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[PATCH] ALSA: usb-audio: US16x08: fix value count for level meters
by Hector Martin 27 Nov '20

27 Nov '20

The level meter control returns 34 integers of info. This fixes: snd-usb-audio 3-1:1.0: control 2:0:0:Level Meter:0: access overflow Cc: stable(a)vger.kernel.org Signed-off-by: Hector Martin <marcan(a)marcan.st> --- sound/usb/mixer_us16x08.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 92b1a6d9c931..bd63a9ce6a70 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -607,7 +607,7 @@ static int snd_us16x08_eq_put(struct snd_kcontrol *kcontrol, static int snd_us16x08_meter_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->count = 1; + uinfo->count = 34; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.max = 0x7FFF; uinfo->value.integer.min = 0; -- 2.28.0

2 1

pcm: ioplug: Limit transfer size to buffer boundary
by GitHub pull_request - opened 27 Nov '20

27 Nov '20

alsa-project/alsa-lib pull request #103 was opened from aditpape: Commit 1714332719fc91507ca24dd3567e50d7094b3001 introduced 2nd transfer() call to transfer all remaining available frames. If the prior calculated avail value exceeds the buffer size a too large size value is passed to the underlaying plugin and results in memory corruption if not blocked by plugin internally. Avail values > buffer size can happen if e.g. xrun detection is disabled, as avail is calculated by pure difference between hw and app position. This patch limits 2nd transfer call to remaining rest of a buffer size. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/103 Patch URL : https://github.com/alsa-project/alsa-lib/pull/103.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] Asoc: qcom: Fix plaback recover problem in suspend resume
by Srinivasa Rao Mandadapu 27 Nov '20

27 Nov '20

To support playback continuation after hard suspend(bypass powerd) and resume: Prepare device in platform trigger callback. Make I2s and DMA control registers as non volatile. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since v1: -- Subject lines changed sound/soc/qcom/lpass-cpu.c | 8 ++------ sound/soc/qcom/lpass-platform.c | 5 +++-- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af684fd..c99be03 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -454,20 +454,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->i2s_ports; ++i) - if (reg == LPAIF_I2SCTL_REG(v, i)) - return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || - reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) return true; return false; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 80b09de..2b0a7c1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -481,8 +481,9 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -ENOTRECOVERABLE; } switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + lpass_platform_pcmops_prepare(component, substream); + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_fields_write(dmactl->enable, id, LPAIF_DMACTL_ENABLE_ON); @@ -592,7 +593,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; } - return 0; + return ret; } static snd_pcm_uframes_t lpass_platform_pcmops_pointer( -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 0

new codec support of ALC897
by Kailang 27 Nov '20

27 Nov '20

Hi Takashi, Attach was the enable patch for ALC897. BR, Kailang

2 1

[PATCH] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 27 Nov '20

27 Nov '20

If stop by underrun and DPCM BE disconnection is run simultaneously, data abort can be occurred by the sequence below. /* In core X, running dpcm_be_dai_trigger() */ for_each_dpcm_be(fe, stream, dpcm) { /* In core Y, running dpcm_be_disconnect() */ 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); /* In core X, running dpcm_be_dai_trigger() */ struct snd_soc_pcm_runtime *be = dpcm->be; <== Accessing freed memory To prevent this situation, dpcm_lock should be acquired during iteration of dpcm list in dpcm_do_trigger(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..7c5d950a8628 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2073,6 +2073,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, struct snd_pcm_substream *substream, int cmd) { @@ -2092,8 +2095,10 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; + unsigned long flags; int ret = 0; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; @@ -2113,7 +2118,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2123,7 +2128,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2133,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; @@ -2142,12 +2147,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; @@ -2155,12 +2160,12 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_SUSPEND; break; @@ -2168,17 +2173,20 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) continue; - if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) + if (!dpcm_can_be_free_stop(fe, be, stream)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); if (ret) - return ret; + break; be->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; break; } + if (ret) + break; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -2916,10 +2924,9 @@ 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); + lockdep_assert_held(&fe->card->dpcm_lock); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -2933,17 +2940,12 @@ 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; } -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, +static int dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { @@ -2954,6 +2956,23 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + unsigned long flags; + int ret; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = dpcm_can_be_free_stop(fe, be, stream); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2963,6 +2982,9 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { + unsigned long flags; + int ret; + const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, SND_SOC_DPCM_STATE_PAUSED, @@ -2970,6 +2992,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, SND_SOC_DPCM_STATE_PREPARE, }; - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + ret = snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0

[PATCH] configure: Replace PKG_CONFIG calls with PKG_CHECK_VAR
by Ismael Luceno 26 Nov '20

26 Nov '20

Signed-off-by: Ismael Luceno <ismael(a)iodev.co.uk> --- configure.ac | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/configure.ac b/configure.ac index 29d043f18950..8e130b3f0775 100644 --- a/configure.ac +++ b/configure.ac @@ -204,12 +204,8 @@ AC_ARG_WITH( [udev-rules-dir], AS_HELP_STRING([--with-udev-rules-dir=DIR],[Directory where to install udev rules to (default=auto)]), [udevrulesdir="$withval"], - [udevdir=$($PKG_CONFIG udev --variable=udevdir) - if test "x$udevdir" = "x"; then - udevrulesdir="/lib/udev/rules.d" - else - udevrulesdir="$udevdir/rules.d" - fi]) + [PKG_CHECK_VAR([udevdir], [udev], [udevdir]) + udevrulesdir="${udevdir:-/lib/udev}/rules.d"]) AC_SUBST(udevrulesdir) dnl Checks for header files. @@ -404,10 +400,10 @@ PKG_CHECK_MODULES(SYSTEMD, [systemd >= 18], [have_min_systemd="no"]) AC_ARG_WITH([systemdsystemunitdir], AS_HELP_STRING([--with-systemdsystemunitdir=DIR], [Directory for systemd service files]), - [], [with_systemdsystemunitdir=$($PKG_CONFIG --variable=systemdsystemunitdir systemd)]) -if test "x$with_systemdsystemunitdir" != xno; then - AC_SUBST([systemdsystemunitdir], [$with_systemdsystemunitdir]) -fi + [], [PKG_CHECK_VAR([with_systemdsystemunitdir], [systemd], [systemdsystemunitdir])]) +AS_IF([test "x$with_systemdsystemunitdir" != xno], + [AC_SUBST([systemdsystemunitdir], ["$with_systemdsystemunitdir"])]) + AM_CONDITIONAL(HAVE_SYSTEMD, [test "$have_min_systemd" = "yes" \ -a -n "$with_systemdsystemunitdir" -a "x$with_systemdsystemunitdir" != xno ]) -- 2.28.0

1 0

[PATCH] configure: Improve failures due to missing pkg-config
by Ismael Luceno 26 Nov '20

26 Nov '20

Output a warning at the beginning, and fill PKG_CONFIG with "false" so that any auto-configuration attempts fail without bogus messages. Signed-off-by: Ismael Luceno <ismael(a)iodev.co.uk> --- configure.ac | 3 +++ 1 file changed, 3 insertions(+) diff --git a/configure.ac b/configure.ac index 1f119c5baeeb..29d043f18950 100644 --- a/configure.ac +++ b/configure.ac @@ -19,6 +19,9 @@ AC_PROG_MKDIR_P AC_PROG_LN_S AC_PROG_SED PKG_PROG_PKG_CONFIG +AS_IF([test -z "$PKG_CONFIG"], + [AC_MSG_WARN([pkg-config is needed for auto-configuration of dependencies]) + PKG_CONFIG=false]) AM_PATH_ALSA(1.0.27) if test "x$enable_alsatest" = "xyes"; then AC_CHECK_FUNC([snd_ctl_elem_add_enumerated], -- 2.28.0

1 0

[PATCH] regmap: sdw: add required header files
by Bard Liao 26 Nov '20

26 Nov '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Explicitly add header files used by regmap SoundWire support. Suggested-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/base/regmap/regmap-sdw.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/base/regmap/regmap-sdw.c b/drivers/base/regmap/regmap-sdw.c index c92d614b4943..c83be26434e7 100644 --- a/drivers/base/regmap/regmap-sdw.c +++ b/drivers/base/regmap/regmap-sdw.c @@ -2,7 +2,9 @@ // Copyright(c) 2015-17 Intel Corporation. #include <linux/device.h> +#include <linux/errno.h> #include <linux/module.h> +#include <linux/regmap.h> #include <linux/soundwire/sdw.h> #include "internal.h" -- 2.17.1

4 3

[PATCH] ASoC: rt5682: change SAR voltage threshold
by shumingf＠realtek.com 26 Nov '20

26 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> To fix errors in some 4 poles headset detection cases, this patch adjusts the voltage threshold for mic detection. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index f299b30f4f59..4d865edadd7e 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -43,6 +43,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, + {RT5682_SAR_IL_CMD_3, 0x8365}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.29.0

2 1

[PATCH] ASoC: fsl: Fix config name of CONFIG_ARCH_MXC
by Shengjiu Wang 26 Nov '20

26 Nov '20

CONFIG_ARCH_MXC should be ARCH_MXC Fixes: 674226db62ec ("ASoC: fsl: SND_SOC_FSL_AUD2HTX should depend on ARCH_MXC") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7d48f4f98e8b..835a14821360 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -107,7 +107,7 @@ config SND_SOC_FSL_XCVR config SND_SOC_FSL_AUD2HTX tristate "AUDIO TO HDMI TX module support" - depends on CONFIG_ARCH_MXC || COMPILE_TEST + depends on ARCH_MXC || COMPILE_TEST help Say Y if you want to add AUDIO TO HDMI TX support for NXP. -- 2.27.0

2 1

[PATCH] ASoC: hdmi-codec: Add RX support
by Shengjiu Wang 26 Nov '20

26 Nov '20

HDMI interface can also be used as receiver, this patch is to add such support. The most difference compare with TX is that RX don't need to get edid information. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/hdmi-codec.c | 33 ++++++++++++++++++++++++++++----- 1 file changed, 28 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e8410b2433de..d5fcc4db8284 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -282,6 +282,7 @@ struct hdmi_codec_priv { static const struct snd_soc_dapm_widget hdmi_widgets[] = { SND_SOC_DAPM_OUTPUT("TX"), + SND_SOC_DAPM_OUTPUT("RX"), }; enum { @@ -389,6 +390,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; mutex_lock(&hcp->lock); @@ -404,7 +406,7 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, goto err; } - if (hcp->hcd.ops->get_eld) { + if (tx && hcp->hcd.ops->get_eld) { ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->hcd.data, hcp->eld, sizeof(hcp->eld)); if (ret) @@ -660,14 +662,20 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai) { struct snd_soc_dapm_context *dapm; struct hdmi_codec_daifmt *daifmt; - struct snd_soc_dapm_route route = { - .sink = "TX", - .source = dai->driver->playback.stream_name, + struct snd_soc_dapm_route route[] = { + { + .sink = "TX", + .source = dai->driver->playback.stream_name, + }, + { + .sink = dai->driver->capture.stream_name, + .source = "RX", + }, }; int ret; dapm = snd_soc_component_get_dapm(dai->component); - ret = snd_soc_dapm_add_routes(dapm, &route, 1); + ret = snd_soc_dapm_add_routes(dapm, route, 2); if (ret) return ret; @@ -757,6 +765,14 @@ static const struct snd_soc_dai_driver hdmi_i2s_dai = { .formats = I2S_FORMATS, .sig_bits = 24, }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 8, + .rates = HDMI_RATES, + .formats = I2S_FORMATS, + .sig_bits = 24, + }, .ops = &hdmi_codec_i2s_dai_ops, .pcm_new = hdmi_codec_pcm_new, }; @@ -773,6 +789,13 @@ static const struct snd_soc_dai_driver hdmi_spdif_dai = { .rates = HDMI_RATES, .formats = SPDIF_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = HDMI_RATES, + .formats = SPDIF_FORMATS, + }, .ops = &hdmi_codec_spdif_dai_ops, .pcm_new = hdmi_codec_pcm_new, }; -- 2.27.0

2 1

[PATCH 0/5] ASoC: merge soc_compr_open() rollback and soc_compr_free()
by Kuninori Morimoto 26 Nov '20

26 Nov '20

Hi Mark soc_compr_open() does rollback when failed (A), but, it is almost same as soc_compr_free(). static int soc_compr_open(xxx) { ... if (ret < 0) goto xxx_err; ... return 0; ^ machine_err: | ... | out: (A) ... | pm_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_compr_free() and rollback. Kuninori Morimoto (5): ASoC: soc-compress: move soc_compr_free() next to soc_compr_open() ASoC: soc-dai: add mark for snd_soc_dai_compr_startup/shutdown() ASoC: soc-component: add mark for snd_soc_component_compr_open/free() ASoC: soc-component: add mark for snd_soc_link_compr_startup/shutdown() ASoC: soc-compress: add soc_compr_clean() and call it from soc_compr_open/free() include/sound/soc-component.h | 6 +- include/sound/soc-dai.h | 4 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 1 + sound/soc/soc-component.c | 17 +++-- sound/soc/soc-compress.c | 115 +++++++++++++++++----------------- sound/soc/soc-dai.c | 13 +++- sound/soc/soc-link.c | 11 +++- 8 files changed, 95 insertions(+), 75 deletions(-) -- 2.25.1

2 6

[PATCH v2 00/39] ASoC: fix !OF compile test warnings
by Krzysztof Kozlowski 26 Nov '20

26 Nov '20

Hi, Changes since v1: 1. New patch 2/39: ASoC: bd28623: mark OF related data as maybe unused, 2. Patches 1-13: add maybe_unused after discussions with Mark, 3. Add Sylwester's review to two patches (one changed so please review one more time). Best regards, Krzysztof Krzysztof Kozlowski (39): ASoC: ak5558: mark OF related data as maybe unused ASoC: bd28623: mark OF related data as maybe unused ASoC: gtm601: mark OF related data as maybe unused ASoC: inno_rk3036: mark OF related data as maybe unused ASoC: rk3328: mark OF related data as maybe unused ASoC: tas571x: mark OF related data as maybe unused ASoC: kirkwood: armada-370-db: mark OF related data as maybe unused ASoC: meson: t9015: mark OF related data as maybe unused ASoC: qcom: mark OF related data as maybe unused ASoC: samsung: smdk_wm8994: mark OF related data as maybe unused ASoC: rockchip: mark OF related data as maybe unused ASoC: ti: davinci: mark OF related data as maybe unused ASoC: uniphier: mark OF related data as maybe unused ASoC: ak4118: skip of_device_id table when !CONFIG_OF ASoC: alc5623: skip of_device_id table when !CONFIG_OF ASoC: alc5632: skip of_device_id table when !CONFIG_OF ASoC: da7218: skip of_device_id table when !CONFIG_OF ASoC: da7219: skip of_device_id table when !CONFIG_OF ASoC: da9055: skip of_device_id table when !CONFIG_OF ASoC: es8316: skip of_device_id table when !CONFIG_OF ASoC: max98090: skip of_device_id table when !CONFIG_OF ASoC: max98095: skip of_device_id table when !CONFIG_OF ASoC: max98371: skip of_device_id table when !CONFIG_OF ASoC: max9867: skip of_device_id table when !CONFIG_OF ASoC: max98925: skip of_device_id table when !CONFIG_OF ASoC: max98926: skip of_device_id table when !CONFIG_OF ASoC: pcm1789: skip of_device_id table when !CONFIG_OF ASoC: pcm179x: skip of_device_id table when !CONFIG_OF ASoC: rt5660: skip of_device_id table when !CONFIG_OF ASoC: tas2562: skip of_device_id table when !CONFIG_OF ASoC: tlv320: skip of_device_id table when !CONFIG_OF ASoC: ts3a227e: skip of_device_id table when !CONFIG_OF ASoC: es7134: mark OF related data as maybe unused ASoC: es7241: mark OF related data as maybe unused ASoC: samsung: i2s: mark OF related data as maybe unused ASoC: max98371: drop driver pm=NULL assignment ASoC: max98925: drop driver pm=NULL assignment ASoC: max98926: drop driver pm=NULL assignment ASoC: samsung: smdk_wm8994: remove redundant of_match_ptr() sound/soc/codecs/ak4118.c | 2 ++ sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/alc5623.c | 2 ++ sound/soc/codecs/alc5632.c | 2 ++ sound/soc/codecs/bd28623.c | 2 +- sound/soc/codecs/da7218.c | 2 ++ sound/soc/codecs/da7219.c | 2 ++ sound/soc/codecs/da9055.c | 2 ++ sound/soc/codecs/es7134.c | 4 ++-- sound/soc/codecs/es7241.c | 2 +- sound/soc/codecs/es8316.c | 2 ++ sound/soc/codecs/gtm601.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/max98090.c | 2 ++ sound/soc/codecs/max98095.c | 2 ++ sound/soc/codecs/max98371.c | 3 ++- sound/soc/codecs/max9867.c | 2 ++ sound/soc/codecs/max98925.c | 3 ++- sound/soc/codecs/max98926.c | 3 ++- sound/soc/codecs/pcm1789-i2c.c | 2 ++ sound/soc/codecs/pcm179x-i2c.c | 2 ++ sound/soc/codecs/rk3328_codec.c | 2 +- sound/soc/codecs/rt5660.c | 2 ++ sound/soc/codecs/tas2562.c | 2 ++ sound/soc/codecs/tas571x.c | 4 ++-- sound/soc/codecs/tlv320adcx140.c | 2 ++ sound/soc/codecs/tlv320aic23-i2c.c | 2 ++ sound/soc/codecs/ts3a227e.c | 2 ++ sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/meson/t9015.c | 2 +- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s.c | 8 ++++---- sound/soc/samsung/smdk_wm8994.c | 4 ++-- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 43 files changed, 69 insertions(+), 32 deletions(-) -- 2.25.1

