May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: Add quirk for HP EliteBook G10 laptops
by Vitaly Rodionov 11 May '23

11 May '23

Add support for HP EliteBook 835/845/845W/865 G10 laptops with CS35L41 amplifiers on I2C/SPI bus connected to Realtek codec. Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 172ffc2c332b..9e509356790f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9458,7 +9458,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b42, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), @@ -9469,8 +9469,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b47, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b63, "HP Elite Dragonfly 13.5 inch G4", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8b77, "HP ElieBook 865 G10", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b87, "HP", ALC236_FIXUP_HP_GPIO_LED), @@ -9481,6 +9486,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c26, "HP HP EliteBook 800G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.34.1

2 1

Newest Update for Fedora 37 (alsa-lib-1.2.9-1.fc37) causes crackling with device
by GitHub issues - opened 11 May '23

11 May '23

alsa-project/alsa-ucm-conf issue #314 was opened from ClaraCrazy: First bug report here + not skilled with debugging, so please go easy on me :smile: OS: Fedora 37 Alsa-lib: 1.2.9-1 Issue: Audio crackling Audio Device: `Logic3 TX101 5.1 Soundbar` connected via `Scarlett 2i2 3rd Gen audio interface (USB)` Reverted lib back to `alsa-lib.1.2.8-2.fc37` (which in turn also downgraded alsa-ucm and alsa-utils again) and issues are fixed. I have no idea how to debug this, but would love to help in any way possible, sadly there was no issue template that would give me infos on what logs are useful / where to get those so uh.. please someone just point me in the right direction :) Thanks for your time Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/314 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Newest Update for Fedora 37 (alsa-lib-1.2.9-1.fc37) causes crackling with device
by GitHub issues - edited 11 May '23

11 May '23

alsa-project/alsa-ucm-conf issue #314 was edited from ClaraCrazy: First bug report here + not skilled with debugging, so please go easy on me :smile: OS: Fedora 37 Alsa-lib: 1.2.9-1 Issue: Audio crackling Audio Device: `Logic3 TX101 5.1 Soundbar` connected via `Scarlett 2i2 3rd Gen audio interface (USB)` Reverted lib back to `alsa-lib.1.2.8-2.fc37` (which in turn also downgraded alsa-ucm and alsa-utils again) and issues are fixed. I have no idea how to debug this, but would love to help in any way possible, sadly there was no issue template that would give me infos on what logs are useful / where to get those so uh.. please someone just point me in the right direction :) Thanks for your time Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/314 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: soc-pcm: Fix and cleanup DPCM locking
by yixuanjiang 11 May '23

