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[PATCH] ASoC: wm8978: Support the recording function of codec
by Mengchen Li 12 Sep '22

12 Sep '22

If we want to use the microphone function of the wm8978 codec, we must enable the microphone bias circuit and power up the bias circuit,otherwise the microphone can not vibrate for sound source sampling.The bit that controls the output voltage of the bias circuit is the fourth bit of the WM8978_POWER_MANAGEMENT_1 register,so this bit must be set to 1.This patch can support the recording function of the wm8978 codec. Signed-off-by: Mengchen Li <mengchenli64(a)gmail.com> --- sound/soc/codecs/wm8978.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index a682f80..fdd5429 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -822,7 +822,9 @@ static int wm8978_hw_params(struct snd_pcm_substream *substream, snd_soc_component_write(component, WM8978_AUDIO_INTERFACE, iface_ctl); snd_soc_component_write(component, WM8978_ADDITIONAL_CONTROL, add_ctl); - + /* Enable MICBEN */ + snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_1, + snd_soc_component_read(component, WM8978_POWER_MANAGEMENT_1) | 0x10); if (wm8978->sysclk != current_clk_id) { if (wm8978->sysclk == WM8978_PLL) /* Run CODEC from PLL instead of MCLK */ -- 2.7.4

2 1

[PATCH v1] ALSA: dummy: Add customizable volume min/max.
by YJ Lee 12 Sep '22

12 Sep '22

Add module parameters to support customized min/max volume leveling, which will be useful to test devices with different volume granularity. Signed-off-by: YJ Lee <yunjunlee(a)chromium.org> --- sound/drivers/dummy.c | 34 ++++++++++++++++++++++++---------- 1 file changed, 24 insertions(+), 10 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 2a7fc49c1a7c5..64fb2778f1e9a 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -42,6 +42,8 @@ MODULE_LICENSE("GPL"); #define USE_CHANNELS_MAX 2 #define USE_PERIODS_MIN 1 #define USE_PERIODS_MAX 1024 +#define USE_MIXER_VOLUME_LEVEL_MIN -50 +#define USE_MIXER_VOLUME_LEVEL_MAX 100 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -50,6 +52,8 @@ static char *model[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = NULL}; static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; +static int mixer_volume_level_min = USE_MIXER_VOLUME_LEVEL_MIN; +static int mixer_volume_level_max = USE_MIXER_VOLUME_LEVEL_MAX; #ifdef CONFIG_HIGH_RES_TIMERS static bool hrtimer = 1; #endif @@ -69,6 +73,10 @@ module_param_array(pcm_substreams, int, NULL, 0444); MODULE_PARM_DESC(pcm_substreams, "PCM substreams # (1-128) for dummy driver."); //module_param_array(midi_devs, int, NULL, 0444); //MODULE_PARM_DESC(midi_devs, "MIDI devices # (0-2) for dummy driver."); +module_param(mixer_volume_level_min, int, 0444); +MODULE_PARM_DESC(mixer_volume_level_min, "Minimum mixer volume level for dummy driver. Default: -50"); +module_param(mixer_volume_level_max, int, 0444); +MODULE_PARM_DESC(mixer_volume_level_max, "Maximum mixer volume level for dummy driver. Default: 100"); module_param(fake_buffer, bool, 0444); MODULE_PARM_DESC(fake_buffer, "Fake buffer allocations."); #ifdef CONFIG_HIGH_RES_TIMERS @@ -713,8 +721,8 @@ static int snd_dummy_volume_info(struct snd_kcontrol *kcontrol, { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; - uinfo->value.integer.min = -50; - uinfo->value.integer.max = 100; + uinfo->value.integer.min = mixer_volume_level_min; + uinfo->value.integer.max = mixer_volume_level_max; return 0; } @@ -739,15 +747,15 @@ static int snd_dummy_volume_put(struct snd_kcontrol *kcontrol, int left, right; left = ucontrol->value.integer.value[0]; - if (left < -50) - left = -50; - if (left > 100) - left = 100; + if (left < mixer_volume_level_min) + left = mixer_volume_level_min; + if (left > mixer_volume_level_max) + left = mixer_volume_level_max; right = ucontrol->value.integer.value[1]; - if (right < -50) - right = -50; - if (right > 100) - right = 100; + if (right < mixer_volume_level_min) + right = mixer_volume_level_min; + if (right > mixer_volume_level_max) + right = mixer_volume_level_max; spin_lock_irq(&dummy->mixer_lock); change = dummy->mixer_volume[addr][0] != left || dummy->mixer_volume[addr][1] != right; @@ -1076,6 +1084,12 @@ static int snd_dummy_probe(struct platform_device *devptr) dummy->pcm_hw.channels_max = m->channels_max; } + if (mixer_volume_level_min > mixer_volume_level_max) { + pr_warn("snd-dummy: Invalid mixer volume level: min=%d, max=%d. Fall back to default value.\n", + mixer_volume_level_min, mixer_volume_level_max); + mixer_volume_level_min = USE_MIXER_VOLUME_LEVEL_MIN; + mixer_volume_level_max = USE_MIXER_VOLUME_LEVEL_MAX; + } err = snd_card_dummy_new_mixer(dummy); if (err < 0) return err; -- 2.31.0