2 40

Re: [PATCH 000/141] Fix fall-through warnings for Clang
by Miguel Ojeda 26 Nov '20

26 Nov '20

On Wed, Nov 25, 2020 at 11:44 PM Edward Cree <ecree.xilinx(a)gmail.com> wrote: > > To make the intent clear, you have to first be certain that you > understand the intent; otherwise by adding either a break or a > fallthrough to suppress the warning you are just destroying the > information that "the intent of this code is unknown". If you don't know what the intent of your own code is, then you *already* have a problem in your hands. > Figuring out the intent of a piece of unfamiliar code takes more > than 1 minute; just because > case foo: > thing; > case bar: > break; > produces identical code to > case foo: > thing; > break; > case bar: > break; > doesn't mean that *either* is correct — maybe the author meant What takes 1 minute is adding it *mechanically* by the author, i.e. so that you later compare whether codegen is the same. > to write > case foo: > return thing; > case bar: > break; > and by inserting that break you've destroyed the marker that > would direct someone who knew what the code was about to look > at that point in the code and spot the problem. Then it means you already have a bug. This patchset gives the maintainer a chance to notice it, which is a good thing. The "you've destroyed the market" claim is bogus, because: 1. you were not looking into it 2. you didn't notice the bug so far 3. is implicit -- harder to spot 4. is only useful if you explicitly take a look at this kind of bug. So why don't you do it now? > Thus, you *always* have to look at more than just the immediate > mechanical context of the code, to make a proper judgement that > yes, this was the intent. I find that is the responsibility of the maintainers and reviewers for tree-wide patches like this, assuming they want. They can also keep the behavior (and the bugs) without spending time. Their choice. > If you think that that sort of thing > can be done in an *average* time of one minute, then I hope you > stay away from code I'm responsible for! Please don't accuse others of recklessness or incompetence, especially if you didn't understand what they said. > A warning is only useful because it makes you *think* about the > code. If you suppress the warning without doing that thinking, > then you made the warning useless; and if the warning made you > think about code that didn't *need* it, then the warning was > useless from the start. We are not suppressing the warning. Quite the opposite, in fact. > So make your mind up: does Clang's stricter -Wimplicit-fallthrough > flag up code that needs thought (in which case the fixes take > effort both to author and to review) As I said several times already, it does take time to review if the maintainer wants to take the chance to see if they had a bug to begin with, but it does not require thought for the author if they just go for equivalent codegen. > or does it flag up code > that can be mindlessly "fixed" (in which case the warning is > worthless)? Proponents in this thread seem to be trying to > have it both ways. A warning is not worthless just because you can mindlessly fix it. There are many counterexamples, e.g. many checkpatch/lint/lang-format/indentation warnings, functional ones like the `if (a = b)` warning... Cheers, Miguel

3 3

pcm: snd_pcm_readi - plugin type multi, the overun appears when read
by GitHub pull_request - opened 26 Nov '20

26 Nov '20

alsa-project/alsa-lib pull request #102 was opened from ouwenchen: After type route use type multi, merge the two input streams.There was PCM error due to insufficient readable data.Cause the underlying read pointer cannot be updated, and then it will be overrun. Request URL : https://github.com/alsa-project/alsa-lib/pull/102 Patch URL : https://github.com/alsa-project/alsa-lib/pull/102.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 0/1] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 26 Nov '20

26 Nov '20

On Wed, 25 Nov 2020 16:28:05 +0530, Vinod Koul wrote: >On 27-10-20, 10:57, Gyeongtaek Lee wrote: >> diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c >> index 0e53f6f31916..a071485383ed 100644 >> --- a/sound/core/compress_offload.c >> +++ b/sound/core/compress_offload.c >> @@ -708,11 +708,24 @@ static int snd_compr_pause(struct snd_compr_stream *stream) >> { >> int retval; >> >> - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) >> + switch (stream->runtime->state) { >> + case SNDRV_PCM_STATE_RUNNING: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_PUSH); > >this seems to fit single line with new 100char limit and makes it a >better read, can we please do that here and few places below .. The line >split is making it look bit ugly IMO > >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; >> + break; >> + case SNDRV_PCM_STATE_DRAINING: >> + if (!stream->device->use_pause_in_draining) >> + return -EPERM; > >I am expecting patches to tinycompress to handle pause while drain. Esp >this case.. > >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> + if (!retval) >> + stream->pause_in_draining = true; >> + break; >> + default: >> return -EPERM; >> - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); >> - if (!retval) >> - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; >> + } >> return retval; >> } >> >> @@ -720,11 +733,25 @@ static int snd_compr_resume(struct snd_compr_stream *stream) >> { >> int retval; >> >> - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) >> + switch (stream->runtime->state) { >> + case SNDRV_PCM_STATE_PAUSED: >> + retval = stream->ops->trigger(stream, >> + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); >> + if (!retval) >> + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; >> + break; >> + case SNDRV_PCM_STATE_DRAINING: >> + if (!stream->device->use_pause_in_draining || >> + !stream->pause_in_draining) >> + return -EPERM; > >does this condition make sense for resume part..? We have already >checked for this while going into pause. I am not expecting drain state >to change while we are paused :) > I have fixed the patch as your review comment and send version 3. For your comment about tinycompress, I'll try to make a patch after study about tinycompress. Thanks, Gyeongtaek Gyeongtaek Lee (1): ALSA: compress: allow pause and resume during draining include/sound/compress_driver.h | 16 ++++++++++++++ sound/core/compress_offload.c | 39 ++++++++++++++++++++++++++------- 2 files changed, 47 insertions(+), 8 deletions(-) base-commit: 418baf2c28f3473039f2f7377760bd8f6897ae18 -- 2.21.0

1 0

Re: drivers/soundwire/qcom.c:767: undefined reference to `slimbus_bus'
by Randy Dunlap 26 Nov '20

26 Nov '20

On 11/2/20 11:47 AM, kernel test robot wrote: > tree: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: 3cea11cd5e3b00d91caf0b4730194039b45c5891 > commit: 09309093d5e8f8774e4a3a0d42b73cf47e9421cf soundwire: qcom: fix SLIBMUS/SLIMBUS typo > date: 8 weeks ago > config: openrisc-randconfig-r005-20201102 (attached as .config) > compiler: or1k-linux-gcc (GCC) 9.3.0 > reproduce (this is a W=1 build): > wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… > git remote add linus https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git > git fetch --no-tags linus master > git checkout 09309093d5e8f8774e4a3a0d42b73cf47e9421cf > # save the attached .config to linux build tree > COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=openrisc > > If you fix the issue, kindly add following tag as appropriate > Reported-by: kernel test robot <lkp(a)intel.com> > > All errors (new ones prefixed by >>): > > or1k-linux-ld: drivers/soundwire/qcom.o: in function `qcom_swrm_probe': >>> drivers/soundwire/qcom.c:767: undefined reference to `slimbus_bus' >>> or1k-linux-ld: drivers/soundwire/qcom.c:771: undefined reference to `slimbus_bus' > > vim +767 drivers/soundwire/qcom.c > > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 756 > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 757 static int qcom_swrm_probe(struct platform_device *pdev) > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 758 { > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 759 struct device *dev = &pdev->dev; > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 760 struct sdw_master_prop *prop; > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 761 struct sdw_bus_params *params; > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 762 struct qcom_swrm_ctrl *ctrl; > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 763 int ret; > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 764 u32 val; > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 765 > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 766 ctrl = devm_kzalloc(dev, sizeof(*ctrl), GFP_KERNEL); > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 @767 if (!ctrl) > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 768 return -ENOMEM; > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 769 > 09309093d5e8f87 Jonathan Marek 2020-09-08 770 #if IS_ENABLED(CONFIG_SLIMBUS) > 02efb49aa805cee Srinivas Kandagatla 2020-01-13 @771 if (dev->parent->bus == &slimbus_bus) { > 5bd773242f75da3 Jonathan Marek 2020-09-05 772 #else > 5bd773242f75da3 Jonathan Marek 2020-09-05 773 if (false) { > 5bd773242f75da3 Jonathan Marek 2020-09-05 774 #endif > :::::: The code at line 767 was first introduced by commit > :::::: 02efb49aa805cee643a643ab61a1118c2fd08b80 soundwire: qcom: add support for SoundWire controller > > :::::: TO: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> > :::::: CC: Vinod Koul <vkoul(a)kernel.org> config SOUNDWIRE_QCOM tristate "Qualcomm SoundWire Master driver" imply SLIMBUS depends on SND_SOC The kernel config that was attached has: CONFIG_SOUNDWIRE_QCOM=y CONFIG_SLIMBUS=m I expected that "imply" would make SLIMBUS=y since SOUNDWIRE_QCOM=y, but I guess that's not the case. :( Any ideas about what to do here? -- ~Randy

3 4

[PATCH] ASoC: fsl_xcvr: fix potential resource leak
by Viorel Suman (OSS) 25 Nov '20

25 Nov '20

From: Viorel Suman <viorel.suman(a)nxp.com> "fw" variable must be relased before return. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 2a28810d0e29..3d58c88ea603 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -706,6 +706,7 @@ static int fsl_xcvr_load_firmware(struct fsl_xcvr *xcvr) /* RAM is 20KiB = 16KiB code + 4KiB data => max 10 pages 2KiB each */ if (rem > 16384) { dev_err(dev, "FW size %d is bigger than 16KiB.\n", rem); + release_firmware(fw); return -ENOMEM; } -- 2.26.2

2 1

[PATCH] ASoC: wm_adsp: fix error return code in wm_adsp_load()
by Luo Meng 25 Nov '20

25 Nov '20

Fix to return a negative error code from the error handling case instead of 0 in function wm_adsp_load(), as done elsewhere in this function. Fixes: 170b1e123f38 ("ASoC: wm_adsp: Add support for new Halo core DSPs") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Luo Meng <luomeng12(a)huawei.com> --- sound/soc/codecs/wm_adsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bcf18bf15a02..e61d00486c65 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1937,6 +1937,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) mem = wm_adsp_find_region(dsp, type); if (!mem) { adsp_err(dsp, "No region of type: %x\n", type); + ret = -EINVAL; goto out_fw; } -- 2.25.4

3 2

[PATCH 0/4] ASoC: SOF: Add 'memory_info' file to debugfs
by Kai Vehmanen 25 Nov '20

25 Nov '20

Series that adds a new debugfs entry to query status of SOF FW memory allocation at runtime. Information on used and free memory is shown separately for each zone. A new IPC message is added for this purpose and ABI minor revision is bumped accordingly to 3.18. To implement this, additional FW configuration data needs to be parsed from the firmware file and this is done in the first patch. A few cleanups to surrounding code is done in separate patches. Karol Trzcinski (4): ASoC: SOF: ext_manifest: Parse firmware config dictionary ASoC: SOF: Improve code alignment in header.h ASoC: SOF: Change section comment for SOF_IPC_TEST_ ASoC: SOF: Add `memory_info` file to debugfs include/sound/sof/debug.h | 41 +++++++++++ include/sound/sof/ext_manifest.h | 20 ++++++ include/sound/sof/header.h | 10 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/debug.c | 117 +++++++++++++++++++++++++++++++ sound/soc/sof/ipc.c | 9 +++ sound/soc/sof/loader.c | 51 ++++++++++++++ sound/soc/sof/sof-priv.h | 2 + 8 files changed, 248 insertions(+), 4 deletions(-) create mode 100644 include/sound/sof/debug.h -- 2.28.0

2 5

[PATCH v2] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 25 Nov '20

25 Nov '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 17 +++++++++++++ sound/core/compress_offload.c | 44 +++++++++++++++++++++++++++------ 2 files changed, 53 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..5a0d6737de5e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,19 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining( + struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 0e53f6f31916..a071485383ed 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -708,11 +708,24 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -720,11 +733,25 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining || + !stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, + SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -767,6 +794,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

3 5

[PATCH] slimbus: qcom-ngd-ctrl: Avoid sending power requests without QMI
by Bjorn Andersson 25 Nov '20

25 Nov '20

Attempting to send a power request during PM operations, when the QMI handle isn't initialized results in a NULL pointer dereference. So check if the QMI handle has been initialized before attempting to post the power requests. Fixes: 917809e2280b ("slimbus: ngd: Add qcom SLIMBus NGD driver") Signed-off-by: Bjorn Andersson <bjorn.andersson(a)linaro.org> --- drivers/slimbus/qcom-ngd-ctrl.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index d8decb345e9d..c502e9e93965 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1229,6 +1229,9 @@ static int qcom_slim_ngd_runtime_resume(struct device *dev) struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev); int ret = 0; + if (!ctrl->qmi.handle) + return 0; + if (ctrl->state >= QCOM_SLIM_NGD_CTRL_ASLEEP) ret = qcom_slim_ngd_power_up(ctrl); if (ret) { @@ -1616,6 +1619,9 @@ static int __maybe_unused qcom_slim_ngd_runtime_suspend(struct device *dev) struct qcom_slim_ngd_ctrl *ctrl = dev_get_drvdata(dev); int ret = 0; + if (!ctrl->qmi.handle) + return 0; + ret = qcom_slim_qmi_power_request(ctrl, false); if (ret && ret != -EBUSY) dev_info(ctrl->dev, "slim resource not idle:%d\n", ret); -- 2.29.2

