October 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: aoa: Fix I2S device accounting
by Takashi Iwai 27 Oct '22

27 Oct '22

i2sbus_add_dev() is supposed to return the number of probed devices, i.e. either 1 or 0. However, i2sbus_add_dev() has one error handling that returns -ENODEV; this will screw up the accumulation number counted in the caller, i2sbus_probe(). Fix the return value to 0 and add the comment for better understanding for readers. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/aoa/soundbus/i2sbus/core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index f6841daf9e3b..51ed2f34b276 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -147,6 +147,7 @@ static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index, return rc; } +/* Returns 1 if added, 0 for otherwise; don't return a negative value! */ /* FIXME: look at device node refcounting */ static int i2sbus_add_dev(struct macio_dev *macio, struct i2sbus_control *control, @@ -213,7 +214,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, * either as the second one in that case is just a modem. */ if (!ok) { kfree(dev); - return -ENODEV; + return 0; } mutex_init(&dev->lock); -- 2.35.3

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[GIT PULL] ASoC fixes for v6.1-rc2
by Mark Brown 27 Oct '22

27 Oct '22

The following changes since commit 9abf2313adc1ca1b6180c508c25f22f9395cc780: Linux 6.1-rc1 (2022-10-16 15:36:24 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.1-rc2 for you to fetch changes up to e9441675edc1bb8dbfadacf68aafacca60d65a25: ASoC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init() (2022-10-21 13:04:14 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.1 Quite a few fixes here, a lot driver specific, plus some new quirks. There was a bit of a mess with the runtime PM handling due to some confusion in the API there which resulted in a number of commits and reverts but that should all be stable now. ---------------------------------------------------------------- Aidan MacDonald (1): ASoC: simple-card: Fix up checks for HW param fixups Colin Ian King (1): ASoC: codecs: jz4725b: Fix spelling mistake "Sourc" -> "Source", "Routee" -> "Route" Derek Fang (2): ASoC: rt5682s: Fix the TDM Tx settings ASoC: rt1019: Fix the TDM settings Geert Uytterhoeven (1): ASoC: codecs: tlv320adc3xxx: Wrap adc3xxx_i2c_remove() in __exit_p() Jiangshan Yi (1): ASoC: cx2072x: fix spelling typo in comment Kai Vehmanen (1): ASoC: SOF: ipc4-mtrace: protect per-core nodes against multiple open Leohearts (1): ASoC: amd: yc: Add Lenovo Thinkbook 14+ 2022 21D0 to quirks table Mark Brown (2): ASoC: wmxxxx: Revert old "ASoC: wmxxxx: Fix PM disable depth imbalance in wmxxxx_probe" ASoC: jz4752b: Capture fixes Peter Ujfalusi (1): Revert "ASoC: soc-component: using pm_runtime_resume_and_get instead of pm_runtime_get_sync" Pierre-Louis Bossart (3): ASoC: Intel: sof_sdw: add quirk variant for LAPBC710 NUC15 ASoC: SOF: Intel: pci-mtl: fix firmware name ASoC: SOF: Intel: pci-tgl: fix ADL-N descriptor Randy Dunlap (2): ASoC: codec: tlv320adc3xxx: add GPIOLIB dependency ASoC: qcom: SND_SOC_SC7180 optionally depends on SOUNDWIRE Shuming Fan (2): ASoC: rt1308-sdw: update the preset settings ASoC: rt1308-sdw: add the default value of some registers Siarhei Volkau (4): ASoC: codecs: jz4725b: add missed Line In power control bit ASoC: codecs: jz4725b: fix reported volume for Master ctl ASoC: codecs: jz4725b: use right control for Capture Volume ASoC: codecs: jz4725b: fix capture selector naming Srinivasa Rao Mandadapu (2): ASoC: qcom: lpass-cpu: mark HDMI TX registers as volatile ASoC: qcom: lpass-cpu: Mark HDMI TX parity register as volatile Xiaolei Wang (1): ASoC: wm8962: Add an event handler for TEMP_HP and TEMP_SPK Yang Yingliang (2): ASoC: Intel: Skylake: fix possible memory leak in skl_codec_device_init() ASoC: SOF: Intel: hda-codec: fix possible memory leak in hda_codec_device_init() Yong Zhi (1): ASoC: Intel: sof_rt5682: Add quirk for Rex board Zhang Qilong (7): ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe ASoC: wm5102: Revert "ASoC: wm5102: Fix PM disable depth imbalance in wm5102_probe" ASoC: wm5110: Revert "ASoC: wm5110: Fix PM disable depth imbalance in wm5110_probe" ASoC: wm8997: Revert "ASoC: wm8997: Fix PM disable depth imbalance in wm8997_probe" ASoC: mt6660: Keep the pm_runtime enables before component stuff in mt6660_i2c_probe linkt (1): ASoC: amd: yc: Adding Lenovo ThinkBook 14 Gen 4+ ARA and Lenovo ThinkBook 16 Gen 4+ ARA to the Quirks List include/sound/simple_card_utils.h | 1 + sound/soc/amd/yc/acp6x-mach.c | 21 ++++++++++++++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cx2072x.h | 2 +- sound/soc/codecs/jz4725b.c | 34 ++++++++++++---------- sound/soc/codecs/mt6660.c | 8 +++--- sound/soc/codecs/rt1019.c | 20 +++++++------ sound/soc/codecs/rt1019.h | 6 ++++ sound/soc/codecs/rt1308-sdw.c | 17 +++++++++-- sound/soc/codecs/rt1308-sdw.h | 3 ++ sound/soc/codecs/rt1308.h | 5 ++++ sound/soc/codecs/rt5682s.c | 15 ++++++++-- sound/soc/codecs/rt5682s.h | 1 + sound/soc/codecs/tlv320adc3xxx.c | 2 +- sound/soc/codecs/wm5102.c | 7 +++-- sound/soc/codecs/wm5110.c | 7 +++-- sound/soc/codecs/wm8962.c | 54 +++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8997.c | 7 +++-- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/simple-card-utils.c | 15 ++++++++++ sound/soc/generic/simple-card.c | 3 +- sound/soc/intel/boards/sof_rt5682.c | 12 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++ sound/soc/intel/skylake/skl.c | 8 +----- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/lpass-cpu.c | 10 +++++++ sound/soc/soc-component.c | 6 ++-- sound/soc/sof/intel/hda-codec.c | 8 +----- sound/soc/sof/intel/pci-mtl.c | 2 +- sound/soc/sof/intel/pci-tgl.c | 30 ++++++++++++++++++- sound/soc/sof/ipc4-mtrace.c | 20 +++++++++++-- 31 files changed, 270 insertions(+), 69 deletions(-)

