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

[PATCH AUTOSEL 5.17 58/66] ASoC: Intel: Revert "ASoC: Intel: sof_es8336: add quirk for Huawei D15 2021"
by Sasha Levin 31 Mar '22

31 Mar '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 1b5283483a782f6560999d8d5965b1874d104812 ] This reverts commit ce6a70bfce21bb4edb7c0f29ecfb0522fa34ab71. The next patch will add run-time detection of the required SSP and this hard-coded quirk is not needed. Acked-by: Mauro Carvalho Chehab <mchehab(a)kernel.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20220308192610.392950-14-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_es8336.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index e6d599f0cd26..20d577eaab6d 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -247,14 +247,6 @@ static const struct dmi_system_id sof_es8336_quirk_table[] = { SOF_ES8336_TGL_GPIO_QUIRK | SOF_ES8336_ENABLE_DMIC) }, - { - .callback = sof_es8336_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "HUAWEI"), - DMI_MATCH(DMI_BOARD_NAME, "BOHB-WAX9-PCB-B2"), - }, - .driver_data = (void *)SOF_ES8336_SSP_CODEC(0) - }, {} }; -- 2.34.1

2 2

[PATCH AUTOSEL 5.17 64/66] ASoC: ak4642: Use of_device_get_match_data()
by Sasha Levin 31 Mar '22

31 Mar '22

From: Minghao Chi <chi.minghao(a)zte.com.cn> [ Upstream commit 835ca59799f5c60b4b54bdc7aa785c99552f63e4 ] Use of_device_get_match_data() to simplify the code. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: Minghao Chi <chi.minghao(a)zte.com.cn> Link: https://lore.kernel.org/r/20220315023226.2118354-1-chi.minghao@zte.com.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/ak4613.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 4d2e78101f28..034195c83bd7 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -653,15 +653,10 @@ static int ak4613_i2c_probe(struct i2c_client *i2c, struct ak4613_priv *priv; regmap_cfg = NULL; - if (np) { - const struct of_device_id *of_id; - - of_id = of_match_device(ak4613_of_match, dev); - if (of_id) - regmap_cfg = of_id->data; - } else { + if (np) + regmap_cfg = of_device_get_match_data(dev); + else regmap_cfg = (const struct regmap_config *)id->driver_data; - } if (!regmap_cfg) return -EINVAL; -- 2.34.1

2 2

Re: [RFC PATCH 0/5] Apple Macs machine-level ASoC driver
by Mark Brown 31 Mar '22

