March 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH alsa-lib] conf: Fix invalid free at parse_args()
by Takashi Iwai 18 Mar '21

18 Mar '21

The previous fix for memory leaks introduced a few regression. The major one is the assert hit in the error path reaching with NULL or uninitialized sub object. Also, in other code paths, it's possible that an already released sub object gets freed again. Fix those bugs by initializing the sub object properly and add a NULL check before calling snd_config_delete(). Fixes: ad5f255b4767 ("conf: fix memory leak on the error path in parse_args()") Reported-and-tested-by: Mark Hills <mark(a)xwax.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/conf.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/src/conf.c b/src/conf.c index 14b14b597e16..1bcd65c87b94 100644 --- a/src/conf.c +++ b/src/conf.c @@ -5080,6 +5080,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) const char *new = str; const char *tmp; char *val = NULL; + + sub = NULL; err = parse_arg(&new, &varlen, &val); if (err < 0) goto _err; @@ -5104,6 +5106,7 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) err = snd_config_search(subs, var, &sub); if (err >= 0) snd_config_delete(sub); + sub = NULL; err = snd_config_search(def, "type", &typ); if (err < 0) { _invalid_type: @@ -5169,7 +5172,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) err = snd_config_add(subs, sub); if (err < 0) { _err: - snd_config_delete(sub); + if (sub) + snd_config_delete(sub); free(val); return err; } -- 2.26.2

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ASoC topology loading vs card bind
by Srinivas Kandagatla 18 Mar '21

18 Mar '21

Hi All, Am trying to load a FrontEnd/pcm dai definition from ASoC topology however I hit a catch 22 situation here. Topology is loaded as part of component probe() but component probe is only called as part of sound-card bind(). Any pointers on how is this supposed to work? Thanks, srini

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[PATCH 0/3] Fix SND_HDA_PREALLOC issue
by Amadeusz Sławiński 18 Mar '21

18 Mar '21

For context it started with user reporting failures when running arecord without any error or warning in dmesg (after fixing some configuration problems thet they had). https://bugzilla.kernel.org/show_bug.cgi?id=201251#c279 After spending time investigating the issue it was narrowed to quite big setting of CONFIG_SND_HDA_PREALLOC_SIZE (4096). When looking at code https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… there is a limit of memory per card: max_alloc_per_card = 32UL * 1024UL * 1024UL When SND_HDA_PREALLOC_SIZE is set to 4096 it only has memory to alloc for 8 frontends, while Skylake HDA card has 10 of them (6 playback and 4 capture), so preallocated memory is exhausted while probing. In consequence 2 of FEs end without allocated memory. It can be workarounded on user side with setting SND_HDA_PREALLOC_SIZE to lower value, other is changing memory limit per card. However in order to not waste user memory, change maximum allocation size on HDA controller to 4MB and force automatical memory allocation insted of preallocated one. First patch adds prints, so similar issues can be easily identified in the future. Second changes maximum size of hda buffer to reasonable value of 4MB. And last one reverts patch which allowed setting prealloc size on X86 platforms. Amadeusz Sławiński (3): ALSA: pcm: Add debug print on memory allocation failure ALSA: hda: Change AZX_MAX_BUF_SIZE from 1GB to 4MB ALSA: hda: Revert "ALSA: hda: Allow setting preallocation again for x86" include/sound/hda_register.h | 4 ++-- sound/core/pcm_memory.c | 8 ++++++++ sound/hda/Kconfig | 7 +++---- 3 files changed, 13 insertions(+), 6 deletions(-) -- 2.25.1

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[PATCH] ALSA: usb-audio: Fix unintentional sign extension issue
by Colin King 18 Mar '21

18 Mar '21

From: Colin Ian King <colin.king(a)canonical.com> The shifting of the u8 integer device by 24 bits to the left will be promoted to a 32 bit signed int and then sign-extended to a 64 bit unsigned long. In the event that the top bit of device is set then all then all the upper 32 bits of the unsigned long will end up as also being set because of the sign-extension. Fix this by casting device to an unsigned long before the shift. Addresses-Coverity: ("Unintended sign extension") Fixes: a07df82c7990 ("ALSA: usb-audio: Add DJM750 to Pioneer mixer quirk") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer_quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 5b77e8140ba6..f13cdfd8bd20 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2926,7 +2926,7 @@ static int snd_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_v u8 group = (private_value & SND_DJM_GROUP_MASK) >> SND_DJM_GROUP_SHIFT; u16 value = elem->value.enumerated.item[0]; - kctl->private_value = ((device << SND_DJM_DEVICE_SHIFT) | + kctl->private_value = (((unsigned long)device << SND_DJM_DEVICE_SHIFT) | (group << SND_DJM_GROUP_SHIFT) | value); @@ -2964,7 +2964,7 @@ static int snd_djm_controls_create(struct usb_mixer_interface *mixer, value = device->controls[i].default_value; knew.name = device->controls[i].name; knew.private_value = ( - (device_idx << SND_DJM_DEVICE_SHIFT) | + ((unsigned long)device_idx << SND_DJM_DEVICE_SHIFT) | (i << SND_DJM_GROUP_SHIFT) | value); err = snd_djm_controls_update(mixer, device_idx, i, value); -- 2.30.2

