UAPI and wireless changes were intentionally excluded from this patch and will be sent out separately. [1] https://en.wikipedia.org/wiki/Flexible_array_member [2] https://www.kernel.org/doc/html/v5.16/process/deprecated.html#zero-length-a… Link: https://github.com/KSPP/linux/issues/78 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- Hi all, I'm expecting to carry this patch in my tree, so it'd be great to get some Acks. And given the size of the patch, I'm only sending this to mailing lists. Thanks! arch/alpha/include/asm/hwrpb.h | 2 +- arch/ia64/include/asm/sal.h | 2 +- arch/s390/include/asm/ccwgroup.h | 2 +- arch/s390/include/asm/chsc.h | 2 +- arch/s390/include/asm/eadm.h | 2 +- arch/s390/include/asm/fcx.h | 4 ++-- arch/s390/include/asm/idals.h | 2 +- arch/s390/include/asm/sclp.h | 2 +- arch/s390/include/asm/sysinfo.h | 6 +++--- arch/sh/include/asm/thread_info.h | 2 +- arch/sparc/include/asm/vio.h | 10 +++++----- arch/um/include/shared/net_kern.h | 2 +- arch/x86/include/asm/microcode_amd.h | 2 +- arch/x86/include/asm/microcode_intel.h | 4 ++-- arch/x86/include/asm/pci.h | 2 +- arch/x86/include/asm/pci_x86.h | 2 +- arch/xtensa/include/asm/bootparam.h | 2 +- drivers/crypto/caam/pdb.h | 2 +- drivers/gpu/drm/i915/gt/uc/intel_guc_ads.c | 2 +- drivers/gpu/drm/nouveau/include/nvfw/hs.h | 2 +- .../hwtracing/coresight/coresight-config.h | 2 +- drivers/misc/bcm-vk/bcm_vk.h | 2 +- .../misc/habanalabs/include/common/cpucp_if.h | 6 +++--- .../habanalabs/include/gaudi/gaudi_packets.h | 4 ++-- .../habanalabs/include/goya/goya_packets.h | 4 ++-- .../net/ethernet/freescale/enetc/enetc_hw.h | 2 +- drivers/net/ethernet/i825xx/sun3_82586.h | 2 +- .../net/ethernet/marvell/octeontx2/af/npc.h | 6 +++--- drivers/net/ethernet/qlogic/qed/qed_mfw_hsi.h | 2 +- drivers/net/ethernet/ti/davinci_mdio.c | 2 +- drivers/scsi/dpt/dpti_i2o.h | 2 +- drivers/scsi/elx/libefc_sli/sli4.h | 20 +++++++++---------- drivers/scsi/mpi3mr/mpi3mr.h | 2 +- drivers/scsi/qla2xxx/qla_bsg.h | 4 ++-- drivers/scsi/qla2xxx/qla_def.h | 2 +- drivers/scsi/qla2xxx/qla_edif_bsg.h | 4 ++-- drivers/scsi/qla2xxx/qla_fw.h | 2 +- drivers/scsi/qla4xxx/ql4_fw.h | 2 +- drivers/staging/r8188eu/include/ieee80211.h | 6 +++--- drivers/staging/r8188eu/include/rtw_cmd.h | 10 +++++----- drivers/staging/rtl8712/rtl871x_cmd.h | 8 ++++---- drivers/staging/rtl8723bs/include/ieee80211.h | 2 +- drivers/staging/rtl8723bs/include/rtw_cmd.h | 2 +- .../include/linux/raspberrypi/vchiq.h | 2 +- drivers/visorbus/vbuschannel.h | 2 +- fs/cifs/ntlmssp.h | 2 +- fs/ext4/fast_commit.h | 4 ++-- fs/ksmbd/ksmbd_netlink.h | 2 +- fs/ksmbd/ntlmssp.h | 6 +++--- fs/ksmbd/smb2pdu.h | 8 ++++---- fs/ksmbd/transport_rdma.c | 2 +- fs/ksmbd/xattr.h | 2 +- fs/xfs/scrub/attr.h | 2 +- include/acpi/actbl2.h | 2 +- include/asm-generic/tlb.h | 4 ++-- include/linux/greybus/greybus_manifest.h | 4 ++-- include/linux/greybus/hd.h | 2 +- include/linux/greybus/module.h | 2 +- include/linux/i3c/ccc.h | 6 +++--- include/linux/mlx5/mlx5_ifc_fpga.h | 2 +- include/linux/platform_data/brcmfmac.h | 2 +- .../linux/platform_data/cros_ec_commands.h | 2 +- include/net/bluetooth/mgmt.h | 2 +- include/net/ioam6.h | 2 +- include/sound/sof/channel_map.h | 4 ++-- scripts/dtc/libfdt/fdt.h | 4 ++-- sound/soc/intel/atom/sst-mfld-dsp.h | 4 ++-- sound/soc/intel/skylake/skl-topology.h | 2 +- tools/lib/perf/include/perf/event.h | 2 +- 69 files changed, 116 insertions(+), 116 deletions(-) diff --git a/arch/alpha/include/asm/hwrpb.h b/arch/alpha/include/asm/hwrpb.h index d8180e527a1e..fc76f36265ad 100644 --- a/arch/alpha/include/asm/hwrpb.h +++ b/arch/alpha/include/asm/hwrpb.h @@ -152,7 +152,7 @@ struct memdesc_struct { unsigned long chksum; unsigned long optional_pa; unsigned long numclusters; - struct memclust_struct cluster[0]; + struct memclust_struct cluster[]; }; struct dsr_struct { diff --git a/arch/ia64/include/asm/sal.h b/arch/ia64/include/asm/sal.h index 78f4f7b40435..22749a201e92 100644 --- a/arch/ia64/include/asm/sal.h +++ b/arch/ia64/include/asm/sal.h @@ -420,7 +420,7 @@ typedef struct sal_log_processor_info { * The rest of this structure consists of variable-length arrays, which can't be * expressed in C. */ - sal_log_mod_error_info_t info[0]; + sal_log_mod_error_info_t info[]; /* * This is what the rest looked like if C supported variable-length arrays: * diff --git a/arch/s390/include/asm/ccwgroup.h b/arch/s390/include/asm/ccwgroup.h index aa995d91cd1d..11d2fb3de4f5 100644 --- a/arch/s390/include/asm/ccwgroup.h +++ b/arch/s390/include/asm/ccwgroup.h @@ -25,7 +25,7 @@ struct ccwgroup_device { unsigned int count; struct device dev; struct work_struct ungroup_work; - struct ccw_device *cdev[0]; + struct ccw_device *cdev[]; }; /** diff --git a/arch/s390/include/asm/chsc.h b/arch/s390/include/asm/chsc.h index ae4d2549cd67..bb48ea380c0d 100644 --- a/arch/s390/include/asm/chsc.h +++ b/arch/s390/include/asm/chsc.h @@ -63,7 +63,7 @@ struct chsc_pnso_area { struct chsc_header response; u32:32; struct chsc_pnso_naihdr naihdr; - struct chsc_pnso_naid_l2 entries[0]; + struct chsc_pnso_naid_l2 entries[]; } __packed __aligned(PAGE_SIZE); #endif /* _ASM_S390_CHSC_H */ diff --git a/arch/s390/include/asm/eadm.h b/arch/s390/include/asm/eadm.h index 445fe4c8184a..06f795855af7 100644 --- a/arch/s390/include/asm/eadm.h +++ b/arch/s390/include/asm/eadm.h @@ -78,7 +78,7 @@ struct aob { struct aob_rq_header { struct scm_device *scmdev; - char data[0]; + char data[]; }; struct scm_device { diff --git a/arch/s390/include/asm/fcx.h b/arch/s390/include/asm/fcx.h index cff0749e9657..b8a028a36173 100644 --- a/arch/s390/include/asm/fcx.h +++ b/arch/s390/include/asm/fcx.h @@ -214,7 +214,7 @@ struct dcw_intrg_data { u32 :32; u64 time; u64 prog_id; - u8 prog_data[0]; + u8 prog_data[]; } __attribute__ ((packed)); #define DCW_FLAGS_CC (1 << (7 - 1)) @@ -241,7 +241,7 @@ struct dcw { u32 :8; u32 cd_count:8; u32 count; - u8 cd[0]; + u8 cd[]; } __attribute__ ((packed)); #define TCCB_FORMAT_DEFAULT 0x7f diff --git a/arch/s390/include/asm/idals.h b/arch/s390/include/asm/idals.h index 6fb7aced104a..40eae2c08d61 100644 --- a/arch/s390/include/asm/idals.h +++ b/arch/s390/include/asm/idals.h @@ -108,7 +108,7 @@ clear_normalized_cda(struct ccw1 * ccw) struct idal_buffer { size_t size; size_t page_order; - void *data[0]; + void *data[]; }; /* diff --git a/arch/s390/include/asm/sclp.h b/arch/s390/include/asm/sclp.h index c68ea35de498..22b3213c6c9d 100644 --- a/arch/s390/include/asm/sclp.h +++ b/arch/s390/include/asm/sclp.h @@ -112,7 +112,7 @@ struct zpci_report_error_header { * (OpenCrypto Successful Diagnostics Execution) */ u16 length; /* Length of Subsequent Data (up to 4K – SCLP header */ - u8 data[0]; /* Subsequent Data passed verbatim to SCLP ET 24 */ + u8 data[]; /* Subsequent Data passed verbatim to SCLP ET 24 */ } __packed; extern char *sclp_early_sccb; diff --git a/arch/s390/include/asm/sysinfo.h b/arch/s390/include/asm/sysinfo.h index fe7b3f8f0791..ab1c6316055c 100644 --- a/arch/s390/include/asm/sysinfo.h +++ b/arch/s390/include/asm/sysinfo.h @@ -67,12 +67,12 @@ struct sysinfo_1_2_2 { unsigned short cpus_configured; unsigned short cpus_standby; unsigned short cpus_reserved; - unsigned short adjustment[0]; + unsigned short adjustment[]; }; struct sysinfo_1_2_2_extension { unsigned int alt_capability; - unsigned short alt_adjustment[0]; + unsigned short alt_adjustment[]; }; struct sysinfo_2_2_1 { @@ -181,7 +181,7 @@ struct sysinfo_15_1_x { unsigned char reserved1; unsigned char mnest; unsigned char reserved2[4]; - union topology_entry tle[0]; + union topology_entry tle[]; }; int stsi(void *sysinfo, int fc, int sel1, int sel2); diff --git a/arch/sh/include/asm/thread_info.h b/arch/sh/include/asm/thread_info.h index 598d0184ffea..3a2d50d61fc9 100644 --- a/arch/sh/include/asm/thread_info.h +++ b/arch/sh/include/asm/thread_info.h @@ -33,7 +33,7 @@ struct thread_info { mm_segment_t addr_limit; /* thread address space */ unsigned long previous_sp; /* sp of previous stack in case of nested IRQ stacks */ - __u8 supervisor_stack[0]; + __u8 supervisor_stack[]; }; #endif diff --git a/arch/sparc/include/asm/vio.h b/arch/sparc/include/asm/vio.h index 8a1a83bbb6d5..2d7bdf665fd3 100644 --- a/arch/sparc/include/asm/vio.h +++ b/arch/sparc/include/asm/vio.h @@ -70,7 +70,7 @@ struct vio_dring_register { #define VIO_RX_DRING_DATA 0x0004 u16 resv; u32 num_cookies; - struct ldc_trans_cookie cookies[0]; + struct ldc_trans_cookie cookies[]; }; struct vio_dring_unregister { @@ -161,7 +161,7 @@ struct vio_disk_desc { u64 size; u32 ncookies; u32 resv2; - struct ldc_trans_cookie cookies[0]; + struct ldc_trans_cookie cookies[]; }; #define VIO_DISK_VNAME_LEN 8 @@ -200,13 +200,13 @@ struct vio_disk_devid { u16 resv; u16 type; u32 len; - char id[0]; + char id[]; }; struct vio_disk_efi { u64 lba; u64 len; - char data[0]; + char data[]; }; /* VIO net specific structures and defines */ @@ -246,7 +246,7 @@ struct vio_net_desc { struct vio_dring_hdr hdr; u32 size; u32 ncookies; - struct ldc_trans_cookie cookies[0]; + struct ldc_trans_cookie cookies[]; }; struct vio_net_dext { diff --git a/arch/um/include/shared/net_kern.h b/arch/um/include/shared/net_kern.h index 441a8a309329..67b2e9a1f2e5 100644 --- a/arch/um/include/shared/net_kern.h +++ b/arch/um/include/shared/net_kern.h @@ -39,7 +39,7 @@ struct uml_net_private { void (*add_address)(unsigned char *, unsigned char *, void *); void (*delete_address)(unsigned char *, unsigned char *, void *); - char user[0]; + char user[]; }; struct net_kern_info { diff --git a/arch/x86/include/asm/microcode_amd.h b/arch/x86/include/asm/microcode_amd.h index 7063b5a43220..ac31f9140d07 100644 --- a/arch/x86/include/asm/microcode_amd.h +++ b/arch/x86/include/asm/microcode_amd.h @@ -38,7 +38,7 @@ struct microcode_header_amd { struct microcode_amd { struct microcode_header_amd hdr; - unsigned int mpb[0]; + unsigned int mpb[]; }; #define PATCH_MAX_SIZE (3 * PAGE_SIZE) diff --git a/arch/x86/include/asm/microcode_intel.h b/arch/x86/include/asm/microcode_intel.h index d85a07d7154f..4c92cea7e4b5 100644 --- a/arch/x86/include/asm/microcode_intel.h +++ b/arch/x86/include/asm/microcode_intel.h @@ -19,7 +19,7 @@ struct microcode_header_intel { struct microcode_intel { struct microcode_header_intel hdr; - unsigned int bits[0]; + unsigned int bits[]; }; /* microcode format is extended from prescott processors */ @@ -33,7 +33,7 @@ struct extended_sigtable { unsigned int count; unsigned int cksum; unsigned int reserved[3]; - struct extended_signature sigs[0]; + struct extended_signature sigs[]; }; #define DEFAULT_UCODE_DATASIZE (2000) diff --git a/arch/x86/include/asm/pci.h b/arch/x86/include/asm/pci.h index d2c76c8d8cfd..f3fd5928bcbb 100644 --- a/arch/x86/include/asm/pci.h +++ b/arch/x86/include/asm/pci.h @@ -135,7 +135,7 @@ struct pci_setup_rom { unsigned long bus; unsigned long device; unsigned long function; - uint8_t romdata[0]; + uint8_t romdata[]; }; #endif /* _ASM_X86_PCI_H */ diff --git a/arch/x86/include/asm/pci_x86.h b/arch/x86/include/asm/pci_x86.h index 490411dba438..3fb6fc596095 100644 --- a/arch/x86/include/asm/pci_x86.h +++ b/arch/x86/include/asm/pci_x86.h @@ -87,7 +87,7 @@ struct irq_routing_table { u32 miniport_data; /* Crap */ u8 rfu[11]; u8 checksum; /* Modulo 256 checksum must give 0 */ - struct irq_info slots[0]; + struct irq_info slots[]; } __attribute__((packed)); extern unsigned int pcibios_irq_mask; diff --git a/arch/xtensa/include/asm/bootparam.h b/arch/xtensa/include/asm/bootparam.h index 892aab399ac8..6333bd1eb9d2 100644 --- a/arch/xtensa/include/asm/bootparam.h +++ b/arch/xtensa/include/asm/bootparam.h @@ -34,7 +34,7 @@ typedef struct bp_tag { unsigned short id; /* tag id */ unsigned short size; /* size of this record excluding the structure*/ - unsigned long data[0]; /* data */ + unsigned long data[]; /* data */ } bp_tag_t; struct bp_meminfo { diff --git a/drivers/crypto/caam/pdb.h b/drivers/crypto/caam/pdb.h index 8ccc22075043..4b1bcf53f7ac 100644 --- a/drivers/crypto/caam/pdb.h +++ b/drivers/crypto/caam/pdb.h @@ -144,7 +144,7 @@ struct ipsec_encap_pdb { }; u32 spi; u32 ip_hdr_len; - u32 ip_hdr[0]; + u32 ip_hdr[]; }; /** diff --git a/drivers/gpu/drm/i915/gt/uc/intel_guc_ads.c b/drivers/gpu/drm/i915/gt/uc/intel_guc_ads.c index 1a1edae67e4e..3acee0060e23 100644 --- a/drivers/gpu/drm/i915/gt/uc/intel_guc_ads.c +++ b/drivers/gpu/drm/i915/gt/uc/intel_guc_ads.c @@ -51,7 +51,7 @@ struct __guc_ads_blob { struct guc_gt_system_info system_info; struct guc_engine_usage engine_usage; /* From here on, location is dynamic! Refer to above diagram. */ - struct guc_mmio_reg regset[0]; + struct guc_mmio_reg regset[]; } __packed; static u32 guc_ads_regset_size(struct intel_guc *guc) diff --git a/drivers/gpu/drm/nouveau/include/nvfw/hs.h b/drivers/gpu/drm/nouveau/include/nvfw/hs.h index 64d0d32200c2..b53bbc4cd130 100644 --- a/drivers/gpu/drm/nouveau/include/nvfw/hs.h +++ b/drivers/gpu/drm/nouveau/include/nvfw/hs.h @@ -23,7 +23,7 @@ struct nvfw_hs_load_header { u32 data_dma_base; u32 data_size; u32 num_apps; - u32 apps[0]; + u32 apps[]; }; const struct nvfw_hs_load_header * diff --git a/drivers/hwtracing/coresight/coresight-config.h b/drivers/hwtracing/coresight/coresight-config.h index 9bd44b940add..2e1670523461 100644 --- a/drivers/hwtracing/coresight/coresight-config.h +++ b/drivers/hwtracing/coresight/coresight-config.h @@ -231,7 +231,7 @@ struct cscfg_config_csdev { bool enabled; struct list_head node; int nr_feat; - struct cscfg_feature_csdev *feats_csdev[0]; + struct cscfg_feature_csdev *feats_csdev[]; }; /** diff --git a/drivers/misc/bcm-vk/bcm_vk.h b/drivers/misc/bcm-vk/bcm_vk.h index a1338f375589..25d51222eedf 100644 --- a/drivers/misc/bcm-vk/bcm_vk.h +++ b/drivers/misc/bcm-vk/bcm_vk.h @@ -311,7 +311,7 @@ struct bcm_vk_peer_log { u32 wr_idx; u32 buf_size; u32 mask; - char data[0]; + char data[]; }; /* max buf size allowed */ diff --git a/drivers/misc/habanalabs/include/common/cpucp_if.h b/drivers/misc/habanalabs/include/common/cpucp_if.h index 737c39f33f05..f9c4acc9bf5a 100644 --- a/drivers/misc/habanalabs/include/common/cpucp_if.h +++ b/drivers/misc/habanalabs/include/common/cpucp_if.h @@ -540,19 +540,19 @@ struct cpucp_packet { struct cpucp_unmask_irq_arr_packet { struct cpucp_packet cpucp_pkt; __le32 length; - __le32 irqs[0]; + __le32 irqs[]; }; struct cpucp_nic_status_packet { struct cpucp_packet cpucp_pkt; __le32 length; - __le32 data[0]; + __le32 data[]; }; struct cpucp_array_data_packet { struct cpucp_packet cpucp_pkt; __le32 length; - __le32 data[0]; + __le32 data[]; }; enum cpucp_packet_rc { diff --git a/drivers/misc/habanalabs/include/gaudi/gaudi_packets.h b/drivers/misc/habanalabs/include/gaudi/gaudi_packets.h index 6e097ace2e96..66fc083a7c6a 100644 --- a/drivers/misc/habanalabs/include/gaudi/gaudi_packets.h +++ b/drivers/misc/habanalabs/include/gaudi/gaudi_packets.h @@ -54,7 +54,7 @@ struct gaudi_packet { /* The rest of the packet data follows. Use the corresponding * packet_XXX struct to deference the data, based on packet type */ - u8 contents[0]; + u8 contents[]; }; struct packet_nop { @@ -75,7 +75,7 @@ struct packet_wreg32 { struct packet_wreg_bulk { __le32 size64; __le32 ctl; - __le64 values[0]; /* data starts here */ + __le64 values[]; /* data starts here */ }; #define GAUDI_PKT_LONG_CTL_OP_SHIFT 20 diff --git a/drivers/misc/habanalabs/include/goya/goya_packets.h b/drivers/misc/habanalabs/include/goya/goya_packets.h index ef54bad20509..50ce5175b63a 100644 --- a/drivers/misc/habanalabs/include/goya/goya_packets.h +++ b/drivers/misc/habanalabs/include/goya/goya_packets.h @@ -62,7 +62,7 @@ struct goya_packet { /* The rest of the packet data follows. Use the corresponding * packet_XXX struct to deference the data, based on packet type */ - u8 contents[0]; + u8 contents[]; }; struct packet_nop { @@ -86,7 +86,7 @@ struct packet_wreg32 { struct packet_wreg_bulk { __le32 size64; __le32 ctl; - __le64 values[0]; /* data starts here */ + __le64 values[]; /* data starts here */ }; struct packet_msg_long { diff --git a/drivers/net/ethernet/freescale/enetc/enetc_hw.h b/drivers/net/ethernet/freescale/enetc/enetc_hw.h index 1514e6a4a3ff..ce5b677e8c2f 100644 --- a/drivers/net/ethernet/freescale/enetc/enetc_hw.h +++ b/drivers/net/ethernet/freescale/enetc/enetc_hw.h @@ -881,7 +881,7 @@ struct sgcl_data { u32 bth; u32 ct; u32 cte; - struct sgce sgcl[0]; + struct sgce sgcl[]; }; #define ENETC_CBDR_FMI_MR BIT(0) diff --git a/drivers/net/ethernet/i825xx/sun3_82586.h b/drivers/net/ethernet/i825xx/sun3_82586.h index 79aef681ac85..451cb3d26cb5 100644 --- a/drivers/net/ethernet/i825xx/sun3_82586.h +++ b/drivers/net/ethernet/i825xx/sun3_82586.h @@ -250,7 +250,7 @@ struct mcsetup_cmd_struct unsigned short cmd_cmd; unsigned short cmd_link; unsigned short mc_cnt; /* number of bytes in the MC-List */ - unsigned char mc_list[0][6]; /* pointer to 6 bytes entries */ + unsigned char mc_list[][6]; /* pointer to 6 bytes entries */ }; /* diff --git a/drivers/net/ethernet/marvell/octeontx2/af/npc.h b/drivers/net/ethernet/marvell/octeontx2/af/npc.h index 77fd39e2c8db..9b6e587e78b4 100644 --- a/drivers/net/ethernet/marvell/octeontx2/af/npc.h +++ b/drivers/net/ethernet/marvell/octeontx2/af/npc.h @@ -455,7 +455,7 @@ struct npc_coalesced_kpu_prfl { u8 name[NPC_NAME_LEN]; /* KPU Profile name */ u64 version; /* KPU firmware/profile version */ u8 num_prfl; /* No of NPC profiles. */ - u16 prfl_sz[0]; + u16 prfl_sz[]; }; struct npc_mcam_kex { @@ -482,7 +482,7 @@ struct npc_kpu_fwdata { * struct npc_kpu_profile_cam[entries]; * struct npc_kpu_profile_action[entries]; */ - u8 data[0]; + u8 data[]; } __packed; struct npc_lt_def { @@ -572,7 +572,7 @@ struct npc_kpu_profile_fwdata { * Custom KPU CAM and ACTION configuration entries. * struct npc_kpu_fwdata kpu[kpus]; */ - u8 data[0]; + u8 data[]; } __packed; struct rvu_npc_mcam_rule { diff --git a/drivers/net/ethernet/qlogic/qed/qed_mfw_hsi.h b/drivers/net/ethernet/qlogic/qed/qed_mfw_hsi.h index b70ee8200e15..6459dd3feb37 100644 --- a/drivers/net/ethernet/qlogic/qed/qed_mfw_hsi.h +++ b/drivers/net/ethernet/qlogic/qed/qed_mfw_hsi.h @@ -2470,6 +2470,6 @@ struct nvm_meta_bin_t { u32 version; #define NVM_META_BIN_VERSION 1 u32 num_options; - u32 options[0]; + u32 options[]; }; #endif diff --git a/drivers/net/ethernet/ti/davinci_mdio.c b/drivers/net/ethernet/ti/davinci_mdio.c index a4efd5e35158..fce2626e34fa 100644 --- a/drivers/net/ethernet/ti/davinci_mdio.c +++ b/drivers/net/ethernet/ti/davinci_mdio.c @@ -70,7 +70,7 @@ struct davinci_mdio_regs { #define USERACCESS_DATA (0xffff) u32 physel; - } user[0]; + } user[]; }; static const struct mdio_platform_data default_pdata = { diff --git a/drivers/scsi/dpt/dpti_i2o.h b/drivers/scsi/dpt/dpti_i2o.h index bf0daeeb50a9..e1fbbf55c09d 100644 --- a/drivers/scsi/dpt/dpti_i2o.h +++ b/drivers/scsi/dpt/dpti_i2o.h @@ -123,7 +123,7 @@ struct i2o_sys_tbl u32 change_ind; u32 reserved2; u32 reserved3; - struct i2o_sys_tbl_entry iops[0]; + struct i2o_sys_tbl_entry iops[]; }; /* diff --git a/drivers/scsi/elx/libefc_sli/sli4.h b/drivers/scsi/elx/libefc_sli/sli4.h index ee2a9e65a88d..38af166cc786 100644 --- a/drivers/scsi/elx/libefc_sli/sli4.h +++ b/drivers/scsi/elx/libefc_sli/sli4.h @@ -609,7 +609,7 @@ struct sli4_rqst_cmn_create_cq_v2 { __le16 cqe_count; __le16 rsvd30; __le32 rsvd32; - struct sli4_dmaaddr page_phys_addr[0]; + struct sli4_dmaaddr page_phys_addr[]; }; enum sli4_create_cqset_e { @@ -634,7 +634,7 @@ struct sli4_rqst_cmn_create_cq_set_v0 { __le16 num_cq_req; __le16 dw6w1_flags; __le16 eq_id[16]; - struct sli4_dmaaddr page_phys_addr[0]; + struct sli4_dmaaddr page_phys_addr[]; }; /* CQE count */ @@ -764,7 +764,7 @@ struct sli4_rqst_cmn_create_mq_ext { __le32 dw7_val; __le32 dw8_flags; __le32 rsvd36; - struct sli4_dmaaddr page_phys_addr[0]; + struct sli4_dmaaddr page_phys_addr[]; }; struct sli4_rsp_cmn_create_mq_ext { @@ -802,7 +802,7 @@ struct sli4_rqst_cmn_create_cq_v0 { __le32 dw6_flags; __le32 rsvd28; __le32 rsvd32; - struct sli4_dmaaddr page_phys_addr[0]; + struct sli4_dmaaddr page_phys_addr[]; }; enum sli4_create_rq_e { @@ -887,7 +887,7 @@ struct sli4_rqst_rq_create_v2 { __le16 base_cq_id; __le16 rsvd26; __le32 rsvd42; - struct sli4_dmaaddr page_phys_addr[0]; + struct sli4_dmaaddr page_phys_addr[]; }; struct sli4_rsp_rq_create_v2 { @@ -3168,7 +3168,7 @@ struct sli4_rqst_cmn_read_object { __le32 read_offset; u8 object_name[104]; __le32 host_buffer_descriptor_count; - struct sli4_bde host_buffer_descriptor[0]; + struct sli4_bde host_buffer_descriptor[]; }; #define RSP_COM_READ_OBJ_EOF 0x80000000 @@ -3191,7 +3191,7 @@ struct sli4_rqst_cmn_write_object { __le32 write_offset; u8 object_name[104]; __le32 host_buffer_descriptor_count; - struct sli4_bde host_buffer_descriptor[0]; + struct sli4_bde host_buffer_descriptor[]; }; #define RSP_CHANGE_STATUS 0xff @@ -3217,7 +3217,7 @@ struct sli4_rqst_cmn_read_object_list { __le32 read_offset; u8 object_name[104]; __le32 host_buffer_descriptor_count; - struct sli4_bde host_buffer_descriptor[0]; + struct sli4_bde host_buffer_descriptor[]; }; enum sli4_rqst_set_dump_flags { @@ -3342,7 +3342,7 @@ struct sli4_rspource_descriptor_v1 { u8 descriptor_type; u8 descriptor_length; __le16 rsvd16; - __le32 type_specific[0]; + __le32 type_specific[]; }; enum sli4_pcie_desc_flags { @@ -3474,7 +3474,7 @@ struct sli4_rqst_post_hdr_templates { struct sli4_rqst_hdr hdr; __le16 rpi_offset; __le16 page_count; - struct sli4_dmaaddr page_descriptor[0]; + struct sli4_dmaaddr page_descriptor[]; }; #define SLI4_HDR_TEMPLATE_SIZE 64 diff --git a/drivers/scsi/mpi3mr/mpi3mr.h b/drivers/scsi/mpi3mr/mpi3mr.h index fc4eaf6d1e47..fb7d8775c9f7 100644 --- a/drivers/scsi/mpi3mr/mpi3mr.h +++ b/drivers/scsi/mpi3mr/mpi3mr.h @@ -878,7 +878,7 @@ struct mpi3mr_fwevt { bool process_evt; u32 evt_ctx; struct kref ref_count; - char event_data[0] __aligned(4); + char event_data[] __aligned(4); }; diff --git a/drivers/scsi/qla2xxx/qla_bsg.h b/drivers/scsi/qla2xxx/qla_bsg.h index 0f8a4c7e52a2..6d2b0a7436c1 100644 --- a/drivers/scsi/qla2xxx/qla_bsg.h +++ b/drivers/scsi/qla2xxx/qla_bsg.h @@ -157,7 +157,7 @@ struct qla84_msg_mgmt { uint16_t rsrvd; struct qla84_mgmt_param mgmtp;/* parameters for cmd */ uint32_t len; /* bytes in payload following this struct */ - uint8_t payload[0]; /* payload for cmd */ + uint8_t payload[]; /* payload for cmd */ }; struct qla_bsg_a84_mgmt { @@ -216,7 +216,7 @@ struct qla_image_version { struct qla_image_version_list { uint32_t count; - struct qla_image_version version[0]; + struct qla_image_version version[]; } __packed; struct qla_status_reg { diff --git a/drivers/scsi/qla2xxx/qla_def.h b/drivers/scsi/qla2xxx/qla_def.h index 9ebf4a234d9a..b6434c72dee3 100644 --- a/drivers/scsi/qla2xxx/qla_def.h +++ b/drivers/scsi/qla2xxx/qla_def.h @@ -5410,7 +5410,7 @@ struct ql_vnd_stat_entry { struct ql_vnd_stats { u64 entry_count; /* Num of entries */ u64 rservd; - struct ql_vnd_stat_entry entry[0]; /* Place holder of entries */ + struct ql_vnd_stat_entry entry[]; /* Place holder of entries */ } __packed; struct ql_vnd_host_stats_resp { diff --git a/drivers/scsi/qla2xxx/qla_edif_bsg.h b/drivers/scsi/qla2xxx/qla_edif_bsg.h index 53026d82ebff..5a26c77157da 100644 --- a/drivers/scsi/qla2xxx/qla_edif_bsg.h +++ b/drivers/scsi/qla2xxx/qla_edif_bsg.h @@ -121,7 +121,7 @@ struct app_pinfo { struct app_pinfo_reply { uint8_t port_count; uint8_t reserved[VND_CMD_APP_RESERVED_SIZE]; - struct app_pinfo ports[0]; + struct app_pinfo ports[]; } __packed; struct app_sinfo_req { @@ -140,7 +140,7 @@ struct app_sinfo { struct app_stats_reply { uint8_t elem_count; - struct app_sinfo elem[0]; + struct app_sinfo elem[]; } __packed; struct qla_sa_update_frame { diff --git a/drivers/scsi/qla2xxx/qla_fw.h b/drivers/scsi/qla2xxx/qla_fw.h index 073d06e88c58..0bb1d562f0bf 100644 --- a/drivers/scsi/qla2xxx/qla_fw.h +++ b/drivers/scsi/qla2xxx/qla_fw.h @@ -1706,7 +1706,7 @@ struct qla_flt_header { __le16 length; __le16 checksum; __le16 unused; - struct qla_flt_region region[0]; + struct qla_flt_region region[]; }; #define FLT_REGION_SIZE 16 diff --git a/drivers/scsi/qla4xxx/ql4_fw.h b/drivers/scsi/qla4xxx/ql4_fw.h index 4e1764df0a73..860ec61b51b9 100644 --- a/drivers/scsi/qla4xxx/ql4_fw.h +++ b/drivers/scsi/qla4xxx/ql4_fw.h @@ -1028,7 +1028,7 @@ struct crash_record { uint8_t out_RISC_reg_dump[256]; /* 80 -17F */ uint8_t in_RISC_reg_dump[256]; /*180 -27F */ - uint8_t in_out_RISC_stack_dump[0]; /*280 - ??? */ + uint8_t in_out_RISC_stack_dump[]; /*280 - ??? */ }; struct conn_event_log_entry { diff --git a/drivers/staging/r8188eu/include/ieee80211.h b/drivers/staging/r8188eu/include/ieee80211.h index 3a23d5299314..5e7de631b8ef 100644 --- a/drivers/staging/r8188eu/include/ieee80211.h +++ b/drivers/staging/r8188eu/include/ieee80211.h @@ -185,7 +185,7 @@ struct ieee_param { struct ieee_param_ex { u32 cmd; u8 sta_addr[ETH_ALEN]; - u8 data[0]; + u8 data[]; }; struct sta_data { @@ -713,7 +713,7 @@ struct ieee80211_info_element_hdr { struct ieee80211_info_element { u8 id; u8 len; - u8 data[0]; + u8 data[]; } __packed; /* @@ -776,7 +776,7 @@ struct ieee80211_txb { u16 reserved; u16 frag_size; u16 payload_size; - struct sk_buff *fragments[0]; + struct sk_buff *fragments[]; }; /* SWEEP TABLE ENTRIES NUMBER*/ diff --git a/drivers/staging/r8188eu/include/rtw_cmd.h b/drivers/staging/r8188eu/include/rtw_cmd.h index cf0945ae11c1..f8991a0493d0 100644 --- a/drivers/staging/r8188eu/include/rtw_cmd.h +++ b/drivers/staging/r8188eu/include/rtw_cmd.h @@ -73,7 +73,7 @@ struct c2h_evt_hdr { u8 id:4; u8 plen:4; u8 seq; - u8 payload[0]; + u8 payload[]; }; #define c2h_evt_exist(c2h_evt) ((c2h_evt)->id || (c2h_evt)->plen) @@ -662,25 +662,25 @@ struct getcurtxpwrlevel_rspi { struct setprobereqextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct setassocreqextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct setproberspextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct setassocrspextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct addBaReq_parm { diff --git a/drivers/staging/rtl8712/rtl871x_cmd.h b/drivers/staging/rtl8712/rtl871x_cmd.h index ddd69c4ae208..95e9ea5b2d98 100644 --- a/drivers/staging/rtl8712/rtl871x_cmd.h +++ b/drivers/staging/rtl8712/rtl871x_cmd.h @@ -657,25 +657,25 @@ struct setra_parm { struct setprobereqextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct setassocreqextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct setproberspextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct setassocrspextraie_parm { unsigned char e_id; unsigned char ie_len; - unsigned char ie[0]; + unsigned char ie[]; }; struct addBaReq_parm { diff --git a/drivers/staging/rtl8723bs/include/ieee80211.h b/drivers/staging/rtl8723bs/include/ieee80211.h index c11d7e2d2347..1e627dc0044d 100644 --- a/drivers/staging/rtl8723bs/include/ieee80211.h +++ b/drivers/staging/rtl8723bs/include/ieee80211.h @@ -204,7 +204,7 @@ struct ieee_param { struct ieee_param_ex { u32 cmd; u8 sta_addr[ETH_ALEN]; - u8 data[0]; + u8 data[]; }; struct sta_data { diff --git a/drivers/staging/rtl8723bs/include/rtw_cmd.h b/drivers/staging/rtl8723bs/include/rtw_cmd.h index 28d2d2732374..1bf030cbbbbe 100644 --- a/drivers/staging/rtl8723bs/include/rtw_cmd.h +++ b/drivers/staging/rtl8723bs/include/rtw_cmd.h @@ -94,7 +94,7 @@ struct c2h_evt_hdr { u8 id:4; u8 plen:4; u8 seq; - u8 payload[0]; + u8 payload[]; }; struct c2h_evt_hdr_88xx { diff --git a/drivers/staging/vc04_services/include/linux/raspberrypi/vchiq.h b/drivers/staging/vc04_services/include/linux/raspberrypi/vchiq.h index 81db7fb76d6d..c93f2f3e87bb 100644 --- a/drivers/staging/vc04_services/include/linux/raspberrypi/vchiq.h +++ b/drivers/staging/vc04_services/include/linux/raspberrypi/vchiq.h @@ -45,7 +45,7 @@ struct vchiq_header { /* Size of message data. */ unsigned int size; - char data[0]; /* message */ + char data[]; /* message */ }; struct vchiq_element { diff --git a/drivers/visorbus/vbuschannel.h b/drivers/visorbus/vbuschannel.h index 4aaf6564eb9f..98711fb6d66e 100644 --- a/drivers/visorbus/vbuschannel.h +++ b/drivers/visorbus/vbuschannel.h @@ -89,7 +89,7 @@ struct visor_vbus_channel { struct visor_vbus_headerinfo hdr_info; struct visor_vbus_deviceinfo chp_info; struct visor_vbus_deviceinfo bus_info; - struct visor_vbus_deviceinfo dev_info[0]; + struct visor_vbus_deviceinfo dev_info[]; } __packed; #endif diff --git a/fs/cifs/ntlmssp.h b/fs/cifs/ntlmssp.h index 298458404252..55758b9ec877 100644 --- a/fs/cifs/ntlmssp.h +++ b/fs/cifs/ntlmssp.h @@ -107,7 +107,7 @@ struct negotiate_message { SECURITY_BUFFER WorkstationName; /* RFC 1001 and ASCII */ struct ntlmssp_version Version; /* SECURITY_BUFFER */ - char DomainString[0]; + char DomainString[]; /* followed by WorkstationString */ } __packed; diff --git a/fs/ext4/fast_commit.h b/fs/ext4/fast_commit.h index 083ad1cb705a..07e8b734c4fd 100644 --- a/fs/ext4/fast_commit.h +++ b/fs/ext4/fast_commit.h @@ -55,13 +55,13 @@ struct ext4_fc_del_range { struct ext4_fc_dentry_info { __le32 fc_parent_ino; __le32 fc_ino; - __u8 fc_dname[0]; + __u8 fc_dname[]; }; /* Value structure for EXT4_FC_TAG_INODE and EXT4_FC_TAG_INODE_PARTIAL. */ struct ext4_fc_inode { __le32 fc_ino; - __u8 fc_raw_inode[0]; + __u8 fc_raw_inode[]; }; /* Value structure for tag EXT4_FC_TAG_TAIL. */ diff --git a/fs/ksmbd/ksmbd_netlink.h b/fs/ksmbd/ksmbd_netlink.h index 71bfb7de4472..ebe6ca08467a 100644 --- a/fs/ksmbd/ksmbd_netlink.h +++ b/fs/ksmbd/ksmbd_netlink.h @@ -241,7 +241,7 @@ struct ksmbd_rpc_command { struct ksmbd_spnego_authen_request { __u32 handle; __u16 spnego_blob_len; /* the length of spnego_blob */ - __u8 spnego_blob[0]; /* + __u8 spnego_blob[]; /* * the GSS token from SecurityBuffer of * SMB2 SESSION SETUP request */ diff --git a/fs/ksmbd/ntlmssp.h b/fs/ksmbd/ntlmssp.h index adaf4c0cbe8f..f13153c18b4e 100644 --- a/fs/ksmbd/ntlmssp.h +++ b/fs/ksmbd/ntlmssp.h @@ -95,7 +95,7 @@ struct security_buffer { struct target_info { __le16 Type; __le16 Length; - __u8 Content[0]; + __u8 Content[]; } __packed; struct negotiate_message { @@ -108,7 +108,7 @@ struct negotiate_message { * struct security_buffer for version info not present since we * do not set the version is present flag */ - char DomainString[0]; + char DomainString[]; /* followed by WorkstationString */ } __packed; @@ -140,7 +140,7 @@ struct authenticate_message { * struct security_buffer for version info not present since we * do not set the version is present flag */ - char UserString[0]; + char UserString[]; } __packed; struct ntlmv2_resp { diff --git a/fs/ksmbd/smb2pdu.h b/fs/ksmbd/smb2pdu.h index 725b800c29c8..d49468426576 100644 --- a/fs/ksmbd/smb2pdu.h +++ b/fs/ksmbd/smb2pdu.h @@ -759,7 +759,7 @@ struct smb2_file_rename_info { /* encoding of request for level 10 */ __u8 Reserved[7]; __u64 RootDirectory; /* MBZ for network operations (why says spec?) */ __le32 FileNameLength; - char FileName[0]; /* New name to be assigned */ + char FileName[]; /* New name to be assigned */ } __packed; /* level 10 Set */ struct smb2_file_link_info { /* encoding of request for level 11 */ @@ -768,7 +768,7 @@ struct smb2_file_link_info { /* encoding of request for level 11 */ __u8 Reserved[7]; __u64 RootDirectory; /* MBZ for network operations (why says spec?) */ __le32 FileNameLength; - char FileName[0]; /* Name to be assigned to new link */ + char FileName[]; /* Name to be assigned to new link */ } __packed; /* level 11 Set */ /* @@ -810,7 +810,7 @@ struct smb2_file_basic_info { /* data block encoding of response to level 18 */ struct smb2_file_alt_name_info { __le32 FileNameLength; - char FileName[0]; + char FileName[]; } __packed; struct smb2_file_stream_info { @@ -818,7 +818,7 @@ struct smb2_file_stream_info { __le32 StreamNameLength; __le64 StreamSize; __le64 StreamAllocationSize; - char StreamName[0]; + char StreamName[]; } __packed; struct smb2_file_eof_info { /* encoding of request for level 10 */ diff --git a/fs/ksmbd/transport_rdma.c b/fs/ksmbd/transport_rdma.c index 3c1ec1ac0b27..9976d39c6ed8 100644 --- a/fs/ksmbd/transport_rdma.c +++ b/fs/ksmbd/transport_rdma.c @@ -211,7 +211,7 @@ struct smb_direct_rdma_rw_msg { struct completion *completion; struct rdma_rw_ctx rw_ctx; struct sg_table sgt; - struct scatterlist sg_list[0]; + struct scatterlist sg_list[]; }; static inline int get_buf_page_count(void *buf, int size) diff --git a/fs/ksmbd/xattr.h b/fs/ksmbd/xattr.h index 8857c01093d9..16499ca5c82d 100644 --- a/fs/ksmbd/xattr.h +++ b/fs/ksmbd/xattr.h @@ -76,7 +76,7 @@ struct xattr_acl_entry { struct xattr_smb_acl { int count; int next; - struct xattr_acl_entry entries[0]; + struct xattr_acl_entry entries[]; }; /* 64bytes hash in xattr_ntacl is computed with sha256 */ diff --git a/fs/xfs/scrub/attr.h b/fs/xfs/scrub/attr.h index 1719e1c4da59..3590e10e3e62 100644 --- a/fs/xfs/scrub/attr.h +++ b/fs/xfs/scrub/attr.h @@ -24,7 +24,7 @@ struct xchk_xattr_buf { * space bitmap follows immediately after; and we have a third buffer * for storing intermediate bitmap results. */ - uint8_t buf[0]; + uint8_t buf[]; }; /* A place to store attribute values. */ diff --git a/include/acpi/actbl2.h b/include/acpi/actbl2.h index 16847c8d9d5f..9ee4d1b39125 100644 --- a/include/acpi/actbl2.h +++ b/include/acpi/actbl2.h @@ -2319,7 +2319,7 @@ struct acpi_table_rgrt { u16 version; u8 image_type; u8 reserved; - u8 image[0]; + u8 image[]; }; /* image_type values */ diff --git a/include/asm-generic/tlb.h b/include/asm-generic/tlb.h index 2c68a545ffa7..fd7feb5c7894 100644 --- a/include/asm-generic/tlb.h +++ b/include/asm-generic/tlb.h @@ -180,7 +180,7 @@ struct mmu_table_batch { struct rcu_head rcu; #endif unsigned int nr; - void *tables[0]; + void *tables[]; }; #define MAX_TABLE_BATCH \ @@ -227,7 +227,7 @@ struct mmu_gather_batch { struct mmu_gather_batch *next; unsigned int nr; unsigned int max; - struct page *pages[0]; + struct page *pages[]; }; #define MAX_GATHER_BATCH \ diff --git a/include/linux/greybus/greybus_manifest.h b/include/linux/greybus/greybus_manifest.h index 6e62fe478712..bef9eb2093e9 100644 --- a/include/linux/greybus/greybus_manifest.h +++ b/include/linux/greybus/greybus_manifest.h @@ -100,7 +100,7 @@ enum { struct greybus_descriptor_string { __u8 length; __u8 id; - __u8 string[0]; + __u8 string[]; } __packed; /* @@ -175,7 +175,7 @@ struct greybus_manifest_header { struct greybus_manifest { struct greybus_manifest_header header; - struct greybus_descriptor descriptors[0]; + struct greybus_descriptor descriptors[]; } __packed; #endif /* __GREYBUS_MANIFEST_H */ diff --git a/include/linux/greybus/hd.h b/include/linux/greybus/hd.h index d3faf0c1a569..718e2857054e 100644 --- a/include/linux/greybus/hd.h +++ b/include/linux/greybus/hd.h @@ -58,7 +58,7 @@ struct gb_host_device { struct gb_svc *svc; /* Private data for the host driver */ - unsigned long hd_priv[0] __aligned(sizeof(s64)); + unsigned long hd_priv[] __aligned(sizeof(s64)); }; #define to_gb_host_device(d) container_of(d, struct gb_host_device, dev) diff --git a/include/linux/greybus/module.h b/include/linux/greybus/module.h index 47b839af145d..3efe2133acfd 100644 --- a/include/linux/greybus/module.h +++ b/include/linux/greybus/module.h @@ -23,7 +23,7 @@ struct gb_module { bool disconnected; - struct gb_interface *interfaces[0]; + struct gb_interface *interfaces[]; }; #define to_gb_module(d) container_of(d, struct gb_module, dev) diff --git a/include/linux/i3c/ccc.h b/include/linux/i3c/ccc.h index 73b0982cc519..ad59a4ae60d1 100644 --- a/include/linux/i3c/ccc.h +++ b/include/linux/i3c/ccc.h @@ -132,7 +132,7 @@ struct i3c_ccc_dev_desc { struct i3c_ccc_defslvs { u8 count; struct i3c_ccc_dev_desc master; - struct i3c_ccc_dev_desc slaves[0]; + struct i3c_ccc_dev_desc slaves[]; } __packed; /** @@ -240,7 +240,7 @@ struct i3c_ccc_bridged_slave_desc { */ struct i3c_ccc_setbrgtgt { u8 count; - struct i3c_ccc_bridged_slave_desc bslaves[0]; + struct i3c_ccc_bridged_slave_desc bslaves[]; } __packed; /** @@ -318,7 +318,7 @@ enum i3c_ccc_setxtime_subcmd { */ struct i3c_ccc_setxtime { u8 subcmd; - u8 data[0]; + u8 data[]; } __packed; #define I3C_CCC_GETXTIME_SYNC_MODE BIT(0) diff --git a/include/linux/mlx5/mlx5_ifc_fpga.h b/include/linux/mlx5/mlx5_ifc_fpga.h index 07d77323f78a..cac4e94abd3a 100644 --- a/include/linux/mlx5/mlx5_ifc_fpga.h +++ b/include/linux/mlx5/mlx5_ifc_fpga.h @@ -185,7 +185,7 @@ struct mlx5_ifc_fpga_access_reg_bits { u8 address[0x40]; - u8 data[0][0x8]; + u8 data[][0x8]; }; enum mlx5_ifc_fpga_qp_state { diff --git a/include/linux/platform_data/brcmfmac.h b/include/linux/platform_data/brcmfmac.h index 2b5676ff35be..f922a192fe58 100644 --- a/include/linux/platform_data/brcmfmac.h +++ b/include/linux/platform_data/brcmfmac.h @@ -178,7 +178,7 @@ struct brcmfmac_platform_data { void (*power_off)(void); char *fw_alternative_path; int device_count; - struct brcmfmac_pd_device devices[0]; + struct brcmfmac_pd_device devices[]; }; diff --git a/include/linux/platform_data/cros_ec_commands.h b/include/linux/platform_data/cros_ec_commands.h index 271bd87bff0a..728735aed980 100644 --- a/include/linux/platform_data/cros_ec_commands.h +++ b/include/linux/platform_data/cros_ec_commands.h @@ -5644,7 +5644,7 @@ struct ec_response_typec_discovery { uint8_t svid_count; /* Number of SVIDs partner sent */ uint16_t reserved; uint32_t discovery_vdo[6]; /* Max VDOs allowed after VDM header is 6 */ - struct svid_mode_info svids[0]; + struct svid_mode_info svids[]; } __ec_align1; /* USB Type-C commands for AP-controlled device policy. */ diff --git a/include/net/bluetooth/mgmt.h b/include/net/bluetooth/mgmt.h index 107b25deae68..9607ec289fd0 100644 --- a/include/net/bluetooth/mgmt.h +++ b/include/net/bluetooth/mgmt.h @@ -696,7 +696,7 @@ struct mgmt_cp_set_blocked_keys { #define MGMT_READ_CONTROLLER_CAP_SIZE 0 struct mgmt_rp_read_controller_cap { __le16 cap_len; - __u8 cap[0]; + __u8 cap[]; } __packed; #define MGMT_OP_READ_EXP_FEATURES_INFO 0x0049 diff --git a/include/net/ioam6.h b/include/net/ioam6.h index 3f45ba37a2c6..781d2d8b2f29 100644 --- a/include/net/ioam6.h +++ b/include/net/ioam6.h @@ -35,7 +35,7 @@ struct ioam6_schema { int len; __be32 hdr; - u8 data[0]; + u8 data[]; }; struct ioam6_pernet_data { diff --git a/include/sound/sof/channel_map.h b/include/sound/sof/channel_map.h index fd3a30fcf756..d363f0ca6979 100644 --- a/include/sound/sof/channel_map.h +++ b/include/sound/sof/channel_map.h @@ -39,7 +39,7 @@ struct sof_ipc_channel_map { uint32_t ext_id; uint32_t ch_mask; uint32_t reserved; - int32_t ch_coeffs[0]; + int32_t ch_coeffs[]; } __packed; /** @@ -55,7 +55,7 @@ struct sof_ipc_stream_map { struct sof_ipc_cmd_hdr hdr; uint32_t num_ch_map; uint32_t reserved[3]; - struct sof_ipc_channel_map ch_map[0]; + struct sof_ipc_channel_map ch_map[]; } __packed; #endif /* __IPC_CHANNEL_MAP_H__ */ diff --git a/scripts/dtc/libfdt/fdt.h b/scripts/dtc/libfdt/fdt.h index f2e68807f277..0c91aa7f67b5 100644 --- a/scripts/dtc/libfdt/fdt.h +++ b/scripts/dtc/libfdt/fdt.h @@ -35,14 +35,14 @@ struct fdt_reserve_entry { struct fdt_node_header { fdt32_t tag; - char name[0]; + char name[]; }; struct fdt_property { fdt32_t tag; fdt32_t len; fdt32_t nameoff; - char data[0]; + char data[]; }; #endif /* !__ASSEMBLY */ diff --git a/sound/soc/intel/atom/sst-mfld-dsp.h b/sound/soc/intel/atom/sst-mfld-dsp.h index 8d9e29b16e57..c8f0816edb53 100644 --- a/sound/soc/intel/atom/sst-mfld-dsp.h +++ b/sound/soc/intel/atom/sst-mfld-dsp.h @@ -427,7 +427,7 @@ struct snd_sst_drop_response { struct snd_sst_async_msg { u32 msg_id; /* Async msg id */ - u32 payload[0]; + u32 payload[]; }; struct snd_sst_async_err_msg { @@ -514,7 +514,7 @@ struct snd_sst_bytes_v2 { u8 pipe_id; u8 rsvd; u16 len; - char bytes[0]; + char bytes[]; }; #define MAX_VTSV_FILES 2 diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index 22963634fbea..a5bccf2fcd88 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -164,7 +164,7 @@ struct skl_base_cfg_ext { u8 reserved[8]; u32 priv_param_length; /* Input pin formats followed by output ones. */ - struct skl_pin_format pins_fmt[0]; + struct skl_pin_format pins_fmt[]; } __packed; struct skl_algo_cfg { diff --git a/tools/lib/perf/include/perf/event.h b/tools/lib/perf/include/perf/event.h index 75ee385fb078..e7758707cadd 100644 --- a/tools/lib/perf/include/perf/event.h +++ b/tools/lib/perf/include/perf/event.h @@ -240,7 +240,7 @@ struct id_index_entry { struct perf_record_id_index { struct perf_event_header header; __u64 nr; - struct id_index_entry entries[0]; + struct id_index_entry entries[]; }; struct perf_record_auxtrace_info { -- 2.27.0

4 6

[PATCH v2] ASoC: cs4265: Fix the duplicated control name
by Fabio Estevam 16 Feb '22

16 Feb '22

From: Fabio Estevam <festevam(a)denx.de> Currently, the following error messages are seen during boot: asoc-simple-card sound: control 2:0:0:SPDIF Switch:0 is already present cs4265 1-004f: ASoC: failed to add widget SPDIF dapm kcontrol SPDIF Switch: -16 Quoting Mark Brown: "The driver is just plain buggy, it defines both a regular SPIDF Switch control and a SND_SOC_DAPM_SWITCH() called SPDIF both of which will create an identically named control, it can never have loaded without error. One or both of those has to be renamed or they need to be merged into one thing." Fix the duplicated control name by combining the two SPDIF controls here and move the register bits onto the DAPM widget and have DAPM control them. Fixes: f853d6b3ba34 ("ASoC: cs4265: Add a S/PDIF enable switch") Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- Changes since v1: - Combine the two controls (Charles) sound/soc/codecs/cs4265.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index 4aaee1873a11..4415fb364d4d 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -150,7 +150,6 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, 6, 1, 0), SOC_ENUM("C Data Access", cam_mode_enum), - SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1), SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), @@ -186,7 +185,7 @@ static const struct snd_soc_dapm_widget cs4265_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Loopback", SND_SOC_NOPM, 0, 0, &loopback_ctl), - SND_SOC_DAPM_SWITCH("SPDIF", SND_SOC_NOPM, 0, 0, + SND_SOC_DAPM_SWITCH("SPDIF", CS4265_SPDIF_CTL2, 5, 1, &spdif_switch), SND_SOC_DAPM_SWITCH("DAC", CS4265_PWRCTL, 1, 1, &dac_switch), -- 2.25.1

