December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4 00/11] soundwire: update ASoC interfaces
by Pierre-Louis Bossart 11 Dec '19

11 Dec '19

We need new fields in existing structures to a) deal with race conditions on codec probe/enumeration b) allow for multi-step ACPI scan/probe/startup on Intel plaforms c) deal with MSI issues using a single handler/threads for all audio interrupts d) deal with access to registers shared across multiple links on Intel platforms These structures for a) will be used by the SOF driver as well as codec drivers. The b) c) and d) cases are only for the Intel-specific implementation. To avoid conflicts between ASoC and Soundwire trees, these 11 patches are provided out-of-order, before the functionality enabled in these header files is added in follow-up patch series which can be applied separately in the ASoC and Soundwire trees. As discussed earlier, Vinod would need to provide an immutable tag for Mark Brown, and the integration on the ASoC side of SOF changes and new codecs drivers can proceed in parallel with SoundWire core changes. I had multiple offline discussions with Vinod/Mark/Takashi on how to proceed withe volume of SoundWire changes. Now that v5.5-rc1 is out we should go ahead with these interface changes. The next patchset has not changed, the series "[PATCH v3 00/15] soundwire: intel: implement new ASoC interfaces" can still be reviewed. An update for the series "[PATCH v3 00/22] soundwire: code hardening and suspend-resume support" will be provided later this week (one last minute issue to fix) Comments and feedback welcome ~Pierre Changes since v3: Reordered code and added kernel doc comments Added prototypes and fields needed to deal with SoundWire interrupts in a single handler/thread, following what is already done on the DSP side. Added mutex to control access to registers shared across links Added initial definitions for clock stop support on Intel platforms. Depending on power and latency requirements, different "quirks" can be supported. Changes since v2: Added new field to deal with a race condition leading to a timeout when the codec goes through a pm_runtime suspend/resume transition while the Master remains active. Clarified commit messages with detailed explanations what those race conditions are and why the changes were introduced. Reordered fields for Intel routines Added kernel-doc definitions for structures Modified the patch subjects to make the mapping between interface definition and implementation straightforward. Changes since v1 (no feedback received since October 23) additional initialization_complete utility to help codec drivers with their resume operation, waiting for the enumeration to complete is not always enough. Bard Liao (2): soundwire: intel: update headers for interrupts soundwire: intel: add link_list to handle interrupts with a single thread Pierre-Louis Bossart (7): soundwire: sdw_slave: add probe_complete structure and new fields soundwire: sdw_slave: add enumeration_complete structure soundwire: sdw_slave: add initialization_complete definition soundwire: sdw_slave: track unattach_request to handle all init sequences soundwire: intel: update interfaces between ASoC and SoundWire soundwire: intel: add mutex for shared SHIM register access soundwire: intel: add clock stop quirks Rander Wang (2): soundwire: intel: update stream callbacks for hwparams/free stream operations soundwire: intel: add prototype for WAKEEN interrupt processing drivers/soundwire/intel.c | 20 ++-- drivers/soundwire/intel.h | 13 ++- drivers/soundwire/intel_init.c | 31 ++---- include/linux/soundwire/sdw.h | 19 ++++ include/linux/soundwire/sdw_intel.h | 154 ++++++++++++++++++++++++++-- 5 files changed, 194 insertions(+), 43 deletions(-) base-commit: e42617b825f8073569da76dc4510bfa019b1c35a -- 2.20.1

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Re: [alsa-devel] [v6, 5/6] ASoC: sgtl5000: Fix of unmute outputs on probe
by Alison Wang 11 Dec '19