11 May '23

From: Takashi Iwai <tiwai(a)suse.de> The existing locking for DPCM has several issues a) a confusing mix of card->mutex and card->pcm_mutex. b) a dpcm_lock spinlock added inconsistently and on paths that could be recursively taken. The use of irqsave/irqrestore was also overkill. The suggested model is: 1) The pcm_mutex is the top-most protection of BE links in the FE. The pcm_mutex is applied always on either the top PCM callbacks or the external call from DAPM, not taken in the internal functions. 2) the FE stream lock is taken in higher levels before invoking dpcm_be_dai_trigger() 3) when adding and deleting a BE, both the pcm_mutex and FE stream lock are taken. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> [clarification of commit message by plbossart] Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20211207173745.15850-4-pierre-louis.bossart@linux… Cc: stable(a)vger.kernel.org # 5.15.x Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 2 - sound/soc/soc-core.c | 1 - sound/soc/soc-pcm.c | 229 ++++++++++++++++++++++++++++--------------- 3 files changed, 152 insertions(+), 80 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e6dd8a257c5..5872a8864f3b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -893,8 +893,6 @@ struct snd_soc_card { struct mutex pcm_mutex; enum snd_soc_pcm_subclass pcm_subclass; - spinlock_t dpcm_lock; - int (*probe)(struct snd_soc_card *card); int (*late_probe)(struct snd_soc_card *card); int (*remove)(struct snd_soc_card *card); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b1749b920f4..a5b3ee69fb88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2339,7 +2339,6 @@ int snd_soc_register_card(struct snd_soc_card *card) mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); mutex_init(&card->pcm_mutex); - spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 3b673477f621..07ad98934c7a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,6 +27,31 @@ #include <sound/soc-link.h> #include <sound/initval.h> +static inline void snd_soc_dpcm_mutex_lock(struct snd_soc_pcm_runtime *rtd) +{ + mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); +} + +static inline void snd_soc_dpcm_mutex_unlock(struct snd_soc_pcm_runtime *rtd) +{ + mutex_unlock(&rtd->card->pcm_mutex); +} + +#define snd_soc_dpcm_mutex_assert_held(rtd) \ + lockdep_assert_held(&(rtd)->card->pcm_mutex) + +static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, + int stream) +{ + snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); +} + +static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, + int stream) +{ + snd_pcm_stream_unlock_irq(snd_soc_dpcm_get_substream(rtd, stream)); +} + #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -73,7 +98,6 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; struct snd_soc_dpcm *dpcm; ssize_t offset = 0; - unsigned long flags; /* FE state */ offset += scnprintf(buf + offset, size - offset, @@ -101,7 +125,6 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -122,7 +145,6 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); out: return offset; } @@ -145,11 +167,13 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + snd_soc_dpcm_mutex_lock(fe); for_each_pcm_streams(stream) if (snd_soc_dai_stream_valid(asoc_rtd_to_cpu(fe, 0), stream)) offset += dpcm_show_state(fe, stream, buf + offset, out_count - offset); + snd_soc_dpcm_mutex_unlock(fe); ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); @@ -221,14 +245,14 @@ static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); - snd_pcm_stream_lock_irq(substream); + snd_soc_dpcm_stream_lock_irq(fe, stream); if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { dpcm_fe_dai_do_trigger(substream, fe->dpcm[stream].trigger_pending - 1); fe->dpcm[stream].trigger_pending = 0; } fe->dpcm[stream].runtime_update = state; - snd_pcm_stream_unlock_irq(substream); + snd_soc_dpcm_stream_unlock_irq(fe, stream); } static void dpcm_set_be_update_state(struct snd_soc_pcm_runtime *be, @@ -256,7 +280,7 @@ void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai; int i; - lockdep_assert_held(&rtd->card->pcm_mutex); + snd_soc_dpcm_mutex_assert_held(rtd); for_each_rtd_dais(rtd, i, dai) snd_soc_dai_action(dai, stream, action); @@ -309,6 +333,8 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, { struct snd_soc_dpcm *dpcm; + snd_soc_dpcm_mutex_assert_held(fe); + for_each_dpcm_be(fe, dir, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -646,14 +672,14 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, return ret; } -static int soc_pcm_clean(struct snd_pcm_substream *substream, int rollback) +static int soc_pcm_clean(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream, int rollback) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_component *component; struct snd_soc_dai *dai; int i; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); if (!rollback) snd_soc_runtime_deactivate(rtd, substream->stream); @@ -665,9 +691,6 @@ static int soc_pcm_clean(struct snd_pcm_substream *substream, int rollback) soc_pcm_components_close(substream, rollback); - - mutex_unlock(&rtd->card->pcm_mutex); - snd_soc_pcm_component_pm_runtime_put(rtd, substream, rollback); for_each_rtd_components(rtd, i, component) @@ -682,9 +705,21 @@ static int soc_pcm_clean(struct snd_pcm_substream *substream, int rollback) * freed here. The cpu DAI, codec DAI, machine and components are also * shutdown. */ +static int __soc_pcm_close(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + return soc_pcm_clean(rtd, substream, 0); +} + +/* PCM close ops for non-DPCM streams */ static int soc_pcm_close(struct snd_pcm_substream *substream) { - return soc_pcm_clean(substream, 0); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + + snd_soc_dpcm_mutex_lock(rtd); + soc_pcm_clean(rtd, substream, 0); + snd_soc_dpcm_mutex_unlock(rtd); + return 0; } static int soc_hw_sanity_check(struct snd_pcm_substream *substream) @@ -730,21 +765,21 @@ static int soc_hw_sanity_check(struct snd_pcm_substream *substream) * then initialized and any private data can be allocated. This also calls * startup for the cpu DAI, component, machine and codec DAI. */ -static int soc_pcm_open(struct snd_pcm_substream *substream) +static int __soc_pcm_open(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_component *component; struct snd_soc_dai *dai; int i, ret = 0; + snd_soc_dpcm_mutex_assert_held(rtd); + for_each_rtd_components(rtd, i, component) pinctrl_pm_select_default_state(component->dev); ret = snd_soc_pcm_component_pm_runtime_get(rtd, substream); if (ret < 0) - goto pm_err; - - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + goto err; ret = soc_pcm_components_open(substream); if (ret < 0) @@ -786,16 +821,26 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_runtime_activate(rtd, substream->stream); ret = 0; err: - mutex_unlock(&rtd->card->pcm_mutex); -pm_err: if (ret < 0) { - soc_pcm_clean(substream, 1); + soc_pcm_clean(rtd, substream, 1); dev_err(rtd->dev, "%s() failed (%d)", __func__, ret); } return ret; } +/* PCM open ops for non-DPCM streams */ +static int soc_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_open(rtd, substream); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; +} + static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { /* @@ -811,13 +856,13 @@ static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) * rate, etc. This function is non atomic and can be called multiple times, * it can refer to the runtime info. */ -static int soc_pcm_prepare(struct snd_pcm_substream *substream) +static int __soc_pcm_prepare(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *dai; int i, ret = 0; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); ret = snd_soc_link_prepare(substream); if (ret < 0) @@ -845,14 +890,24 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(dai, 0, substream->stream); out: - mutex_unlock(&rtd->card->pcm_mutex); - if (ret < 0) dev_err(rtd->dev, "ASoC: %s() failed (%d)\n", __func__, ret); return ret; } +/* PCM prepare ops for non-DPCM streams */ +static int soc_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_prepare(rtd, substream); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; +} + static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, unsigned int mask) { @@ -864,13 +919,13 @@ static void soc_pcm_codec_params_fixup(struct snd_pcm_hw_params *params, interval->max = channels; } -static int soc_pcm_hw_clean(struct snd_pcm_substream *substream, int rollback) +static int soc_pcm_hw_clean(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream, int rollback) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *dai; int i; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); /* clear the corresponding DAIs parameters when going to be inactive */ for_each_rtd_dais(rtd, i, dai) { @@ -900,16 +955,28 @@ static int soc_pcm_hw_clean(struct snd_pcm_substream *substream, int rollback) snd_soc_dai_hw_free(dai, substream, rollback); } - mutex_unlock(&rtd->card->pcm_mutex); return 0; } /* * Frees resources allocated by hw_params, can be called multiple times */ +static int __soc_pcm_hw_free(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + return soc_pcm_hw_clean(rtd, substream, 0); +} + +/* hw_free PCM ops for non-DPCM streams */ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) { - return soc_pcm_hw_clean(substream, 0); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_hw_free(rtd, substream); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; } /* @@ -917,15 +984,15 