3 2

[PATCH v7 0/7] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 16 Feb '22

16 Feb '22

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V6: -- Update conditional clock voting to optional clock voting. -- Update Kconfig depends on field with select. -- Fix typo errors. Changes Since V5: -- Create new patch by updating macro name to lpi specific. -- Create new patch by updating lpi pin group structure with core group_desc structure. -- Fix typo errors. -- Sort macros in the make file and configuration file. Changes Since V4: -- Update commit message and description of the chip specific extraction patch. -- Sort macros in kconfig and makefile. -- Update optional clock voting to conditional clock voting. -- Fix typo errors. -- Move to quicinc domain email id's. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (7): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Update macro name to LPI specific pinctrl: qcom: Update lpi pin group structure pinctrl: qcom: Extract chip specific LPASS LPI code pinctrl: qcom: Add SC7280 lpass pin configuration pinctrl: qcom: Update clock voting as optional Tested this on SM8250 MTP with WSA and WCD codecs. Tested-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 133 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 133 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 245 +-------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 86 ++++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 168 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 166 ++++++++++++++ 9 files changed, 696 insertions(+), 368 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- 2.7.4

1 6

[PATCH v2 0/6] Add Tegra234 HDA support
by Mohan Kumar 16 Feb '22

16 Feb '22

This series add the support for TEGRA234 HDA driver support Mohan Kumar (6): ALSA: hda/tegra: Add Tegra234 hda driver support ALSA: hda/tegra: Hardcode GCAP ISS value on T234 ALSA: hda/tegra: Update scratch reg. communication dt-bindings: Add HDA support for Tegra234 dt-bindings: Document Tegra234 HDA support arm64: tegra: Add hda dts node for Tegra234 .../bindings/sound/nvidia,tegra30-hda.yaml | 3 + .../nvidia/tegra234-p3737-0000+p3701-0000.dts | 6 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 18 +++ include/dt-bindings/clock/tegra234-clock.h | 4 + include/dt-bindings/memory/tegra234-mc.h | 6 + .../dt-bindings/power/tegra234-powergate.h | 9 ++ include/dt-bindings/reset/tegra234-reset.h | 2 + sound/pci/hda/hda_tegra.c | 33 ++++- sound/pci/hda/patch_hdmi.c | 118 ++++++++++++++---- 9 files changed, 175 insertions(+), 24 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.17.1

5 12

[PATCH] ASoC: core: unregister clients and machine drivers in .shutdown
by Kai Vehmanen 16 Feb '22

16 Feb '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> On a platform shutdown, the expectation for most drivers is that userspace tasks will release all resources. When those sequences do not complete, it can be the case that PCM devices exposed by ALSA cards are used *after* the DSP shutdown completes, leading to a platform hang. When the clients and machine drivers provide an _unregister callback, let's invoke it in the shutdown sequence. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/core.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index d99ecbb4282d..2a35d8ddf43e 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -463,10 +463,19 @@ EXPORT_SYMBOL(snd_sof_device_remove); int snd_sof_device_shutdown(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_sof_pdata *pdata = sdev->pdata; if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); + /* + * make sure clients and machine driver(s) are unregistered to force + * all userspace devices to be closed prior to the DSP shutdown sequence + */ + sof_unregister_clients(sdev); + + snd_sof_machine_unregister(sdev, pdata); + if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) return snd_sof_shutdown(sdev); base-commit: f7d344a2bd5ec81fbd1ce76928fd059e57ec9bea -- 2.35.1

2 2

[PATCH] ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2
by Kai Vehmanen 16 Feb '22

16 Feb '22

Implement HDA keep alive (KAE) support for Intel display codecs. When no audio stream is active, the display codec will provide a continuous clock and a valid but silent audio stream to any connected HDMI/DP receiver. Without this, upon starting a new playback stream, initial samples may be lost as many receivers require time to initialize for new clock. This is a new feature in Intel AlderLake-P display codec implementation and replaces the Intel i915 silent-stream extension that has been used on older hardware. Main benefit of the new method is that codec no longer needs to be kept in D0 power state. This patch depends on commit 112a87c48e83 ("drm/i915/display: program audio CDCLK-TS for keepalives"). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jyri Sarha <jyri.sarha(a)intel.com> --- Hi, with newer Intel hardware, we can support the HDMI/DP silent-stream/keep-alive with standard HDA KAE bit. This patch only enables KAE for Intel hardware, but the patch is done in a way to allow other vendor's to opt-in and use the same KAE code. As I can't validate this for other hw, I only enable the feature for hw I have. The Intel specific implementation is put under SILENT_STREAM_I915 and it continues to be used for older hardware. sound/pci/hda/Kconfig | 19 ++--- sound/pci/hda/patch_hdmi.c | 152 +++++++++++++++++++++++++++---------- 2 files changed, 122 insertions(+), 49 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index febe1c2b7d9a..9f6c99c1d87b 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -285,15 +285,16 @@ config SND_HDA_INTEL_HDMI_SILENT_STREAM bool "Enable Silent Stream always for HDMI" depends on SND_HDA_INTEL help - Intel hardware has a feature called 'silent stream', that - keeps external HDMI receiver's analog circuitry powered on - avoiding 2-3 sec silence during playback start. This mechanism - relies on setting channel_id as 0xf, sending info packet and - preventing codec D3 entry (increasing platform static power - consumption when HDMI receiver is plugged-in). 2-3 sec silence - at the playback start is expected whenever there is format change. - (default is 2 channel format). - Say Y to enable Silent Stream feature. + Say Y to enable HD-Audio Keep Alive (KAE) aka Silent Stream + for HDMI on hardware that supports the feature. + + When enabled, the HDMI/DisplayPort codec will continue to provide + a continuous clock and a valid but silent data stream to + any connected external receiver. This allows to avoid gaps + at start of playback. Many receivers require multiple seconds + to start playing audio after the clock has been stopped. + This feature can impact power consumption as resources + are kept reserved both at transmitter and receiver. endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 64fe025fda86..584af2aa2f41 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -120,6 +120,12 @@ struct hdmi_pcm { struct snd_kcontrol *eld_ctl; }; +enum { + SILENT_STREAM_OFF = 0, + SILENT_STREAM_KAE, /* use standard HDA Keep-Alive */ + SILENT_STREAM_I915, /* Intel i915 extension */ +}; + struct hdmi_spec { struct hda_codec *codec; int num_cvts; @@ -179,7 +185,7 @@ struct hdmi_spec { hda_nid_t vendor_nid; const int *port_map; int port_num; - bool send_silent_stream; /* Flag to enable silent stream feature */ + int silent_stream_type; }; #ifdef CONFIG_SND_HDA_COMPONENT @@ -1665,18 +1671,71 @@ static void hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, #define I915_SILENT_FORMAT_BITS 16 #define I915_SILENT_FMT_MASK 0xf +static void silent_stream_enable_i915(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin) +{ + unsigned int format; + + snd_hdac_sync_audio_rate(&codec->core, per_pin->pin_nid, + per_pin->dev_id, I915_SILENT_RATE); + + /* trigger silent stream generation in hw */ + format = snd_hdac_calc_stream_format(I915_SILENT_RATE, I915_SILENT_CHANNELS, + I915_SILENT_FORMAT, I915_SILENT_FORMAT_BITS, 0); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, + I915_SILENT_FMT_MASK, I915_SILENT_FMT_MASK, format); + usleep_range(100, 200); + snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, I915_SILENT_FMT_MASK, 0, format); + + per_pin->channels = I915_SILENT_CHANNELS; + hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); +} + +static void silent_stream_set_kae(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + bool enable) +{ + unsigned int param; + + codec_dbg(codec, "HDMI: KAE %d cvt-NID=0x%x\n", enable, per_pin->cvt_nid); + + param = snd_hda_codec_read(codec, per_pin->cvt_nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + param = (param >> 16) & 0xff; + + if (enable) + param |= AC_DIG3_KAE; + else + param &= ~AC_DIG3_KAE; + + snd_hda_codec_write(codec, per_pin->cvt_nid, 0, AC_VERB_SET_DIGI_CONVERT_3, param); +} + static void silent_stream_enable(struct hda_codec *codec, struct hdmi_spec_per_pin *per_pin) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_cvt *per_cvt; int cvt_idx, pin_idx, err; - unsigned int format; + int keep_power = 0; + + /* + * Power-up will call hdmi_present_sense, so the PM calls + * have to be done without mutex held. + */ + + err = snd_hda_power_up_pm(codec); + if (err < 0 && err != -EACCES) { + codec_err(codec, + "Failed to power up codec for silent stream enable ret=[%d]\n", err); + snd_hda_power_down_pm(codec); + return; + } mutex_lock(&per_pin->lock); if (per_pin->setup) { codec_dbg(codec, "hdmi: PCM already open, no silent stream\n"); + err = -EBUSY; goto unlock_out; } @@ -1703,22 +1762,23 @@ static void silent_stream_enable(struct hda_codec *codec, /* configure unused pins to choose other converters */ pin_cvt_fixup(codec, per_pin, 0); - snd_hdac_sync_audio_rate(&codec->core, per_pin->pin_nid, - per_pin->dev_id, I915_SILENT_RATE); - - /* trigger silent stream generation in hw */ - format = snd_hdac_calc_stream_format(I915_SILENT_RATE, I915_SILENT_CHANNELS, - I915_SILENT_FORMAT, I915_SILENT_FORMAT_BITS, 0); - snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, - I915_SILENT_FMT_MASK, I915_SILENT_FMT_MASK, format); - usleep_range(100, 200); - snd_hda_codec_setup_stream(codec, per_pin->cvt_nid, I915_SILENT_FMT_MASK, 0, format); - - per_pin->channels = I915_SILENT_CHANNELS; - hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); + switch (spec->silent_stream_type) { + case SILENT_STREAM_KAE: + silent_stream_set_kae(codec, per_pin, true); + break; + case SILENT_STREAM_I915: + silent_stream_enable_i915(codec, per_pin); + keep_power = 1; + break; + default: + break; + } unlock_out: mutex_unlock(&per_pin->lock); + + if (err || !keep_power) + snd_hda_power_down_pm(codec); } static void silent_stream_disable(struct hda_codec *codec, @@ -1726,7 +1786,16 @@ static void silent_stream_disable(struct hda_codec *codec, { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_cvt *per_cvt; - int cvt_idx; + int cvt_idx, err; + + err = snd_hda_power_up_pm(codec); + if (err < 0 && err != -EACCES) { + codec_err(codec, + "Failed to power up codec for silent stream disable ret=[%d]\n", + err); + snd_hda_power_down_pm(codec); + return; + } mutex_lock(&per_pin->lock); if (!per_pin->silent_stream) @@ -1741,11 +1810,16 @@ static void silent_stream_disable(struct hda_codec *codec, per_cvt->assigned = 0; } + if (spec->silent_stream_type == SILENT_STREAM_KAE) + silent_stream_set_kae(codec, per_pin, false); + per_pin->cvt_nid = 0; per_pin->silent_stream = false; unlock_out: mutex_unlock(&per_pin->lock); + + snd_hda_power_down_pm(codec); } /* update ELD and jack state via audio component */ @@ -1767,29 +1841,11 @@ static void sync_eld_via_acomp(struct hda_codec *codec, monitor_next = per_pin->sink_eld.monitor_present; mutex_unlock(&per_pin->lock); - /* - * Power-up will call hdmi_present_sense, so the PM calls - * have to be done without mutex held. - */ - - if (spec->send_silent_stream) { - int pm_ret; - - if (!monitor_prev && monitor_next) { - pm_ret = snd_hda_power_up_pm(codec); - if (pm_ret < 0) - codec_err(codec, - "Monitor plugged-in, Failed to power up codec ret=[%d]\n", - pm_ret); + if (spec->silent_stream_type) { + if (!monitor_prev && monitor_next) silent_stream_enable(codec, per_pin); - } else if (monitor_prev && !monitor_next) { + else if (monitor_prev && !monitor_next) silent_stream_disable(codec, per_pin); - pm_ret = snd_hda_power_down_pm(codec); - if (pm_ret < 0) - codec_err(codec, - "Monitor plugged-out, Failed to power down codec ret=[%d]\n", - pm_ret); - } } } @@ -2982,7 +3038,7 @@ static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid, * module param or Kconfig option */ if (send_silent_stream) - spec->send_silent_stream = true; + spec->silent_stream_type = SILENT_STREAM_I915; return parse_intel_hdmi(codec); } @@ -3035,6 +3091,22 @@ static int patch_i915_tgl_hdmi(struct hda_codec *codec) return ret; } +static int patch_i915_adlp_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int res; + + res = patch_i915_tgl_hdmi(codec); + if (!res) { + spec = codec->spec; + + if (spec->silent_stream_type) + spec->silent_stream_type = SILENT_STREAM_KAE; + } + + return res; +} + /* Intel Baytrail and Braswell; with eld notifier */ static int patch_i915_byt_hdmi(struct hda_codec *codec) { @@ -4391,10 +4463,10 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862818, "Raptorlake HDMI", patch_i915_tgl_hdmi), -HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), -HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), base-commit: fc30f8dd54d4b42ace4e0bcf356ba2f2e1d8a2c7 -- 2.35.1

3 3

[PATCH v17 0/2] mtk: add ADSP mailbox controller for MT8195
by allen-kh.cheng 15 Feb '22

15 Feb '22

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> Mediatek ADSP IPC is used to send notification or short message between processors with dsp. It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp. This patchset was tested and confirmed on MT8195 cherry board. Based on linux-next changes since v16: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v15: - add paragraphs to describe the config symbols fully changes since v14: - add inline for get_mtk_adsp_mbox_priv changes since v13: - rebase on v5.16-rc8 - add back ptr check from of_device_get_match_data changes since v12: - remove of_device_get_match_data ptr check: of_device_get_match_data(dev) will never going to return NULL. driver probe with compatible mediatek,mt8195-adsp-mbox. changes since v11: - remove useless MODULE_LICENSE changes since v10: - split up v9 into two separate submissions changes since v9: - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (2): dt-bindings: mailbox: add mtk adsp-mbox document mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/mailbox/Kconfig | 9 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ 4 files changed, 239 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c -- 2.18.0

2 4

[PATCH] ASoC: tegra20: spdif: make const array rates static
by Colin Ian King 15 Feb '22

15 Feb '22

Don't populate the read-only const array rates on the stack but instead it static. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/tegra/tegra20_spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index d09cd7ee6879..64c2f304f254 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -186,7 +186,7 @@ static int tegra20_spdif_filter_rates(struct snd_pcm_hw_params *params, struct snd_soc_dai *dai = rule->private; struct tegra20_spdif *spdif = dev_get_drvdata(dai->dev); struct clk *parent = clk_get_parent(spdif->clk_spdif_out); - const unsigned int rates[] = { 32000, 44100, 48000 }; + static const unsigned int rates[] = { 32000, 44100, 48000 }; long i, parent_rate, valid_rates = 0; parent_rate = clk_get_rate(parent); -- 2.34.1

2 1

[PATCH] ASoC: soc-core: skip zero num_dai component in searching dai name
by Shengjiu Wang 15 Feb '22

15 Feb '22

In the case like dmaengine which's not a dai but as a component, the num_dai is zero, dmaengine component has the same component_of_node as cpu dai, when cpu dai component is not ready, but dmaengine component is ready, try to get cpu dai name, the snd_soc_get_dai_name() return -EINVAL, not -EPROBE_DEFER, that cause below error: asoc-simple-card <card name>: parse error -22 asoc-simple-card: probe of <card name> failed with error -22 The sound card failed to probe. So this patch fixes the issue above by skipping the zero num_dai component in searching dai name. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 434e61b46983..a088bc9f7dd7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3233,7 +3233,7 @@ int snd_soc_get_dai_name(const struct of_phandle_args *args, for_each_component(pos) { struct device_node *component_of_node = soc_component_to_node(pos); - if (component_of_node != args->np) + if (component_of_node != args->np || !pos->num_dai) continue; ret = snd_soc_component_of_xlate_dai_name(pos, args, dai_name); -- 2.17.1

2 1

[PATCH v2] ASoC: SOF: Makefile: Fix randconfig sof-client build when SND_SOC_SOF=y
by Peter Ujfalusi 15 Feb '22

15 Feb '22

Intel's kernel test robot found the following randconfig combination: SND_SOC_SOF=y SND_SOC_SOF_CLIENT=m In this the sof-client object is not going to be built into the snd-sof.o and we will have undefined references to the sof-client functions. Fixes: 6955d9512d0e ("ASoC: SOF: Introduce IPC SOF client support") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Peter Ujfalusi <Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com>> --- Hi Mark, Pierre, Changes since v1: - fix author and SoB from gmail.com to linux.intel.com When the client support was added we did some permutation tests (at least I did) but missed this combination. It is a valid one for sure and the proper fix is via the Makefile. Sorry for mising this Regards, Peter sound/soc/sof/Makefile | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 4d31282c847d..a0459f06c68a 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,7 +2,9 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o -snd-sof-$(CONFIG_SND_SOC_SOF_CLIENT) += sof-client.o +ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) +snd-sof-objs += sof-client.o +endif snd-sof-$(CONFIG_SND_SOC_SOF_COMPRESS) += compress.o -- 2.35.1

2 1

[PATCH] ASoC: cs4265: Fix the duplicated control name
by Fabio Estevam 15 Feb '22

15 Feb '22

From: Fabio Estevam <festevam(a)denx.de> Currently, the following error messages are seen during boot: asoc-simple-card sound: control 2:0:0:SPDIF Switch:0 is already present cs4265 1-004f: ASoC: failed to add widget SPDIF dapm kcontrol SPDIF Switch: -16 Quoting Mark Brown: "The driver is just plain buggy, it defines both a regular SPIDF Switch control and a SND_SOC_DAPM_SWITCH() called SPDIF both of which will create an identically named control, it can never have loaded without error. One or both of those has to be renamed." Rename it from "SPDIF Switch" to "SPDIF" to avoid the name confict. Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/codecs/cs4265.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index c338c9f6cf91..8891e9275cba 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -150,7 +150,7 @@ static const struct snd_kcontrol_new cs4265_snd_controls[] = { SOC_SINGLE("E to F Buffer Disable Switch", CS4265_SPDIF_CTL1, 6, 1, 0), SOC_ENUM("C Data Access", cam_mode_enum), - SOC_SINGLE("SPDIF Switch", CS4265_SPDIF_CTL2, 5, 1, 1), + SOC_SINGLE("SPDIF", CS4265_SPDIF_CTL2, 5, 1, 1), SOC_SINGLE("Validity Bit Control Switch", CS4265_SPDIF_CTL2, 3, 1, 0), SOC_ENUM("SPDIF Mono/Stereo", spdif_mono_stereo_enum), -- 2.25.1

2 3

[PATCH] ASoC: codec: wm8960: complete discharge on BIAS OFF->STANDBY
by Viorel Suman (OSS) 15 Feb '22

15 Feb '22

From: Viorel Suman <viorel.suman(a)nxp.com> On BIAS STANDBY->OFF transition the current implementation sleeps 600ms on suspend in order to discharge the chip. The suspend is propagated from "snd_soc_suspend" call for all audio cards in a serial fashion, thus in case of boards like i.MX8DXL EVK which has 3 distinct WM8960 codecs the total cumulated sleep on suspend is 1.8 seconds. On the other hand the BIAS OFF->STANDBY transition happens asynchronously with regard to "snd_soc_resume" - the call is propagated from "soc_resume_deferred" which is just scheduled from "snd_soc_resume", each card having its own work scheduled to execute "soc_resume_deferred" call. The patch performs discharge completion on BIAS OFF->STANDBY transition so that the cumulated effect on suspend described above is avoided and discharge is completed in paralel in case of multiple WM8960 codecs on the board. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/codecs/wm8960.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 75de6a81465df..b7a2246184e08 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -45,6 +45,8 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +#define WM8960_DSCH_TOUT 600 /* discharge timeout, ms */ + static bool is_pll_freq_available(unsigned int source, unsigned int target); static int wm8960_set_pll(struct snd_soc_component *component, unsigned int freq_in, unsigned int freq_out); @@ -133,6 +135,7 @@ struct wm8960_priv { int freq_in; bool is_stream_in_use[2]; struct wm8960_data pdata; + ktime_t dsch_start; }; #define wm8960_reset(c) regmap_write(c, WM8960_RESET, 0) @@ -898,6 +901,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_component *component, struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component); u16 pm2 = snd_soc_component_read(component, WM8960_POWER2); int ret; + ktime_t tout; switch (level) { case SND_SOC_BIAS_ON: @@ -944,6 +948,11 @@ static int wm8960_set_bias_level_out3(struct snd_soc_component *component, case SND_SOC_BIAS_STANDBY: if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* ensure discharge is complete */ + tout = WM8960_DSCH_TOUT - ktime_ms_delta(ktime_get(), wm8960->dsch_start); + if (tout > 0) + msleep(tout); + regcache_sync(wm8960->regmap); /* Enable anti-pop features */ @@ -973,9 +982,9 @@ static int wm8960_set_bias_level_out3(struct snd_soc_component *component, WM8960_POBCTRL | WM8960_SOFT_ST | WM8960_BUFDCOPEN | WM8960_BUFIOEN); - /* Disable VMID and VREF, let them discharge */ + /* Disable VMID and VREF, mark discharge */ snd_soc_component_write(component, WM8960_POWER1, 0); - msleep(600); + wm8960->dsch_start = ktime_get(); break; } -- 2.26.3

3 2

[PATCH] conf: Remove left justification for SND_CONFIG_TYPE_REAL
by Simon Christensson 15 Feb '22

15 Feb '22

In order to avoid extra padding replace data type "%-16g" (left justification) with default right justification "%16g" when printing snd_config type SND_CONFIG_TYPE_REAL. Example for control "min_dB" available in softvol: Before patch: "min_dB -50 " After patch: "min_dB -50.0"

1 0

[PATCH 0/2] Add Euro Headset support for wcd938x codec
by Srinivasa Rao Mandadapu 15 Feb '22

15 Feb '22

This patch set is to add switch control for selecting CTIA/OMTP Headset Srinivasa Rao Mandadapu (2): ASoC: codec: wcd938x: Add switch control for selecting CTIA/OMTP Headset ASoC: dt-bindings: wcd938x: Add gpio property for selecting CTIA/OMTP headset .../devicetree/bindings/sound/qcom,wcd938x.yaml | 4 +++ sound/soc/codecs/wcd938x.c | 38 ++++++++++++++++++++++ 2 files changed, 42 insertions(+) -- 2.7.4

3 5

[PATCH 0/9] use GFP_KERNEL
by Julia Lawall 15 Feb '22

15 Feb '22

Platform_driver and pci_driver probe functions aren't called with locks held and thus don't need GFP_ATOMIC. Use GFP_KERNEL instead. All changes have been compile-tested. --- drivers/ata/pata_mpc52xx.c | 2 +- drivers/crypto/ux500/cryp/cryp_core.c | 2 +- drivers/crypto/ux500/hash/hash_core.c | 2 +- drivers/media/pci/cx18/cx18-driver.c | 2 +- drivers/media/platform/fsl-viu.c | 2 +- drivers/message/fusion/mptspi.c | 2 +- drivers/mfd/sta2x11-mfd.c | 2 +- drivers/mtd/devices/spear_smi.c | 2 +- drivers/net/ethernet/moxa/moxart_ether.c | 4 ++-- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- 10 files changed, 11 insertions(+), 11 deletions(-)

5 5

[PATCH 20/23] ASoC: codecs: wcd938x: Make use of the helper component_compare/release_of
by Yong Wu 15 Feb '22

15 Feb '22

Use the common compare/release helpers from component. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yong Wu <yong.wu(a)mediatek.com> --- sound/soc/codecs/wcd938x.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index eff200a07d9f..54f4a41ceca2 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -4417,16 +4417,6 @@ static const struct component_master_ops wcd938x_comp_ops = { .unbind = wcd938x_unbind, }; -static int wcd938x_compare_of(struct device *dev, void *data) -{ - return dev->of_node == data; -} - -static void wcd938x_release_of(struct device *dev, void *data) -{ - of_node_put(data); -} - static int wcd938x_add_slave_components(struct wcd938x_priv *wcd938x, struct device *dev, struct component_match **matchptr) @@ -4442,8 +4432,8 @@ static int wcd938x_add_slave_components(struct wcd938x_priv *wcd938x, } of_node_get(wcd938x->rxnode); - component_match_add_release(dev, matchptr, wcd938x_release_of, - wcd938x_compare_of, wcd938x->rxnode); + component_match_add_release(dev, matchptr, component_release_of, + component_compare_of, wcd938x->rxnode); wcd938x->txnode = of_parse_phandle(np, "qcom,tx-device", 0); if (!wcd938x->txnode) { @@ -4451,8 +4441,8 @@ static int wcd938x_add_slave_components(struct wcd938x_priv *wcd938x, return -ENODEV; } of_node_get(wcd938x->txnode); - component_match_add_release(dev, matchptr, wcd938x_release_of, - wcd938x_compare_of, wcd938x->txnode); + component_match_add_release(dev, matchptr, component_release_of, + component_compare_of, wcd938x->txnode); return 0; } -- 2.18.0

3 3

cs4265: failed to add widget SPDIF dapm kcontrol
by Fabio Estevam 14 Feb '22

14 Feb '22

Hi Charles, I have a custom board with a cs4265 codec and the audio plays fine. During boot, the following error messages are seen: asoc-simple-card sound: control 2:0:0:SPDIF Switch:0 is already present cs4265 1-004f: ASoC: failed to add widget SPDIF dapm kcontrol SPDIF Switch: -16 SPDIF is not used on this board. Would you have any recommendations for getting rid of such error messages? Thanks, Fabio Estevam

2 3

[PATCH v13 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 14 Feb '22

14 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V12: -- Fix typo errors. Changes Since V11: -- Fix kernel robot issue on arguments type mismatch. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 304 ++++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 447 +++++++++++++++ sound/soc/qcom/lpass.h | 145 +++++ 9 files changed, 1887 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

2 5

[GIT PULL] ASoC fixes for v5.17-rc4
by Mark Brown 14 Feb '22

14 Feb '22

The following changes since commit a4f399a1416f645ac701064a55b0cb5203707ac9: Input: wm97xx: Simplify resource management (2022-01-31 13:17:24 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.17-rc4 for you to fetch changes up to a887f9c7a4d37a8e874ba8415a42a92a1b5139fc: ASoC: wm_adsp: Correct control read size when parsing compressed buffer (2022-02-10 17:26:43 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.18 More fixes that have arrived in the past few -rcs, plus a MAINTAINERS update. The biggest update here is the fix for control change notifications in ASoC generic controls found by mixer-test. ---------------------------------------------------------------- Charles Keepax (1): ASoC: wm_adsp: Correct control read size when parsing compressed buffer Guenter Roeck (1): ASoC: Revert "ASoC: mediatek: Check for error clk pointer" Kai Vehmanen (3): ASoC: rt5682s: do not block workqueue if card is unbound ASoC: rt5668: do not block workqueue if card is unbound ASoC: rt5682: do not block workqueue if card is unbound Mark Brown (5): ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw() ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw_sx() ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw_range() ASoC: ops: Fix stereo change notifications in snd_soc_put_xr_sx() ASoC: ops: Fix stereo change notifications Martin Povišer (1): ASoC: tas2770: Insert post reset delay Stephen Boyd (1): ASoC: qcom: Actually clear DMA interrupt register for HDMI Tzung-Bi Shih (1): MAINTAINERS: update cros_ec_codec maintainers V sujith kumar Reddy (1): ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver .../bindings/sound/google,cros-ec-codec.yaml | 1 + MAINTAINERS | 1 + drivers/soc/mediatek/mtk-scpsys.c | 15 +++----- sound/soc/amd/acp/acp-mach.h | 1 - sound/soc/amd/acp/acp-sof-mach.c | 4 +-- sound/soc/codecs/rt5668.c | 12 ++++--- sound/soc/codecs/rt5682.c | 12 ++++--- sound/soc/codecs/rt5682s.c | 12 ++++--- sound/soc/codecs/tas2770.c | 7 +++- sound/soc/codecs/wm_adsp.c | 3 +- sound/soc/qcom/lpass-platform.c | 8 ++--- sound/soc/soc-ops.c | 41 ++++++++++++++++------ 12 files changed, 72 insertions(+), 45 deletions(-)

2 1

[PATCH] ASoC: wm8731: Delete empty remove() function
by Mark Brown 14 Feb '22

14 Feb '22

The I2C remove function is empty for the wm8731 driver, it can just be deleted. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm8731.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 86b1f6eaa599..5d4949c2ec9b 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -777,11 +777,6 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, return 0; } -static int wm8731_i2c_remove(struct i2c_client *client) -{ - return 0; -} - static const struct i2c_device_id wm8731_i2c_id[] = { { "wm8731", 0 }, { } @@ -794,7 +789,6 @@ static struct i2c_driver wm8731_i2c_driver = { .of_match_table = wm8731_of_match, }, .probe = wm8731_i2c_probe, - .remove = wm8731_i2c_remove, .id_table = wm8731_i2c_id, }; #endif -- 2.30.2

2 2

[PATCH 0/4] ASoC: pcm3168a: code cleanup
by Nikita Yushchenko 14 Feb '22

14 Feb '22

These patches clean up pcm3168a driver, without introducing any functional change. Nikita Yushchenko (4): ASoC: pcm3168a: cleanup unintuitive mask usage ASoC: pcm3168a: refactor hw_params routine ASoC: pcm3168a: refactor format handling ASoC: pcm3168a: remove numeric PCM3168A_NUM_SUPPLIES sound/soc/codecs/pcm3168a.c | 150 +++++++++++++++++------------------- 1 file changed, 70 insertions(+), 80 deletions(-) -- 2.30.2

2 5

[PATCH v2 1/2] platform: make platform_get_irq_optional() optional
by Sergey Shtylyov 14 Feb '22

14 Feb '22

This patch is based on the former Andy Shevchenko's patch: https://lore.kernel.org/lkml/20210331144526.19439-1-andriy.shevchenko@linux… Currently platform_get_irq_optional() returns an error code even if IRQ resource simply has not been found. It prevents the callers from being error code agnostic in their error handling: ret = platform_get_irq_optional(...); if (ret < 0 && ret != -ENXIO) return ret; // respect deferred probe if (ret > 0) ...we get an IRQ... All other *_optional() APIs seem to return 0 or NULL in case an optional resource is not available. Let's follow this good example, so that the callers would look like: ret = platform_get_irq_optional(...); if (ret < 0) return ret; if (ret > 0) ...we get an IRQ... Reported-by: Matthias Schiffer <matthias.schiffer(a)ew.tq-group.com> Signed-off-by: Sergey Shtylyov <s.shtylyov(a)omp.ru> --- Changes in version 2: - added the error check using dev_err_probe() to platform_get_irq_optional(); - fixed up the IRQ checks in 'drivers/char/ipmi/bt-bmc.c'; - fixed up the IRQ check in 'drivers/thermal/rcar_gen3_thermal.c'; - removed from the patch 'drivers/edac/xgene_edac.c and 'drivers/power/supply/mp2629_charger.c' as they were switched to calling platform_get_irq(); - shortened the IRQ checking code in 'drivers/vfio/platform/vfio_platform.c'; - rebased atop of the recent platform_get_irq_byname() patch; - reformatted the patch description. drivers/base/platform.c | 60 +++++++++++++++--------- drivers/char/ipmi/bt-bmc.c | 6 +-- drivers/counter/interrupt-cnt.c | 4 +- drivers/gpio/gpio-altera.c | 3 +- drivers/gpio/gpio-mvebu.c | 2 +- drivers/gpio/gpio-tqmx86.c | 2 +- drivers/i2c/busses/i2c-brcmstb.c | 8 ++-- drivers/i2c/busses/i2c-ocores.c | 4 +- drivers/mmc/host/sh_mmcif.c | 4 +- drivers/mtd/nand/raw/brcmnand/brcmnand.c | 4 +- drivers/net/ethernet/davicom/dm9000.c | 2 +- drivers/net/ethernet/freescale/fec_ptp.c | 2 +- drivers/phy/renesas/phy-rcar-gen3-usb2.c | 4 +- drivers/platform/chrome/cros_ec_lpc.c | 2 +- drivers/platform/x86/intel/punit_ipc.c | 2 +- drivers/spi/spi-hisi-sfc-v3xx.c | 2 +- drivers/spi/spi-mtk-nor.c | 3 +- drivers/thermal/rcar_gen3_thermal.c | 2 + drivers/tty/serial/8250/8250_mtk.c | 4 +- drivers/tty/serial/sh-sci.c | 6 +-- drivers/uio/uio_pdrv_genirq.c | 2 +- drivers/vfio/platform/vfio_platform.c | 3 +- sound/soc/dwc/dwc-i2s.c | 4 +- 23 files changed, 76 insertions(+), 59 deletions(-) diff --git a/drivers/base/platform.c b/drivers/base/platform.c index 7d08cd8947be..52a8356f8261 100644 --- a/drivers/base/platform.c +++ b/drivers/base/platform.c @@ -148,25 +148,7 @@ devm_platform_ioremap_resource_byname(struct platform_device *pdev, EXPORT_SYMBOL_GPL(devm_platform_ioremap_resource_byname); #endif /* CONFIG_HAS_IOMEM */ -/** - * platform_get_irq_optional - get an optional IRQ for a device - * @dev: platform device - * @num: IRQ number index - * - * Gets an IRQ for a platform device. Device drivers should check the return - * value for errors so as to not pass a negative integer value to the - * request_irq() APIs. This is the same as platform_get_irq(), except that it - * does not print an error message if an IRQ can not be obtained. - * - * For example:: - * - * int irq = platform_get_irq_optional(pdev, 0); - * if (irq < 0) - * return irq; - * - * Return: non-zero IRQ number on success, negative error number on failure. - */ -int platform_get_irq_optional(struct platform_device *dev, unsigned int num) +static int __platform_get_irq(struct platform_device *dev, unsigned int num) { int ret; #ifdef CONFIG_SPARC @@ -235,6 +217,42 @@ int platform_get_irq_optional(struct platform_device *dev, unsigned int num) return -EINVAL; return ret; } + +/** + * platform_get_irq_optional - get an optional IRQ for a device + * @dev: platform device + * @num: IRQ number index + * + * Gets an IRQ for a platform device. Device drivers should check the return + * value for errors so as to not pass a negative integer value to the + * request_irq() APIs. This is the same as platform_get_irq(), except that it + * does not print an error message if an IRQ can not be obtained and returns + * 0 when IRQ resource has not been found. + * + * For example:: + * + * int irq = platform_get_irq_optional(pdev, 0); + * if (irq < 0) + * return irq; + * if (irq > 0) + * ...we have IRQ line defined... + * + * Return: non-zero IRQ number on success, 0 if IRQ wasn't found, negative error + * number on failure. + */ +int platform_get_irq_optional(struct platform_device *dev, unsigned int num) +{ + int ret; + + ret = __platform_get_irq(dev, num); + if (ret == -ENXIO) + return 0; + if (ret < 0) + return dev_err_probe(&dev->dev, ret, + "IRQ index %u not found\n", num); + + return ret; +} EXPORT_SYMBOL_GPL(platform_get_irq_optional); /** @@ -258,7 +276,7 @@ int platform_get_irq(struct platform_device *dev, unsigned int num) { int ret; - ret = platform_get_irq_optional(dev, num); + ret = __platform_get_irq(dev, num); if (ret < 0) return dev_err_probe(&dev->dev, ret, "IRQ index %u not found\n", num); @@ -277,7 +295,7 @@ int platform_irq_count(struct platform_device *dev) { int ret, nr = 0; - while ((ret = platform_get_irq_optional(dev, nr)) >= 0) + while ((ret = __platform_get_irq(dev, nr)) >= 0) nr++; if (ret == -EPROBE_DEFER) diff --git a/drivers/char/ipmi/bt-bmc.c b/drivers/char/ipmi/bt-bmc.c index 7450904e330a..289acaf4e720 100644 --- a/drivers/char/ipmi/bt-bmc.c +++ b/drivers/char/ipmi/bt-bmc.c @@ -380,7 +380,7 @@ static int bt_bmc_config_irq(struct bt_bmc *bt_bmc, u32 reg; bt_bmc->irq = platform_get_irq_optional(pdev, 0); - if (bt_bmc->irq < 0) + if (bt_bmc->irq <= 0) return bt_bmc->irq; rc = devm_request_irq(dev, bt_bmc->irq, bt_bmc_irq, IRQF_SHARED, @@ -438,7 +438,7 @@ static int bt_bmc_probe(struct platform_device *pdev) bt_bmc_config_irq(bt_bmc, pdev); - if (bt_bmc->irq >= 0) { + if (bt_bmc->irq > 0) { dev_info(dev, "Using IRQ %d\n", bt_bmc->irq); } else { dev_info(dev, "No IRQ; using timer\n"); @@ -464,7 +464,7 @@ static int bt_bmc_remove(struct platform_device *pdev) struct bt_bmc *bt_bmc = dev_get_drvdata(&pdev->dev); misc_deregister(&bt_bmc->miscdev); - if (bt_bmc->irq < 0) + if (bt_bmc->irq <= 0) del_timer_sync(&bt_bmc->poll_timer); return 0; } diff --git a/drivers/counter/interrupt-cnt.c b/drivers/counter/interrupt-cnt.c index 9e99702470c2..a2443c66330b 100644 --- a/drivers/counter/interrupt-cnt.c +++ b/drivers/counter/interrupt-cnt.c @@ -157,9 +157,7 @@ static int interrupt_cnt_probe(struct platform_device *pdev) priv = counter_priv(counter); priv->irq = platform_get_irq_optional(pdev, 0); - if (priv->irq == -ENXIO) - priv->irq = 0; - else if (priv->irq < 0) + if (priv->irq < 0) return dev_err_probe(dev, priv->irq, "failed to get IRQ\n"); priv->gpio = devm_gpiod_get_optional(dev, NULL, GPIOD_IN); diff --git a/drivers/gpio/gpio-altera.c b/drivers/gpio/gpio-altera.c index b59fae993626..02a2995aa368 100644 --- a/drivers/gpio/gpio-altera.c +++ b/drivers/gpio/gpio-altera.c @@ -267,8 +267,7 @@ static int altera_gpio_probe(struct platform_device *pdev) altera_gc->mmchip.gc.parent = &pdev->dev; altera_gc->mapped_irq = platform_get_irq_optional(pdev, 0); - - if (altera_gc->mapped_irq < 0) + if (altera_gc->mapped_irq <= 0) goto skip_irq; if (of_property_read_u32(node, "altr,interrupt-type", &reg)) { diff --git a/drivers/gpio/gpio-mvebu.c b/drivers/gpio/gpio-mvebu.c index 4c1f9e1091b7..d94594807697 100644 --- a/drivers/gpio/gpio-mvebu.c +++ b/drivers/gpio/gpio-mvebu.c @@ -1293,7 +1293,7 @@ static int mvebu_gpio_probe(struct platform_device *pdev) for (i = 0; i < 4; i++) { int irq = platform_get_irq_optional(pdev, i); - if (irq < 0) + if (irq <= 0) continue; irq_set_chained_handler_and_data(irq, mvebu_gpio_irq_handler, mvchip); diff --git a/drivers/gpio/gpio-tqmx86.c b/drivers/gpio/gpio-tqmx86.c index 5b103221b58d..dc0f83236ce8 100644 --- a/drivers/gpio/gpio-tqmx86.c +++ b/drivers/gpio/gpio-tqmx86.c @@ -237,7 +237,7 @@ static int tqmx86_gpio_probe(struct platform_device *pdev) int ret, irq; irq = platform_get_irq_optional(pdev, 0); - if (irq < 0 && irq != -ENXIO) + if (irq < 0) return irq; res = platform_get_resource(pdev, IORESOURCE_IO, 0); diff --git a/drivers/i2c/busses/i2c-brcmstb.c b/drivers/i2c/busses/i2c-brcmstb.c index 490ee3962645..69395ae27a1a 100644 --- a/drivers/i2c/busses/i2c-brcmstb.c +++ b/drivers/i2c/busses/i2c-brcmstb.c @@ -250,7 +250,7 @@ static int brcmstb_i2c_wait_for_completion(struct brcmstb_i2c_dev *dev) int ret = 0; unsigned long timeout = msecs_to_jiffies(I2C_TIMEOUT); - if (dev->irq >= 0) { + if (dev->irq > 0) { if (!wait_for_completion_timeout(&dev->done, timeout)) ret = -ETIMEDOUT; } else { @@ -297,7 +297,7 @@ static int brcmstb_send_i2c_cmd(struct brcmstb_i2c_dev *dev, return rc; /* only if we are in interrupt mode */ - if (dev->irq >= 0) + if (dev->irq > 0) reinit_completion(&dev->done); /* enable BSC CTL interrupt line */ @@ -652,7 +652,7 @@ static int brcmstb_i2c_probe(struct platform_device *pdev) brcmstb_i2c_enable_disable_irq(dev, INT_DISABLE); /* register the ISR handler */ - if (dev->irq >= 0) { + if (dev->irq > 0) { rc = devm_request_irq(&pdev->dev, dev->irq, brcmstb_i2c_isr, IRQF_SHARED, int_name ? int_name : pdev->name, @@ -696,7 +696,7 @@ static int brcmstb_i2c_probe(struct platform_device *pdev) dev_info(dev->device, "%s@%dhz registered in %s mode\n", int_name ? int_name : " ", dev->clk_freq_hz, - (dev->irq >= 0) ? "interrupt" : "polling"); + (dev->irq > 0) ? "interrupt" : "polling"); return 0; diff --git a/drivers/i2c/busses/i2c-ocores.c b/drivers/i2c/busses/i2c-ocores.c index a0af027db04c..1f4d5e52ff42 100644 --- a/drivers/i2c/busses/i2c-ocores.c +++ b/drivers/i2c/busses/i2c-ocores.c @@ -691,10 +691,10 @@ static int ocores_i2c_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "sifive,fu540-c000-i2c")) { i2c->flags |= OCORES_FLAG_BROKEN_IRQ; - irq = -ENXIO; + irq = 0; } - if (irq == -ENXIO) { + if (!irq) { ocores_algorithm.master_xfer = ocores_xfer_polling; } else { if (irq < 0) diff --git a/drivers/mmc/host/sh_mmcif.c b/drivers/mmc/host/sh_mmcif.c index bcc595c70a9f..f558b9862032 100644 --- a/drivers/mmc/host/sh_mmcif.c +++ b/drivers/mmc/host/sh_mmcif.c @@ -1465,14 +1465,14 @@ static int sh_mmcif_probe(struct platform_device *pdev) sh_mmcif_sync_reset(host); sh_mmcif_writel(host->addr, MMCIF_CE_INT_MASK, MASK_ALL); - name = irq[1] < 0 ? dev_name(dev) : "sh_mmc:error"; + name = irq[1] <= 0 ? dev_name(dev) : "sh_mmc:error"; ret = devm_request_threaded_irq(dev, irq[0], sh_mmcif_intr, sh_mmcif_irqt, 0, name, host); if (ret) { dev_err(dev, "request_irq error (%s)\n", name); goto err_clk; } - if (irq[1] >= 0) { + if (irq[1] > 0) { ret = devm_request_threaded_irq(dev, irq[1], sh_mmcif_intr, sh_mmcif_irqt, 0, "sh_mmc:int", host); diff --git a/drivers/mtd/nand/raw/brcmnand/brcmnand.c b/drivers/mtd/nand/raw/brcmnand/brcmnand.c index f75929783b94..ac222985efde 100644 --- a/drivers/mtd/nand/raw/brcmnand/brcmnand.c +++ b/drivers/mtd/nand/raw/brcmnand/brcmnand.c @@ -1521,7 +1521,7 @@ static irqreturn_t brcmnand_ctlrdy_irq(int irq, void *data) /* check if you need to piggy back on the ctrlrdy irq */ if (ctrl->edu_pending) { - if (irq == ctrl->irq && ((int)ctrl->edu_irq >= 0)) + if (irq == ctrl->irq && ((int)ctrl->edu_irq > 0)) /* Discard interrupts while using dedicated edu irq */ return IRQ_HANDLED; @@ -2956,7 +2956,7 @@ static int brcmnand_edu_setup(struct platform_device *pdev) brcmnand_edu_init(ctrl); ctrl->edu_irq = platform_get_irq_optional(pdev, 1); - if (ctrl->edu_irq < 0) { + if (ctrl->edu_irq <= 0) { dev_warn(dev, "FLASH EDU enabled, using ctlrdy irq\n"); } else { diff --git a/drivers/net/ethernet/davicom/dm9000.c b/drivers/net/ethernet/davicom/dm9000.c index 0985ab216566..740c660a9411 100644 --- a/drivers/net/ethernet/davicom/dm9000.c +++ b/drivers/net/ethernet/davicom/dm9000.c @@ -1509,7 +1509,7 @@ dm9000_probe(struct platform_device *pdev) } db->irq_wake = platform_get_irq_optional(pdev, 1); - if (db->irq_wake >= 0) { + if (db->irq_wake > 0) { dev_dbg(db->dev, "wakeup irq %d\n", db->irq_wake); ret = request_irq(db->irq_wake, dm9000_wol_interrupt, diff --git a/drivers/net/ethernet/freescale/fec_ptp.c b/drivers/net/ethernet/freescale/fec_ptp.c index af99017a5453..de1d23808b6c 100644 --- a/drivers/net/ethernet/freescale/fec_ptp.c +++ b/drivers/net/ethernet/freescale/fec_ptp.c @@ -616,7 +616,7 @@ void fec_ptp_init(struct platform_device *pdev, int irq_idx) /* Failure to get an irq is not fatal, * only the PTP_CLOCK_PPS clock events should stop */ - if (irq >= 0) { + if (irq > 0) { ret = devm_request_irq(&pdev->dev, irq, fec_pps_interrupt, 0, pdev->name, ndev); if (ret < 0) diff --git a/drivers/phy/renesas/phy-rcar-gen3-usb2.c b/drivers/phy/renesas/phy-rcar-gen3-usb2.c index 9de617ca9daa..4914d6aca208 100644 --- a/drivers/phy/renesas/phy-rcar-gen3-usb2.c +++ b/drivers/phy/renesas/phy-rcar-gen3-usb2.c @@ -439,7 +439,7 @@ static int rcar_gen3_phy_usb2_init(struct phy *p) u32 val; int ret; - if (!rcar_gen3_is_any_rphy_initialized(channel) && channel->irq >= 0) { + if (!rcar_gen3_is_any_rphy_initialized(channel) && channel->irq > 0) { INIT_WORK(&channel->work, rcar_gen3_phy_usb2_work); ret = request_irq(channel->irq, rcar_gen3_phy_usb2_irq, IRQF_SHARED, dev_name(channel->dev), channel); @@ -486,7 +486,7 @@ static int rcar_gen3_phy_usb2_exit(struct phy *p) val &= ~USB2_INT_ENABLE_UCOM_INTEN; writel(val, usb2_base + USB2_INT_ENABLE); - if (channel->irq >= 0 && !rcar_gen3_is_any_rphy_initialized(channel)) + if (channel->irq > 0 && !rcar_gen3_is_any_rphy_initialized(channel)) free_irq(channel->irq, channel); return 0; diff --git a/drivers/platform/chrome/cros_ec_lpc.c b/drivers/platform/chrome/cros_ec_lpc.c index d6306d2a096f..91686d306534 100644 --- a/drivers/platform/chrome/cros_ec_lpc.c +++ b/drivers/platform/chrome/cros_ec_lpc.c @@ -400,7 +400,7 @@ static int cros_ec_lpc_probe(struct platform_device *pdev) irq = platform_get_irq_optional(pdev, 0); if (irq > 0) ec_dev->irq = irq; - else if (irq != -ENXIO) { + else if (irq < 0) { dev_err(dev, "couldn't retrieve IRQ number (%d)\n", irq); return irq; } diff --git a/drivers/platform/x86/intel/punit_ipc.c b/drivers/platform/x86/intel/punit_ipc.c index 66bb39fd0ef9..f3cf5ee1466f 100644 --- a/drivers/platform/x86/intel/punit_ipc.c +++ b/drivers/platform/x86/intel/punit_ipc.c @@ -278,7 +278,7 @@ static int intel_punit_ipc_probe(struct platform_device *pdev) platform_set_drvdata(pdev, punit_ipcdev); irq = platform_get_irq_optional(pdev, 0); - if (irq < 0) { + if (irq <= 0) { dev_warn(&pdev->dev, "Invalid IRQ, using polling mode\n"); } else { ret = devm_request_irq(&pdev->dev, irq, intel_punit_ioc, diff --git a/drivers/spi/spi-hisi-sfc-v3xx.c b/drivers/spi/spi-hisi-sfc-v3xx.c index d3a23b1c2a4c..476ddc081c60 100644 --- a/drivers/spi/spi-hisi-sfc-v3xx.c +++ b/drivers/spi/spi-hisi-sfc-v3xx.c @@ -467,7 +467,7 @@ static int hisi_sfc_v3xx_probe(struct platform_device *pdev) dev_err(dev, "failed to request irq%d, ret = %d\n", host->irq, ret); host->irq = 0; } - } else { + } else if (host->irq < 0) { host->irq = 0; } diff --git a/drivers/spi/spi-mtk-nor.c b/drivers/spi/spi-mtk-nor.c index 5c93730615f8..2422b0545936 100644 --- a/drivers/spi/spi-mtk-nor.c +++ b/drivers/spi/spi-mtk-nor.c @@ -829,8 +829,7 @@ static int mtk_nor_probe(struct platform_device *pdev) mtk_nor_init(sp); irq = platform_get_irq_optional(pdev, 0); - - if (irq < 0) { + if (irq <= 0) { dev_warn(sp->dev, "IRQ not available."); } else { ret = devm_request_irq(sp->dev, irq, mtk_nor_irq_handler, 0, diff --git a/drivers/thermal/rcar_gen3_thermal.c b/drivers/thermal/rcar_gen3_thermal.c index 43eb25b167bc..359b9941c42b 100644 --- a/drivers/thermal/rcar_gen3_thermal.c +++ b/drivers/thermal/rcar_gen3_thermal.c @@ -432,6 +432,8 @@ static int rcar_gen3_thermal_request_irqs(struct rcar_gen3_thermal_priv *priv, irq = platform_get_irq_optional(pdev, i); if (irq < 0) return irq; + if (!irq) + return -ENXIO; irqname = devm_kasprintf(dev, GFP_KERNEL, "%s:ch%d", dev_name(dev), i); diff --git a/drivers/tty/serial/8250/8250_mtk.c b/drivers/tty/serial/8250/8250_mtk.c index fb65dc601b23..328ab074fd89 100644 --- a/drivers/tty/serial/8250/8250_mtk.c +++ b/drivers/tty/serial/8250/8250_mtk.c @@ -621,7 +621,7 @@ static int __maybe_unused mtk8250_suspend(struct device *dev) serial8250_suspend_port(data->line); pinctrl_pm_select_sleep_state(dev); - if (irq >= 0) { + if (irq > 0) { err = enable_irq_wake(irq); if (err) { dev_err(dev, @@ -641,7 +641,7 @@ static int __maybe_unused mtk8250_resume(struct device *dev) struct mtk8250_data *data = dev_get_drvdata(dev); int irq = data->rx_wakeup_irq; - if (irq >= 0) + if (irq > 0) disable_irq_wake(irq); pinctrl_pm_select_default_state(dev); diff --git a/drivers/tty/serial/sh-sci.c b/drivers/tty/serial/sh-sci.c index 968967d722d4..96c3146ff6a4 100644 --- a/drivers/tty/serial/sh-sci.c +++ b/drivers/tty/serial/sh-sci.c @@ -1915,7 +1915,7 @@ static int sci_request_irq(struct sci_port *port) * Certain port types won't support all of the * available interrupt sources. */ - if (unlikely(irq < 0)) + if (unlikely(irq <= 0)) continue; } @@ -1963,7 +1963,7 @@ static void sci_free_irq(struct sci_port *port) * Certain port types won't support all of the available * interrupt sources. */ - if (unlikely(irq < 0)) + if (unlikely(irq <= 0)) continue; /* Check if already freed (irq was muxed) */ @@ -2875,7 +2875,7 @@ static int sci_init_single(struct platform_device *dev, if (sci_port->irqs[0] < 0) return -ENXIO; - if (sci_port->irqs[1] < 0) + if (sci_port->irqs[1] <= 0) for (i = 1; i < ARRAY_SIZE(sci_port->irqs); i++) sci_port->irqs[i] = sci_port->irqs[0]; diff --git a/drivers/uio/uio_pdrv_genirq.c b/drivers/uio/uio_pdrv_genirq.c index 63258b6accc4..7fd275fc6ceb 100644 --- a/drivers/uio/uio_pdrv_genirq.c +++ b/drivers/uio/uio_pdrv_genirq.c @@ -162,7 +162,7 @@ static int uio_pdrv_genirq_probe(struct platform_device *pdev) if (!uioinfo->irq) { ret = platform_get_irq_optional(pdev, 0); uioinfo->irq = ret; - if (ret == -ENXIO) + if (!ret) uioinfo->irq = UIO_IRQ_NONE; else if (ret == -EPROBE_DEFER) return ret; diff --git a/drivers/vfio/platform/vfio_platform.c b/drivers/vfio/platform/vfio_platform.c index 68a1c87066d7..f7423d10cefd 100644 --- a/drivers/vfio/platform/vfio_platform.c +++ b/drivers/vfio/platform/vfio_platform.c @@ -32,8 +32,9 @@ static struct resource *get_platform_resource(struct vfio_platform_device *vdev, static int get_platform_irq(struct vfio_platform_device *vdev, int i) { struct platform_device *pdev = (struct platform_device *) vdev->opaque; + int ret = platform_get_irq_optional(pdev, i); - return platform_get_irq_optional(pdev, i); + return ret ? ret : -ENXIO; } static int vfio_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 5cb58929090d..ff19c5130459 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -643,7 +643,7 @@ static int dw_i2s_probe(struct platform_device *pdev) dev->dev = &pdev->dev; irq = platform_get_irq_optional(pdev, 0); - if (irq >= 0) { + if (irq > 0) { ret = devm_request_irq(&pdev->dev, irq, i2s_irq_handler, 0, pdev->name, dev); if (ret < 0) { @@ -697,7 +697,7 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (!pdata) { - if (irq >= 0) { + if (irq > 0) { ret = dw_pcm_register(pdev); dev->use_pio = true; } else { -- 2.26.3