2 1

[PATCH 0/5] soundwire: only clear valid interrupts
by Bard Liao 25 Nov '20

25 Nov '20

We wrote 1 to the handled interrupts bits along with 0 to all other bits to the SoundWire DPx interrupt register. However, DP0 has reserved fields and the read-only SDCA_CASCADE bit. DPN also has reserved fields. We should not try to write values in these fields. Besides, we deal with pending interrupts in a loop but we didn't reset the slave_notify status. Pierre-Louis Bossart (5): soundwire: bus: add comments to explain interrupt loop filter soundwire: bus: reset slave_notify status at each loop soundwire: registers: add definitions for clearable interrupt fields soundwire: bus: only clear valid DP0 interrupts soundwire: bus: only clear valid DPN interrupts drivers/soundwire/bus.c | 27 +++++++++++++++++-------- include/linux/soundwire/sdw_registers.h | 11 ++++++++++ 2 files changed, 30 insertions(+), 8 deletions(-) -- 2.17.1

2 6

Dear akm.com, please publish/share AK4375[A] DAC datasheet
by Vincent Knecht 25 Nov '20

25 Nov '20

Thanks in advance

1 0

SB Omni Surround 5.1
by GitHub issues - opened 24 Nov '20

24 Nov '20

alsa-project/alsa-lib issue #101 was opened from witoldo7: Hi, SB Omni Surround 5.1 use default hw 1 for iec958.0 please update USB-Audio.conf, 0 is used for dac(jack connectors) Issue URL : https://github.com/alsa-project/alsa-lib/issues/101 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

jack: add option to allow non-jack-aligned period size
by GitHub pull_request - opened 24 Nov '20

24 Nov '20

alsa-project/alsa-plugins pull request #14 was opened from aditpape: Commit 72337f1dcb2df6a2b7fdae93e88d7dd3a7e97ab4 forces ALSA period size to be exact multiple of the JACK daemon period size. There are good reasons to align the period size, but completely disallowing non aligned periods is too restrictive. This patch adds an option "align_psize" which allows to optionally revert to original behavior. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-plugins/pull/14 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/14.patch Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: hda/realtek: Enable headset of ASUS UX482EG & B9400CEA with ALC294
by Jian-Hong Pan 24 Nov '20

24 Nov '20

Some laptops like ASUS UX482EG & B9400CEA's headset audio does not work until the quirk ALC294_FIXUP_ASUS_HPE is applied. Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 739dbaf54517..1aafc55f1505 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8585,6 +8585,9 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x13, 0x90a60140}), + SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_HPE, + {0x17, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0294, 0x1043, "ASUS", ALC294_FIXUP_ASUS_MIC, {0x14, 0x90170110}, {0x1b, 0x90a70130}, -- 2.29.2

2 1

[PATCH] slimbus: qcom-ngd-ctrl: Constify static structs
by Rikard Falkeborn 24 Nov '20

24 Nov '20

qcom_slim_qmi_msg_handlers[] and qcom_slim_ngd_qmi_svc_event_ops are only used as input arguments to qmi_handle_init() which accepts const pointers to both qmi_ops and qmi_msg_handler. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index d8decb345e9d..ea8f605b5c99 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -434,7 +434,7 @@ static int qcom_slim_qmi_send_power_request(struct qcom_slim_ngd_ctrl *ctrl, return 0; } -static struct qmi_msg_handler qcom_slim_qmi_msg_handlers[] = { +static const struct qmi_msg_handler qcom_slim_qmi_msg_handlers[] = { { .type = QMI_RESPONSE, .msg_id = SLIMBUS_QMI_POWER_RESP_V01, @@ -1309,7 +1309,7 @@ static void qcom_slim_ngd_qmi_del_server(struct qmi_handle *hdl, qmi->svc_info.sq_port = 0; } -static struct qmi_ops qcom_slim_ngd_qmi_svc_event_ops = { +static const struct qmi_ops qcom_slim_ngd_qmi_svc_event_ops = { .new_server = qcom_slim_ngd_qmi_new_server, .del_server = qcom_slim_ngd_qmi_del_server, }; -- 2.29.2

2 1

[PATCH] soundwire: SDCA: detect sdca_cascade interrupt
by Bard Liao 24 Nov '20

24 Nov '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The SoundWire 1.2 specification defines an "SDCA cascade" bit which handles a logical OR of all SDCA interrupt sources (up to 30 defined). Due to limitations of the addressing space, this bit is located in the SDW_DP0_INT register when DP0 is used, or alternatively in the DP0_SDCA_Support_INTSTAT register when DP0 is not used. To allow for both cases to be handled, this bit will be checked in the main device-level interrupt handling code. This will result in the register being read twice if DP0 is enabled, but it's not clear how to optimize this case. It's also more logical to deal with this interrupt at the device than the port level, this bit is really not DP0 specific and its location in the DP0_INTSTAT bit is only due to the lack of free space in SCP_INTSTAT_1. The SDCA_Cascade bit cannot be masked or cleared, so the interrupt handling only forwards the detection to the Slave driver, which will deal with reading the relevant SDCA status bits and clearing them. The bus driver only signals the detection. The communication with the Slave driver is based on the same interrupt callback, with only an extension to provide the status of the sdca_cascade bit. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 28 +++++++++++++++++++++++++++- include/linux/soundwire/sdw.h | 4 ++++ 2 files changed, 31 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 8eaf31e76677..ffe4600fd95b 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1424,6 +1424,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; + u8 sdca_cascade = 0; u8 buf, buf2[2], _buf, _buf2[2]; bool parity_check; bool parity_quirk; @@ -1453,6 +1454,16 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) goto io_err; } + if (slave->prop.is_sdca) { + ret = sdw_read(slave, SDW_DP0_INT); + if (ret < 0) { + dev_err(slave->bus->dev, + "SDW_DP0_INT read failed:%d\n", ret); + goto io_err; + } + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + } + do { /* * Check parity, bus clash and Slave (impl defined) @@ -1489,6 +1500,10 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) clear |= SDW_SCP_INT1_IMPL_DEF; } + /* the SDCA interrupts are cleared in the codec driver .interrupt_callback() */ + if (sdca_cascade) + slave_notify = true; + /* Check port 0 - 3 interrupts */ port = buf & SDW_SCP_INT1_PORT0_3; @@ -1526,6 +1541,7 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) /* Update the Slave driver */ if (slave_notify && slave->ops && slave->ops->interrupt_callback) { + slave_intr.sdca_cascade = sdca_cascade; slave_intr.control_port = clear; memcpy(slave_intr.port, &port_status, sizeof(slave_intr.port)); @@ -1563,11 +1579,21 @@ static int sdw_handle_slave_alerts(struct sdw_slave *slave) goto io_err; } + if (slave->prop.is_sdca) { + ret = sdw_read(slave, SDW_DP0_INT); + if (ret < 0) { + dev_err(slave->bus->dev, + "SDW_DP0_INT read failed:%d\n", ret); + goto io_err; + } + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + } + /* Make sure no interrupts are pending */ buf &= _buf; buf2[0] &= _buf2[0]; buf2[1] &= _buf2[1]; - stat = buf || buf2[0] || buf2[1]; + stat = buf || buf2[0] || buf2[1] || sdca_cascade; /* * Exit loop if Slave is continuously in ALERT state even diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 41cc1192f9aa..f0b01b728640 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -359,6 +359,7 @@ struct sdw_dpn_prop { * @sink_dpn_prop: Sink Data Port N properties * @scp_int1_mask: SCP_INT1_MASK desired settings * @quirks: bitmask identifying deltas from the MIPI specification + * @is_sdca: the Slave supports the SDCA specification */ struct sdw_slave_prop { u32 mipi_revision; @@ -382,6 +383,7 @@ struct sdw_slave_prop { struct sdw_dpn_prop *sink_dpn_prop; u8 scp_int1_mask; u32 quirks; + bool is_sdca; }; #define SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY BIT(0) @@ -479,10 +481,12 @@ struct sdw_slave_id { /** * struct sdw_slave_intr_status - Slave interrupt status + * @sdca_cascade: set if the Slave device reports an SDCA interrupt * @control_port: control port status * @port: data port status */ struct sdw_slave_intr_status { + bool sdca_cascade; u8 control_port; u8 port[15]; }; -- 2.17.1

2 1

[PATCH] soundwire: Fix DEBUG_LOCKS_WARN_ON for uninitialized attribute
by Srinivas Kandagatla 24 Nov '20

24 Nov '20

running kernel with CONFIG_DEBUG_LOCKS_ALLOC enabled will below warning: BUG: key ffff502e09807098 has not been registered! DEBUG_LOCKS_WARN_ON(1) WARNING: CPU: 5 PID: 129 at kernel/locking/lockdep.c:4623 lockdep_init_map_waits+0xe8/0x250 Modules linked in: CPU: 5 PID: 129 Comm: kworker/5:1 Tainted: G W 5.10.0-rc1-00277-ged49f224ca3f-dirty #1210 Hardware name: Qualcomm Technologies, Inc. Robotics RB5 (DT) Workqueue: events deferred_probe_work_func pstate: 80c00005 (Nzcv daif +PAN +UAO -TCO BTYPE=--) pc : lockdep_init_map_waits+0xe8/0x250 lr : lockdep_init_map_waits+0xe8/0x250 [ Trimmed ] Call trace: lockdep_init_map_waits+0xe8/0x250 __kernfs_create_file+0x78/0x180 sysfs_add_file_mode_ns+0x94/0x1c8 internal_create_group+0x110/0x3e0 sysfs_create_group+0x18/0x28 devm_device_add_group+0x4c/0xb0 add_all_attributes+0x438/0x490 sdw_slave_sysfs_dpn_init+0x128/0x138 sdw_slave_sysfs_init+0x80/0xa0 sdw_drv_probe+0x94/0x170 really_probe+0x118/0x3e0 driver_probe_device+0x5c/0xc0 [ Trimmed ] CPU: 5 PID: 129 Comm: kworker/5:1 Tainted: G W 5.10.0-rc1-00277-ged49f224ca3f-dirty #1210 Hardware name: Qualcomm Technologies, Inc. Robotics RB5 (DT) Workqueue: events deferred_probe_work_func Call trace: dump_backtrace+0x0/0x1c0 show_stack+0x18/0x68 dump_stack+0xd8/0x134 __warn+0xa0/0x158 report_bug+0xc8/0x178 bug_handler+0x20/0x78 brk_handler+0x70/0xc8 [ Trimmed ] Fix this by initializing dynamically allocated sysfs attribute to keep lockdep happy! Fixes: bcac59029955 ("soundwire: add Slave sysfs support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/sysfs_slave_dpn.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/sysfs_slave_dpn.c b/drivers/soundwire/sysfs_slave_dpn.c index 05a721ea9830..c4b6543c09fd 100644 --- a/drivers/soundwire/sysfs_slave_dpn.c +++ b/drivers/soundwire/sysfs_slave_dpn.c @@ -37,6 +37,7 @@ static int field##_attribute_alloc(struct device *dev, \ return -ENOMEM; \ dpn_attr->N = N; \ dpn_attr->dir = dir; \ + sysfs_attr_init(&dpn_attr->dev_attr.attr); \ dpn_attr->format_string = format_string; \ dpn_attr->dev_attr.attr.name = __stringify(field); \ dpn_attr->dev_attr.attr.mode = 0444; \ -- 2.21.0

3 2

[PATCH] ASoC: imx-audmux: Remove unused .id_table
by Fabio Estevam 23 Nov '20

23 Nov '20

Since 5.10-rc1 i.MX is a devicetree-only platform and the existing .id_table support in this driver was only useful for old non-devicetree platforms. Get rid of the .id_table since it is no longer used. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/imx-audmux.c | 28 ++++------------------------ 1 file changed, 4 insertions(+), 24 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 25c18b9e348f..dfa05d40b276 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -170,22 +170,9 @@ static enum imx_audmux_type { IMX31_AUDMUX, } audmux_type; -static const struct platform_device_id imx_audmux_ids[] = { - { - .name = "imx21-audmux", - .driver_data = IMX21_AUDMUX, - }, { - .name = "imx31-audmux", - .driver_data = IMX31_AUDMUX, - }, { - /* sentinel */ - } -}; -MODULE_DEVICE_TABLE(platform, imx_audmux_ids); - static const struct of_device_id imx_audmux_dt_ids[] = { - { .compatible = "fsl,imx21-audmux", .data = &imx_audmux_ids[0], }, - { .compatible = "fsl,imx31-audmux", .data = &imx_audmux_ids[1], }, + { .compatible = "fsl,imx21-audmux", .data = (void *)IMX21_AUDMUX, }, + { .compatible = "fsl,imx31-audmux", .data = (void *)IMX31_AUDMUX, }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, imx_audmux_dt_ids); @@ -300,9 +287,6 @@ static int imx_audmux_parse_dt_defaults(struct platform_device *pdev, static int imx_audmux_probe(struct platform_device *pdev) { - const struct of_device_id *of_id = - of_match_device(imx_audmux_dt_ids, &pdev->dev); - audmux_base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(audmux_base)) return PTR_ERR(audmux_base); @@ -314,9 +298,7 @@ static int imx_audmux_probe(struct platform_device *pdev) audmux_clk = NULL; } - if (of_id) - pdev->id_entry = of_id->data; - audmux_type = pdev->id_entry->driver_data; + audmux_type = (enum imx_audmux_type)of_device_get_match_data(&pdev->dev); switch (audmux_type) { case IMX31_AUDMUX: @@ -335,8 +317,7 @@ static int imx_audmux_probe(struct platform_device *pdev) if (!regcache) return -ENOMEM; - if (of_id) - imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); return 0; } @@ -386,7 +367,6 @@ static const struct dev_pm_ops imx_audmux_pm = { static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, .remove = imx_audmux_remove, - .id_table = imx_audmux_ids, .driver = { .name = DRIVER_NAME, .pm = &imx_audmux_pm, -- 2.17.1

2 1

[PATCH 1/4] ASoC: meson: depend on COMMON_CLK to fix compile tests
by Krzysztof Kozlowski 23 Nov '20

23 Nov '20

The Meson SoC sound drivers use Common Clock Framework thus they cannot be built on platforms without it (e.g. compile test on MIPS with RALINK and SOC_RT305X): /usr/bin/mips-linux-gnu-ld: sound/soc/meson/aiu-encoder-spdif.o: in function `aiu_encoder_spdif_startup': aiu-encoder-spdif.c:(.text+0x3a0): undefined reference to `clk_set_parent' /usr/bin/mips-linux-gnu-ld: sound/soc/meson/axg-tdm-formatter.o: in function `axg_tdm_formatter_event': (.text+0x7ec): undefined reference to `clk_set_parent' Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/meson/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 363dc3b1bbe4..dd8a05e61f58 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only menu "ASoC support for Amlogic platforms" - depends on ARCH_MESON || COMPILE_TEST + depends on ARCH_MESON || COMPILE_TEST && COMMON_CLK config SND_MESON_AIU tristate "Amlogic AIU" -- 2.25.1

2 5

[PATCH] Asoc: qcom: Fix enabling BCLK and LRCLK in LPAIF invalid state
by Srinivasa Rao Mandadapu 23 Nov '20