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) * function can also be called multiple times and can allocate buffers * (using snd_pcm_lib_* ). It's non-atomic. */ -static int soc_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int __soc_pcm_hw_params(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai; struct snd_soc_dai *codec_dai; int i, ret = 0; - mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); + snd_soc_dpcm_mutex_assert_held(rtd); ret = soc_pcm_params_symmetry(substream, params); if (ret) @@ -997,16 +1064,27 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_pcm_component_hw_params(substream, params); out: - mutex_unlock(&rtd->card->pcm_mutex); - if (ret < 0) { - soc_pcm_hw_clean(substream, 1); + soc_pcm_hw_clean(rtd, substream, 1); dev_err(rtd->dev, "ASoC: %s() failed (%d)\n", __func__, ret); } return ret; } +/* hw_params PCM ops for non-DPCM streams */ +static int soc_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + int ret; + + snd_soc_dpcm_mutex_lock(rtd); + ret = __soc_pcm_hw_params(rtd, substream, params); + snd_soc_dpcm_mutex_unlock(rtd); + return ret; +} + static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); @@ -1124,7 +1202,8 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { struct snd_soc_dpcm *dpcm; - unsigned long flags; + + snd_soc_dpcm_mutex_assert_held(fe); /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1140,10 +1219,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); + snd_soc_dpcm_stream_lock_irq(fe, stream); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + snd_soc_dpcm_stream_unlock_irq(fe, stream); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1188,8 +1267,10 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; - unsigned long flags; + snd_soc_dpcm_mutex_assert_held(fe); + + snd_soc_dpcm_stream_lock_irq(fe, stream); for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", stream ? "capture" : "playback", @@ -1207,12 +1288,11 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) dpcm_remove_debugfs_state(dpcm); - spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); } + snd_soc_dpcm_stream_unlock_irq(fe, stream); } /* get BE for DAI widget and stream */ @@ -1426,12 +1506,9 @@ int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; - unsigned long flags; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) dpcm_set_be_update_state(dpcm->be, stream, SND_SOC_DPCM_UPDATE_NO); - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); } void dpcm_be_dai_stop(struct snd_soc_pcm_runtime *fe, int stream, @@ -1467,12 +1544,12 @@ void dpcm_be_dai_stop(struct snd_soc_pcm_runtime *fe, int stream, continue; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) { - soc_pcm_hw_free(be_substream); + __soc_pcm_hw_free(be, be_substream); be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; } } - soc_pcm_close(be_substream); + __soc_pcm_close(be, be_substream); be_substream->runtime = NULL; be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; } @@ -1520,7 +1597,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) stream ? "capture" : "playback", be->dai_link->name); be_substream->runtime = be->dpcm[stream].runtime; - err = soc_pcm_open(be_substream); + err = __soc_pcm_open(be, be_substream); if (err < 0) { be->dpcm[stream].users--; if (be->dpcm[stream].users < 0) @@ -1764,7 +1841,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dev_dbg(fe->dev, "ASoC: open FE %s\n", fe->dai_link->name); /* start the DAI frontend */ - ret = soc_pcm_open(fe_substream); + ret = __soc_pcm_open(fe, fe_substream); if (ret < 0) goto unwind; @@ -1795,6 +1872,8 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int stream = substream->stream; + snd_soc_dpcm_mutex_assert_held(fe); + dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ @@ -1803,7 +1882,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name); /* now shutdown the frontend */ - soc_pcm_close(substream); + __soc_pcm_close(fe, substream); /* run the stream stop event */ dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); @@ -1848,7 +1927,7 @@ void dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) dev_dbg(be->dev, "ASoC: hw_free BE %s\n", be->dai_link->name); - soc_pcm_hw_free(be_substream); + __soc_pcm_hw_free(be, be_substream); be->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; } @@ -1859,13 +1938,13 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int stream = substream->stream; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); dev_dbg(fe->dev, "ASoC: hw_free FE %s\n", fe->dai_link->name); /* call hw_free on the frontend */ - soc_pcm_hw_free(substream); + soc_pcm_hw_clean(fe, substream, 0); /* only hw_params backends that are either sinks or sources * to this frontend DAI */ @@ -1874,7 +1953,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_HW_FREE; dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); return 0; } @@ -1918,7 +1997,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) dev_dbg(be->dev, "ASoC: hw_params BE %s\n", be->dai_link->name); - ret = soc_pcm_hw_params(be_substream, &dpcm->hw_params); + ret = __soc_pcm_hw_params(be, be_substream, &dpcm->hw_params); if (ret < 0) goto unwind; @@ -1948,7 +2027,7 @@ int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; - soc_pcm_hw_free(be_substream); + __soc_pcm_hw_free(be, be_substream); } return ret; @@ -1960,7 +2039,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int ret, stream = substream->stream; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); memcpy(&fe->dpcm[stream].hw_params, params, @@ -1974,7 +2053,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, params_channels(params), params_format(params)); /* call hw_params on the frontend */ - ret = soc_pcm_hw_params(substream, params); + ret = __soc_pcm_hw_params(fe, substream, params); if (ret < 0) dpcm_be_dai_hw_free(fe, stream); else @@ -1982,7 +2061,7 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, out: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); if (ret < 0) dev_err(fe->dev, "ASoC: %s failed (%d)\n", __func__, ret); @@ -2253,7 +2332,7 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) dev_dbg(be->dev, "ASoC: prepare BE %s\n", be->dai_link->name); - ret = soc_pcm_prepare(be_substream); + ret = __soc_pcm_prepare(be, be_substream); if (ret < 0) break; @@ -2271,7 +2350,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(substream); int stream = substream->stream, ret = 0; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); dev_dbg(fe->dev, "ASoC: prepare FE %s\n", fe->dai_link->name); @@ -2290,7 +2369,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; /* call prepare on the frontend */ - ret = soc_pcm_prepare(substream); + ret = __soc_pcm_prepare(fe, substream); if (ret < 0) goto out; @@ -2298,7 +2377,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) out: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); if (ret < 0) dev_err(fe->dev, "ASoC: %s() failed (%d)\n", __func__, ret); @@ -2349,7 +2428,6 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; - unsigned long flags; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); @@ -2418,7 +2496,6 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any pending BEs */ - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2430,7 +2507,6 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) be->dpcm[stream].state == SND_SOC_DPCM_STATE_NEW) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); if (ret < 0) dev_err(fe->dev, "ASoC: %s() failed (%d)\n", __func__, ret); @@ -2505,7 +2581,7 @@ int snd_soc_dpcm_runtime_update(struct snd_soc_card *card) struct snd_soc_pcm_runtime *fe; int ret = 0; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + mutex_lock_nested(&card->pcm_mutex, card->pcm_subclass); /* shutdown all old paths first */ for_each_card_rtds(card, fe) { ret = soc_dpcm_fe_runtime_update(fe, 0); @@ -2521,7 +2597,7 @@ int snd_soc_dpcm_runtime_update(struct snd_soc_card *card) } out: - mutex_unlock(&card->mutex); + mutex_unlock(&card->pcm_mutex); return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_runtime_update); @@ -2532,6 +2608,8 @@ static void dpcm_fe_dai_cleanup(struct snd_pcm_substream *fe_substream) struct snd_soc_dpcm *dpcm; int stream = fe_substream->stream; + snd_soc_dpcm_mutex_assert_held(fe); + /* mark FE's links ready to prune */ for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; @@ -2546,12 +2624,12 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) struct snd_soc_pcm_runtime *fe = asoc_substream_to_rtd(fe_substream); int ret; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); ret = dpcm_fe_dai_shutdown(fe_substream); dpcm_fe_dai_cleanup(fe_substream); - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); return ret; } @@ -2562,7 +2640,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) int ret; int stream = fe_substream->stream; - mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + snd_soc_dpcm_mutex_lock(fe); fe->dpcm[stream].runtime = fe_substream->runtime; ret = dpcm_path_get(fe, stream, &list); @@ -2579,7 +2657,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); open_end: - mutex_unlock(&fe->card->mutex); + snd_soc_dpcm_mutex_unlock(fe); return ret; } @@ -2840,10 +2918,8 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_dpcm *dpcm; int state; int ret = 1; - unsigned long flags; int i; - spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2857,7 +2933,6 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, } } } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to do this BE DAI */ return ret; -- 2.40.1.495.gc816e09b53d-goog