2 2

[PATCH v1] ALSA: dummy: Fix trailing whitespaces.
by YJ Lee 12 Sep '22

12 Sep '22

Fix checkpatch.pl ERROR: trailing whitespaces. Signed-off-by: YJ Lee <yunjunlee(a)chromium.org> --- sound/drivers/dummy.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 2a7fc49c1a7c5..fcf1ee00bd214 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -296,7 +296,7 @@ static void dummy_systimer_callback(struct timer_list *t) struct dummy_systimer_pcm *dpcm = from_timer(dpcm, t, timer); unsigned long flags; int elapsed = 0; - + spin_lock_irqsave(&dpcm->lock, flags); dummy_systimer_update(dpcm); dummy_systimer_rearm(dpcm); @@ -717,7 +717,7 @@ static int snd_dummy_volume_info(struct snd_kcontrol *kcontrol, uinfo->value.integer.max = 100; return 0; } - + static int snd_dummy_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -766,7 +766,7 @@ static const DECLARE_TLV_DB_SCALE(db_scale_dummy, -4500, 30, 0); .private_value = addr } #define snd_dummy_capsrc_info snd_ctl_boolean_stereo_info - + static int snd_dummy_capsrc_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1100,7 +1100,7 @@ static int snd_dummy_suspend(struct device *pdev) snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); return 0; } - + static int snd_dummy_resume(struct device *pdev) { struct snd_card *card = dev_get_drvdata(pdev); -- 2.37.2.789.g6183377224-goog

2 2

[PATCH] ASoC: ts3a227e: allow enabling the jack detect in driver
by Astrid Rost 12 Sep '22

12 Sep '22

ti,jack-detect enables the jack detection input device Signed-off-by: Astrid Rost <astrid.rost(a)axis.com> --- sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/ts3a227e.c | 62 ++++++++++++++++++++++++++++++++++++- 2 files changed, 63 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..cb86e52cd02f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1671,6 +1671,8 @@ config SND_SOC_TLV320ADCX140 config SND_SOC_TS3A227E tristate "TI Headset/Mic detect and keypress chip" depends on I2C + select SND_JACK + select SND_JACK_INPUT_DEV config SND_SOC_TSCS42XX tristate "Tempo Semiconductor TSCS42xx CODEC" diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index d8ab0810fceb..b5d0d32cafdb 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -38,6 +38,32 @@ static const int ts3a227e_buttons[] = { SND_JACK_BTN_3, }; +/* Headphones jack detection DAPM pin */ +static struct snd_soc_jack_pin ts3a227e_jack_pins[] = { + + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Internal Speaker", + /* disable speaker when hp jack is inserted */ + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Internal Mic", + /* disable microphone when microphone jack is inserted */ + .mask = SND_JACK_MICROPHONE, + .invert = 1, + }, + +}; + #define TS3A227E_NUM_BUTTONS 4 #define TS3A227E_JACK_MASK (SND_JACK_HEADPHONE | \ SND_JACK_MICROPHONE | \ @@ -250,7 +276,12 @@ int ts3a227e_enable_jack_detect(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(ts3a227e_enable_jack_detect); -static struct snd_soc_component_driver ts3a227e_soc_driver; +static int ts3a227e_probe(struct snd_soc_component *component); + +static const struct snd_soc_component_driver ts3a227e_soc_driver = { + .name = "Audio Accessory Detection ts3a227e", + .probe = ts3a227e_probe, +}; static const struct regmap_config ts3a227e_regmap_config = { .val_bits = 8, @@ -355,6 +386,35 @@ static int ts3a227e_resume(struct device *dev) } #endif +static int ts3a227e_probe(struct snd_soc_component *component) +{ + int ret = 0; + bool enable = 0; + struct snd_soc_card *card = component->card; + struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); + + enable = device_property_read_bool(component->dev, "ti,jack-detect"); + if (enable) { + /* Enable Headset and 4 Buttons Jack detection */ + ts3a227e->jack = devm_kzalloc(component->dev, + sizeof(struct snd_soc_jack), GFP_KERNEL); + + ret = snd_soc_card_jack_new_pins(card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + ts3a227e->jack, + ts3a227e_jack_pins, + ARRAY_SIZE(ts3a227e_jack_pins)); + if (ret) + return ret; + + ret = ts3a227e_enable_jack_detect(component, ts3a227e->jack); + } + + return ret; +} + static const struct dev_pm_ops ts3a227e_pm = { SET_SYSTEM_SLEEP_PM_OPS(ts3a227e_suspend, ts3a227e_resume) }; -- 2.20.1