23 Nov '20

Fix enabling BCLK and LRCLK only when LPAIF is invalid state and bit clock in enable state. In device suspend/resume scenario LPAIF is going to reset state. which is causing LRCLK disable and BCLK enable. Avoid such inconsitency by removing unnecessary cpu dai prepare API, which is doing LRCLK enable, and by maintaining BLCK state information. Fixes: 7e6799d8f87d ("ASoC: qcom: lpass-cpu: Enable MI2S BCLK and LRCLK together") Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 62 ++++++++++++++++++++-------------------- sound/soc/qcom/lpass-lpaif-reg.h | 7 +++++ sound/soc/qcom/lpass-platform.c | 20 +++++++++++-- sound/soc/qcom/lpass.h | 1 + 4 files changed, 57 insertions(+), 33 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index d33eae6..af684fd 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -263,28 +263,6 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, return 0; } -static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; - unsigned int id = dai->driver->id; - int ret; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = regmap_fields_write(i2sctl->spken, id, - LPAIF_I2SCTL_SPKEN_ENABLE); - } else { - ret = regmap_fields_write(i2sctl->micen, id, - LPAIF_I2SCTL_MICEN_ENABLE); - } - - if (ret) - dev_err(dai->dev, "error writing to i2sctl enable: %d\n", ret); - - return ret; -} - static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -292,6 +270,18 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; unsigned int id = dai->driver->id; int ret = -EINVAL; + unsigned int val = 0; + + ret = regmap_read(drvdata->lpaif_map, + LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), &val); + if (ret) { + dev_err(dai->dev, "error reading from i2sctl reg: %d\n", ret); + return ret; + } + if (val == LPAIF_I2SCTL_RESET_STATE) { + dev_err(dai->dev, "error in i2sctl register state\n"); + return -ENOTRECOVERABLE; + } switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -308,11 +298,14 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - ret = clk_enable(drvdata->mi2s_bit_clk[id]); - if (ret) { - dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); - clk_disable(drvdata->mi2s_osr_clk[id]); - return ret; + if (drvdata->bit_clk_state[id] == LPAIF_BIT_CLK_DISABLE) { + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + drvdata->bit_clk_state[id] = LPAIF_BIT_CLK_ENABLE; } break; @@ -329,7 +322,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, if (ret) dev_err(dai->dev, "error writing to i2sctl reg: %d\n", ret); - clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + if (drvdata->bit_clk_state[id] == LPAIF_BIT_CLK_ENABLE) { + clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); + drvdata->bit_clk_state[id] = LPAIF_BIT_CLK_DISABLE; + } break; } @@ -341,7 +337,6 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, - .prepare = lpass_cpu_daiops_prepare, .trigger = lpass_cpu_daiops_trigger, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); @@ -459,16 +454,20 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; + for (i = 0; i < v->i2s_ports; ++i) + if (reg == LPAIF_I2SCTL_REG(v, i)) + return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || + reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) return true; return false; @@ -861,6 +860,7 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + drvdata->bit_clk_state[dai_id] = LPAIF_BIT_CLK_DISABLE; } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-lpaif-reg.h b/sound/soc/qcom/lpass-lpaif-reg.h index 08f3fe5..4055428 100644 --- a/sound/soc/qcom/lpass-lpaif-reg.h +++ b/sound/soc/qcom/lpass-lpaif-reg.h @@ -60,6 +60,13 @@ #define LPAIF_I2SCTL_BITWIDTH_24 1 #define LPAIF_I2SCTL_BITWIDTH_32 2 +#define LPAIF_BIT_CLK_DISABLE 0 +#define LPAIF_BIT_CLK_ENABLE 1 + +#define LPAIF_I2SCTL_RESET_STATE 0x003C0004 +#define LPAIF_DMACTL_RESET_STATE 0x00200000 + + /* LPAIF IRQ */ #define LPAIF_IRQ_REG_ADDR(v, addr, port) \ (v->irq_reg_base + (addr) + v->irq_reg_stride * (port)) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 7a3fdf8..80b09de 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -110,6 +110,7 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component, struct regmap *map; unsigned int dai_id = cpu_dai->driver->id; + component->id = dai_id; data = kzalloc(sizeof(*data), GFP_KERNEL); if (!data) return -ENOMEM; @@ -451,19 +452,34 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, unsigned int reg_irqclr = 0, val_irqclr = 0; unsigned int reg_irqen = 0, val_irqen = 0, val_mask = 0; unsigned int dai_id = cpu_dai->driver->id; + unsigned int dma_ctrl_reg = 0; ch = pcm_data->dma_ch; if (dir == SNDRV_PCM_STREAM_PLAYBACK) { id = pcm_data->dma_ch; - if (dai_id == LPASS_DP_RX) + if (dai_id == LPASS_DP_RX) { dmactl = drvdata->hdmi_rd_dmactl; - else + map = drvdata->hdmiif_map; + } else { dmactl = drvdata->rd_dmactl; + map = drvdata->lpaif_map; + } } else { dmactl = drvdata->wr_dmactl; id = pcm_data->dma_ch - v->wrdma_channel_start; + map = drvdata->lpaif_map; + } + ret = regmap_read(map, LPAIF_DMACTL_REG(v, ch, dir, dai_id), &dma_ctrl_reg); + if (ret) { + dev_err(soc_runtime->dev, "error reading from rdmactl reg: %d\n", ret); + return ret; } + if (dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE || + dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE + 1) { + dev_err(soc_runtime->dev, "error in rdmactl register state\n"); + return -ENOTRECOVERABLE; + } switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 32a68c4..0195372 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -68,6 +68,7 @@ struct lpass_data { unsigned int mi2s_playback_sd_mode[LPASS_MAX_MI2S_PORTS]; unsigned int mi2s_capture_sd_mode[LPASS_MAX_MI2S_PORTS]; int hdmi_port_enable; + int bit_clk_state[LPASS_MAX_MI2S_PORTS]; /* low-power audio interface (LPAIF) registers */ void __iomem *lpaif; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH v2] ASoC: hdmi-codec: Get ELD in before reporting plugged event
by Cheng-Yi Chiang 23 Nov '20

23 Nov '20

In plugged callback, ELD should be updated from display driver so that user space can query information from ELD immediately after receiving jack plugged event. When jack is unplugged, clear ELD buffer so that user space does not get obsolete information of unplugged HDMI. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Change from v1 to v2: - Report jack unplug event then clear ELD. This makes sure user will not get a cleared ELD during their usage of jack. sound/soc/codecs/hdmi-codec.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 403d4c6a49a8..e8410b2433de 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -692,10 +692,16 @@ static void plugged_cb(struct device *dev, bool plugged) { struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); - if (plugged) + if (plugged) { + if (hcp->hcd.ops->get_eld) { + hcp->hcd.ops->get_eld(dev->parent, hcp->hcd.data, + hcp->eld, sizeof(hcp->eld)); + } hdmi_codec_jack_report(hcp, SND_JACK_LINEOUT); - else + } else { hdmi_codec_jack_report(hcp, 0); + memset(hcp->eld, 0, sizeof(hcp->eld)); + } } static int hdmi_codec_set_jack(struct snd_soc_component *component, -- 2.29.2.299.gdc1121823c-goog

2 1

[RFC] MAINTAINERS tag for cleanup robot
by trix＠redhat.com 23 Nov '20

23 Nov '20

A difficult part of automating commits is composing the subsystem preamble in the commit log. For the ongoing effort of a fixer producing one or two fixes a release the use of 'treewide:' does not seem appropriate. It would be better if the normal prefix was used. Unfortunately normal is not consistent across the tree. So I am looking for comments for adding a new tag to the MAINTAINERS file D: Commit subsystem prefix ex/ for FPGA DFL DRIVERS D: fpga: dfl: Continuing with cleaning up clang's -Wextra-semi-stmt A significant number of warnings are caused by function like macros with a trailing semicolon. For example. #define FOO(a) a++; <-- extra, unneeded semicolon void bar() { int v = 0; FOO(a); } Clang will warn at the FOO(a); expansion location. Instead of removing the semicolon there, the fixer removes semicolon from the macro definition. After the fixer, the code will be: #define FOO(a) a++ void bar() { int v = 0; FOO(a); } The fixer review is https://reviews.llvm.org/D91789 A run over allyesconfig for x86_64 finds 62 issues, 5 are false positives. The false positives are caused by macros passed to other macros and by some macro expansions that did not have an extra semicolon. This cleans up about 1,000 of the current 10,000 -Wextra-semi-stmt warnings in linux-next. An update to [RFC] clang tooling cleanup This change adds the clang-tidy-fix as a top level target and uses it to do the cleaning. The next iteration will do a loop of cleaners. This will mean breaking clang-tidy-fix out into its own processing function 'run_fixers'. Makefile: Add toplevel target clang-tidy-fix to makefile Calls clang-tidy with -fix option for a set of checkers that programatically fixes the kernel source in place, treewide. Signed-off-by: Tom Rix <trix(a)redhat.com> --- Makefile | 7 ++++--- scripts/clang-tools/run-clang-tools.py | 20 +++++++++++++++++--- 2 files changed, 21 insertions(+), 6 deletions(-) diff --git a/Makefile b/Makefile index 47a8add4dd28..57756dbb767b 100644 --- a/Makefile +++ b/Makefile @@ -1567,20 +1567,21 @@ help: echo '' @echo 'Static analysers:' @echo ' checkstack - Generate a list of stack hogs' @echo ' versioncheck - Sanity check on version.h usage' @echo ' includecheck - Check for duplicate included header files' @echo ' export_report - List the usages of all exported symbols' @echo ' headerdep - Detect inclusion cycles in headers' @echo ' coccicheck - Check with Coccinelle' @echo ' clang-analyzer - Check with clang static analyzer' @echo ' clang-tidy - Check with clang-tidy' + @echo ' clang-tidy-fix - Check and fix with clang-tidy' @echo '' @echo 'Tools:' @echo ' nsdeps - Generate missing symbol namespace dependencies' @echo '' @echo 'Kernel selftest:' @echo ' kselftest - Build and run kernel selftest' @echo ' Build, install, and boot kernel before' @echo ' running kselftest on it' @echo ' Run as root for full coverage' @echo ' kselftest-all - Build kernel selftest' @@ -1842,30 +1843,30 @@ nsdeps: modules quiet_cmd_gen_compile_commands = GEN $@ cmd_gen_compile_commands = $(PYTHON3) $< -a $(AR) -o $@ $(filter-out $<, $(real-prereqs)) $(extmod-prefix)compile_commands.json: scripts/clang-tools/gen_compile_commands.py \ $(if $(KBUILD_EXTMOD),,$(KBUILD_VMLINUX_OBJS) $(KBUILD_VMLINUX_LIBS)) \ $(if $(CONFIG_MODULES), $(MODORDER)) FORCE $(call if_changed,gen_compile_commands) targets += $(extmod-prefix)compile_commands.json -PHONY += clang-tidy clang-analyzer +PHONY += clang-tidy-fix clang-tidy clang-analyzer ifdef CONFIG_CC_IS_CLANG quiet_cmd_clang_tools = CHECK $< cmd_clang_tools = $(PYTHON3) $(srctree)/scripts/clang-tools/run-clang-tools.py $@ $< -clang-tidy clang-analyzer: $(extmod-prefix)compile_commands.json +clang-tidy-fix clang-tidy clang-analyzer: $(extmod-prefix)compile_commands.json $(call cmd,clang_tools) else -clang-tidy clang-analyzer: +clang-tidy-fix clang-tidy clang-analyzer: @echo "$@ requires CC=clang" >&2 @false endif # Scripts to check various things for consistency # --------------------------------------------------------------------------- PHONY += includecheck versioncheck coccicheck export_report includecheck: diff --git a/scripts/clang-tools/run-clang-tools.py b/scripts/clang-tools/run-clang-tools.py index fa7655c7cec0..c177ca822c56 100755 --- a/scripts/clang-tools/run-clang-tools.py +++ b/scripts/clang-tools/run-clang-tools.py @@ -22,43 +22,57 @@ def parse_arguments(): Returns: args: Dict of parsed args Has keys: [path, type] """ usage = """Run clang-tidy or the clang static-analyzer on a compilation database.""" parser = argparse.ArgumentParser(description=usage) type_help = "Type of analysis to be performed" parser.add_argument("type", - choices=["clang-tidy", "clang-analyzer"], + choices=["clang-tidy-fix", "clang-tidy", "clang-analyzer"], help=type_help) path_help = "Path to the compilation database to parse" parser.add_argument("path", type=str, help=path_help) return parser.parse_args() def init(l, a): global lock global args lock = l args = a def run_analysis(entry): # Disable all checks, then re-enable the ones we want checks = "-checks=-*," - if args.type == "clang-tidy": + fix = "" + header_filter = "" + if args.type == "clang-tidy-fix": + checks += "linuxkernel-macro-trailing-semi" + # + # Fix this + # #define M(a) a++; <-- clang-tidy fixes the problem here + # int f() { + # int v = 0; + # M(v); <-- clang reports problem here + # return v; + # } + fix += "-fix" + header_filter += "-header-filter=.*" + elif args.type == "clang-tidy": checks += "linuxkernel-*" else: checks += "clang-analyzer-*" - p = subprocess.run(["clang-tidy", "-p", args.path, checks, entry["file"]], + p = subprocess.run(["clang-tidy", "-p", args.path, checks, header_filter, fix, entry["file"]], stdout=subprocess.PIPE, stderr=subprocess.STDOUT, cwd=entry["directory"]) with lock: sys.stderr.buffer.write(p.stdout) def main(): args = parse_arguments() -- 2.18.4

8 16

[PATCH 01/38] ASoC: ak5558: drop of_match_ptr from of_device_id table
by Krzysztof Kozlowski 23 Nov '20

23 Nov '20

The driver can match only via the DT table so the table should be always used and the of_match_ptr does not have any sense (this also allows ACPI matching via PRP0001, even though it is not relevant here). This fixes compile warning (!CONFIG_OF on x86_64): sound/soc/codecs/ak5558.c:418:34: warning: ‘ak5558_i2c_dt_ids’ defined but not used [-Wunused-const-variable=] Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/codecs/ak5558.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 2f076d5ee284..427d7d51bf53 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -423,7 +423,7 @@ static const struct of_device_id ak5558_i2c_dt_ids[] = { static struct i2c_driver ak5558_i2c_driver = { .driver = { .name = "ak5558", - .of_match_table = of_match_ptr(ak5558_i2c_dt_ids), + .of_match_table = ak5558_i2c_dt_ids, .pm = &ak5558_pm, }, .probe_new = ak5558_i2c_probe, -- 2.25.1

5 56

Poor performace on mmap reading arm64 on audio device
by Michael Nazzareno Trimarchi 23 Nov '20

23 Nov '20

Hi all I'm trying to figure out how to increase performance on audio reading using the mmap interface. Right now what I understand it's that allocation comes from core/memalloc.c ops that allocate the memory for dma under driver/dma. The reference platform I have is an imx8mm and the allocation in arm64 is: 0xffff800011ff5000-0xffff800012005000 64K PTE RW NX SHD AF UXN MEM/NORMAL-NC This is the reason that is allocated for dma interface. Now access linear on the multichannel interface the performance is bad but worse if I try to access a channel a time on read. So it looks like it is better to copy the block using memcpy on a cached area and then operate on a single channel sample. If it's correct what I'm saying the mmap_begin and mmap_commit basically they don't do anything on cache level so the page mapping and way is used is always the same. Can the interface be modified to allow cache the area during read and restore in the commit phase? Michael