4 7

[PATCH] ALSA: hda/realtek: Fix deadlock by COEF mutex
by Takashi Iwai 14 Feb '22

14 Feb '22

The recently introduced coef_mutex for Realtek codec seems causing a deadlock when the relevant code is invoked from the power-off state; then the HD-audio core tries to power-up internally, and this kicks off the codec runtime PM code that tries to take the same coef_mutex. In order to avoid the deadlock, do the temporary power up/down around the coef_mutex acquisition and release. This assures that the power-up sequence runs before the mutex, hence no re-entrance will happen. Fixes: b837a9f5ab3b ("ALSA: hda: realtek: Fix race at concurrent COEF updates") Reported-and-tested-by: Julian Wollrath <jwollrath(a)web.de> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20220214132838.4db10fca@schienar Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 39 +++++++++++++++++++++-------------- 1 file changed, 24 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9473fb76ff19..3a42457984e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -138,6 +138,22 @@ struct alc_spec { * COEF access helper functions */ +static void coef_mutex_lock(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + snd_hda_power_up_pm(codec); + mutex_lock(&spec->coef_mutex); +} + +static void coef_mutex_unlock(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + mutex_unlock(&spec->coef_mutex); + snd_hda_power_down_pm(codec); +} + static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx) { @@ -151,12 +167,11 @@ static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx) { - struct alc_spec *spec = codec->spec; unsigned int val; - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); val = __alc_read_coefex_idx(codec, nid, coef_idx); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); return val; } @@ -173,11 +188,9 @@ static void __alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int coef_val) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); __alc_write_coefex_idx(codec, nid, coef_idx, coef_val); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } #define alc_write_coef_idx(codec, coef_idx, coef_val) \ @@ -198,11 +211,9 @@ static void alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int mask, unsigned int bits_set) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); __alc_update_coefex_idx(codec, nid, coef_idx, mask, bits_set); - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } #define alc_update_coef_idx(codec, coef_idx, mask, bits_set) \ @@ -235,9 +246,7 @@ struct coef_fw { static void alc_process_coef_fw(struct hda_codec *codec, const struct coef_fw *fw) { - struct alc_spec *spec = codec->spec; - - mutex_lock(&spec->coef_mutex); + coef_mutex_lock(codec); for (; fw->nid; fw++) { if (fw->mask == (unsigned short)-1) __alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val); @@ -245,7 +254,7 @@ static void alc_process_coef_fw(struct hda_codec *codec, __alc_update_coefex_idx(codec, fw->nid, fw->idx, fw->mask, fw->val); } - mutex_unlock(&spec->coef_mutex); + coef_mutex_unlock(codec); } /* -- 2.31.1

1 0

[PATCH] ALSA: usb-audio: Don't abort resume upon errors
by Takashi Iwai 14 Feb '22

14 Feb '22

The default mixer resume code treats the errors at restoring the modified mixer items as a fatal error, and it returns back to the caller. This ends up in the resume failure, and the device will be come unavailable, although basically those errors are intermittent and can be safely ignored. The problem itself has been present from the beginning, but it didn't hit usually because the code tries to resume only the modified items. But now with the recent commit to forcibly initialize each item at the probe time, the problem surfaced more often, hence it appears as a regression. This patch fixes the regression simply by ignoring the errors at resume. Fixes: b96681bd5827 ("ALSA: usb-audio: Initialize every feature unit once at probe time") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215561 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 630766ba259f..a5641956ef10 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3678,17 +3678,14 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) err = snd_usb_set_cur_mix_value(cval, c + 1, idx, cval->cache_val[idx]); if (err < 0) - return err; + break; } idx++; } } else { /* master */ - if (cval->cached) { - err = snd_usb_set_cur_mix_value(cval, 0, 0, *cval->cache_val); - if (err < 0) - return err; - } + if (cval->cached) + snd_usb_set_cur_mix_value(cval, 0, 0, *cval->cache_val); } return 0; -- 2.31.1

1 0

[PATCH] ASoC: SOF: compr: Mark snd_compress_ops static
by Daniel Baluta 14 Feb '22

14 Feb '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> Functions won't be directly used outside of compress.c file so mark them as static. This will also fix warnings reported by kernel test robot: >> sound/soc/sof/compress.c:91:5: warning: no previous prototype for function 'sof_compr_open' [-Wmissing-prototypes] int sof_compr_open(struct snd_soc_component *component, ^ sound/soc/sof/compress.c:91:1: note: declare 'static' if the function is not intended to be used outside of this translation unit int sof_compr_open(struct snd_soc_component *component, Fixes: 6324cf901e14 ("SoC: SOF: compr: Add compress ops implementation") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/compress.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 2137d1d65dd9..2af8d75204e9 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -89,8 +89,8 @@ static int create_page_table(struct snd_soc_component *component, spcm->stream[dir].page_table.area, size); } -int sof_compr_open(struct snd_soc_component *component, - struct snd_compr_stream *cstream) +static int sof_compr_open(struct snd_soc_component *component, + struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_compr_runtime *crtd = cstream->runtime; @@ -125,8 +125,8 @@ int sof_compr_open(struct snd_soc_component *component, return 0; } -int sof_compr_free(struct snd_soc_component *component, - struct snd_compr_stream *cstream) +static int sof_compr_free(struct snd_soc_component *component, + struct snd_compr_stream *cstream) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_compr_tstamp *tstamp = cstream->runtime->private_data; @@ -159,8 +159,8 @@ int sof_compr_free(struct snd_soc_component *component, return ret; } -int sof_compr_set_params(struct snd_soc_component *component, - struct snd_compr_stream *cstream, struct snd_compr_params *params) +static int sof_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_compr_params *params) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -222,8 +222,8 @@ int sof_compr_set_params(struct snd_soc_component *component, return 0; } -int sof_compr_get_params(struct snd_soc_component *component, - struct snd_compr_stream *cstream, struct snd_codec *params) +static int sof_compr_get_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_codec *params) { /* TODO: we don't query the supported codecs for now, if the * application asks for an unsupported codec the set_params() will fail. @@ -231,8 +231,8 @@ int sof_compr_get_params(struct snd_soc_component *component, return 0; } -int sof_compr_trigger(struct snd_soc_component *component, - struct snd_compr_stream *cstream, int cmd) +static int sof_compr_trigger(struct snd_soc_component *component, + struct snd_compr_stream *cstream, int cmd) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_pcm_runtime *rtd = cstream->private_data; @@ -271,9 +271,9 @@ int sof_compr_trigger(struct snd_soc_component *component, &reply, sizeof(reply)); } -int sof_compr_copy(struct snd_soc_component *component, - struct snd_compr_stream *cstream, - char __user *buf, size_t count) +static int sof_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + char __user *buf, size_t count) { struct snd_compr_runtime *rtd = cstream->runtime; unsigned int offset, n; -- 2.27.0

2 1

[PATCH 1/3] ASoC: wm_adsp: Correct control read size when parsing compressed buffer
by Charles Keepax 14 Feb '22

14 Feb '22

When parsing the compressed stream the whole buffer descriptor is now read in a single cs_dsp_coeff_read_ctrl; on older firmwares this descriptor is just 4 bytes but on more modern firmwares it is 24 bytes. The current code reads the full 24 bytes regardless, this was working but reading junk for the last 20 bytes. However commit f444da38ac92 ("firmware: cs_dsp: Add offset to cs_dsp read/write") added a size check into cs_dsp_coeff_read_ctrl, causing the older firmwares to now return an error. Update the code to only read the amount of data appropriate for the firmware loaded. Fixes: 04ae08596737 ("ASoC: wm_adsp: Switch to using wm_coeff_read_ctrl for compressed buffers") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3672e3d1703e..0582585236a28 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1441,7 +1441,8 @@ static int wm_adsp_buffer_parse_coeff(struct cs_dsp_coeff_ctl *cs_ctl) int ret, i; for (i = 0; i < 5; ++i) { - ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, &coeff_v1, sizeof(coeff_v1)); + ret = cs_dsp_coeff_read_ctrl(cs_ctl, 0, &coeff_v1, + min(cs_ctl->len, sizeof(coeff_v1))); if (ret < 0) return ret; -- 2.11.0

2 4

[PATCH] ALSA: pcm: accept the OPEN state for snd_pcm_stop()
by Jaroslav Kysela 14 Feb '22

14 Feb '22

It may be useful to completely invalidate streaming under some circumstances like the USB gadget detach. This case is a bit different than the complete disconnection. The applications should be notified that the PCM streaming is no longer available, but the recovery may be expected. This patch adds support for SNDRV_PCM_STATE_OPEN passed to snd_pcm_stop() function. Only the hw_free ioctl is allowed to free the buffers in this state for a full recovery operation or the PCM file handle must be closed. In the OPEN state, ioctls return EBADFD error (with the added hw_free exception above). The applications which are not aware about this new state transition (and recovery) will fail with an error. This operation is expected. Link: https://lore.kernel.org/alsa-devel/4e17c99b-1d8a-8ebe-972c-9b1299952ff8@ivi… Cc: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/pcm.h | 1 + sound/core/pcm.c | 1 + sound/core/pcm_native.c | 12 +++++++++++- 3 files changed, 13 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 33451f8ff755..4de1c2c91109 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -467,6 +467,7 @@ struct snd_pcm_substream { /* -- assigned files -- */ int ref_count; atomic_t mmap_count; + atomic_t queued_hw_free; unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); struct pid *pid; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6fd3677685d7..8dc7e99b2113 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -694,6 +694,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) snd_pcm_group_init(&substream->self_group); list_add_tail(&substream->link_list, &substream->self_group.substreams); atomic_set(&substream->mmap_count, 0); + atomic_set(&substream->queued_hw_free, 0); prev = substream; } return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 621883e71194..118ad3f26f4a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -686,10 +686,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: + if (atomic_read(&substream->queued_hw_free)) + goto __badfd; case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: break; default: +__badfd: snd_pcm_stream_unlock_irq(substream); return -EBADFD; } @@ -829,6 +832,7 @@ static int do_hw_free(struct snd_pcm_substream *substream) result = substream->ops->hw_free(substream); if (substream->managed_buffer_alloc) snd_pcm_lib_free_pages(substream); + atomic_set(&substream->queued_hw_free, 0); return result; } @@ -1454,6 +1458,8 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); + if (state == SNDRV_PCM_STATE_OPEN) + atomic_set(&substream->queued_hw_free, 1); } static const struct action_ops snd_pcm_action_stop = { @@ -1469,6 +1475,9 @@ static const struct action_ops snd_pcm_action_stop = { * * The state of each stream is then changed to the given state unconditionally. * + * If the requested state is OPEN, the stream is invalidated and + * the application must call hw_free to recover the operation. + * * Return: Zero if successful, or a negative error code. */ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) @@ -2637,7 +2646,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN || + atomic_read(&substream->queued_hw_free)) do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; -- 2.33.1

5 7

[PATCH 1/2] ALSA: hda: Fix regression on forced probe mask option
by Takashi Iwai 14 Feb '22

14 Feb '22

The forced probe mask via probe_mask 0x100 bit doesn't work any longer as expected since the bus init code was moved and it's clearing the codec_mask value that was set beforehand. This patch fixes the long-time regression by moving the check_probe_mask() call. Fixes: a41d122449be ("ALSA: hda - Embed bus into controller object") Reported-by: dmummenschanz(a)web.de Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/trinity-f018660b-95c9-442b-a2a8-c92a56eb07ed-1644… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4b0338c4c543..18b795220b52 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1798,8 +1798,6 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, assign_position_fix(chip, check_position_fix(chip, position_fix[dev])); - check_probe_mask(chip, dev); - if (single_cmd < 0) /* allow fallback to single_cmd at errors */ chip->fallback_to_single_cmd = 1; else /* explicitly set to single_cmd or not */ @@ -1825,6 +1823,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, chip->bus.core.needs_damn_long_delay = 1; } + check_probe_mask(chip, dev); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); if (err < 0) { dev_err(card->dev, "Error creating device [card]!\n"); -- 2.31.1

1 1

Cannon Lake PCH cAVS (ALC892) not detected by kernel driver
by dmummenschanz＠web.de 14 Feb '22

14 Feb '22

Hello, the audio chip Cannon Lake PCH cAVS ALC892 does not get recognized on my system. Only the HDMI devices are recognized. lspci shows the following audio device: 00:1f.3 Audio device: Intel Corporation Cannon Lake PCH cAVS (rev 10) dmesg only shows: [ 189.559786] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_exit [i915]) [ 189.581395] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input20 [ 189.581503] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input21 [ 189.581628] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input22 [ 189.581711] input: HDA Intel PCH HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input23 [ 189.581849] input: HDA Intel PCH HDMI/DP,pcm=10 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input24 ALSA-info file: http://alsa-project.org/db/?f=96102ee84f258d1cac83ed33795356550181511c There is a workaround availble: echo auto | tee /sys/devices/pci0000\:00/0000\:00\:1f.3/power/control echo 1 | tee /sys/devices/pci0000\:00/0000\:00\:1f.3/remove echo 1 | tee /sys/bus/pci/rescan unfortunately this doesn't work for me. The device never shows up. Sound on winX works fine on this device btw. I'm happy to assist with any attempts to make the sound work.

3 13

[PATCH 0/4] ALSA: hda: Expose codec organization functions
by Cezary Rojewski 14 Feb '22

14 Feb '22

Changes expose several function that are currently unavailable for HDA-DSP drivers for use. Those functions are: snd_hda_codec_cleanup_for_unbind() snd_hda_codec_set_power_save() snd_hda_codec_register() snd_hda_codec_unregister() snd_hda_codec_device_init() This allows upcoming AVS driver [1] to re-use even mode of HDA related code that is currently available in sound/pci/hda and sound/hda and prevent any code duplication within avs-driver that would otherwise had to happen. Last patch in the series provides snd_hdac_ext_bus_link_at() - a helper function which allows for retrieval of HDA segment (link) based on codec address directly. This is simpler than parsing codec-name first to extract the address what is the case for snd_hdac_ext_bus_get_link(). The latter function is updated to re-use newly added one so core logic is not duplicated after the addition. [1]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (4): ALSA: hda: Add snd_hdac_ext_bus_link_at() helper ALSA: hda: Update and expose snd_hda_codec_device_init() ALSA: hda: Update and expose codec register procedures ALSA: hda: Expose codec cleanup and power-save functions include/sound/hda_codec.h | 11 +++- include/sound/hdaudio_ext.h | 1 + sound/hda/ext/hdac_ext_controller.c | 31 +++++++--- sound/pci/hda/hda_codec.c | 94 ++++++++++++++++++++--------- sound/pci/hda/hda_local.h | 2 - sound/soc/codecs/hdac_hda.c | 2 +- 6 files changed, 100 insertions(+), 41 deletions(-) -- 2.25.1

4 18

[PATCH] ALSA: hda/realtek: Add quirk for Legion Y9000X 2019
by Yu Huang 14 Feb '22

14 Feb '22

Legion Y9000X 2019 has the same speaker with Y9000X 2020, but with a different quirk address. Add one quirk entry to make the speaker work on Y9000X 2019 too. Signed-off-by: Yu Huang <diwang90(a)gmail.com> --- 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 8315bf7d4c38..9473fb76ff19 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9170,6 +9170,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x383d, "Legion Y9000X 2019", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), SND_PCI_QUIRK(0x17aa, 0x3847, "Legion 7 16ACHG6", ALC287_FIXUP_LEGION_16ACHG6), SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), -- 2.34.1

2 1

Re: [PATCH 1/1] ALSA: usb-audio: revert to IMPLICIT_FB_FIXED_DEV for M-Audio FastTrack Ultra
by Takashi Iwai 14 Feb '22

14 Feb '22

On Fri, 11 Feb 2022 23:49:13 +0100, Matteo Martelli wrote: > > Commit 83b7dcbc51c930fc2079ab6c6fc9d719768321f1 introduced a generic > implicit feedback parser, which fails to execute for M-Audio FastTrack > Ultra sound cards. The issue is with the ENDPOINT_SYNCTYPE check in > add_generic_implicit_fb() where the SYNCTYPE is ADAPTIVE instead of ASYNC. > The reason is that the sync type of the FastTrack output endpoints are > set to adaptive in the quirks table since commit > 65f04443c96dbda11b8fff21d6390e082846aa3c. > > Signed-off-by: Matteo Martelli <matteomartelli3(a)gmail.com> Thanks, applied now. Takashi

1 0

[PATCH v12 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 14 Feb '22

14 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V11: -- Fix kernel robot issue on arguments type mismatch. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 304 ++++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 447 +++++++++++++++ sound/soc/qcom/lpass.h | 145 +++++ 9 files changed, 1887 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

1 1

[PATCH] ASoC: SOF: Makefile: Fix randconfig sof-client build when SND_SOC_SOF=y
by Peter Ujfalusi 14 Feb '22

14 Feb '22

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Intel's kernel test robot found the following randconfig combination: SND_SOC_SOF=y SND_SOC_SOF_CLIENT=m In this the sof-client object is not going to be built into the snd-sof.o and we will have undefined references to the sof-client functions. Fixes: 6955d9512d0e ("ASoC: SOF: Introduce IPC SOF client support") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> --- Hi Mark, Pierre, When the client support was added we did some permutation tests (at least I did) but missed this combination. It is a valid one for sure and the proper fix is via the Makefile. Sorry for mising this Regards, Peter sound/soc/sof/Makefile | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 4d31282c847d..a0459f06c68a 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,7 +2,9 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o iomem-utils.o sof-audio.o stream-ipc.o -snd-sof-$(CONFIG_SND_SOC_SOF_CLIENT) += sof-client.o +ifneq ($(CONFIG_SND_SOC_SOF_CLIENT),) +snd-sof-objs += sof-client.o +endif snd-sof-$(CONFIG_SND_SOC_SOF_COMPRESS) += compress.o -- 2.35.1

1 0

[PATCH 23/23] ALSA: hda/realtek: Make use of the helper component_compare_dev_name
by Yong Wu 14 Feb '22

14 Feb '22

Use the common compare helper from component. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jeremy Szu <jeremy.szu(a)canonical.com> Cc: Werner Sembach <wse(a)tuxedocomputers.com> Cc: Hui Wang <hui.wang(a)canonical.com> Cc: Cameron Berkenpas <cam(a)neo-zeon.de> Cc: Kailang Yang <kailang(a)realtek.com> Cc: Lucas Tanure <tanureal(a)opensource.cirrus.com> Cc: Sami Loone <sami(a)loone.fi> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Yong Wu <yong.wu(a)mediatek.com> --- sound/pci/hda/patch_realtek.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 668274e52674..9da004d99cdb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6530,11 +6530,6 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, } } -static int comp_match_dev_name(struct device *dev, void *data) -{ - return strcmp(dev_name(dev), data) == 0; -} - static int find_comp_by_dev_name(struct alc_spec *spec, const char *name) { int i; @@ -6595,7 +6590,7 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char "%s-%s:00-cs35l41-hda.%d", bus, hid, i); if (!name) return; - component_match_add(dev, &spec->match, comp_match_dev_name, name); + component_match_add(dev, &spec->match, component_compare_dev_name, name); } ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); if (ret) @@ -6644,9 +6639,9 @@ static void alc287_fixup_legion_16achg6_speakers(struct hda_codec *cdc, const st switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - component_match_add(dev, &spec->match, comp_match_dev_name, + component_match_add(dev, &spec->match, component_compare_dev_name, "i2c-CLSA0100:00-cs35l41-hda.0"); - component_match_add(dev, &spec->match, comp_match_dev_name, + component_match_add(dev, &spec->match, component_compare_dev_name, "i2c-CLSA0100:00-cs35l41-hda.1"); ret = component_master_add_with_match(dev, &comp_master_ops, spec->match); if (ret) -- 2.18.0

1 0

[PATCH v11 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 13 Feb '22

13 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 304 ++++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 447 +++++++++++++++ sound/soc/qcom/lpass.h | 145 +++++ 9 files changed, 1887 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

2 11

Conflictuous devices name
by GitHub issues - opened 12 Feb '22

12 Feb '22

alsa-project/alsa-lib issue #212 was opened from nm2107: Hello, I'm using [mixxx](https://github.com/mixxxdj/mixxx) (which uses [portmidi](https://github.com/PortMidi/portmidi), which uses ALSA) with a pair of Denon SC3900 decks connected via USB MIDI. However, I'm running into an issue where I cannot distinguish both SC3900 units anymore, as they are reported by alsa with the same name : ```bash $ amidi -l Dir Device Name IO hw:2,0,0 SC3900 MIDI 1 IO hw:3,0,0 SC3900 MIDI 1 ``` (they were previously differentiated by `SC3900 MIDI 1` and `SC3900 MIDI 2` names). PortMIDI (and so mixxx) are relying on the device name to identify them, so I'm not able to distinguish the two plugged in units (only one device is listed in mixxx, as the name is common). I'm using alsa-libs 1.2.6.1. I tried to downgrade to version 1.2.5.1, but is hasn't solved my issue. Along with this downgrade, I also tried to downgrade my linux kernel to v5.13 (I was on v5.16), but it hasn't shown any improvement. Do you have an idea of what may cause this naming conflict ? Thank you :) Issue URL : https://github.com/alsa-project/alsa-lib/issues/212 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/3] firewire: assist unit driver to compute packet time stamp
by Takashi Sakamoto 12 Feb '22

12 Feb '22

Hi, Current implementation of Linux FireWire subsystem doesn't allow unit driver to operate content of packet in IR context according to time stamp. Additionally it doesn't allow unit driver to read current value of CYCLE_TIME register in OHCI 1394 controller. It brings disadvantages to drivers in Linux sound subsystem in regards of handling time for sampled data such as PCM frames and MIDI messages. This rerolled patchset is first step to improve the situation. Changes in v2: * Rebase v1 patchset to v5.16 release V1: * https://lore.kernel.org/lkml/20211202113457.24011-1-o-takashi@sakamocchi.jp/ Hector Martin (1): firewire: Add dummy read_csr/write_csr functions Takashi Sakamoto (2): firewire: add kernel API to access CYCLE_TIME register firewire: add kernel API to access packet structure in request structure for AR context drivers/firewire/core-card.c | 39 +++++++++++++++++++++++++++++ drivers/firewire/core-cdev.c | 6 +++-- drivers/firewire/core-transaction.c | 18 +++++++++++++ include/linux/firewire.h | 3 +++ 4 files changed, 64 insertions(+), 2 deletions(-) -- 2.32.0

1 3

[PATCH] ALSA: cleanup double word in comment
by trix＠redhat.com 11 Feb '22

11 Feb '22

From: Tom Rix <trix(a)redhat.com> Remove the second 'device'. Signed-off-by: Tom Rix <trix(a)redhat.com> --- include/sound/hda_verbs.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h index e36b77531c5c3..006d358acce25 100644 --- a/include/sound/hda_verbs.h +++ b/include/sound/hda_verbs.h @@ -461,7 +461,7 @@ enum { #define AC_DE_ELDV (1<<1) #define AC_DE_IA (1<<2) -/* device device types (0x0-0xf) */ +/* device types (0x0-0xf) */ enum { AC_JACK_LINE_OUT, AC_JACK_SPEAKER, -- 2.26.3

2 1

[PATCH] ALSA: hda: Add PCI and HDMI IDs for Intel Raptor Lake
by Kai Vehmanen 11 Feb '22

11 Feb '22

Add a set of HD Audio PCI IDs, and the HDMI codec VID, for Intel Raptor Lake. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 11 +++++++++++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a2922233e85f..a3c4d6e293e1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2510,6 +2510,17 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x4b58), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Raptor Lake */ + { PCI_DEVICE(0x8086, 0x7a50), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x51ca), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x51cb), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x51ce), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x51cf), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 92df4f243ec6..64fe025fda86 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4390,6 +4390,7 @@ HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862818, "Raptorlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), base-commit: 44cb545b0c2afe21b38044b3595a2579f3a8d7b2 -- 2.35.1

3 2

Re: [PATCH 31/31] sound: soc: sof: changing LED_* from enum led_brightness to actual value
by Mark Brown 11 Feb '22

11 Feb '22

On Fri, Jan 21, 2022 at 01:54:36PM -0300, Luiz Sampaio wrote: > The enum led_brightness, which contains the declaration of LED_OFF, > LED_ON, LED_HALF and LED_FULL is obsolete, as the led class now supports > max_brightness. > --- You've not provided a Signed-off-by for this so I can't do anything with it, please see Documentation/process/submitting-patches.rst for details on what this is and why it's important. Please submit patches using subject lines reflecting the style for the subsystem, this makes it easier for people to identify relevant patches. Look at what existing commits in the area you're changing are doing and make sure your subject lines visually resemble what they're doing. There's no need to resubmit to fix this alone.

1 0

[PATCH 00/19] soundwire: stream: cleanup of 'stream' support
by Bard Liao 11 Feb '22

11 Feb '22

This series revisits the SoundWire 'sdw_stream' support to split allocation and configuration steps. This is necessary if for example the routines are called multiple times from the hw_params stage. This also helps with better error handling. Pierre-Louis Bossart (19): soundwire: stream: remove unused parameter in sdw_stream_add_slave soundwire: stream: add slave runtime to list earlier soundwire: stream: simplify check on port range soundwire: stream: add alloc/config/free helpers for ports soundwire: stream: split port allocation and configuration loops soundwire: stream: split alloc and config in two functions soundwire: stream: add 'slave' prefix for port range checks soundwire: stream: group sdw_port and sdw_master/slave_port functions soundwire: stream: simplify sdw_alloc_master_rt() soundwire: stream: split sdw_alloc_master_rt() in alloc and config soundwire: stream: move sdw_alloc_slave_rt() before 'master' helpers soundwire: stream: split sdw_alloc_slave_rt() in alloc and config soundwire: stream: group sdw_stream_ functions soundwire: stream: rename and move master/slave_rt_free routines soundwire: stream: move list addition to sdw_slave_alloc_rt() soundwire: stream: separate alloc and config within sdw_stream_add_xxx() soundwire: stream: introduce sdw_slave_rt_find() helper soundwire: stream: sdw_stream_add_ functions can be called multiple times soundwire: stream: make enable/disable/deprepare idempotent drivers/soundwire/stream.c | 960 +++++++++++++++++++++---------------- 1 file changed, 547 insertions(+), 413 deletions(-) -- 2.17.1

2 20

[PATCH] ASoC: tlv320adc3xxx: Add IIR filter configuration
by Ricard Wanderlof 10 Feb '22

10 Feb '22

The TLV320ADC3001/3101 have an internal DSP, which can either be used in various preset configurations (called "Processing Blocks" in the data sheet), or as a freely programmable (using the "PurePath Studio" graphical programming tool from TI) but rather small DSP ("miniDSP"). Using the default configuration (PRB_R1) it's possible to set up filtering using a first-order IIR, which can be useful for adding a digital high pass filter to the signal chain, for instance. This patch adds support for configuring the IIR filter coefficients. The filter itself is always enabled; the default coefficients implement a pass-through function. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/tlv320adc3xxx.c | 160 +++++++++++++++++++++++++++++-- 1 file changed, 153 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index 4baf3d881633..d0d5206db0c7 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -169,6 +169,23 @@ #define ADC3XXX_ANALOG_PGA_FLAGS ADC3XXX_REG(1, 62) /* 63-127 Reserved */ +/* + * Page 4 registers. First page of coefficient memory for the miniDSP. + */ +#define ADC3XXX_LEFT_ADC_IIR_COEFF_N0_MSB ADC3XXX_REG(4, 8) +#define ADC3XXX_LEFT_ADC_IIR_COEFF_N0_LSB ADC3XXX_REG(4, 9) +#define ADC3XXX_LEFT_ADC_IIR_COEFF_N1_MSB ADC3XXX_REG(4, 10) +#define ADC3XXX_LEFT_ADC_IIR_COEFF_N1_LSB ADC3XXX_REG(4, 11) +#define ADC3XXX_LEFT_ADC_IIR_COEFF_D1_MSB ADC3XXX_REG(4, 12) +#define ADC3XXX_LEFT_ADC_IIR_COEFF_D1_LSB ADC3XXX_REG(4, 13) + +#define ADC3XXX_RIGHT_ADC_IIR_COEFF_N0_MSB ADC3XXX_REG(4, 72) +#define ADC3XXX_RIGHT_ADC_IIR_COEFF_N0_LSB ADC3XXX_REG(4, 73) +#define ADC3XXX_RIGHT_ADC_IIR_COEFF_N1_MSB ADC3XXX_REG(4, 74) +#define ADC3XXX_RIGHT_ADC_IIR_COEFF_N1_LSB ADC3XXX_REG(4, 75) +#define ADC3XXX_RIGHT_ADC_IIR_COEFF_D1_MSB ADC3XXX_REG(4, 76) +#define ADC3XXX_RIGHT_ADC_IIR_COEFF_D1_LSB ADC3XXX_REG(4, 77) + /* * Register bits. */ @@ -373,22 +390,63 @@ static const struct reg_default adc3xxx_defaults[] = { { 180, 0xff }, { 181, 0x00 }, { 182, 0x3f }, { 183, 0xff }, { 184, 0x00 }, { 185, 0x3f }, { 186, 0x00 }, { 187, 0x80 }, { 188, 0x80 }, { 189, 0x00 }, { 190, 0x00 }, { 191, 0x00 }, + + /* Page 4 */ + { 1024, 0x00 }, { 1026, 0x01 }, { 1027, 0x17 }, + { 1028, 0x01 }, { 1029, 0x17 }, { 1030, 0x7d }, { 1031, 0xd3 }, + { 1032, 0x7f }, { 1033, 0xff }, { 1034, 0x00 }, { 1035, 0x00 }, + { 1036, 0x00 }, { 1037, 0x00 }, { 1038, 0x7f }, { 1039, 0xff }, + { 1040, 0x00 }, { 1041, 0x00 }, { 1042, 0x00 }, { 1043, 0x00 }, + { 1044, 0x00 }, { 1045, 0x00 }, { 1046, 0x00 }, { 1047, 0x00 }, + { 1048, 0x7f }, { 1049, 0xff }, { 1050, 0x00 }, { 1051, 0x00 }, + { 1052, 0x00 }, { 1053, 0x00 }, { 1054, 0x00 }, { 1055, 0x00 }, + { 1056, 0x00 }, { 1057, 0x00 }, { 1058, 0x7f }, { 1059, 0xff }, + { 1060, 0x00 }, { 1061, 0x00 }, { 1062, 0x00 }, { 1063, 0x00 }, + { 1064, 0x00 }, { 1065, 0x00 }, { 1066, 0x00 }, { 1067, 0x00 }, + { 1068, 0x7f }, { 1069, 0xff }, { 1070, 0x00 }, { 1071, 0x00 }, + { 1072, 0x00 }, { 1073, 0x00 }, { 1074, 0x00 }, { 1075, 0x00 }, + { 1076, 0x00 }, { 1077, 0x00 }, { 1078, 0x7f }, { 1079, 0xff }, + { 1080, 0x00 }, { 1081, 0x00 }, { 1082, 0x00 }, { 1083, 0x00 }, + { 1084, 0x00 }, { 1085, 0x00 }, { 1086, 0x00 }, { 1087, 0x00 }, + { 1088, 0x00 }, { 1089, 0x00 }, { 1090, 0x00 }, { 1091, 0x00 }, + { 1092, 0x00 }, { 1093, 0x00 }, { 1094, 0x00 }, { 1095, 0x00 }, + { 1096, 0x00 }, { 1097, 0x00 }, { 1098, 0x00 }, { 1099, 0x00 }, + { 1100, 0x00 }, { 1101, 0x00 }, { 1102, 0x00 }, { 1103, 0x00 }, + { 1104, 0x00 }, { 1105, 0x00 }, { 1106, 0x00 }, { 1107, 0x00 }, + { 1108, 0x00 }, { 1109, 0x00 }, { 1110, 0x00 }, { 1111, 0x00 }, + { 1112, 0x00 }, { 1113, 0x00 }, { 1114, 0x00 }, { 1115, 0x00 }, + { 1116, 0x00 }, { 1117, 0x00 }, { 1118, 0x00 }, { 1119, 0x00 }, + { 1120, 0x00 }, { 1121, 0x00 }, { 1122, 0x00 }, { 1123, 0x00 }, + { 1124, 0x00 }, { 1125, 0x00 }, { 1126, 0x00 }, { 1127, 0x00 }, + { 1128, 0x00 }, { 1129, 0x00 }, { 1130, 0x00 }, { 1131, 0x00 }, + { 1132, 0x00 }, { 1133, 0x00 }, { 1134, 0x00 }, { 1135, 0x00 }, + { 1136, 0x00 }, { 1137, 0x00 }, { 1138, 0x00 }, { 1139, 0x00 }, + { 1140, 0x00 }, { 1141, 0x00 }, { 1142, 0x00 }, { 1143, 0x00 }, + { 1144, 0x00 }, { 1145, 0x00 }, { 1146, 0x00 }, { 1147, 0x00 }, + { 1148, 0x00 }, { 1149, 0x00 }, { 1150, 0x00 }, { 1151, 0x00 }, }; static bool adc3xxx_volatile_reg(struct device *dev, unsigned int reg) { - switch (reg) { - case ADC3XXX_RESET: + if (reg == ADC3XXX_RESET) return true; - default: - return false; - } + + /* + * Coefficient RAM registers for miniDSP are marked as volatile + * mainly because they must be written in pairs, so we don't want + * them to be cached. Updates are not likely to occur very often, + * so the performance penalty is minimal. + */ + if (reg >= ADC3XXX_REG(4, 2) && reg <= ADC3XXX_REG(4, 128)) + return true; + + return false; } static const struct regmap_range_cfg adc3xxx_ranges[] = { { .range_min = 0, - .range_max = 2 * ADC3XXX_PAGE_SIZE, + .range_max = 5 * ADC3XXX_PAGE_SIZE, .selector_reg = ADC3XXX_PAGE_SELECT, .selector_mask = 0xff, .selector_shift = 0, @@ -410,7 +468,7 @@ static const struct regmap_config adc3xxx_regmap = { .ranges = adc3xxx_ranges, .num_ranges = ARRAY_SIZE(adc3xxx_ranges), - .max_register = 2 * ADC3XXX_PAGE_SIZE, + .max_register = 5 * ADC3XXX_PAGE_SIZE, }; struct adc3xxx_rate_divs { @@ -497,6 +555,83 @@ static int adc3xxx_pll_delay(struct snd_soc_dapm_widget *w, return 0; } +static int adc3xxx_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoeff = kcontrol->private_value >> 16; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = numcoeff; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xffff; /* all coefficients are 16 bit */ + return 0; +} + +static int adc3xxx_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + int numcoeff = kcontrol->private_value >> 16; + int reg = kcontrol->private_value & 0xffff; + int index = 0; + + while (index < numcoeff) { + unsigned int value_msb, value_lsb, value; + + value_msb = snd_soc_component_read(component, reg++); + if ((int)value_msb < 0) + return (int)value_msb; + + value_lsb = snd_soc_component_read(component, reg++); + if ((int)value_lsb < 0) + return (int)value_lsb; + + value = (value_msb << 8) | value_lsb; + ucontrol->value.integer.value[index++] = value; + } + + return 0; +} + +static int adc3xxx_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + int numcoeff = kcontrol->private_value >> 16; + int reg = kcontrol->private_value & 0xffff; + int index = 0; + int ret; + + while (index < numcoeff) { + unsigned int value = ucontrol->value.integer.value[index++]; + unsigned int value_msb = (value >> 8) & 0xff; + unsigned int value_lsb = value & 0xff; + + ret = snd_soc_component_write(component, reg++, value_msb); + if (ret) + return ret; + + ret = snd_soc_component_write(component, reg++, value_lsb); + if (ret) + return ret; + } + + return 0; +} + +/* All on-chip filters have coefficients which are expressed in terms of + * 16 bit values, so represent them as strings of 16-bit integers. + */ +#define TI_COEFFICIENTS(xname, reg, numcoeffs) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = adc3xxx_coefficient_info, \ + .get = adc3xxx_coefficient_get,\ + .put = adc3xxx_coefficient_put, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .private_value = reg | (numcoeffs << 16) \ +} + static const char * const adc_softstepping_text[] = { "1 step", "2 step", "off" }; static SOC_ENUM_SINGLE_DECL(adc_softstepping_enum, ADC3XXX_ADC_DIGITAL, 0, adc_softstepping_text); @@ -640,6 +775,17 @@ static const struct snd_kcontrol_new adc3xxx_snd_controls[] = { SOC_SINGLE("Right ADC Unselected CM Bias Capture Switch", ADC3XXX_RIGHT_PGA_SEL_2, 6, 1, 0), SOC_ENUM("Dither Control DC Offset", dither_dc_offset_enum), + + /* Coefficient memory for miniDSP. */ + /* For the default PRB_R1 processing block, the only available + * filter is the first order IIR. + */ + + TI_COEFFICIENTS("Left ADC IIR Coefficients N0 N1 D1", + ADC3XXX_LEFT_ADC_IIR_COEFF_N0_MSB, 3), + + TI_COEFFICIENTS("Right ADC IIR Coefficients N0 N1 D1", + ADC3XXX_RIGHT_ADC_IIR_COEFF_N0_MSB, 3), }; /* Left input selection, Single Ended inputs and Differential inputs */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 8

[PATCH v2 0/9] ASoC: SOF: IPC client infrastructure
by Peter Ujfalusi 10 Feb '22

10 Feb '22

Hi, Changes since v1: - Fix compilation error in sound/soc/sof/compress.c: error: implicit declaration of function 'snd_sof_create_page_table' The patch introducing this is not in sof-dev, and I have missed cross compiling for aarch64. Original Cover letter The Linux SOF implementation is historically monolithic in a sense that all features accessible in the firmware can be used via the snd_sof_dev struct in one way or another. Support for features can not be added or removed runtime and with the current way of things it is hard if not impossible to implement support for dynamic feature support when based on the firmware manifest we can easily enable/access independent modules with the SOF. In order to be able to support such modularity this series introduces a small framework within SOF for client support using the Auxiliary bus. Client drivers can be removed runtime and later re-loaded if needed without affecting the core's behaviour, but it is the core's and the platform's duty to create the Auxiliary devices usable in the platform and via the firmware. There is still a need for SOF manifest update to convey information about features to really make the full dynamic client device creation. The series will introduce the core SOF client support and converts the generic ipc flood test, ipc message injector and the probes (Intel HDA only) to a client driver. Regards, Peter --- Peter Ujfalusi (8): ASoC: SOF: Drop unused DSP power states: D3_HOT and D3_COLD ASoC: SOF: Move the definition of enum sof_dsp_power_states to global header ASoC: SOF: ipc: Read and pass the whole message to handlers for IPC events ASoC: SOF: Split up utils.c into sof-utils and iomem-utils ASoC: SOF: Introduce IPC SOF client support ASoC: SOF: sof-client: Add support for clients not managed by pm framework ASoC: SOF: Convert the generic IPC message injector into SOF client ASoC: SOF: Convert the generic probe support to SOF client Ranjani Sridharan (1): ASoC: SOF: Convert the generic IPC flood test into SOF client include/sound/sof.h | 8 + sound/soc/sof/Kconfig | 33 +- sound/soc/sof/Makefile | 15 +- sound/soc/sof/compress.c | 1 + sound/soc/sof/core.c | 50 +- sound/soc/sof/debug.c | 565 -------------- sound/soc/sof/intel/Kconfig | 19 +- sound/soc/sof/intel/apl.c | 13 +- sound/soc/sof/intel/cnl.c | 13 +- sound/soc/sof/intel/hda-dai.c | 19 - sound/soc/sof/intel/hda-dsp.c | 6 - sound/soc/sof/intel/hda-probes.c | 72 +- sound/soc/sof/intel/hda.c | 10 + sound/soc/sof/intel/hda.h | 49 +- sound/soc/sof/intel/icl.c | 13 +- sound/soc/sof/intel/tgl.c | 13 +- sound/soc/sof/{utils.c => iomem-utils.c} | 61 +- sound/soc/sof/ipc.c | 88 ++- sound/soc/sof/ops.h | 43 - sound/soc/sof/pcm.c | 7 +- sound/soc/sof/pm.c | 13 +- sound/soc/sof/sof-client-ipc-flood-test.c | 396 ++++++++++ sound/soc/sof/sof-client-ipc-msg-injector.c | 192 +++++ sound/soc/sof/sof-client-probes.c | 821 ++++++++++++++++++++ sound/soc/sof/sof-client-probes.h | 31 + sound/soc/sof/sof-client.c | 469 +++++++++++ sound/soc/sof/sof-client.h | 67 ++ sound/soc/sof/sof-priv.h | 139 ++-- sound/soc/sof/sof-probes.c | 364 --------- sound/soc/sof/sof-probes.h | 38 - sound/soc/sof/sof-utils.c | 77 ++ sound/soc/sof/sof-utils.h | 19 + sound/soc/sof/trace.c | 1 + 33 files changed, 2416 insertions(+), 1309 deletions(-) rename sound/soc/sof/{utils.c => iomem-utils.c} (59%) create mode 100644 sound/soc/sof/sof-client-ipc-flood-test.c create mode 100644 sound/soc/sof/sof-client-ipc-msg-injector.c create mode 100644 sound/soc/sof/sof-client-probes.c create mode 100644 sound/soc/sof/sof-client-probes.h create mode 100644 sound/soc/sof/sof-client.c create mode 100644 sound/soc/sof/sof-client.h delete mode 100644 sound/soc/sof/sof-probes.c delete mode 100644 sound/soc/sof/sof-probes.h create mode 100644 sound/soc/sof/sof-utils.c create mode 100644 sound/soc/sof/sof-utils.h -- 2.35.1

2 10

[PATCH 0/9] ASoC: SOF: IPC client infrastructure
by Peter Ujfalusi 10 Feb '22

10 Feb '22

Hi, The Linux SOF implementation is historically monolithic in a sense that all features accessible in the firmware can be used via the snd_sof_dev struct in one way or another. Support for features can not be added or removed runtime and with the current way of things it is hard if not impossible to implement support for dynamic feature support when based on the firmware manifest we can easily enable/access independent modules with the SOF. In order to be able to support such modularity this series introduces a small framework within SOF for client support using the Auxiliary bus. Client drivers can be removed runtime and later re-loaded if needed without affecting the core's behaviour, but it is the core's and the platform's duty to create the Auxiliary devices usable in the platform and via the firmware. There is still a need for SOF manifest update to convey information about features to really make the full dynamic client device creation. The series will introduce the core SOF client support and converts the generic ipc flood test, ipc message injector and the probes (Intel HDA only) to a client driver. Regards, Peter --- Peter Ujfalusi (8): ASoC: SOF: Drop unused DSP power states: D3_HOT and D3_COLD ASoC: SOF: Move the definition of enum sof_dsp_power_states to global header ASoC: SOF: ipc: Read and pass the whole message to handlers for IPC events ASoC: SOF: Split up utils.c into sof-utils and iomem-utils ASoC: SOF: Introduce IPC SOF client support ASoC: SOF: sof-client: Add support for clients not managed by pm framework ASoC: SOF: Convert the generic IPC message injector into SOF client ASoC: SOF: Convert the generic probe support to SOF client Ranjani Sridharan (1): ASoC: SOF: Convert the generic IPC flood test into SOF client include/sound/sof.h | 8 + sound/soc/sof/Kconfig | 33 +- sound/soc/sof/Makefile | 15 +- sound/soc/sof/core.c | 50 +- sound/soc/sof/debug.c | 565 -------------- sound/soc/sof/intel/Kconfig | 19 +- sound/soc/sof/intel/apl.c | 13 +- sound/soc/sof/intel/cnl.c | 13 +- sound/soc/sof/intel/hda-dai.c | 19 - sound/soc/sof/intel/hda-dsp.c | 6 - sound/soc/sof/intel/hda-probes.c | 72 +- sound/soc/sof/intel/hda.c | 10 + sound/soc/sof/intel/hda.h | 49 +- sound/soc/sof/intel/icl.c | 13 +- sound/soc/sof/intel/tgl.c | 13 +- sound/soc/sof/{utils.c => iomem-utils.c} | 61 +- sound/soc/sof/ipc.c | 88 ++- sound/soc/sof/ops.h | 43 - sound/soc/sof/pcm.c | 7 +- sound/soc/sof/pm.c | 13 +- sound/soc/sof/sof-client-ipc-flood-test.c | 396 ++++++++++ sound/soc/sof/sof-client-ipc-msg-injector.c | 192 +++++ sound/soc/sof/sof-client-probes.c | 821 ++++++++++++++++++++ sound/soc/sof/sof-client-probes.h | 31 + sound/soc/sof/sof-client.c | 469 +++++++++++ sound/soc/sof/sof-client.h | 67 ++ sound/soc/sof/sof-priv.h | 139 ++-- sound/soc/sof/sof-probes.c | 364 --------- sound/soc/sof/sof-probes.h | 38 - sound/soc/sof/sof-utils.c | 77 ++ sound/soc/sof/sof-utils.h | 19 + sound/soc/sof/trace.c | 1 + 32 files changed, 2415 insertions(+), 1309 deletions(-) rename sound/soc/sof/{utils.c => iomem-utils.c} (59%) create mode 100644 sound/soc/sof/sof-client-ipc-flood-test.c create mode 100644 sound/soc/sof/sof-client-ipc-msg-injector.c create mode 100644 sound/soc/sof/sof-client-probes.c create mode 100644 sound/soc/sof/sof-client-probes.h create mode 100644 sound/soc/sof/sof-client.c create mode 100644 sound/soc/sof/sof-client.h delete mode 100644 sound/soc/sof/sof-probes.c delete mode 100644 sound/soc/sof/sof-probes.h create mode 100644 sound/soc/sof/sof-utils.c create mode 100644 sound/soc/sof/sof-utils.h -- 2.35.1

2 13

[PATCH v2 0/6] ASoC: dt-bindings: samsung: convert to dtschema
by Krzysztof Kozlowski 10 Feb '22

10 Feb '22

Hi, The patchset is based on Rob's sound-dai changes: https://lore.kernel.org/all/20220126231427.1638089-1-robh@kernel.org/ Changes since v1: 1. Correct samsung,snow cpu/sound-dai. Best regards, Krzysztof Krzysztof Kozlowski (6): ASoC: dt-bindings: samsung,aries-wm8994: require sound-dai property ASoC: dt-bindings: samsung,arndale: convert to dtschema ASoC: dt-bindings: samsung,arndale: document ALC5631 ASoC: dt-bindings: samsung,smdk5250: convert to dtschema ASoC: dt-bindings: samsung,snow: convert to dtschema ASoC: dt-bindings: samsung,tm2: convert to dtschema .../devicetree/bindings/sound/arndale.txt | 25 ------ .../bindings/sound/samsung,aries-wm8994.yaml | 4 + .../bindings/sound/samsung,arndale.yaml | 45 +++++++++++ .../bindings/sound/samsung,smdk-wm8994.txt | 14 ---- .../bindings/sound/samsung,smdk5250.yaml | 38 +++++++++ .../bindings/sound/samsung,snow.yaml | 74 +++++++++++++++++ .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2.yaml | 80 +++++++++++++++++++ .../devicetree/bindings/sound/snow.txt | 31 ------- 9 files changed, 241 insertions(+), 112 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,arndale.yaml delete mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml create mode 100644 Documentation/devicetree/bindings/sound/samsung,snow.yaml delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/snow.txt -- 2.32.0

3 13

[PATCH v9 0/3] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 10 Feb '22

10 Feb '22

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (3): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: Add macro for lpass DAI id's max limit ASoC: qcom: SC7280: Add machine driver Changes Since V8: -- Update snd_soc_dai_get_sdw_stream api compatible to latest kernel. -- Fix bot errors in dt-bindings file. -- Add co-developer email id. Changes Since V7: -- Create separate patch for dai id's macro. -- Remove unused dapm widget structure. -- Move to quicinc domain email id's. Changes Since V6: -- Remove redundant headers. -- Move max ports macro to lpass.h header. -- Arrange structure alignment. -- Fix indentation errors. -- Update module license. Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 180 +++++++++++++ sound/soc/qcom/Kconfig | 14 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass.h | 1 + sound/soc/qcom/sc7280.c | 284 +++++++++++++++++++++ 5 files changed, 481 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- 2.7.4

2 4

[PATCH] ASoC: rt5640: Remove the sysclk and sysclk_src checking
by Oder Chiou 10 Feb '22

10 Feb '22

Remove the sysclk and sysclk_src checking in the function set_sysclk() to prevent the PLL power off. It is not getting re-programmed during subsequent runs after the first run (in BIAS_OFF stage). Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e7a82565b905..30c2e7cb7ed2 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1839,9 +1839,6 @@ static int rt5640_set_dai_sysclk(struct snd_soc_dai *dai, unsigned int reg_val = 0; unsigned int pll_bit = 0; - if (freq == rt5640->sysclk && clk_id == rt5640->sysclk_src) - return 0; - switch (clk_id) { case RT5640_SCLK_S_MCLK: reg_val |= RT5640_SCLK_SRC_MCLK; -- 2.35.1

2 1

[RESEND v8 0/3] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 10 Feb '22

10 Feb '22

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (3): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: Add macro for lpass DAI id's max limit ASoC: qcom: SC7280: Add machine driver Changes Since V7: -- Create separate patch for dai id's macro. -- Remove unused dapm widget structure. -- Move to quicinc domain email id's. Changes Since V6: -- Remove redundant headers. -- Move max ports macro to lpass.h header. -- Arrange structure alignment. -- Fix indentation errors. -- Update module license. Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 171 +++++++++++++ sound/soc/qcom/Kconfig | 14 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass.h | 1 + sound/soc/qcom/sc7280.c | 284 +++++++++++++++++++++ 5 files changed, 472 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- 2.7.4

3 6

[PATCH] soundwire: Replace acpi_bus_get_device()
by Rafael J. Wysocki 10 Feb '22

10 Feb '22

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Replace acpi_bus_get_device() that is going to be dropped with acpi_fetch_acpi_dev(). No intentional functional impact. Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- drivers/soundwire/intel_init.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) Index: linux-pm/drivers/soundwire/intel_init.c =================================================================== --- linux-pm.orig/drivers/soundwire/intel_init.c +++ linux-pm/drivers/soundwire/intel_init.c @@ -180,7 +180,8 @@ static struct sdw_intel_ctx if (!res) return NULL; - if (acpi_bus_get_device(res->handle, &adev)) + adev = acpi_fetch_acpi_dev(res->handle); + if (!adev) return NULL; if (!res->count) @@ -294,13 +295,13 @@ err: static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { - struct acpi_device *adev; + struct acpi_device *adev = acpi_fetch_acpi_dev(ctx->handle); struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; - if (acpi_bus_get_device(ctx->handle, &adev)) + if (!adev) return -EINVAL; /* Check SNDWLCAP.LCOUNT */

2 1

[PATCH] soundwire: bus: add dev_warn() messages to track UNATTACHED devices
by Bard Liao 10 Feb '22

10 Feb '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> In rare cases, some devices seem to lose sync and never re-attach on the bus. This seems to happen only when there are more than one device per link, which suggests either an electrical issue, a race condition or a state machine issue. Add two dev_warn() messages to identify the sequence by which the devices become UNATTACHED. BugLink: https://github.com/thesofproject/linux/issues/3063 BugLink: https://github.com/thesofproject/linux/issues/3325 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 67369e941d0d..354d3f89366f 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1749,8 +1749,11 @@ int sdw_handle_slave_status(struct sdw_bus *bus, continue; if (status[i] == SDW_SLAVE_UNATTACHED && - slave->status != SDW_SLAVE_UNATTACHED) + slave->status != SDW_SLAVE_UNATTACHED) { + dev_warn(&slave->dev, "Slave %d state check1: UNATTACHED, status was %d\n", + i, slave->status); sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); + } } if (status[0] == SDW_SLAVE_ATTACHED) { @@ -1785,6 +1788,9 @@ int sdw_handle_slave_status(struct sdw_bus *bus, if (slave->status == SDW_SLAVE_UNATTACHED) break; + dev_warn(&slave->dev, "Slave %d state check2: UNATTACHED, status was %d\n", + i, slave->status); + sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); break; -- 2.17.1

2 1

[PATCH] soundwire: intel: fix wrong register name in intel_shim_wake
by Bard Liao 10 Feb '22

10 Feb '22

From: Libin Yang <libin.yang(a)intel.com> When clearing the sdw wakests status, we should use SDW_SHIM_WAKESTS. Fixes: 4a17c441c7cb ("soundwire: intel: revisit SHIM programming sequences.") Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 122f7a29d8ca..63101f1ba271 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -448,8 +448,8 @@ static void intel_shim_wake(struct sdw_intel *sdw, bool wake_enable) /* Clear wake status */ wake_sts = intel_readw(shim, SDW_SHIM_WAKESTS); - wake_sts |= (SDW_SHIM_WAKEEN_ENABLE << link_id); - intel_writew(shim, SDW_SHIM_WAKESTS_STATUS, wake_sts); + wake_sts |= (SDW_SHIM_WAKESTS_STATUS << link_id); + intel_writew(shim, SDW_SHIM_WAKESTS, wake_sts); } mutex_unlock(sdw->link_res->shim_lock); } -- 2.17.1

2 1

[PATCH] ASoC: qcom: Actually clear DMA interrupt register for HDMI
by Stephen Boyd 10 Feb '22

10 Feb '22

In commit da0363f7bfd3 ("ASoC: qcom: Fix for DMA interrupt clear reg overwriting") we changed regmap_write() to regmap_update_bits() so that we can avoid overwriting bits that we didn't intend to modify. Unfortunately this change breaks the case where a register is writable but not readable, which is exactly how the HDMI irq clear register is designed (grep around LPASS_HDMITX_APP_IRQCLEAR_REG to see how it's write only). That's because regmap_update_bits() tries to read the register from the hardware and if it isn't readable it looks in the regmap cache to see what was written there last time to compare against what we want to write there. Eventually, we're unable to modify this register at all because the bits that we're trying to set are already set in the cache. This is doubly bad for the irq clear register because you have to write the bit to clear an interrupt. Given the irq is level triggered, we see an interrupt storm upon plugging in an HDMI cable and starting audio playback. The irq storm is so great that performance degrades significantly, leading to CPU soft lockups. Fix it by using regmap_write_bits() so that we really do write the bits in the clear register that we want to. This brings the number of irqs handled by lpass_dma_interrupt_handler() down from ~150k/sec to ~10/sec. Fixes: da0363f7bfd3 ("ASoC: qcom: Fix for DMA interrupt clear reg overwriting") Cc: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- sound/soc/qcom/lpass-platform.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index a59e9d20cb46..4b1773c1fb95 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -524,7 +524,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -EINVAL; } - ret = regmap_update_bits(map, reg_irqclr, val_irqclr, val_irqclr); + ret = regmap_write_bits(map, reg_irqclr, val_irqclr, val_irqclr); if (ret) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", ret); return ret; @@ -665,7 +665,7 @@ static irqreturn_t lpass_dma_interrupt_handler( return -EINVAL; } if (interrupts & LPAIF_IRQ_PER(chan)) { - rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_PER(chan) | val)); + rv = regmap_write_bits(map, reg, mask, (LPAIF_IRQ_PER(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -676,7 +676,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_XRUN(chan)) { - rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_XRUN(chan) | val)); + rv = regmap_write_bits(map, reg, mask, (LPAIF_IRQ_XRUN(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); @@ -688,7 +688,7 @@ static irqreturn_t lpass_dma_interrupt_handler( } if (interrupts & LPAIF_IRQ_ERR(chan)) { - rv = regmap_update_bits(map, reg, mask, (LPAIF_IRQ_ERR(chan) | val)); + rv = regmap_write_bits(map, reg, mask, (LPAIF_IRQ_ERR(chan) | val)); if (rv) { dev_err(soc_runtime->dev, "error writing to irqclear reg: %d\n", rv); base-commit: dfd42facf1e4ada021b939b4e19c935dcdd55566 -- https://chromeos.dev

2 1

[PATCH 1/2] ASoC: bindings: fsl-asoc-card: Add compatible for tlv320aic31xx codec
by Apertis package maintainers 10 Feb '22

