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

[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

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[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

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

10 Sep '22

alsa-project/alsa-utils issue #170 was opened from BlastBolt5: I have been working on a script to control pipewire sources and sinks using faders on my midi keyboard but it 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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