4 7

[PATCH v3] ASoC: dt-bindings: nau8825: Convert to dtschema
by David Lin 11 May '23

11 May '23

Convert the NAU8825 audio CODEC bindings to DT schema. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> Changes: V2 -> V3: - refine node name from "nau8825" to "codec" for generic purpose V1 -> V2: - add interrupts properties - add maximum to nuvoton,jack-insert-debounce and nuvoton,jack-eject-debounce properites - add a enum item for nuvoton,short-key-debounce properites - add default value for most properites - add maxItems to clocks properites and mclk entries to clock-names properites - refine wrong indentation from clocks and clock-names - refine dts example for interrupts and clocks - remove headset label from dts example --- .../devicetree/bindings/sound/nau8825.txt | 111 -------- .../bindings/sound/nuvoton,nau8825.yaml | 242 ++++++++++++++++++ 2 files changed, 242 insertions(+), 111 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt deleted file mode 100644 index a9c34526f4cb..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8825.txt +++ /dev/null @@ -1,111 +0,0 @@ -Nuvoton NAU8825 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8825" - - - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-threshold-num: Number of buttons supported - - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as - SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) - where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. - Refer datasheet section 10.2 for more information about threshold calculation. - - - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. - - - nuvoton,sar-voltage: Reference voltage for button impedance measurement. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,sar-compare-time: SAR compare time - 0 - 500 ns - 1 - 1 us - 2 - 2 us - 3 - 4 us - - - nuvoton,sar-sampling-time: SAR sampling time - 0 - 2 us - 1 - 4 us - 2 - 8 us - 3 - 16 us - - - nuvoton,short-key-debounce: Button short key press debounce time. - 0 - 30 ms - 1 - 50 ms - 2 - 100 ms - 3 - 30 ms - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,crosstalk-enable: make crosstalk function enable if set. - - - nuvoton,adcout-drive-strong: make the drive strength of ADCOUT IO PIN strong if set. - Otherwise, the drive keeps normal strength. - - - nuvoton,adc-delay-ms: Delay (in ms) to make input path stable and avoid pop noise. The - default value is 125 and range between 125 to 500 ms. - - - 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 - -Example: - - headset: nau8825@1a { - compatible = "nuvoton,nau8825"; - reg = <0x1a>; - interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - // Setup 4 buttons impedance according to Android specification - nuvoton,sar-threshold-num = <4>; - nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; - nuvoton,sar-hysteresis = <1>; - nuvoton,sar-voltage = <0>; - nuvoton,sar-compare-time = <0>; - nuvoton,sar-sampling-time = <0>; - nuvoton,short-key-debounce = <2>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,crosstalk-enable; - - clock-names = "mclk"; - clocks = <&tegra_pmc TEGRA_PMC_CLK_OUT_2>; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml new file mode 100644 index 000000000000..a68c553d7464 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8825.yaml @@ -0,0 +1,242 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8825.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU8825 audio CODEC + +maintainers: + - John Hsu <KCHSU0(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - nuvoton,nau8825 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + description: + The CODEC's interrupt output. + + nuvoton,jkdet-enable: + description: + Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: + Enable JKDET pin pull. + If set - pin pull enabled, otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: + Pull-up JKDET pin. + If set then JKDET pin is pull up, otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + JKDET pin polarity. + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,vref-impedance: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + VREF Impedance selection. + enum: + - 0 # Open + - 1 # 25 kOhm + - 2 # 125 kOhm + - 3 # 2.5 kOhm + default: 2 + + nuvoton,micbias-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Micbias voltage level. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-threshold-num: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Number of buttons supported. + minimum: 1 + maximum: 4 + default: 4 + + nuvoton,sar-threshold: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: + Impedance threshold for each button. Array that contains up to 8 buttons + configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) where MICBIAS is + configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by + 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold + calculation. + minItems: 1 + maxItems: 4 + items: + minimum: 0 + maximum: 255 + + nuvoton,sar-hysteresis: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button impedance measurement hysteresis. + default: 0 + + nuvoton,sar-voltage: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Reference voltage for button impedance measurement. + enum: + - 0 # VDDA + - 1 # VDDA + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,sar-compare-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR compare time. + enum: + - 0 # 500 ns + - 1 # 1 us + - 2 # 2 us + - 3 # 4 us + default: 1 + + nuvoton,sar-sampling-time: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + SAR sampling time. + enum: + - 0 # 2 us + - 1 # 4 us + - 2 # 8 us + - 3 # 16 us + default: 1 + + nuvoton,short-key-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Button short key press debounce time. + enum: + - 0 # 30 ms + - 1 # 50 ms + - 2 # 100 ms + - 3 # 30 ms + default: 3 + + nuvoton,jack-insert-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms. + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + number from 0 to 7 that sets debounce time to 2^(n+2) ms + maximum: 7 + default: 0 + + nuvoton,crosstalk-enable: + description: + make crosstalk function enable if set. + type: boolean + + nuvoton,adcout-drive-strong: + description: + make the drive strength of ADCOUT IO PIN strong if set. + Otherwise, the drive keeps normal strength. + type: boolean + + nuvoton,adc-delay-ms: + description: + Delay (in ms) to make input path stable and avoid pop noise. + The default value is 125 and range between 125 to 500 ms. + minimum: 125 + maximum: 500 + default: 125 + + clocks: + description: + list of phandle and clock specifier pairs according to common clock + bindings for the clocks described in clock-names. + maxItems: 1 + + clock-names: + description: + should include "mclk" for the MCLK master clock. + items: + - const: mclk + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <38 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,sar-compare-time = <0>; + nuvoton,sar-sampling-time = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,crosstalk-enable; + + clock-names = "mclk"; + clocks = <&tegra_pmc 1>; + }; + }; -- 2.25.1