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[Regression] Bug 216613 - Sound stopped working with v6.0.3 on Lenovo T14 Gen2i: ASoC: error at snd_soc_component_probe
by Thorsten Leemhuis 27 Oct '22

27 Oct '22

Hi, this is your Linux kernel regression tracker speaking. I noticed a regression report in bugzilla.kernel.org. As many (most?) kernel developer don't keep an eye on it, I decided to forward it by mail. Quoting from https://bugzilla.kernel.org/show_bug.cgi?id=216613 : > Grzegorz Alibożek 2022-10-21 19:26:43 UTC > > After upgrade kernel from 6.0.2 to 6.0.3 on Lenovo T14 Gen2i, sound stopped working. > dmesg: > > paź 21 21:11:45 kernel: snd_hda_codec_hdmi ehdaudio0D2: failed to create hda codec -12 > paź 21 21:11:45 kernel: snd_hda_codec_hdmi ehdaudio0D2: ASoC: error at snd_soc_component_probe on ehdaudio0D2: -12 > paź 21 21:11:45 kernel: skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: failed to instantiate card -12 > > [reply] [−] Comment 1 Grzegorz Alibożek 2022-10-21 19:56:43 UTC > > Created attachment 303070 [details] > trace See the ticket for more details. BTW, let me use this mail to also add the report to the list of tracked regressions to ensure it's doesn't fall through the cracks: #regzbot introduced: v6.0.2..v6.0.3 https://bugzilla.kernel.org/show_bug.cgi?id=216613 #regzbot ignore-activity Ciao, Thorsten (wearing his 'the Linux kernel's regression tracker' hat) P.S.: As the Linux kernel's regression tracker I deal with a lot of reports and sometimes miss something important when writing mails like this. If that's the case here, don't hesitate to tell me in a public reply, it's in everyone's interest to set the public record straight.