10 Feb '22

From: Ariel D'Alessandro <ariel.dalessandro(a)collabora.com> Commit 8c9b9cfb7724 ("ASoC: fsl-asoc-card: Support fsl,imx-audio-tlv320aic31xx codec")' added support for tlv320aic31xx codec to fsl-asoc-card, but missed the related device-tree compatible string documentation. Fix this. Signed-off-by: Ariel D'Alessandro <ariel.dalessandro(a)collabora.com> --- Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index b219626a5403..8b4f4015cfe4 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -40,6 +40,8 @@ The compatible list for this generic sound card currently: "fsl,imx-audio-tlv320aic32x4" + "fsl,imx-audio-tlv320aic31xx" + "fsl,imx-audio-si476x" "fsl,imx-audio-wm8958" -- 2.34.1

3 4

[PATCH] ALSA: ca0106: Rename register macro names
by Takashi Iwai 10 Feb '22

10 Feb '22

ca0106 driver code uses too generic names for its register definitions such as PTR, DATA, IPR, etc, which may eventually conflict with other code. This patch renames (some of) those register definitions with CA0106_ prefix to avoid the conflicts. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ca0106/ca0106.h | 18 ++++----- sound/pci/ca0106/ca0106_main.c | 70 ++++++++++++++++----------------- sound/pci/ca0106/ca0106_mixer.c | 16 ++++---- 3 files changed, 52 insertions(+), 52 deletions(-) diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index f246e6094289..991b1c5d41d5 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -59,15 +59,15 @@ /* PCI function 0 registers, address = <val> + PCIBASE0 */ /************************************************************************************************/ -#define PTR 0x00 /* Indexed register set pointer register */ +#define CA0106_PTR 0x00 /* Indexed register set pointer register */ /* NOTE: The CHANNELNUM and ADDRESS words can */ /* be modified independently of each other. */ /* CNL[1:0], ADDR[27:16] */ -#define DATA 0x04 /* Indexed register set data register */ +#define CA0106_DATA 0x04 /* Indexed register set data register */ /* DATA[31:0] */ -#define IPR 0x08 /* Global interrupt pending register */ +#define CA0106_IPR 0x08 /* Global interrupt pending register */ /* Clear pending interrupts by writing a 1 to */ /* the relevant bits and zero to the other bits */ #define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ @@ -88,7 +88,7 @@ #define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ #define IPR_PCI 0x00000001 /* PCI Bus error */ -#define INTE 0x0c /* Interrupt enable register */ +#define CA0106_INTE 0x0c /* Interrupt enable register */ #define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ #define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ @@ -108,8 +108,8 @@ #define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ #define INTE_PCI 0x00000001 /* PCI Bus error */ -#define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ -#define HCFG 0x14 /* Hardware config register */ +#define CA0106_UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ +#define CA0106_HCFG 0x14 /* Hardware config register */ /* 0x1000 causes AC3 to fails. It adds a dither bit. */ #define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */ @@ -133,7 +133,7 @@ #define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */ /* Should be set to 1 when the EMU10K1 is */ /* completely initialized. */ -#define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */ +#define CA0106_GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */ /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */ /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */ /* SB Live 24bit: @@ -152,9 +152,9 @@ * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF) * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin. */ -#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */ +#define CA0106_AC97DATA 0x1c /* AC97 register set data register (16 bit) */ -#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */ +#define CA0106_AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */ /********************************************************************************************************/ /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 36fb150b72fb..8577f9fa5ea6 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -338,8 +338,8 @@ unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, regptr = (reg << 16) | chn; spin_lock_irqsave(&emu->emu_lock, flags); - outl(regptr, emu->port + PTR); - val = inl(emu->port + DATA); + outl(regptr, emu->port + CA0106_PTR); + val = inl(emu->port + CA0106_DATA); spin_unlock_irqrestore(&emu->emu_lock, flags); return val; } @@ -355,8 +355,8 @@ void snd_ca0106_ptr_write(struct snd_ca0106 *emu, regptr = (reg << 16) | chn; spin_lock_irqsave(&emu->emu_lock, flags); - outl(regptr, emu->port + PTR); - outl(data, emu->port + DATA); + outl(regptr, emu->port + CA0106_PTR); + outl(data, emu->port + CA0106_DATA); spin_unlock_irqrestore(&emu->emu_lock, flags); } @@ -455,8 +455,8 @@ static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb) unsigned int intr_enable; spin_lock_irqsave(&emu->emu_lock, flags); - intr_enable = inl(emu->port + INTE) | intrenb; - outl(intr_enable, emu->port + INTE); + intr_enable = inl(emu->port + CA0106_INTE) | intrenb; + outl(intr_enable, emu->port + CA0106_INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } @@ -466,8 +466,8 @@ static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb unsigned int intr_enable; spin_lock_irqsave(&emu->emu_lock, flags); - intr_enable = inl(emu->port + INTE) & ~intrenb; - outl(intr_enable, emu->port + INTE); + intr_enable = inl(emu->port + CA0106_INTE) & ~intrenb; + outl(intr_enable, emu->port + CA0106_INTE); spin_unlock_irqrestore(&emu->emu_lock, flags); } @@ -786,9 +786,9 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) hcfg_set = 0; break; } - hcfg = inl(emu->port + HCFG) ; + hcfg = inl(emu->port + CA0106_HCFG) ; hcfg = (hcfg & ~hcfg_mask) | hcfg_set; - outl(hcfg, emu->port + HCFG); + outl(hcfg, emu->port + CA0106_HCFG); reg40 = snd_ca0106_ptr_read(emu, 0x40, 0); reg40 = (reg40 & ~reg40_mask) | reg40_set; snd_ca0106_ptr_write(emu, 0x40, 0, reg40); @@ -888,9 +888,9 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) hcfg_set = 0; break; } - hcfg = inl(emu->port + HCFG) ; + hcfg = inl(emu->port + CA0106_HCFG) ; hcfg = (hcfg & ~hcfg_mask) | hcfg_set; - outl(hcfg, emu->port + HCFG); + outl(hcfg, emu->port + CA0106_HCFG); reg71 = snd_ca0106_ptr_read(emu, 0x71, 0); reg71 = (reg71 & ~reg71_mask) | reg71_set; snd_ca0106_ptr_write(emu, 0x71, 0, reg71); @@ -1142,8 +1142,8 @@ static unsigned short snd_ca0106_ac97_read(struct snd_ac97 *ac97, unsigned short val; spin_lock_irqsave(&emu->emu_lock, flags); - outb(reg, emu->port + AC97ADDRESS); - val = inw(emu->port + AC97DATA); + outb(reg, emu->port + CA0106_AC97ADDRESS); + val = inw(emu->port + CA0106_AC97DATA); spin_unlock_irqrestore(&emu->emu_lock, flags); return val; } @@ -1155,8 +1155,8 @@ static void snd_ca0106_ac97_write(struct snd_ac97 *ac97, unsigned long flags; spin_lock_irqsave(&emu->emu_lock, flags); - outb(reg, emu->port + AC97ADDRESS); - outw(val, emu->port + AC97DATA); + outb(reg, emu->port + CA0106_AC97ADDRESS); + outw(val, emu->port + CA0106_AC97DATA); spin_unlock_irqrestore(&emu->emu_lock, flags); } @@ -1200,7 +1200,7 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) unsigned int stat76; struct snd_ca0106_channel *pchannel; - status = inl(chip->port + IPR); + status = inl(chip->port + CA0106_IPR); if (! status) return IRQ_NONE; @@ -1255,7 +1255,7 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) } // acknowledge the interrupt if necessary - outl(status, chip->port+IPR); + outl(status, chip->port + CA0106_IPR); return IRQ_HANDLED; } @@ -1383,7 +1383,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) int ch; unsigned int def_bits; - outl(0, chip->port + INTE); + outl(0, chip->port + CA0106_INTE); /* * Init to 0x02109204 : @@ -1420,8 +1420,8 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000); /* Write 0x8000 to AC97_REC_GAIN to mute it. */ - outb(AC97_REC_GAIN, chip->port + AC97ADDRESS); - outw(0x8000, chip->port + AC97DATA); + outb(AC97_REC_GAIN, chip->port + CA0106_AC97ADDRESS); + outw(0x8000, chip->port + CA0106_AC97DATA); #if 0 /* FIXME: what are these? */ snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006); snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006); @@ -1495,30 +1495,30 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) /* FIXME: Still need to find out what the other GPIO bits do. * E.g. For digital spdif out. */ - outl(0x0, chip->port+GPIO); - /* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */ - outl(0x005f5301, chip->port+GPIO); /* Analog */ + outl(0x0, chip->port + CA0106_GPIO); + /* outl(0x00f0e000, chip->port + CA0106_GPIO); */ /* Analog */ + outl(0x005f5301, chip->port + CA0106_GPIO); /* Analog */ } else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */ /* FIXME: Still need to find out what the other GPIO bits do. * E.g. For digital spdif out. */ - outl(0x0, chip->port+GPIO); - /* outl(0x00f0e000, chip->port+GPIO); */ /* Analog */ - outl(0x005f5301, chip->port+GPIO); /* Analog */ + outl(0x0, chip->port + CA0106_GPIO); + /* outl(0x00f0e000, chip->port + CA0106_GPIO); */ /* Analog */ + outl(0x005f5301, chip->port + CA0106_GPIO); /* Analog */ } else { - outl(0x0, chip->port+GPIO); - outl(0x005f03a3, chip->port+GPIO); /* Analog */ - /* outl(0x005f02a2, chip->port+GPIO); */ /* SPDIF */ + outl(0x0, chip->port + CA0106_GPIO); + outl(0x005f03a3, chip->port + CA0106_GPIO); /* Analog */ + /* outl(0x005f02a2, chip->port + CA0106_GPIO); */ /* SPDIF */ } snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */ /* outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); */ /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ - /* outl(0x00001409, chip->port+HCFG); */ - /* outl(0x00000009, chip->port+HCFG); */ + /* outl(0x00001409, chip->port + CA0106_HCFG); */ + /* outl(0x00000009, chip->port + CA0106_HCFG); */ /* AC97 2.0, Enable outputs. */ - outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); + outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port + CA0106_HCFG); if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ @@ -1560,12 +1560,12 @@ static void ca0106_stop_chip(struct snd_ca0106 *chip) { /* disable interrupts */ snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0); - outl(0, chip->port + INTE); + outl(0, chip->port + CA0106_INTE); snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0); udelay(1000); /* disable audio */ /* outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); */ - outl(0, chip->port + HCFG); + outl(0, chip->port + CA0106_HCFG); /* FIXME: We need to stop and DMA transfers here. * But as I am not sure how yet, we cannot from the dma pages. * So we can fix: snd-malloc: Memory leak? pages not freed = 8 diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index c852c6a75b91..05f56015ddd8 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -70,8 +70,8 @@ static void ca0106_spdif_enable(struct snd_ca0106 *emu) snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000); val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000; snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val); - val = inl(emu->port + GPIO) & ~0x101; - outl(val, emu->port + GPIO); + val = inl(emu->port + CA0106_GPIO) & ~0x101; + outl(val, emu->port + CA0106_GPIO); } else { /* Analog */ @@ -79,8 +79,8 @@ static void ca0106_spdif_enable(struct snd_ca0106 *emu) snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000); val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000; snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val); - val = inl(emu->port + GPIO) | 0x101; - outl(val, emu->port + GPIO); + val = inl(emu->port + CA0106_GPIO) | 0x101; + outl(val, emu->port + CA0106_GPIO); } } @@ -119,14 +119,14 @@ static void ca0106_set_capture_mic_line_in(struct snd_ca0106 *emu) if (emu->capture_mic_line_in) { /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */ - tmp = inl(emu->port+GPIO) & ~0x400; + tmp = inl(emu->port + CA0106_GPIO) & ~0x400; tmp = tmp | 0x400; - outl(tmp, emu->port+GPIO); + outl(tmp, emu->port + CA0106_GPIO); /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */ } else { /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */ - tmp = inl(emu->port+GPIO) & ~0x400; - outl(tmp, emu->port+GPIO); + tmp = inl(emu->port + CA0106_GPIO) & ~0x400; + outl(tmp, emu->port + CA0106_GPIO); /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */ } } -- 2.31.1

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[PATCH 0/2] ALSA: memalloc: Fixes for sync check and SG helper
by Takashi Iwai 10 Feb '22

10 Feb '22

The recent changes and additions in ALSA core memalloc helpers had incorrect usages of dma_need_sync() and the order of SG page invalidation. This patch corrects those. Takashi === Takashi Iwai (2): ALSA: memalloc: Fix dma_need_sync() checks ALSA: memalloc: invalidate SG pages before sync sound/core/memalloc.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) -- 2.31.1

1 2

[bug report] ASoC: SOF: compr: Add compress ops implementation
by Dan Carpenter 10 Feb '22

10 Feb '22

Hello Daniel Baluta, This is a semi-automatic email about new static checker warnings. The patch 6324cf901e14: "ASoC: SOF: compr: Add compress ops implementation" from Jan 20, 2022, leads to the following Smatch complaint: sound/soc/sof/compress.c:54 snd_sof_compr_fragment_elapsed() warn: variable dereferenced before check 'cstream' (see line 48) sound/soc/sof/compress.c 47 { 48 struct snd_soc_pcm_runtime *rtd = cstream->private_data; 49 struct snd_compr_runtime *crtd = cstream->runtime; ^^^^^^^^^^^^^^^^ The patch adds a new dereference 50 struct snd_soc_component *component; 51 struct snd_compr_tstamp *tstamp; 52 struct snd_sof_pcm *spcm; 53 54 if (!cstream) ^^^^^^^^ The old code assumed "cstream" could be NULL 55 return; 56 regards, dan carpenter

1 0

[PATCH AUTOSEL 5.16 01/42] ASoC: mediatek: fix unmet dependency on GPIOLIB for SND_SOC_DMIC
by Sasha Levin 09 Feb '22

09 Feb '22

From: Julian Braha <julianbraha(a)gmail.com> [ Upstream commit 579b2c8f72d974f27d85bbd53846f34675ee3b01 ] When SND_SOC_MT8195_MT6359_RT1011_RT5682 is selected, and GPIOLIB is not selected, Kbuild gives the following warning: WARNING: unmet direct dependencies detected for SND_SOC_DMIC Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && GPIOLIB [=n] Selected by [y]: - SND_SOC_MT8195_MT6359_RT1011_RT5682 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && I2C [=y] && SND_SOC_MT8195 [=y] && MTK_PMIC_WRAP [=y] This is because SND_SOC_MT8195_MT6359_RT1011_RT5682 selects SND_SOC_DMIC without selecting or depending on GPIOLIB, depsite SND_SOC_DMIC depending on GPIOLIB. This unmet dependency bug was detected by Kismet, a static analysis tool for Kconfig. Please advise if this is not the appropriate solution. Signed-off-by: Julian Braha <julianbraha(a)gmail.com> Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20220117050324.68371-1-julianbraha@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 3b1ddea26a9ef..76f191ec7bf84 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_MT8195_MT6359_RT1019_RT5682 config SND_SOC_MT8195_MT6359_RT1011_RT5682 tristate "ASoC Audio driver for MT8195 with MT6359 RT1011 RT5682 codec" - depends on I2C + depends on I2C && GPIOLIB depends on SND_SOC_MT8195 && MTK_PMIC_WRAP select SND_SOC_MT6359 select SND_SOC_RT1011 -- 2.34.1

1 0

[PATCH v6] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 09 Feb '22

09 Feb '22

Convert the bindings for McASP controllers for TI SoCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains', '#sound-dai-cells' and 'port' which were missing from the txt file. Removes properties 'sram-size-playback' and 'sram-size-capture' since they are not used. Adds 'dmas' and 'dma-names' in the example which were missing from the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Changes 'op-modes', 'serial-dir' to optional properties as they are not needed if the McASP is used only as GPIO. Changes 'tdm-slots' to required property only for I2S operation mode. Adds the yaml file in the 'MAINTAINERS' under the heading 'TEXAS INSTRUMENTS ASoC DRIVERS' Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> --- Changelog: v6: - adds 'reviewed-by' and 'acked-by' from v5 patch v5: - changes the commit message - changes description for 'reg', 'op-mode', 'tx-num-evt', 'rx-num-evt' and 'interrupts' - adds minimum and maximum for 'tdm-slots' - removes default from 'serial-dir' - removes maxItems from 'tdm-slots', 'tx-num-evt', 'rx-num-evt' and 'auxclk-fs-ratio' as they are of type uint32 - adds 'tdm-slots' as conditionally required property for I2S mode of operation v4: - changes the commit message - adds type and description to relevant properties - changes maxItems for 'serial-dir' - removes properties 'sram-size-playback' and 'sram-size-capture' as they are not used - removes 'function-gpios' - removes 'num-serializer' - marks 'tdm-slots', 'op-mode' and 'serial-dir' as optional properties - adds the yaml file in MAINTAINERS v3: - remove maxItems from 'clock-names' v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'ports' and 'num-serializer' as node properties .../bindings/sound/davinci-mcasp-audio.txt | 86 -------- .../bindings/sound/davinci-mcasp-audio.yaml | 201 ++++++++++++++++++ MAINTAINERS | 1 + 3 files changed, 202 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..f46c66bc6b2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,201 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: 0 - I2S or 1 - DIT operation mode + enum: + - 0 + - 1 + + tdm-slots: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number of channels over one serializer + the property is ignored in DIT mode + minimum: 2 + maximum: 32 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + All AXR pins should be present in the array even if inactive + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 25 + items: + minimum: 0 + maximum: 2 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures WFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + rx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures RFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + dismod: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + specify the drive on TX pin during inactive time slots + 0 - 3-state, 2 - logic low, 3 - logic high + enum: + - 0 + - 2 + - 3 + default: 2 + + interrupts: + anyOf: + - minItems: 1 + items: + - description: TX interrupt + - description: RX interrupt + - items: + - description: common/combined interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + $ref: /schemas/types.yaml#/definitions/string + description: parent clock name for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + $ref: /schemas/types.yaml#/definitions/uint32 + description: ratio of AUCLK and FS rate if applicable + + gpio-controller: true + + "#gpio-cells": + const: 2 + + clocks: + minItems: 1 + items: + - description: functional clock + - description: module specific optional ahclkx clock + - description: module specific optional ahclkr clock + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + description: phandle to the corresponding power-domain + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + port: + description: connection for when McASP is used via graph card + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + +allOf: + - if: + properties: + opmode: + enum: + - 0 + + then: + required: + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 35ca7a4a65d8..d8f99c617ad5 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -19157,6 +19157,7 @@ TEXAS INSTRUMENTS ASoC DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ TEXAS INSTRUMENTS' DAC7612 DAC DRIVER -- 2.17.1

2 1

[PATCH] ASoC: SOF: Intel: hdac_ext_stream: consistent prefixes for variables/members
by Peter Ujfalusi 09 Feb '22

09 Feb '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The existing code maximizes confusion by using 'stream' and 'hstream' variables of different types, e.g: struct hdac_stream *stream; struct hdac_ext_stream *stream; struct hdac_stream *hstream; struct hdac_ext_stream *hstream; This confusion is partly inherited from legacy code but SOF contributors added their own creative spin, e.g. struct hdac_ext_stream *link_dev; struct hdac_ext_stream *dsp_stream; struct hdac_ext_stream hda_stream; and my personal favorite: stream = &hda_stream->hda_stream; This patch suggests a consistent naming across all Intel code related to HDAudio stream management. The convention is - by hierarchical order: struct sof_intel_hda_stream *hda_stream; struct hdac_ext_stream *hext_stream; struct hdac_stream *hstream; No functionality change - just renaming of variables/members. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 92 +++++++++++------------ sound/soc/sof/intel/hda-dsp.c | 14 ++-- sound/soc/sof/intel/hda-ipc.c | 10 +-- sound/soc/sof/intel/hda-loader.c | 50 ++++++------- sound/soc/sof/intel/hda-pcm.c | 12 +-- sound/soc/sof/intel/hda-probes.c | 34 ++++----- sound/soc/sof/intel/hda-stream.c | 122 +++++++++++++++---------------- sound/soc/sof/intel/hda-trace.c | 6 +- sound/soc/sof/intel/hda.h | 17 +++-- 9 files changed, 179 insertions(+), 178 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 28a54145c150..af0c85e4e299 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -62,7 +62,7 @@ static struct hdac_ext_stream * const struct sof_intel_dsp_desc *chip; struct snd_sof_dev *sdev; struct hdac_ext_stream *res = NULL; - struct hdac_stream *stream = NULL; + struct hdac_stream *hstream = NULL; int stream_dir = substream->stream; @@ -72,39 +72,39 @@ static struct hdac_ext_stream * } spin_lock_irq(&bus->reg_lock); - list_for_each_entry(stream, &bus->stream_list, list) { - struct hdac_ext_stream *hstream = - stream_to_hdac_ext_stream(stream); - if (stream->direction != substream->stream) + list_for_each_entry(hstream, &bus->stream_list, list) { + struct hdac_ext_stream *hext_stream = + stream_to_hdac_ext_stream(hstream); + if (hstream->direction != substream->stream) continue; - hda_stream = hstream_to_sof_hda_stream(hstream); + hda_stream = hstream_to_sof_hda_stream(hext_stream); sdev = hda_stream->sdev; chip = get_chip_info(sdev->pdata); /* check if link is available */ - if (!hstream->link_locked) { + if (!hext_stream->link_locked) { /* * choose the first available link for platforms that do not have the * PROCEN_FMT_QUIRK set. */ if (!(chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK)) { - res = hstream; + res = hext_stream; break; } - if (stream->opened) { + if (hstream->opened) { /* * check if the stream tag matches the stream * tag of one of the connected FEs */ if (hda_check_fes(rtd, stream_dir, - stream->stream_tag)) { - res = hstream; + hstream->stream_tag)) { + res = hext_stream; break; } } else { - res = hstream; + res = hext_stream; /* * This must be a hostless stream. @@ -132,17 +132,17 @@ static struct hdac_ext_stream * return res; } -static int hda_link_dma_params(struct hdac_ext_stream *stream, +static int hda_link_dma_params(struct hdac_ext_stream *hext_stream, struct hda_pipe_params *params) { - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; unsigned char stream_tag = hstream->stream_tag; struct hdac_bus *bus = hstream->bus; struct hdac_ext_link *link; unsigned int format_val; - snd_hdac_ext_stream_decouple(bus, stream, true); - snd_hdac_ext_link_stream_reset(stream); + snd_hdac_ext_stream_decouple(bus, hext_stream, true); + snd_hdac_ext_link_stream_reset(hext_stream); format_val = snd_hdac_calc_stream_format(params->s_freq, params->ch, params->format, @@ -151,9 +151,9 @@ static int hda_link_dma_params(struct hdac_ext_stream *stream, dev_dbg(bus->dev, "format_val=%d, rate=%d, ch=%d, format=%d\n", format_val, params->s_freq, params->ch, params->format); - snd_hdac_ext_link_stream_setup(stream, format_val); + snd_hdac_ext_link_stream_setup(hext_stream, format_val); - if (stream->hstream.direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (hext_stream->hstream.direction == SNDRV_PCM_STREAM_PLAYBACK) { list_for_each_entry(link, &bus->hlink_list, list) { if (link->index == params->link_index) snd_hdac_ext_link_set_stream_id(link, @@ -161,7 +161,7 @@ static int hda_link_dma_params(struct hdac_ext_stream *stream, } } - stream->link_prepared = 1; + hext_stream->link_prepared = 1; return 0; } @@ -218,7 +218,7 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, { struct hdac_stream *hstream = substream->runtime->private_data; struct hdac_bus *bus = hstream->bus; - struct hdac_ext_stream *link_dev; + struct hdac_ext_stream *hext_stream; struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); struct sof_intel_hda_stream *hda_stream; @@ -229,18 +229,18 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, int ret; /* get stored dma data if resuming from system suspend */ - link_dev = snd_soc_dai_get_dma_data(dai, substream); - if (!link_dev) { - link_dev = hda_link_stream_assign(bus, substream); - if (!link_dev) + hext_stream = snd_soc_dai_get_dma_data(dai, substream); + if (!hext_stream) { + hext_stream = hda_link_stream_assign(bus, substream); + if (!hext_stream) return -EBUSY; - snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); + snd_soc_dai_set_dma_data(dai, substream, (void *)hext_stream); } - stream_tag = hdac_stream(link_dev)->stream_tag; + stream_tag = hdac_stream(hext_stream)->stream_tag; - hda_stream = hstream_to_sof_hda_stream(link_dev); + hda_stream = hstream_to_sof_hda_stream(hext_stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) w = dai->playback_widget; @@ -257,7 +257,7 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, return -EINVAL; /* set the hdac_stream in the codec dai */ - snd_soc_dai_set_stream(codec_dai, hdac_stream(link_dev), substream->stream); + snd_soc_dai_set_stream(codec_dai, hdac_stream(hext_stream), substream->stream); p_params.s_fmt = snd_pcm_format_width(params_format(params)); p_params.ch = params_channels(params); @@ -271,20 +271,20 @@ static int hda_link_hw_params(struct snd_pcm_substream *substream, else p_params.link_bps = codec_dai->driver->capture.sig_bits; - return hda_link_dma_params(link_dev, &p_params); + return hda_link_dma_params(hext_stream, &p_params); } static int hda_link_pcm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct hdac_ext_stream *link_dev = + struct hdac_ext_stream *hext_stream = snd_soc_dai_get_dma_data(dai, substream); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(dai->component); struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int stream = substream->stream; - if (link_dev->link_prepared) + if (hext_stream->link_prepared) return 0; dev_dbg(sdev->dev, "hda: prepare stream dir %d\n", substream->stream); @@ -326,7 +326,7 @@ static int hda_link_dai_config_pause_push_ipc(struct snd_soc_dapm_widget *w) static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct hdac_ext_stream *link_dev = + struct hdac_ext_stream *hext_stream = snd_soc_dai_get_dma_data(dai, substream); struct sof_intel_hda_stream *hda_stream; struct snd_soc_pcm_runtime *rtd; @@ -345,7 +345,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, if (!link) return -EINVAL; - hda_stream = hstream_to_sof_hda_stream(link_dev); + hda_stream = hstream_to_sof_hda_stream(hext_stream); dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd); @@ -354,11 +354,11 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - snd_hdac_ext_link_stream_start(link_dev); + snd_hdac_ext_link_stream_start(hext_stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: - snd_hdac_ext_link_stream_clear(link_dev); + snd_hdac_ext_link_stream_clear(hext_stream); /* * free DAI widget during stop/suspend to keep widget use_count's balanced. @@ -368,14 +368,14 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, return ret; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - stream_tag = hdac_stream(link_dev)->stream_tag; + stream_tag = hdac_stream(hext_stream)->stream_tag; snd_hdac_ext_link_clear_stream_id(link, stream_tag); } - link_dev->link_prepared = 0; + hext_stream->link_prepared = 0; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_hdac_ext_link_stream_clear(link_dev); + snd_hdac_ext_link_stream_clear(hext_stream); ret = hda_link_dai_config_pause_push_ipc(w); if (ret < 0) @@ -396,22 +396,22 @@ static int hda_link_hw_free(struct snd_pcm_substream *substream, struct hdac_ext_link *link; struct hdac_stream *hstream; struct snd_soc_pcm_runtime *rtd; - struct hdac_ext_stream *link_dev; + struct hdac_ext_stream *hext_stream; struct snd_soc_dapm_widget *w; int ret; hstream = substream->runtime->private_data; bus = hstream->bus; rtd = asoc_substream_to_rtd(substream); - link_dev = snd_soc_dai_get_dma_data(dai, substream); + hext_stream = snd_soc_dai_get_dma_data(dai, substream); - if (!link_dev) { + if (!hext_stream) { dev_dbg(dai->dev, - "%s: link_dev is not assigned\n", __func__); + "%s: hext_stream is not assigned\n", __func__); return -EINVAL; } - hda_stream = hstream_to_sof_hda_stream(link_dev); + hda_stream = hstream_to_sof_hda_stream(hext_stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) w = dai->playback_widget; @@ -428,13 +428,13 @@ static int hda_link_hw_free(struct snd_pcm_substream *substream, return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - stream_tag = hdac_stream(link_dev)->stream_tag; + stream_tag = hdac_stream(hext_stream)->stream_tag; snd_hdac_ext_link_clear_stream_id(link, stream_tag); } snd_soc_dai_set_dma_data(dai, substream, NULL); - snd_hdac_ext_stream_release(link_dev, HDAC_EXT_STREAM_TYPE_LINK); - link_dev->link_prepared = 0; + snd_hdac_ext_stream_release(hext_stream, HDAC_EXT_STREAM_TYPE_LINK); + hext_stream->link_prepared = 0; /* free the host DMA channel reserved by hostless streams */ hda_stream->host_reserved = 0; diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 916a257ea96b..0fe522549c91 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -904,7 +904,7 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_bus *bus = sof_to_bus(sdev); struct snd_soc_pcm_runtime *rtd; - struct hdac_ext_stream *stream; + struct hdac_ext_stream *hext_stream; struct hdac_ext_link *link; struct hdac_stream *s; const char *name; @@ -912,7 +912,7 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) /* set internal flag for BE */ list_for_each_entry(s, &bus->stream_list, list) { - stream = stream_to_hdac_ext_stream(s); + hext_stream = stream_to_hdac_ext_stream(s); /* * clear stream. This should already be taken care for running @@ -920,20 +920,20 @@ int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) * streams do not get suspended, so this needs to be done * explicitly during suspend. */ - if (stream->link_substream) { - rtd = asoc_substream_to_rtd(stream->link_substream); + if (hext_stream->link_substream) { + rtd = asoc_substream_to_rtd(hext_stream->link_substream); name = asoc_rtd_to_codec(rtd, 0)->component->name; link = snd_hdac_ext_bus_get_link(bus, name); if (!link) return -EINVAL; - stream->link_prepared = 0; + hext_stream->link_prepared = 0; - if (hdac_stream(stream)->direction == + if (hdac_stream(hext_stream)->direction == SNDRV_PCM_STREAM_CAPTURE) continue; - stream_tag = hdac_stream(stream)->stream_tag; + stream_tag = hdac_stream(hext_stream)->stream_tag; snd_hdac_ext_link_clear_stream_id(link, stream_tag); } } diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index f0cf8019d72d..a8c452144168 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -255,13 +255,13 @@ int hda_ipc_msg_data(struct snd_sof_dev *sdev, hda_stream = container_of(hstream, struct sof_intel_hda_stream, - hda_stream.hstream); + hext_stream.hstream); /* The stream might already be closed */ if (!hstream) return -ESTRPIPE; - sof_mailbox_read(sdev, hda_stream->stream.posn_offset, p, sz); + sof_mailbox_read(sdev, hda_stream->sof_intel_stream.posn_offset, p, sz); } return 0; @@ -277,17 +277,17 @@ int hda_ipc_pcm_params(struct snd_sof_dev *sdev, size_t posn_offset = reply->posn_offset; hda_stream = container_of(hstream, struct sof_intel_hda_stream, - hda_stream.hstream); + hext_stream.hstream); /* check for unaligned offset or overflow */ if (posn_offset > sdev->stream_box.size || posn_offset % sizeof(struct sof_ipc_stream_posn) != 0) return -EINVAL; - hda_stream->stream.posn_offset = sdev->stream_box.offset + posn_offset; + hda_stream->sof_intel_stream.posn_offset = sdev->stream_box.offset + posn_offset; dev_dbg(sdev->dev, "pcm: stream dir %d, posn mailbox offset is %zu", - substream->stream, hda_stream->stream.posn_offset); + substream->stream, hda_stream->sof_intel_stream.posn_offset); return 0; } diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 6af3325b7e40..3c0d7cbdf174 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -47,18 +47,18 @@ static struct hdac_ext_stream *cl_stream_prepare(struct snd_sof_dev *sdev, unsig unsigned int size, struct snd_dma_buffer *dmab, int direction) { - struct hdac_ext_stream *dsp_stream; + struct hdac_ext_stream *hext_stream; struct hdac_stream *hstream; struct pci_dev *pci = to_pci_dev(sdev->dev); int ret; - dsp_stream = hda_dsp_stream_get(sdev, direction, 0); + hext_stream = hda_dsp_stream_get(sdev, direction, 0); - if (!dsp_stream) { + if (!hext_stream) { dev_err(sdev->dev, "error: no stream available\n"); return ERR_PTR(-ENODEV); } - hstream = &dsp_stream->hstream; + hstream = &hext_stream->hstream; hstream->substream = NULL; /* allocate DMA buffer */ @@ -73,21 +73,21 @@ static struct hdac_ext_stream *cl_stream_prepare(struct snd_sof_dev *sdev, unsig hstream->bufsize = size; if (direction == SNDRV_PCM_STREAM_CAPTURE) { - ret = hda_dsp_iccmax_stream_hw_params(sdev, dsp_stream, dmab, NULL); + ret = hda_dsp_iccmax_stream_hw_params(sdev, hext_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: iccmax stream prepare failed: %d\n", ret); goto error; } } else { - ret = hda_dsp_stream_hw_params(sdev, dsp_stream, dmab, NULL); + ret = hda_dsp_stream_hw_params(sdev, hext_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: hdac prepare failed: %d\n", ret); goto error; } - hda_dsp_stream_spib_config(sdev, dsp_stream, HDA_DSP_SPIB_ENABLE, size); + hda_dsp_stream_spib_config(sdev, hext_stream, HDA_DSP_SPIB_ENABLE, size); } - return dsp_stream; + return hext_stream; error: hda_dsp_stream_put(sdev, direction, hstream->stream_tag); @@ -209,9 +209,9 @@ static int cl_dsp_init(struct snd_sof_dev *sdev, int stream_tag) } static int cl_trigger(struct snd_sof_dev *sdev, - struct hdac_ext_stream *stream, int cmd) + struct hdac_ext_stream *hext_stream, int cmd) { - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); /* code loader is special case that reuses stream ops */ @@ -231,19 +231,19 @@ static int cl_trigger(struct snd_sof_dev *sdev, hstream->running = true; return 0; default: - return hda_dsp_stream_trigger(sdev, stream, cmd); + return hda_dsp_stream_trigger(sdev, hext_stream, cmd); } } static int cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, - struct hdac_ext_stream *stream) + struct hdac_ext_stream *hext_stream) { - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); int ret = 0; if (hstream->direction == SNDRV_PCM_STREAM_PLAYBACK) - ret = hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_DISABLE, 0); + ret = hda_dsp_stream_spib_config(sdev, hext_stream, HDA_DSP_SPIB_DISABLE, 0); else snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, sd_offset, SOF_HDA_SD_CTL_DMA_START, 0); @@ -267,12 +267,12 @@ static int cl_cleanup(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, return ret; } -static int cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream) +static int cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream) { unsigned int reg; int ret, status; - ret = cl_trigger(sdev, stream, SNDRV_PCM_TRIGGER_START); + ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_START); if (ret < 0) { dev_err(sdev->dev, "error: DMA trigger start failed\n"); return ret; @@ -296,7 +296,7 @@ static int cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream) __func__); } - ret = cl_trigger(sdev, stream, SNDRV_PCM_TRIGGER_STOP); + ret = cl_trigger(sdev, hext_stream, SNDRV_PCM_TRIGGER_STOP); if (ret < 0) { dev_err(sdev->dev, "error: DMA trigger stop failed\n"); if (!status) @@ -392,7 +392,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) struct snd_sof_pdata *plat_data = sdev->pdata; const struct sof_dev_desc *desc = plat_data->desc; const struct sof_intel_dsp_desc *chip_info; - struct hdac_ext_stream *stream; + struct hdac_ext_stream *hext_stream; struct firmware stripped_firmware; int ret, ret1, i; @@ -417,11 +417,11 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) init_waitqueue_head(&sdev->boot_wait); /* prepare DMA for code loader stream */ - stream = cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, stripped_firmware.size, - &sdev->dmab, SNDRV_PCM_STREAM_PLAYBACK); - if (IS_ERR(stream)) { + hext_stream = cl_stream_prepare(sdev, HDA_CL_STREAM_FORMAT, stripped_firmware.size, + &sdev->dmab, SNDRV_PCM_STREAM_PLAYBACK); + if (IS_ERR(hext_stream)) { dev_err(sdev->dev, "error: dma prepare for fw loading failed\n"); - return PTR_ERR(stream); + return PTR_ERR(hext_stream); } memcpy(sdev->dmab.area, stripped_firmware.data, @@ -433,7 +433,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) "Attempting iteration %d of Core En/ROM load...\n", i); hda->boot_iteration = i + 1; - ret = cl_dsp_init(sdev, stream->hstream.stream_tag); + ret = cl_dsp_init(sdev, hext_stream->hstream.stream_tag); /* don't retry anymore if successful */ if (!ret) @@ -472,7 +472,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) * Continue with code loading and firmware boot */ hda->boot_iteration = HDA_FW_BOOT_ATTEMPTS; - ret = cl_copy_fw(sdev, stream); + ret = cl_copy_fw(sdev, hext_stream); if (!ret) dev_dbg(sdev->dev, "Firmware download successful, booting...\n"); else @@ -485,7 +485,7 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) * This should be done even if firmware loading fails. * If the cleanup also fails, we return the initial error */ - ret1 = cl_cleanup(sdev, &sdev->dmab, stream); + ret1 = cl_cleanup(sdev, &sdev->dmab, hext_stream); if (ret1 < 0) { dev_err(sdev->dev, "error: Code loader DSP cleanup failed\n"); diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index d78aa5d8552d..eec83ca557a1 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -96,7 +96,7 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, struct sof_ipc_stream_params *ipc_params) { struct hdac_stream *hstream = substream->runtime->private_data; - struct hdac_ext_stream *stream = stream_to_hdac_ext_stream(hstream); + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct snd_dma_buffer *dmab; struct sof_ipc_fw_version *v = &sdev->fw_ready.version; @@ -118,7 +118,7 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, (params->info & SNDRV_PCM_INFO_NO_PERIOD_WAKEUP) && (params->flags & SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP); - ret = hda_dsp_stream_hw_params(sdev, stream, dmab, params); + ret = hda_dsp_stream_hw_params(sdev, hext_stream, dmab, params); if (ret < 0) { dev_err(sdev->dev, "error: hdac prepare failed: %d\n", ret); return ret; @@ -126,9 +126,9 @@ int hda_dsp_pcm_hw_params(struct snd_sof_dev *sdev, /* enable SPIB when rewinds are disabled */ if (hda_disable_rewinds) - hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_ENABLE, 0); + hda_dsp_stream_spib_config(sdev, hext_stream, HDA_DSP_SPIB_ENABLE, 0); else - hda_dsp_stream_spib_config(sdev, stream, HDA_DSP_SPIB_DISABLE, 0); + hda_dsp_stream_spib_config(sdev, hext_stream, HDA_DSP_SPIB_DISABLE, 0); /* update no_stream_position flag for ipc params */ if (hda && hda->no_ipc_position) { @@ -174,9 +174,9 @@ int hda_dsp_pcm_trigger(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, int cmd) { struct hdac_stream *hstream = substream->runtime->private_data; - struct hdac_ext_stream *stream = stream_to_hdac_ext_stream(hstream); + struct hdac_ext_stream *hext_stream = stream_to_hdac_ext_stream(hstream); - return hda_dsp_stream_trigger(sdev, stream, cmd); + return hda_dsp_stream_trigger(sdev, hext_stream, cmd); } snd_pcm_uframes_t hda_dsp_pcm_pointer(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/intel/hda-probes.c b/sound/soc/sof/intel/hda-probes.c index fe2f3f7d236b..35e548da3609 100644 --- a/sound/soc/sof/intel/hda-probes.c +++ b/sound/soc/sof/intel/hda-probes.c @@ -23,34 +23,34 @@ int hda_probe_compr_assign(struct snd_sof_dev *sdev, struct snd_compr_stream *cstream, struct snd_soc_dai *dai) { - struct hdac_ext_stream *stream; + struct hdac_ext_stream *hext_stream; - stream = hda_dsp_stream_get(sdev, cstream->direction, 0); - if (!stream) + hext_stream = hda_dsp_stream_get(sdev, cstream->direction, 0); + if (!hext_stream) return -EBUSY; - hdac_stream(stream)->curr_pos = 0; - hdac_stream(stream)->cstream = cstream; - cstream->runtime->private_data = stream; + hdac_stream(hext_stream)->curr_pos = 0; + hdac_stream(hext_stream)->cstream = cstream; + cstream->runtime->private_data = hext_stream; - return hdac_stream(stream)->stream_tag; + return hdac_stream(hext_stream)->stream_tag; } int hda_probe_compr_free(struct snd_sof_dev *sdev, struct snd_compr_stream *cstream, struct snd_soc_dai *dai) { - struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + struct hdac_ext_stream *hext_stream = hda_compr_get_stream(cstream); int ret; ret = hda_dsp_stream_put(sdev, cstream->direction, - hdac_stream(stream)->stream_tag); + hdac_stream(hext_stream)->stream_tag); if (ret < 0) { dev_dbg(sdev->dev, "stream put failed: %d\n", ret); return ret; } - hdac_stream(stream)->cstream = NULL; + hdac_stream(hext_stream)->cstream = NULL; cstream->runtime->private_data = NULL; return 0; @@ -61,8 +61,8 @@ int hda_probe_compr_set_params(struct snd_sof_dev *sdev, struct snd_compr_params *params, struct snd_soc_dai *dai) { - struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); - struct hdac_stream *hstream = hdac_stream(stream); + struct hdac_ext_stream *hext_stream = hda_compr_get_stream(cstream); + struct hdac_stream *hstream = hdac_stream(hext_stream); struct snd_dma_buffer *dmab; u32 bits, rate; int bps, ret; @@ -80,7 +80,7 @@ int hda_probe_compr_set_params(struct snd_sof_dev *sdev, hstream->period_bytes = cstream->runtime->fragment_size; hstream->no_period_wakeup = 0; - ret = hda_dsp_stream_hw_params(sdev, stream, dmab, NULL); + ret = hda_dsp_stream_hw_params(sdev, hext_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: hdac prepare failed: %d\n", ret); return ret; @@ -93,9 +93,9 @@ int hda_probe_compr_trigger(struct snd_sof_dev *sdev, struct snd_compr_stream *cstream, int cmd, struct snd_soc_dai *dai) { - struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + struct hdac_ext_stream *hext_stream = hda_compr_get_stream(cstream); - return hda_dsp_stream_trigger(sdev, stream, cmd); + return hda_dsp_stream_trigger(sdev, hext_stream, cmd); } int hda_probe_compr_pointer(struct snd_sof_dev *sdev, @@ -103,11 +103,11 @@ int hda_probe_compr_pointer(struct snd_sof_dev *sdev, struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) { - struct hdac_ext_stream *stream = hda_compr_get_stream(cstream); + struct hdac_ext_stream *hext_stream = hda_compr_get_stream(cstream); struct snd_soc_pcm_stream *pstream; pstream = &dai->driver->capture; - tstamp->copied_total = hdac_stream(stream)->curr_pos; + tstamp->copied_total = hdac_stream(hext_stream)->curr_pos; tstamp->sampling_rate = snd_pcm_rate_bit_to_rate(pstream->rates); return 0; diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index ba60807fbd8f..daeb64c495e4 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -57,7 +57,7 @@ static char *hda_hstream_dbg_get_stream_info_str(struct hdac_stream *hstream) */ static int hda_setup_bdle(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, - struct hdac_stream *stream, + struct hdac_stream *hstream, struct sof_intel_dsp_bdl **bdlp, int offset, int size, int ioc) { @@ -68,7 +68,7 @@ static int hda_setup_bdle(struct snd_sof_dev *sdev, dma_addr_t addr; int chunk; - if (stream->frags >= HDA_DSP_MAX_BDL_ENTRIES) { + if (hstream->frags >= HDA_DSP_MAX_BDL_ENTRIES) { dev_err(sdev->dev, "error: stream frags exceeded\n"); return -EINVAL; } @@ -91,11 +91,11 @@ static int hda_setup_bdle(struct snd_sof_dev *sdev, size -= chunk; bdl->ioc = (size || !ioc) ? 0 : cpu_to_le32(0x01); bdl++; - stream->frags++; + hstream->frags++; offset += chunk; dev_vdbg(sdev->dev, "bdl, frags:%d, chunk size:0x%x;\n", - stream->frags, chunk); + hstream->frags, chunk); } *bdlp = bdl; @@ -108,47 +108,47 @@ static int hda_setup_bdle(struct snd_sof_dev *sdev, */ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, - struct hdac_stream *stream) + struct hdac_stream *hstream) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; struct sof_intel_dsp_bdl *bdl; int i, offset, period_bytes, periods; int remain, ioc; - period_bytes = stream->period_bytes; + period_bytes = hstream->period_bytes; dev_dbg(sdev->dev, "%s: period_bytes:0x%x\n", __func__, period_bytes); if (!period_bytes) - period_bytes = stream->bufsize; + period_bytes = hstream->bufsize; - periods = stream->bufsize / period_bytes; + periods = hstream->bufsize / period_bytes; dev_dbg(sdev->dev, "%s: periods:%d\n", __func__, periods); - remain = stream->bufsize % period_bytes; + remain = hstream->bufsize % period_bytes; if (remain) periods++; /* program the initial BDL entries */ - bdl = (struct sof_intel_dsp_bdl *)stream->bdl.area; + bdl = (struct sof_intel_dsp_bdl *)hstream->bdl.area; offset = 0; - stream->frags = 0; + hstream->frags = 0; /* * set IOC if don't use position IPC * and period_wakeup needed. */ ioc = hda->no_ipc_position ? - !stream->no_period_wakeup : 0; + !hstream->no_period_wakeup : 0; for (i = 0; i < periods; i++) { if (i == (periods - 1) && remain) /* set the last small entry */ offset = hda_setup_bdle(sdev, dmab, - stream, &bdl, offset, + hstream, &bdl, offset, remain, 0); else offset = hda_setup_bdle(sdev, dmab, - stream, &bdl, offset, + hstream, &bdl, offset, period_bytes, ioc); } @@ -156,10 +156,10 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, } int hda_dsp_stream_spib_config(struct snd_sof_dev *sdev, - struct hdac_ext_stream *stream, + struct hdac_ext_stream *hext_stream, int enable, u32 size) { - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; u32 mask; if (!sdev->bar[HDA_DSP_SPIB_BAR]) { @@ -175,7 +175,7 @@ int hda_dsp_stream_spib_config(struct snd_sof_dev *sdev, enable << hstream->index); /* set the SPIB value */ - sof_io_write(sdev, stream->spib_addr, size); + sof_io_write(sdev, hext_stream->spib_addr, size); return 0; } @@ -186,7 +186,7 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags) { struct hdac_bus *bus = sof_to_bus(sdev); struct sof_intel_hda_stream *hda_stream; - struct hdac_ext_stream *stream = NULL; + struct hdac_ext_stream *hext_stream = NULL; struct hdac_stream *s; spin_lock_irq(&bus->reg_lock); @@ -194,10 +194,10 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags) /* get an unused stream */ list_for_each_entry(s, &bus->stream_list, list) { if (s->direction == direction && !s->opened) { - stream = stream_to_hdac_ext_stream(s); - hda_stream = container_of(stream, + hext_stream = stream_to_hdac_ext_stream(s); + hda_stream = container_of(hext_stream, struct sof_intel_hda_stream, - hda_stream); + hext_stream); /* check if the host DMA channel is reserved */ if (hda_stream->host_reserved) continue; @@ -210,11 +210,11 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags) spin_unlock_irq(&bus->reg_lock); /* stream found ? */ - if (!stream) { + if (!hext_stream) { dev_err(sdev->dev, "error: no free %s streams\n", direction == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); - return stream; + return hext_stream; } hda_stream->flags = flags; @@ -229,7 +229,7 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags) HDA_VS_INTEL_EM2, HDA_VS_INTEL_EM2_L1SEN, 0); - return stream; + return hext_stream; } /* free a stream */ @@ -237,7 +237,7 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) { struct hdac_bus *bus = sof_to_bus(sdev); struct sof_intel_hda_stream *hda_stream; - struct hdac_ext_stream *stream; + struct hdac_ext_stream *hext_stream; struct hdac_stream *s; bool dmi_l1_enable = true; bool found = false; @@ -249,8 +249,8 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) * that are DMI L1 incompatible. */ list_for_each_entry(s, &bus->stream_list, list) { - stream = stream_to_hdac_ext_stream(s); - hda_stream = container_of(stream, struct sof_intel_hda_stream, hda_stream); + hext_stream = stream_to_hdac_ext_stream(s); + hda_stream = container_of(hext_stream, struct sof_intel_hda_stream, hext_stream); if (!s->opened) continue; @@ -319,9 +319,9 @@ static int hda_dsp_stream_reset(struct snd_sof_dev *sdev, struct hdac_stream *hs } int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, - struct hdac_ext_stream *stream, int cmd) + struct hdac_ext_stream *hext_stream, int cmd) { - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); u32 dma_start = SOF_HDA_SD_CTL_DMA_START; int ret = 0; @@ -396,17 +396,17 @@ int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, } /* minimal recommended programming for ICCMAX stream */ -int hda_dsp_iccmax_stream_hw_params(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream, +int hda_dsp_iccmax_stream_hw_params(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params) { struct hdac_bus *bus = sof_to_bus(sdev); - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); int ret; u32 mask = 0x1 << hstream->index; - if (!stream) { + if (!hext_stream) { dev_err(sdev->dev, "error: no stream available\n"); return -ENODEV; } @@ -467,20 +467,20 @@ int hda_dsp_iccmax_stream_hw_params(struct snd_sof_dev *sdev, struct hdac_ext_st * and normal stream. */ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, - struct hdac_ext_stream *stream, + struct hdac_ext_stream *hext_stream, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params) { const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); struct hdac_bus *bus = sof_to_bus(sdev); - struct hdac_stream *hstream = &stream->hstream; + struct hdac_stream *hstream = &hext_stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); int ret; u32 dma_start = SOF_HDA_SD_CTL_DMA_START; u32 mask; u32 run; - if (!stream) { + if (!hext_stream) { dev_err(sdev->dev, "error: no stream available\n"); return -ENODEV; } @@ -659,28 +659,28 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream) { - struct hdac_stream *stream = substream->runtime->private_data; - struct hdac_ext_stream *link_dev = container_of(stream, - struct hdac_ext_stream, - hstream); + struct hdac_stream *hstream = substream->runtime->private_data; + struct hdac_ext_stream *hext_stream = container_of(hstream, + struct hdac_ext_stream, + hstream); struct hdac_bus *bus = sof_to_bus(sdev); - u32 mask = 0x1 << stream->index; + u32 mask = 0x1 << hstream->index; int ret; - ret = hda_dsp_stream_reset(sdev, stream); + ret = hda_dsp_stream_reset(sdev, hstream); if (ret < 0) return ret; spin_lock_irq(&bus->reg_lock); /* couple host and link DMA if link DMA channel is idle */ - if (!link_dev->link_locked) + if (!hext_stream->link_locked) snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, mask, 0); spin_unlock_irq(&bus->reg_lock); - hda_dsp_stream_spib_config(sdev, link_dev, HDA_DSP_SPIB_DISABLE, 0); + hda_dsp_stream_spib_config(sdev, hext_stream, HDA_DSP_SPIB_DISABLE, 0); - stream->substream = NULL; + hstream->substream = NULL; return 0; } @@ -808,7 +808,7 @@ irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) int hda_dsp_stream_init(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); - struct hdac_ext_stream *stream; + struct hdac_ext_stream *hext_stream; struct hdac_stream *hstream; struct pci_dev *pci = to_pci_dev(sdev->dev); struct sof_intel_hda_dev *sof_hda = bus_to_sof_hda(bus); @@ -872,27 +872,27 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) hda_stream->sdev = sdev; - stream = &hda_stream->hda_stream; + hext_stream = &hda_stream->hext_stream; - stream->pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + hext_stream->pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + SOF_HDA_PPHC_BASE + SOF_HDA_PPHC_INTERVAL * i; - stream->pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + hext_stream->pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + SOF_HDA_PPLC_BASE + SOF_HDA_PPLC_MULTI * num_total + SOF_HDA_PPLC_INTERVAL * i; /* do we support SPIB */ if (sdev->bar[HDA_DSP_SPIB_BAR]) { - stream->spib_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + hext_stream->spib_addr = sdev->bar[HDA_DSP_SPIB_BAR] + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + SOF_HDA_SPIB_SPIB; - stream->fifo_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + hext_stream->fifo_addr = sdev->bar[HDA_DSP_SPIB_BAR] + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + SOF_HDA_SPIB_MAXFIFO; } - hstream = &stream->hstream; + hstream = &hext_stream->hstream; hstream->bus = bus; hstream->sd_int_sta_mask = 1 << i; hstream->index = i; @@ -927,28 +927,28 @@ int hda_dsp_stream_init(struct snd_sof_dev *sdev) hda_stream->sdev = sdev; - stream = &hda_stream->hda_stream; + hext_stream = &hda_stream->hext_stream; /* we always have DSP support */ - stream->pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + + hext_stream->pphc_addr = sdev->bar[HDA_DSP_PP_BAR] + SOF_HDA_PPHC_BASE + SOF_HDA_PPHC_INTERVAL * i; - stream->pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + + hext_stream->pplc_addr = sdev->bar[HDA_DSP_PP_BAR] + SOF_HDA_PPLC_BASE + SOF_HDA_PPLC_MULTI * num_total + SOF_HDA_PPLC_INTERVAL * i; /* do we support SPIB */ if (sdev->bar[HDA_DSP_SPIB_BAR]) { - stream->spib_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + hext_stream->spib_addr = sdev->bar[HDA_DSP_SPIB_BAR] + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + SOF_HDA_SPIB_SPIB; - stream->fifo_addr = sdev->bar[HDA_DSP_SPIB_BAR] + + hext_stream->fifo_addr = sdev->bar[HDA_DSP_SPIB_BAR] + SOF_HDA_SPIB_BASE + SOF_HDA_SPIB_INTERVAL * i + SOF_HDA_SPIB_MAXFIFO; } - hstream = &stream->hstream; + hstream = &hext_stream->hstream; hstream->bus = bus; hstream->sd_int_sta_mask = 1 << i; hstream->index = i; @@ -983,7 +983,7 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *s, *_s; - struct hdac_ext_stream *stream; + struct hdac_ext_stream *hext_stream; struct sof_intel_hda_stream *hda_stream; /* free position buffer */ @@ -1003,9 +1003,9 @@ void hda_dsp_stream_free(struct snd_sof_dev *sdev) if (s->bdl.area) snd_dma_free_pages(&s->bdl); list_del(&s->list); - stream = stream_to_hdac_ext_stream(s); - hda_stream = container_of(stream, struct sof_intel_hda_stream, - hda_stream); + hext_stream = stream_to_hdac_ext_stream(s); + hda_stream = container_of(hext_stream, struct sof_intel_hda_stream, + hext_stream); devm_kfree(sdev->dev, hda_stream); } } diff --git a/sound/soc/sof/intel/hda-trace.c b/sound/soc/sof/intel/hda-trace.c index 1791ec045a54..755ef1d835e0 100644 --- a/sound/soc/sof/intel/hda-trace.c +++ b/sound/soc/sof/intel/hda-trace.c @@ -22,14 +22,14 @@ static int hda_dsp_trace_prepare(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab) { struct sof_intel_hda_dev *hda = sdev->pdata->hw_pdata; - struct hdac_ext_stream *stream = hda->dtrace_stream; - struct hdac_stream *hstream = &stream->hstream; + struct hdac_ext_stream *hext_stream = hda->dtrace_stream; + struct hdac_stream *hstream = &hext_stream->hstream; int ret; hstream->period_bytes = 0;/* initialize period_bytes */ hstream->bufsize = dmab->bytes; - ret = hda_dsp_stream_hw_params(sdev, stream, dmab, NULL); + ret = hda_dsp_stream_hw_params(sdev, hext_stream, dmab, NULL); if (ret < 0) dev_err(sdev->dev, "error: hdac prepare failed: %d\n", ret); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 2390561906dd..324418582044 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -471,14 +471,14 @@ static inline struct hda_bus *sof_to_hbus(struct snd_sof_dev *s) struct sof_intel_hda_stream { struct snd_sof_dev *sdev; - struct hdac_ext_stream hda_stream; - struct sof_intel_stream stream; + struct hdac_ext_stream hext_stream; + struct sof_intel_stream sof_intel_stream; int host_reserved; /* reserve host DMA channel */ u32 flags; }; #define hstream_to_sof_hda_stream(hstream) \ - container_of(hstream, struct sof_intel_hda_stream, hda_stream) + container_of(hstream, struct sof_intel_hda_stream, hext_stream) #define bus_to_sof_hda(bus) \ container_of(bus, struct sof_intel_hda_dev, hbus.core) @@ -545,18 +545,19 @@ int hda_dsp_pcm_ack(struct snd_sof_dev *sdev, struct snd_pcm_substream *substrea int hda_dsp_stream_init(struct snd_sof_dev *sdev); void hda_dsp_stream_free(struct snd_sof_dev *sdev); int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, - struct hdac_ext_stream *stream, + struct hdac_ext_stream *hext_stream, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params); -int hda_dsp_iccmax_stream_hw_params(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream, +int hda_dsp_iccmax_stream_hw_params(struct snd_sof_dev *sdev, + struct hdac_ext_stream *hext_stream, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params); int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, - struct hdac_ext_stream *stream, int cmd); + struct hdac_ext_stream *hext_stream, int cmd); irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context); int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, - struct hdac_stream *stream); + struct hdac_stream *hstream); bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); @@ -564,7 +565,7 @@ 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); int hda_dsp_stream_spib_config(struct snd_sof_dev *sdev, - struct hdac_ext_stream *stream, + struct hdac_ext_stream *hext_stream, int enable, u32 size); int hda_ipc_msg_data(struct snd_sof_dev *sdev, -- 2.35.1