31 Mar '22

On Thu, Mar 31, 2022 at 03:28:12PM +0200, Martin Povišer wrote: > > On 31. 3. 2022, at 14:34, Mark Brown <broonie(a)kernel.org> wrote: > > The broad issue here is that what you consider ridiculous someone else > > might have some bright ideas for configuring dynamically - if things are > > being exposed for dynamic configuration it's probably because someone > > wanted them, if the control is genuinely useless then it should just be > Well but these are codec drivers reused on different systems, it can both > be 'not genuinely useless’ on some system and ridiculous to leave open on > the systems I am trying to write drivers for. It wouldn't be the first time that we've had someone turn up with a new idea for how to configure an already existing bit of hardware, part of the reason for this approach is that people do get surprised by user creativity with their systems. > > The TDM swap thing you're mentioning looks like it's a left/right > > selection which people do use sometimes as a way of doing mono mixes and > > reorientation. The ISENSE/VSENSE is less obvious, though it's possible > > there's issues with not having enough slots on a heavily used TDM bus or > > sometimes disabling the speaker protection processing for whatever > > reason. > Not only that. On TAS2770 the default value for ‘ASI1 Sel’ is ‘I2C offset’ > meaning the speaker amp driver ignores my set_tdm_slot calls. If you tell > me it’s okay to change that behaviour and it won’t be considered backwards > compatibility breaking, that would be part of the solution I am seeking > here. Having the default state be muted or not routed is quite common, UCM files or equivalent are typically required for embedded style hardware like this. > But even then, what for example if the system has a single speaker (as it > does on the Mac mini to be covered by this driver) and the I2S bus is left > undriven for the duration of unused TDM slots? That may genuinely pose > a risk of people blowing their speakers by switching something in alsamixer. Right, so that's a more sensible and valid use case. We do have the platform_max feature available for precisely this reason - that's probably more appropriate here since if there's a danger of people blowing their speaker with a floating input they could also blow their speaker with just a very loud audio signal so limiting the volume people can set on the speaker driver seems sensible and would also cover them for misrouting. Whatever the device might pick up from noise on an undriven bus could also be played as audio down the bus. This does become a little fun with speaker protection as we'd want to raise the kernel limit so that userspace can dynamically manage the volume to contorl power (though that might be done with software control), but it's easy enoguh to raise limits later. On the other hand it seems like userspace might reasonably choose to do a mono mix for this output entirely in software, in which case telling the speaker amp to pick up one channel would make sense, or to just play out a stereo signal over I2S and have the amplifier do a mono mix and I'm not seeing why we'd force one or the other in the machine driver. > The ISENSE/VSENSE controls are also actually useless on these systems as we > are not doing anything to pick up the measured values (which are sent back > over the I2S lines). I don’t know if there can be driver conflict between Presumably someone might want to work on figuring that out though, and from a hardware safety point of view it would be better if they did. > two speaker amps trying to drive the I2S lines at the same time should > the user happen to enable SENSE facilities on more than one of them. > Now I can grudgingly study that and rule it out but I would rather hide > the controls altogether. Yes, having two devices driving the bus at the same time wouldn't be great. How is the TDM slot selection for the signals done in the hardware, I'm not seeing anything immediately obvious in the driver? I'd have thought that things would be implemented such that you could implement speaker protection on all speakers simultaneously but perhaps not. > That’s the reasoning anyway. To reiterate, seems to me the controls > are useless/confusing at best and dangerous at worst. I'm just not seeing an issue for the slot selection.

1 0

Re: [RFC PATCH 5/5] ASoC: Add macaudio machine driver
by Mark Brown 31 Mar '22

31 Mar '22

On Thu, Mar 31, 2022 at 02:08:51PM +0200, Martin Povišer wrote: > > On 31. 3. 2022, at 13:59, Mark Brown <broonie(a)kernel.org> wrote: > >> + for_each_rtd_components(rtd, i, component) > >> + snd_soc_component_set_jack(component, &ma->jack, NULL); > > What is the jack configuration this is attempting to describe? It looks > > like you have some dedicated speaker driver devices which are going to > > get attached to jacks here for example. > We know the speakers will ignore the set_jack call. There’s one jack and > this way we know the jack codec will attach to it, for speakers it’s a no-op. > (If you prefer I will special-case it to the jack codec.) It would be better to special case, this looks obviously wrong and will break if someone adds error handling. > >> + return !strcmp(name, pattern); > >> +} > > This looks worryingly like use case configuration. > I go over this in the cover letter! This is fixing the TDM slot selection > and disabling voltage/current sensing on the speaker amp codecs, which have > no business being exposed to userspace as options. This is not use case, > this not letting people blow their speakers from userspace. Your comments in the cover letter are all pretty vague too, that just says that these controls are "ridiculous" which isn't terribly specific about what the actual goal is. If it's just "I can't see why anyone would want to configure this" then that's a decision you're taking about what people might want to do which is broadly a use case configuration and the control should be left there in case someone comes up with an idea.