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Thanks to review question [PATCH V2 1/2] ASoc:codes:Add Awinic AW883XX audio amplifier driver
by wangweidong.a＠awinic.com 27 Oct '22

27 Oct '22

Hi Mark Brown Thank you for your valuable advice to let me understand the problem. I will fix this issue in the next patch Best regards Weidong Wang On Mon, Oct 24, 2022 at 10:41:03AM +0800, wangweidong.a(a)awinic.com wrote: > > Then it's not a mute function, the goal of the mute function is to > > run > before all the power management code to minimise glitches during power > management. Just implement the power management via the standard ASoC > power > > > management APIs. > The essence of calling mute is not switch dsp, but switch PA. We think > that PA has only two states. When no audio stream enters, turn off the > PA and turn off the PA's dsp. When the audio stream enters, the PA is > turned on, and the dsp is turned on at the same time > That's not what the mute function is for, the interface is specifically for muting the stream while power management changes are > going on. What you're describing is power management so should be controlled via DAPM. > If your device doesn't have any support for a separate mute function then it should just not implement anything for that in the > driver.

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[PATCH v2 0/9] ASoC: qdsp6: audioreach: add multi-port, SAL and MFC support
by Srinivas Kandagatla 26 Oct '22

26 Oct '22

This patchset adds support to multi-port connections between AudioReach Modules which is required for sophisticated graphs like ECNS or Speaker Protection. Also as part of ECNS testing new module support for SAL and MFC are added. Tested on SM8450 with ECNS. Thanks, Srini Changes since v1: Fixed two warnings on unused and unintialized variable. Srinivas Kandagatla (9): ASoC: qdsp6: audioreach: topology use idr_alloc_u32 ASoC: qdsp6: audioreach: remove unused connection_list ASoC: qdsp6: audioreach: update dapm kcontrol private data ASoC: qdsp6: audioreach: Simplify handing FE and BE graph connections ASoC: qdsp6: audioreach: simplify module_list sz calculation ASoC: qdsp6: audioreach: add support for more port connections ASoC: qdsp6: audioreach: add support to enable SAL Module ASoC: qdsp6: audioreach: add support for MFC Module ASoC: qdsp6: audioreach: add support to enable module command include/uapi/sound/snd_ar_tokens.h | 27 +++ sound/soc/qcom/qdsp6/audioreach.c | 310 ++++++++++++++++++++--------- sound/soc/qcom/qdsp6/audioreach.h | 47 +++-- sound/soc/qcom/qdsp6/q6apm.c | 84 +------- sound/soc/qcom/qdsp6/q6apm.h | 6 +- sound/soc/qcom/qdsp6/topology.c | 243 +++++++++++++++++++--- 6 files changed, 489 insertions(+), 228 deletions(-) -- 2.21.0

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[PATCH v1 1/2] ASoC: simple-card: Support custom DAI system clock IDs
by Aidan MacDonald 26 Oct '22