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[GIT PULL] ASoC fixes for v5.12-rc2
by Mark Brown 18 Mar '21

18 Mar '21

The following changes since commit a38fd8748464831584a19438cbb3082b5a2dab15: Linux 5.12-rc2 (2021-03-05 17:33:41 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.12-rc2 for you to fetch changes up to f9dc51cc6676e0360dfcb1836b3d426a3ccf3093: Merge series "Do not handle MCLK device clock in simple-card-utils" from Sameer Pujar <spujar(a)nvidia.com>: (2021-03-16 17:55:37 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.12 Quite a lot of mostly platform specific fixes here, the only one which is generic is a fix for regressions on devices with more complex clocking support with simple-card. There's also a few new device IDs and platform quirks. ---------------------------------------------------------------- Alexander Shiyan (1): ASoC: fsl_ssi: Fix TDM slot setup for I2S mode Bard Liao (1): ASoC: rt711: add snd_soc_component remove callback Benjamin Rood (1): ASoC: sgtl5000: set DAP_AVC_CTRL register to correct default value on probe Colin Ian King (1): ASoC: codecs: lpass-rx-macro: Fix uninitialized variable ec_tx Hans de Goede (8): ASoC: rt5670: Remove 'OUT Channel Switch' control ASoC: rt5670: Remove 'HP Playback Switch' control ASoC: rt5670: Remove ADC vol-ctrl mute bits poking from Sto1 ADC mixer settings ASoC: rt5670: Add emulated 'DAC1 Playback Switch' control ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 10-p0XX OVCD current threshold ASoC: rt5640: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: rt5651: Fix dac- and adc- vol-tlv values being off by a factor of 10 ASoC: es8316: Simplify adc_pga_gain_tlv table Jack Yu (2): ASoC: rt1015: fix i2c communication error ASoC: rt1015: enable BCLK detection after calibration Jiaxin Yu (1): ASoC: mediatek: mt8192: fix tdm out data is valid on rising edge Jon Hunter (1): ASoC: soc-core: Prevent warning if no DMI table is present Jonathan Marek (2): ASoC: codecs: lpass-va-macro: mute/unmute all active decimators ASoC: codecs: lpass-wsa-macro: fix RX MIX input controls Lucas Tanure (5): ASoC: cs42l42: Fix Bitclock polarity inversion ASoC: cs42l42: Fix channel width support ASoC: cs42l42: Fix mixer volume control ASoC: cs42l42: Don't enable/disable regulator at Bias Level ASoC: cs42l42: Always wait at least 3ms after reset Mark Brown (5): Merge series "ASoC: rt5670: Various kcontrol fixes" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "AsoC: rt5640/rt5651: Volume control fixes" from Hans de Goede <hdegoede(a)redhat.com>: Merge series "Report jack and button detection + Capture Support" from Lucas Tanure <tanureal(a)opensource.cirrus.com>: Merge series "ASoC: sdm845: array out of bound issues" from Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>: Merge series "Do not handle MCLK device clock in simple-card-utils" from Sameer Pujar <spujar(a)nvidia.com>: Meng Li (1): spi: cadence: set cqspi to the driver_data field of struct device Pan Xiuli (1): ASoC: SOF: intel: fix wrong poll bits in dsp power down Peter Robinson (1): ASoC: remove remnants of sirf prima/atlas audio codec Pierre-Louis Bossart (1): ASoC: SOF: Intel: unregister DMIC device on probe error Sameer Pujar (2): ASoC: simple-card-utils: Do not handle device clock ASoC: rt5659: Update MCLK rate in set_sysclk() Shengjiu Wang (3): ASoC: ak4458: Add MODULE_DEVICE_TABLE ASoC: ak5558: Add MODULE_DEVICE_TABLE ASoC: dt-bindings: fsl_spdif: Add compatible string for new platforms Srinivas Kandagatla (3): ASoC: qcom: sdm845: Fix array out of bounds access ASoC: qcom: sdm845: Fix array out of range on rx slim channels ASoC: codecs: wcd934x: add a sanity check in set channel map Srinivasa Rao Mandadapu (1): ASoC: qcom: lpass-cpu: Fix lpass dai ids parse .../devicetree/bindings/sound/fsl,spdif.yaml | 4 + drivers/spi/spi-cadence-quadspi.c | 1 + sound/soc/codecs/Kconfig | 5 - sound/soc/codecs/ak4458.c | 1 + sound/soc/codecs/ak5558.c | 1 + sound/soc/codecs/cs42l42.c | 112 ++++++------------- sound/soc/codecs/cs42l42.h | 13 ++- sound/soc/codecs/es8316.c | 9 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 28 +++-- sound/soc/codecs/lpass-wsa-macro.c | 20 ++-- sound/soc/codecs/rt1015.c | 2 + sound/soc/codecs/rt5640.c | 4 +- sound/soc/codecs/rt5651.c | 4 +- sound/soc/codecs/rt5659.c | 5 + sound/soc/codecs/rt5670.c | 110 +++++++++++++++--- sound/soc/codecs/rt5670.h | 9 +- sound/soc/codecs/rt711.c | 8 ++ sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/sirf-audio-codec.h | 124 --------------------- sound/soc/codecs/wcd934x.c | 6 + sound/soc/fsl/fsl_ssi.c | 6 +- sound/soc/generic/simple-card-utils.c | 13 ++- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 4 +- sound/soc/mediatek/mt8192/mt8192-reg.h | 8 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/sdm845.c | 6 +- sound/soc/soc-core.c | 4 + sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda.c | 1 + 31 files changed, 232 insertions(+), 286 deletions(-) delete mode 100644 sound/soc/codecs/sirf-audio-codec.h

