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Re: [PATCH 6/6] ASoC: Intel: boards: add missing dependency from realtek-common
by Mark Brown 27 Oct '22

27 Oct '22

On Mon, Oct 17, 2022 at 08:17:12AM +0200, Adam Borowski wrote: > These boards use code from sound/soc/sof/sof-audio.c > > Fixes: e1d5e13324020c4b405e63cae34560c7992bec2e > Signed-off-by: Adam Borowski <kilobyte(a)angband.pl> Please include human readable descriptions of things like commits and issues being discussed in e-mail in your mails, this makes them much easier for humans to read especially when they have no internet access. I do frequently catch up on my mail on flights or while otherwise travelling so this is even more pressing for me than just being about making things a bit easier to read. You've not incldued a cover letter for this and only copied me on some patches, are there any dependency issues?

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[PATCH v3 0/9] ASoC: Intel: avs: PCM power management
by Cezary Rojewski 27 Oct '22

27 Oct '22

Goal of the series is implementation of suspend/resume operations for a PCM stream along with all the collaterals connected to the subject. Start with splitting avs_dai_fe_hw_free() as ideally we would like to reuse as much of existing code as possible but snd_pcm_lib_free_pages() is not desired part of the function when speaking of suspend operation. The actual implementation of suspend/resume() for component drivers follows. For most scenarios, the PM flow is similar to standard streaming one, except for the part where the position register are being saved and the lack of PCM pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and avs_dai_resume_XXX() functions reuse their non-PM equivalents. Order of operations is affected by the fact that path binding/unbinding happens only in FE part of the stream. Above essentially unlocks SX+streaming scenarios i.e.: power transitions with an ongoing stream. As some streams are allowed to run in low power state, support is provided for S0iX state. The handlers check ACPI capabilities and the number of active low-power paths before deciding between SX and S0iX flows. The last portion of the patchset is addition of power/clock gating overrides. There is no single set of registers that ensures AudioDSP firmware loads 100% of time on every single configuration. By having them exposed, user can have the loading procedure behavior adjusted for their configuration without having to recompile the kernel. Changes in v3: - fixed unused-but-set-variable warnings reported by the test robot Changes in v2: - moved DRSM reg polling to separate function, snd_hdac_stream_wait_drsm() as suggested by Pierre - moved the acpi_gbl_FADT under AVS_S0IX_SUPPORTED macro and relocated the checks from pcm.c to topology.c so that the low-power streams are filtered as early as possible - fixed compilation when CONFIG_ACPI is disabled as reported by the test robot - simplified all the avs_dai_resume/suspend_xxx() by moving ->ignore_suspend check one level up, to avs_component_pm_op() (patch 0002) Amadeusz Sławiński (1): ASoC: Intel: avs: Handle SUSPEND and RESUME triggers Cezary Rojewski (7): ASoC: Intel: avs: Split pcm pages freeing operation from hw_free() ASoC: Intel: avs: Introduce PCM power management routines ALSA: hda: Introduce snd_hdac_stream_wait_drsm() ASoC: Intel: avs: Restart instead of resuming HDA capture streams ASoC: Intel: avs: Count low power streams ASoC: Intel: avs: Power and clock gating policy overriding ASoC: Intel: avs: Enact power gating policy Piotr Maziarz (1): ASoC: Intel: avs: Standby power-state support include/sound/hdaudio.h | 1 + include/sound/hdaudio_ext.h | 5 + sound/hda/hdac_stream.c | 21 ++ sound/soc/intel/avs/avs.h | 8 + sound/soc/intel/avs/core.c | 97 +++++++-- sound/soc/intel/avs/loader.c | 10 + sound/soc/intel/avs/pcm.c | 368 +++++++++++++++++++++++++++++---- sound/soc/intel/avs/topology.c | 10 + 8 files changed, 460 insertions(+), 60 deletions(-) -- 2.25.1

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[PATCH] ASoC: sof_es8336: reduce pop noise on speaker
by Zhu Ning 27 Oct '22

27 Oct '22

The Speaker GPIO needs to be turned on slightly behind the codec turned on. It also need to be turned off slightly before the codec turned down. Current code uses delay in DAPM_EVENT to do it but the mdelay delays the DAPM itself and thus has no effect. A delayed_work is added to turn on the speaker. The Speaker is turned off in .trigger since trigger is called slightly before the DAPM events. Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> --- sound/soc/intel/boards/sof_es8336.c | 58 ++++++++++++++++++++++------- 1 file changed, 44 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index fbb42e54947a..7fde36a02d11 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -63,6 +63,7 @@ struct sof_es8336_private { struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; + struct delayed_work pcm_pop_work; }; struct sof_hdmi_pcm { @@ -111,6 +112,44 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headset at mic1 port enabled\n"); } +static void pcm_pop_work_events(struct work_struct *work) +{ + struct sof_es8336_private *priv = + container_of(work, struct sof_es8336_private, pcm_pop_work.work); + + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + +} + +static int sof_8336_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_card *card = rtd->card; + struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + if (priv->speaker_en == false) + if (substream->stream == 0) + gpiod_set_value_cansleep(priv->gpio_speakers, true); + break; + default: + return -EINVAL; + } + + return 0; +} + static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -122,19 +161,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); - - if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) - return 0; - - if (SND_SOC_DAPM_EVENT_ON(event)) - msleep(70); - - gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); - + queue_delayed_work(system_wq, &priv->pcm_pop_work, msecs_to_jiffies(70)); return 0; } @@ -344,6 +371,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, /* machine stream operations */ static struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, + .trigger = sof_8336_trigger, }; static struct snd_soc_dai_link_component platform_component[] = { @@ -723,7 +751,8 @@ static int sof_es8336_probe(struct platform_device *pdev) } INIT_LIST_HEAD(&priv->hdmi_pcm_list); - + INIT_DELAYED_WORK(&priv->pcm_pop_work, + pcm_pop_work_events); snd_soc_card_set_drvdata(card, priv); if (mach->mach_params.dmic_num > 0) { @@ -752,6 +781,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + cancel_delayed_work(&priv->pcm_pop_work); gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- 2.37.2

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[PATCH] ALSA: aoa: i2sbus: fix possible memory leak in i2sbus_add_dev()
by Yang Yingliang 27 Oct '22

27 Oct '22

dev_set_name() in soundbus_add_one() allocates memory for name, it need be freed when of_device_register() fails, call soundbus_dev_put() to give up the reference that hold in device_initialize(), so that it can be freed in kobject_cleanup() when the refcount hit to 0. And other resources are also freed in i2sbus_release_dev(), so it can return 0 directly. Fixes: f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/aoa/soundbus/i2sbus/core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index faf6b03131ee..f6841daf9e3b 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -302,6 +302,10 @@ static int i2sbus_add_dev(struct macio_dev *macio, if (soundbus_add_one(&dev->sound)) { printk(KERN_DEBUG "i2sbus: device registration error!\n"); + if (dev->sound.ofdev.dev.kobj.state_initialized) { + soundbus_dev_put(&dev->sound); + return 0; + } goto err; } -- 2.25.1

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[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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