2 6

[PATCH v5 0/6] Tegra210 audio graph card
by Sameer Pujar 23 Nov '20

23 Nov '20

This series adds audio graph based sound card support for Tegra210 platforms like Jetson-TX1 an Jetson-Nano. The following preparatory audio graph enhancement series is already merged. * https://patchwork.kernel.org/project/alsa-devel/list/?series=375629&state=* Following are the summary of changes: * Add graph/audio-graph based schemas or schema updates for Tegra210 component and machine drivers. * Add Tegra audio graph machine driver. * Add required DT support for Jetson-TX1/Nano. This work is based on earlier discussion of DPCM usage for Tegra and simple card driver updates. * https://lkml.org/lkml/2020/4/30/519 * https://lkml.org/lkml/2020/6/27/4 This series has dependency over following graph and audio-graph series. * http://patchwork.ozlabs.org/project/devicetree-bindings/patch/2020110220365… * https://patchwork.kernel.org/project/alsa-devel/list/?series=382009&state=* Changelog ========= v4 -> v5 -------- * Audio graph related changes were sent in separate v5 series as mentioned above and are dropped from current series. * Graph and audio graph doc patches are dropped from this series and are sent separately as mentioned above. * Minor change with phandle label for TX1 and Nano platform DT files. * No changes in other patches. v3 -> v4 -------- * Added new patches to convert graph.txt and audio-graph-card.txt to corresponding json-schema files. Later these references are used in Tegra audio graph schema. * AHUB component binding docs are updated to reflect the usage of ports/port/endpoint * More common stuff is moved into graph_parse_of() and this is used by both generic and Tegra audio graph. * DT binding for Tegra audio graph is updated to included "ports { }" * As per the suggestion 'void *data' member is dropped from 'asoc_simple_priv' and instead container method is used to maintain required custom data internal to Tegra audio graph. v2 -> v3 -------- * Dropped new compatible addition in generic graph driver after reviewing it with Morimoto-san. Instead added Tegra audio graph driver and new compatibles are added in the same. * Added new patches to expose new members for customization in audio graph driver. * Added new patch for Tegra audio graph driver and related documentation. * Minor change in below commit where mutex version of helper is used "ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM" * DT binding is updated to use the newly exposed compatibles * No changes in other patches v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. Sameer Pujar (6): ASoC: dt-bindings: tegra: Add graph bindings ASoC: dt-bindings: tegra: Add json-schema for Tegra audio graph card ASoC: tegra: Add audio graph based card driver arm64: defconfig: Enable Tegra audio graph card driver arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../sound/nvidia,tegra-audio-graph-card.yaml | 187 +++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 6 + .../bindings/sound/nvidia,tegra210-admaif.yaml | 6 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 13 +- .../bindings/sound/nvidia,tegra210-dmic.yaml | 6 + .../bindings/sound/nvidia,tegra210-i2s.yaml | 6 + .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 153 ++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 262 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 146 ++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_audio_graph_card.c | 255 ++++++++++++++++++++ 13 files changed, 1050 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c -- 2.7.4

3 11

[PATCH] ALSA: usb-audio : Add support for the Pioneer DJ DDJ-RR controller.
by Daniel 21 Nov '20

21 Nov '20

--- sound/usb/quirks-table.h | 56 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 55 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 3c1697f6b60c..a39233cb4d72 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3530,7 +3530,61 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, - +{ + /* + * PIONEER DJ DDJ-RR + * PCM is 6 channels out & 4 channels in @ 44.1 fixed + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000d), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, //Master, Headphones & Booth + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, //2x RCA inputs (CH1 & CH2) + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-900NXS2 -- 2.29.2

3 2

[PATCH V2] ALSA: emu10k1: Use dma_set_mask_and_coherent to simplify code
by Christophe JAILLET 21 Nov '20

21 Nov '20

'pci_set_dma_mask()' + 'pci_set_consistent_dma_mask()' can be replaced by an equivalent 'dma_set_mask_and_coherent()' which is much less verbose. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- v2: do not update the dev_err message, to keep consistency with other dev_err --- sound/pci/emu10k1/emu10k1x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index def8161cde4c..f2b6fd434474 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -894,8 +894,8 @@ static int snd_emu10k1x_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || - pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { + + if (dma_set_mask_and_coherent(&pci->dev, DMA_BIT_MASK(28)) < 0) { dev_err(card->dev, "error to set 28bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; -- 2.27.0

2 1

[PATCH] ALSA: aloop: Constify ops structs
by Rikard Falkeborn 21 Nov '20

21 Nov '20

The only usage of the ops field in the loopback_cable struct is to call its members, the field it self is never changed. Make it a pointer to const. This allows us to constify two static loopback_ops structs to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/drivers/aloop.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index c91356326699..702f91b9c60f 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -105,7 +105,7 @@ struct loopback_cable { unsigned int running; unsigned int pause; /* timer specific */ - struct loopback_ops *ops; + const struct loopback_ops *ops; /* If sound timer is used */ struct { int stream; @@ -1021,7 +1021,7 @@ static int loopback_jiffies_timer_open(struct loopback_pcm *dpcm) return 0; } -static struct loopback_ops loopback_jiffies_timer_ops = { +static const struct loopback_ops loopback_jiffies_timer_ops = { .open = loopback_jiffies_timer_open, .start = loopback_jiffies_timer_start, .stop = loopback_jiffies_timer_stop, @@ -1172,7 +1172,7 @@ static int loopback_snd_timer_open(struct loopback_pcm *dpcm) /* stop_sync() is not required for sound timer because it does not need to be * restarted in loopback_prepare() on Xrun recovery */ -static struct loopback_ops loopback_snd_timer_ops = { +static const struct loopback_ops loopback_snd_timer_ops = { .open = loopback_snd_timer_open, .start = loopback_snd_timer_start, .stop = loopback_snd_timer_stop, -- 2.29.2

2 1

[PATCH] ALSA: emu10k1: Use dma_set_mask_and_coherent to simplify code
by Christophe JAILLET 21 Nov '20

21 Nov '20

'pci_set_dma_mask()' + 'pci_set_consistent_dma_mask()' can be replaced by an equivalent 'dma_set_mask_and_coherent()' which is much less verbose. With at it replace the 'card->dev' parameter passed to a 'dev_err()' call by an equivalent '&pci->dev' which is more consistent with the test just one line above. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/pci/emu10k1/emu10k1x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index def8161cde4c..f2b6fd434474 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -894,9 +894,9 @@ static int snd_emu10k1x_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || - pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { - dev_err(card->dev, "error to set 28bit mask DMA\n"); + + if (dma_set_mask_and_coherent(&pci->dev, DMA_BIT_MASK(28)) < 0) { + dev_err(&pci->dev, "error to set 28bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; } -- 2.27.0

2 1

Announcement for alsa-gobject v0.2.0 release
by Takashi Sakamoto 21 Nov '20

21 Nov '20

Hi, This is an announcement for the second official release of alsa-gobject project[1]. The release is available in below URL: * https://github.com/alsa-project/alsa-gobject/releases/tag/v0.2.0 It includes some enhancements and bug fixes. * Some implementations of GLib GQuark is exported as error domain and corresponding enumerations are added. This improves error reporting in each libraries. It closes issue #47[2]. * Selection of substream in ALSARawmidi.StreamPair.open() now works well. This closes issue #53[3]. * Prototypes of some methods in ALSACtl.Card object for data of TLV(Type-Length-Value) in ALSA control interface are changed without backward compatibility. Corresponding exported symbols are maintained by new symbol version. In detail, refer to #55[4]. * Error handling is changed so that function call with invalid argument just returns with warnings instead of generating GError, according to convention for GLib application. Memory allocation failure is supposed to be unrecoverable error and aborts program, as well. * It's possible to build for older version Linux kernel, therefore build dependency is now Linux kernel v4.5 or later. For the new release, documentations are also updated. They're auto-generated by gtk-doc[5] from function annotations: * https://alsa-project.github.io/alsa-gobject-docs/ The new release of Rust crates are available as well in repository. They're auto-generated by gtk-rs/gir[6]: * https://github.com/alsa-project/alsa-gobject-rs/releases/tag/v0.2.0 I note that ALSA PCM interface is not supported yet. Please refer to my announcement for v0.1.0 for the reason[7]. [1] https://github.com/alsa-project/alsa-gobject [2] https://github.com/alsa-project/alsa-gobject/issues/47 [3] https://github.com/alsa-project/alsa-gobject/issues/53 [4] https://github.com/alsa-project/alsa-gobject/pull/55 [5] https://developer.gnome.org/gtk-doc-manual/ [6] https://github.com/gtk-rs/gir [7] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-June/169488.html Regards Takashi Sakamoto

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[PATCH] ASoC: SOF: IPC: fix implicit type overflow
by Kai Vehmanen 20 Nov '20

20 Nov '20

From: Curtis Malainey <cujomalainey(a)chromium.org> Implicit values may have a length of 15bits (s16) so we need to declare the proper size so we don't get undefined behaviour. This appears to be arch and compiler dependent. This commit is to keep the headers aligned between the firmware and kernel. UBSan discovered this bug in the firmware. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/sof/header.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 2d35997ace40..571e1dd54b89 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -31,12 +31,12 @@ /* Global Message - Generic */ #define SOF_GLB_TYPE_SHIFT 28 -#define SOF_GLB_TYPE_MASK (0xf << SOF_GLB_TYPE_SHIFT) +#define SOF_GLB_TYPE_MASK (0xfL << SOF_GLB_TYPE_SHIFT) #define SOF_GLB_TYPE(x) ((x) << SOF_GLB_TYPE_SHIFT) /* Command Message - Generic */ #define SOF_CMD_TYPE_SHIFT 16 -#define SOF_CMD_TYPE_MASK (0xfff << SOF_CMD_TYPE_SHIFT) +#define SOF_CMD_TYPE_MASK (0xfffL << SOF_CMD_TYPE_SHIFT) #define SOF_CMD_TYPE(x) ((x) << SOF_CMD_TYPE_SHIFT) /* Global Message Types */ -- 2.28.0

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[PATCH 0/2] ASoC: mt6359: Fix regulator_dev_lookup() fails for id "LDO_VAUD18"
by Jiaxin Yu 20 Nov '20

20 Nov '20

This series fixed "LDO_VAUD18-supply" regulator register fail. We can see the error log "mt6359-sound supply LDO_VAUD18 not found, using dummy regulator" when register the DAPM widget "LDO_VAUD18" Otherwise, the power can not be turned on correctly when recording. Jiaxin Yu (2): ASoC: mediatek: mt6359: Fix regulator_dev_lookup() fails for id "LDO_VAUD18" dt-bindings: mediatek: mt6359: remove unused property for mt6359 Documentation/devicetree/bindings/sound/mt6359.yaml | 9 --------- sound/soc/codecs/mt6359.c | 7 ++++--- 2 files changed, 4 insertions(+), 12 deletions(-) -- 2.18.0

2 3

[PATCH] ASoC: qcom: sm8250: fix HDMI audio playback
by Dmitry Baryshkov 20 Nov '20

20 Nov '20

Current code does not setup CPU dai (causing -EIO errors on playback) and does not pass SND_SOC_DAIFMT_I2S to codec fmt (causing i2s-hifi errors). Fix both errors to enable HDMI audio playback on SM8250. Tested on RB5 platform. Signed-off-by: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> Fixes: aa2e2785545a ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/sm8250.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 35c9ad11edff..314e3e566d6d 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -36,6 +36,7 @@ static int sm8250_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, static int sm8250_snd_startup(struct snd_pcm_substream *substream) { + unsigned int fmt = SND_SOC_DAIFMT_CBS_CFS; unsigned int codec_dai_fmt = SND_SOC_DAIFMT_CBS_CFS; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); @@ -43,10 +44,11 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream) switch (cpu_dai->id) { case TERTIARY_MI2S_RX: - codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF; + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT, MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, fmt); snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); break; default: -- 2.29.2

3 2

[PATCH] ASoC: mmp-sspa: set phase two word length register
by Kyle Russell 20 Nov '20

20 Nov '20

If hw params enables dual phase transmission, then the word length for the second phase should be set to match the sample format instead of remaining at the reset default. This matches the configuration already being done for the first phase. This driver already sets the phase two sample size, so this should complete the phase two configuration. Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> --- sound/soc/pxa/mmp-sspa.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 52d4d8ace1c3..4803972ee655 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -246,6 +246,9 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; sspa_ctrl |= SSPA_CTL_XWDLEN1(bitval); + sspa_ctrl &= ~SSPA_CTL_XWDLEN2_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN2(bitval); + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; sspa_ctrl |= SSPA_CTL_XSSZ1(bitval); -- 2.25.1

2 1

[PATCH] ASoC: codecs: lpass-va-macro: add missing MODULE_DEVICE_TABLE
by Srinivas Kandagatla 20 Nov '20

20 Nov '20

Fix module loading due by adding missing MODULE_DEVICE_TABLE. Fixes: 908e6b1df26e ("ASoC: codecs: lpass-va-macro: Add support to VA Macro") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/lpass-va-macro.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b604de07e650..3e6bbef26dcb 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1487,6 +1487,7 @@ static const struct of_device_id va_macro_dt_match[] = { { .compatible = "qcom,sm8250-lpass-va-macro" }, {} }; +MODULE_DEVICE_TABLE(of, va_macro_dt_match); static struct platform_driver va_macro_driver = { .driver = { -- 2.21.0

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[PATCH] ASoC: dt-bindings: renesas, rsnd: Fix duplicate 'allOf' entries
by Rob Herring 20 Nov '20

20 Nov '20

Commit e52f3f291152 ("ASoC: audio-graph-card: Refactor schema") added an 'allOf' entry, but one is already present in the schema. Multiple keys is not valid and results in an error: ruamel.yaml.constructor.DuplicateKeyError: while constructing a mapping in "<unicode string>", line 4, column 1 found duplicate key "allOf" with value "[]" (original value: "[]") in "<unicode string>", line 262, column 1 Fixes: e52f3f291152 ("ASoC: audio-graph-card: Refactor schema") Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index cbfd5914b432..0fd37aa84947 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -9,10 +9,6 @@ title: Renesas R-Car Sound Driver Device Tree Bindings maintainers: - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> -allOf: - - $ref: audio-graph.yaml# - - $ref: audio-graph-port.yaml# - properties: compatible: @@ -260,6 +256,8 @@ required: - "#sound-dai-cells" allOf: + - $ref: audio-graph.yaml# + - $ref: audio-graph-port.yaml# - if: properties: compatible: -- 2.25.1

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[PATCH] ASoC: SOF: control: override volume info callback
by Kai Vehmanen 20 Nov '20

20 Nov '20

From: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> ASoC dapm controls currently don't support more than 2 channels. This is a problem for SOF-based devices where individual volume control cannot be provided on the 4 DMIC input path. If we want to provide controls for more than 2 channels, this patch suggests a simple solution based on an override of the info callback. For example, in the case with 4 channel DMIC PGAs, a sof_info callback would be used. Mono and stereo cases will keep using the existing dapm info callback. A longer-term solution would be to remove the limits to 2 channels in ASoC/DAPM/topology. This is a topic Intel is currently looking into, e.g. by removing the use of 'reg' and 'rreg' fields and use arrays instead. Such changes will be rather intrusive and touch multiple codec and platform drivers. Removing restrictions is the right thing to do, but this will need to be done in steps with lots of validation. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/control.c | 22 ++++++++++++++++++++++ sound/soc/sof/sof-audio.h | 2 ++ sound/soc/sof/topology.c | 9 +++++++++ 3 files changed, 33 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 056c86ad5a47..a5dd728c580a 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -114,6 +114,28 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, return change; } +int snd_sof_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *sm = (struct soc_mixer_control *)kcontrol->private_value; + struct snd_sof_control *scontrol = sm->dobj.private; + unsigned int channels = scontrol->num_channels; + int platform_max; + + if (!sm->platform_max) + sm->platform_max = sm->max; + platform_max = sm->platform_max; + + if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = channels; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - sm->min; + return 0; +} + int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9f645a2e5a6c..ddcfd46617f8 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -124,6 +124,8 @@ int snd_sof_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_sof_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *ucontrol); int snd_sof_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_sof_switch_put(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 44fddeda6043..b0132fa56fe2 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1041,6 +1041,15 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, goto out; } + /* + * If control has more than 2 channels we need to override the info. This is because even if + * ASoC layer has defined topology's max channel count to SND_SOC_TPLG_MAX_CHAN = 8, the + * pre-defined dapm control types (and related functions) creating the actual control + * restrict the channels only to mono or stereo. + */ + if (le32_to_cpu(mc->num_channels) > 2) + kc->info = snd_sof_volume_info; + /* init the volume get/put data */ scontrol->size = struct_size(scontrol->control_data, chanv, le32_to_cpu(mc->num_channels)); -- 2.28.0