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[PATCH v4 2/3] dt-bindings: Add Rockchip rk817 audio CODEC support
by Chris Morgan 18 Mar '21

18 Mar '21

Create dt-binding documentation to document rk817 codec. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- .../bindings/sound/rockchip,rk817-codec.yaml | 63 +++++++++++++++++++ 1 file changed, 63 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml new file mode 100644 index 000000000000..986cebcd0832 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk817-codec.yaml @@ -0,0 +1,63 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rockchip-rk817.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Rockchip rk817 audio codec + +description: + The rk817 codec is an I2C codec integrated with every Rockchip + rk817 PMIC MFD. + +properties: + compatible: + enum: + - rockchip,rk817-codec + + "#sound-dai-cells": + const: 1 + + clocks: + maxItems: 1 + + clock-names: + const: "mclk" + + mic-in-differential: + description: the microphone is in differential mode. + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - clocks + - clock-names + +additionalProperties: false + +examples: + - | +rk817: pmic@20 { + compatible = "rockchip,rk817"; + reg = <0x20>; + interrupt-parent = <&gpio0>; + interrupts = <7 IRQ_TYPE_LEVEL_LOW>; + pinctrl-names = "default"; + pinctrl-0 = <&pmic_int>; + + ........ + + rk817_codec: codec { + #sound-dai-cells = <0>; + compatible = "rockchip,rk817-codec"; + clocks = <&cru SCLK_I2S_8CH_OUT>; + clock-names = "mclk"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s_8ch_mclk>; + mic-in-differential; + status = "okay"; + }; + + ........ + +}; -- 2.25.1

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Re: [PATCH] ASoC: rt715: add main capture switch and main capture volume
by Mark Brown 18 Mar '21

18 Mar '21

On Thu, Mar 18, 2021 at 12:19:18PM +0000, Jack Yu wrote: > > This change is literally just removing some brackets that don't do anything, the > > regmap API has no impact here. > Sorry, I misunderstand your reply, and yes, this modification just remove some brackets. > Should I send it with separate patch? Yes, sure.