26 Oct '22

Some DAIs have multiple system clock sources, which can be chosen using the "clk_id" argument to snd_soc_dai_set_sysclk(). Currently this is hardcoded to 0 when using simple cards, but that choice is not always suitable. Add the "system-clock-id" property to allow selecting a different clock ID on a per-DAI basis. To simplify the logic on DPCM cards, add a dummy "asoc_simple_dai" instance and use that for the dummy components on DPCM links. This ensures that when we're iterating over DAIs in the PCM runtime there is always a matching "asoc_simple_dai" we can dereference. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- include/sound/simple_card_utils.h | 2 ++ sound/soc/generic/simple-card-utils.c | 26 ++++++++++++++++++++------ 2 files changed, 22 insertions(+), 6 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index a0b827f0c2f6..9f9a72299637 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -26,6 +26,7 @@ struct asoc_simple_dai { const char *name; unsigned int sysclk; int clk_direction; + int sysclk_id; int slots; int slot_width; unsigned int tx_slot_mask; @@ -67,6 +68,7 @@ struct asoc_simple_priv { struct prop_nums num; unsigned int mclk_fs; } *dai_props; + struct asoc_simple_dai dummy_dai; struct asoc_simple_jack hp_jack; struct asoc_simple_jack mic_jack; struct snd_soc_dai_link *dai_link; diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index bef16833c487..d4d898e06e76 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -262,6 +262,9 @@ int asoc_simple_parse_clk(struct device *dev, if (of_property_read_bool(node, "system-clock-direction-out")) simple_dai->clk_direction = SND_SOC_CLOCK_OUT; + if (!of_property_read_u32(node, "system-clock-id", &val)) + simple_dai->sysclk_id = val; + return 0; } EXPORT_SYMBOL_GPL(asoc_simple_parse_clk); @@ -355,7 +358,7 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) if (props->mclk_fs && !dai->clk_fixed && !snd_soc_dai_active(cpu_dai)) snd_soc_dai_set_sysclk(cpu_dai, - 0, 0, SND_SOC_CLOCK_OUT); + dai->sysclk_id, 0, SND_SOC_CLOCK_OUT); asoc_simple_clk_disable(dai); } @@ -364,7 +367,7 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) if (props->mclk_fs && !dai->clk_fixed && !snd_soc_dai_active(codec_dai)) snd_soc_dai_set_sysclk(codec_dai, - 0, 0, SND_SOC_CLOCK_IN); + dai->sysclk_id, 0, SND_SOC_CLOCK_IN); asoc_simple_clk_disable(dai); } @@ -439,7 +442,7 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream, struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *props = simple_priv_to_props(priv, rtd->num); unsigned int mclk, mclk_fs = 0; - int i, ret; + int i, ret, sysclk_id; if (props->mclk_fs) mclk_fs = props->mclk_fs; @@ -472,13 +475,21 @@ int asoc_simple_hw_params(struct snd_pcm_substream *substream, } for_each_rtd_codec_dais(rtd, i, sdai) { - ret = snd_soc_dai_set_sysclk(sdai, 0, mclk, SND_SOC_CLOCK_IN); + pdai = simple_props_to_dai_codec(props, i); + sysclk_id = pdai->sysclk_id; + + ret = snd_soc_dai_set_sysclk(sdai, sysclk_id, mclk, + SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) return ret; } for_each_rtd_cpu_dais(rtd, i, sdai) { - ret = snd_soc_dai_set_sysclk(sdai, 0, mclk, SND_SOC_CLOCK_OUT); + pdai = simple_props_to_dai_cpu(props, i); + sysclk_id = pdai->sysclk_id; + + ret = snd_soc_dai_set_sysclk(sdai, pdai->sysclk_id, mclk, + SND_SOC_CLOCK_OUT); if (ret && ret != -ENOTSUPP) return ret; } @@ -523,7 +534,8 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; if (simple_dai->sysclk) { - ret = snd_soc_dai_set_sysclk(dai, 0, simple_dai->sysclk, + ret = snd_soc_dai_set_sysclk(dai, simple_dai->sysclk_id, + simple_dai->sysclk, simple_dai->clk_direction); if (ret && ret != -ENOTSUPP) { dev_err(dai->dev, "simple-card: set_sysclk error\n"); @@ -858,6 +870,7 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dai_link[i].cpus = &priv->dummy; dai_props[i].num.cpus = dai_link[i].num_cpus = 1; + dai_props[i].cpu_dai = &priv->dummy_dai; } if (li->num[i].codecs) { @@ -882,6 +895,7 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, dai_link[i].codecs = &priv->dummy; dai_props[i].num.codecs = dai_link[i].num_codecs = 1; + dai_props[i].codec_dai = &priv->dummy_dai; } if (li->num[i].platforms) { -- 2.38.1

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[PATCH v6 0/9] ASoC: cleanups and improvements for jz4740-i2s
by Aidan MacDonald 26 Oct '22

26 Oct '22

This series is a preparatory cleanup of the jz4740-i2s driver before adding support for a new SoC. The two improvements are lifting unnecessary restrictions on sample rates and formats -- the existing ones appear to be derived from the limitations of the JZ4740's internal codec and don't reflect the actual capabilities of the I2S controller. I'm unable to test the series on any JZ47xx SoCs, but I have tested on an X1000 (which is the SoC I'll be adding in a followup series). Changes in v6: * Added Cc: stable for flush bits fix. Changes in v5: * Drop 'mem' resource removal patch already upstream. * Update FIFO flush bits fix to address Paul's review comments. * Drop PLL clock name patch, that needs a different approach. Link for v4: https://lore.kernel.org/alsa-devel/20220708160244.21933-1-aidanmacdonald.0x… Aidan MacDonald (9): ASoC: jz4740-i2s: Handle independent FIFO flush bits ASoC: jz4740-i2s: Convert to regmap API ASoC: jz4740-i2s: Simplify using regmap fields ASoC: jz4740-i2s: Use FIELD_PREP() macros in hw_params callback ASoC: jz4740-i2s: Align macro values and sort includes ASoC: jz4740-i2s: Support S20_LE and S24_LE sample formats ASoC: jz4740-i2s: Support continuous sample rate ASoC: jz4740-i2s: Move component functions near the component driver ASoC: jz4740-i2s: Refactor DAI probe/remove ops as component ops sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 455 ++++++++++++++++++---------------- 2 files changed, 243 insertions(+), 213 deletions(-) -- 2.38.1