1 0

[PATCH v5 1/4] rename tas2764 to tas27xx
by Raphael-Xu 31 Mar '22

31 Mar '22

Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- sound/soc/codecs/Kconfig | 6 +++--- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{tas2764.c => tas27xx.c} | 0 sound/soc/codecs/{tas2764.h => tas27xx.h} | 0 4 files changed, 5 insertions(+), 5 deletions(-) rename sound/soc/codecs/{tas2764.c => tas27xx.c} (100%) rename sound/soc/codecs/{tas2764.h => tas27xx.h} (100%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f46a22660103..2c4595f6f7ec 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -213,7 +213,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_STI_SAS imply SND_SOC_TAS2552 imply SND_SOC_TAS2562 - imply SND_SOC_TAS2764 + imply SND_SOC_TAS27XX imply SND_SOC_TAS2770 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X @@ -1472,8 +1472,8 @@ config SND_SOC_TAS2562 tristate "Texas Instruments TAS2562 Mono Audio amplifier" depends on I2C -config SND_SOC_TAS2764 - tristate "Texas Instruments TAS2764 Mono Audio amplifier" +config SND_SOC_TAS27XX + tristate "Texas Instruments TAS2764/TAS2780 Mono Audio amplifier" depends on I2C config SND_SOC_TAS2770 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8637e9e869e3..15ceb95d42f5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -337,7 +337,7 @@ snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o snd-soc-tas2562-objs := tas2562.o -snd-soc-tas2764-objs := tas2764.o +snd-soc-tas27xx-objs := tas27xx.o # Mux snd-soc-simple-mux-objs := simple-mux.o @@ -574,7 +574,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o -obj-$(CONFIG_SND_SOC_TAS2764) += snd-soc-tas2764.o +obj-$(CONFIG_SND_SOC_TAS27XX) += snd-soc-tas27xx.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas27xx.c similarity index 100% rename from sound/soc/codecs/tas2764.c rename to sound/soc/codecs/tas27xx.c diff --git a/sound/soc/codecs/tas2764.h b/sound/soc/codecs/tas27xx.h similarity index 100% rename from sound/soc/codecs/tas2764.h rename to sound/soc/codecs/tas27xx.h -- 2.35.1

4 8

[PATCH] ALSA: pcm: Fix potential AB/BA lock with buffer_mutex and mmap_lock
by Takashi Iwai 31 Mar '22