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Buffers used in ASoC Intel Atom SST driver
by Takashi Iwai 11 Sep '22

11 Sep '22

Hi, while cleaning up / fixing the memory allocation code, I noticed that Intel SST driver is the only driver that currently uses the SNDRV_DMA_TYPE_CONTINUOUS pages for its PCM buffer. Is there any reason not to use the standard device buffer (SNDRV_DMA_TYPE_DEV)? i.e. the change like below. thanks, Takashi -- 8< -- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ASoC: Intel: sst: Switch to standard device pages Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index a56dd48c045f..c75616a5fd0a 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -676,10 +676,9 @@ static int sst_soc_pcm_new(struct snd_soc_component *component, if (dai->driver->playback.channels_min || dai->driver->capture.channels_min) { - snd_pcm_set_managed_buffer_all(pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_DMA), - SST_MIN_BUFFER, SST_MAX_BUFFER); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, + pcm->card->dev, + SST_MIN_BUFFER, SST_MAX_BUFFER); } return 0; } -- 2.35.3

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[PATCH] ASoC: nau8825: Add ADCOUT IO drive strength control
by David Lin 11 Sep '22

11 Sep '22

Add a property to control the driving of ADCOUT. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- Documentation/devicetree/bindings/sound/nau8825.txt | 2 ++ sound/soc/codecs/nau8825.c | 6 ++++++ sound/soc/codecs/nau8825.h | 3 +++ 3 files changed, 11 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt index 388a7bc60b1f..54972cc2ca1a 100644 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -71,6 +71,8 @@ Optional properties: - nuvoton,crosstalk-enable: make crosstalk function enable if set. + - nuvoton,adcout-drive-str: ADCOUT IO drive strength. 0 - normal, 1 - stronger. + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the clocks described in clock-names - clock-names: should include "mclk" for the MCLK master clock diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 8213273f501e..fc00e600cf2c 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1983,6 +1983,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825) /* Disable short Frame Sync detection logic */ regmap_update_bits(regmap, NAU8825_REG_LEFT_TIME_SLOT, NAU8825_DIS_FS_SHORT_DET, NAU8825_DIS_FS_SHORT_DET); + /* ADCDAT IO drive strength control */ + regmap_update_bits(remap, NAU8825_REG_CHARGE_PUMP, + NAU8825_ADCOUT_DS_MASK, + nau8825->adcout_ds << NAU8825_ADCOUT_DS_SFT); } static const struct regmap_config nau8825_regmap_config = { @@ -2521,6 +2525,7 @@ static void nau8825_print_device_properties(struct nau8825 *nau8825) nau8825->jack_eject_debounce); dev_dbg(dev, "crosstalk-enable: %d\n", nau8825->xtalk_enable); + dev_dbg(dev, "adcout-drive-str: %d\n", nau8825->adcout_ds); } static int nau8825_read_device_properties(struct device *dev, @@ -2587,6 +2592,7 @@ static int nau8825_read_device_properties(struct device *dev, nau8825->jack_eject_debounce = 0; nau8825->xtalk_enable = device_property_read_bool(dev, "nuvoton,crosstalk-enable"); + nau8825->adcout_ds = device_property_read_bool(dev, "nuvoton,adcout-drive-str"); nau8825->mclk = devm_clk_get(dev, "mclk"); if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 887bbff03ec6..6d112b6145df 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -418,6 +418,8 @@ #define NAU8825_POWERUP_HP_DRV_L (1 << 0) /* CHARGE_PUMP (0x80) */ +#define NAU8825_ADCOUT_DS_SFT 12 +#define NAU8825_ADCOUT_DS_MASK (1 << NAU8825_ADCOUT_DS_SFT) #define NAU8825_JAMNODCLOW (1 << 10) #define NAU8825_POWER_DOWN_DACR (1 << 9) #define NAU8825_POWER_DOWN_DACL (1 << 8) @@ -477,6 +479,7 @@ struct nau8825 { int imp_rms[NAU8825_XTALK_IMM]; int xtalk_enable; bool xtalk_baktab_initialized; /* True if initialized. */ + bool adcout_ds; }; int nau8825_enable_jack_detect(struct snd_soc_component *component, -- 2.25.1

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[PATCH] ASoC: wm8731: fix build when I2C is not enabled
by Randy Dunlap 11 Sep '22