11 Dec '19

Hi, Oleksandr, I have a question about this patch. Could you give me some ideas? Thanks. In this patch, snd_soc_component_update_bits is used instead of snd_soc_component_write. Although EN_HP_ZCD and EN_ADC_ZCD are enabled in ANA_CTRL register, MUTE_LO, MUTE_HP and MUTE_ADC bits are remained as the default value. It causes LO, HP and ADC are all muted after driver probe. BTW, in the original code, snd_soc_component_write is used and MUTE_LO, MUTE_HP and MUTE_ADC are all set as zero (unmute). I saw the above phenomenon on the latest linux-next. For LO and HP are muted, no sound can be heard. Best Regards, Alison Wang > Subject: [v6,5/6] ASoC: sgtl5000: Fix of unmute outputs on probe > > To enable "zero cross detect" for ADC/HP, change HP_ZCD_EN/ADC_ZCD_EN > bits only instead of writing the whole CHIP_ANA_CTRL register. > > Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> > Reviewed-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> > Reviewed-by: Igor Opaniuk <igor.opaniuk(a)toradex.com> > Reviewed-by: Fabio Estevam <festevam(a)gmail.com> > > --- > > Changes in v6: None > Changes in v5: None > Changes in v4: None > Changes in v3: None > Changes in v2: > - Fix patch formatting > > sound/soc/codecs/sgtl5000.c | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > > > diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index > b65232521ea8..23f4ae2f0723 100644 > --- a/sound/soc/codecs/sgtl5000.c > +++ b/sound/soc/codecs/sgtl5000.c > @@ -1453,6 +1453,7 @@ static int sgtl5000_probe(struct > snd_soc_component *component) > int ret; > u16 reg; > struct sgtl5000_priv *sgtl5000 = > snd_soc_component_get_drvdata(component); > + unsigned int zcd_mask = SGTL5000_HP_ZCD_EN | > SGTL5000_ADC_ZCD_EN; > > /* power up sgtl5000 */ > ret = sgtl5000_set_power_regs(component); > @@ -1480,9 +1481,8 @@ static int sgtl5000_probe(struct > snd_soc_component *component) > 0x1f); > snd_soc_component_write(component, SGTL5000_CHIP_PAD_STRENGTH, > reg); > > - snd_soc_component_write(component, SGTL5000_CHIP_ANA_CTRL, > - SGTL5000_HP_ZCD_EN | > - SGTL5000_ADC_ZCD_EN); > + snd_soc_component_update_bits(component, > SGTL5000_CHIP_ANA_CTRL, > + zcd_mask, zcd_mask); > > snd_soc_component_update_bits(component, > SGTL5000_CHIP_MIC_CTRL, > SGTL5000_BIAS_R_MASK, >

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[alsa-devel] [PATCH] ALSA: hda/realtek - Line-out jack doesn't work on a Dell AIO
by Hui Wang 11 Dec '19

11 Dec '19

After applying the fixup ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, the Line-out jack works well. And instead of adding a new set of pin definition in the pin_fixup_tbl, we put a more generic matching entry in the fallback_pin_fixup_tbl. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d6e34b3b3aa..dbfafee97931 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7643,11 +7643,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x90a70130}, {0x1b, 0x90170110}, {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, - {0x12, 0xb7a60130}, - {0x13, 0xb8a61140}, - {0x16, 0x90170110}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, @@ -7841,6 +7836,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + {0x19, 0x40000000}, + {0x1a, 0x40000000}), {} }; -- 2.17.1

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[alsa-devel] [alsa-lib][PATCH] ucm: setup conf_format after getting ALSA_CONFIG_UCM_VAR successfully
by libin.yang＠linux.intel.com 11 Dec '19

11 Dec '19

From: Libin Yang <libin.yang(a)linux.intel.com> Set the conf_format to 1 after getting ALSA_CONFIG_UCM_VAR successfully. Otherwise, the conf_format is not set in this scenario. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- src/ucm/parser.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index ba50027..d61124a 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -89,6 +89,8 @@ static void configuration_filename(snd_use_case_mgr_t *uc_mgr, env = getenv(ALSA_CONFIG_UCM2_VAR); if (env == NULL) { env = getenv(ALSA_CONFIG_UCM_VAR); + if (env) + uc_mgr->conf_format = 1; } else { uc_mgr->conf_format = 2; } -- 2.7.4

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[alsa-devel] [PATCH 00/23] ALSA: Drop superfluous ioctl PCM ops (for 5.6)
by Takashi Iwai 11 Dec '19