2 1

[PATCH] MAINTAINERS: update cros_ec_codec maintainers
by Tzung-Bi Shih 09 Feb '22

09 Feb '22

Updates cros_ec_codec maintainers. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 1 + MAINTAINERS | 1 + 2 files changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index 77adbebed824..c3e9f3485449 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,6 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> + - Tzung-Bi Shih <tzungbi(a)google.com> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/MAINTAINERS b/MAINTAINERS index 568152aa9973..a1f4fd15e770 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4573,6 +4573,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> +M: Tzung-Bi Shih <tzungbi(a)google.com> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.35.0.263.gb82422642f-goog

4 3

[PATCH] ALSA: hda: Fix driver index handling at re-binding
by Takashi Iwai 09 Feb '22

09 Feb '22

HD-audio driver handles the multiple instances and keeps the static index that is incremented at each probe. This becomes a problem when user tries to re-bind the device via sysfs multiple times; as the device index isn't cleared unlike rmmod case, it points to the next element at re-binding, and eventually later you can't probe any more when it reaches to SNDRV_CARDS_MAX (usually 32). This patch is an attempt to improve the handling at rebinding. Instead of a static device index, now we keep a bitmap and assigns to the first zero bit position. At the driver remove, in return, the bitmap slot is cleared again, so that it'll be available for the next probe. Reported-by: Alexander Sergeyev <sergeev917(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4b0338c4c543..a2922233e85f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2064,14 +2064,16 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; +static DECLARE_BITMAP(probed_devs, SNDRV_CARDS); + static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { - static int dev; struct snd_card *card; struct hda_intel *hda; struct azx *chip; bool schedule_probe; + int dev; int err; if (pci_match_id(driver_denylist, pci)) { @@ -2079,10 +2081,11 @@ static int azx_probe(struct pci_dev *pci, return -ENODEV; } + dev = find_first_zero_bit(probed_devs, SNDRV_CARDS); if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { - dev++; + set_bit(dev, probed_devs); return -ENOENT; } @@ -2149,7 +2152,7 @@ static int azx_probe(struct pci_dev *pci, if (schedule_probe) schedule_delayed_work(&hda->probe_work, 0); - dev++; + set_bit(dev, probed_devs); if (chip->disabled) complete_all(&hda->probe_wait); return 0; @@ -2372,6 +2375,7 @@ static void azx_remove(struct pci_dev *pci) cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); + clear_bit(chip->dev_index, probed_devs); pci_set_drvdata(pci, NULL); snd_card_free(card); } -- 2.34.1

1 0

[PATCH v1 0/6] Add Tegra234 HDA support
by Mohan Kumar 09 Feb '22

09 Feb '22

This series add the support for TEGRA234 HDA driver support Mohan Kumar (6): ALSA: hda/tegra: Add Tegra234 hda driver support ALSA: hda/tegra: Hardcode GCAP ISS value on T234 ALSA: hda/tegra: Update scratch reg. communication dt-bindings: Add HDA support for Tegra234 dt-bindings: Document Tegra234 HDA support arm64: tegra: Add hda dts node for Tegra234 .../bindings/sound/nvidia,tegra30-hda.yaml | 3 + .../nvidia/tegra234-p3737-0000+p3701-0000.dts | 6 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 18 +++ include/dt-bindings/clock/tegra234-clock.h | 4 + include/dt-bindings/memory/tegra234-mc.h | 6 + .../dt-bindings/power/tegra234-powergate.h | 9 ++ include/dt-bindings/reset/tegra234-reset.h | 2 + include/sound/hda_codec.h | 4 + sound/pci/hda/hda_tegra.c | 33 ++++- sound/pci/hda/patch_hdmi.c | 115 ++++++++++++++---- 10 files changed, 176 insertions(+), 24 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.17.1

2 7

[PATCH 1/4] ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8
by Jeremy Szu 08 Feb '22

08 Feb '22

The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the fixup, the mute/micmute LEDs work good. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- 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 552e2cb73291..9d68f591c6bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8291,6 +8291,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.31.1

3 32

[PATCH] ASoC: fsl_sai: Enable combine mode soft
by Sascha Hauer 08 Feb '22

08 Feb '22

The fsl_sai driver calculates the number of pins used and enables multiple channels if necessary. This means the SAI expects data in one FIFO per pin. The SDMA engine only services a single FIFO, so multi pin support doesn't work at all. This patch enables the software combine mode in chips that support it. With this the SAI presents only a single FIFO to the outside and distributes the data into the different FIFOs internally. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_sai.c | 9 +++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 10544fa27dc0..cab015d96889 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -517,6 +517,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } + if (sai->soc_data->pins > 1) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FCOMB_MASK, FSL_SAI_CR4_FCOMB_SOFT); + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE((1 << pins) - 1)); @@ -1195,6 +1199,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, .use_edma = false, .fifo_depth = 32, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, @@ -1204,6 +1209,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 32, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = true, .flags = 0, @@ -1213,6 +1219,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 16, + .pins = 2, .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = PMQOS_CPU_LATENCY, @@ -1222,6 +1229,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 128, + .pins = 8, .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = 0, @@ -1231,6 +1239,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .use_edma = true, .fifo_depth = 64, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 9aaf231bc024..410f6e6a9137 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -223,6 +223,7 @@ struct fsl_sai_soc_data { bool use_edma; bool mclk0_is_mclk1; unsigned int fifo_depth; + unsigned int pins; unsigned int reg_offset; unsigned int flags; }; -- 2.30.2

2 2

[PATCH v1 0/6] ASoC: Fix gpiolib dependencies
by Mark Brown 08 Feb '22

08 Feb '22

This series removes a bunch of spurious selects of gpiolib that were causing noise in randconfig build tests. Mark Brown (6): ASoC: dmic: Remove spurious gpiolib select ASoC: rt9120: Remove spurious gpiolib select ASoC: simple-amplifier: Remove spurious gpiolib select ASoC: max9759: Remove spurious gpiolib select ASoC: zl38060: Remove spurious gpiolib select ASoC: simple-mux: Depend on gpiolib rather than selecting it sound/soc/codecs/Kconfig | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) base-commit: e783362eb54cd99b2cac8b3a9aeac942e6f6ac07 -- 2.30.2

1 7

[PATCH v3] ASoC: SOF: compr: Add compress ops implementation
by Daniel Baluta 08 Feb '22

08 Feb '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> Implement snd_compress_ops. There are a lot of similarities with PCM implementation. For now we use sof_ipc_pcm_params to transfer compress parameters to SOF firmware. This will be changed in the future once we either add new compress parameters to SOF or enhance existing sof_ipc_pcm_params structure to support all native compress params. Note that get_caps and get_codec_caps are missing and will be added later. This is because we need to find a way to advertise DSP capabilities depending on supported platforms. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v2: - dropped patch 2/2 because deciding on how to support get_caps and get_codec_caps needs some rework. - annotated commit message for 1/2 to mention about get_caps and get_codec_caps Changes since v1: (https://lore.kernel.org/lkml/20220113161341.371345-1-daniel.baluta@oss.nxp.…) - Addressed review from Cezary and Pierre - fixed layout of declaration blocks to be more consistent - avoid using rtd and runtime simultaneously inside a function (always used rtd and crtd) - check return code for create_page_table - completely remove sof_compr_stream and use snd_compr_tstmap instead to keep compress stream related info.· - add get_caps and get_codec_caps implementations (in patch 2)· sound/soc/sof/compress.c | 273 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 271 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 01ca85f0b87f..91a9c95929cd 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -10,6 +10,22 @@ #include "sof-audio.h" #include "sof-priv.h" +static void sof_set_transferred_bytes(struct snd_compr_tstamp *tstamp, + u64 host_pos, u64 buffer_size) +{ + u64 prev_pos; + unsigned int copied; + + div64_u64_rem(tstamp->copied_total, buffer_size, &prev_pos); + + if (host_pos < prev_pos) + copied = (buffer_size - prev_pos) + host_pos; + else + copied = host_pos - prev_pos; + + tstamp->copied_total += copied; +} + static void snd_sof_compr_fragment_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = @@ -29,14 +45,16 @@ void snd_sof_compr_init_elapsed_work(struct work_struct *work) */ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) { + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; struct snd_soc_component *component; - struct snd_soc_pcm_runtime *rtd; + struct snd_compr_tstamp *tstamp; struct snd_sof_pcm *spcm; if (!cstream) return; - rtd = cstream->private_data; + tstamp = crtd->private_data; component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); spcm = snd_sof_find_spcm_dai(component, rtd); @@ -46,6 +64,257 @@ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) return; } + sof_set_transferred_bytes(tstamp, spcm->stream[cstream->direction].posn.host_posn, + crtd->buffer_size); + /* use the same workqueue-based solution as for PCM, cf. snd_sof_pcm_elapsed */ schedule_work(&spcm->stream[cstream->direction].period_elapsed_work); } + +static int create_page_table(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + unsigned char *dma_area, size_t size) +{ + struct snd_dma_buffer *dmab = cstream->runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + int dir = cstream->direction; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + return snd_sof_create_page_table(component->dev, dmab, + spcm->stream[dir].page_table.area, size); +} + +int sof_compr_open(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; + struct snd_compr_tstamp *tstamp; + struct snd_sof_pcm *spcm; + int dir; + + tstamp = kzalloc(sizeof(*tstamp), GFP_KERNEL); + if (!tstamp) + return -ENOMEM; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) { + kfree(tstamp); + return -EINVAL; + } + + dir = cstream->direction; + + if (spcm->stream[dir].cstream) { + kfree(tstamp); + return -EBUSY; + } + + spcm->stream[dir].cstream = cstream; + spcm->stream[dir].posn.host_posn = 0; + spcm->stream[dir].posn.dai_posn = 0; + spcm->prepared[dir] = false; + + crtd->private_data = tstamp; + + return 0; +} + +int sof_compr_free(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_compr_tstamp *tstamp = cstream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_sof_pcm *spcm; + int ret = 0; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + if (spcm->prepared[cstream->direction]) { + ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); + if (!ret) + spcm->prepared[cstream->direction] = false; + } + + cancel_work_sync(&spcm->stream[cstream->direction].period_elapsed_work); + spcm->stream[cstream->direction].cstream = NULL; + kfree(tstamp); + + return ret; +} + +int sof_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_compr_params *params) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; + struct sof_ipc_pcm_params_reply ipc_params_reply; + struct snd_compr_tstamp *tstamp; + struct sof_ipc_pcm_params pcm; + struct snd_sof_pcm *spcm; + int ret; + + tstamp = crtd->private_data; + + spcm = snd_sof_find_spcm_dai(component, rtd); + + if (!spcm) + return -EINVAL; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, crtd->buffer_size); + if (ret < 0) + return ret; + + ret = create_page_table(component, cstream, crtd->dma_area, crtd->dma_bytes); + if (ret < 0) + return ret; + + memset(&pcm, 0, sizeof(pcm)); + + pcm.params.buffer.pages = PFN_UP(crtd->dma_bytes); + pcm.hdr.size = sizeof(pcm); + pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; + + pcm.comp_id = spcm->stream[cstream->direction].comp_id; + pcm.params.hdr.size = sizeof(pcm.params); + pcm.params.buffer.phy_addr = spcm->stream[cstream->direction].page_table.addr; + pcm.params.buffer.size = crtd->dma_bytes; + pcm.params.direction = cstream->direction; + pcm.params.channels = params->codec.ch_out; + pcm.params.rate = params->codec.sample_rate; + pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; + pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE; + pcm.params.sample_container_bytes = + snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32) >> 3; + pcm.params.host_period_bytes = params->buffer.fragment_size; + + ret = sof_ipc_tx_message(sdev->ipc, pcm.hdr.cmd, &pcm, sizeof(pcm), + &ipc_params_reply, sizeof(ipc_params_reply)); + if (ret < 0) { + dev_err(component->dev, "error ipc failed\n"); + return ret; + } + + tstamp->byte_offset = sdev->stream_box.offset + ipc_params_reply.posn_offset; + tstamp->sampling_rate = params->codec.sample_rate; + + spcm->prepared[cstream->direction] = true; + + return 0; +} + +int sof_compr_get_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_codec *params) +{ + /* TODO: we don't query the supported codecs for now, if the + * application asks for an unsupported codec the set_params() will fail. + */ + return 0; +} + +int sof_compr_trigger(struct snd_soc_component *component, + struct snd_compr_stream *cstream, int cmd) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE; + break; + default: + dev_err(component->dev, "error: unhandled trigger cmd %d\n", cmd); + break; + } + + return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); +} + +int sof_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_available, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_from_user(ptr, buf, count); + } else { + ret = copy_from_user(ptr, buf, n); + ret += copy_from_user(rtd->dma_area, buf + n, count - n); + } + + return count - ret; +} + +static int sof_compr_pointer(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_tstamp *pstamp = cstream->runtime->private_data; + + tstamp->sampling_rate = pstamp->sampling_rate; + tstamp->copied_total = pstamp->copied_total; + + return 0; +} + +struct snd_compress_ops sof_compressed_ops = { + .open = sof_compr_open, + .free = sof_compr_free, + .set_params = sof_compr_set_params, + .get_params = sof_compr_get_params, + .trigger = sof_compr_trigger, + .pointer = sof_compr_pointer, + .copy = sof_compr_copy, +}; +EXPORT_SYMBOL(sof_compressed_ops); -- 2.27.0

2 1

[PATCH 0/2] ASoC: SOF: dma-trace: Change trace_init() ops parameter list
by Peter Ujfalusi 08 Feb '22

08 Feb '22

Hi, the DMA trace implementation on AMD platform assumes that the stream_tag pointer is pointing the stream_tag member of struct sof_ipc_dma_trace_params_ext, which is true at the moment, but it can not be guarantied and a change in the dtrace core can cause out of bound accesses for AMD. For this reason, change the API to pass the struct itself which will remove the assumption and makes it clear from both sides what is expected to be sent via the parameter list. This opens up a window to clean up the intel and AMD implementation at the same time. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: intel: hda-trace: Pass the dma buffer pointer to hda_dsp_trace_prepare ASoC: SOF: dma-trace: Pass pointer to params_ext struct in trace_init() sound/soc/sof/amd/acp-trace.c | 38 ++++++++------------------------- sound/soc/sof/amd/acp.h | 3 ++- sound/soc/sof/intel/hda-trace.c | 17 ++++++++------- sound/soc/sof/intel/hda.h | 3 ++- sound/soc/sof/ops.h | 4 ++-- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sof/trace.c | 2 +- 7 files changed, 26 insertions(+), 43 deletions(-) -- 2.35.0

2 3

[PATCH v5] ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema
by Jayesh Choudhary 08 Feb '22

08 Feb '22

Convert the bindings for McASP controllers for TI SOCs from txt to YAML schema. Adds additional properties 'clocks', 'clock-names', 'power-domains', '#sound-dai-cells' and 'port' which were missing from the txt file. Removes properties 'sram-size-playback' and 'sram-size-capture' since they are not used. Adds 'dmas' and 'dma-names' in the example which were missing from the txt file. Changes 'interrupts' and 'interrupt-names' from optional to required properties. Changes 'op-modes', 'serial-dir' to optional properties as they are not needed if the McASP is used only as GPIO. Changes 'tdm-slots' to required property only for I2S operation mode. Adds the yaml file in the 'MAINTAINERS' under the heading 'TEXAS INSTRUMENTS ASoC DRIVERS' Signed-off-by: Jayesh Choudhary <j-choudhary(a)ti.com> --- Changelog: v5: - changes the commit message - changes description for 'reg', 'op-mode', 'tx-num-evt', 'rx-num-evt' and 'interrupts' - adds minimum and maximum for 'tdm-slots' - removes default from 'serial-dir' - removes maxItems from 'tdm-slots', 'tx-num-evt', 'rx-num-evt' and 'auxclk-fs-ratio' as they are of type uint32 - adds 'tdm-slots' as conditionally required property for I2S mode of operation v4: - changes the commit message - adds type and description to relevant properties - changes maxItems for 'serial-dir' - removes properties 'sram-size-playback' and 'sram-size-capture' as they are not used - removes 'function-gpios' - removes 'num-serializer' - marks 'tdm-slots', 'op-mode' and 'serial-dir' as optional properties - adds the yaml file in MAINTAINERS v3: - remove maxItems from 'clock-names' v2: - changes the commit message - modifies the properties 'clocks', 'clock-names', 'dma-names', 'dmas', 'interrupts' and 'interrupt-names' according to the arm SOCs - adds 'ports' and 'num-serializer' as node properties .../bindings/sound/davinci-mcasp-audio.txt | 86 -------- .../bindings/sound/davinci-mcasp-audio.yaml | 201 ++++++++++++++++++ MAINTAINERS | 1 + 3 files changed, 202 insertions(+), 86 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt deleted file mode 100644 index bd863bd69501..000000000000 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ /dev/null @@ -1,86 +0,0 @@ -Texas Instruments McASP controller - -Required properties: -- compatible : - "ti,dm646x-mcasp-audio" : for DM646x platforms - "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, AM43xx, TI81xx) - "ti,dra7-mcasp-audio" : for DRA7xx platforms - "ti,omap4-mcasp-audio" : for OMAP4 - -- reg : Should contain reg specifiers for the entries in the reg-names property. -- reg-names : Should contain: - * "mpu" for the main registers (required). For compatibility with - existing software, it is recommended this is the first entry. - * "dat" for separate data port register access (optional). -- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, - IEC60958-1, and AES-3 formats. -- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted - or received over one serializer. -- serial-dir : A list of serializer configuration. Each entry is a number - indication for serializer pin direction. - (0 - INACTIVE, 1 - TX, 2 - RX) -- dmas: two element list of DMA controller phandles and DMA request line - ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. The dma - identifiers must be "rx" and "tx". - -Optional properties: - -- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0 -- tx-num-evt : FIFO levels. -- rx-num-evt : FIFO levels. -- dismod : Specify the drive on TX pin during inactive slots - 0 : 3-state - 2 : logic low - 3 : logic high - Defaults to 'logic low' when the property is not present -- sram-size-playback : size of sram to be allocated during playback -- sram-size-capture : size of sram to be allocated during capture -- interrupts : Interrupt numbers for McASP -- interrupt-names : Known interrupt names are "tx" and "rx" -- pinctrl-0: Should specify pin control group used for this controller. -- pinctrl-names: Should contain only one value - "default", for more details - please refer to pinctrl-bindings.txt -- fck_parent : Should contain a valid clock name which will be used as parent - for the McASP fck -- auxclk-fs-ratio: When McASP is bus master indicates the ratio between AUCLK - and FS rate if applicable: - AUCLK rate = auxclk-fs-ratio * FS rate - -Optional GPIO support: -If any McASP pin need to be used as GPIO then the McASP node must have: -... - gpio-controller - #gpio-cells = <2>; -... - -When requesting a GPIO, the first parameter is the PIN index in McASP_P* -registers. -For example to request the AXR2 pin of mcasp8: -function-gpios = <&mcasp8 2 0>; - -Or to request the ACLKR pin of mcasp8: -function-gpios = <&mcasp8 29 0>; - -For generic gpio information, please refer to bindings/gpio/gpio.txt - -Example: - -mcasp0: mcasp0@1d00000 { - compatible = "ti,da830-mcasp-audio"; - reg = <0x100000 0x3000>; - reg-names "mpu"; - interrupts = <82>, <83>; - interrupt-names = "tx", "rx"; - op-mode = <0>; /* MCASP_IIS_MODE */ - tdm-slots = <2>; - serial-dir = < - 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ - 0 0 0 0 - 0 0 0 1 - 2 0 0 0 >; - tx-num-evt = <1>; - rx-num-evt = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml new file mode 100644 index 000000000000..f46c66bc6b2d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -0,0 +1,201 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/davinci-mcasp-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: McASP Controller for TI SoCs + +maintainers: + - Jayesh Choudhary <j-choudhary(a)ti.com> + +properties: + compatible: + enum: + - ti,dm646x-mcasp-audio + - ti,da830-mcasp-audio + - ti,am33xx-mcasp-audio + - ti,dra7-mcasp-audio + - ti,omap4-mcasp-audio + + reg: + minItems: 1 + items: + - description: CFG registers + - description: data registers + + reg-names: + minItems: 1 + items: + - const: mpu + - const: dat + + op-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: 0 - I2S or 1 - DIT operation mode + enum: + - 0 + - 1 + + tdm-slots: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number of channels over one serializer + the property is ignored in DIT mode + minimum: 2 + maximum: 32 + + serial-dir: + description: + A list of serializer configuration + Entry is indication for serializer pin direction + 0 - Inactive, 1 - TX, 2 - RX + All AXR pins should be present in the array even if inactive + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 1 + maxItems: 25 + items: + minimum: 0 + maximum: 2 + + dmas: + minItems: 1 + items: + - description: transmission DMA channel + - description: reception DMA channel + + dma-names: + minItems: 1 + items: + - const: tx + - const: rx + + ti,hwmods: + $ref: /schemas/types.yaml#/definitions/string + description: Name of hwmod associated with McASP + maxItems: 1 + deprecated: true + + tx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures WFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + rx-num-evt: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + configures RFIFO threshold + 0 disables the FIFO use + if property is missing, then also FIFO use is disabled + + dismod: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + specify the drive on TX pin during inactive time slots + 0 - 3-state, 2 - logic low, 3 - logic high + enum: + - 0 + - 2 + - 3 + default: 2 + + interrupts: + anyOf: + - minItems: 1 + items: + - description: TX interrupt + - description: RX interrupt + - items: + - description: common/combined interrupt + + interrupt-names: + oneOf: + - minItems: 1 + items: + - const: tx + - const: rx + - const: common + + fck_parent: + $ref: /schemas/types.yaml#/definitions/string + description: parent clock name for McASP fck + maxItems: 1 + + auxclk-fs-ratio: + $ref: /schemas/types.yaml#/definitions/uint32 + description: ratio of AUCLK and FS rate if applicable + + gpio-controller: true + + "#gpio-cells": + const: 2 + + clocks: + minItems: 1 + items: + - description: functional clock + - description: module specific optional ahclkx clock + - description: module specific optional ahclkr clock + + clock-names: + minItems: 1 + items: + - const: fck + - const: ahclkx + - const: ahclkr + + power-domains: + description: phandle to the corresponding power-domain + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + port: + description: connection for when McASP is used via graph card + type: object + +required: + - compatible + - reg + - reg-names + - dmas + - dma-names + - interrupts + - interrupt-names + +allOf: + - if: + properties: + opmode: + enum: + - 0 + + then: + required: + - tdm-slots + +additionalProperties: false + +examples: + - | + mcasp0: mcasp0@1d00000 { + compatible = "ti,da830-mcasp-audio"; + reg = <0x100000 0x3000>; + reg-names = "mpu"; + interrupts = <82>, <83>; + interrupt-names = "tx", "rx"; + op-mode = <0>; /* MCASP_IIS_MODE */ + tdm-slots = <2>; + dmas = <&main_udmap 0xc400>, <&main_udmap 0x4400>; + dma-names = "tx", "rx"; + serial-dir = < + 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ + 0 0 0 0 + 0 0 0 1 + 2 0 0 0 >; + tx-num-evt = <1>; + rx-num-evt = <1>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 5ea5655a29c3..ec6bde4efcae 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -18995,6 +18995,7 @@ TEXAS INSTRUMENTS ASoC DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Maintained +F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ TEXAS INSTRUMENTS' DAC7612 DAC DRIVER -- 2.17.1

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[PATCH 1/3] ASoC: rt5682s: do not block workqueue if card is unbound
by Kai Vehmanen 08 Feb '22

08 Feb '22

The current rt5682s_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index efa1016831dd..1e662d1be2b3 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -824,11 +824,13 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5682s_priv, jack_detect_work.work); int val, btn_type; - while (!rt5682s->component) - usleep_range(10000, 15000); - - while (!rt5682s->component->card->instantiated) - usleep_range(10000, 15000); + if (!rt5682s->component || !rt5682s->component->card || + !rt5682s->component->card->instantiated) { + /* card not yet ready, try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5682s->jack_detect_work, msecs_to_jiffies(15)); + return; + } mutex_lock(&rt5682s->jdet_mutex); mutex_lock(&rt5682s->calibrate_mutex); base-commit: d466706b9d0dae4754a47c34fd5f4654ee54f609 -- 2.35.1

2 3

[PATCH v2] ASoC: tas2770: Insert post reset delay
by Martin Povišer 08 Feb '22

08 Feb '22

Per TAS2770 datasheet there must be a 1 ms delay from reset to first command. So insert delays into the driver where appropriate. Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- In previous submission of this patch some people got their copy encrypted. That was by accident, pardon that. I have changed my email provider since. v1: https://lore.kernel.org/alsa-devel/20220201234612.74401-1-povik+lin@protonm… sound/soc/codecs/tas2770.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index b52c0861edc8..b397a2122dd4 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -38,10 +38,12 @@ static void tas2770_reset(struct tas2770_priv *tas2770) gpiod_set_value_cansleep(tas2770->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2770->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2770->component, TAS2770_SW_RST, TAS2770_RST); + usleep_range(1000, 2000); } static int tas2770_set_bias_level(struct snd_soc_component *component, @@ -110,6 +112,7 @@ static int tas2770_codec_resume(struct snd_soc_component *component) if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); } else { ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, @@ -529,8 +532,10 @@ static int tas2770_codec_probe(struct snd_soc_component *component) tas2770->component = component; - if (tas2770->sdz_gpio) + if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2770_reset(tas2770); -- 2.33.0

2 1

Re: [PATCH 5/5] spi: make remove callback a void function
by Mark Brown 08 Feb '22

08 Feb '22

On Sun, Jan 23, 2022 at 06:52:01PM +0100, Uwe Kleine-König wrote: > The value returned by an spi driver's remove function is mostly ignored. > (Only an error message is printed if the value is non-zero that the > error is ignored.) > > So change the prototype of the remove function to return no value. This > way driver authors are not tempted to assume that passing an error to > the upper layer is a good idea. All drivers are adapted accordingly. > There is no intended change of behaviour, all callbacks were prepared to > return 0 before. I was going to apply this but it needs rebasing against current code unfortunately.

1 0

[PATCH v1] ASoC: nau8822: export right speaker inversion
by Francesco Dolcini 08 Feb '22

08 Feb '22

From: Max Krummenacher <max.krummenacher(a)toradex.com> This control inverts the signal fed to the right amplifier. This allows to connect one loudspeaker to the two speaker amplifiers in a bridge-tied load configuration getting higher output power. Signed-off-by: Max Krummenacher <max.krummenacher(a)toradex.com> Signed-off-by: Francesco Dolcini <francesco.dolcini(a)toradex.com> --- sound/soc/codecs/nau8822.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/nau8822.c b/sound/soc/codecs/nau8822.c index 58123390c7a3..bf1155903421 100644 --- a/sound/soc/codecs/nau8822.c +++ b/sound/soc/codecs/nau8822.c @@ -351,6 +351,8 @@ static const struct snd_kcontrol_new nau8822_snd_controls[] = { NAU8822_REG_LHP_VOLUME, NAU8822_REG_RHP_VOLUME, 0, 63, 0, spk_tlv), + SOC_SINGLE("Speaker RInversion Switch", + NAU8822_REG_RIGHT_SPEAKER_CONTROL, 4, 1, 0), SOC_DOUBLE_R("Speaker ZC Switch", NAU8822_REG_LSPKOUT_VOLUME, NAU8822_REG_RSPKOUT_VOLUME, 7, 1, 0), -- 2.25.1

2 5

Re: Bug 215561 - snd-usb-audio:reset_resume error -22 after suspend (fwd from bugzilla)
by Thorsten Leemhuis 08 Feb '22

08 Feb '22

Hi, this is your Linux kernel regression tracker speaking. Top-posting for once, to make this easy accessible to everyone. Below issue is one of those those reports that seem to get stranded in bugzilla.kernel.org Greg, Jaroslav, Takashi, could one of you please take a look? The reporter submitted it as USB bug, but maybe that wasn't a wise choice. Ciao, Thorsten (wearing his 'the Linux kernel's regression tracker' hat) P.S.: As the Linux kernel's regression tracker I'm getting a lot of reports on my table. I can only look briefly into most of them and lack knowledge about most of the areas they concern. I thus unfortunately will sometimes get things wrong or miss something important. I hope that's not the case here; if you think it is, don't hesitate to tell me in a public reply, it's in everyone's interest to set the public record straight. #regzbot poke On 03.02.22 15:59, Thorsten Leemhuis wrote: > Hi, this is your Linux kernel regression tracker speaking. > > There is a regression in bugzilla.kernel.org I'd like to add to the > tracking: > > #regzbot introduced: v5.15..v5.16 > #regzbot from: antifermion(a)protonmail.com <antifermion(a)protonmail.com> > #regzbot title: snd: usb: snd-usb-audio:reset_resume error -22 after suspen > #regzbot link: https://bugzilla.kernel.org/show_bug.cgi?id=215561 > > Quote: > >> If I use the microphone, then suspend and wake up again, I get the error >> >> ``` >> snd-usb-audio 1-5:1.0: reset_resume error -22 >> ``` >> >> Aftewards, audio stops working completely. Both pipewire and pulseaudio (tried both; not at the same time) freeze. >> >> After running >> >> ``` >> rmmod -f snd_usb_audio >> ``` >> >> and replugging the microphone, audio works again. >> Here is the relevant log: >> >> ``` >> [ 157.791937] usbcore: deregistering interface driver snd-usb-audio >> [ 228.517855] usb 1-5: USB disconnect, device number 2 >> [ 236.464334] usb 1-5: new high-speed USB device number 3 using xhci_hcd >> [ 236.798863] usb 1-5: New USB device found, idVendor=0d8c, idProduct=0171, bcdDevice= 1.04 >> [ 236.798867] usb 1-5: New USB device strings: Mfr=3, Product=1, SerialNumber=0 >> [ 236.798869] usb 1-5: Product: USB 2.0 HD Audio >> [ 236.798871] usb 1-5: Manufacturer: C-Media Electronics Inc. >> [ 236.826966] input: C-Media Electronics Inc. USB 2.0 HD Audio Consumer Control as /devices/pci0000:00/0000:00:01.3/0000:02:00.0/usb1/1-5/1-5:1.3/0003:0D8C:0171.000B/input/input40 >> [ 236.878380] input: C-Media Electronics Inc. USB 2.0 HD Audio as /devices/pci0000:00/0000:00:01.3/0000:02:00.0/usb1/1-5/1-5:1.3/0003:0D8C:0171.000B/input/input41 >> [ 236.878447] hid-generic 0003:0D8C:0171.000B: input,hiddev96,hidraw2: USB HID v1.11 Device [C-Media Electronics Inc. USB 2.0 HD Audio] on usb-0000:02:00.0-5/input3 >> [ 242.306593] usbcore: registered new interface driver snd-usb-audio >> >> ``` >> >> I'm running Arch Linux (5.16.4.arch1-1). The linux-lts kernel (5.15.18-1) works fine. >> >> Other uses report the same error: https://bbs.archlinux.org/viewtopic.php?id=273469 > > Ciao, Thorsten (wearing his 'Linux kernel regression tracker' hat) > > P.S.: As a Linux kernel regression tracker I'm getting a lot of reports > on my table. I can only look briefly into most of them. Unfortunately > therefore I sometimes will get things wrong or miss something important. > I hope that's not the case here; if you think it is, don't hesitate to > tell me about it in a public reply, that's in everyone's interest. > > BTW, I have no personal interest in this issue, which is tracked using > regzbot, my Linux kernel regression tracking bot > (https://linux-regtracking.leemhuis.info/regzbot/). I'm only posting > this mail to get things rolling again and hence don't need to be CC on > all further activities wrt to this regression. > > --- > Additional information about regzbot: > > If you want to know more about regzbot, check out its web-interface, the > getting start guide, and/or the references documentation: > > https://linux-regtracking.leemhuis.info/regzbot/ > https://gitlab.com/knurd42/regzbot/-/blob/main/docs/getting_started.md > https://gitlab.com/knurd42/regzbot/-/blob/main/docs/reference.md > > The last two documents will explain how you can interact with regzbot > yourself if your want to. > > Hint for reporters: when reporting a regression it's in your interest to > tell #regzbot about it in the report, as that will ensure the regression > gets on the radar of regzbot and the regression tracker. That's in your > interest, as they will make sure the report won't fall through the > cracks unnoticed. > > Hint for developers: you normally don't need to care about regzbot once > it's involved. Fix the issue as you normally would, just remember to > include a 'Link:' tag to the report in the commit message, as explained > in Documentation/process/submitting-patches.rst > That aspect was recently was made more explicit in commit 1f57bd42b77c: > https://git.kernel.org/linus/1f57bd42b77c

1 0

[PATCH][next] kselftest: alsa: fix spelling mistake "desciptor" -> "descriptor"
by Colin Ian King 08 Feb '22

08 Feb '22

There are some spelling mistakes in some ksft messages. Fix them. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- tools/testing/selftests/alsa/mixer-test.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index d0b788b8d287..eb2213540fe3 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -198,7 +198,7 @@ static void find_controls(void) err = snd_ctl_poll_descriptors_count(card_data->handle); if (err != 1) { - ksft_exit_fail_msg("Unexpected desciptor count %d for card %d\n", + ksft_exit_fail_msg("Unexpected descriptor count %d for card %d\n", err, card); } @@ -248,12 +248,12 @@ static int wait_for_event(struct ctl_data *ctl, int timeout) &(ctl->card->pollfd), 1, &revents); if (err < 0) { - ksft_print_msg("snd_ctl_poll_desciptors_revents() failed for %s: %d\n", + ksft_print_msg("snd_ctl_poll_descriptors_revents() failed for %s: %d\n", ctl->name, err); return err; } if (revents & POLLERR) { - ksft_print_msg("snd_ctl_poll_desciptors_revents() reported POLLERR for %s\n", + ksft_print_msg("snd_ctl_poll_descriptors_revents() reported POLLERR for %s\n", ctl->name); return -1; } -- 2.34.1

3 2

[PATCH] ALSA: usb-audio: remove redundant assignment to variable c
by Colin Ian King 07 Feb '22

07 Feb '22

The variable c is being initialized in an outer for-loop and also re-initialized inside an inner for-loop. The first initialization is redundant and can be removed. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/usb/mixer_s1810c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c index 0255089c9efb..fac4bbc6b275 100644 --- a/sound/usb/mixer_s1810c.c +++ b/sound/usb/mixer_s1810c.c @@ -221,7 +221,7 @@ static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) e = 0xbc; for (n = 0; n < 2; n++) { off = n * 18; - for (b = off, c = 0; b < 18 + off; b++) { + for (b = off; b < 18 + off; b++) { /* This channel to all outputs ? */ for (c = 0; c <= 8; c++) { snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); -- 2.34.1

2 1

[RFC 00/37] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 07 Feb '22

07 Feb '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Several functions found within HDAudio and ASoC framework have been exported and are re-used later by the avs-driver to prevent code being duplicated in this solution. All of these act as driver dependencies and are found at the beginning of the series to make it clear what's needed for avs-driver to compile. Note: this series is dependent upon NHLT-series [6] which was recently merged by Takashi to his for-next branch yet is unavailable on current broonie/for-next. Note: this series does not add fully functional driver as its size would get out of control. Focus is put on adding new code. A SND_SOC_INTEL_SKYLAKE_FAMILY=n dependency is added so no probing attempts are performed. Until all patches are merged, Skylake-driver remains the recommended option. Series which will later follow this one, focus on filling all the remaining functionality gaps and provide filesystems support. Also, a range of machine boards will be added to match range of configurations supported by the avs-driver. Once avs-driver and its collaterals are merged, skylake-driver deprecation and removal process begins, along with all other components that are connected to it e.g.: skylake-driver-only machine boards. As AudioDSP firmware which avs-driver communicates with supports a wide range of audio formats, module configurations and multi-pipeline streams, couple of new concepts are introduce to enable all those functionalities: - Path template and path variants - Runtime path - Conditional path - Granular sound cards (as opposed to 'super-cards') These are later better explained by their respective patches: 'Topology parsing', 'Path management', 'Conditional-path support' and 'Machine board registration'. A 'path' represents a DSP side of audio stream in runtime - is a logical container for pipelines which are themselves composed of modules - processing units. Due to high range of possible audio format combinations, there can be more variants of given path (and thus, its pipelines and modules) than total number of pipelines and module instances which firmware supports concurrently, all the instance IDs are allocated dynamically with help of IDA interface. 'Path templates' come from topology file and describe a pattern which is later used to actually create runtime 'path'. To support modern audio usecases such as WaveOnVoice and EchoReference, conditional paths concept came into existence. These work similarly to standard paths except are a consequence of other paths being created or deleted, rather then being created when userspace app opens specific FE for streaming. Their state machine is controlled by source and sink paths which created them in the first place. Granular machine boards is a contrast to 'super-card' idea which is currently widely used throughout Intel ASoC drivers. Major reasons are: complexity reduction (each board now focuses on a single, concrete device) and overall reduction of topology file size when entire configuration is taken into account. This has functional benefits too: one card failing won't prevent others from probing and being operative. Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/all/YaDq7L1Mu++3UBL7@sirena.org.uk/T/ Cezary Rojewski (37): ALSA: hda: Add snd_hdac_ext_bus_link_at() helper ALSA: hda: Update and expose snd_hda_codec_device_init() ALSA: hda: Update and expose codec register procedures ALSA: hda: Expose codec cleanup and power-save functions ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: Topology parsing ASoC: Intel: avs: Path management ASoC: Intel: avs: Conditional-path support ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA ASoC: Intel: avs: Generic soc component driver ASoC: Intel: avs: Generic PCM FE operations ASoC: Intel: avs: non-HDA PCM BE operations ASoC: Intel: avs: HDA PCM BE operations ASoC: Intel: avs: Coredump and recovery flow ASoC: Intel: avs: Prepare for firmware tracing ASoC: Intel: avs: D0ix power state support ASoC: Intel: avs: Event tracing ASoC: Intel: avs: Machine board registration ASoC: Intel: avs: PCI driver implementation ASoC: Intel: avs: Power management ASoC: Intel: avs: SKL-based platforms support ASoC: Intel: avs: APL-based platforms support .../ABI/testing/sysfs-bus-pci-devices-avs | 24 + include/sound/hda_codec.h | 11 +- include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 50 + include/sound/soc-acpi.h | 2 + include/sound/soc-dapm.h | 1 + include/uapi/sound/intel/avs/tokens.h | 147 ++ sound/hda/ext/hdac_ext_controller.c | 31 +- sound/pci/hda/hda_codec.c | 93 +- sound/pci/hda/hda_local.h | 2 - sound/soc/codecs/hdac_hda.c | 2 +- sound/soc/intel/Kconfig | 19 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 12 + sound/soc/intel/avs/apl.c | 244 +++ sound/soc/intel/avs/avs.h | 331 ++++ sound/soc/intel/avs/board_selection.c | 459 +++++ sound/soc/intel/avs/cldma.c | 328 ++++ sound/soc/intel/avs/cldma.h | 29 + sound/soc/intel/avs/core.c | 737 +++++++ sound/soc/intel/avs/dsp.c | 326 ++++ sound/soc/intel/avs/ipc.c | 612 ++++++ sound/soc/intel/avs/loader.c | 672 +++++++ sound/soc/intel/avs/messages.c | 674 +++++++ sound/soc/intel/avs/messages.h | 813 ++++++++ sound/soc/intel/avs/path.c | 1287 +++++++++++++ sound/soc/intel/avs/path.h | 85 + sound/soc/intel/avs/pcm.c | 1240 ++++++++++++ sound/soc/intel/avs/registers.h | 83 + sound/soc/intel/avs/skl.c | 127 ++ sound/soc/intel/avs/topology.c | 1700 +++++++++++++++++ sound/soc/intel/avs/topology.h | 207 ++ sound/soc/intel/avs/trace.c | 34 + sound/soc/intel/avs/trace.h | 158 ++ sound/soc/intel/avs/utils.c | 305 +++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 37 files changed, 10824 insertions(+), 40 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 include/uapi/sound/intel/avs/tokens.h create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/apl.c create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/board_selection.c create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/path.c create mode 100644 sound/soc/intel/avs/path.h create mode 100644 sound/soc/intel/avs/pcm.c create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/skl.c create mode 100644 sound/soc/intel/avs/topology.c create mode 100644 sound/soc/intel/avs/topology.h create mode 100644 sound/soc/intel/avs/trace.c create mode 100644 sound/soc/intel/avs/trace.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

4 56

Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which was developed by many developers ( https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 ) and is BSD-3-Clause licensed. I have tested that with this PR, the PinePhone has the expected HiFi and VoiceCall profiles.This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in this PR: https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. See also: * `alsa-info`: https://alsa-project.org/db/?f=5e331f5ef10bd167b3bd7640bfcf0e879100f80d * `pa-info`: [pa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8015083/pa… * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which is BSD-3-Clause licensed. I have tested that with this PR, the PinePhone has the expected HiFi and VoiceCall profiles.This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in this PR: https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. `alsa-info`: https://alsa-project.org/db/?f=5e331f5ef10bd167b3bd7640bfcf0e879100f80d `pa-info`: [pa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8015083/pa… See also: * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which is BSD-3-Clause licensed. I have tested that with this PR, the PinePhone has the expected HiFi and VoiceCall profiles.This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in this PR: https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. See also: * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which is BSD-3-Clause licensed. I have tested that with this PR, the PinePhone has the expected HiFi and VoiceCall profiles.This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. See also: * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which is BSD-3-Clause licensed. This UCM has been tested via Pulseaudio's ALSA_CONFIG_UCM2 environment variable, but I've not yet tested that this UCM, when included in this package, is correctly recognised on a PinePhone as part of this Todo: * [x] Test that this UCM is correctly recognised on a PinePhone. **Done**. This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. See also: * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

WIP: Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which is BSD-3-Clause licensed. This UCM has been tested via Pulseaudio's ALSA_CONFIG_UCM2 environment variable, but I've not yet tested that this UCM, when included in this package, is correctly recognised on a PinePhone as part of this Todo: * [x] Test that this UCM is correctly recognised on a PinePhone. **Done**. This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. See also: * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

WIP: Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which is BSD-3-Clause licensed. This UCM has been tested via Pulseaudio's ALSA_CONFIG_UCM2 environment variable, but I've not yet tested that this UCM, when included in this package, is correctly recognised on a PinePhone as part of this Todo: * [x] Test that this UCM is correctly recognised on a PinePhone. Done. This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. See also: * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

WIP: Add UCM for PinePhone
by GitHub pull_request - edited 07 Feb '22

07 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The DTS configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which is BSD-3-Clause licensed. This UCM has been tested via Pulseaudio's ALSA_CONFIG_UCM2 environment variable, but I've not yet tested that this UCM, when included in this package, is correctly recognised on a PinePhone as part of this Todo: * [x] Test that this UCM is correctly recognised on a PinePhone. See also: * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Alsa applies noise cancelling when not desired
by GitHub issues - opened 04 Feb '22

04 Feb '22

alsa-project/alsa-lib issue #211 was opened from Martmists-GH: In my [original issue](https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1306) on pulseaudio I was told it was an alsa issue. Issue URL : https://github.com/alsa-project/alsa-lib/issues/211 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

`alsatplg` compile error `undefined source widget/stream` and `undefined sink widget/stream`
by GitHub issues - edited 04 Feb '22

04 Feb '22

alsa-project/alsa-utils issue #142 was edited from LazyGeniusMan: Hi, I'm trying to do enable HDA+DMIC in my Kabylake R laptop using a guide[^1]. It's said to compile hda-dsp in [alsa-topology-conf](https://github.com/alsa-project/alsa-topology-conf), but I got this error: ``` [lazygeniusman@archlinux hda-dsp]$ alsatplg -c skl_hda_dsp_generic-tplg.conf -o skl_hda_dsp_generic-tplg.bin ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'hda-dsp-analog-playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'hda-dsp-digital-playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'hda-dsp-alt-analog-playback' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'hda-dsp-audio-analog-capture' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'hda-dsp-audio-digital-capture' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'hda-dsp-audio-alt-analog-capture' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'HDMI1 Playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'HDMI2 Playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'HDMI3 Playback' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'DMIC-Capture' ``` it's also have same error when I run `make firmware` in the repo root dir Could someone help me solve it? EDIT1: error -13 solved but the undefined source and sink error still there [^1]: https://gist.github.com/crojewsk/4e6382bfb0dbfaaf60513174211f29cb#topology Issue URL : https://github.com/alsa-project/alsa-utils/issues/142 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3 1/2] kselftest: alsa: Check for event generation when we write to controls
by Mark Brown 04 Feb '22

04 Feb '22

Add some coverage of event generation to mixer-test. Rather than doing a separate set of writes designed to trigger events we add a step to the existing write_and_verify() which checks to see if the value we read back from non-volatile controls matches the value before writing and that an event is or isn't generated as appropriate. The "tests" for events then simply check that no spurious or missing events were detected. This avoids needing further logic to generate appropriate values for each control type and maximises coverage. When checking for events we use a timeout of 0. This relies on the kernel generating any event prior to returning to userspace when setting a control. That is currently the case and it is difficult to see it changing, if it does the test will need to be updated. Using a delay of 0 means that we don't slow things down unduly when checking for no event or when events fail to be generated. We don't check behaviour for volatile controls since we can't tell what the behaviour is supposed to be for any given control. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Shuah Khan <skhan(a)linuxfoundation.org> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> --- v3: - Add a check for a removal event when monitoring for events, if one is seen then return an error. v2: - Get the numid from the API rather than using the control index. tools/testing/selftests/alsa/mixer-test.c | 154 +++++++++++++++++++++- 1 file changed, 151 insertions(+), 3 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 0e88f4f3d802..6edb7dca32af 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -3,7 +3,7 @@ // kselftest for the ALSA mixer API // // Original author: Mark Brown <broonie(a)kernel.org> -// Copyright (c) 2021 Arm Limited +// Copyright (c) 2021-2 Arm Limited // This test will iterate over all cards detected in the system, exercising // every mixer control it can find. This may conflict with other system @@ -27,11 +27,12 @@ #include "../kselftest.h" -#define TESTS_PER_CONTROL 4 +#define TESTS_PER_CONTROL 6 struct card_data { snd_ctl_t *handle; int card; + struct pollfd pollfd; int num_ctls; snd_ctl_elem_list_t *ctls; struct card_data *next; @@ -43,6 +44,8 @@ struct ctl_data { snd_ctl_elem_info_t *info; snd_ctl_elem_value_t *def_val; int elem; + int event_missing; + int event_spurious; struct card_data *card; struct ctl_data *next; }; @@ -149,6 +152,7 @@ void find_controls(void) if (!ctl_data) ksft_exit_fail_msg("Out of memory\n"); + memset(ctl_data, 0, sizeof(*ctl_data)); ctl_data->card = card_data; ctl_data->elem = ctl; ctl_data->name = snd_ctl_elem_list_get_name(card_data->ctls, @@ -184,6 +188,26 @@ void find_controls(void) ctl_list = ctl_data; } + /* Set up for events */ + err = snd_ctl_subscribe_events(card_data->handle, true); + if (err < 0) { + ksft_exit_fail_msg("snd_ctl_subscribe_events() failed for card %d: %d\n", + card, err); + } + + err = snd_ctl_poll_descriptors_count(card_data->handle); + if (err != 1) { + ksft_exit_fail_msg("Unexpected desciptor count %d for card %d\n", + err, card); + } + + err = snd_ctl_poll_descriptors(card_data->handle, + &card_data->pollfd, 1); + if (err != 1) { + ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n", + card, err); + } + next_card: if (snd_card_next(&card) < 0) { ksft_print_msg("snd_card_next"); @@ -194,6 +218,79 @@ void find_controls(void) snd_config_delete(config); } +/* + * Block for up to timeout ms for an event, returns a negative value + * on error, 0 for no event and 1 for an event. + */ +int wait_for_event(struct ctl_data *ctl, int timeout) +{ + unsigned short revents; + snd_ctl_event_t *event; + int count, err; + unsigned int mask = 0; + unsigned int ev_id; + + snd_ctl_event_alloca(&event); + + do { + err = poll(&(ctl->card->pollfd), 1, timeout); + if (err < 0) { + ksft_print_msg("poll() failed for %s: %s (%d)\n", + ctl->name, strerror(errno), errno); + return -1; + } + /* Timeout */ + if (err == 0) + return 0; + + err = snd_ctl_poll_descriptors_revents(ctl->card->handle, + &(ctl->card->pollfd), + 1, &revents); + if (err < 0) { + ksft_print_msg("snd_ctl_poll_desciptors_revents() failed for %s: %d\n", + ctl->name, err); + return err; + } + if (revents & POLLERR) { + ksft_print_msg("snd_ctl_poll_desciptors_revents() reported POLLERR for %s\n", + ctl->name); + return -1; + } + /* No read events */ + if (!(revents & POLLIN)) { + ksft_print_msg("No POLLIN\n"); + continue; + } + + err = snd_ctl_read(ctl->card->handle, event); + if (err < 0) { + ksft_print_msg("snd_ctl_read() failed for %s: %d\n", + ctl->name, err); + return err; + } + + if (snd_ctl_event_get_type(event) != SND_CTL_EVENT_ELEM) + continue; + + /* The ID returned from the event is 1 less than numid */ + mask = snd_ctl_event_elem_get_mask(event); + ev_id = snd_ctl_event_elem_get_numid(event); + if (ev_id != snd_ctl_elem_info_get_numid(ctl->info)) { + ksft_print_msg("Event for unexpected ctl %s\n", + snd_ctl_event_elem_get_name(event)); + continue; + } + + if ((mask & SND_CTL_EVENT_MASK_REMOVE) == SND_CTL_EVENT_MASK_REMOVE) { + ksft_print_msg("Removal event for %s\n", + ctl->name); + return -1; + } + } while ((mask & SND_CTL_EVENT_MASK_VALUE) != SND_CTL_EVENT_MASK_VALUE); + + return 1; +} + bool ctl_value_index_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val, int index) { @@ -428,7 +525,8 @@ int write_and_verify(struct ctl_data *ctl, { int err, i; bool error_expected, mismatch_shown; - snd_ctl_elem_value_t *read_val, *w_val; + snd_ctl_elem_value_t *initial_val, *read_val, *w_val; + snd_ctl_elem_value_alloca(&initial_val); snd_ctl_elem_value_alloca(&read_val); snd_ctl_elem_value_alloca(&w_val); @@ -446,6 +544,18 @@ int write_and_verify(struct ctl_data *ctl, snd_ctl_elem_value_copy(expected_val, write_val); } + /* Store the value before we write */ + if (snd_ctl_elem_info_is_readable(ctl->info)) { + snd_ctl_elem_value_set_id(initial_val, ctl->id); + + err = snd_ctl_elem_read(ctl->card->handle, initial_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + return err; + } + } + /* * Do the write, if we have an expected value ignore the error * and carry on to validate the expected value. @@ -470,6 +580,30 @@ int write_and_verify(struct ctl_data *ctl, return err; } + /* + * Check for an event if the value changed, or confirm that + * there was none if it didn't. We rely on the kernel + * generating the notification before it returns from the + * write, this is currently true, should that ever change this + * will most likely break and need updating. + */ + if (!snd_ctl_elem_info_is_volatile(ctl->info)) { + err = wait_for_event(ctl, 0); + if (snd_ctl_elem_value_compare(initial_val, read_val)) { + if (err < 1) { + ksft_print_msg("No event generated for %s\n", + ctl->name); + ctl->event_missing++; + } + } else { + if (err != 0) { + ksft_print_msg("Spurious event generated for %s\n", + ctl->name); + ctl->event_spurious++; + } + } + } + /* * Use the libray to compare values, if there's a mismatch * carry on and try to provide a more useful diagnostic than @@ -898,6 +1032,18 @@ void test_ctl_write_invalid(struct ctl_data *ctl) ctl->card->card, ctl->elem); } +void test_ctl_event_missing(struct ctl_data *ctl) +{ + ksft_test_result(!ctl->event_missing, "event_missing.%d.%d\n", + ctl->card->card, ctl->elem); +} + +void test_ctl_event_spurious(struct ctl_data *ctl) +{ + ksft_test_result(!ctl->event_spurious, "event_spurious.%d.%d\n", + ctl->card->card, ctl->elem); +} + int main(void) { struct ctl_data *ctl; @@ -917,6 +1063,8 @@ int main(void) test_ctl_write_default(ctl); test_ctl_write_valid(ctl); test_ctl_write_invalid(ctl); + test_ctl_event_missing(ctl); + test_ctl_event_spurious(ctl); } ksft_exit_pass(); -- 2.30.2