31 Mar '22

syzbot caught a potential deadlock between the PCM runtime->buffer_mutex and the mm->mmap_lock. It was brought by the recent fix to cover the racy read/write and other ioctls, and in that commit, I overlooked a (hopefully only) corner case that may take the revert lock, namely, the OSS mmap. The OSS mmap operation exceptionally allows to re-configure the parameters inside the OSS mmap syscall, where mm->mmap_mutex is already held. Meanwhile, the copy_from/to_user calls at read/write operations also take the mm->mmap_lock internally, hence it may lead to a AB/BA deadlock. A similar problem was already seen in the past and we fixed it with a refcount (in commit b248371628aa). The former fix covered only the call paths with OSS read/write and OSS ioctls, while we need to cover the concurrent access via both ALSA and OSS APIs now. This patch addresses the problem above by replacing the buffer_mutex lock in the read/write operations with a refcount similar as we've used for OSS. The new field, runtime->buffer_accessing, keeps the number of concurrent read/write operations. Unlike the former buffer_mutex protection, this protects only around the copy_from/to_user() calls; the other codes are basically protected by the PCM stream lock. The refcount can be a negative, meaning blocked by the ioctls. If a negative value is seen, the read/write aborts with -EBUSY. In the ioctl side, OTOH, they check this refcount, too, and set to a negative value for blocking unless it's already being accessed. Reported-by: syzbot+6e5c88838328e99c7e1c(a)syzkaller.appspotmail.com Fixes: dca947d4d26d ("ALSA: pcm: Fix races among concurrent read/write and buffer changes") Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/000000000000381a0d05db622a81@google.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/pcm.h | 1 + sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 9 +++++---- sound/core/pcm_native.c | 39 ++++++++++++++++++++++++++++++++------- 4 files changed, 39 insertions(+), 11 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 314f2779cab5..6b99310b5b88 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -402,6 +402,7 @@ struct snd_pcm_runtime { struct fasync_struct *fasync; bool stop_operating; /* sync_stop will be called */ struct mutex buffer_mutex; /* protect for buffer changes */ + atomic_t buffer_accessing; /* >0: in r/w operation, <0: blocked */ /* -- private section -- */ void *private_data; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index edd9849210f2..977d54320a5c 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -970,6 +970,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, runtime->status->state = SNDRV_PCM_STATE_OPEN; mutex_init(&runtime->buffer_mutex); + atomic_set(&runtime->buffer_accessing, 0); substream->runtime = runtime; substream->private_data = pcm->private_data; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a40a35e51fad..1fc7c50ffa62 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1906,11 +1906,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (avail >= runtime->twake) break; snd_pcm_stream_unlock_irq(substream); - mutex_unlock(&runtime->buffer_mutex); tout = schedule_timeout(wait_time); - mutex_lock(&runtime->buffer_mutex); snd_pcm_stream_lock_irq(substream); set_current_state(TASK_INTERRUPTIBLE); switch (runtime->status->state) { @@ -2221,7 +2219,6 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, nonblock = !!(substream->f_flags & O_NONBLOCK); - mutex_lock(&runtime->buffer_mutex); snd_pcm_stream_lock_irq(substream); err = pcm_accessible_state(runtime); if (err < 0) @@ -2276,6 +2273,10 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, err = -EINVAL; goto _end_unlock; } + if (!atomic_inc_unless_negative(&runtime->buffer_accessing)) { + err = -EBUSY; + goto _end_unlock; + } snd_pcm_stream_unlock_irq(substream); if (!is_playback) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_CPU); @@ -2284,6 +2285,7 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (is_playback) snd_pcm_dma_buffer_sync(substream, SNDRV_DMA_SYNC_DEVICE); snd_pcm_stream_lock_irq(substream); + atomic_dec(&runtime->buffer_accessing); if (err < 0) goto _end_unlock; err = pcm_accessible_state(runtime); @@ -2313,7 +2315,6 @@ snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - mutex_unlock(&runtime->buffer_mutex); return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } EXPORT_SYMBOL(__snd_pcm_lib_xfer); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 704fdc9ebf91..4adaee62ef33 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -685,6 +685,24 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm, return 0; } +/* acquire buffer_mutex; if it's in r/w operation, return -EBUSY, otherwise + * block the further r/w operations + */ +static int snd_pcm_buffer_access_lock(struct snd_pcm_runtime *runtime) +{ + if (!atomic_dec_unless_positive(&runtime->buffer_accessing)) + return -EBUSY; + mutex_lock(&runtime->buffer_mutex); + return 0; /* keep buffer_mutex, unlocked by below */ +} + +/* release buffer_mutex and clear r/w access flag */ +static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime) +{ + mutex_unlock(&runtime->buffer_mutex); + atomic_inc(&runtime->buffer_accessing); +} + #if IS_ENABLED(CONFIG_SND_PCM_OSS) #define is_oss_stream(substream) ((substream)->oss.oss) #else @@ -695,14 +713,16 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime; - int err = 0, usecs; + int err, usecs; unsigned int bits; snd_pcm_uframes_t frames; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - mutex_lock(&runtime->buffer_mutex); + err = snd_pcm_buffer_access_lock(runtime); + if (err < 0) + return err; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: @@ -820,7 +840,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_lib_free_pages(substream); } unlock: - mutex_unlock(&runtime->buffer_mutex); + snd_pcm_buffer_access_unlock(runtime); return err; } @@ -865,7 +885,9 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - mutex_lock(&runtime->buffer_mutex); + result = snd_pcm_buffer_access_lock(runtime); + if (result < 0) + return result; snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_SETUP: @@ -884,7 +906,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); cpu_latency_qos_remove_request(&substream->latency_pm_qos_req); unlock: - mutex_unlock(&runtime->buffer_mutex); + snd_pcm_buffer_access_unlock(runtime); return result; } @@ -1369,12 +1391,15 @@ static int snd_pcm_action_nonatomic(const struct action_ops *ops, /* Guarantee the group members won't change during non-atomic action */ down_read(&snd_pcm_link_rwsem); - mutex_lock(&substream->runtime->buffer_mutex); + res = snd_pcm_buffer_access_lock(substream->runtime); + if (res < 0) + goto unlock; if (snd_pcm_stream_linked(substream)) res = snd_pcm_action_group(ops, substream, state, false); else res = snd_pcm_action_single(ops, substream, state); - mutex_unlock(&substream->runtime->buffer_mutex); + snd_pcm_buffer_access_unlock(substream->runtime); + unlock: up_read(&snd_pcm_link_rwsem); return res; } -- 2.31.1