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[PATCH] ASoC: SOF: nocodec: modify DAI link definitions
by Kai Vehmanen 20 Nov '20

20 Nov '20

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> The ignore_machine field in the component driver is used to ignore the FE DAI links defined in the machine driver, override BE fixups and set the stream names for the DAI links defined in the machine driver. This is required to make SOF compatible with the legacy machine drivers. In the case of the nocodec machine driver in SOF, there is no need to rely upon this ignore_machine logic in the core. Modify the machine driver to set DAI link stream names and the BE hw_params_fixup callback appropriately. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/sof.h | 6 ++++-- sound/soc/sof/nocodec.c | 18 ++++++++++++++---- sound/soc/sof/pcm.c | 3 +-- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-audio.h | 3 +++ 5 files changed, 23 insertions(+), 9 deletions(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index 9aa055289dcc..646a655c3c6b 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -100,6 +100,8 @@ struct sof_dev_desc { const struct snd_sof_dsp_ops *ops; }; -int sof_nocodec_setup(struct device *dev, - const struct snd_sof_dsp_ops *ops); +int sof_nocodec_setup(struct device *dev, const struct snd_sof_dsp_ops *ops, + int (*pcm_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params)); + #endif diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index 9e922df6a710..3b9bb2e83a86 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -10,17 +10,21 @@ #include <linux/module.h> #include <sound/sof.h> +#include "sof-audio.h" #include "sof-priv.h" static struct snd_soc_card sof_nocodec_card = { .name = "nocodec", /* the sof- prefix is added by the core */ + .topology_shortname = "sof-nocodec", .owner = THIS_MODULE }; static int sof_nocodec_bes_setup(struct device *dev, const struct snd_sof_dsp_ops *ops, struct snd_soc_dai_link *links, - int link_num, struct snd_soc_card *card) + int link_num, struct snd_soc_card *card, + int (*pcm_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params)) { struct snd_soc_dai_link_component *dlc; int i; @@ -39,6 +43,8 @@ static int sof_nocodec_bes_setup(struct device *dev, if (!links[i].name) return -ENOMEM; + links[i].stream_name = links[i].name; + links[i].cpus = &dlc[0]; links[i].codecs = &dlc[1]; links[i].platforms = &dlc[2]; @@ -57,6 +63,8 @@ static int sof_nocodec_bes_setup(struct device *dev, links[i].dpcm_playback = 1; if (ops->drv[i].capture.channels_min) links[i].dpcm_capture = 1; + + links[i].be_hw_params_fixup = pcm_dai_link_fixup; } card->dai_link = links; @@ -65,8 +73,9 @@ static int sof_nocodec_bes_setup(struct device *dev, return 0; } -int sof_nocodec_setup(struct device *dev, - const struct snd_sof_dsp_ops *ops) +int sof_nocodec_setup(struct device *dev, const struct snd_sof_dsp_ops *ops, + int (*pcm_dai_link_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params)) { struct snd_soc_dai_link *links; @@ -77,7 +86,7 @@ int sof_nocodec_setup(struct device *dev, return -ENOMEM; return sof_nocodec_bes_setup(dev, ops, links, ops->num_drv, - &sof_nocodec_card); + &sof_nocodec_card, pcm_dai_link_fixup); } EXPORT_SYMBOL(sof_nocodec_setup); @@ -86,6 +95,7 @@ static int sof_nocodec_probe(struct platform_device *pdev) struct snd_soc_card *card = &sof_nocodec_card; card->dev = &pdev->dev; + card->topology_shortname_created = true; return devm_snd_soc_register_card(&pdev->dev, card); } diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 37d12162e448..0dc39fbcd81d 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -620,8 +620,7 @@ static int sof_pcm_new(struct snd_soc_component *component, } /* fixup the BE DAI link to match any values from topology */ -static int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) +int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 9a8ce25a8fc8..3277489fee5e 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -468,7 +468,7 @@ int sof_machine_check(struct snd_sof_dev *sdev) mach->drv_name = "sof-nocodec"; sof_pdata->tplg_filename = desc->nocodec_tplg_filename; - ret = sof_nocodec_setup(sdev->dev, desc->ops); + ret = sof_nocodec_setup(sdev->dev, desc->ops, sof_pcm_dai_link_fixup); if (ret < 0) return ret; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 9f645a2e5a6c..5b016f00956b 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -212,6 +212,9 @@ int snd_sof_ipc_set_get_comp_data(struct snd_sof_control *scontrol, enum sof_ipc_ctrl_cmd ctrl_cmd, bool send); +/* DAI link fixup */ +int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); + /* PM */ int sof_restore_pipelines(struct device *dev); int sof_set_hw_params_upon_resume(struct device *dev); -- 2.28.0

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ucm2: add support to for Qualcomm RB5 Platform
by GitHub pull_request - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-ucm-conf pull request #63 was opened from Srinivas-Kandagatla: This patchset adds support to The Qualcomm Robotics RB5 Platform Audio ports. RB5 has 2 WSA881X smart speakers attached via Soundwire, HDMI and a DMIC0 on the board. All the audio related patches are merged in ASoC tree. Also to note that the driver name and long card names are fixed properly on all new and old machine drivers so that we can organise the ucm2 directories correctly. That is the reason for moving sm8250 out of Qualcomm folder. Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/63 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/63.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Sm8250 v1
by GitHub pull_request - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-ucm-conf pull request #62 was opened from Srinivas-Kandagatla: This patchset adds support to The Qualcomm Robotics RB5 Platform Audio ports. RB5 has 2 WSA881X smart speakers attached via Soundwire, HDMI and a DMIC0 on the board. All the audio related patches are merged in ASoC tree. Also to note that the driver name and long card names are fixed properly on all new and old machine drivers so that we can organise the ucm2 directories correctly. That is the reason for moving sm8250 out of Qualcomm folder. Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/62 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/62.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Dell AIO wrong sound tone
by Kailang 20 Nov '20

20 Nov '20

Hi Takashi, Dell AIO only had one audio jack. It will get wrong sound tone with replug speaker or HP or headset. This patch will solve it. Many thanks. BR, Kailang

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[RFC PATCH] ASoC: ak4458: use reset control instead of reset gpio
by Viorel Suman (OSS) 20 Nov '20

20 Nov '20

From: Viorel Suman <viorel.suman(a)nxp.com> Using GPIO API seems not a way to go for the case when the same reset GPIO is used to control several codecs. For a such case we can use the "gpio-reset" driver and the "shared" reset API to manage the reset GPIO - to deassert the reset when the first codec is powered up and assert it when there is no codec in use. DTS is supposed to look as follows: / { ak4458_reset: gpio-reset { compatible = "gpio-reset"; reset-gpios = <&pca6416 4 GPIO_ACTIVE_LOW>; #reset-cells = <0>; initially-in-reset; }; }; &i2c3 { pca6416: gpio@20 { compatible = "ti,tca6416"; reg = <0x20>; gpio-controller; #gpio-cells = <2>; }; ak4458_1: ak4458@10 { compatible = "asahi-kasei,ak4458"; reg = <0x10>; resets = <&ak4458_reset>; }; ak4458_2: ak4458@11 { compatible = "asahi-kasei,ak4458"; reg = <0x11>; resets = <&ak4458_reset>; }; ak4458_3: ak4458@12 { compatible = "asahi-kasei,ak4458"; reg = <0x12>; resets = <&ak4458_reset>; }; }; Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/codecs/ak4458.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 1010c9ee2e83..f27727cb1382 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -13,6 +13,7 @@ #include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/reset.h> #include <linux/slab.h> #include <sound/initval.h> #include <sound/pcm_params.h> @@ -45,7 +46,7 @@ struct ak4458_priv { const struct ak4458_drvdata *drvdata; struct device *dev; struct regmap *regmap; - struct gpio_desc *reset_gpiod; + struct reset_control *reset; struct gpio_desc *mute_gpiod; int digfil; /* SSLOW, SD, SLOW bits */ int fs; /* sampling rate */ @@ -597,17 +598,17 @@ static struct snd_soc_dai_driver ak4497_dai = { static void ak4458_power_off(struct ak4458_priv *ak4458) { - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 0); - usleep_range(1000, 2000); + if (ak4458->reset) { + reset_control_assert(ak4458->reset); + msleep(20); } } static void ak4458_power_on(struct ak4458_priv *ak4458) { - if (ak4458->reset_gpiod) { - gpiod_set_value_cansleep(ak4458->reset_gpiod, 1); - usleep_range(1000, 2000); + if (ak4458->reset) { + reset_control_deassert(ak4458->reset); + msleep(20); } } @@ -685,7 +686,6 @@ static int __maybe_unused ak4458_runtime_resume(struct device *dev) if (ak4458->mute_gpiod) gpiod_set_value_cansleep(ak4458->mute_gpiod, 1); - ak4458_power_off(ak4458); ak4458_power_on(ak4458); regcache_cache_only(ak4458->regmap, false); @@ -771,10 +771,9 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) ak4458->drvdata = of_device_get_match_data(&i2c->dev); - ak4458->reset_gpiod = devm_gpiod_get_optional(ak4458->dev, "reset", - GPIOD_OUT_LOW); - if (IS_ERR(ak4458->reset_gpiod)) - return PTR_ERR(ak4458->reset_gpiod); + ak4458->reset = devm_reset_control_get_optional_shared(ak4458->dev, NULL); + if (IS_ERR(ak4458->reset)) + return PTR_ERR(ak4458->reset); ak4458->mute_gpiod = devm_gpiod_get_optional(ak4458->dev, "mute", GPIOD_OUT_LOW); -- 2.26.2

4 8

Lenovo t14s amd only Dummy Device as output in pulseaudio
by GitHub issues - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-ucm-conf issue #61 was opened from J-Bu: Hi, since upgrading to alsa-ucm 1.2.4 I only have a "Dummy Device" instead of the internal speaker. When using alsa-ucm 1.2.3 all is fine. I'm not quite sure if this is a pulseaudio or alsa issue maybe you can guide me in the right direction Output of lspci: ``` 06:00.1 Audio device: Advanced Micro Devices, Inc. [AMD/ATI] Device 1637 Subsystem: Lenovo Device 5082 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 32 bytes Interrupt: pin B routed to IRQ 85 Region 0: Memory at fd3c8000 (32-bit, non-prefetchable) [size=16K] Capabilities: [48] Vendor Specific Information: Len=08 <?> Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1+,D2+,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable+ DSel=0 DScale=0 PME+ Capabilities: [64] Express (v2) Legacy Endpoint, MSI 00 DevCap: MaxPayload 256 bytes, PhantFunc 0, Latency L0s <4us, L1 unlimited ExtTag+ AttnBtn- AttnInd- PwrInd- RBE+ FLReset- DevCtl: CorrErr- NonFatalErr- FatalErr- UnsupReq- RlxdOrd+ ExtTag+ PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 512 bytes DevSta: CorrErr- NonFatalErr- FatalErr- UnsupReq- AuxPwr- TransPend- LnkCap: Port #0, Speed 16GT/s, Width x16, ASPM L0s L1, Exit Latency L0s <64ns, L1 <1us ClockPM- Surprise- LLActRep- BwNot- ASPMOptComp+ LnkCtl: ASPM Disabled; RCB 64 bytes, Disabled- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 16GT/s (ok), Width x16 (ok) TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- DevCap2: Completion Timeout: Range ABCD, TimeoutDis+ NROPrPrP- LTR- 10BitTagComp+ 10BitTagReq- OBFF Not Supported, ExtFmt+ EETLPPrefix+, MaxEETLPPrefixes 1 EmergencyPowerReduction Not Supported, EmergencyPowerReductionInit- FRS- AtomicOpsCap: 32bit- 64bit- 128bitCAS- DevCtl2: Completion Timeout: 50us to 50ms, TimeoutDis- LTR- OBFF Disabled, AtomicOpsCtl: ReqEn- LnkSta2: Current De-emphasis Level: -3.5dB, EqualizationComplete- EqualizationPhase1- EqualizationPhase2- EqualizationPhase3- LinkEqualizationRequest- Retimer- 2Retimers- CrosslinkRes: unsupported Capabilities: [a0] MSI: Enable+ Count=1/1 Maskable- 64bit+ Address: 00000000fee09000 Data: 4024 Capabilities: [100 v1] Vendor Specific Information: ID=0001 Rev=1 Len=010 <?> Kernel driver in use: snd_hda_intel ``` I bisected the issue: ``` git bisect start # good: [dbac4c3e7ad05b4eb17964688fd9f07c61b8d336] ucm2: fix more indentation issues git bisect good dbac4c3e7ad05b4eb17964688fd9f07c61b8d336 # bad: [334b12e9a796cddf70779b68418ee7ad52ba1d14] sof-soundwire: use the codecs/hda/hdmi.conf macro git bisect bad 334b12e9a796cddf70779b68418ee7ad52ba1d14 # bad: [8a9609cc694cb1de501a67d7c091edaa9a338932] cht-bsw-rt5672: Add Lenovo ThinkPad 10 specific configuration git bisect bad 8a9609cc694cb1de501a67d7c091edaa9a338932 # good: [9f2dae6a560c60593c24aa4b85aed07418f9dd8e] sof-soundwire: initial UCM2 version git bisect good 9f2dae6a560c60593c24aa4b85aed07418f9dd8e # bad: [73c105b4707b601de63d1f24641e65e5ac23f3a6] DB820c/DB845c: move to Qualcomm/ tree git bisect bad 73c105b4707b601de63d1f24641e65e5ac23f3a6 # good: [38e5906cd1b18b6f4df5a1d35ca535325e7b65a0] sof-hda-dsp: fix the device order (Hdmi devices) git bisect good 38e5906cd1b18b6f4df5a1d35ca535325e7b65a0 # bad: [6b8518c431cac18a825bd7342875f54b44cb7afc] DAISY-I2S: move to Samsung/snow/snow.conf git bisect bad 6b8518c431cac18a825bd7342875f54b44cb7afc # bad: [dcef48f13d4f5db79b006755074940b94730a883] HDA-Intel: add support for AMD acp microphone devices git bisect bad dcef48f13d4f5db79b006755074940b94730a883 # first bad commit: [dcef48f13d4f5db79b006755074940b94730a883] HDA-Intel: add support for AMD acp microphone devices ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/61 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/4] alsa-gobject: bump project version up to v0.2.0 for new release
by GitHub pull_request - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-gobject pull request #58 was opened from takaswie: This patchset is a preparation for new release, including slight changes. ``` Takashi Sakamoto (4): hwdep: fix metadata and documentation for the type of interface upgrade CI operationg system to Ubuntu 20.04 LTS amd64 upgrade CI operationg system to Fedora 33 bump version up to 0.2.0 to release .github/workflows/build.yml | 4 ++-- README.rst | 7 ++++--- meson.build | 2 +- src/hwdep/alsahwdep-enum-types.h | 22 +++++++++++++--------- 4 files changed, 20 insertions(+), 15 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/58 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/58.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/3] alsa-gobject: buildable in kernel 4.5 or later
by GitHub pull_request - opened 20 Nov '20

20 Nov '20

alsa-project/alsa-gobject pull request #57 was opened from takaswie: Some users would like to build alsa-gobject in former versions of Linux kernel. https://sourceforge.net/p/alsa/mailman/alsa-user/thread/d95fce3f-5244-0e5f-… This patchset use some macro from Linux kernel for conditional build ``` Takashi Sakamoto (3): seq: client_info: conditional build for Linux kernel v4.6 or later hwdep: conditional build for Linux kernel v4.9 or later hwdep: conditional build for Linux kernel 4.12 or later src/hwdep/alsahwdep-enum-types.h | 16 +++++++++++++--- src/seq/client-info.c | 9 +++++++-- 2 files changed, 20 insertions(+), 5 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/57 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/57.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[GIT PULL] ASoC fixes for v5.10-rc4
by Mark Brown 19 Nov '20