2 3

Re: [syzbot] possible deadlock in snd_timer_interrupt (2)
by syzbot 04 Feb '22

04 Feb '22

syzbot has found a reproducer for the following issue on: HEAD commit: 1f2cfdd349b7 printk: Fix incorrect __user type in proc_doi.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=146fce24700000 kernel config: https://syzkaller.appspot.com/x/.config?x=b4a89edfcc8f7c74 dashboard link: https://syzkaller.appspot.com/bug?extid=1ee0910eca9c94f71f25 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1007e462700000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com ======================================================== WARNING: possible irq lock inversion dependency detected 5.17.0-rc2-syzkaller-00071-g1f2cfdd349b7 #0 Not tainted -------------------------------------------------------- syz-executor.3/4250 just changed the state of lock: ffff88814a62e148 (&timer->lock){..-.}-{2:2}, at: snd_timer_interrupt.part.0+0x34/0xcf0 sound/core/timer.c:856 but this lock took another, SOFTIRQ-READ-unsafe lock in the past: (tasklist_lock){.+.+}-{2:2} and interrupts could create inverse lock ordering between them. other info that might help us debug this: Chain exists of: &timer->lock --> &new->fa_lock --> tasklist_lock Possible interrupt unsafe locking scenario: CPU0 CPU1 ---- ---- lock(tasklist_lock); local_irq_disable(); lock(&timer->lock); lock(&new->fa_lock); <Interrupt> lock(&timer->lock); *** DEADLOCK *** 2 locks held by syz-executor.3/4250: #0: ffff88807efef828 (&mm->mmap_lock#2){++++}-{3:3}, at: __might_fault+0xa1/0x170 mm/memory.c:5271 #1: ffffc90000dc0d70 ((&priv->tlist)){+.-.}-{0:0}, at: lockdep_copy_map include/linux/lockdep.h:35 [inline] #1: ffffc90000dc0d70 ((&priv->tlist)){+.-.}-{0:0}, at: call_timer_fn+0xd5/0x6b0 kernel/time/timer.c:1411 the shortest dependencies between 2nd lock and 1st lock: -> (tasklist_lock){.+.+}-{2:2} { HARDIRQ-ON-R at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 do_wait+0x284/0xce0 kernel/exit.c:1518 kernel_wait+0x9c/0x150 kernel/exit.c:1708 call_usermodehelper_exec_sync kernel/umh.c:139 [inline] call_usermodehelper_exec_work+0xf5/0x180 kernel/umh.c:166 process_one_work+0x9ac/0x1650 kernel/workqueue.c:2307 worker_thread+0x657/0x1110 kernel/workqueue.c:2454 kthread+0x2e9/0x3a0 kernel/kthread.c:377 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:295 SOFTIRQ-ON-R at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 do_wait+0x284/0xce0 kernel/exit.c:1518 kernel_wait+0x9c/0x150 kernel/exit.c:1708 call_usermodehelper_exec_sync kernel/umh.c:139 [inline] call_usermodehelper_exec_work+0xf5/0x180 kernel/umh.c:166 process_one_work+0x9ac/0x1650 kernel/workqueue.c:2307 worker_thread+0x657/0x1110 kernel/workqueue.c:2454 kthread+0x2e9/0x3a0 kernel/kthread.c:377 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:295 INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_write_lock_irq include/linux/rwlock_api_smp.h:195 [inline] _raw_write_lock_irq+0x32/0x50 kernel/locking/spinlock.c:326 copy_process+0x47da/0x7300 kernel/fork.c:2284 kernel_clone+0xe7/0xab0 kernel/fork.c:2555 kernel_thread+0xb5/0xf0 kernel/fork.c:2607 rest_init+0x23/0x3e0 init/main.c:690 start_kernel+0x47a/0x49b init/main.c:1138 secondary_startup_64_no_verify+0xc3/0xcb INITIAL READ USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 do_wait+0x284/0xce0 kernel/exit.c:1518 kernel_wait+0x9c/0x150 kernel/exit.c:1708 call_usermodehelper_exec_sync kernel/umh.c:139 [inline] call_usermodehelper_exec_work+0xf5/0x180 kernel/umh.c:166 process_one_work+0x9ac/0x1650 kernel/workqueue.c:2307 worker_thread+0x657/0x1110 kernel/workqueue.c:2454 kthread+0x2e9/0x3a0 kernel/kthread.c:377 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:295 } ... key at: [<ffffffff8b80a098>] tasklist_lock+0x18/0x40 ... acquired at: __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 send_sigio+0xab/0x380 fs/fcntl.c:810 kill_fasync_rcu fs/fcntl.c:1021 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x1f8/0x470 fs/fcntl.c:1028 sock_wake_async+0xd2/0x160 net/socket.c:1372 sk_wake_async include/net/sock.h:2444 [inline] sk_wake_async include/net/sock.h:2440 [inline] sock_def_readable+0x349/0x4e0 net/core/sock.c:3149 unix_dgram_sendmsg+0xf30/0x1a10 net/unix/af_unix.c:2029 sock_sendmsg_nosec net/socket.c:705 [inline] sock_sendmsg+0xcf/0x120 net/socket.c:725 ____sys_sendmsg+0x331/0x810 net/socket.c:2413 ___sys_sendmsg+0xf3/0x170 net/socket.c:2467 __sys_sendmmsg+0x195/0x470 net/socket.c:2553 __do_sys_sendmmsg net/socket.c:2582 [inline] __se_sys_sendmmsg net/socket.c:2579 [inline] __x64_sys_sendmmsg+0x99/0x100 net/socket.c:2579 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&f->f_owner.lock){....}-{2:2} { INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_write_lock_irq include/linux/rwlock_api_smp.h:195 [inline] _raw_write_lock_irq+0x32/0x50 kernel/locking/spinlock.c:326 f_modown+0x2a/0x390 fs/fcntl.c:91 __f_setown fs/fcntl.c:110 [inline] f_setown+0xd7/0x230 fs/fcntl.c:138 sock_ioctl+0x37e/0x640 net/socket.c:1182 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL READ USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 send_sigio+0x24/0x380 fs/fcntl.c:796 kill_fasync_rcu fs/fcntl.c:1021 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x1f8/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff9057ea40>] __key.5+0x0/0x40 ... acquired at: __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 send_sigio+0x24/0x380 fs/fcntl.c:796 kill_fasync_rcu fs/fcntl.c:1021 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x1f8/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&new->fa_lock){....}-{2:2} { INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_write_lock_irq include/linux/rwlock_api_smp.h:195 [inline] _raw_write_lock_irq+0x32/0x50 kernel/locking/spinlock.c:326 fasync_remove_entry+0xb6/0x1f0 fs/fcntl.c:891 fasync_helper+0x9e/0xb0 fs/fcntl.c:994 __fput+0x846/0x9f0 fs/file_table.c:308 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 tracehook_notify_resume include/linux/tracehook.h:188 [inline] exit_to_user_mode_loop kernel/entry/common.c:175 [inline] exit_to_user_mode_prepare+0x27e/0x290 kernel/entry/common.c:207 __syscall_exit_to_user_mode_work kernel/entry/common.c:289 [inline] syscall_exit_to_user_mode+0x19/0x60 kernel/entry/common.c:300 do_syscall_64+0x42/0xb0 arch/x86/entry/common.c:86 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL READ USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 kill_fasync_rcu fs/fcntl.c:1014 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x136/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff9057f820>] __key.0+0x0/0x40 ... acquired at: __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 kill_fasync_rcu fs/fcntl.c:1014 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x136/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&timer->lock){..-.}-{2:2} { IN-SOFTIRQ-W at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x39/0x50 kernel/locking/spinlock.c:162 snd_timer_interrupt.part.0+0x34/0xcf0 sound/core/timer.c:856 snd_timer_interrupt sound/core/timer.c:1154 [inline] snd_timer_s_function+0x14b/0x200 sound/core/timer.c:1154 call_timer_fn+0x1a5/0x6b0 kernel/time/timer.c:1421 expire_timers kernel/time/timer.c:1466 [inline] __run_timers.part.0+0x67c/0xa30 kernel/time/timer.c:1734 __run_timers kernel/time/timer.c:1715 [inline] run_timer_softirq+0xb3/0x1d0 kernel/time/timer.c:1747 __do_softirq+0x29b/0x9c2 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x123/0x180 kernel/softirq.c:637 irq_exit_rcu+0x5/0x20 kernel/softirq.c:649 sysvec_apic_timer_interrupt+0x93/0xc0 arch/x86/kernel/apic/apic.c:1097 asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 kasan_check_range+0x0/0x180 mm/kasan/generic.c:349 instrument_atomic_read include/linux/instrumented.h:71 [inline] test_bit include/asm-generic/bitops/instrumented-non-atomic.h:134 [inline] cpumask_test_cpu include/linux/cpumask.h:379 [inline] cpu_online include/linux/cpumask.h:921 [inline] trace_lock_release include/trace/events/lock.h:58 [inline] lock_release+0xa1/0x720 kernel/locking/lockdep.c:5650 __might_fault mm/memory.c:5272 [inline] __might_fault+0x142/0x170 mm/memory.c:5257 _copy_from_user+0x27/0x180 lib/usercopy.c:13 copy_from_user include/linux/uaccess.h:192 [inline] snd_seq_oss_write+0x38b/0x780 sound/core/seq/oss/seq_oss_rw.c:93 odev_write+0x55/0x90 sound/core/seq/oss/seq_oss.c:168 vfs_write+0x28e/0xae0 fs/read_write.c:588 ksys_write+0x12d/0x250 fs/read_write.c:643 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x39/0x50 kernel/locking/spinlock.c:162 snd_timer_resolution+0x55/0x100 sound/core/timer.c:489 snd_timer_user_params.isra.0+0x18e/0x8c0 sound/core/timer.c:1851 __snd_timer_user_ioctl.isra.0+0x1020/0x2490 sound/core/timer.c:2100 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff9087d3c0>] __key.12+0x0/0x40 ... acquired at: mark_usage kernel/locking/lockdep.c:4500 [inline] __lock_acquire+0x11d2/0x5470 kernel/locking/lockdep.c:4981 lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x39/0x50 kernel/locking/spinlock.c:162 snd_timer_interrupt.part.0+0x34/0xcf0 sound/core/timer.c:856 snd_timer_interrupt sound/core/timer.c:1154 [inline] snd_timer_s_function+0x14b/0x200 sound/core/timer.c:1154 call_timer_fn+0x1a5/0x6b0 kernel/time/timer.c:1421 expire_timers kernel/time/timer.c:1466 [inline] __run_timers.part.0+0x67c/0xa30 kernel/time/timer.c:1734 __run_timers kernel/time/timer.c:1715 [inline] run_timer_softirq+0xb3/0x1d0 kernel/time/timer.c:1747 __do_softirq+0x29b/0x9c2 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x123/0x180 kernel/softirq.c:637 irq_exit_rcu+0x5/0x20 kernel/softirq.c:649 sysvec_apic_timer_interrupt+0x93/0xc0 arch/x86/kernel/apic/apic.c:1097 asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 kasan_check_range+0x0/0x180 mm/kasan/generic.c:349 instrument_atomic_read include/linux/instrumented.h:71 [inline] test_bit include/asm-generic/bitops/instrumented-non-atomic.h:134 [inline] cpumask_test_cpu include/linux/cpumask.h:379 [inline] cpu_online include/linux/cpumask.h:921 [inline] trace_lock_release include/trace/events/lock.h:58 [inline] lock_release+0xa1/0x720 kernel/locking/lockdep.c:5650 __might_fault mm/memory.c:5272 [inline] __might_fault+0x142/0x170 mm/memory.c:5257 _copy_from_user+0x27/0x180 lib/usercopy.c:13 copy_from_user include/linux/uaccess.h:192 [inline] snd_seq_oss_write+0x38b/0x780 sound/core/seq/oss/seq_oss_rw.c:93 odev_write+0x55/0x90 sound/core/seq/oss/seq_oss.c:168 vfs_write+0x28e/0xae0 fs/read_write.c:588 ksys_write+0x12d/0x250 fs/read_write.c:643 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae stack backtrace: CPU: 1 PID: 4250 Comm: syz-executor.3 Not tainted 5.17.0-rc2-syzkaller-00071-g1f2cfdd349b7 #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106

1 0

[PATCH] ASoC: madera: Add dependencies on MFD
by Charles Keepax 03 Feb '22

03 Feb '22

The Madera CODECs use regmap_irq functions but nothing ensures that regmap_irq is built into the kernel. Add dependencies on the ASoC symbols for the relevant MFD component. There is no point in building the ASoC driver if the MFD doesn't support it and the MFD part contains the necessary dependencies to ensure everything is built into the kernel. Reported-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/Kconfig | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d6b8f5cb6ef8b..68ad04ab3349c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -733,6 +733,7 @@ config SND_SOC_CS4349 config SND_SOC_CS47L15 tristate + depends on MFD_CS47L15 config SND_SOC_CS47L24 tristate @@ -740,15 +741,19 @@ config SND_SOC_CS47L24 config SND_SOC_CS47L35 tristate + depends on MFD_CS47L35 config SND_SOC_CS47L85 tristate + depends on MFD_CS47L85 config SND_SOC_CS47L90 tristate + depends on MFD_CS47L90 config SND_SOC_CS47L92 tristate + depends on MFD_CS47L92 # Cirrus Logic Quad-Channel ADC config SND_SOC_CS53L30 -- 2.11.0

2 1

[PATCH 0/3] ASoC: sun4i-i2s: Support for Allwinner R329 and D1 SoCs
by Samuel Holland 03 Feb '22

03 Feb '22

This series extends the sun4i-i2s binding and driver to support some newer versions of the hardware. Each instance of the hardwar now has multiple input/output pins, and channels can be muxed between them. Since so far the driver only supports a "default" linear channel map, the driver changes are minimal. Samuel Holland (3): ASoC: dt-bindings: sun4i-i2s: Add compatibles for R329 and D1 ASoC: sun4i-i2s: Update registers for more channels ASoC: sun4i-i2s: Add support for the R329/D1 variant .../sound/allwinner,sun4i-a10-i2s.yaml | 5 ++ sound/soc/sunxi/sun4i-i2s.c | 68 +++++++++++++++---- 2 files changed, 59 insertions(+), 14 deletions(-) -- 2.33.1

3 5

[PATCH] ASoC: samsung: Explicitly include gpiolib header
by Mark Brown 03 Feb '22

03 Feb '22

midas_wm811 uses gpiolib but relies on the header being implicitly included which can lead to build failures in some configurations, explicitly pull the header in to avoid problems. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/samsung/midas_wm1811.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index a2019535a0b1..5e9dc18687cc 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -6,6 +6,7 @@ // Copyright (C) 2020 Samsung Electronics Co., Ltd. #include <linux/clk.h> +#include <linux/gpio/consumer.h> #include <linux/mfd/wm8994/registers.h> #include <linux/module.h> #include <linux/of.h> -- 2.30.2

2 2

[PATCH] ASoC: amd: acp: Set gpio_spkr_en to None for max speaker amplifer in machine driver
by V sujith kumar Reddy 03 Feb '22

03 Feb '22

Maxim codec driver already enabling/disabling spk_en_gpio in form of sd_mode gpio hence remove such gpio access control from machine driver to avoid conflict Signed-off-by: V sujith kumar Reddy <vsujithkumar.reddy(a)amd.com> --- sound/soc/amd/acp/acp-mach.h | 1 - sound/soc/amd/acp/acp-sof-mach.c | 4 ++-- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index fd6299844ebe..c855f50d6b34 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -21,7 +21,6 @@ #include <linux/gpio/consumer.h> #define EN_SPKR_GPIO_GB 0x11F -#define EN_SPKR_GPIO_NK 0x146 #define EN_SPKR_GPIO_NONE -EINVAL enum be_id { diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index c1d9650fc222..3346677949e3 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -37,7 +37,7 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, + .gpio_spkr_en = EN_SPKR_GPIO_NONE, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +56,7 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, + .gpio_spkr_en = EN_SPKR_GPIO_NONE, }; static const struct snd_kcontrol_new acp_controls[] = { -- 2.25.1

3 4

[PATCH] ASoC: cs47l15: Ensure regmap_irq is built
by Mark Brown 03 Feb '22

03 Feb '22

The cs47l15 driver uses regmap_irq but does not ensure that it is built, causing build failures if nothing else being build requires the helpers. Add the appropriate select. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2fd21004688c..69fd11b2b046 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -733,6 +733,7 @@ config SND_SOC_CS4349 config SND_SOC_CS47L15 tristate + select REGMAP_IRQ config SND_SOC_CS47L24 tristate -- 2.30.2

2 2

[PATCH] ASoC: wm8962: Allow switching between analog and digital inputs
by Charles Keepax 03 Feb '22

03 Feb '22

When the DMIC_ENA bit is set the analogue inputs are disconnected from the digital core of the chip, in favour of the digital microphones. Currently the driver will always enable DMIC_ENA whilst the GPIOs are configured for the DMIC function, this means the user can't currently use both the analog inputs and the digital inputs in one system. Add an additional DAPM mutex that allows switching between analog and digital inputs into the digital core. Reported-by: Martin Kepplinger <martin.kepplinger(a)puri.sm> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm8962.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a5584ba962dcf..2c41d31956aa8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2049,6 +2049,13 @@ static SOC_ENUM_SINGLE_DECL(hpoutl_enum, static const struct snd_kcontrol_new hpoutl_mux = SOC_DAPM_ENUM("HPOUTL Mux", hpoutl_enum); +static const char * const input_mode_text[] = { "Analog", "Digital" }; + +static SOC_ENUM_SINGLE_VIRT_DECL(input_mode_enum, input_mode_text); + +static const struct snd_kcontrol_new input_mode_mux = + SOC_DAPM_ENUM("Input Mode", input_mode_enum); + static const struct snd_kcontrol_new inpgal[] = { SOC_DAPM_SINGLE("IN1L Switch", WM8962_LEFT_INPUT_PGA_CONTROL, 3, 1, 0), SOC_DAPM_SINGLE("IN2L Switch", WM8962_LEFT_INPUT_PGA_CONTROL, 2, 1, 0), @@ -2147,6 +2154,9 @@ SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0, SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), +SND_SOC_DAPM_MUX("Input Mode L", SND_SOC_NOPM, 0, 0, &input_mode_mux), +SND_SOC_DAPM_MUX("Input Mode R", SND_SOC_NOPM, 0, 0, &input_mode_mux), + SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0), @@ -2226,16 +2236,19 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "DMIC_ENA", NULL, "DMICDAT" }, + { "Input Mode L", "Analog", "MIXINL" }, + { "Input Mode L", "Digital", "DMIC_ENA" }, + { "Input Mode R", "Analog", "MIXINR" }, + { "Input Mode R", "Digital", "DMIC_ENA" }, + { "ADCL", NULL, "SYSCLK" }, { "ADCL", NULL, "TOCLK" }, - { "ADCL", NULL, "MIXINL" }, - { "ADCL", NULL, "DMIC_ENA" }, + { "ADCL", NULL, "Input Mode L" }, { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, - { "ADCR", NULL, "MIXINR" }, - { "ADCR", NULL, "DMIC_ENA" }, + { "ADCR", NULL, "Input Mode R" }, { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, -- 2.11.0

3 2

`alsatplg` compile error `undefined source widget/stream` and `undefined sink widget/stream`
by GitHub issues - opened 03 Feb '22

03 Feb '22

alsa-project/alsa-utils issue #142 was opened from LazyGeniusMan: Hi, I'm trying to do enable HDA+DMIC in my Kabylake R laptop using a guide[^1]. It's said to compile hda-dsp in [alsa-topology-conf](https://github.com/alsa-project/alsa-topology-conf), but I got this error: ``` [lazygeniusman@archlinux hda-dsp]$ alsatplg -c skl_hda_dsp_generic-tplg.conf -o skl_hda_dsp_generic-tplg.bin ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'hda-dsp-analog-playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'hda-dsp-digital-playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'hda-dsp-alt-analog-playback' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'hda-dsp-audio-analog-capture' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'hda-dsp-audio-digital-capture' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'hda-dsp-audio-alt-analog-capture' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'HDMI1 Playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'HDMI2 Playback' ALSA lib dapm.c:247:(tplg_build_routes) undefined source widget/stream 'HDMI3 Playback' ALSA lib dapm.c:225:(tplg_build_routes) undefined sink widget/stream 'DMIC-Capture' Unable to open output file 'skl_hda_dsp_generic-tplg.bin.new': Unknown error -13 ``` it's also have same error when I run `make firmware` in the repo root dir Could someone help me solve it? [^1]: https://gist.github.com/crojewsk/4e6382bfb0dbfaaf60513174211f29cb#topology Issue URL : https://github.com/alsa-project/alsa-utils/issues/142 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[GIT PULL] Immutable branch with spi in i2c-multi-instantiate changes
by Hans de Goede 02 Feb '22

02 Feb '22

Hi Takashi, Rafael, As discussed here is a pull-req for an immutable branch with all the changes from the "[PATCH v6 0/9] Support Spi in i2c-multi-instantiate driver" series, including one ALSA change, in case you want to merge this to avoid conflicts. Regards, Hans The following changes since commit e783362eb54cd99b2cac8b3a9aeac942e6f6ac07: Linux 5.17-rc1 (2022-01-23 10:12:53 +0200) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/pdx86/platform-drivers-x86.git tags/platform-drivers-x86-serial-multi-instantiate-1 for you to fetch changes up to d9c01c530cc5e3b6d5bdfeae12c3d0f33fae7498: ACPI / scan: Create platform device for CS35L41 (2022-02-02 18:12:41 +0100) ---------------------------------------------------------------- This branch contains 5.17-rc1 + the SPI tree's spi-acpi-helpers tag + the other patches from the "[PATCH v6 0/9] Support Spi in i2c-multi-instantiate driver" series. ---------------------------------------------------------------- Lucas Tanure (4): platform/x86: i2c-multi-instantiate: Rename it for a generic serial driver name platform/x86: serial-multi-instantiate: Reorganize I2C functions ALSA: hda/realtek: Add support for HP Laptops ACPI / scan: Create platform device for CS35L41 Stefan Binding (5): spi: Make spi_alloc_device and spi_add_device public again spi: Create helper API to lookup ACPI info for spi device spi: Support selection of the index of the ACPI Spi Resource before alloc spi: Add API to count spi acpi resources platform/x86: serial-multi-instantiate: Add SPI support MAINTAINERS | 4 +- drivers/acpi/scan.c | 16 +- drivers/platform/x86/Kconfig | 12 +- drivers/platform/x86/Makefile | 2 +- drivers/platform/x86/i2c-multi-instantiate.c | 174 ------------ drivers/platform/x86/serial-multi-instantiate.c | 348 ++++++++++++++++++++++++ drivers/spi/spi.c | 137 ++++++++-- include/linux/spi/spi.h | 20 ++ sound/pci/hda/patch_realtek.c | 43 ++- 9 files changed, 550 insertions(+), 206 deletions(-) delete mode 100644 drivers/platform/x86/i2c-multi-instantiate.c create mode 100644 drivers/platform/x86/serial-multi-instantiate.c

1 0

How to add support for A_LAW pcm format to ALSA Loop driver ?
by Christophe Leroy 02 Feb '22

02 Feb '22

Hi, I see that the ALSA Loop driver supports only the following formats: .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE | SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE), I'd like to have Loopback service with A_LAW format, I though it would just require to add SNDRV_PCM_FMTBIT_A_LAW to the format above as the Loopback driver just loops the samples. But it doesn't work, I still get only the following formats: ALSA ERROR hw_params: set (FORMAT) value = A_LAW : Invalid argument ACCESS: RW_INTERLEAVED FORMAT: S16_LE S16_BE S24_LE S32_LE S32_BE S24_3LE SUBFORMAT: STD SAMPLE_BITS: [16 32] FRAME_BITS: [16 1024] CHANNELS: [1 32] RATE: [8000 192000] PERIOD_TIME: (5 65536000] PERIOD_SIZE: [1 524288] PERIOD_BYTES: [2 67108864] PERIODS: [1 1024] BUFFER_TIME: (5 131072000] BUFFER_SIZE: [1 1048576] BUFFER_BYTES: [2 134217728] TICK_TIME: ALL ALSA lib pcm.c:8644:(snd_pcm_set_params) Sample format not available for CAPTURE: Invalid argument What am I missing, what needs to be added to the Loop driver to support A_LAW format ? Thanks for your help Christophe

1 0

[PATCH v6 0/9] Support Spi in i2c-multi-instantiate driver
by Stefan Binding 02 Feb '22

02 Feb '22

Add support for SPI bus in the i2c-multi-instantiate driver as upcoming laptops will need to multi instantiate SPI devices from a single device node, which has multiple SpiSerialBus entries at the ACPI table. With the new SPI support, i2c-multi-instantiate becomes bus-multi-instantiate and is moved to the ACPI folder. The intention is to support the SPI bus by re-using the current I2C multi instantiate, instead of creating a new SPI multi instantiate, to make it possible for peripherals that can be controlled by I2C or SPI to have the same HID at the ACPI table. The new driver (serial multi instantiate, smi) checks for the hard-coded bus type and returns -ENODEV in case of zero devices found for that bus. In the case of automatic bus detection, the driver will give preference to I2C. The expectation is for a device node in the ACPI table to have multiple I2cSerialBus only or multiple SpiSerialBus only, not a mix of both; and for the case where there are both entries in one device node, only the I2C ones would be probed. This new serial multi instantiate will be used in CS35L41 HDA new driver. Changes since V5: - comment, commit message and Kconfig description fixes - minor fixes in serial-multi-instantiate - use lowercase for SSIDs in patch_realtek.c Lucas Tanure (4): platform/x86: i2c-multi-instantiate: Rename it for a generic serial driver name platform/x86: serial-multi-instantiate: Reorganize I2C functions ALSA: hda/realtek: Add support for HP Laptops ACPI / scan: Create platform device for CS35L41 Stefan Binding (5): spi: Make spi_alloc_device and spi_add_device public again spi: Create helper API to lookup ACPI info for spi device spi: Support selection of the index of the ACPI Spi Resource before alloc spi: Add API to count spi acpi resources platform/x86: serial-multi-instantiate: Add SPI support MAINTAINERS | 4 +- drivers/acpi/scan.c | 16 +- drivers/platform/x86/Kconfig | 12 +- drivers/platform/x86/Makefile | 2 +- drivers/platform/x86/i2c-multi-instantiate.c | 174 --------- .../platform/x86/serial-multi-instantiate.c | 349 ++++++++++++++++++ drivers/spi/spi.c | 137 ++++++- include/linux/spi/spi.h | 20 + sound/pci/hda/patch_realtek.c | 43 ++- 9 files changed, 551 insertions(+), 206 deletions(-) delete mode 100644 drivers/platform/x86/i2c-multi-instantiate.c create mode 100644 drivers/platform/x86/serial-multi-instantiate.c -- 2.25.1

5 28

[PATCH v2 1/2] kselftest: alsa: Check for event generation when we write to controls
by Mark Brown 02 Feb '22

02 Feb '22

Add some coverage of event generation to mixer-test. Rather than doing a separate set of writes designed to trigger events we add a step to the existing write_and_verify() which checks to see if the value we read back from non-volatile controls matches the value before writing and that an event is or isn't generated as appropriate. The "tests" for events then simply check that no spurious or missing events were detected. This avoids needing further logic to generate appropriate values for each control type and maximises coverage. When checking for events we use a timeout of 0. This relies on the kernel generating any event prior to returning to userspace when setting a control. That is currently the case and it is difficult to see it changing, if it does the test will need to be updated. Using a delay of 0 means that we don't slow things down unduly when checking for no event or when events fail to be generated. We don't check behaviour for volatile controls since we can't tell what the behaviour is supposed to be for any given control. Signed-off-by: Mark Brown <broonie(a)kernel.org> Reviewed-by: Shuah Khan <skhan(a)linuxfoundation.org> --- v2: - Get the numid from the API rather than using the control index. tools/testing/selftests/alsa/mixer-test.c | 148 +++++++++++++++++++++- 1 file changed, 145 insertions(+), 3 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 0e88f4f3d802..79ffbca429ee 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -3,7 +3,7 @@ // kselftest for the ALSA mixer API // // Original author: Mark Brown <broonie(a)kernel.org> -// Copyright (c) 2021 Arm Limited +// Copyright (c) 2021-2 Arm Limited // This test will iterate over all cards detected in the system, exercising // every mixer control it can find. This may conflict with other system @@ -27,11 +27,12 @@ #include "../kselftest.h" -#define TESTS_PER_CONTROL 4 +#define TESTS_PER_CONTROL 6 struct card_data { snd_ctl_t *handle; int card; + struct pollfd pollfd; int num_ctls; snd_ctl_elem_list_t *ctls; struct card_data *next; @@ -43,6 +44,8 @@ struct ctl_data { snd_ctl_elem_info_t *info; snd_ctl_elem_value_t *def_val; int elem; + int event_missing; + int event_spurious; struct card_data *card; struct ctl_data *next; }; @@ -149,6 +152,7 @@ void find_controls(void) if (!ctl_data) ksft_exit_fail_msg("Out of memory\n"); + memset(ctl_data, 0, sizeof(*ctl_data)); ctl_data->card = card_data; ctl_data->elem = ctl; ctl_data->name = snd_ctl_elem_list_get_name(card_data->ctls, @@ -184,6 +188,26 @@ void find_controls(void) ctl_list = ctl_data; } + /* Set up for events */ + err = snd_ctl_subscribe_events(card_data->handle, true); + if (err < 0) { + ksft_exit_fail_msg("snd_ctl_subscribe_events() failed for card %d: %d\n", + card, err); + } + + err = snd_ctl_poll_descriptors_count(card_data->handle); + if (err != 1) { + ksft_exit_fail_msg("Unexpected desciptor count %d for card %d\n", + err, card); + } + + err = snd_ctl_poll_descriptors(card_data->handle, + &card_data->pollfd, 1); + if (err != 1) { + ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n", + card, err); + } + next_card: if (snd_card_next(&card) < 0) { ksft_print_msg("snd_card_next"); @@ -194,6 +218,73 @@ void find_controls(void) snd_config_delete(config); } +/* + * Block for up to timeout ms for an event, returns a negative value + * on error, 0 for no event and 1 for an event. + */ +int wait_for_event(struct ctl_data *ctl, int timeout) +{ + unsigned short revents; + snd_ctl_event_t *event; + int count, err; + unsigned int mask = 0; + unsigned int ev_id; + + snd_ctl_event_alloca(&event); + + do { + err = poll(&(ctl->card->pollfd), 1, timeout); + if (err < 0) { + ksft_print_msg("poll() failed for %s: %s (%d)\n", + ctl->name, strerror(errno), errno); + return -1; + } + /* Timeout */ + if (err == 0) + return 0; + + err = snd_ctl_poll_descriptors_revents(ctl->card->handle, + &(ctl->card->pollfd), + 1, &revents); + if (err < 0) { + ksft_print_msg("snd_ctl_poll_desciptors_revents() failed for %s: %d\n", + ctl->name, err); + return err; + } + if (revents & POLLERR) { + ksft_print_msg("snd_ctl_poll_desciptors_revents() reported POLLERR for %s\n", + ctl->name); + return -1; + } + /* No read events */ + if (!(revents & POLLIN)) { + ksft_print_msg("No POLLIN\n"); + continue; + } + + err = snd_ctl_read(ctl->card->handle, event); + if (err < 0) { + ksft_print_msg("snd_ctl_read() failed for %s: %d\n", + ctl->name, err); + return err; + } + + if (snd_ctl_event_get_type(event) != SND_CTL_EVENT_ELEM) + continue; + + /* The ID returned from the event is 1 less than numid */ + mask = snd_ctl_event_elem_get_mask(event); + ev_id = snd_ctl_event_elem_get_numid(event); + if (ev_id != snd_ctl_elem_info_get_numid(ctl->info)) { + ksft_print_msg("Event for unexpected ctl %s\n", + snd_ctl_event_elem_get_name(event)); + continue; + } + } while ((mask & SND_CTL_EVENT_MASK_VALUE) != SND_CTL_EVENT_MASK_VALUE); + + return 1; +} + bool ctl_value_index_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val, int index) { @@ -428,7 +519,8 @@ int write_and_verify(struct ctl_data *ctl, { int err, i; bool error_expected, mismatch_shown; - snd_ctl_elem_value_t *read_val, *w_val; + snd_ctl_elem_value_t *initial_val, *read_val, *w_val; + snd_ctl_elem_value_alloca(&initial_val); snd_ctl_elem_value_alloca(&read_val); snd_ctl_elem_value_alloca(&w_val); @@ -446,6 +538,18 @@ int write_and_verify(struct ctl_data *ctl, snd_ctl_elem_value_copy(expected_val, write_val); } + /* Store the value before we write */ + if (snd_ctl_elem_info_is_readable(ctl->info)) { + snd_ctl_elem_value_set_id(initial_val, ctl->id); + + err = snd_ctl_elem_read(ctl->card->handle, initial_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + return err; + } + } + /* * Do the write, if we have an expected value ignore the error * and carry on to validate the expected value. @@ -470,6 +574,30 @@ int write_and_verify(struct ctl_data *ctl, return err; } + /* + * Check for an event if the value changed, or confirm that + * there was none if it didn't. We rely on the kernel + * generating the notification before it returns from the + * write, this is currently true, should that ever change this + * will most likely break and need updating. + */ + if (!snd_ctl_elem_info_is_volatile(ctl->info)) { + err = wait_for_event(ctl, 0); + if (snd_ctl_elem_value_compare(initial_val, read_val)) { + if (err < 1) { + ksft_print_msg("No event generated for %s\n", + ctl->name); + ctl->event_missing++; + } + } else { + if (err != 0) { + ksft_print_msg("Spurious event generated for %s\n", + ctl->name); + ctl->event_spurious++; + } + } + } + /* * Use the libray to compare values, if there's a mismatch * carry on and try to provide a more useful diagnostic than @@ -898,6 +1026,18 @@ void test_ctl_write_invalid(struct ctl_data *ctl) ctl->card->card, ctl->elem); } +void test_ctl_event_missing(struct ctl_data *ctl) +{ + ksft_test_result(!ctl->event_missing, "event_missing.%d.%d\n", + ctl->card->card, ctl->elem); +} + +void test_ctl_event_spurious(struct ctl_data *ctl) +{ + ksft_test_result(!ctl->event_spurious, "event_spurious.%d.%d\n", + ctl->card->card, ctl->elem); +} + int main(void) { struct ctl_data *ctl; @@ -917,6 +1057,8 @@ int main(void) test_ctl_write_default(ctl); test_ctl_write_valid(ctl); test_ctl_write_invalid(ctl); + test_ctl_event_missing(ctl); + test_ctl_event_spurious(ctl); } ksft_exit_pass(); -- 2.30.2

2 3

[PATCH] Revert "ASoC: amd: acp: Power on/off the speaker enable gpio pin based on DAPM callback."
by Curtis Malainey 02 Feb '22

02 Feb '22

This reverts commit 5c5f08f7fc0bee9a1bc3fbdcb7a21cfd0648ab14. This commit hard codes GPIO for max98357a drivers at a board specific level in the machine driver in the kernel itself. When used with a board that properly defines the GPIO pins in ACPI there is a resource contention over the GPIO pin and the card fails to probe. The amplifier driver has handled this pin lookup long before the this change landed and it the pin should continue to be owned by the amplifier as it is specific to that component. As a result this should be reverted. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Rob Barnes <robbarnes(a)google.com> Cc: Eric Peers <epeers(a)google.com> --- sound/soc/amd/acp/Kconfig | 6 +++--- sound/soc/amd/acp/acp-legacy-mach.c | 19 +++---------------- sound/soc/amd/acp/acp-mach-common.c | 25 ------------------------- sound/soc/amd/acp/acp-mach.h | 10 +--------- sound/soc/amd/acp/acp-sof-mach.c | 21 +++------------------ 5 files changed, 10 insertions(+), 71 deletions(-) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index d5838df3064b..154be5e70821 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -32,7 +32,7 @@ config SND_AMD_ASOC_RENOIR config SND_SOC_AMD_MACH_COMMON tristate - depends on X86 && PCI && I2C && GPIOLIB + depends on X86 && PCI && I2C select CLK_FIXED_FCH select SND_SOC_RT5682_I2C select SND_SOC_DMIC @@ -44,14 +44,14 @@ config SND_SOC_AMD_MACH_COMMON config SND_SOC_AMD_LEGACY_MACH tristate "AMD Legacy Machine Driver Support" - depends on X86 && PCI && I2C && GPIOLIB + depends on X86 && PCI && I2C select SND_SOC_AMD_MACH_COMMON help This option enables legacy sound card support for ACP audio. config SND_SOC_AMD_SOF_MACH tristate "AMD SOF Machine Driver Support" - depends on X86 && PCI && I2C && GPIOLIB + depends on X86 && PCI && I2C select SND_SOC_AMD_MACH_COMMON help This option enables SOF sound card support for ACP audio. diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 0ad1cf41b308..de0f8024e2fb 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -27,7 +27,6 @@ static struct acp_card_drvdata rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = NONE, - .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -42,16 +41,15 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", event_spkr_handler), - SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), - SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), }; static int acp_asoc_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; - unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -69,20 +67,9 @@ static int acp_asoc_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; - spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_legacy_dai_links_create(card); - if (gpio_is_valid(spkr_gpio)) { - ret = devm_gpio_request(dev, spkr_gpio, "spkren"); - if (ret) { - dev_err(dev, "(%s) gpio request failed: %d\n", - __func__, ret); - return ret; - } - gpio_direction_output(spkr_gpio, 0); - } - ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index c9caade5cb74..7386e5bb61b5 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -71,31 +71,6 @@ static const struct snd_soc_dapm_route rt5682_map[] = { { "IN1P", NULL, "Headset Mic" }, }; -int event_spkr_handler(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct acp_card_drvdata *drvdata = snd_soc_card_get_drvdata(card); - - if (!gpio_is_valid(drvdata->gpio_spkr_en)) - return 0; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - gpio_set_value(drvdata->gpio_spkr_en, 1); - break; - case SND_SOC_DAPM_PRE_PMD: - gpio_set_value(drvdata->gpio_spkr_en, 0); - break; - default: - dev_warn(card->dev, "%s invalid setting\n", __func__); - break; - } - return 0; -} -EXPORT_SYMBOL_NS_GPL(event_spkr_handler, SND_SOC_AMD_MACH); - /* Define card ops for RT5682 CODEC */ static int acp_card_rt5682_init(struct snd_soc_pcm_runtime *rtd) { diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index fd6299844ebe..5dc47cfbff10 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -17,12 +17,6 @@ #include <linux/input.h> #include <linux/module.h> #include <sound/soc.h> -#include <linux/gpio.h> -#include <linux/gpio/consumer.h> - -#define EN_SPKR_GPIO_GB 0x11F -#define EN_SPKR_GPIO_NK 0x146 -#define EN_SPKR_GPIO_NONE -EINVAL enum be_id { HEADSET_BE_ID = 0, @@ -55,11 +49,9 @@ struct acp_card_drvdata { unsigned int dai_fmt; struct clk *wclk; struct clk *bclk; - unsigned int gpio_spkr_en; }; int acp_sofdsp_dai_links_create(struct snd_soc_card *card); int acp_legacy_dai_links_create(struct snd_soc_card *card); -int event_spkr_handler(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event); + #endif diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index c1d9650fc222..dc2786cc81ef 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -27,7 +27,6 @@ static struct acp_card_drvdata sof_rt5682_rt1019_data = { .hs_codec_id = RT5682, .amp_codec_id = RT1019, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_GB, }; static struct acp_card_drvdata sof_rt5682_max_data = { @@ -37,7 +36,6 @@ static struct acp_card_drvdata sof_rt5682_max_data = { .hs_codec_id = RT5682, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static struct acp_card_drvdata sof_rt5682s_rt1019_data = { @@ -56,7 +54,6 @@ static struct acp_card_drvdata sof_rt5682s_max_data = { .hs_codec_id = RT5682S, .amp_codec_id = MAX98360A, .dmic_codec_id = DMIC, - .gpio_spkr_en = EN_SPKR_GPIO_NK, }; static const struct snd_kcontrol_new acp_controls[] = { @@ -70,16 +67,15 @@ static const struct snd_kcontrol_new acp_controls[] = { static const struct snd_soc_dapm_widget acp_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", event_spkr_handler), - SND_SOC_DAPM_SPK("Left Spk", event_spkr_handler), - SND_SOC_DAPM_SPK("Right Spk", event_spkr_handler), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), }; static int acp_sof_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; struct device *dev = &pdev->dev; - unsigned int spkr_gpio; int ret; if (!pdev->id_entry) @@ -97,20 +93,9 @@ static int acp_sof_probe(struct platform_device *pdev) card->controls = acp_controls; card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; - spkr_gpio = ((struct acp_card_drvdata *)(card->drvdata))->gpio_spkr_en; acp_sofdsp_dai_links_create(card); - if (gpio_is_valid(spkr_gpio)) { - ret = devm_gpio_request(dev, spkr_gpio, "spkren"); - if (ret) { - dev_err(dev, "(%s) gpio request failed: %d\n", - __func__, ret); - return ret; - } - gpio_direction_output(spkr_gpio, 0); - } - ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, -- 2.32.0

4 6

[PATCH] ASoC: tas2770: Insert post reset delay
by Martin Povišer 02 Feb '22

02 Feb '22

Per TAS2770 datasheet there must be a 1 ms delay from reset to first command. So insert delays into the driver where appropriate. Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Martin Povišer <povik+lin(a)protonmail.com> --- sound/soc/codecs/tas2770.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index b52c0861edc8..b397a2122dd4 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -38,10 +38,12 @@ static void tas2770_reset(struct tas2770_priv *tas2770) gpiod_set_value_cansleep(tas2770->reset_gpio, 0); msleep(20); gpiod_set_value_cansleep(tas2770->reset_gpio, 1); + usleep_range(1000, 2000); } snd_soc_component_write(tas2770->component, TAS2770_SW_RST, TAS2770_RST); + usleep_range(1000, 2000); } static int tas2770_set_bias_level(struct snd_soc_component *component, @@ -110,6 +112,7 @@ static int tas2770_codec_resume(struct snd_soc_component *component) if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); } else { ret = snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, @@ -529,8 +532,10 @@ static int tas2770_codec_probe(struct snd_soc_component *component) tas2770->component = component; - if (tas2770->sdz_gpio) + if (tas2770->sdz_gpio) { gpiod_set_value_cansleep(tas2770->sdz_gpio, 1); + usleep_range(1000, 2000); + } tas2770_reset(tas2770); -- 2.33.0

2 1

[PATCH] kselftest: alsa: Check for event generation when we write to controls
by Mark Brown 02 Feb '22

02 Feb '22

Add some coverage of event generation to mixer-test. Rather than doing a separate set of writes designed to trigger events we add a step to the existing write_and_verify() which checks to see if the value we read back from non-volatile controls matches the value before writing and that an event is or isn't generated as appropriate. The "tests" for events then simply check that no spurious or missing events were detected. This avoids needing further logic to generate appropriate values for each control type and maximises coverage. When checking for events we use a timeout of 0. This relies on the kernel generating any event prior to returning to userspace when setting a control. That is currently the case and it is difficult to see it changing, if it does the test will need to be updated. Using a delay of 0 means that we don't slow things down unduly when checking for no event or when events fail to be generated. We don't check behaviour for volatile controls since we can't tell what the behaviour is supposed to be for any given control. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- tools/testing/selftests/alsa/mixer-test.c | 148 +++++++++++++++++++++- 1 file changed, 145 insertions(+), 3 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 0e88f4f3d802..42cf3b724586 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -3,7 +3,7 @@ // kselftest for the ALSA mixer API // // Original author: Mark Brown <broonie(a)kernel.org> -// Copyright (c) 2021 Arm Limited +// Copyright (c) 2021-2 Arm Limited // This test will iterate over all cards detected in the system, exercising // every mixer control it can find. This may conflict with other system @@ -27,11 +27,12 @@ #include "../kselftest.h" -#define TESTS_PER_CONTROL 4 +#define TESTS_PER_CONTROL 6 struct card_data { snd_ctl_t *handle; int card; + struct pollfd pollfd; int num_ctls; snd_ctl_elem_list_t *ctls; struct card_data *next; @@ -43,6 +44,8 @@ struct ctl_data { snd_ctl_elem_info_t *info; snd_ctl_elem_value_t *def_val; int elem; + int event_missing; + int event_spurious; struct card_data *card; struct ctl_data *next; }; @@ -149,6 +152,7 @@ void find_controls(void) if (!ctl_data) ksft_exit_fail_msg("Out of memory\n"); + memset(ctl_data, 0, sizeof(*ctl_data)); ctl_data->card = card_data; ctl_data->elem = ctl; ctl_data->name = snd_ctl_elem_list_get_name(card_data->ctls, @@ -184,6 +188,26 @@ void find_controls(void) ctl_list = ctl_data; } + /* Set up for events */ + err = snd_ctl_subscribe_events(card_data->handle, true); + if (err < 0) { + ksft_exit_fail_msg("snd_ctl_subscribe_events() failed for card %d: %d\n", + card, err); + } + + err = snd_ctl_poll_descriptors_count(card_data->handle); + if (err != 1) { + ksft_exit_fail_msg("Unexpected desciptor count %d for card %d\n", + err, card); + } + + err = snd_ctl_poll_descriptors(card_data->handle, + &card_data->pollfd, 1); + if (err != 1) { + ksft_exit_fail_msg("snd_ctl_poll_descriptors() failed for %d\n", + card, err); + } + next_card: if (snd_card_next(&card) < 0) { ksft_print_msg("snd_card_next"); @@ -194,6 +218,73 @@ void find_controls(void) snd_config_delete(config); } +/* + * Block for up to timeout ms for an event, returns a negative value + * on error, 0 for no event and 1 for an event. + */ +int wait_for_event(struct ctl_data *ctl, int timeout) +{ + unsigned short revents; + snd_ctl_event_t *event; + int count, err; + unsigned int mask = 0; + unsigned int ev_id; + + snd_ctl_event_alloca(&event); + + do { + err = poll(&(ctl->card->pollfd), 1, timeout); + if (err < 0) { + ksft_print_msg("poll() failed for %s: %s (%d)\n", + ctl->name, strerror(errno), errno); + return -1; + } + /* Timeout */ + if (err == 0) + return 0; + + err = snd_ctl_poll_descriptors_revents(ctl->card->handle, + &(ctl->card->pollfd), + 1, &revents); + if (err < 0) { + ksft_print_msg("snd_ctl_poll_desciptors_revents() failed for %s: %d\n", + ctl->name, err); + return err; + } + if (revents & POLLERR) { + ksft_print_msg("snd_ctl_poll_desciptors_revents() reported POLLERR for %s\n", + ctl->name); + return -1; + } + /* No read events */ + if (!(revents & POLLIN)) { + ksft_print_msg("No POLLIN\n"); + continue; + } + + err = snd_ctl_read(ctl->card->handle, event); + if (err < 0) { + ksft_print_msg("snd_ctl_read() failed for %s: %d\n", + ctl->name, err); + return err; + } + + if (snd_ctl_event_get_type(event) != SND_CTL_EVENT_ELEM) + continue; + + /* The ID returned from the event is 1 less than numid */ + mask = snd_ctl_event_elem_get_mask(event); + ev_id = snd_ctl_event_elem_get_numid(event); + if (ev_id != ctl->elem + 1) { + ksft_print_msg("Event for unexpected ctl %s\n", + snd_ctl_event_elem_get_name(event)); + continue; + } + } while ((mask & SND_CTL_EVENT_MASK_VALUE) != SND_CTL_EVENT_MASK_VALUE); + + return 1; +} + bool ctl_value_index_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val, int index) { @@ -428,7 +519,8 @@ int write_and_verify(struct ctl_data *ctl, { int err, i; bool error_expected, mismatch_shown; - snd_ctl_elem_value_t *read_val, *w_val; + snd_ctl_elem_value_t *initial_val, *read_val, *w_val; + snd_ctl_elem_value_alloca(&initial_val); snd_ctl_elem_value_alloca(&read_val); snd_ctl_elem_value_alloca(&w_val); @@ -446,6 +538,18 @@ int write_and_verify(struct ctl_data *ctl, snd_ctl_elem_value_copy(expected_val, write_val); } + /* Store the value before we write */ + if (snd_ctl_elem_info_is_readable(ctl->info)) { + snd_ctl_elem_value_set_id(initial_val, ctl->id); + + err = snd_ctl_elem_read(ctl->card->handle, initial_val); + if (err < 0) { + ksft_print_msg("snd_ctl_elem_read() failed: %s\n", + snd_strerror(err)); + return err; + } + } + /* * Do the write, if we have an expected value ignore the error * and carry on to validate the expected value. @@ -470,6 +574,30 @@ int write_and_verify(struct ctl_data *ctl, return err; } + /* + * Check for an event if the value changed, or confirm that + * there was none if it didn't. We rely on the kernel + * generating the notification before it returns from the + * write, this is currently true, should that ever change this + * will most likely break and need updating. + */ + if (!snd_ctl_elem_info_is_volatile(ctl->info)) { + err = wait_for_event(ctl, 0); + if (snd_ctl_elem_value_compare(initial_val, read_val)) { + if (err < 1) { + ksft_print_msg("No event generated for %s\n", + ctl->name); + ctl->event_missing++; + } + } else { + if (err != 0) { + ksft_print_msg("Spurious event generated for %s\n", + ctl->name); + ctl->event_spurious++; + } + } + } + /* * Use the libray to compare values, if there's a mismatch * carry on and try to provide a more useful diagnostic than @@ -898,6 +1026,18 @@ void test_ctl_write_invalid(struct ctl_data *ctl) ctl->card->card, ctl->elem); } +void test_ctl_event_missing(struct ctl_data *ctl) +{ + ksft_test_result(!ctl->event_missing, "event_missing.%d.%d\n", + ctl->card->card, ctl->elem); +} + +void test_ctl_event_spurious(struct ctl_data *ctl) +{ + ksft_test_result(!ctl->event_spurious, "event_spurious.%d.%d\n", + ctl->card->card, ctl->elem); +} + int main(void) { struct ctl_data *ctl; @@ -917,6 +1057,8 @@ int main(void) test_ctl_write_default(ctl); test_ctl_write_valid(ctl); test_ctl_write_invalid(ctl); + test_ctl_event_missing(ctl); + test_ctl_event_spurious(ctl); } ksft_exit_pass(); -- 2.30.2

3 5

Re: [PATCH] wm8962: add a simple DMIC enable control
by Charles Keepax 02 Feb '22

02 Feb '22

On Wed, Feb 02, 2022 at 11:17:34AM +0100, Martin Kepplinger wrote: > Am Mittwoch, dem 02.02.2022 um 09:53 +0000 schrieb Charles Keepax: > > On Tue, Feb 01, 2022 at 04:01:13PM +0100, Martin Kepplinger wrote: > Do you have a code example from a different codec that has roughly what > is missing here? (the sound subsystem is new to me) Full disclosure this is complete untested, but it should be pretty close. Let me know if it does the trick and I will send a proper patch to the list. I do have a Librem 5 in a draw somewhere so can pull that out if we get really stuck, but that might have to wait until the weekend :-). I don't know if you guys are using the analogue bypass paths around the digital core on the chip. I think those will still work with the mics set to digital, so I have left the routes as is, but that might require some checking at some point. Thanks, Charles diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index a5584ba962dcf..2c41d31956aa8 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2049,6 +2049,13 @@ static SOC_ENUM_SINGLE_DECL(hpoutl_enum, static const struct snd_kcontrol_new hpoutl_mux = SOC_DAPM_ENUM("HPOUTL Mux", hpoutl_enum); +static const char * const input_mode_text[] = { "Analog", "Digital" }; + +static SOC_ENUM_SINGLE_VIRT_DECL(input_mode_enum, input_mode_text); + +static const struct snd_kcontrol_new input_mode_mux = + SOC_DAPM_ENUM("Input Mode", input_mode_enum); + static const struct snd_kcontrol_new inpgal[] = { SOC_DAPM_SINGLE("IN1L Switch", WM8962_LEFT_INPUT_PGA_CONTROL, 3, 1, 0), SOC_DAPM_SINGLE("IN2L Switch", WM8962_LEFT_INPUT_PGA_CONTROL, 2, 1, 0), @@ -2147,6 +2154,9 @@ SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0, SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), +SND_SOC_DAPM_MUX("Input Mode L", SND_SOC_NOPM, 0, 0, &input_mode_mux), +SND_SOC_DAPM_MUX("Input Mode R", SND_SOC_NOPM, 0, 0, &input_mode_mux), + SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0), @@ -2226,16 +2236,19 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "DMIC_ENA", NULL, "DMICDAT" }, + { "Input Mode L", "Analog", "MIXINL" }, + { "Input Mode L", "Digital", "DMIC_ENA" }, + { "Input Mode R", "Analog", "MIXINR" }, + { "Input Mode R", "Digital", "DMIC_ENA" }, + { "ADCL", NULL, "SYSCLK" }, { "ADCL", NULL, "TOCLK" }, - { "ADCL", NULL, "MIXINL" }, - { "ADCL", NULL, "DMIC_ENA" }, + { "ADCL", NULL, "Input Mode L" }, { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, - { "ADCR", NULL, "MIXINR" }, - { "ADCR", NULL, "DMIC_ENA" }, + { "ADCR", NULL, "Input Mode R" }, { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, -- 2.11.0

1 0

Re: [PATCH] wm8962: add a simple DMIC enable control
by Charles Keepax 02 Feb '22

02 Feb '22

On Tue, Feb 01, 2022 at 04:01:13PM +0100, Martin Kepplinger wrote: > In order to use the external mic we need to switch DMIC_ENA off. When you say external mic do you mean an analogue mic connected to the INxy pins on the chip? > I know when DMIC is not used at all, the codec driver does > snd_soc_dapm_nc_pin(dapm, "DMICDAT"); but I'm not sure how I'd create > a control based on that. snd_soc_dapm_nc_pin should be called on DMICDAT if the GPIOs are not configured for use of a DMIC. And if called DMIC_ENA should never be set by the driver, since it can't be used as the pins are not configured to operate as DMICs. > I'm not yet looking into detection - only making it work when selected > manually, via ucm. Although I need detection too later. > > While this works when I set the control in ucm, AFAIK the way I do it > here is not correct though due to dapm. > > I guess this conflicts with the widget: > SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), > > Do you have any advice for me on how to do what I want? > Yes the DMIC enable should be controlled through DAPM when the relevant audio path is enabled. Just to check I understand the problem correctly. You have a system that has both analogue and digital mics connected, and the problem is that DMIC_ENA is then permanently enabled, meaning you can't access the audio from the analogue mics? Assuming I am correct above, looking through the DAPM graph it does look like the DMIC is oddly wired. It does appear to be hard wired into ADCL, which would indeed cause it to be permanently enabled if the pins are configured for DMIC. Assuming there isn't some reason the chip can't switch between digital and analogue modes (I can't see an obvious one in the datasheet), I think really there is a DAPM mux missing here. There should be a mux connecting both MIXINL and DMIC_ENA to ADCL/ADCR, rather than them both being directly connected that would let the user switch between analogue and digital inputs. Thanks, Charles