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Re: [PATCH] conf: fix memory leak on the error path in parse_args()
by Mark Hills 18 Mar '21

18 Mar '21

Having a little trouble which bisected to this patch. First noticed it's causing Chromium to crash out one of its subprocesses (stack trace below) Can actually be replicated with a simple "aplay -L": aplay: conf.c:2207: snd_config_delete: Assertion `config' failed. Aborted (core dumped) Program received signal SIGABRT, Aborted. 0x00007ffff7984c7b in raise () from /lib64/libc.so.6 (gdb) bt #0 0x00007ffff7984c7b in raise () from /lib64/libc.so.6 #1 0x00007ffff7965548 in abort () from /lib64/libc.so.6 #2 0x00007ffff796542f in __assert_fail_base.cold () from /lib64/libc.so.6 #3 0x00007ffff7975fc2 in __assert_fail () from /lib64/libc.so.6 #4 0x00007ffff7cce7e2 in snd_config_delete (config=<optimized out>) at conf.c:2207 #5 0x00007ffff7cd004d in parse_args (subs=0x6c66f0, str=<optimized out>, str@entry=0x641fa5 "CARD=Layla3G,DEV=0", defs=<optimized out>) at conf.c:5172 #6 0x00007ffff7cd0c31 in snd_config_expand (config=0x6594a0, root=root@entry=0x639330, args=args@entry=0x641fa5 "CARD=Layla3G,DEV=0", private_data=private_data@entry=0x0, result=result@entry=0x7fffffffdf80) at conf.c:5227 #7 0x00007ffff7cd13e2 in snd_config_search_definition (config=config@entry=0x639330, base=base@entry=0x40cf74 "pcm", name=name@entry=0x641fa0 "plug:CARD=Layla3G,DEV=0", result=result@entry=0x7fffffffdf80) at conf.c:5312 #8 0x00007ffff7cd8154 in try_config (config=config@entry=0x639330, list=list@entry=0x7fffffffe260, base=0x40cf74 "pcm", name=<optimized out>) at namehint.c:269 #9 0x00007ffff7cd8e8a in add_card (config=<optimized out>, rw_config=0x639330, list=list@entry=0x7fffffffe260, card=0) at namehint.c:488 #10 0x00007ffff7cd9410 in snd_device_name_hint (card=0, card@entry=-1, iface=<optimized out>, iface@entry=0x40cf74 "pcm", hints=hints@entry=0x7fffffffe308) at namehint.c:639 #11 0x000000000040442b in pcm_list () at aplay.c:339 #12 0x000000000040a638 in main (argc=2, argv=0x7fffffffe5c8) at aplay.c:824 I saw "namehint" in the stack; I'm using this fragment of asoundrc (it's the only way to get chosen ALSA devices available in Chrome) but it seems to be unrelated; commenting it out and the assertion still fails. I haven't had time to look any deeper. namehint.pcm [ "layla1|Layla: Microphone input 1" "layla2|Layla: Microphone input 2" "layla12|Layla: Microphone inputs 1,2" "layla34|Layla: Line inputs 3,4 (DJ mixer)" "layla|Layla: Analogue inputs 1,2,3,4" "plug:shure|Shure X2U: Microphone input" "plug:dj1|Audio 8 DJ: Input 1 (Left deck)" "plug:dj2|Audio 8 DJ: Input 2 (Right deck)" "plug:local|Local audio bus" ] -- Mark >From ad5f255b4767d52f7aaafd3eb348c680e990f0b8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela <perex(a)perex.cz> Date: Wed, 10 Mar 2021 18:10:57 +0100 Subject: [PATCH] conf: fix memory leak on the error path in parse_args() Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- src/conf.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf.c b/src/conf.c index 38eefbf8..14b14b59 100644 --- a/src/conf.c +++ b/src/conf.c @@ -5169,6 +5169,7 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) err = snd_config_add(subs, sub); if (err < 0) { _err: + snd_config_delete(sub); free(val); return err; } -- 2.17.5 chrome: conf.c:2207: snd_config_delete: Assertion `config' failed. Received signal 6 #0 0x55c2535f35e9 base::debug::CollectStackTrace() #1 0x55c25355e723 base::debug::StackTrace::StackTrace() #2 0x55c2535f3190 base::debug::(anonymous namespace)::StackDumpSignalHandler() #3 0x7f35b16e6690 (/lib64/libpthread-2.30.so+0x1368f) #4 0x7f35ac583c7b __GI_raise #5 0x7f35ac564548 __GI_abort #6 0x7f35ac56442f __assert_fail_base.cold #7 0x7f35ac574fc2 __GI___assert_fail #8 0x7f35acda27e2 snd_config_delete #9 0x7f35acda404d parse_args #10 0x7f35acda4c31 snd_config_expand #11 0x7f35acda53e2 snd_config_search_definition #12 0x7f35acdac154 try_config #13 0x7f35acdace8a add_card #14 0x7f35acdad13c snd_device_name_hint #15 0x55c250eb3b13 media::AudioManagerAlsa::HasAnyAlsaAudioDevice() #16 0x55c250ea9445 media::AudioSystemHelper::ComputeOutputParameters() #17 0x55c250ea93e0 media::AudioSystemHelper::GetOutputStreamParameters() #18 0x55c251d38db3 audio::SystemInfo::GetOutputStreamParameters() #19 0x55c2511bd455 audio::mojom::SystemInfoStubDispatch::AcceptWithResponder() #20 0x55c253d0bad3 mojo::InterfaceEndpointClient::HandleValidatedMessage() #21 0x55c253d0e8bd mojo::MessageDispatcher::Accept() #22 0x55c253d12380 mojo::internal::MultiplexRouter::ProcessIncomingMessage() #23 0x55c253d11b09 mojo::internal::MultiplexRouter::Accept() #24 0x55c253d0e8f9 mojo::MessageDispatcher::Accept() #25 0x55c253d0a184 mojo::Connector::DispatchMessage() #26 0x55c253d0a8b7 mojo::Connector::ReadAllAvailableMessages() #27 0x55c25056d034 base::internal::Invoker<>::RunOnce() #28 0x55c2535b52b2 base::TaskAnnotator::RunTask() #29 0x55c2535c6c11 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl() #30 0x55c2535c68ea base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork() #31 0x55c25357669a base::MessagePumpDefault::Run() #32 0x55c2535c725b base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run() #33 0x55c25359820b base::RunLoop::Run() #34 0x55c2534d01da content::UtilityMain() #35 0x55c2534fba9c content::RunZygote() #36 0x55c2534fcd6b content::ContentMainRunnerImpl::Run() #37 0x55c2534fa35d content::RunContentProcess() #38 0x55c2534facfd content::ContentMain() #39 0x55c25056a068 ChromeMain #40 0x7f35ac565e6b __libc_start_main #41 0x55c250569eaa _start r8: 0000000000000000 r9: 00007ffd94a02da0 r10: 0000000000000008 r11: 0000000000000246 r12: 00007f35ace23e21 r13: 000000000000089f r14: 00007f35ace2af41 r15: 0000000000000004 di: 0000000000000002 si: 00007ffd94a02da0 bp: 00007f35ac6ebf38 bx: 00007f35a5d83f00 dx: 0000000000000000 ax: 0000000000000000 cx: 00007f35ac583c7b sp: 00007ffd94a02e18 ip: 00007f35ac583c7b efl: 0000000000000246 cgf: 002b000000000033 erf: 0000000000000000 trp: 0000000000000000 msk: 0000000000000000 cr2: 0000000000000000 [end of stack trace] Calling _exit(1). Core file will not be generated.