19 Nov '20

The following changes since commit f9d7c6eb23f7e55e7a0ca5451da06909bdfdd0e4: ASoC: mchp-spdiftx: Do not set Validity bit(s) (2020-11-04 17:50:22 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.10-rc4 for you to fetch changes up to 879ee8b6f2bae0cc4a25536f8841db1dbc969523: ASOC: Intel: kbl_rt5663_rt5514_max98927: Do not try to disable disabled clock (2020-11-18 17:59:23 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.11 A collection of driver specific fixes, mostly for x86 systems (or CODECs used mostly on x86) and all for relatively minor issues, the biggest one being fixing S24_LE format on Keem Bay systems. ---------------------------------------------------------------- Cezary Rojewski (2): ASoC: Intel: catpt: Skip position update for unprepared streams ASoC: Intel: catpt: Correct clock selection for dai trigger Guenter Roeck (1): ASOC: Intel: kbl_rt5663_rt5514_max98927: Do not try to disable disabled clock Jack Yu (1): ASoC: rt1015: add delay to fix pop noise from speaker Mark Brown (1): Merge Intel catpt DSP fixes into asoc-5.10 Michael Sit Wei Hong (1): ASoC: Intel: KMB: Fix S24_LE configuration Shuming Fan (1): ASoC: rt1015: increase the time to detect BCLK Srinivasa Rao Mandadapu (1): ASoC: qcom: lpass-platform: Fix memory leak Documentation/devicetree/bindings/sound/rt1015.txt | 6 ++++++ include/sound/rt1015.h | 15 +++++++++++++++ sound/soc/codecs/rt1015.c | 22 +++++++++++++++++++++- sound/soc/codecs/rt1015.h | 2 ++ .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 ++ sound/soc/intel/catpt/pcm.c | 9 ++++----- sound/soc/intel/keembay/kmb_platform.c | 6 +++--- sound/soc/qcom/lpass-platform.c | 5 ++++- 8 files changed, 57 insertions(+), 10 deletions(-) create mode 100644 include/sound/rt1015.h

2 1

[PATCH v1] ona10iv: add ona10iv smart PA kernel driver
by Jamie Meacham 19 Nov '20