3 2

[PATCH 0/4] Add support for IIO devices in ASoC
by Herve Codina 11 May '23

11 May '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already under review [2]. This series introduces simple-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Herve Codina (4): dt-bindings: sound: Add simple-iio-aux iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Add a helper to build a DAPM widget dynamically ASoC: codecs: Add support for the generic IIO auxiliary devices .../bindings/sound/simple-iio-aux.yaml | 65 ++++ drivers/iio/inkern.c | 67 ++++ include/linux/iio/consumer.h | 11 + include/sound/soc-dapm.h | 12 +- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/simple-iio-aux.c | 307 ++++++++++++++++++ 7 files changed, 475 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-iio-aux.yaml create mode 100644 sound/soc/codecs/simple-iio-aux.c -- 2.39.2

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Can't play 44100hz, 88200hz files Natively.
by GitHub issues - opened 11 May '23

11 May '23

alsa-project/alsa-ucm-conf issue #313 was opened from RenEH01: Can't play audio files at native sample rate. Everything using the Hifi 2.0 output configuration forces a sample rate of 48000hz. When I switch to the Pro Audio driver under pavucontrol, 44100 hz can play natively. Is there a forced resampling occuring in the UCM2 driver? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/313 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/4] ASoC: qcom: add display port module support.
by Srinivas Kandagatla 11 May '23

11 May '23

This patchset adds support to displayport on AudioReach. Patches are tested on X13s with two display ports. Srinivas Kandagatla (4): ASoC: qcom: q6dsp-common: move channel allocation to common ASoC: qcom: audioreach: add support for DISPLAY PORT SINK module ASoC: qcom: q6dsp: add support to more display ports ASoC: qcom: q6apm: add support to display ports in lpass dais .../sound/qcom,q6dsp-lpass-ports.h | 8 ++ sound/soc/qcom/qdsp6/audioreach.c | 75 +++++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 12 +++ sound/soc/qcom/qdsp6/q6afe-dai.c | 34 ++------- sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 39 ++++++++++ sound/soc/qcom/qdsp6/q6dsp-common.c | 35 +++++++++ sound/soc/qcom/qdsp6/q6dsp-common.h | 1 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 43 +++++++---- 8 files changed, 204 insertions(+), 43 deletions(-) -- 2.21.0

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[alsa-devel@alsa-project.org: [PATCH 3/5] ASoC: mediatek: mt8195-afe-pcm: Simplify runtime PM during probe]
by Mark Brown 11 May '23

11 May '23

Hi, Here's another mail (one of several in this series) that got completely mangled by the alsa-project.org mailman to the point of unusability. I didn't see any response to my last mail about this, is there any news on fixing mailman to not do this? It's extremely disruptive to working with lore. Thanks, Mark

