[alsa-devel] [PATCH AUTOSEL 4.14 060/130] ALSA: hda - Fix pending unsol events at shutdown

[alsa-devel] [PATCH AUTOSEL 4.14 060/130] ALSA: hda - Fix pending unsol events at shutdown

[Sasha Levin]

From: Takashi Iwai <tiwai@suse.de>

[ Upstream commit ca58f55108fee41d87c9123f85ad4863e5de7f45 ]

This is an alternative fix attemp for the issue reported in the commit
caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling") that was
reverted later due to regressions.  Instead of tweaking the hardware
disablement order and the enforced irq flushing, do calling
cancel_work_sync() of the unsol work early enough, and explicitly
ignore the unsol events during the shutdown by checking the
bus->shutdown flag.

Fixes: caa8422d01e9 ("ALSA: hda: Flush interrupts on disabling")
Cc: Chris Wilson <chris@chris-wilson.co.uk>
Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/pci/hda/hda_bind.c  | 4 ++++
 sound/pci/hda/hda_intel.c | 3 +++
 2 files changed, 7 insertions(+)

diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 8db1890605f60..c175b2cf63f77 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -42,6 +42,10 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev)
 {
 	struct hda_codec *codec = container_of(dev, struct hda_codec, core);
 
+	/* ignore unsol events during shutdown */
+	if (codec->bus->shutdown)
+		return;
+
 	if (codec->patch_ops.unsol_event)
 		codec->patch_ops.unsol_event(codec, ev);
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 96e9b3944b925..890793ad85ca1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1450,8 +1450,11 @@ static int azx_free(struct azx *chip)
 static int azx_dev_disconnect(struct snd_device *device)
 {
 	struct azx *chip = device->device_data;
+	struct hdac_bus *bus = azx_bus(chip);
 
 	chip->bus.shutdown = 1;
+	cancel_work_sync(&bus->unsol_work);
+
 	return 0;
 }
 
-- 
2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 080/130] ALSA: timer: Limit max amount of slave instances

[Sasha Levin]

From: Takashi Iwai <tiwai@suse.de>

[ Upstream commit fdea53fe5de532969a332d6e5e727f2ad8bf084d ]

The fuzzer tries to open the timer instances as much as possible, and
this may cause a system hiccup easily.  We've already introduced the
cap for the max number of available instances for the h/w timers, and
we should put such a limit also to the slave timers, too.

This patch introduces the limit to the multiple opened slave timers.
The upper limit is hard-coded to 1000 for now, which should suffice
for any practical usages up to now.

Link: https://lore.kernel.org/r/20191106154257.5853-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/core/timer.c | 10 ++++++++++
 1 file changed, 10 insertions(+)

diff --git a/sound/core/timer.c b/sound/core/timer.c
index c60dfd52e8a6d..22589a073423b 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -88,6 +88,9 @@ static LIST_HEAD(snd_timer_slave_list);
 /* lock for slave active lists */
 static DEFINE_SPINLOCK(slave_active_lock);
 
+#define MAX_SLAVE_INSTANCES	1000
+static int num_slaves;
+
 static DEFINE_MUTEX(register_mutex);
 
 static int snd_timer_free(struct snd_timer *timer);
@@ -266,6 +269,10 @@ int snd_timer_open(struct snd_timer_instance **ti,
 			err = -EINVAL;
 			goto unlock;
 		}
+		if (num_slaves >= MAX_SLAVE_INSTANCES) {
+			err = -EBUSY;
+			goto unlock;
+		}
 		timeri = snd_timer_instance_new(owner, NULL);
 		if (!timeri) {
 			err = -ENOMEM;
@@ -275,6 +282,7 @@ int snd_timer_open(struct snd_timer_instance **ti,
 		timeri->slave_id = tid->device;
 		timeri->flags |= SNDRV_TIMER_IFLG_SLAVE;
 		list_add_tail(&timeri->open_list, &snd_timer_slave_list);
+		num_slaves++;
 		err = snd_timer_check_slave(timeri);
 		if (err < 0) {
 			snd_timer_close_locked(timeri, &card_dev_to_put);
@@ -364,6 +372,8 @@ static int snd_timer_close_locked(struct snd_timer_instance *timeri,
 	struct snd_timer_instance *slave, *tmp;
 
 	list_del(&timeri->open_list);
+	if (timeri->flags & SNDRV_TIMER_IFLG_SLAVE)
+		num_slaves--;
 
 	/* force to stop the timer */
 	snd_timer_stop(timeri);
-- 
2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 103/130] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile

[Sasha Levin]

From: Ben Zhang <benzh@chromium.org>

[ Upstream commit eabf424f7b60246c76dcb0ea6f1e83ef9abbeaa6 ]

The codec dies when RT5677_PWR_ANLG2(MX-64h) is set to 0xACE1
while it's streaming audio over SPI. The DSP firmware turns
on PLL2 (MX-64 bit 8) when SPI streaming starts.  However regmap
does not believe that register can change by itself. When
BST1 (bit 15) is turned on with regmap_update_bits(), it doesn't
read the register first before write, so PLL2 power bit is
cleared by accident.

Marking MX-64h as volatile in regmap solved the issue.

Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/codecs/rt5677.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c
index 1cd20b88a3a9c..82ee8f4b965be 100644
--- a/sound/soc/codecs/rt5677.c
+++ b/sound/soc/codecs/rt5677.c
@@ -297,6 +297,7 @@ static bool rt5677_volatile_register(struct device *dev, unsigned int reg)
 	case RT5677_I2C_MASTER_CTRL7:
 	case RT5677_I2C_MASTER_CTRL8:
 	case RT5677_HAP_GENE_CTRL2:
+	case RT5677_PWR_ANLG2: /* Modified by DSP firmware */
 	case RT5677_PWR_DSP_ST:
 	case RT5677_PRIV_DATA:
 	case RT5677_ASRC_22:
-- 
2.20.1

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[alsa-devel] [PATCH AUTOSEL 4.14 104/130] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint

[Sasha Levin]

From: Yu-Hsuan Hsu <yuhsuan@chromium.org>

[ Upstream commit e2db787bdcb4f2722ecf410168f0583764634e45 ]

On KBL platform, the microphone is attached to external codec(rt5514)
instead of PCH. However, TDM slot between PCH and codec is 16 bits only.
In order to avoid setting wrong format, we should add a constraint to
force to use 16 bits format forever.

Signed-off-by: Yu-Hsuan Hsu <yuhsuan@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190923162940.199580-1-yuhsuan@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
---
 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 41cb1fefbd427..405196283688c 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -422,6 +422,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream)
 	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 			dmic_constraints);
 
+	runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16);
+
 	return snd_pcm_hw_constraint_list(substream->runtime, 0,
 			SNDRV_PCM_HW_PARAM_RATE, &constraints_rates);
 }
-- 
2.20.1

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