2 1

[v1] ASoC: qcom: Add driver support for ALC5682I-VS.
by Judy Hsiao 31 Mar '22

31 Mar '22

Add Machine driver support for ALC5682I-VS codec. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> --- sound/soc/qcom/Kconfig | 2 + sound/soc/qcom/sc7280.c | 104 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 106 insertions(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 28d0dfb4033c..750653404ba3 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -197,6 +197,8 @@ config SND_SOC_SC7280 select SND_SOC_LPASS_MACRO_COMMON imply SND_SOC_LPASS_RX_MACRO imply SND_SOC_LPASS_TX_MACRO + select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S help Add support for audio on Qualcomm Technologies Inc. SC7280 SoC-based systems. diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index bd0bf9c8cb28..a3e6f61e92cb 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -12,14 +12,20 @@ #include <sound/jack.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/rt5682s.h> #include <linux/soundwire/sdw.h> +#include "../codecs/rt5682.h" +#include "../codecs/rt5682s.h" #include "common.h" #include "lpass.h" +#define DEFAULT_MCLK_RATE 19200000 +#define RT5682_PLL1_FREQ (48000 * 512) struct sc7280_snd_data { struct snd_soc_card card; struct sdw_stream_runtime *sruntime[LPASS_MAX_PORTS]; + u32 pri_mi2s_clk_count; struct snd_soc_jack hs_jack; struct snd_soc_jack hdmi_jack; bool jack_setup; @@ -69,6 +75,7 @@ static int sc7280_headset_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } switch (cpu_dai->id) { + case MI2S_PRIMARY: case LPASS_CDC_DMA_RX0: case LPASS_CDC_DMA_TX3: for_each_rtd_codec_dais(rtd, i, codec_dai) { @@ -110,11 +117,49 @@ static int sc7280_hdmi_init(struct snd_soc_pcm_runtime *rtd) return snd_soc_component_set_jack(component, &pdata->hdmi_jack, NULL); } +static int sc7280_rt5682_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct snd_soc_card *card = rtd->card; + struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); + int ret; + + if (++data->pri_mi2s_clk_count == 1) { + snd_soc_dai_set_sysclk(cpu_dai, + LPASS_MCLK0, + DEFAULT_MCLK_RATE, + SNDRV_PCM_STREAM_PLAYBACK); + } + snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_CBS_CFS | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_I2S); + + ret = snd_soc_dai_set_pll(codec_dai, RT5682S_PLL1, RT5682S_PLL_S_BCLK1, + 1536000, RT5682_PLL1_FREQ); + if (ret) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682S_SCLK_S_PLL1, + RT5682_PLL1_FREQ, + SND_SOC_CLOCK_IN); + + if (ret) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", + ret); + + return 0; +} + static int sc7280_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); switch (cpu_dai->id) { + case MI2S_PRIMARY: case LPASS_CDC_DMA_TX3: return sc7280_headset_init(rtd); case LPASS_CDC_DMA_RX0: @@ -227,10 +272,66 @@ static int sc7280_snd_hw_free(struct snd_pcm_substream *substream) return 0; } +static void sc7280_snd_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + + switch (cpu_dai->id) { + case MI2S_PRIMARY: + if (--data->pri_mi2s_clk_count == 0) { + snd_soc_dai_set_sysclk(cpu_dai, + LPASS_MCLK0, + 0, + SNDRV_PCM_STREAM_PLAYBACK); + } + break; + case MI2S_SECONDARY: + break; + case LPASS_DP_RX: + break; + default: + dev_err(rtd->dev, "%s: invalid dai id 0x%x\n", __func__, + cpu_dai->id); + break; + } +} + +static int sc7280_snd_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int ret = 0; + + switch (cpu_dai->id) { + case MI2S_PRIMARY: + ret = sc7280_rt5682_init(rtd); + break; + case MI2S_SECONDARY: + break; + case LPASS_DP_RX: + break; + default: + dev_err(rtd->dev, "%s: invalid dai id 0x%x\n", __func__, + cpu_dai->id); + return -EINVAL; + } + return ret; +} + static const struct snd_soc_ops sc7280_ops = { + .startup = sc7280_snd_startup, .hw_params = sc7280_snd_hw_params, .hw_free = sc7280_snd_hw_free, .prepare = sc7280_snd_prepare, + .shutdown = sc7280_snd_shutdown, +}; + +static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), }; static int sc7280_snd_platform_probe(struct platform_device *pdev) @@ -252,6 +353,9 @@ static int sc7280_snd_platform_probe(struct platform_device *pdev) card->driver_name = "SC7280"; card->dev = dev; + card->dapm_widgets = sc7280_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sc7280_snd_widgets); + ret = qcom_snd_parse_of(card); if (ret) return ret; -- 2.35.1.1021.g381101b075-goog