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[PATCH v1] ASoC: jz4740-i2s: Make I2S divider calculations more robust
by Aidan MacDonald 11 May '23

11 May '23

When the CPU supplies bit/frame clocks, the system clock (clk_i2s) is divided to produce the bit clock. This is a simple 1/N divider with a fairly limited range, so for a given system clock frequency only a few sample rates can be produced. Usually a wider range of sample rates is supported by varying the system clock frequency. The old calculation method was not very robust and could easily produce the wrong clock rate, especially with non-standard rates. For example, if the system clock is 1.99x the target bit clock rate, the divider would be calculated as 1 instead of the more accurate 2. Instead, use a more accurate method that considers two adjacent divider settings and selects the one that produces the least error versus the requested rate. If the error is 5% or higher then the rate setting is rejected to prevent garbled audio. Skip divider calculation when the codec is supplying both the bit and frame clock; in that case, the divider outputs are unused and we don't want to constrain the sample rate. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 54 ++++++++++++++++++++++++++++++++--- 1 file changed, 50 insertions(+), 4 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6d9cfe0a5041..d0f6c945d9ae 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -218,18 +218,48 @@ static int jz4740_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int jz4740_i2s_get_i2sdiv(unsigned long mclk, unsigned long rate, + unsigned long i2sdiv_max) +{ + unsigned long div, rate1, rate2, err1, err2; + + div = mclk / (64 * rate); + if (div == 0) + div = 1; + + rate1 = mclk / (64 * div); + rate2 = mclk / (64 * (div + 1)); + + err1 = abs(rate1 - rate); + err2 = abs(rate2 - rate); + + /* + * Choose the divider that produces the smallest error in the + * output rate and reject dividers with a 5% or higher error. + * In the event that both dividers are outside the acceptable + * error margin, reject the rate to prevent distorted audio. + * (The number 5% is arbitrary.) + */ + if (div <= i2sdiv_max && err1 <= err2 && err1 < rate/20) + return div; + if (div < i2sdiv_max && err2 < rate/20) + return div + 1; + + return -EINVAL; +} + static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct jz4740_i2s *i2s = snd_soc_dai_get_drvdata(dai); struct regmap_field *div_field; + unsigned long i2sdiv_max; unsigned int sample_size; - uint32_t ctrl; - int div; + uint32_t ctrl, conf; + int div = 1; regmap_read(i2s->regmap, JZ_REG_AIC_CTRL, &ctrl); - - div = clk_get_rate(i2s->clk_i2s) / (64 * params_rate(params)); + regmap_read(i2s->regmap, JZ_REG_AIC_CONF, &conf); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: @@ -258,11 +288,27 @@ static int jz4740_i2s_hw_params(struct snd_pcm_substream *substream, ctrl &= ~JZ_AIC_CTRL_MONO_TO_STEREO; div_field = i2s->field_i2sdiv_playback; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_playback.msb, + i2s->soc_info->field_i2sdiv_playback.lsb); } else { ctrl &= ~JZ_AIC_CTRL_INPUT_SAMPLE_SIZE; ctrl |= FIELD_PREP(JZ_AIC_CTRL_INPUT_SAMPLE_SIZE, sample_size); div_field = i2s->field_i2sdiv_capture; + i2sdiv_max = GENMASK(i2s->soc_info->field_i2sdiv_capture.msb, + i2s->soc_info->field_i2sdiv_capture.lsb); + } + + /* + * Only calculate I2SDIV if we're supplying the bit or frame clock. + * If the codec is supplying both clocks then the divider output is + * unused, and we don't want it to limit the allowed sample rates. + */ + if (conf & (JZ_AIC_CONF_BIT_CLK_MASTER | JZ_AIC_CONF_SYNC_CLK_MASTER)) { + div = jz4740_i2s_get_i2sdiv(clk_get_rate(i2s->clk_i2s), + params_rate(params), i2sdiv_max); + if (div < 0) + return div; } regmap_write(i2s->regmap, JZ_REG_AIC_CTRL, ctrl); -- 2.39.2

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