1 0

[PATCH] kselftest: alsa: Declare most functions static
by Mark Brown 01 Feb '22

01 Feb '22

This program has only one file so most functions can be static. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- This depends on my previously posted patch adding coverage of events: https://lore.kernel.org/all/20220201151551.21497-1-broonie@kernel.org/ tools/testing/selftests/alsa/mixer-test.c | 58 ++++++++++++----------- 1 file changed, 30 insertions(+), 28 deletions(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 42cf3b724586..8faae89b7b48 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -71,7 +71,8 @@ struct ctl_data *ctl_list = NULL; #endif #ifndef LIB_HAS_LOAD_STRING -int snd_config_load_string(snd_config_t **config, const char *s, size_t size) +static int snd_config_load_string(snd_config_t **config, const char *s, + size_t size) { snd_input_t *input; snd_config_t *dst; @@ -99,7 +100,7 @@ int snd_config_load_string(snd_config_t **config, const char *s, size_t size) } #endif -void find_controls(void) +static void find_controls(void) { char name[32]; int card, ctl, err; @@ -222,7 +223,7 @@ void find_controls(void) * Block for up to timeout ms for an event, returns a negative value * on error, 0 for no event and 1 for an event. */ -int wait_for_event(struct ctl_data *ctl, int timeout) +static int wait_for_event(struct ctl_data *ctl, int timeout) { unsigned short revents; snd_ctl_event_t *event; @@ -285,8 +286,9 @@ int wait_for_event(struct ctl_data *ctl, int timeout) return 1; } -bool ctl_value_index_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val, - int index) +static bool ctl_value_index_valid(struct ctl_data *ctl, + snd_ctl_elem_value_t *val, + int index) { long int_val; long long int64_val; @@ -397,7 +399,7 @@ bool ctl_value_index_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val, * Check that the provided value meets the constraints for the * provided control. */ -bool ctl_value_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val) +static bool ctl_value_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val) { int i; bool valid = true; @@ -413,7 +415,7 @@ bool ctl_value_valid(struct ctl_data *ctl, snd_ctl_elem_value_t *val) * Check that we can read the default value and it is valid. Write * tests use the read value to restore the default. */ -void test_ctl_get_value(struct ctl_data *ctl) +static void test_ctl_get_value(struct ctl_data *ctl) { int err; @@ -448,9 +450,9 @@ void test_ctl_get_value(struct ctl_data *ctl) ctl->card->card, ctl->elem); } -bool show_mismatch(struct ctl_data *ctl, int index, - snd_ctl_elem_value_t *read_val, - snd_ctl_elem_value_t *expected_val) +static bool show_mismatch(struct ctl_data *ctl, int index, + snd_ctl_elem_value_t *read_val, + snd_ctl_elem_value_t *expected_val) { long long expected_int, read_int; @@ -513,9 +515,9 @@ bool show_mismatch(struct ctl_data *ctl, int index, * the write to fail, for verifying that invalid writes don't corrupt * anything. */ -int write_and_verify(struct ctl_data *ctl, - snd_ctl_elem_value_t *write_val, - snd_ctl_elem_value_t *expected_val) +static int write_and_verify(struct ctl_data *ctl, + snd_ctl_elem_value_t *write_val, + snd_ctl_elem_value_t *expected_val) { int err, i; bool error_expected, mismatch_shown; @@ -622,7 +624,7 @@ int write_and_verify(struct ctl_data *ctl, * Make sure we can write the default value back to the control, this * should validate that at least some write works. */ -void test_ctl_write_default(struct ctl_data *ctl) +static void test_ctl_write_default(struct ctl_data *ctl) { int err; @@ -655,7 +657,7 @@ void test_ctl_write_default(struct ctl_data *ctl) ctl->card->card, ctl->elem); } -bool test_ctl_write_valid_boolean(struct ctl_data *ctl) +static bool test_ctl_write_valid_boolean(struct ctl_data *ctl) { int err, i, j; bool fail = false; @@ -676,7 +678,7 @@ bool test_ctl_write_valid_boolean(struct ctl_data *ctl) return !fail; } -bool test_ctl_write_valid_integer(struct ctl_data *ctl) +static bool test_ctl_write_valid_integer(struct ctl_data *ctl) { int err; int i; @@ -706,7 +708,7 @@ bool test_ctl_write_valid_integer(struct ctl_data *ctl) return !fail; } -bool test_ctl_write_valid_integer64(struct ctl_data *ctl) +static bool test_ctl_write_valid_integer64(struct ctl_data *ctl) { int err, i; long long j, step; @@ -734,7 +736,7 @@ bool test_ctl_write_valid_integer64(struct ctl_data *ctl) return !fail; } -bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) +static bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) { int err, i, j; bool fail = false; @@ -755,7 +757,7 @@ bool test_ctl_write_valid_enumerated(struct ctl_data *ctl) return !fail; } -void test_ctl_write_valid(struct ctl_data *ctl) +static void test_ctl_write_valid(struct ctl_data *ctl) { bool pass; int err; @@ -808,8 +810,8 @@ void test_ctl_write_valid(struct ctl_data *ctl) ctl->card->card, ctl->elem); } -bool test_ctl_write_invalid_value(struct ctl_data *ctl, - snd_ctl_elem_value_t *val) +static bool test_ctl_write_invalid_value(struct ctl_data *ctl, + snd_ctl_elem_value_t *val) { int err; long val_read; @@ -830,7 +832,7 @@ bool test_ctl_write_invalid_value(struct ctl_data *ctl, return !ctl_value_valid(ctl, val); } -bool test_ctl_write_invalid_boolean(struct ctl_data *ctl) +static bool test_ctl_write_invalid_boolean(struct ctl_data *ctl) { int err, i; long val_read; @@ -849,7 +851,7 @@ bool test_ctl_write_invalid_boolean(struct ctl_data *ctl) return !fail; } -bool test_ctl_write_invalid_integer(struct ctl_data *ctl) +static bool test_ctl_write_invalid_integer(struct ctl_data *ctl) { int i; bool fail = false; @@ -895,7 +897,7 @@ bool test_ctl_write_invalid_integer(struct ctl_data *ctl) return !fail; } -bool test_ctl_write_invalid_integer64(struct ctl_data *ctl) +static bool test_ctl_write_invalid_integer64(struct ctl_data *ctl) { int i; bool fail = false; @@ -941,7 +943,7 @@ bool test_ctl_write_invalid_integer64(struct ctl_data *ctl) return !fail; } -bool test_ctl_write_invalid_enumerated(struct ctl_data *ctl) +static bool test_ctl_write_invalid_enumerated(struct ctl_data *ctl) { int err, i; unsigned int val_read; @@ -973,7 +975,7 @@ bool test_ctl_write_invalid_enumerated(struct ctl_data *ctl) } -void test_ctl_write_invalid(struct ctl_data *ctl) +static void test_ctl_write_invalid(struct ctl_data *ctl) { bool pass; int err; @@ -1026,13 +1028,13 @@ void test_ctl_write_invalid(struct ctl_data *ctl) ctl->card->card, ctl->elem); } -void test_ctl_event_missing(struct ctl_data *ctl) +static void test_ctl_event_missing(struct ctl_data *ctl) { ksft_test_result(!ctl->event_missing, "event_missing.%d.%d\n", ctl->card->card, ctl->elem); } -void test_ctl_event_spurious(struct ctl_data *ctl) +static void test_ctl_event_spurious(struct ctl_data *ctl) { ksft_test_result(!ctl->event_spurious, "event_spurious.%d.%d\n", ctl->card->card, ctl->elem); -- 2.30.2

2 1

[GIT PULL] ASoC fixes for v5.17-rc2
by Mark Brown 01 Feb '22

01 Feb '22

The following changes since commit f517ba4924ad026f2583553db02f3c8bc69de88b: ASoC: cs35l41: Add support for hibernate memory retention mode (2022-01-07 17:14:27 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.17-rc2 for you to fetch changes up to a4f399a1416f645ac701064a55b0cb5203707ac9: Input: wm97xx: Simplify resource management (2022-01-31 13:17:24 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.17 Quite a few fixes here, including an unusually large set in the core spurred on by various testing efforts as well as the usual small driver fixes. There are quite a few fixes for out of bounds writes in both the core and the various Qualcomm drivers, plus a couple of fixes for locking in the DPCM code. ---------------------------------------------------------------- Ajit Kumar Pandey (1): ASoC: amd: acp-mach: Fix Left and Right rt1019 amp devices Christophe JAILLET (1): Input: wm97xx: Simplify resource management Dan Carpenter (1): ASoC: max9759: fix underflow in speaker_gain_control_put() Dmitry Osipenko (1): ASoC: hdmi-codec: Fix OOB memory accesses Jiasheng Jiang (1): ASoC: cpcap: Check for NULL pointer after calling of_get_child_by_name Julian Braha (1): ASoC: mediatek: fix unmet dependency on GPIOLIB for SND_SOC_DMIC Mark Brown (5): ASoC: ops: Reject out of bounds values in snd_soc_put_volsw() ASoC: ops: Reject out of bounds values in snd_soc_put_volsw_sx() ASoC: ops: Reject out of bounds values in snd_soc_put_xr_sx() ASoC DPCM lockdep fixes ASoC: ops: Check for negative values before reading them Miaoqian Lin (1): ASoC: fsl: Add missing error handling in pcm030_fabric_probe Peter Ujfalusi (1): ASoC: rt5682: Fix deadlock on resume Robert Hancock (2): ASoC: xilinx: xlnx_formatter_pcm: Make buffer bytes multiple of period bytes ASoC: simple-card: fix probe failure on platform component Srinivas Kandagatla (4): ASoC: codecs: wcd938x: fix incorrect used of portid ASoC: codecs: lpass-rx-macro: fix sidetone register offsets ASoC: codecs: wcd938x: fix return value of mixer put function ASoC: qdsp6: q6apm-dai: only stop graphs that are started Takashi Iwai (2): ASoC: soc-pcm: Fix DPCM lockdep warning due to nested stream locks ASoC: soc-pcm: Move debugfs removal out of spinlock drivers/input/touchscreen/wm97xx-core.c | 12 +++--------- include/sound/pcm.h | 15 +++++++++++++++ include/uapi/sound/asound.h | 4 +++- sound/core/pcm_native.c | 13 +++++++++++++ sound/soc/amd/acp/acp-mach-common.c | 4 ++-- sound/soc/codecs/cpcap.c | 2 ++ sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 8 ++++---- sound/soc/codecs/max9759.c | 3 ++- sound/soc/codecs/rt5682-i2c.c | 15 ++++----------- sound/soc/codecs/rt5682.c | 24 ++++++++---------------- sound/soc/codecs/rt5682.h | 2 -- sound/soc/codecs/wcd938x.c | 31 +++++++++++++++++-------------- sound/soc/fsl/pcm030-audio-fabric.c | 11 ++++++++--- sound/soc/generic/simple-card.c | 26 +++++++++++++++++++++++++- sound/soc/mediatek/Kconfig | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 7 +++++-- sound/soc/soc-ops.c | 29 ++++++++++++++++++++++++++--- sound/soc/soc-pcm.c | 18 ++++++++++++------ sound/soc/xilinx/xlnx_formatter_pcm.c | 27 ++++++++++++++++++++++++--- 20 files changed, 175 insertions(+), 80 deletions(-)

2 1

[PATCH] ALSA: hda: Skip codec shutdown in case the codec is not registered
by Peter Ujfalusi 01 Feb '22

01 Feb '22

If the codec->registered is not set then it means that pm_runtime is not yet enabled and the codec->pcm_list_head has not been initialized. The access to the not initialized pcm_list_head will lead a kernel crash during shutdown. Reported-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Tested-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7016b48227bf..f552785d301e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3000,6 +3000,10 @@ void snd_hda_codec_shutdown(struct hda_codec *codec) { struct hda_pcm *cpcm; + /* Skip the shutdown if codec is not registered */ + if (!codec->registered) + return; + list_for_each_entry(cpcm, &codec->pcm_list_head, list) snd_pcm_suspend_all(cpcm->pcm); -- 2.35.0

2 1

No speaker output ALC294 Asus Zephyrus S17 GX701LWS
by kernel＠killick.email 01 Feb '22

01 Feb '22

I am having no speaker output on the attached device. Headphones work through jack, but just not the internal speakers. Attached alsa-info. If you need any further information please let me know.

1 0

[PATCH] ALSA: usb-audio: Correct quirk for VF0770
by Jonas Hahnfeld 01 Feb '22

01 Feb '22

This device provides both audio and video. The original quirk added in commit 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770") used USB_DEVICE to match the vendor and product ID. Depending on module order, if snd-usb-audio was asked first, it would match the entire device and uvcvideo wouldn't get to see it. Change the matching to USB_AUDIO_DEVICE to restore uvcvideo matching in all cases. Fixes: 48827e1d6af5 ("ALSA: usb-audio: Add quirk for VF0770") Reported-by: Jukka Heikintalo <heikintalo.jukka(a)gmail.com> Tested-by: Jukka Heikintalo <heikintalo.jukka(a)gmail.com> Reported-by: Paweł Susicki <pawel.susicki(a)gmail.com> Tested-by: Paweł Susicki <pawel.susicki(a)gmail.com> Cc: <stable(a)vger.kernel.org> # 5.4, 5.10, 5.14, 5.15 Signed-off-by: Jonas Hahnfeld <hahnjo(a)hahnjo.de> --- sound/usb/quirks-table.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b1522e43173e..0ea39565e623 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -84,7 +84,7 @@ * combination. */ { - USB_DEVICE(0x041e, 0x4095), + USB_AUDIO_DEVICE(0x041e, 0x4095), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, -- 2.35.1

2 1

[PATCH] sound: Replace acpi_bus_get_device()
by Rafael J. Wysocki 01 Feb '22

01 Feb '22

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Replace acpi_bus_get_device() that is going to be dropped with acpi_fetch_acpi_dev(). No intentional functional impact. Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- sound/hda/intel-sdw-acpi.c | 7 +++---- sound/soc/soc-acpi.c | 7 ++----- 2 files changed, 5 insertions(+), 9 deletions(-) Index: linux-pm/sound/hda/intel-sdw-acpi.c =================================================================== --- linux-pm.orig/sound/hda/intel-sdw-acpi.c +++ linux-pm/sound/hda/intel-sdw-acpi.c @@ -50,11 +50,11 @@ static bool is_link_enabled(struct fwnod static int sdw_intel_scan_controller(struct sdw_intel_acpi_info *info) { - struct acpi_device *adev; + struct acpi_device *adev = acpi_fetch_acpi_dev(info->handle); int ret, i; u8 count; - if (acpi_bus_get_device(info->handle, &adev)) + if (!adev) return -EINVAL; /* Found controller, find links supported */ @@ -119,7 +119,6 @@ static acpi_status sdw_intel_acpi_cb(acp void *cdata, void **return_value) { struct sdw_intel_acpi_info *info = cdata; - struct acpi_device *adev; acpi_status status; u64 adr; @@ -127,7 +126,7 @@ static acpi_status sdw_intel_acpi_cb(acp if (ACPI_FAILURE(status)) return AE_OK; /* keep going */ - if (acpi_bus_get_device(handle, &adev)) { + if (!acpi_fetch_acpi_dev(handle)) { pr_err("%s: Couldn't find ACPI handle\n", __func__); return AE_NOT_FOUND; } Index: linux-pm/sound/soc/soc-acpi.c =================================================================== --- linux-pm.orig/sound/soc/soc-acpi.c +++ linux-pm/sound/soc/soc-acpi.c @@ -55,16 +55,13 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_find_mach static acpi_status snd_soc_acpi_find_package(acpi_handle handle, u32 level, void *context, void **ret) { - struct acpi_device *adev; + struct acpi_device *adev = acpi_fetch_acpi_dev(handle); acpi_status status; struct snd_soc_acpi_package_context *pkg_ctx = context; pkg_ctx->data_valid = false; - if (acpi_bus_get_device(handle, &adev)) - return AE_OK; - - if (adev->status.present && adev->status.functional) { + if (adev && adev->status.present && adev->status.functional) { struct acpi_buffer buffer = {ACPI_ALLOCATE_BUFFER, NULL}; union acpi_object *myobj = NULL;

2 1

[PATCH] conf: fix memory leaks in parse_def()
by Sebastian Berger 01 Feb '22

01 Feb '22

For all execution paths in parse_def(), free the id string before returning. Previous implementations fail to do this if the configuration: (1) tries to reference the child of a non-compound node, or (2) does not provide a valid argument after an assignment ('=') operator. For example, the invocations: (1) snd_config_load_string(&conf, "foo 0 foo.a 1", 0) (2) snd_config_load_string(&conf, "bar =", 0) would leak the strings "foo" or "bar", respectively. Signed-off-by: Sebastian Berger <sebastian.berger(a)mailbox.org> --- src/conf.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/src/conf.c b/src/conf.c index 70f0e773..8a09505b 100644 --- a/src/conf.c +++ b/src/conf.c @@ -1403,7 +1403,8 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid if (mode != OVERRIDE) { if (n->type != SND_CONFIG_TYPE_COMPOUND) { SNDERR("%s is not a compound", id); - return -EINVAL; + err = -EINVAL; + goto __end; } n->u.compound.join = true; parent = n; @@ -1425,8 +1426,10 @@ static int parse_def(snd_config_t *parent, input_t *input, int skip, int overrid } if (c == '=') { c = get_nonwhite(input); - if (c < 0) - return c; + if (c < 0) { + err = c; + goto __end; + } } if (!skip) { if (_snd_config_search(parent, id, -1, &n) == 0) { -- 2.35.1

1 0

[PATCH v6 0/1] firmware: mtk: add adsp ipc protocol for sof
by allen-kh.cheng 31 Jan '22

31 Jan '22

From: Allen-KH Cheng <Allen-KH.Cheng(a)mediatek.com> This patch provides mtk adsp ipc support for sof. ADSP IPC protocol offers (send/recv) interfaces using mediatek-mailbox APIs. This was tested and confirmed on MT8195 cherry board with sof. Based on linux-next changes since v5: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v4: - add error message for wrong mbox chan changes since v3: - rebase on v5.16-rc8 - update reviewers changes since v2: - add out tag for two memory free phases changes since v1: - add comments for mtk_adsp_ipc_send and mtk_adsp_ipc_recv - remove useless MODULE_LICENSE - change label name to out_free Allen-KH Cheng (1): firmware: mediatek: add adsp ipc protocol interface drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/mediatek/Kconfig | 9 + drivers/firmware/mediatek/Makefile | 2 + drivers/firmware/mediatek/mtk-adsp-ipc.c | 161 ++++++++++++++++++ .../linux/firmware/mediatek/mtk-adsp-ipc.h | 65 +++++++ 6 files changed, 239 insertions(+) create mode 100644 drivers/firmware/mediatek/Kconfig create mode 100644 drivers/firmware/mediatek/Makefile create mode 100644 drivers/firmware/mediatek/mtk-adsp-ipc.c create mode 100644 include/linux/firmware/mediatek/mtk-adsp-ipc.h -- 2.18.0

2 2

[PATCH] ASoC: max98927: add missing header file
by Randy Dunlap 31 Jan '22

31 Jan '22

Add a header file that provides the missing function prototypes and macro to fix these build errors (seen on arch/alpha/): ../sound/soc/codecs/max98927.c: In function 'max98927_i2c_probe': ../sound/soc/codecs/max98927.c:902:19: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_regulator_get_optional'? [-Werror=implicit-function-declaration] 902 | = devm_gpiod_get_optional(&i2c->dev, "reset", GPIOD_OUT_HIGH); | ^~~~~~~~~~~~~~~~~~~~~~~ ../sound/soc/codecs/max98927.c:902:63: error: 'GPIOD_OUT_HIGH' undeclared (first use in this function); did you mean 'GPIOF_INIT_HIGH'? 902 | = devm_gpiod_get_optional(&i2c->dev, "reset", GPIOD_OUT_HIGH); | ^~~~~~~~~~~~~~ ../sound/soc/codecs/max98927.c:909:17: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 909 | gpiod_set_value_cansleep(max98927->reset_gpio, 0); | ^~~~~~~~~~~~~~~~~~~~~~~~ Fixes: 4d67dc1998f1 ("ASoC: max98927: Handle reset gpio when probing i2c") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Alejandro Tafalla <atafalla(a)dnyon.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/max98927.c | 1 + 1 file changed, 1 insertion(+) --- linux-next-20220128.orig/sound/soc/codecs/max98927.c +++ linux-next-20220128/sound/soc/codecs/max98927.c @@ -16,6 +16,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/of_gpio.h> #include <sound/tlv.h> #include "max98927.h"

2 1

[PATCH] ALSA: hda/realtek: Add quirk for ASUS GU603
by Albert Geanta 31 Jan '22

31 Jan '22

The ASUS GU603 (Zephyrus M16 - SSID 1043:16b2) requires a quirk similar to other ASUS devices for correctly routing the 4 integrated speakers. This fixes it by adding a corresponding quirk entry, which connects the bass speakers to the proper DAC. Signed-off-by: Albert Geantă <albertgeanta(a)gmail.com> --- 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 668274e52674..61b314c7dbc9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8969,6 +8969,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.35.1

3 2

[PATCH 0/3] hda/realtek: Add/Improve several Gigabyte X570(S) audio quirks
by Christian Lachner 31 Jan '22

31 Jan '22

This patch series cleans up and improves my previous work for the Gigabyte X570 audio quirks and adds more mainboards to make use of it. Christian Lachner (3): hda/realtek: Add missing fixup-model for Gigabyte X570 ALC1220 quirks hda/realtek: Fix silent output on Gigabyte X570S Aorus Master (newer chipset) hda/realtek: Fix silent output on Gigabyte X570 Aorus Xtreme after reboot from Windows sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) -- 2.34.1

2 4

[PATCH] ALSA: hda: realtek: Fix race at concurrent COEF updates
by Takashi Iwai 31 Jan '22

31 Jan '22

The COEF access is done with two steps: setting the index then read or write the data. When multiple COEF accesses are performed concurrently, the index and data might be paired unexpectedly. In most cases, this isn't a big problem as the COEF setup is done at the initialization, but some dynamic changes like the mute LED may hit such a race. For avoiding the racy COEF accesses, this patch introduces a new mutex coef_mutex to alc_spec, and wrap the COEF accessing functions with it. Reported-by: Alexander Sergeyev <sergeev917(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20220111195229.a77wrpjclqwrx4bx@localhost.localdo… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 61 ++++++++++++++++++++++++++++------- 1 file changed, 50 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 668274e52674..a5677be0a405 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -98,6 +98,7 @@ struct alc_spec { unsigned int gpio_mic_led_mask; struct alc_coef_led mute_led_coef; struct alc_coef_led mic_led_coef; + struct mutex coef_mutex; hda_nid_t headset_mic_pin; hda_nid_t headphone_mic_pin; @@ -137,8 +138,8 @@ struct alc_spec { * COEF access helper functions */ -static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, - unsigned int coef_idx) +static int __alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx) { unsigned int val; @@ -147,28 +148,61 @@ static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, return val; } +static int alc_read_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx) +{ + struct alc_spec *spec = codec->spec; + unsigned int val; + + mutex_lock(&spec->coef_mutex); + val = __alc_read_coefex_idx(codec, nid, coef_idx); + mutex_unlock(&spec->coef_mutex); + return val; +} + #define alc_read_coef_idx(codec, coef_idx) \ alc_read_coefex_idx(codec, 0x20, coef_idx) -static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, - unsigned int coef_idx, unsigned int coef_val) +static void __alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_val) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_COEF_INDEX, coef_idx); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PROC_COEF, coef_val); } +static void alc_write_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int coef_val) +{ + struct alc_spec *spec = codec->spec; + + mutex_lock(&spec->coef_mutex); + __alc_write_coefex_idx(codec, nid, coef_idx, coef_val); + mutex_unlock(&spec->coef_mutex); +} + #define alc_write_coef_idx(codec, coef_idx, coef_val) \ alc_write_coefex_idx(codec, 0x20, coef_idx, coef_val) +static void __alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid, + unsigned int coef_idx, unsigned int mask, + unsigned int bits_set) +{ + unsigned int val = __alc_read_coefex_idx(codec, nid, coef_idx); + + if (val != -1) + __alc_write_coefex_idx(codec, nid, coef_idx, + (val & ~mask) | bits_set); +} + static void alc_update_coefex_idx(struct hda_codec *codec, hda_nid_t nid, unsigned int coef_idx, unsigned int mask, unsigned int bits_set) { - unsigned int val = alc_read_coefex_idx(codec, nid, coef_idx); + struct alc_spec *spec = codec->spec; - if (val != -1) - alc_write_coefex_idx(codec, nid, coef_idx, - (val & ~mask) | bits_set); + mutex_lock(&spec->coef_mutex); + __alc_update_coefex_idx(codec, nid, coef_idx, mask, bits_set); + mutex_unlock(&spec->coef_mutex); } #define alc_update_coef_idx(codec, coef_idx, mask, bits_set) \ @@ -201,13 +235,17 @@ struct coef_fw { static void alc_process_coef_fw(struct hda_codec *codec, const struct coef_fw *fw) { + struct alc_spec *spec = codec->spec; + + mutex_lock(&spec->coef_mutex); for (; fw->nid; fw++) { if (fw->mask == (unsigned short)-1) - alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val); + __alc_write_coefex_idx(codec, fw->nid, fw->idx, fw->val); else - alc_update_coefex_idx(codec, fw->nid, fw->idx, - fw->mask, fw->val); + __alc_update_coefex_idx(codec, fw->nid, fw->idx, + fw->mask, fw->val); } + mutex_unlock(&spec->coef_mutex); } /* @@ -1153,6 +1191,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) codec->spdif_status_reset = 1; codec->forced_resume = 1; codec->patch_ops = alc_patch_ops; + mutex_init(&spec->coef_mutex); err = alc_codec_rename_from_preset(codec); if (err < 0) { -- 2.31.1

1 0

[BUG] ALSA: core: possible deadlock involving waiting and locking operations
by Jia-Ju Bai 29 Jan '22

29 Jan '22

Hello, My static analysis tool reports a possible deadlock in the sound driver in Linux 5.10: snd_card_disconnect_sync() spin_lock_irq(&card->files_lock); --> Line 461 (Lock A) wait_event_lock_irq(card->remove_sleep, ...); --> Line 462 (Wait X) spin_unlock_irq(&card->files_lock); --> Line 465 (Unlock A) snd_hwdep_release() mutex_lock(&hw->open_mutex); --> Line 152 (Lock B) mutex_unlock(&hw->open_mutex); --> Line 157 (Unlock B) snd_card_file_remove() wake_up_all(&card->remove_sleep); --> Line 976 (Wake X) snd_hwdep_open() mutex_lock(&hw->open_mutex); --> Line 95 (Lock B) snd_card_file_add() spin_lock(&card->files_lock); --> Line 932 (Lock A) spin_unlock(&card->files_lock); --> Line 940 (Unlock A) mutex_unlock(&hw->open_mutex); --> Line 139 (Unlock B) When snd_card_disconnect_sync() is executed, "Wait X" is performed by holding "Lock A". If snd_hwdep_open() is executed at this time, it holds "Lock B" and then waits for acquiring "Lock A". If snd_hwdep_release() is executed at this time, it waits for acquiring "Lock B", and thus "Wake X" cannot be performed to wake up "Wait X" in snd_card_disconnect_sync(), causing a possible deadlock. I am not quite sure whether this possible problem is real and how to fix it if it is real. Any feedback would be appreciated, thanks :) Best wishes, Jia-Ju Bai

3 4

[PATCH v2] ASoC: cs42l51: Improve error handling in cs42l51_remove()
by Uwe Kleine-König 29 Jan '22

29 Jan '22

When disabling a regulator fails while the device goes away, there is little we can do and the machine is probably in enough trouble that any action we'd want to take fails anyhow. The return value used to be passed on in cs42l51_i2c_remove() (i.e. the i2c device remove callback). But the i2c core ignores the error code (apart from emitting a generic warning) and removes the device anyhow. So return 0 unconditionally in cs42l51_i2c_remove(), and instead of returning the error code to the upper layer emit a more helpful warning message. After that nobody is interested any more in the actual error code, so let cs42l51_remove() return void. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, compared to v1 (Message-Id: 20211021103627.70975-1-u.kleine-koenig(a)pengutronix.de) I'm using regulator_bulk_disable() instead of regulator_bulk_force_disable(). Best regards Uwe sound/soc/codecs/cs42l51-i2c.c | 4 +++- sound/soc/codecs/cs42l51.c | 11 ++++++++--- sound/soc/codecs/cs42l51.h | 2 +- 3 files changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index 70260e0a8f09..3cb21a2ba29f 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -31,7 +31,9 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c, static int cs42l51_i2c_remove(struct i2c_client *i2c) { - return cs42l51_remove(&i2c->dev); + cs42l51_remove(&i2c->dev); + + return 0; } static const struct dev_pm_ops cs42l51_pm_ops = { diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index c61b17dc2af8..e9c3cb4e2bfc 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -793,14 +793,19 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) } EXPORT_SYMBOL_GPL(cs42l51_probe); -int cs42l51_remove(struct device *dev) +void cs42l51_remove(struct device *dev) { struct cs42l51_private *cs42l51 = dev_get_drvdata(dev); + int ret; gpiod_set_value_cansleep(cs42l51->reset_gpio, 1); - return regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), - cs42l51->supplies); + ret = regulator_bulk_disable(ARRAY_SIZE(cs42l51->supplies), + cs42l51->supplies); + if (ret) + dev_warn(dev, "Failed to disable all regulators (%pe)\n", + ERR_PTR(ret)); + } EXPORT_SYMBOL_GPL(cs42l51_remove); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 9d06cf7f8876..a79343e8a54e 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -13,7 +13,7 @@ struct device; extern const struct regmap_config cs42l51_regmap; int cs42l51_probe(struct device *dev, struct regmap *regmap); -int cs42l51_remove(struct device *dev); +void cs42l51_remove(struct device *dev); int __maybe_unused cs42l51_suspend(struct device *dev); int __maybe_unused cs42l51_resume(struct device *dev); extern const struct of_device_id cs42l51_of_match[]; base-commit: df0cc57e057f18e44dac8e6c18aba47ab53202f9 -- 2.34.1

2 1

[PATCH 0/3] SoC: SOF: ipc: Optimizations for tx message
by Peter Ujfalusi 29 Jan '22

29 Jan '22

Hi, The series will drop the internal use of 'header' parameter which is always set to hdr->cmd. The other simplification is to use the provided message directly as it is guarantied to be valid throughout the message sending and we can save memory by not allocating a temporary buffer, also saving on needles memcpy() operations. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: cnl: Use pm_gate->hdr.cmd in cnl_compact_ipc_compress() ASoC: SOF: ipc: Drop header parameter from sof_ipc_tx_message_unlocked() ASoC: SOF: ipc: Do not allocate buffer for msg_data sound/soc/sof/intel/cnl.c | 6 ++---- sound/soc/sof/ipc.c | 24 ++++++++++++------------ 2 files changed, 14 insertions(+), 16 deletions(-) -- 2.35.0

2 4

[PATCH v2 0/3] (Re)enable DP/HDMI audio for RK3399 Gru
by Brian Norris 29 Jan '22

29 Jan '22

This series fixes DP/HDMI audio for RK3399 Gru systems. First, there was a regression with the switch to SPDIF. Patch 1 can be taken separately as a regression fix if desired. But it's not quite so useful (at least on Chrome OS systems) without the second part. Second, jack detection was never upstreamed, because the hdmi-codec dependencies were still being worked out when this platform was first supported. Patches cover a few subsystems. Perhaps this is something for arm-soc? Changes in v2: - (Un)set pinctrl, because the default assumes we're routing out to external pins Brian Norris (3): arm64: dts: rockchip: Switch RK3399-Gru DP to SPDIF output drm/rockchip: cdn-dp: Support HDMI codec plug-change callback ASoC: rk3399_gru_sound: Wire up DP jack detection arch/arm64/boot/dts/rockchip/rk3399-gru.dtsi | 17 ++++++++---- drivers/gpu/drm/rockchip/cdn-dp-core.c | 28 ++++++++++++++++++++ drivers/gpu/drm/rockchip/cdn-dp-core.h | 4 +++ sound/soc/rockchip/rk3399_gru_sound.c | 20 ++++++++++++++ 4 files changed, 64 insertions(+), 5 deletions(-) -- 2.34.1.703.g22d0c6ccf7-goog

5 11

[PATCH v3 0/6] ASOC: amd: acp: Add generic PDM and PCI driver support for ACP
by Ajit Kumar Pandey 29 Jan '22

29 Jan '22

changes since v2: - add dmic prepare callback and adopted few nitpicks comments - use read_poll_timeout in place of while loop - added check for unused variable as pointed out - few cleanups Ajit Kumar Pandey (6): ASoC: amd: acp: Add generic support for PDM controller on ACP ASoC: amd: acp: Add PDM controller based dmic dai for Renoir ASoC: amd: acp: Add generic PCI driver module for ACP device ASoC: amd: acp: Add ACP init()/deinit() callback for Renoir. ASoC: amd: acp: acp-legacy: Add DMIC dai link support for Renoir ASoC: amd: renoir: Add check for acp configuration flags sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp/Kconfig | 10 ++ sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-legacy-mach.c | 4 +- sound/soc/amd/acp/acp-mach-common.c | 21 +++ sound/soc/amd/acp/acp-pci.c | 160 ++++++++++++++++++++++ sound/soc/amd/acp/acp-pdm.c | 193 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 160 ++++++++++++++++++++++ sound/soc/amd/acp/amd.h | 23 +++- sound/soc/amd/acp/chip_offset_byte.h | 26 ++++ sound/soc/amd/mach-config.h | 1 + sound/soc/amd/renoir/rn-pci-acp3x.c | 7 +- sound/soc/amd/renoir/rn_acp3x.h | 3 + 13 files changed, 609 insertions(+), 4 deletions(-) create mode 100644 sound/soc/amd/acp/acp-pci.c create mode 100644 sound/soc/amd/acp/acp-pdm.c -- 2.25.1

2 7

[PATCH] ASoC: ops: Check for negative values before reading them
by Mark Brown 29 Jan '22

29 Jan '22

The controls allow inputs to be specified as negative but our manipulating them into register fields need to be done on unsigned variables so the checks for negative numbers weren't taking effect properly. Do the checks for negative values on the variable in the ABI struct rather than on our local unsigned copy. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index dc0e7c8d31f3..9833611b83d1 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -316,26 +316,26 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, if (sign_bit) mask = BIT(sign_bit + 1) - 1; + if (ucontrol->value.integer.value[0] < 0) + return -EINVAL; val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; - if (val < 0) - return -EINVAL; val = (val + min) & mask; if (invert) val = max - val; val_mask = mask << shift; val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { + if (ucontrol->value.integer.value[1] < 0) + return -EINVAL; val2 = ucontrol->value.integer.value[1]; if (mc->platform_max && val2 > mc->platform_max) return -EINVAL; if (val2 > max - min) return -EINVAL; - if (val2 < 0) - return -EINVAL; val2 = (val2 + min) & mask; if (invert) val2 = max - val2; @@ -423,13 +423,13 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int err = 0; unsigned int val, val_mask; + if (ucontrol->value.integer.value[0] < 0) + return -EINVAL; val = ucontrol->value.integer.value[0]; if (mc->platform_max && val > mc->platform_max) return -EINVAL; if (val > max - min) return -EINVAL; - if (val < 0) - return -EINVAL; val_mask = mask << shift; val = (val + min) & mask; val = val << shift; -- 2.30.2

1 1

[PATCH v1] ASoC: hdmi-codec: Fix OOB memory accesses
by Dmitry Osipenko 29 Jan '22

29 Jan '22

Correct size of iec_status array by changing it to the size of status array of the struct snd_aes_iec958. This fixes out-of-bounds slab read accesses made by memcpy() of the hdmi-codec driver. This problem is reported by KASAN. Cc: stable(a)vger.kernel.org Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- include/uapi/sound/asound.h | 4 +++- sound/soc/codecs/hdmi-codec.c | 2 +- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index ef0cafe295b2..2d3e5df39a59 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -56,8 +56,10 @@ * * ****************************************************************************/ +#define AES_IEC958_STATUS_SIZE 24 + struct snd_aes_iec958 { - unsigned char status[24]; /* AES/IEC958 channel status bits */ + unsigned char status[AES_IEC958_STATUS_SIZE]; /* AES/IEC958 channel status bits */ unsigned char subcode[147]; /* AES/IEC958 subcode bits */ unsigned char pad; /* nothing */ unsigned char dig_subframe[4]; /* AES/IEC958 subframe bits */ diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index b61f980cabdc..b07607a9ecea 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -277,7 +277,7 @@ struct hdmi_codec_priv { bool busy; struct snd_soc_jack *jack; unsigned int jack_status; - u8 iec_status[5]; + u8 iec_status[AES_IEC958_STATUS_SIZE]; }; static const struct snd_soc_dapm_widget hdmi_widgets[] = { -- 2.33.1

2 1

[PATCH v2 0/2] ASoC DPCM lockdep fixes
by Takashi Iwai 29 Jan '22

29 Jan '22

Hi, this is the revised patches for addressing ASoC lockdep warnings due to the recent DPCM locking refactoring. Takashi v1->v2: - minor correction in the changelog for nested lock - debugfs removal workaround === Takashi Iwai (2): ASoC: soc-pcm: Fix DPCM lockdep warning due to nested stream locks ASoC: soc-pcm: Move debugfs removal out of spinlock include/sound/pcm.h | 15 +++++++++++++++ sound/core/pcm_native.c | 13 +++++++++++++ sound/soc/soc-pcm.c | 18 ++++++++++++------ 3 files changed, 40 insertions(+), 6 deletions(-) -- 2.31.1

2 3

[PATCH] ASoC: rt5682: Fix deadlock on resume
by Peter Ujfalusi 29 Jan '22

29 Jan '22

On resume from suspend the following chain of events can happen: A rt5682_resume() -> mod_delayed_work() for jack_detect_work B DAPM sequence starts ( DAPM is locked now) A1. rt5682_jack_detect_handler() scheduled - Takes both jdet_mutex and calibrate_mutex - Calls in to rt5682_headset_detect() which tries to take DAPM lock, it starts to wait for it as B path took it already. B1. DAPM sequence reaches the "HP Amp", rt5682_hp_event() tries to take the jdet_mutex, but it is locked in A1, so it waits. Deadlock. To solve the deadlock, drop the jdet_mutex, use the jack_detect_work to do the jack removal handling, move the dapm lock up one level to protect the most of the rt5682_jack_detect_handler(), but not the jack reporting as it might trigger a DAPM sequence. The rt5682_headset_detect() can be changed to static as well. Fixes: 8deb34a90f063 ("ASoC: rt5682: fix the wrong jack type detected") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, Recently we started to see resume failures in SOF CI: https://github.com/thesofproject/linux/issues/3370 Which turned out to be caused by a lockdep in rt5682 codec driver on at resume time. This patch is fixing the issue while not introducing other failures, checked by our CI: https://github.com/thesofproject/linux/pull/3392 Regards, Peter sound/soc/codecs/rt5682-i2c.c | 15 ++++----------- sound/soc/codecs/rt5682.c | 24 ++++++++---------------- sound/soc/codecs/rt5682.h | 2 -- 3 files changed, 12 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 20e0f90ea498..20fc0f3766de 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -59,18 +59,12 @@ static void rt5682_jd_check_handler(struct work_struct *work) struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, jd_check_work.work); - if (snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) - & RT5682_JDH_RS_MASK) { + if (snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) & RT5682_JDH_RS_MASK) /* jack out */ - rt5682->jack_type = rt5682_headset_detect(rt5682->component, 0); - - snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, - SND_JACK_HEADSET | - SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3); - } else { + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, 0); + else schedule_delayed_work(&rt5682->jd_check_work, 500); - } } static irqreturn_t rt5682_irq(int irq, void *data) @@ -198,7 +192,6 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, } mutex_init(&rt5682->calibrate_mutex); - mutex_init(&rt5682->jdet_mutex); rt5682_calibrate(rt5682); rt5682_apply_patch_list(rt5682, &i2c->dev); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 415ec564c82e..0a0ec4a021e1 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -922,15 +922,13 @@ static void rt5682_enable_push_button_irq(struct snd_soc_component *component, * * Returns detect status. */ -int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) +static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = &component->dapm; unsigned int val, count; if (jack_insert) { - snd_soc_dapm_mutex_lock(dapm); - snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); @@ -981,8 +979,6 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) snd_soc_component_update_bits(component, RT5682_MICBIAS_2, RT5682_PWR_CLK25M_MASK | RT5682_PWR_CLK1M_MASK, RT5682_PWR_CLK25M_PU | RT5682_PWR_CLK1M_PU); - - snd_soc_dapm_mutex_unlock(dapm); } else { rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, @@ -1011,7 +1007,6 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) dev_dbg(component->dev, "jack_type = %d\n", rt5682->jack_type); return rt5682->jack_type; } -EXPORT_SYMBOL_GPL(rt5682_headset_detect); static int rt5682_set_jack_detect(struct snd_soc_component *component, struct snd_soc_jack *hs_jack, void *data) @@ -1094,6 +1089,7 @@ void rt5682_jack_detect_handler(struct work_struct *work) { struct rt5682_priv *rt5682 = container_of(work, struct rt5682_priv, jack_detect_work.work); + struct snd_soc_dapm_context *dapm; int val, btn_type; while (!rt5682->component) @@ -1102,7 +1098,9 @@ void rt5682_jack_detect_handler(struct work_struct *work) while (!rt5682->component->card->instantiated) usleep_range(10000, 15000); - mutex_lock(&rt5682->jdet_mutex); + dapm = snd_soc_component_get_dapm(rt5682->component); + + snd_soc_dapm_mutex_lock(dapm); mutex_lock(&rt5682->calibrate_mutex); val = snd_soc_component_read(rt5682->component, RT5682_AJD1_CTRL) @@ -1162,6 +1160,9 @@ void rt5682_jack_detect_handler(struct work_struct *work) rt5682->irq_work_delay_time = 50; } + mutex_unlock(&rt5682->calibrate_mutex); + snd_soc_dapm_mutex_unlock(dapm); + snd_soc_jack_report(rt5682->hs_jack, rt5682->jack_type, SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | @@ -1174,9 +1175,6 @@ void rt5682_jack_detect_handler(struct work_struct *work) else cancel_delayed_work_sync(&rt5682->jd_check_work); } - - mutex_unlock(&rt5682->calibrate_mutex); - mutex_unlock(&rt5682->jdet_mutex); } EXPORT_SYMBOL_GPL(rt5682_jack_detect_handler); @@ -1526,7 +1524,6 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -1538,17 +1535,12 @@ static int rt5682_hp_event(struct snd_soc_dapm_widget *w, RT5682_DEPOP_1, 0x60, 0x60); snd_soc_component_update_bits(component, RT5682_DAC_ADC_DIG_VOL1, 0x00c0, 0x0080); - - mutex_lock(&rt5682->jdet_mutex); - snd_soc_component_update_bits(component, RT5682_HP_CTRL_2, RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN, RT5682_HP_C2_DAC_L_EN | RT5682_HP_C2_DAC_R_EN); usleep_range(5000, 10000); snd_soc_component_update_bits(component, RT5682_CHARGE_PUMP_1, RT5682_CP_SW_SIZE_MASK, RT5682_CP_SW_SIZE_L); - - mutex_unlock(&rt5682->jdet_mutex); break; case SND_SOC_DAPM_POST_PMD: diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index c917c76200ea..52ff0d9c36c5 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1463,7 +1463,6 @@ struct rt5682_priv { int jack_type; int irq_work_delay_time; - struct mutex jdet_mutex; }; extern const char *rt5682_supply_names[RT5682_NUM_SUPPLIES]; @@ -1473,7 +1472,6 @@ int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev); -int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert); void rt5682_jack_detect_handler(struct work_struct *work); bool rt5682_volatile_register(struct device *dev, unsigned int reg); -- 2.34.1

2 1

[PATCH] ASoC: soc-pcm: Fix DPCM lockdep warning due to nested stream locks
by Takashi Iwai 29 Jan '22

29 Jan '22

The recent change for DPCM locking caused spurious lockdep warnings. Actually the warnings are false-positive, as those are triggered due to the nested stream locks for FE and BE. Since both locks belong to the same lock class, lockdep sees it as if a deadlock. For fixing this, we need take PCM stream locks for BE with the nested lock primitives. Since currently snd_pcm_stream_lock*() helper assumes only the top-level single locking, a new helper function snd_pcm_stream_lock_irqsave_nested() is defined for a single-depth nested lock, which is now used in the BE DAI trigger that is always performed inside a FE stream lock. Fixes: b7898396f4bb ("ASoC: soc-pcm: Fix and cleanup DPCM locking") Reported-and-tested-by: Hans de Goede <hdegoede(a)redhat.com> Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Link: https://lore.kernel.org/r/73018f3c-9769-72ea-0325-b3f8e2381e30@redhat.com Link: https://lore.kernel.org/alsa-devel/9a0abddd-49e9-872d-2f00-a1697340f786@sam… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 15 +++++++++++++++ sound/core/pcm_native.c | 13 +++++++++++++ sound/soc/soc-pcm.c | 6 +++--- 3 files changed, 31 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 9b187d86e1bd..36da42cd0774 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -617,6 +617,7 @@ void snd_pcm_stream_unlock(struct snd_pcm_substream *substream); void snd_pcm_stream_lock_irq(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irq(struct snd_pcm_substream *substream); unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); +unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream); /** * snd_pcm_stream_lock_irqsave - Lock the PCM stream @@ -635,6 +636,20 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream); void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, unsigned long flags); +/** + * snd_pcm_stream_lock_irqsave_nested - Single-nested PCM stream locking + * @substream: PCM substream + * @flags: irq flags + * + * This locks the PCM stream like snd_pcm_stream_lock_irqsave() but with + * the single-depth lockdep subclass. + */ +#define snd_pcm_stream_lock_irqsave_nested(substream, flags) \ + do { \ + typecheck(unsigned long, flags); \ + flags = _snd_pcm_stream_lock_irqsave_nested(substream); \ + } while (0) + /** * snd_pcm_group_for_each_entry - iterate over the linked substreams * @s: the iterator diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 621883e71194..a056b3ef3c84 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -172,6 +172,19 @@ unsigned long _snd_pcm_stream_lock_irqsave(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave); +unsigned long _snd_pcm_stream_lock_irqsave_nested(struct snd_pcm_substream *substream) +{ + unsigned long flags = 0; + if (substream->pcm->nonatomic) + mutex_lock_nested(&substream->self_group.mutex, + SINGLE_DEPTH_NESTING); + else + spin_lock_irqsave_nested(&substream->self_group.lock, flags, + SINGLE_DEPTH_NESTING); + return flags; +} +EXPORT_SYMBOL_GPL(_snd_pcm_stream_lock_irqsave_nested); + /** * snd_pcm_stream_unlock_irqrestore - Unlock the PCM stream * @substream: PCM substream diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7abfc48b26ca..e8876e65c649 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -46,8 +46,8 @@ static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); } -#define snd_soc_dpcm_stream_lock_irqsave(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave(snd_soc_dpcm_get_substream(rtd, stream), flags) +#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ + snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, int stream) @@ -2094,7 +2094,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave(be, stream, flags); + snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) -- 2.31.1

3 4

[PATCH] ASoC: SOF: Intel: hda: Remove link assignment limitation
by Peter Ujfalusi 28 Jan '22

28 Jan '22

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The limitation to assign a link DMA channel for a BE iff the corresponding host DMA channel is assigned to a connected FE is only applicable if the PROCEN_FMT_QUIRK is set. So, remove it for platforms that do not enable the quirk. Complements: a792bfc1c2bc ("ASoC: SOF: Intel: hda-stream: limit PROCEN workaround") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index cd12589355ef..28a54145c150 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -59,6 +59,8 @@ static struct hdac_ext_stream * { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct sof_intel_hda_stream *hda_stream; + const struct sof_intel_dsp_desc *chip; + struct snd_sof_dev *sdev; struct hdac_ext_stream *res = NULL; struct hdac_stream *stream = NULL; @@ -77,9 +79,20 @@ static struct hdac_ext_stream * continue; hda_stream = hstream_to_sof_hda_stream(hstream); + sdev = hda_stream->sdev; + chip = get_chip_info(sdev->pdata); /* check if link is available */ if (!hstream->link_locked) { + /* + * choose the first available link for platforms that do not have the + * PROCEN_FMT_QUIRK set. + */ + if (!(chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK)) { + res = hstream; + break; + } + if (stream->opened) { /* * check if the stream tag matches the stream -- 2.35.0

2 1

[PATCH] ASoC: SOF: trace: Simplify count adjustment in trace_read
by Peter Ujfalusi 28 Jan '22

28 Jan '22

The first count check and fixup against "buffer - lpos" can be removed as we will do the adjustment later against the "avail" in sof_dfsentry_trace_read() Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/trace.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index f13024c8ebf2..9b505c4fe794 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -308,9 +308,6 @@ static ssize_t sof_dfsentry_trace_read(struct file *file, char __user *buffer, lpos_64 = lpos; lpos = do_div(lpos_64, buffer_size); - if (count > buffer_size - lpos) /* min() not used to avoid sparse warnings */ - count = buffer_size - lpos; - /* get available count based on current host offset */ avail = sof_wait_trace_avail(sdev, lpos, buffer_size); if (sdev->dtrace_error) { @@ -319,7 +316,8 @@ static ssize_t sof_dfsentry_trace_read(struct file *file, char __user *buffer, } /* make sure count is <= avail */ - count = avail > count ? count : avail; + if (count > avail) + count = avail; /* copy available trace data to debugfs */ rem = copy_to_user(buffer, ((u8 *)(dfse->buf) + lpos), count); -- 2.35.0

2 1

[PATCH v2 0/5] Tegra234 APE support
by Sameer Pujar 28 Jan '22

28 Jan '22

This adds APE support on Tegra234 and enables sound card on Jetson AGX Orin platform. Changelog ========= v1->v2 ------ * Rebased patches on latest linux-next Mohan Kumar (1): ASoC: tegra: Update AHUB driver for Tegra234 Sameer Pujar (4): dt-bindings: Add Tegra234 APE support dt-bindings: Document Tegra234 APE support arm64: tegra: Add audio devices on Tegra234 arm64: tegra: APE sound card for Jetson AGX Orin .../bindings/bus/nvidia,tegra210-aconnect.yaml | 1 + .../bindings/dma/nvidia,tegra210-adma.yaml | 4 +- .../bindings/interrupt-controller/arm,gic.yaml | 1 + .../bindings/sound/nvidia,tegra186-dspk.yaml | 4 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 4 +- .../bindings/sound/nvidia,tegra210-adx.yaml | 1 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 1 + .../bindings/sound/nvidia,tegra210-amx.yaml | 3 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 1 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 1 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 1 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 1 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 1 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 1783 ++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra234.dtsi | 432 +++++ include/dt-bindings/clock/tegra234-clock.h | 74 +- include/dt-bindings/memory/tegra234-mc.h | 7 + include/dt-bindings/power/tegra234-powergate.h | 9 + sound/soc/tegra/tegra210_ahub.c | 146 +- sound/soc/tegra/tegra210_ahub.h | 4 +- 20 files changed, 2473 insertions(+), 6 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.7.4

2 6

[PATCH] ASoC: dt-bindings: realtek, rt5682s: Drop Tegra specifics from example
by Rob Herring 28 Jan '22

28 Jan '22

There's no need to complicate examples with a platform specific macro. It also complicates example parsing to figure out the number of interrupt cells in examples (based on bracketing). Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/sound/realtek,rt5682s.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml index d65c0ed5060c..ca5b8987b749 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -21,6 +21,7 @@ properties: description: I2C address of the device. interrupts: + maxItems: 1 description: The CODEC's interrupt output. realtek,dmic1-data-pin: @@ -94,7 +95,7 @@ required: examples: - | - #include <dt-bindings/gpio/tegra-gpio.h> + #include <dt-bindings/gpio/gpio.h> #include <dt-bindings/interrupt-controller/irq.h> i2c { @@ -104,10 +105,9 @@ examples: codec@1a { compatible = "realtek,rt5682s"; reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; + interrupts = <6 IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = - <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + <&gpio 2 GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; realtek,dmic1-clk-pin = <1>; realtek,jd-src = <1>; -- 2.32.0

2 1

[PATCH] ASoC: fsl: Add missing error handling in pcm030_fabric_probe
by Miaoqian Lin 28 Jan '22

28 Jan '22

Add the missing platform_device_put() and platform_device_del() before return from pcm030_fabric_probe in the error handling case. Fixes: c912fa913446 ("ASoC: fsl: register the wm9712-codec") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/fsl/pcm030-audio-fabric.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index af3c3b90c0ac..02c4ae20c673 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -93,15 +93,22 @@ static int pcm030_fabric_probe(struct platform_device *op) dev_err(&op->dev, "platform_device_alloc() failed\n"); ret = platform_device_add(pdata->codec_device); - if (ret) + if (ret) { dev_err(&op->dev, "platform_device_add() failed: %d\n", ret); + goto exit_device_put; + } ret = snd_soc_register_card(card); if (ret) dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); platform_set_drvdata(op, pdata); + return ret; +exit_release_dev: + platform_device_del(pdata->codec_device); +exit_device_put: + platform_device_put(pdata->codec_device); return ret; } -- 2.17.1

2 3

[PATCH 0/4] ASoC: qcom: fixes for Qualcomm codecs and q6apm
by Srinivas Kandagatla 28 Jan '22

28 Jan '22

Hi Mark, Some recent testing found few issues with wcd938x and lpass-tx codec drivers. WCD938x was accessing array out of boundaries resulting in corruption and system crashes along with not handling kcontrol put return values correctly and rx-macro had incorrect sidetone registers offsets. One final fix in q6apm to add a check if graph started before stopping it. Thanks, srini Srinivas Kandagatla (4): ASoC: codecs: wcd938x: fix incorrect used of portid ASoC: codecs: lpass-rx-macro: fix sidetone register offsets ASoC: codecs: wcd938x: fix return value of mixer put function ASoC: qdsp6: q6apm-dai: only stop graphs that are started sound/soc/codecs/lpass-rx-macro.c | 8 ++++---- sound/soc/codecs/wcd938x.c | 31 +++++++++++++++++-------------- sound/soc/qcom/qdsp6/q6apm-dai.c | 7 +++++-- 3 files changed, 26 insertions(+), 20 deletions(-) -- 2.21.0

2 7

[PATCH 1/2] ASoC: ops: fix signedness bug in snd_soc_put_volsw()
by Dan Carpenter 28 Jan '22

28 Jan '22

The "val" and "val2" variables need to signed for the checking to work as intended. Fixes: 817f7c9335ec ("ASoC: ops: Reject out of bounds values in snd_soc_put_volsw()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-ops.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index dc0e7c8d31f3..0091fa96eb48 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -310,8 +310,9 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int invert = mc->invert; int err; bool type_2r = false; - unsigned int val2 = 0; - unsigned int val, val_mask; + unsigned int val_mask; + int val2 = 0; + int val; if (sign_bit) mask = BIT(sign_bit + 1) - 1; -- 2.20.1

2 5

[PATCH] ASoC: dt-bindings: Centralize the 'sound-dai' definition
by Rob Herring 28 Jan '22