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Re: [PATCH] ASoC: rt715: add main capture switch and main capture volume
by Mark Brown 18 Mar '21

18 Mar '21

On Thu, Mar 18, 2021 at 03:35:31AM +0000, Jack Yu wrote: > > > - *l_val = (val_h << 8); > > > + *l_val = val_h << 8; > > > ret = regmap_read(rt715->regmap, addr_h, l_val); > > > if (ret < 0) > > > pr_err("Failed to get L channel gain.\n"); > > This looks like an unrelated formatting change. > Because regmap has its own callback for read/write, so the formatting change is required for callback. This change is literally just removing some brackets that don't do anything, the regmap API has no impact here. > > > if (mc->invert) { > > > /* for mute status */ > > > - read_ll = !((read_ll & 0x80) >> RT715_MUTE_SFT); > > > - read_rl = !((read_rl & 0x80) >> RT715_MUTE_SFT); > > > + read_ll = !(read_ll & 0x80); > > > + read_rl = !(read_rl & 0x80); > > > } else { > > > /* for gain */ > > > read_ll = read_ll & 0x7f; > > This too. > This change is just to simplify the code. The result will still be the same. Sure, but that should be a separate patch with a separate changelog which explicitly says that this is being done.

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[PATCH] ALSA: hda: Fix spelling mistakes
by hjh 18 Mar '21

18 Mar '21

Signed-off-by: hjh <huangjianghui(a)uniontech.com> --- sound/hda/hdac_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index a6ed3dc35f7e..1eb8563db2df 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -618,7 +618,7 @@ void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger); /** - * snd_hdac_stream_sync - sync with start/strop trigger operation + * snd_hdac_stream_sync - sync with start/stop trigger operation * @azx_dev: HD-audio core stream (master stream) * @start: true = start, false = stop * @streams: bit flags of streams to sync -- 2.20.1

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