11 Dec '19

Hi, this is another series of patches targeted to 5.6 kernel. This one is applied on top of the previous series for managed allocation, and again specific to non-ASoC drivers in sound/*. For ASoC and other subsystems, another patchset will be submitted later. The changes are pretty simple, just removing the snd_pcm_lib_ioctl entries that became superfluous. Takashi === Takashi Iwai (23): ALSA: aoa: Drop superfluous ioctl PCM ops ALSA: arm: Drop superfluous ioctl PCM ops ALSA: atmel: Drop superfluous ioctl PCM ops ALSA: drivers: Drop superfluous ioctl PCM ops ALSA: firewire: Drop superfluous ioctl PCM ops ALSA: es1688: Drop superfluous ioctl PCM ops ALSA: isa: Drop superfluous ioctl PCM ops ALSA: mips: Drop superfluous ioctl PCM ops ALSA: parisc: Drop superfluous ioctl PCM ops ALSA: intel8x0: Drop superfluous ioctl PCM ops ALSA: via82xx: Drop superfluous ioctl PCM ops ALSA: atiixp: Drop superfluous ioctl PCM ops ALSA: asihpi: Drop superfluous ioctl PCM ops ALSA: trident: Drop superfluous ioctl PCM ops ALSA: pci: Drop superfluous ioctl PCM ops ALSA: pcmcia: Drop superfluous ioctl PCM ops ALSA: ppc: Drop superfluous ioctl PCM ops ALSA: sh: Drop superfluous ioctl PCM ops ALSA: sparc: Drop superfluous ioctl PCM ops ALSA: spi: Drop superfluous ioctl PCM ops ALSA: usb: Drop superfluous ioctl PCM ops ALSA: x86: Drop superfluous ioctl PCM ops ALSA: xen: Drop superfluous ioctl PCM ops sound/aoa/soundbus/i2sbus/pcm.c | 2 -- sound/arm/aaci.c | 2 -- sound/arm/pxa2xx-ac97.c | 1 - sound/atmel/ac97c.c | 2 -- sound/drivers/aloop.c | 1 - sound/drivers/dummy.c | 2 -- sound/drivers/ml403-ac97cr.c | 2 -- sound/drivers/pcsp/pcsp_lib.c | 1 - sound/drivers/vx/vx_pcm.c | 2 -- sound/firewire/bebob/bebob_pcm.c | 2 -- sound/firewire/dice/dice-pcm.c | 2 -- sound/firewire/digi00x/digi00x-pcm.c | 2 -- sound/firewire/fireface/ff-pcm.c | 2 -- sound/firewire/fireworks/fireworks_pcm.c | 2 -- sound/firewire/isight.c | 1 - sound/firewire/motu/motu-pcm.c | 2 -- sound/firewire/oxfw/oxfw-pcm.c | 2 -- sound/firewire/tascam/tascam-pcm.c | 2 -- sound/isa/ad1816a/ad1816a_lib.c | 2 -- sound/isa/es1688/es1688_lib.c | 8 -------- sound/isa/es18xx.c | 2 -- sound/isa/gus/gus_pcm.c | 2 -- sound/isa/msnd/msnd.c | 2 -- sound/isa/sb/emu8000_pcm.c | 1 - sound/isa/sb/sb16_main.c | 2 -- sound/isa/sb/sb8_main.c | 2 -- sound/isa/wss/wss_lib.c | 2 -- sound/mips/hal2.c | 2 -- sound/mips/sgio2audio.c | 3 --- sound/parisc/harmony.c | 2 -- sound/pci/ad1889.c | 2 -- sound/pci/ali5451/ali5451.c | 4 ---- sound/pci/als300.c | 2 -- sound/pci/als4000.c | 2 -- sound/pci/asihpi/asihpi.c | 17 ----------------- sound/pci/atiixp.c | 3 --- sound/pci/atiixp_modem.c | 2 -- sound/pci/au88x0/au88x0_pcm.c | 1 - sound/pci/aw2/aw2-alsa.c | 2 -- sound/pci/azt3328.c | 3 --- sound/pci/bt87x.c | 1 - sound/pci/ca0106/ca0106_main.c | 8 -------- sound/pci/cmipci.c | 5 ----- sound/pci/cs4281.c | 2 -- sound/pci/cs46xx/cs46xx_lib.c | 10 ---------- sound/pci/cs5535audio/cs5535audio_pcm.c | 2 -- sound/pci/ctxfi/ctpcm.c | 2 -- sound/pci/echoaudio/echoaudio.c | 4 ---- sound/pci/emu10k1/emu10k1x.c | 2 -- sound/pci/emu10k1/emupcm.c | 6 ------ sound/pci/emu10k1/p16v.c | 2 -- sound/pci/ens1370.c | 3 --- sound/pci/es1938.c | 2 -- sound/pci/es1968.c | 2 -- sound/pci/fm801.c | 2 -- sound/pci/hda/hda_controller.c | 1 - sound/pci/ice1712/ice1712.c | 5 ----- sound/pci/ice1712/ice1724.c | 5 ----- sound/pci/intel8x0.c | 12 ------------ sound/pci/intel8x0m.c | 2 -- sound/pci/lola/lola_pcm.c | 1 - sound/pci/lx6464es/lx6464es.c | 2 -- sound/pci/maestro3.c | 2 -- sound/pci/mixart/mixart.c | 2 -- sound/pci/nm256/nm256.c | 2 -- sound/pci/oxygen/oxygen_pcm.c | 6 ------ sound/pci/pcxhr/pcxhr.c | 1 - sound/pci/riptide/riptide.c | 2 -- sound/pci/rme32.c | 8 -------- sound/pci/rme96.c | 4 ---- sound/pci/sis7019.c | 2 -- sound/pci/sonicvibes.c | 2 -- sound/pci/trident/trident_main.c | 31 ------------------------------- sound/pci/via82xx.c | 5 ----- sound/pci/via82xx_modem.c | 2 -- sound/pci/ymfpci/ymfpci_main.c | 5 ----- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 - sound/ppc/pmac.c | 2 -- sound/ppc/snd_ps3.c | 1 - sound/sh/aica.c | 1 - sound/sh/sh_dac_audio.c | 1 - sound/sparc/amd7930.c | 2 -- sound/sparc/cs4231.c | 2 -- sound/sparc/dbri.c | 1 - sound/spi/at73c213.c | 1 - sound/usb/6fire/pcm.c | 1 - sound/usb/caiaq/audio.c | 1 - sound/usb/hiface/pcm.c | 1 - sound/usb/line6/capture.c | 1 - sound/usb/line6/playback.c | 1 - sound/usb/misc/ua101.c | 2 -- sound/usb/pcm.c | 2 -- sound/usb/usx2y/usbusx2yaudio.c | 1 - sound/usb/usx2y/usx2yhwdeppcm.c | 1 - sound/x86/intel_hdmi_audio.c | 1 - sound/xen/xen_snd_front_alsa.c | 2 -- 96 files changed, 281 deletions(-) -- 2.16.4