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[PATCH 0/7] meson: refactor common part of configuration
by GitHub pull_request - opened 31 Mar '22

31 Mar '22

alsa-project/alsa-gobject pull request #66 was opened from takaswie: In this project, meson configuration for included libraries consists of two parts; common and specific. During development, the common part gains differences between libraries. This patchset refactors meson configuration to increase the same expression in the common part. ``` Takashi Sakamoto (7): timer: code refactoring for search path of alsatimer header meson: aggregate dependency declaration on glib/gobject and libudev meson: aggregate module imports meson: fulfill missing sources for gir scanner which are blank meson: move pkg-config description to metadata part meson: aggregate and move extra gir to metadata part meson: move path to top header to metadata part meson.build | 2 -- src/ctl/meson.build | 37 +++++++++++++++------------------- src/hwdep/meson.build | 39 ++++++++++++++++-------------------- src/meson.build | 15 ++++++++++++++ src/rawmidi/meson.build | 38 ++++++++++++++++------------------- src/seq/meson.build | 44 ++++++++++++++++++----------------------- src/timer/meson.build | 40 +++++++++++++++++-------------------- 7 files changed, 102 insertions(+), 113 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/66 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/66.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[v7 0/4] ASoC: mediatek: mt8192: support rt1015p_rt5682s
by Jiaxin Yu 30 Mar '22

30 Mar '22

From: Jiaxin Yu <jiaxin.yu(a)mediatek.corp-partner.google.com> The series reuses mt8192-mt6359-rt10150rt5682.c for supporting machine driver with rt1015p speaker amplifier and rt5682s headset codec. Changes from v6: - "speaker-codec" changes to "speaker-codecs" due to there may be two speaker codec. Changes from v5: - "mediatek,headset-codec" and "mediatek,speaker-codec" drop prefix and move to properties from patternProperties. Changes form v4: - split a large patch into three small patches for easy reviewing - correct coding style Changes from v3: - fix build error: too many arguments for format [-Werror-format-extra-args] Changes from v2: - fix build warnings such as "data argument not used by format string" Changes from v1: - uses the snd_soc_of_get_dai_link_codecs to complete the configuration of dai_link's codecs - uses definitions to simplifies card name and compatible name Jiaxin Yu (4): ASoC: dt-bindings: mt8192-mt6359: add new compatible and new properties ASoC: mediatek: mt8192: refactor for I2S3 DAI link of speaker ASoC: mediatek: mt8192: refactor for I2S8/I2S9 DAI links of headset ASoC: mediatek: mt8192: support rt1015p_rt5682s .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 32 +++ sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 201 +++++++++++------- 3 files changed, 156 insertions(+), 78 deletions(-) -- 2.18.0

4 14

[GIT PULL] ASoC fixes for v5.18
by Mark Brown 30 Mar '22

30 Mar '22

The following changes since commit 49a24e9d9c740d3bd8b1200f225f67d45e3d68a5: Make the SOF control, PCM and PM code IPC agnostic (2022-03-18 20:11:08 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.18 for you to fetch changes up to 664d66dc0a64b32e60a5ad59a9aebb08676a612b: ASoC: SOF: Intel: Fix build error without SND_SOC_SOF_PCI_DEV (2022-03-25 16:09:28 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.18 A few fixes that came in during the merge window, all fairly routine. ---------------------------------------------------------------- Jiaxin Yu (1): ASoC: mediatek: mt6358: add missing EXPORT_SYMBOLs Meng Tang (2): ASoC: fsl-asoc-card: Fix jack_event() always return 0 ASoC: rockchip: i2s_tdm: Fixup config for SND_SOC_DAIFMT_DSP_A/B Zheng Bin (1): ASoC: SOF: Intel: Fix build error without SND_SOC_SOF_PCI_DEV sound/soc/codecs/mt6358.c | 4 ++++ sound/soc/fsl/fsl-asoc-card.c | 12 ++++-------- sound/soc/rockchip/rockchip_i2s_tdm.c | 10 +++++----- sound/soc/sof/intel/Kconfig | 1 + 4 files changed, 14 insertions(+), 13 deletions(-)

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