28 Jan '22

'sound-dai' is a common property, but has duplicate type definitions. Create a new common definition to define the type and then update all the other occurrences to just define how many entries there are just like other phandle+arg properties. The constraints on the number of entries is based on the examples and could be wrong. Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> Signed-off-by: Rob Herring <robh(a)kernel.org> --- Please ack, this depends on commit abf0fee97313 ("dt-bindings: Improve phandle-array schemas") in my tree. --- .../bindings/sound/amlogic,gx-sound-card.yaml | 4 ++-- .../bindings/sound/google,sc7180-trogdor.yaml | 6 ++++-- .../bindings/sound/imx-audio-card.yaml | 7 +++++-- .../bindings/sound/qcom,sm8250.yaml | 10 +++++++--- .../bindings/sound/samsung,aries-wm8994.yaml | 5 +---- .../bindings/sound/samsung,midas-audio.yaml | 2 -- .../bindings/sound/samsung,odroid.yaml | 9 +++------ .../devicetree/bindings/sound/sound-dai.yaml | 20 +++++++++++++++++++ 8 files changed, 42 insertions(+), 21 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sound-dai.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index 2e35aeaa8781..8b5be4b92f35 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -57,7 +57,7 @@ patternProperties: rate sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle of the CPU DAI patternProperties: @@ -71,7 +71,7 @@ patternProperties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle of the codec DAI required: diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 837e3faa63a9..233caa0ade87 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -62,13 +62,15 @@ patternProperties: description: Holds subnode which indicates cpu dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 codec: description: Holds subnode which indicates codec dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 required: - link-name diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml index d1816dd061cf..bb3a435722c7 100644 --- a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -59,13 +59,16 @@ patternProperties: description: Holds subnode which indicates cpu dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 codec: description: Holds subnode which indicates codec dai. type: object properties: - sound-dai: true + sound-dai: + minItems: 1 + maxItems: 2 fsl,mclk-equal-bclk: description: Indicates mclk can be equal to bclk, especially for sai interface diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index 4bfda04b4608..4ecd4080bb96 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -69,19 +69,23 @@ patternProperties: description: Holds subnode which indicates cpu dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 platform: description: Holds subnode which indicates platform dai. type: object properties: - sound-dai: true + sound-dai: + maxItems: 1 codec: description: Holds subnode which indicates codec dai. type: object properties: - sound-dai: true + sound-dai: + minItems: 1 + maxItems: 4 required: - link-name diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index eb487ed3ca3b..4ffa275b3c49 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -27,9 +27,6 @@ properties: sound-dai: minItems: 2 maxItems: 2 - items: - maxItems: 1 - $ref: /schemas/types.yaml#/definitions/phandle-array description: | phandles to the I2S controller and bluetooth codec, in that order @@ -38,7 +35,7 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array + maxItems: 1 description: phandle to the WM8994 CODEC samsung,audio-routing: diff --git a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml index 095775c598fa..ec50bcb4af5f 100644 --- a/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,midas-audio.yaml @@ -21,7 +21,6 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array maxItems: 1 description: phandle to the I2S controller required: @@ -31,7 +30,6 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array maxItems: 1 description: phandle to the WM1811 CODEC required: diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index e8122bc87362..db2513f3e168 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -37,18 +37,15 @@ properties: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array description: phandles to the I2S controllers codec: type: object properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle-array - description: | - List of phandles to the CODEC nodes, - first entry must be corresponding to the MAX98090 CODEC and - the second entry must be the phandle of the HDMI IP block node. + items: + - description: phandle of the MAX98090 CODEC + - description: phandle of the HDMI IP block node samsung,audio-routing: $ref: /schemas/types.yaml#/definitions/non-unique-string-array diff --git a/Documentation/devicetree/bindings/sound/sound-dai.yaml b/Documentation/devicetree/bindings/sound/sound-dai.yaml new file mode 100644 index 000000000000..61c6f7abc4e7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sound-dai.yaml @@ -0,0 +1,20 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sound-dai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Digital Audio Interface consumer Device Tree Bindings + +maintainers: + - Rob Herring <robh(a)kernel.org> + +select: true + +properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: A phandle plus args to digital audio interface provider(s) + +additionalProperties: true +... -- 2.32.0

4 5

Fix pre-processor defs parsing with recursive includes
by GitHub pull_request - opened 27 Jan '22

27 Jan '22

alsa-project/alsa-utils pull request #140 was opened from ranj063: Since the Define {} config has been removed form the input_cfg, use the tplg_pp->define_cfg_merged for search for variable definitions. Also, fix the parsing logic to first merge all variables from the conf files recursively and then override them with the command line definitions. Request URL : https://github.com/alsa-project/alsa-utils/pull/140 Patch URL : https://github.com/alsa-project/alsa-utils/pull/140.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

acp6x: add initial support for AMD Yellow Carp - ACP6x
by GitHub pull_request - opened 27 Jan '22

27 Jan '22

alsa-project/alsa-ucm-conf pull request #138 was opened from perexg: https://github.com/alsa-project/alsa-ucm-conf/issues/136 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/138 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/138.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Add intial Amlogic GXL support
by GitHub pull_request - opened 27 Jan '22

27 Jan '22

alsa-project/alsa-ucm-conf pull request #137 was opened from jeromebrunet: This adds the support for the GXL-P241 and the LIBRETECH-CC. I have proposed the related ucm-validator PR: https://github.com/alsa-project/alsa-tests/pull/23 I was not successful with txt check: ``` !!Sysfs card info !!--------------- Traceback (most recent call last): File "/home/jbrunet/work/generic/sound/alsa-tests/python/ucm-validator/./ucm.py", line 238, in <module> main(sys.argv) File "/home/jbrunet/work/generic/sound/alsa-tests/python/ucm-validator/./ucm.py", line 233, in main r = globals()[cmd](*argv[2:]) File "/home/jbrunet/work/generic/sound/alsa-tests/python/ucm-validator/./ucm.py", line 185, in do_configs errors, warnings = config_walk(alsainfo_path) File "/home/jbrunet/work/generic/sound/alsa-tests/python/ucm-validator/./ucm.py", line 129, in config_walk errors1, warnings1 = config_walk(path2) File "/home/jbrunet/work/generic/sound/alsa-tests/python/ucm-validator/./ucm.py", line 142, in config_walk info.load(path2) File "/home/jbrunet/work/generic/sound/alsa-tests/python/ucm-validator/../lib/alsainfo.py", line 314, in load raise AlsaInfoError("%s: unknown section %s" % (filename, repr(name))) alsainfo.AlsaInfoError: configs/Meson/LIBRETECH-CC.txt: unknown section 'Sysfs card info' ``` I also check with the dummy json: ``` *** configuration 'configs/Meson/GXL-P241.json' @ AlsaJsonSoundcard(json='GXL-P241', id='GXL-P241', driver='gx-sound-card', name='GXL-P241', longname='GXL-P241') *** Device file 'conf.d/gx-sound-card/GXL-P241.conf' Verb 'HiFi', file 'Meson/p241/p241-HiFi.conf' Device 'Line' Device 'HDMI' ERR: conf.d/gx-sound-card/GXL-P241.conf: Verb=HiFi Device=Line PlaybackPCM defined in SectionVerb ERR: AlsaJsonSoundcard(json='GXL-P241', id='GXL-P241', driver='gx-sound-card', name='GXL-P241', longname='GXL-P241') ERR: total errors: 1 ``` I'm not quite sure where to put the `PlaybackPCM`, if not in `Verb`. The `Device ` section in not a good fit, as far as I understand. On these devices, we have a single PCM interface. HDMI and Line Devices may be enabled on it (possibly simultaneously). Any suggestion ? Looking at pulseaudio, I'm quite happy with the result: ``` $ pactl list sinks Sink #0 State: SUSPENDED Name: alsa_output.platform-sound.HiFi__hw_LIBRETECHCC_0__sink Description: Built-in Audio HDMI + Analog Lineout Driver: module-alsa-card.c Sample Specification: s16le 2ch 44100Hz Channel Map: front-left,front-right Owner Module: 6 Mute: no Volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB balance 0.00 Base Volume: 65536 / 100% / 0.00 dB Monitor Source: alsa_output.platform-sound.HiFi__hw_LIBRETECHCC_0__sink.monitor Latency: 0 usec, configured 0 usec Flags: HARDWARE DECIBEL_VOLUME LATENCY Properties: alsa.resolution_bits = "16" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "" alsa.id = "fe.dai-link-0 (*)" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "LIBRETECH-CC" alsa.long_card_name = "LIBRETECH-CC" alsa.driver_name = "snd_soc_meson_gx_sound_card" device.bus_path = "platform-sound" sysfs.path = "/devices/platform/sound/sound/card0" device.form_factor = "internal" device.string = "_ucm0001.hw:LIBRETECHCC,0" device.buffering.buffer_size = "352768" device.buffering.fragment_size = "176384" device.access_mode = "mmap+timer" device.profile.name = "HiFi: hw:LIBRETECHCC,0: sink" device.profile.description = "HDMI + Analog Lineout" alsa.mixer_device = "_ucm0001.hw:LIBRETECHCC" device.description = "Built-in Audio HDMI + Analog Lineout" module-udev-detect.discovered = "1" device.icon_name = "audio-card" Ports: [Out] HDMI: HDMI (type: HDMI, priority: 100, availability unknown) [Out] Line: Analog Lineout (type: Line, priority: 100, availability unknown) Active Port: [Out] HDMI Formats: pcm ``` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/137 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/137.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Mouse stopped working in xterm and rxvt-unicode
by GitHub issues - opened 27 Jan '22

27 Jan '22

alsa-project/alsa-utils issue #139 was opened from m-ou-se: Since 1.2.6, I can't control alsamixer with my mouse's scroll wheel anymore in xterm and rxvt-unicode. I'm assuming it's related to this commit: https://github.com/alsa-project/alsa-utils/commit/31820c5f239fca2de12c8d1a7… Issue URL : https://github.com/alsa-project/alsa-utils/issues/139 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: hda: Fix signedness of sscanf() arguments
by Takashi Iwai 27 Jan '22

27 Jan '22

The %x format of sscanf() takes an unsigned int pointer, while we pass a signed int pointer. Practically it's OK, but this may result in a compile warning. Let's fix it. Fixes: a235d5b8e550 ("ALSA: hda: Allow model option to specify PCI SSID alias") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_auto_parser.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 82c492b05667..cd1db943b7e0 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -981,7 +981,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, int id = HDA_FIXUP_ID_NOT_SET; const char *name = NULL; const char *type = NULL; - int vendor, device; + unsigned int vendor, device; if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; -- 2.31.1

1 0

[PATCH] ALSA: usb-audio: initialize variables that could ignore errors
by trix＠redhat.com 27 Jan '22

27 Jan '22

From: Tom Rix <trix(a)redhat.com> clang static analysis reports this representative issue mixer.c:1548:35: warning: Assigned value is garbage or undefined ucontrol->value.integer.value[0] = val; ^ ~~~ The filter_error() macro allows errors to be ignored. If errors can be ignored, initialize variables so garbage will not be used. Fixes: 48cc42973509 ("ALSA: usb-audio: Filter error from connector kctl ops, too") Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/usb/mixer.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e8f3f8d622ec5..630766ba259fd 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1527,6 +1527,10 @@ static int get_connector_value(struct usb_mixer_elem_info *cval, usb_audio_err(chip, "cannot get connectors status: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", UAC_GET_CUR, validx, idx, cval->val_type); + + if (val) + *val = 0; + return filter_error(cval, ret); } -- 2.26.3

2 1

[PATCH 0/5] Tegra234 APE support
by Sameer Pujar 27 Jan '22

27 Jan '22

This adds APE support on Tegra234 and enables sound card on Jetson AGX Orin platform. Mohan Kumar (1): ASoC: tegra: Update AHUB driver for Tegra234 Sameer Pujar (4): dt-bindings: Add Tegra234 APE support dt-bindings: Document Tegra234 APE support arm64: tegra: Add audio devices on Tegra234 arm64: tegra: APE sound card for Jetson AGX Orin .../bindings/bus/nvidia,tegra210-aconnect.yaml | 1 + .../bindings/dma/nvidia,tegra210-adma.yaml | 4 +- .../bindings/interrupt-controller/arm,gic.yaml | 1 + .../bindings/sound/nvidia,tegra186-dspk.yaml | 4 +- .../bindings/sound/nvidia,tegra210-admaif.yaml | 4 +- .../bindings/sound/nvidia,tegra210-adx.yaml | 1 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 1 + .../bindings/sound/nvidia,tegra210-amx.yaml | 3 + .../bindings/sound/nvidia,tegra210-dmic.yaml | 1 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 1 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 1 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 1 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 1 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 1781 ++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra234.dtsi | 432 +++++ include/dt-bindings/clock/tegra234-clock.h | 72 + include/dt-bindings/memory/tegra234-mc.h | 5 + include/dt-bindings/power/tegra234-powergate.h | 1 + sound/soc/tegra/tegra210_ahub.c | 147 +- sound/soc/tegra/tegra210_ahub.h | 4 +- 20 files changed, 2461 insertions(+), 5 deletions(-) -- 2.7.4

2 7

Add support for AMD ACP Microphone devices for Yellow Carp platform
by GitHub issues - opened 27 Jan '22

27 Jan '22

alsa-project/alsa-ucm-conf issue #136 was opened from vijendarmukunda: We have tried to add similar changes which were applied for Renoir platform for YC platform by following below link. https://github.com/alsa-project/alsa-ucm-conf/issues/54 But we are unable to create links. Without ALSA UCM changes, we are able to observe similar issues earlier reported on Renoir platform as mentioned below. 1) Fallback profile is getting loaded and capture endpoint name is not listed as "Digital Microphone" 2) Mic Mute Led button is not reflected with Mute status. Attached pa-info and alsa info output. [painfo.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7947396/pai… [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7947401/… a Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/136 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fixes for attribute value validation in the topology pre-processor
by GitHub pull_request - opened 27 Jan '22

27 Jan '22

alsa-project/alsa-utils pull request #138 was opened from ranj063: This PR fixes 2 issues: 1. Allow variable definitions within included conf files 2. Validating attributes after expanding the variables Request URL : https://github.com/alsa-project/alsa-utils/pull/138 Patch URL : https://github.com/alsa-project/alsa-utils/pull/138.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: [PATCH v2] sound/soc: adds TAS5754M digital input amplifier component driver
by Joerg Schambacher 26 Jan '22

26 Jan '22

Mark, Thanks for your useful feedback. I guess my comments on my first reply got lost somewhere on the way .... I've updated and fixed some other issues meanwhile, too. Joerg On Wed, Nov 17, 2021 at 03:13:47PM +0000, Mark Brown wrote: > On Fri, Oct 29, 2021 at 11:57:30AM +0200, Joerg Schambacher wrote: > > > Adds a minimum component driver to run the amplifier in I2S master > > mode only from standard audio clocks. Therefore, it only allows > > 44.1, 88.2, 176.4, 48, 96 and 192ksps with 16, 20, 24 and 32 bits > > sample size. Digital volume control and the -6dB switch are supported. > > This looks pretty clean, there's a bunch of minor stylistic bits below > but other than one major thing this looks good. The major thing is the > use of a had coded init sequence to configure the device rather than > implemenenting the relevant APIs to do so, that's going to make the > driver unsuitable for use on boards other than the specific one you > happen to be looking at at the minute. > > Please submit patches using subject lines reflecting the style for the > subsystem, this makes it easier for people to identify relevant patches. > Look at what existing commits in the area you're changing are doing and > make sure your subject lines visually resemble what they're doing. > > > @@ -0,0 +1,547 @@ > > +// SPDX-License-Identifier: GPL-2.0 > > +/* > > + * Driver for the TAS5754M Audio Amplifier in Master mode (only) > > + * supports only standard audio frequencies 44.1 to 192 ksps > > Please make the entire comment a C++ one so things look more > intentional. > > > +EXPORT_SYMBOL_GPL(tas5754m_regmap); > yes, remainder from some experiments - fixed > There's only one file in the driver, why is this exported? > > > + mclk = clk_get_rate(tas5754m->sclk); > > + bclk = sample_len * 2 * params_rate(params); > > snd_soc_params_to_bclk(). snd_soc_params_to_bclk() does not always gives the necessary value > > > + // set SCLK divider > > + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_CLKDIV_1, > > + bclk_div - 1); > > + > > + // set LRCLK divider > > + ret |= regmap_write(tas5754m->regmap, TAS5754M_MASTER_CLKDIV_2, > > + lrclk_div - 1); > Fixed > Check the return statuses individually or just don't bother, if you or > them together then you could end up with a nonsensical error code which > could cause confusion down the line. > > > + if (mute) { > > + snd_soc_component_write(component, TAS5754M_MUTE, 0x11); > > + } else { > > + usleep_range(1000, 2000); > > + snd_soc_component_write(component, TAS5754M_MUTE, 0x00); > > Why the sleep here? Wait for settling of the clocks before unmute > > > + ret = regmap_multi_reg_write(regmap, tas5754m_init_sequence, > > + ARRAY_SIZE(tas5754m_init_sequence)); > > This init sequence looks an awful lot like it's doing board specific > configuration - it's enabling the PLL, configuring GPIOs and clock > dividers among other things. These should all be done through board > specific configuration or based on the currently configured audio path, > what's suitable for one board may not be suitable for another board. > Doing a reset of the chip is good but the rest of it really looks like > it doesn't belong. Ok, completely re-organized in the next trial > > > + ret = snd_soc_register_component(dev, > > + &tas5754m_soc_component, &tas5754m_dai, 1); > > devm_snd_soc_register_component() > fixed > > +static const struct i2c_device_id tas5754m_i2c_id[] = { > > + { "tas5754m", }, > > + { } > > +}; > > +MODULE_DEVICE_TABLE(i2c, tas5754m_i2c_id); > > + > > +#ifdef CONFIG_OF > > +static const struct of_device_id tas5754m_of_match[] = { > > + { .compatible = "ti,tas5754m", }, > > + { .compatible = "ti,tas5756m", }, > > + { } > > +}; > > +MODULE_DEVICE_TABLE(of, tas5754m_of_match); > > +#endif > > Why not list both I2C IDs as well? Fixed > > > @@ -0,0 +1,259 @@ > > +/* SPDX-License-Identifier: GPL-2.0-only */ > > +/* > > + * Driver for the TAS5754M DAC+amplifier combo devices > > + * > > + * Author: (copied from pcm512x.h) > > + * Mark Brown <broonie(a)kernel.org> > > + * Copyright 2014 Linaro Ltd > > + */ > > If the register map can be copied can't the two drivers be combined? The TI apps team recommended to write a separate driver as there are some differences. I have also aligned some names to the TAS575xM spec in the next patch > > > +#define TAS5754M_VIRT_BASE 0x000 > > +#define TAS5754M_PAGE_LEN 0x80 > > +#define TAS5754M_PAGE_BASE(n) (TAS5754M_VIRT_BASE + (TAS5754M_PAGE_LEN * n)) > > + > > +#define TAS5754M_PAGE 0 > > There's no mention of paging in the regmap description for the driver > which feels like it's asking for problems. I think, it's defined in the correct way. Where/when exactly do you see a problem? > > > + > > +#define TAS5754M_RESET (TAS5754M_PAGE_BASE(0) + 1) > > +#define TAS5754M_POWER (TAS5754M_PAGE_BASE(0) + 2) > > +#define TAS5754M_MUTE (TAS5754M_PAGE_BASE(0) + 3) > > +#define TAS5754M_PLL_EN (TAS5754M_PAGE_BASE(0) + 4) > > +#define TAS5754M_SPI_MISO_FUNCTION (TAS5754M_PAGE_BASE(0) + 6) > > +#define TAS5754M_DSP (TAS5754M_PAGE_BASE(0) + 7) > > +#define TAS5754M_GPIO_EN (TAS5754M_PAGE_BASE(0) + 8) > > +#define TAS5754M_BCLK_LRCLK_CFG (TAS5754M_PAGE_BASE(0) + 9) > > +#define TAS5754M_DSP_GPIO_INPUT (TAS5754M_PAGE_BASE(0) + 10) > > +#define TAS5754M_MASTER_MODE (TAS5754M_PAGE_BASE(0) + 12) > > +#define TAS5754M_PLL_REF (TAS5754M_PAGE_BASE(0) + 13) > > +#define TAS5754M_DAC_REF (TAS5754M_PAGE_BASE(0) + 14) > > +#define TAS5754M_GPIO_DACIN (TAS5754M_PAGE_BASE(0) + 16) > > +#define TAS5754M_GPIO_PLLIN (TAS5754M_PAGE_BASE(0) + 18) > > +#define TAS5754M_SYNCHRONIZE (TAS5754M_PAGE_BASE(0) + 19) > > +#define TAS5754M_PLL_COEFF_0 (TAS5754M_PAGE_BASE(0) + 20) > > +#define TAS5754M_PLL_COEFF_1 (TAS5754M_PAGE_BASE(0) + 21) > > +#define TAS5754M_PLL_COEFF_2 (TAS5754M_PAGE_BASE(0) + 22) > > +#define TAS5754M_PLL_COEFF_3 (TAS5754M_PAGE_BASE(0) + 23) > > +#define TAS5754M_PLL_COEFF_4 (TAS5754M_PAGE_BASE(0) + 24) > > +#define TAS5754M_DSP_CLKDIV (TAS5754M_PAGE_BASE(0) + 27) > > +#define TAS5754M_DAC_CLKDIV (TAS5754M_PAGE_BASE(0) + 28) > > +#define TAS5754M_NCP_CLKDIV (TAS5754M_PAGE_BASE(0) + 29) > > +#define TAS5754M_OSR_CLKDIV (TAS5754M_PAGE_BASE(0) + 30) > > +#define TAS5754M_MASTER_CLKDIV_1 (TAS5754M_PAGE_BASE(0) + 32) > > +#define TAS5754M_MASTER_CLKDIV_2 (TAS5754M_PAGE_BASE(0) + 33) > > +#define TAS5754M_FS_SPEED_MODE (TAS5754M_PAGE_BASE(0) + 34) > > +#define TAS5754M_IDAC_1 (TAS5754M_PAGE_BASE(0) + 35) > > +#define TAS5754M_IDAC_2 (TAS5754M_PAGE_BASE(0) + 36) > > +#define TAS5754M_ERROR_DETECT (TAS5754M_PAGE_BASE(0) + 37) > > +#define TAS5754M_I2S_1 (TAS5754M_PAGE_BASE(0) + 40) > > +#define TAS5754M_I2S_2 (TAS5754M_PAGE_BASE(0) + 41) > > +#define TAS5754M_DAC_ROUTING (TAS5754M_PAGE_BASE(0) + 42) > > +#define TAS5754M_DSP_PROGRAM (TAS5754M_PAGE_BASE(0) + 43) > > +#define TAS5754M_CLKDET (TAS5754M_PAGE_BASE(0) + 44) > > +#define TAS5754M_AUTO_MUTE (TAS5754M_PAGE_BASE(0) + 59) > > +#define TAS5754M_DIGITAL_VOLUME_1 (TAS5754M_PAGE_BASE(0) + 60) > > +#define TAS5754M_DIGITAL_VOLUME_2 (TAS5754M_PAGE_BASE(0) + 61) > > +#define TAS5754M_DIGITAL_VOLUME_3 (TAS5754M_PAGE_BASE(0) + 62) > > +#define TAS5754M_DIGITAL_MUTE_1 (TAS5754M_PAGE_BASE(0) + 63) > > +#define TAS5754M_DIGITAL_MUTE_2 (TAS5754M_PAGE_BASE(0) + 64) > > +#define TAS5754M_DIGITAL_MUTE_3 (TAS5754M_PAGE_BASE(0) + 65) > > +#define TAS5754M_GPIO_OUTPUT_1 (TAS5754M_PAGE_BASE(0) + 80) > > +#define TAS5754M_GPIO_OUTPUT_2 (TAS5754M_PAGE_BASE(0) + 81) > > +#define TAS5754M_GPIO_OUTPUT_3 (TAS5754M_PAGE_BASE(0) + 82) > > +#define TAS5754M_GPIO_OUTPUT_4 (TAS5754M_PAGE_BASE(0) + 83) > > +#define TAS5754M_GPIO_OUTPUT_5 (TAS5754M_PAGE_BASE(0) + 84) > > +#define TAS5754M_GPIO_OUTPUT_6 (TAS5754M_PAGE_BASE(0) + 85) > > +#define TAS5754M_GPIO_CONTROL_1 (TAS5754M_PAGE_BASE(0) + 86) > > +#define TAS5754M_GPIO_CONTROL_2 (TAS5754M_PAGE_BASE(0) + 87) > > +#define TAS5754M_OVERFLOW (TAS5754M_PAGE_BASE(0) + 90) > > +#define TAS5754M_RATE_DET_1 (TAS5754M_PAGE_BASE(0) + 91) > > +#define TAS5754M_RATE_DET_2 (TAS5754M_PAGE_BASE(0) + 92) > > +#define TAS5754M_RATE_DET_3 (TAS5754M_PAGE_BASE(0) + 93) > > +#define TAS5754M_RATE_DET_4 (TAS5754M_PAGE_BASE(0) + 94) > > +#define TAS5754M_CLOCK_STATUS (TAS5754M_PAGE_BASE(0) + 95) > > +#define TAS5754M_ANALOG_MUTE_DET (TAS5754M_PAGE_BASE(0) + 108) > > +#define TAS5754M_GPIN (TAS5754M_PAGE_BASE(0) + 119) > > +#define TAS5754M_DIGITAL_MUTE_DET (TAS5754M_PAGE_BASE(0) + 120) > > + > > +#define TAS5754M_OUTPUT_AMPLITUDE (TAS5754M_PAGE_BASE(1) + 1) > > +#define TAS5754M_ANALOG_GAIN_CTRL (TAS5754M_PAGE_BASE(1) + 2) > > +#define TAS5754M_UNDERVOLTAGE_PROT (TAS5754M_PAGE_BASE(1) + 5) > > +#define TAS5754M_ANALOG_MUTE_CTRL (TAS5754M_PAGE_BASE(1) + 6) > > +#define TAS5754M_ANALOG_GAIN_BOOST (TAS5754M_PAGE_BASE(1) + 7) > > +#define TAS5754M_VCOM_CTRL_1 (TAS5754M_PAGE_BASE(1) + 8) > > +#define TAS5754M_VCOM_CTRL_2 (TAS5754M_PAGE_BASE(1) + 9) > > + > > +#define TAS5754M_CRAM_CTRL (TAS5754M_PAGE_BASE(44) + 1) > > + > > +#define TAS5754M_FLEX_A (TAS5754M_PAGE_BASE(253) + 63) > > +#define TAS5754M_FLEX_B (TAS5754M_PAGE_BASE(253) + 64) > > + > > +#define TAS5754M_MAX_REGISTER (TAS5754M_PAGE_BASE(253) + 64) > > + > > +/* Page 0, Register 1 - reset */ > > +#define TAS5754M_RSTR (1 << 0) > > +#define TAS5754M_RSTM (1 << 4) > > + > > +/* Page 0, Register 2 - power */ > > +#define TAS5754M_RQPD (1 << 0) > > +#define TAS5754M_RQPD_SHIFT 0 > > +#define TAS5754M_RQST (1 << 4) > > +#define TAS5754M_RQST_SHIFT 4 > > + > > +/* Page 0, Register 3 - mute */ > > +#define TAS5754M_RQMR (1 << 0) > > +#define TAS5754M_RQMR_SHIFT 0 > > +#define TAS5754M_RQML (1 << 4) > > +#define TAS5754M_RQML_SHIFT 4 > > + > > +/* Page 0, Register 4 - PLL */ > > +#define TAS5754M_PLLE (1 << 0) > > +#define TAS5754M_PLLE_SHIFT 0 > > +#define TAS5754M_PLCK (1 << 4) > > +#define TAS5754M_PLCK_SHIFT 4 > > + > > +/* Page 0, Register 7 - DSP */ > > +#define TAS5754M_SDSL (1 << 0) > > +#define TAS5754M_SDSL_SHIFT 0 > > +#define TAS5754M_DEMP (1 << 4) > > +#define TAS5754M_DEMP_SHIFT 4 > > + > > +/* Page 0, Register 8 - GPIO output enable */ > > +#define TAS5754M_G1OE (1 << 0) > > +#define TAS5754M_G2OE (1 << 1) > > +#define TAS5754M_G3OE (1 << 2) > > +#define TAS5754M_G4OE (1 << 3) > > +#define TAS5754M_G5OE (1 << 4) > > +#define TAS5754M_G6OE (1 << 5) > > + > > +/* Page 0, Register 9 - BCK, LRCLK configuration */ > > +#define TAS5754M_LRKO (1 << 0) > > +#define TAS5754M_LRKO_SHIFT 0 > > +#define TAS5754M_BCKO (1 << 4) > > +#define TAS5754M_BCKO_SHIFT 4 > > +#define TAS5754M_BCKP (1 << 5) > > +#define TAS5754M_BCKP_SHIFT 5 > > + > > +/* Page 0, Register 12 - Master mode BCK, LRCLK reset */ > > +#define TAS5754M_RLRK (1 << 0) > > +#define TAS5754M_RLRK_SHIFT 0 > > +#define TAS5754M_RBCK (1 << 1) > > +#define TAS5754M_RBCK_SHIFT 1 > > + > > +/* Page 0, Register 13 - PLL reference */ > > +#define TAS5754M_SREF (7 << 4) > > +#define TAS5754M_SREF_SHIFT 4 > > +#define TAS5754M_SREF_SCK (0 << 4) > > +#define TAS5754M_SREF_BCK (1 << 4) > > +#define TAS5754M_SREF_GPIO (3 << 4) > > + > > +/* Page 0, Register 14 - DAC reference */ > > +#define TAS5754M_SDAC (7 << 4) > > +#define TAS5754M_SDAC_SHIFT 4 > > +#define TAS5754M_SDAC_MCK (0 << 4) > > +#define TAS5754M_SDAC_PLL (1 << 4) > > +#define TAS5754M_SDAC_SCK (3 << 4) > > +#define TAS5754M_SDAC_BCK (4 << 4) > > +#define TAS5754M_SDAC_GPIO (5 << 4) > > + > > +/* Page 0, Register 16, 18 - GPIO source for DAC, PLL */ > > +#define TAS5754M_GREF (7 << 0) > > +#define TAS5754M_GREF_SHIFT 0 > > +#define TAS5754M_GREF_GPIO1 (0 << 0) > > +#define TAS5754M_GREF_GPIO2 (1 << 0) > > +#define TAS5754M_GREF_GPIO3 (2 << 0) > > +#define TAS5754M_GREF_GPIO4 (3 << 0) > > +#define TAS5754M_GREF_GPIO5 (4 << 0) > > +#define TAS5754M_GREF_GPIO6 (5 << 0) > > + > > +/* Page 0, Register 19 - synchronize */ > > +#define TAS5754M_RQSY (1 << 0) > > +#define TAS5754M_RQSY_RESUME (0 << 0) > > +#define TAS5754M_RQSY_HALT (1 << 0) > > + > > +/* Page 0, Register 34 - fs speed mode */ > > +#define TAS5754M_FSSP (3 << 0) > > +#define TAS5754M_FSSP_SHIFT 0 > > +#define TAS5754M_FSSP_48KHZ (0 << 0) > > +#define TAS5754M_FSSP_96KHZ (1 << 0) > > +#define TAS5754M_FSSP_192KHZ (2 << 0) > > +#define TAS5754M_FSSP_384KHZ (3 << 0) > > + > > +/* Page 0, Register 37 - Error detection */ > > +#define TAS5754M_IPLK (1 << 0) > > +#define TAS5754M_DCAS (1 << 1) > > +#define TAS5754M_IDCM (1 << 2) > > +#define TAS5754M_IDCH (1 << 3) > > +#define TAS5754M_IDSK (1 << 4) > > +#define TAS5754M_IDBK (1 << 5) > > +#define TAS5754M_IDFS (1 << 6) > > + > > +/* Page 0, Register 40 - I2S configuration */ > > +#define TAS5754M_ALEN (3 << 0) > > +#define TAS5754M_ALEN_SHIFT 0 > > +#define TAS5754M_ALEN_16 (0 << 0) > > +#define TAS5754M_ALEN_20 (1 << 0) > > +#define TAS5754M_ALEN_24 (2 << 0) > > +#define TAS5754M_ALEN_32 (3 << 0) > > +#define TAS5754M_AFMT (3 << 4) > > +#define TAS5754M_AFMT_SHIFT 4 > > +#define TAS5754M_AFMT_I2S (0 << 4) > > +#define TAS5754M_AFMT_DSP (1 << 4) > > +#define TAS5754M_AFMT_RTJ (2 << 4) > > +#define TAS5754M_AFMT_LTJ (3 << 4) > > + > > +/* Page 0, Register 42 - DAC routing */ > > +#define TAS5754M_AUPR_SHIFT 0 > > +#define TAS5754M_AUPL_SHIFT 4 > > + > > +/* Page 0, Register 59 - auto mute */ > > +#define TAS5754M_ATMR_SHIFT 0 > > +#define TAS5754M_ATML_SHIFT 4 > > + > > +/* Page 0, Register 63 - ramp rates */ > > +#define TAS5754M_VNDF_SHIFT 6 > > +#define TAS5754M_VNDS_SHIFT 4 > > +#define TAS5754M_VNUF_SHIFT 2 > > +#define TAS5754M_VNUS_SHIFT 0 > > + > > +/* Page 0, Register 64 - emergency ramp rates */ > > +#define TAS5754M_VEDF_SHIFT 6 > > +#define TAS5754M_VEDS_SHIFT 4 > > + > > +/* Page 0, Register 65 - Digital mute enables */ > > +#define TAS5754M_ACTL_SHIFT 2 > > +#define TAS5754M_AMLE_SHIFT 1 > > +#define TAS5754M_AMRE_SHIFT 0 > > + > > +/* Page 0, Register 80-85, GPIO output selection */ > > +#define TAS5754M_GxSL (31 << 0) > > +#define TAS5754M_GxSL_SHIFT 0 > > +#define TAS5754M_GxSL_OFF (0 << 0) > > +#define TAS5754M_GxSL_DSP (1 << 0) > > +#define TAS5754M_GxSL_REG (2 << 0) > > +#define TAS5754M_GxSL_AMUTB (3 << 0) > > +#define TAS5754M_GxSL_AMUTL (4 << 0) > > +#define TAS5754M_GxSL_AMUTR (5 << 0) > > +#define TAS5754M_GxSL_CLKI (6 << 0) > > +#define TAS5754M_GxSL_SDOUT (7 << 0) > > +#define TAS5754M_GxSL_ANMUL (8 << 0) > > +#define TAS5754M_GxSL_ANMUR (9 << 0) > > +#define TAS5754M_GxSL_PLLLK (10 << 0) > > +#define TAS5754M_GxSL_CPCLK (11 << 0) > > +#define TAS5754M_GxSL_UV0_7 (14 << 0) > > +#define TAS5754M_GxSL_UV0_3 (15 << 0) > > +#define TAS5754M_GxSL_PLLCK (16 << 0) > > + > > +/* Page 1, Register 2 - analog volume control */ > > +#define TAS5754M_RAGN_SHIFT 0 > > +#define TAS5754M_LAGN_SHIFT 4 > > + > > +/* Page 1, Register 7 - analog boost control */ > > +#define TAS5754M_AGBR_SHIFT 0 > > +#define TAS5754M_AGBL_SHIFT 4 > > + > > +#endif > > > > base-commit: f6274b06e326d8471cdfb52595f989a90f5e888f > > -- > > 2.17.1 > >

2 1

[PATCH v5 0/9] Support Spi in i2c-multi-instantiate driver
by Stefan Binding 26 Jan '22

26 Jan '22

Add support for SPI bus in the i2c-multi-instantiate driver as upcoming laptops will need to multi instantiate SPI devices from a single device node, which has multiple SpiSerialBus entries at the ACPI table. With the new SPI support, i2c-multi-instantiate becomes bus-multi-instantiate and is moved to the ACPI folder. The intention is to support the SPI bus by re-using the current I2C multi instantiate, instead of creating a new SPI multi instantiate, to make it possible for peripherals that can be controlled by I2C or SPI to have the same HID at the ACPI table. The new driver (serial multi instantiate, smi) checks for the hard-coded bus type and returns -ENODEV in case of zero devices found for that bus. In the case of automatic bus detection, the driver will give preference to I2C. The expectation is for a device node in the ACPI table to have multiple I2cSerialBus only or multiple SpiSerialBus only, not a mix of both; and for the case where there are both entries in one device node, only the I2C ones would be probed. This new serial multi instantiate will be used in CS35L41 HDA new driver. Changes since V4: - renamed bus-multi-instantiate to serial-multi-instantiate - various minor fixes in serial-multi-instantiate - fix returns in serial-multi-instantiate - removed unnecessary stubs in spi.h - re-order SSIDs in patch_realtek.c Lucas Tanure (4): platform/x86: i2c-multi-instantiate: Rename it for a generic serial driver name platform/x86: serial-multi-instantiate: Reorganize I2C functions ALSA: hda/realtek: Add support for HP Laptops ACPI / scan: Create platform device for CS35L41 Stefan Binding (5): spi: Make spi_alloc_device and spi_add_device public again spi: Create helper API to lookup ACPI info for spi device spi: Support selection of the index of the ACPI Spi Resource before alloc spi: Add API to count spi acpi resources platform/x86: serial-multi-instantiate: Add SPI support MAINTAINERS | 4 +- drivers/acpi/scan.c | 16 +- drivers/platform/x86/Kconfig | 14 +- drivers/platform/x86/Makefile | 2 +- drivers/platform/x86/i2c-multi-instantiate.c | 174 --------- .../platform/x86/serial-multi-instantiate.c | 346 ++++++++++++++++++ drivers/spi/spi.c | 137 ++++++- include/linux/spi/spi.h | 20 + sound/pci/hda/patch_realtek.c | 43 ++- 9 files changed, 549 insertions(+), 207 deletions(-) delete mode 100644 drivers/platform/x86/i2c-multi-instantiate.c create mode 100644 drivers/platform/x86/serial-multi-instantiate.c -- 2.25.1

3 17

[PATCH v2] kselftest: alsa: Add test case for writing invalid values
by Mark Brown 26 Jan '22

26 Jan '22

Attempt to write various invalid values for control types we know about and check that something sensible happens. The ABI isn't quite as clearly defined as one might like, rather than generating an error when an invalid value is written many devices will silently rewrite the value into one that is valid for the control. The exact value chosen is not predictable so in the case the write succeeds we just check that the value we read back is one that is valid for the control. Reviewed-by: Shuah Khan <skhan(a)linuxfoundation.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- v2: - Fix a couple of _integer()/_integer64() confusions. tools/testing/selftests/alsa/mixer-test.c | 222 +++++++++++++++++++++- 1 file changed, 221 insertions(+), 1 deletion(-) diff --git a/tools/testing/selftests/alsa/mixer-test.c b/tools/testing/selftests/alsa/mixer-test.c index 17f158d7a767..0e88f4f3d802 100644 --- a/tools/testing/selftests/alsa/mixer-test.c +++ b/tools/testing/selftests/alsa/mixer-test.c @@ -13,6 +13,7 @@ #include <stdio.h> #include <stdlib.h> #include <stdbool.h> +#include <limits.h> #include <string.h> #include <getopt.h> #include <stdarg.h> @@ -26,7 +27,7 @@ #include "../kselftest.h" -#define TESTS_PER_CONTROL 3 +#define TESTS_PER_CONTROL 4 struct card_data { snd_ctl_t *handle; @@ -679,6 +680,224 @@ void test_ctl_write_valid(struct ctl_data *ctl) ctl->card->card, ctl->elem); } +bool test_ctl_write_invalid_value(struct ctl_data *ctl, + snd_ctl_elem_value_t *val) +{ + int err; + long val_read; + + /* Ideally this will fail... */ + err = snd_ctl_elem_write(ctl->card->handle, val); + if (err < 0) + return false; + + /* ...but some devices will clamp to an in range value */ + err = snd_ctl_elem_read(ctl->card->handle, val); + if (err < 0) { + ksft_print_msg("%s failed to read: %s\n", + ctl->name, snd_strerror(err)); + return true; + } + + return !ctl_value_valid(ctl, val); +} + +bool test_ctl_write_invalid_boolean(struct ctl_data *ctl) +{ + int err, i; + long val_read; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_boolean(val, i, 2); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + } + + return !fail; +} + +bool test_ctl_write_invalid_integer(struct ctl_data *ctl) +{ + int i; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + if (snd_ctl_elem_info_get_min(ctl->info) != LONG_MIN) { + /* Just under range */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer(val, i, + snd_ctl_elem_info_get_min(ctl->info) - 1); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + + /* Minimum representable value */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer(val, i, LONG_MIN); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + } + + if (snd_ctl_elem_info_get_max(ctl->info) != LONG_MAX) { + /* Just over range */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer(val, i, + snd_ctl_elem_info_get_max(ctl->info) + 1); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + + /* Maximum representable value */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer(val, i, LONG_MAX); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + } + } + + return !fail; +} + +bool test_ctl_write_invalid_integer64(struct ctl_data *ctl) +{ + int i; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + if (snd_ctl_elem_info_get_min64(ctl->info) != LLONG_MIN) { + /* Just under range */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer64(val, i, + snd_ctl_elem_info_get_min64(ctl->info) - 1); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + + /* Minimum representable value */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer64(val, i, LLONG_MIN); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + } + + if (snd_ctl_elem_info_get_max64(ctl->info) != LLONG_MAX) { + /* Just over range */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer64(val, i, + snd_ctl_elem_info_get_max64(ctl->info) + 1); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + + /* Maximum representable value */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_integer64(val, i, LLONG_MAX); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + } + } + + return !fail; +} + +bool test_ctl_write_invalid_enumerated(struct ctl_data *ctl) +{ + int err, i; + unsigned int val_read; + bool fail = false; + snd_ctl_elem_value_t *val; + snd_ctl_elem_value_alloca(&val); + + snd_ctl_elem_value_set_id(val, ctl->id); + + for (i = 0; i < snd_ctl_elem_info_get_count(ctl->info); i++) { + /* One beyond maximum */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_enumerated(val, i, + snd_ctl_elem_info_get_items(ctl->info)); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + + /* Maximum representable value */ + snd_ctl_elem_value_copy(val, ctl->def_val); + snd_ctl_elem_value_set_enumerated(val, i, UINT_MAX); + + if (test_ctl_write_invalid_value(ctl, val)) + fail = true; + + } + + return !fail; +} + + +void test_ctl_write_invalid(struct ctl_data *ctl) +{ + bool pass; + int err; + + /* If the control is turned off let's be polite */ + if (snd_ctl_elem_info_is_inactive(ctl->info)) { + ksft_print_msg("%s is inactive\n", ctl->name); + ksft_test_result_skip("write_invalid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + if (!snd_ctl_elem_info_is_writable(ctl->info)) { + ksft_print_msg("%s is not writeable\n", ctl->name); + ksft_test_result_skip("write_invalid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + switch (snd_ctl_elem_info_get_type(ctl->info)) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + pass = test_ctl_write_invalid_boolean(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER: + pass = test_ctl_write_invalid_integer(ctl); + break; + + case SND_CTL_ELEM_TYPE_INTEGER64: + pass = test_ctl_write_invalid_integer64(ctl); + break; + + case SND_CTL_ELEM_TYPE_ENUMERATED: + pass = test_ctl_write_invalid_enumerated(ctl); + break; + + default: + /* No tests for this yet */ + ksft_test_result_skip("write_invalid.%d.%d\n", + ctl->card->card, ctl->elem); + return; + } + + /* Restore the default value to minimise disruption */ + err = write_and_verify(ctl, ctl->def_val, NULL); + if (err < 0) + pass = false; + + ksft_test_result(pass, "write_invalid.%d.%d\n", + ctl->card->card, ctl->elem); +} + int main(void) { struct ctl_data *ctl; @@ -697,6 +916,7 @@ int main(void) test_ctl_get_value(ctl); test_ctl_write_default(ctl); test_ctl_write_valid(ctl); + test_ctl_write_invalid(ctl); } ksft_exit_pass(); -- 2.30.2

2 1

[PATCH] ALSA: hda: Fix UAF of leds class devs at unbinding
by Takashi Iwai 26 Jan '22

26 Jan '22

The LED class devices that are created by HD-audio codec drivers are registered via devm_led_classdev_register() and associated with the HD-audio codec device. Unfortunately, it turned out that the devres release doesn't work for this case; namely, since the codec resource release happens before the devm call chain, it triggers a NULL dereference or a UAF for a stale set_brightness_delay callback. For fixing the bug, this patch changes the LED class device register and unregister in a manual manner without devres, keeping the instances in hda_gen_spec. Reported-by: Alexander Sergeyev <sergeev917(a)gmail.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20220111195229.a77wrpjclqwrx4bx@localhost.localdo… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 17 +++++++++++++++-- sound/pci/hda/hda_generic.h | 3 +++ 2 files changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3bf5e3410703..fc114e522480 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -91,6 +91,12 @@ static void snd_hda_gen_spec_free(struct hda_gen_spec *spec) free_kctls(spec); snd_array_free(&spec->paths); snd_array_free(&spec->loopback_list); +#ifdef CONFIG_SND_HDA_GENERIC_LEDS + if (spec->led_cdevs[LED_AUDIO_MUTE]) + led_classdev_unregister(spec->led_cdevs[LED_AUDIO_MUTE]); + if (spec->led_cdevs[LED_AUDIO_MICMUTE]) + led_classdev_unregister(spec->led_cdevs[LED_AUDIO_MICMUTE]); +#endif } /* @@ -3922,7 +3928,10 @@ static int create_mute_led_cdev(struct hda_codec *codec, enum led_brightness), bool micmute) { + struct hda_gen_spec *spec = codec->spec; struct led_classdev *cdev; + int idx = micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE; + int err; cdev = devm_kzalloc(&codec->core.dev, sizeof(*cdev), GFP_KERNEL); if (!cdev) @@ -3932,10 +3941,14 @@ static int create_mute_led_cdev(struct hda_codec *codec, cdev->max_brightness = 1; cdev->default_trigger = micmute ? "audio-micmute" : "audio-mute"; cdev->brightness_set_blocking = callback; - cdev->brightness = ledtrig_audio_get(micmute ? LED_AUDIO_MICMUTE : LED_AUDIO_MUTE); + cdev->brightness = ledtrig_audio_get(idx); cdev->flags = LED_CORE_SUSPENDRESUME; - return devm_led_classdev_register(&codec->core.dev, cdev); + err = led_classdev_register(&codec->core.dev, cdev); + if (err < 0) + return err; + spec->led_cdevs[idx] = cdev; + return 0; } /** diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 8e1bc8ea74fc..34eba40cc6e6 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -294,6 +294,9 @@ struct hda_gen_spec { struct hda_jack_callback *cb); void (*mic_autoswitch_hook)(struct hda_codec *codec, struct hda_jack_callback *cb); + + /* leds */ + struct led_classdev *led_cdevs[NUM_AUDIO_LEDS]; }; /* values for add_stereo_mix_input flag */ -- 2.31.1

1 0

Re: No sound on gpd pocket 3
by Pierre-Louis Bossart 26 Jan '22

26 Jan '22

On 1/24/22 12:35, anthony tonitch wrote: > HO, I'm really sorry, I did this alsa-info from my main computer over > ssh and I might have typed the command on the wrong terminal :'( > here it is http://alsa-project.org/db/?f=8ec2e745ceafb1a659d71eff99ad3a9eda3eb62d ah, now it's starting to make sense. I am afraid you have the ES8316 codec, or one of its variants. /sys/bus/acpi/devices/ESSX8326:00/status 15 so no you don't want to use options snd-intel-dspcfg dsp_driver=1 but options snd-intel-dspcfg dsp_driver=3 The auto-detection don't work because we use a different codec HID for the driver selection { .flags = FLAG_SOF, .device = 0x5a98, .codec_hid = "ESSX8336", }, as well as the machine driver selection { .id = "ESSX8336", .drv_name = "sof-essx8336", .sof_tplg_filename = "sof-glk-es8336.tplg", }, This is really a problematic set of devices, where absolutely all possible I2S links have been used, along with GPIO settings and DMICs. Please file a bug on https://github.com/thesofproject/linux/issues so that we can keep track of this and suggest solution. Thanks! > Le lun. 24 janv. 2022 à 18:11, Pierre-Louis Bossart > <pierre-louis.bossart(a)linux.intel.com> a écrit : >> >> >> >> On 1/24/22 10:52, anthony tonitch wrote: >>> Hello, >>> >>> I have a problem I guess related to sof... I already asked on the irc >>> #alsa(a)libera.chat and gnarface advised me to come and ask on this >>> mailing list! >> >> using "options snd-intel-dspcfg dsp_driver=1" to avoid using SOF isn't >> very helpful on an AMD platform :-) >> >> 08:00.1 Audio device [0403]: NVIDIA Corporation GP104 High Definition >> Audio Controller [10de:10f0] (rev a1) >> Subsystem: PNY Device [196e:11a0] >> 0a:00.3 Audio device [0403]: Advanced Micro Devices, Inc. [AMD] Family >> 17h (Models 00h-0fh) HD Audio Controller [1022:1457] >> Subsystem: ASUSTeK Computer Inc. Device [1043:86c7] >> >> Best of luck! >> >>> >>> I am struggling to make the sound of my gpd pocket 3 to work on linux >>> (archlinux to be precise)! In fact, the sound is not supposed to work >>> out of the box as other users tested it before me! but they all had >>> the solution to set the dsp_driver and this didn't worked for me! >>> I checked several times to see if the module was loaded properly and >>> yes it is (as you can also check in the alsa-info at the end of the mail)! >>> >>> If I do aplay -l I get a bunch of hdmi connection that is not working >>> at all! I'm really lost here and I don't even know what to look for >>> anymore >>> >>> The sound works great on windows so This doens't seems to be hardware problem ?! >>> and I tried to plug an usb headset which give me sound so I doesn't >>> seems to be config related either ?! (tho the jack doesn't work) >>> >>> I asked on several places already. I post the different links here in >>> case you want to check but I don't have any response to any post >>> unfortunately! >>> >>> - Post on Archlinux BBS: https://bbs.archlinux.org/viewtopic.php?id=273044 >>> - Post on subreddit GPDPocket: >>> https://www.reddit.com/r/GPDPocket/comments/s31qi8/gpd_p3_no_sound_linux/ >>> >>> - Screenshot and bios picture of maybe relevant informations: >>> - https://imgur.com/a/DouSB3m >>> - https://imgur.com/a/K232PLJ >>> >>> I hope there is enough informations to help me debug this but if you >>> need anything I can provide it! this is a small pc and I'm usualy >>> quarrying it around! >>> I know this is better to send mail but if you need me to chat for any >>> reason (it would be easier for instance) I'm usually hanging on the >>> libera.chat irc as tonitch you can pm or tag me on #alsa >>> >>> Here is the alsa-info.sh : >>> http://alsa-project.org/db/?f=6ddb669a19086cebdb121c97c25bbdccb98e856d >>> >>> Thanks a lot for your help! >>> >>> PS: Ubuntu mate have released their official pocket 3 distribution and >>> I tried it but the sound didn't worked either, here is the alsa-info >>> In case this help ?! : >>> http://alsa-project.org/db/?f=df4c183fd00cec224006ec8caa228705399ad873

2 4

[PATCH] ASoC: Intel: atom: Remove redundant check to simplify the code
by Tang Bin 25 Jan '22

25 Jan '22

In the function sst_platform_get_resources(), if platform_get_irq() failed, the return should not be zero, as the example in platform.c is * int irq = platform_get_irq(pdev, 0) * if (irq < 0) * return irq; So remove the redundant check to simplify the code. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3be64430c..696d547c5 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -226,8 +226,8 @@ static int sst_platform_get_resources(struct intel_sst_drv *ctx) /* Find the IRQ */ ctx->irq_num = platform_get_irq(pdev, ctx->pdata->res_info->acpi_ipc_irq_index); - if (ctx->irq_num <= 0) - return ctx->irq_num < 0 ? ctx->irq_num : -EIO; + if (ctx->irq_num < 0) + return ctx->irq_num; return 0; } -- 2.20.1.windows.1

5 7

[PATCH 0/3] ASoC: sh: rz-ssi: Trivial changes
by Lad Prabhakar 25 Jan '22

25 Jan '22

Hi All, This patch series has trivial changes for rz-ssi driver. Patches 1 (partial) & 2 are from series [3]. patch 1/3 is just a cosmetic change which was part of patch [0] where it uses a do-while instead of while-do. Patch 2/3 is from series [1] where a helper function is added and patch 3/3 is a new patch which removes duplicate macros. All the patches apply to [2] (sound -next) [0] https://patchwork.kernel.org/project/alsa-devel/patch/ 20220115012303.29651-2-prabhakar.mahadev-lad.rj(a)bp.renesas.com/ [1] https://patchwork.kernel.org/project/alsa-devel/patch/ 20220115012303.29651-6-prabhakar.mahadev-lad.rj(a)bp.renesas.com/ [2] https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/log/ [3] https://patchwork.kernel.org/project/alsa-devel/list/?series=605672 Cheers, Prabhakar Lad Prabhakar (3): ASoC: sh: rz-ssi: Use a do-while loop in rz_ssi_pio_recv() ASoC: sh: rz-ssi: Add rz_ssi_set_substream() helper function ASoC: sh: rz-ssi: Remove duplicate macros sound/soc/sh/rz-ssi.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) -- 2.17.1

3 5

[PATCH v1 0/3] ASoC: Add bounds checking for written values
by Mark Brown 25 Jan '22

25 Jan '22

This series adds verification that values written to registers are in bounds for controls since the core doesn't validate for us. Mark Brown (3): ASoC: ops: Reject out of bounds values in snd_soc_put_volsw() ASoC: ops: Reject out of bounds values in snd_soc_put_volsw_sx() ASoC: ops: Reject out of bounds values in snd_soc_put_xr_sx() sound/soc/soc-ops.c | 29 ++++++++++++++++++++++++++--- 1 file changed, 26 insertions(+), 3 deletions(-) base-commit: e783362eb54cd99b2cac8b3a9aeac942e6f6ac07 -- 2.30.2

3 10

Re: [PATCH 5/5] spi: make remove callback a void function
by Jérôme Pouiller 25 Jan '22

25 Jan '22

On Sunday 23 January 2022 18:52:01 CET Uwe Kleine-König wrote: > > The value returned by an spi driver's remove function is mostly ignored. > (Only an error message is printed if the value is non-zero that the > error is ignored.) > > So change the prototype of the remove function to return no value. This > way driver authors are not tempted to assume that passing an error to > the upper layer is a good idea. All drivers are adapted accordingly. > There is no intended change of behaviour, all callbacks were prepared to > return 0 before. > > Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> [...] > diff --git a/drivers/staging/wfx/bus_spi.c b/drivers/staging/wfx/bus_spi.c > index 55ffcd7c42e2..fa0ff66a457d 100644 > --- a/drivers/staging/wfx/bus_spi.c > +++ b/drivers/staging/wfx/bus_spi.c > @@ -232,12 +232,11 @@ static int wfx_spi_probe(struct spi_device *func) > return wfx_probe(bus->core); > } > > -static int wfx_spi_remove(struct spi_device *func) > +static void wfx_spi_remove(struct spi_device *func) > { > struct wfx_spi_priv *bus = spi_get_drvdata(func); > > wfx_release(bus->core); > - return 0; > } > > /* For dynamic driver binding, kernel does not use OF to match driver. It only For the wfx driver: Acked-by: Jérôme Pouiller <jerome.pouiller(a)silabs.com> -- Jérôme Pouiller

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