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[alsa-devel] [PATCH AUTOSEL 4.4 56/71] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 11 Dec '19

11 Dec '19

From: Ben Zhang <benzh(a)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(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)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 69d987a9935c9..90f8173123f6c 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -295,6 +295,7 @@ 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_PLL1_CTRL2: -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 33/71] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)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(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)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 7ea201c05e5da..d0d6dfbfcfdf8 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 ef8955abd9186..3e3277100f08a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1310,8 +1310,11 @@ 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 4.9 69/91] ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile
by Sasha Levin 11 Dec '19

11 Dec '19

From: Ben Zhang <benzh(a)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(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Link: https://lore.kernel.org/r/20191106011335.223061-6-cujomalainey@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)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 65ac4518ad060..49ab26e69f2fc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -305,6 +305,7 @@ 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 4.9 50/91] ALSA: timer: Limit max amount of slave instances
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)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(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)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 e944d27f79c34..f8a4b2a2f8f6b 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -87,6 +87,9 @@ 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); @@ -265,6 +268,10 @@ 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; @@ -274,6 +281,7 @@ 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); @@ -363,6 +371,8 @@ 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 4.9 43/91] ALSA: hda - Fix pending unsol events at shutdown
by Sasha Levin 11 Dec '19

11 Dec '19

From: Takashi Iwai <tiwai(a)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(a)chris-wilson.co.uk> Link: https://lore.kernel.org/r/s5h1ruxt9cz.wl-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)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 7ea201c05e5da..d0d6dfbfcfdf8 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 f2f1d9fd848c8..3234e9ca02cec 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1275,8 +1275,11 @@ 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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