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[PATCH] ASoC: dt-bindings: mt8192-mt6359: Set maxItems, not type, for sound-dai
by Nícolas F. R. A. Prado 26 Oct '22

26 Oct '22

sound-dai is a standard property whose type is already set to phandle-array by sound-dai.yaml, so there's no need to set it (and wrongly so for headset-codec) in this binding. What should be set however is the maximum number of items, which for headset-codec should be 1. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- .../devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml index 478be7e3fa29..c6e614c1c30b 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml @@ -34,7 +34,7 @@ properties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 required: - sound-dai @@ -48,7 +48,6 @@ properties: maxItems: 2 items: maxItems: 1 - $ref: /schemas/types.yaml#/definitions/phandle-array required: - sound-dai -- 2.38.1

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[PATCH] ASoC: SOF: ipc4-loader: Return ssize_t from sof_ipc4_fw_parse_ext_man()
by Peter Ujfalusi 26 Oct '22

26 Oct '22

sof_ipc4_fw_parse_ext_man() can return negative error numbers which is not correct for the used size_t type. Change the return value to ssize_t and use the same type where the function is called. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 73c091a2fe96 ("ASoC: SOF: ipc4-loader: Support for loading external libraries") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc4-loader.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc4-loader.c b/sound/soc/sof/ipc4-loader.c index af0018b38cf0..702196774c50 100644 --- a/sound/soc/sof/ipc4-loader.c +++ b/sound/soc/sof/ipc4-loader.c @@ -17,8 +17,8 @@ /* The module ID includes the id of the library it is part of at offset 12 */ #define SOF_IPC4_MOD_LIB_ID_SHIFT 12 -static size_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev, - struct sof_ipc4_fw_library *fw_lib) +static ssize_t sof_ipc4_fw_parse_ext_man(struct snd_sof_dev *sdev, + struct sof_ipc4_fw_library *fw_lib) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; const struct firmware *fw = fw_lib->sof_fw.fw; @@ -141,7 +141,7 @@ static size_t sof_ipc4_fw_parse_basefw_ext_man(struct snd_sof_dev *sdev) { struct sof_ipc4_fw_data *ipc4_data = sdev->private; struct sof_ipc4_fw_library *fw_lib; - size_t payload_offset; + ssize_t payload_offset; int ret; fw_lib = devm_kzalloc(sdev->dev, sizeof(*fw_lib), GFP_KERNEL); @@ -170,7 +170,7 @@ static int sof_ipc4_load_library_by_uuid(struct snd_sof_dev *sdev, struct sof_ipc4_fw_data *ipc4_data = sdev->private; struct sof_ipc4_fw_library *fw_lib; const char *fw_filename; - size_t payload_offset; + ssize_t payload_offset; int ret, i, err; if (!sdev->pdata->fw_lib_prefix) { -- 2.38.1

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[PATCH] ASoC: SOF: Intel: MTL: fix comment error
by Pierre-Louis Bossart 26 Oct '22

26 Oct '22

From: Yong Zhi <yong.zhi(a)intel.com> Fix small cut-and-paste error in comment. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/mtl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 7e8b298786df6..91619036762b4 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -393,7 +393,7 @@ static int mtl_dsp_core_power_down(struct snd_sof_dev *sdev, int core) snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, MTL_DSP2CXCTL_PRIMARY_CORE_SPA_MASK, 0); - /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + /* Wait for unstable CPA read (0 then 1 then 0) just after setting SPA bit */ usleep_range(1000, 1010); ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_DSP2CXCTL_PRIMARY_CORE, dspcxctl, @@ -422,7 +422,7 @@ static int mtl_power_down_dsp(struct snd_sof_dev *sdev) snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFDSSCS, MTL_HFDSSCS_SPA_MASK, 0); - /* Wait for unstable CPA read (1 then 0 then 1) just after setting SPA bit */ + /* Wait for unstable CPA read (0 then 1 then 0) just after setting SPA bit */ usleep_range(1000, 1010); /* poll with timeout to check if operation successful */ -- 2.34.1

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