11 Sep '22

Building wm8731-i2c.c has a kconfig warning and build errors when I2C is not enabled, so prevent that kconfig-uration. WARNING: unmet direct dependencies detected for SND_SOC_WM8731_I2C Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n] Selected by [m]: - SND_SOC_DB1200 [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AU1XPSC [=m] and subsequent build errors: ../sound/soc/codecs/wm8731-i2c.c:64:1: warning: data definition has no type or storage class 64 | module_i2c_driver(wm8731_i2c_driver); ../sound/soc/codecs/wm8731-i2c.c:64:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] ../sound/soc/codecs/wm8731-i2c.c:64:1: warning: parameter names (without types) in function declaration ../sound/soc/codecs/wm8731-i2c.c:55:26: warning: 'wm8731_i2c_driver' defined but not used [-Wunused-variable] 55 | static struct i2c_driver wm8731_i2c_driver = { Fixes: 9dc15f81baf2 ("ASoC: wm8731: Factor out the I2C and SPI bus code into separate modules") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai --- sound/soc/au1x/Kconfig | 1 + 1 file changed, 1 insertion(+) --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_DB1000 config SND_SOC_DB1200 tristate "DB1200/DB1300/DB1550 Audio support" depends on SND_SOC_AU1XPSC + depends on I2C select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC select SND_SOC_WM9712

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[PATCH] ALSA: hda: Fix hang at HD-audio codec unbinding due to refcount saturation
by Takashi Iwai 10 Sep '22

10 Sep '22

We fixed the potential deadlock at dynamic unbinding the HD-audio codec at the commit 7206998f578d ("ALSA: hda: Fix potential deadlock at codec unbinding"), but ironically, this caused another potential deadlock. The current code uses refcount_dec() and waits for the pending task with wait_event for dropping the refcount to 0. This works fine when PCMs are assigned and actually waiting for the refcount drop. Meanwhile, when there was no PCM assigned, the refcount_dec() call itself was supposed to drop to zero -- alas, it doesn't in reality; refcount_dec() complains, spews kernel warning and it saturates instead of dropping to 0, due to the nature of refcount_dec() implementation. This eventually blocks the wait_event() wakeup and the code get stuck there. For avoiding the problem, we call refcount_dec_and_test() and skips the sync-wait if it already reaches to zero. The patch does a slight code reshuffling to make sure to invoke other disconnect calls before the sync-wait, too. Fixes: 7206998f578d ("ALSA: hda: Fix potential deadlock at codec unbinding") Reported-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Tested-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/YxtflWQnslMHVlU7@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_bind.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index cae9a975cbcc..1a868dd9dc4b 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -157,10 +157,10 @@ static int hda_codec_driver_remove(struct device *dev) return codec->bus->core.ext_ops->hdev_detach(&codec->core); } - refcount_dec(&codec->pcm_ref); snd_hda_codec_disconnect_pcms(codec); snd_hda_jack_tbl_disconnect(codec); - wait_event(codec->remove_sleep, !refcount_read(&codec->pcm_ref)); + if (!refcount_dec_and_test(&codec->pcm_ref)) + wait_event(codec->remove_sleep, !refcount_read(&codec->pcm_ref)); snd_power_sync_ref(codec->bus->card); if (codec->patch_ops.free) -- 2.35.3

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hda codec unbind refcount hang
by Ville Syrjälä 10 Sep '22

10 Sep '22

Hi Takashi, commit 7206998f578d ("ALSA: hda: Fix potential deadlock at codec unbinding") introduced a problem on at least one of my older machines. The problem happens when hda_codec_driver_remove() encounters a codec without any pcms (and thus the refcount is 1) and tries to call refcount_dec(). Turns out refcount_dec() doesn't like to be used for dropping the refcount to 0, and instead if spews a warning and does its saturate thing. The subsequent wait_event() is then permanently stuck waiting on the saturated refcount. I've definitely seen the same kind of pattern used elsewhere in the kernel as well, so the fact that refcount_t can't be used to implement it is a bit of surprise to me. I guess most other places still use atomic_t instead. -- Ville Syrjälä Intel

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aseqdump crashing after running for a bit
by GitHub issues - edited 10 Sep '22

10 Sep '22

alsa-project/alsa-utils issue #170 was edited from BlastBolt5: I have been working on a script to control pipewire sources and sinks using faders on my midi keyboard but aseqdump keeps erroring out with > Port unsubscribed 128:0 I don't have any other midi device to test it out and see if it's specific to this conrtoller but I'm using the Korg nanoKontrol2. If i need to provide anything else just let me know. Issue URL : https://github.com/alsa-project/alsa-utils/issues/170 Repository URL: https://github.com/alsa-project/alsa-utils

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