19 Nov '20

From: Jamie Meacham <jamie.meacham(a)onsemi.com> add ona10iv smart PA kernel driver Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Jamie Meacham <jamie.meacham(a)onsemi.com> --- sound/soc/codecs/Kconfig | 8 sound/soc/codecs/Makefile | 2 sound/soc/codecs/ona10iv.c | 699 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/ona10iv.h | 632 +++++++++++++++++++++++++++++++ 4 files changed, 1341 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2b6eceb35..9f0b594e8 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -137,6 +137,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_NAU8824 imply SND_SOC_NAU8825 imply SND_SOC_HDMI_CODEC + imply SND_SOC_ONA10IV imply SND_SOC_PCM1681 imply SND_SOC_PCM1789_I2C imply SND_SOC_PCM179X_I2C @@ -921,6 +922,13 @@ config SND_SOC_MSM8916_WCD_DIGITAL tristate "Qualcomm MSM8916 WCD DIGITAL Codec" select REGMAP_MMIO +config SND_SOC_ONA10IV + tristate "ON Semiconductor ONA10IV Speaker Amplifier" + depends on I2C + help + Adds support for the ON Semiconductor ONA10IV + speaker amplifier with I/V feedback + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 144d9256d..061c2479a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -136,6 +136,7 @@ snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o +snd-soc-ona10iv-objs := ona10iv.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1789-codec-objs := pcm1789.o snd-soc-pcm1789-i2c-objs := pcm1789-i2c.o @@ -446,6 +447,7 @@ obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_ONA10IV) += snd-soc-ona10iv.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o obj-$(CONFIG_SND_SOC_PCM1789_I2C) += snd-soc-pcm1789-i2c.o diff --git a/sound/soc/codecs/ona10iv.c b/sound/soc/codecs/ona10iv.c new file mode 100644 index 000000000..53db45adc --- /dev/null +++ b/sound/soc/codecs/ona10iv.c @@ -0,0 +1,699 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC ON Semiconductor ONA10IV 16W Digital Input Mono Class-D + * Audio Amplifier with Speaker I/V Sense + * + * Copyright (C) 2020 ON Semiconductor - https://www.onsemi.com/ + * Author: Jamie Meacham <jamie.meacham(a)onsemi.com> + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/err.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "ona10iv.h" + +/////////////////////////////////////////////////////// +// Local function prototypes +/////////////////////////////////////////////////////// +static void param_errcheck(int retval, struct device *dev, + const char *errmsg, int param_val); + +/////////////////////////////////////////////////////// +// Local structure definitions +/////////////////////////////////////////////////////// +struct ona10iv_priv { + struct regmap *regmap; + struct snd_soc_component *component; + int volume; +}; + +/////////////////////////////////////////////////////// +// Function implementations +/////////////////////////////////////////////////////// +static void ona10iv_reset(struct ona10iv_priv *ona10iv) +{ + snd_soc_component_write(ona10iv->component, ONA10IV_REG_PWR_CTRL, + ONA10IV_RST); +} + +static int ona10iv_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + int from_state; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + dev_dbg(component->dev, "ONA10IV-->active\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY_RELEASE); + break; + + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + dev_dbg(component->dev, "ONA10IV-->standby\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); + break; + + case SND_SOC_BIAS_OFF: + dev_dbg(component->dev, "ONA10IV-->soft shut-down\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_SHUTDOWN | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); + + break; + default: + return -EINVAL; + } + + if (ret < 0) + return ret; + + return 0; +} + +#ifdef CONFIG_PM +static int ona10iv_codec_suspend(struct snd_soc_component *component) +{ + int from_state; + int ret; + + dev_dbg(component->dev, "ONA10IV-->suspend (standby)\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); // pause for volume ramp to complete + + if (ret < 0) + return ret; + + return 0; +} + +static int ona10iv_codec_resume(struct snd_soc_component *component) +{ + int from_state; + int ret; + + dev_dbg(component->dev, "ONA10IV-->resume (active)\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); // pause for volume ramp to complete + + if (ret < 0) + return ret; + + return 0; +} +#else +#define ona10iv_codec_suspend NULL +#define ona10iv_codec_resume NULL +#endif + +static int ona10iv_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_dbg(component->dev, "ONA10IV-->post power-up\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY_RELEASE); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_dbg(component->dev, "ONA10IV-->pre power-down\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + msleep(255); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, "ONA10IV-->post power-down\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_SHUTDOWN | ONA10IV_STBY_RELEASE); + break; + default: + dev_err(component->dev, "Not supported evevt\n"); + ret = -EINVAL; + } + + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", ONA10IV_REG_DATAO_CTRL2, 7, 1, 0); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", ONA10IV_REG_DATAO_CTRL2, 6, 1, 0); + +static const struct snd_soc_dapm_widget ona10iv_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ona10iv DAI", "DAI Playback", + 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("ISENSE", ONA10IV_REG_DATAO_CTRL2, + 7, 0, &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", ONA10IV_REG_DATAO_CTRL2, + 7, 0, &vsense_switch), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, + 0, 0, ona10iv_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON"), + SND_SOC_DAPM_ADC("VMON", "DAI Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("IMON", "DAI Capture", SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route ona10iv_audio_map[] = { + {"DAC", NULL, "ona10iv DAI"}, + {"OUT", NULL, "DAC"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static int ona10iv_mute(struct snd_soc_dai *dai, int mute, int dir) +{ + int ret; + struct snd_soc_component *component = dai->component; + struct ona10iv_priv *ona10iv; + + ona10iv = snd_soc_component_get_drvdata(component); + + /* using "mute" bit can cause a pop. Instead store volume setting + * and set volume to minimum allowing a smooth ramp-down. + */ + + dev_dbg(component->dev, "mute dir: %d val: %d\n", dir, mute); + + if (dir != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + if (mute) { + ret = regmap_read(ona10iv->regmap, ONA10IV_REG_MAX_VOL, + &ona10iv->volume); + if (ret < 0) { + dev_err(component->dev, "Failed to read: %d\n", ret); + return ret; + } + ret = regmap_write(ona10iv->regmap, ONA10IV_REG_MAX_VOL, + ONA10IV_VOL_AMP_MUTE); + } else { + ret = regmap_write(ona10iv->regmap, ONA10IV_REG_MAX_VOL, + ona10iv->volume); + } + + return ret; +} + +static int ona10iv_set_bitwidth(struct ona10iv_priv *ona10iv, int format) +{ + struct snd_soc_component *component = ona10iv->component; + int slot_width, sample_width; + int ret = 0; + + dev_dbg(component->dev, + "ONA10IV-->set bitwidth for format: %d\n", format); + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + sample_width = ONA10IV_SAMPW_16; + slot_width = ONA10IV_SLOTW_16; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + sample_width = ONA10IV_SAMPW_24; + slot_width = ONA10IV_SLOTW_24; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + sample_width = ONA10IV_SAMPW_32; + slot_width = ONA10IV_SLOTW_32; + break; + default: + ret = -EINVAL; + } + param_errcheck(ret, component->dev, + "error no case match for format", format); + + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL2, ONA10IV_SAMPW_MASK, + sample_width); + param_errcheck(ret, component->dev, + "error writing sample width", sample_width); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL2, ONA10IV_SLOTW_MASK, + slot_width); + param_errcheck(ret, component->dev, + "error writing slot width", slot_width); + if (ret < 0) + return ret; + + return 0; +} + +static int ona10iv_set_samplerate(struct ona10iv_priv *ona10iv, + int samplerate) +{ + struct snd_soc_component *component = ona10iv->component; + int rate_val; + int ret = 0; + + dev_dbg(component->dev, "ONA10IV-->samplerate: %d\n", samplerate); + + switch (samplerate) { + case 16000: + rate_val = ONA10IV_FS_16KHZ; + break; + case 22050: + rate_val = ONA10IV_FS_22KHZ; + break; + case 32000: + rate_val = ONA10IV_FS_32KHZ; + break; + case 44100: + rate_val = ONA10IV_FS_44KHZ; + break; + case 48000: + rate_val = ONA10IV_FS_48KHZ; + break; + case 96000: + rate_val = ONA10IV_FS_96KHZ; + break; + default: + ret = -EINVAL; + } + param_errcheck(ret, component->dev, + "error no case match for rate", samplerate); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL1, ONA10IV_FS_MASK, + rate_val); + param_errcheck(ret, component->dev, + "error writing rate", rate_val); + if (ret < 0) + return ret; + + return 0; +} + +static int ona10iv_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct ona10iv_priv *ona10iv = + snd_soc_component_get_drvdata(component); + int ret; + + dev_dbg(component->dev, "ONA10IV-->setting hw params\n"); + + ret = ona10iv_set_bitwidth(ona10iv, params_format(params)); + param_errcheck(ret, component->dev, + "error setting bitwidth for format", + params_format(params)); + if (ret) + return ret; + + ret = ona10iv_set_samplerate(ona10iv, params_rate(params)); + param_errcheck(ret, component->dev, + "error setting rate", params_rate(params)); + if (ret) + return ret; + + return 0; +} + +static int ona10iv_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + int left_slot, right_slot; + int slot_val; + int i_enable, v_enable; + int i_slot, v_slot; + int reg_val; + int ret; + + dev_dbg(component->dev, + "set dai txm: %x rxm %x slots: %d width: %d\n", + tx_mask, rx_mask, slots, slot_width); + + if (tx_mask == 0 || slots != 2) + return -EINVAL; + + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + + + // right slot checked but not used. + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (left_slot) { + case 1: + slot_val = ONA10IV_A_SLOT_1; + break; + case 2: + slot_val = ONA10IV_A_SLOT_2; + break; + } + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL3, ONA10IV_A_SLOT_MSK, + slot_val); + param_errcheck(ret, component->dev, + "error setting transmit slot", slot_val); + if (ret < 0) + return ret; + + i_slot = 0; + v_slot = 0; + if (rx_mask == 0) { + i_enable = 0; + v_enable = 0; + } else { + i_enable = 1; + i_slot = __ffs(rx_mask); + rx_mask &= ~(1 << i_slot); + if (rx_mask != 0) { + v_enable = 1; + v_slot = __ffs(rx_mask); + rx_mask &= ~(1 << v_slot); + } + } + + if (rx_mask != 0 || i_slot >= slots || v_slot >= slots) + return -EINVAL; + + if (i_enable) { + reg_val = ONA10IV_I_DATAO_TX_ENABLE; + switch (i_slot) { + case 1: + reg_val |= ONA10IV_I_SLOT_1; + break; + case 2: + reg_val |= ONA10IV_I_SLOT_2; + break; + } + ret = snd_soc_component_write(component, + ONA10IV_REG_DATAO_CTRL2, reg_val); + param_errcheck(ret, component->dev, + "error setting current sense slot and enable", + reg_val); + if (ret < 0) + return ret; + } + + if (v_enable) { + reg_val = ONA10IV_V_DATAO_TX_ENABLE; + switch (v_slot) { + case 1: + reg_val |= ONA10IV_V_SLOT_1; + break; + case 2: + reg_val |= ONA10IV_V_SLOT_2; + break; + } + ret = snd_soc_component_write(component, + ONA10IV_REG_DATAO_CTRL2, reg_val); + param_errcheck(ret, component->dev, + "error setting voltage sense slot and enable", + reg_val); + if (ret < 0) + return ret; + } + + return 0; +} + +static void param_errcheck(int retval, struct device *dev, + const char *errmsg, int param_val) +{ + if (retval < 0) + dev_dbg(dev, "Error %d: %s = %d\n", retval, errmsg, param_val); +} + +static struct snd_soc_dai_ops ona10iv_dai_ops = { + .mute_stream = ona10iv_mute, + .hw_params = ona10iv_hw_params, + .set_fmt = NULL, + .set_tdm_slot = ona10iv_set_dai_tdm_slot, + .startup = NULL, + .shutdown = NULL, + .prepare = NULL, +}; + +#define ONA10IV_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +#define ONA10IV_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_driver ona10iv_dai[] = { + { + .name = "ona10iv DAI", + .id = 0, + .playback = { + .stream_name = "DAI Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ONA10IV_RATES, + .formats = ONA10IV_FORMATS, + }, + .capture = { + .stream_name = "DAI Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ONA10IV_RATES, + .formats = ONA10IV_FORMATS, + }, + .ops = &ona10iv_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int ona10iv_codec_probe(struct snd_soc_component *component) +{ + struct ona10iv_priv *ona10iv = + snd_soc_component_get_drvdata(component); + + ona10iv->component = component; + + dev_dbg(component->dev, "ONA10IV-->device probe\n"); + + ona10iv_reset(ona10iv); + + return 0; +} + +static DECLARE_TLV_DB_MINMAX(ona10iv_digital_tlv, 0, 1600); +static DECLARE_TLV_DB_MINMAX(ona10iv_playback_volume, -9562, 0); + +static const struct snd_kcontrol_new ona10iv_snd_controls[] = { + SOC_SINGLE_TLV("Playback Volume", ONA10IV_REG_MAX_VOL, 0, 255, 1, + ona10iv_playback_volume), + SOC_SINGLE_TLV("Amp Gain", ONA10IV_REG_GAIN_CTRL1, 0, 31, 1, + ona10iv_digital_tlv), +}; + +static const struct snd_soc_component_driver ona10iv_component_driver = { + .probe = ona10iv_codec_probe, + .suspend = ona10iv_codec_suspend, + .resume = ona10iv_codec_resume, + .set_bias_level = ona10iv_set_bias_level, + .controls = ona10iv_snd_controls, + .num_controls = ARRAY_SIZE(ona10iv_snd_controls), + .dapm_widgets = ona10iv_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ona10iv_dapm_widgets), + .dapm_routes = ona10iv_audio_map, + .num_dapm_routes = ARRAY_SIZE(ona10iv_audio_map), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .use_pmdown_time = 1, +}; + +static const struct reg_default ona10iv_reg_defaults[] = { + { ONA10IV_REG_PWR_CTRL, 0x00 }, + { ONA10IV_REG_INT_MASK, 0x00 }, + { ONA10IV_REG_ERR_CTRL, 0x00 }, + { ONA10IV_REG_DAI_CTRL1, 0x06 }, + { ONA10IV_REG_DAI_CTRL2, 0x55 }, + { ONA10IV_REG_DAI_CTRL3, 0x00 }, + { ONA10IV_REG_MAX_VOL, 0x00 }, + { ONA10IV_REG_VOL_CTRL, 0x0E }, + { ONA10IV_REG_GAIN_CTRL1, 0x00 }, + { ONA10IV_REG_EMI_CTRL, 0x1C }, + { ONA10IV_REG_AGC_BATT, 0x00 }, + { ONA10IV_REG_AGC_CTRL1, 0x00 }, + { ONA10IV_REG_AGC_CTRL2, 0x00 }, + { ONA10IV_REG_AGC_CTRL3, 0x00 }, + { ONA10IV_REG_MAGC_CTRL1, 0x00 }, + { ONA10IV_REG_MAGC_CTRL2, 0x00 }, + { ONA10IV_REG_MAGC_CTRL3, 0x00 }, + { ONA10IV_REG_SENSE_CTRL, 0x00 }, + { ONA10IV_REG_DATAO_CTRL1, 0x00 }, + { ONA10IV_REG_DATAO_CTRL2, 0x08 }, + { ONA10IV_REG_DATAO_CTRL3, 0x02 }, +}; + +static bool ona10iv_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ONA10IV_REG_PWR_CTRL: /* reset bit is self clearing */ + case ONA10IV_REG_INT_FLAG: + case ONA10IV_REG_ERR_STAT: + case ONA10IV_REG_T_SENSE_OUT1: + case ONA10IV_REG_T_SENSE_OUT2:/* Sticky interrupt flags */ + return true; + } + + return false; +} + +static bool ona10iv_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ONA10IV_REG_DEVICE_ID: + case ONA10IV_REG_ERR_STAT: + case ONA10IV_REG_T_SENSE_OUT1: + case ONA10IV_REG_T_SENSE_OUT2: + return false; + } + + return true; +} + +static const struct regmap_config ona10iv_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = ona10iv_writeable, + .volatile_reg = ona10iv_volatile, + .reg_defaults = ona10iv_reg_defaults, + .max_register = ONA10IV_REG_DATAO_CTRL3, + .num_reg_defaults = ARRAY_SIZE(ona10iv_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static int ona10iv_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ona10iv_priv *ona10iv; + int ret, reg; + + ona10iv = devm_kzalloc(&i2c->dev, sizeof(*ona10iv), GFP_KERNEL); + if (ona10iv == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ona10iv); + + ona10iv->regmap = devm_regmap_init_i2c(i2c, &ona10iv_i2c_regmap); + if (IS_ERR(ona10iv->regmap)) { + ret = PTR_ERR(ona10iv->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + ret = regmap_read(ona10iv->regmap, ONA10IV_REG_DEVICE_ID, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: %d\n", ret); + return ret; + } + dev_dbg(&i2c->dev, "device revision: %x\n", reg); + + ona10iv->volume = 0; + + ret = devm_snd_soc_register_component(&i2c->dev, + &ona10iv_component_driver, + ona10iv_dai, ARRAY_SIZE(ona10iv_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register component: %d\n", ret); + return ret; +} + +static int ona10iv_i2c_remove(struct i2c_client *client) +{ + return 0; +} + +static const struct i2c_device_id ona10iv_i2c_id[] = { + { "ona10iv", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ona10iv_i2c_id); + +static const struct of_device_id ona10iv_of_match[] = { + { .compatible = "onnn,ona10iv" }, + {}, +}; +MODULE_DEVICE_TABLE(of, ona10iv_of_match); + +static struct i2c_driver ona10iv_i2c_driver = { + .driver = { + .name = "ona10iv", + .of_match_table = of_match_ptr(ona10iv_of_match), + }, + .probe = ona10iv_i2c_probe, + .remove = ona10iv_i2c_remove, + .id_table = ona10iv_i2c_id, +}; +module_i2c_driver(ona10iv_i2c_driver); + +MODULE_AUTHOR("Jamie Meacham <jamie.meacham(a)onsemi.com>"); +MODULE_DESCRIPTION("ONA10IV I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ona10iv.h b/sound/soc/codecs/ona10iv.h new file mode 100644 index 000000000..2c46088ac --- /dev/null +++ b/sound/soc/codecs/ona10iv.h @@ -0,0 +1,632 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC ONA10IV codec driver + * + * Copyright (C) 2020 ON Semiconductor - https://www.onsemi.com/ + */ +#ifndef __ONA10IV__ +#define __ONA10IV__ + + +/** ONA10IV Register Map **/ +#define ONA10IV_REG_DEVICE_ID (0x00) + +#define ONA10IV_REG_PWR_CTRL (0x01) +#define ONA10IV_TFB_PD (0x01<<7) +#define ONA10IV_BATT_CFG (0x01<<6) +#define ONA10IV_AGC_PD (0x01<<5) +#define ONA10IV_IVSNS_PD (0x01<<4) +#define ONA10IV_AMP_PD (0x01<<3) +#define ONA10IV_SD_N_SHUTDOWN (0x00<<2) +#define ONA10IV_SD_N_NORMAL (0x01<<2) +#define ONA10IV_STBY_RELEASE (0x00<<1) +#define ONA10IV_STBY (0x01<<1) +#define ONA10IV_PWR_STATE_MASK (0x03<<1) +#define ONA10IV_RST_RELEASE (0x00<<0) +#define ONA10IV_RST (0x01<<0) + +#define ONA10IV_REG_INT_FLAG (0x02) +#define ONA10IV_TFB_I (0x01<<5) +#define ONA10IV_AGC_I (0x01<<4) +#define ONA10IV_CERR_I (0x01<<3) +#define ONA10IV_TERR_I (0x01<<2) +#define ONA10IV_IERR_I (0x01<<1) +#define ONA10IV_VERR_I (0x01<<0) + +#define ONA10IV_REG_INT_MASK (0x03) +#define ONA10IV_ALL_MASK (0x01<<6) +#define ONA10IV_TFB_M (0x01<<5) +#define ONA10IV_AGC_M (0x01<<4) +#define ONA10IV_CERR_M (0x01<<3) +#define ONA10IV_TERR_M (0x01<<2) +#define ONA10IV_IERR_M (0x01<<1) +#define ONA10IV_VERR_M (0x01<<0) + +#define ONA10IV_REG_ERR_STAT (0x04) +#define ONA10IV_TFB (0x01<<5) +#define ONA10IV_AGC (0x01<<4) +#define ONA10IV_CERR (0x01<<3) +#define ONA10IV_TERR (0x01<<2) +#define ONA10IV_IERR (0x01<<1) +#define ONA10IV_VERR (0x01<<0) + +#define ONA10IV_REG_ERR_CTRL (0x05) +#define ONA10IV_MRCV (0x01<<5) +#define ONA10IV_MAX_ARCV_ALWAYS (0x00<<3) +#define ONA10IV_MAX_ARCV_1_ATT (0x01<<3) +#define ONA10IV_MAX_ARCV_3_ATT (0x02<<3) +#define ONA10IV_MAX_ARCV_7_ATT (0x03<<3) +#define ONA10IV_ARCV_T (0x01<<2) +#define ONA10IV_ARCV_I (0x01<<1) +#define ONA10IV_ARCV_C (0x01<<0) + +#define ONA10IV_REG_DAI_CTRL1 (0x06) +#define ONA10IV_BEDGE_DAI_FALLING (0x00<<7) +#define ONA10IV_BEDGE_DAI_RISING (0x01<<7) +#define ONA10IV_FORMAT_I2S_DELAY (0x00<<6) +#define ONA10IV_FORMAT_NO_DELAY (0x01<<6) +#define ONA10IV_DAI_I2S (0x00<<4) +#define ONA10IV_DAI_TDM2 (0x01<<4) +#define ONA10IV_DAI_TDM4 (0x02<<4) +#define ONA10IV_DAI_TDM8 (0x03<<4) +#define ONA10IV_FS_16KHZ (0x02<<0) +#define ONA10IV_FS_22KHZ (0x03<<0) +#define ONA10IV_FS_32KHZ (0x04<<0) +#define ONA10IV_FS_44KHZ (0x05<<0) +#define ONA10IV_FS_48KHZ (0x06<<0) +#define ONA10IV_FS_96KHZ (0x08<<0) +#define ONA10IV_FS_MASK (0x0F<<0) + +#define ONA10IV_REG_DAI_CTRL2 (0x07) +#define ONA10IV_SAMPW_16 (0x00<<6) +#define ONA10IV_SAMPW_24 (0x01<<6) +#define ONA10IV_SAMPW_32 (0x02<<6) +#define ONA10IV_SAMPW_MASK (0x03<<6) +#define ONA10IV_SLOTW_16 (0x00<<4) +#define ONA10IV_SLOTW_24 (0x01<<4) +#define ONA10IV_SLOTW_32 (0x02<<4) +#define ONA10IV_SLOTW_MASK (0x03<<4) +#define ONA10IV_FRM_DLY_0 (0x00<<2) +#define ONA10IV_FRM_DLY_1 (0x01<<2) +#define ONA10IV_FRM_DLY_2 (0x02<<2) +#define ONA10IV_FRM_DLY_3 (0x03<<2) +#define ONA10IV_FRM_POL_NORMAL (0x00<<1) +#define ONA10IV_FRM_POL_INVERT (0x01<<1) +#define ONA10IV_FRCK_MODE_50 (0x00<<0) +#define ONA10IV_FRCK_MODE_PULSE (0x01<<0) + +#define ONA10IV_REG_DAI_CTRL3 (0x08) +#define ONA10IV_DAC_MODE_1 (0x00<<4) +#define ONA10IV_DAC_MODE_2 (0x01<<4) +#define ONA10IV_DAC_MODE_3 (0x02<<4) +#define ONA10IV_DAC_MODE_NOT_USED (0x03<<4) +#define ONA10IV_PDM_MODE_PCM (0x00<<3) +#define ONA10IV_PDM_MODE_PDM (0x01<<3) +#define ONA10IV_A_SLOT_1 (0x00<<0) +#define ONA10IV_A_SLOT_2 (0x01<<0) +#define ONA10IV_A_SLOT_3 (0x02<<0) +#define ONA10IV_A_SLOT_4 (0x03<<0) +#define ONA10IV_A_SLOT_5 (0x04<<0) +#define ONA10IV_A_SLOT_6 (0x05<<0) +#define ONA10IV_A_SLOT_7 (0x06<<0) +#define ONA10IV_A_SLOT_8 (0x07<<0) +#define ONA10IV_A_SLOT_MSK (0x07<<0) + +#define ONA10IV_REG_MAX_VOL (0x09) +#define ONA10IV_VOL_0DB (0x00) +#define ONA10IV_VOL_MINUS_0_375DB (0x01) +#define ONA10IV_VOL_MINUS_0_750DB (0x02) +#define ONA10IV_VOL_MINUS_1_125DB (0x03) +#define ONA10IV_VOL_MINUS_1_500DB (0x04) +#define ONA10IV_VOL_MINUS_1_875DB (0x05) +#define ONA10IV_VOL_MINUS_2_250DB (0x06) +#define ONA10IV_VOL_MINUS_2_625DB (0x07) +#define ONA10IV_VOL_MINUS_3_000DB (0x08) +#define ONA10IV_VOL_MINUS_3_375DB (0x09) +#define ONA10IV_VOL_MINUS_3_750DB (0x0A) +#define ONA10IV_VOL_MINUS_4_125DB (0x0B) +#define ONA10IV_VOL_MINUS_4_500DB (0x0C) +#define ONA10IV_VOL_MINUS_4_875DB (0x0D) +#define ONA10IV_VOL_MINUS_5_250DB (0x0E) +#define ONA10IV_VOL_MINUS_5_625DB (0x0F) +#define ONA10IV_VOL_MINUS_6_000DB (0x10) +#define ONA10IV_VOL_MINUS_6_375DB (0x11) +#define ONA10IV_VOL_MINUS_6_750DB (0x12) +#define ONA10IV_VOL_MINUS_7_125DB (0x13) +#define ONA10IV_VOL_MINUS_7_500DB (0x14) +#define ONA10IV_VOL_MINUS_7_875DB (0x15) +#define ONA10IV_VOL_MINUS_8_250DB (0x16) +#define ONA10IV_VOL_MINUS_8_625DB (0x17) +#define ONA10IV_VOL_MINUS_9_000DB (0x18) +#define ONA10IV_VOL_MINUS_9_375DB (0x19) +#define ONA10IV_VOL_MINUS_9_750DB (0x1A) +#define ONA10IV_VOL_MINUS_10_125DB (0x1B) +#define ONA10IV_VOL_MINUS_10_500DB (0x1C) +#define ONA10IV_VOL_MINUS_10_875DB (0x1D) +#define ONA10IV_VOL_MINUS_11_250DB (0x1E) +#define ONA10IV_VOL_MINUS_11_625DB (0x1F) +#define ONA10IV_VOL_MINUS_12_000DB (0x20) +#define ONA10IV_VOL_MINUS_12_375DB (0x21) +#define ONA10IV_VOL_MINUS_12_750DB (0x22) +#define ONA10IV_VOL_MINUS_13_125DB (0x23) +#define ONA10IV_VOL_MINUS_13_500DB (0x24) +#define ONA10IV_VOL_MINUS_13_875DB (0x25) +#define ONA10IV_VOL_MINUS_14_250DB (0x26) +#define ONA10IV_VOL_MINUS_14_625DB (0x27) +#define ONA10IV_VOL_MINUS_15_000DB (0x28) +#define ONA10IV_VOL_MINUS_15_375DB (0x29) +#define ONA10IV_VOL_MINUS_15_750DB (0x2A) +#define ONA10IV_VOL_MINUS_16_125DB (0x2B) +#define ONA10IV_VOL_MINUS_16_500DB (0x2C) +#define ONA10IV_VOL_MINUS_16_875DB (0x2D) +#define ONA10IV_VOL_MINUS_17_250DB (0x2E) +#define ONA10IV_VOL_MINUS_17_625DB (0x2F) +#define ONA10IV_VOL_MINUS_18_000DB (0x30) +#define ONA10IV_VOL_MINUS_18_375DB (0x31) +#define ONA10IV_VOL_MINUS_18_750DB (0x32) +#define ONA10IV_VOL_MINUS_19_125DB (0x33) +#define ONA10IV_VOL_MINUS_19_500DB (0x34) +#define ONA10IV_VOL_MINUS_19_875DB (0x35) +#define ONA10IV_VOL_MINUS_20_250DB (0x36) +#define ONA10IV_VOL_MINUS_20_625DB (0x37) +#define ONA10IV_VOL_MINUS_21_000DB (0x38) +#define ONA10IV_VOL_MINUS_21_375DB (0x39) +#define ONA10IV_VOL_MINUS_21_750DB (0x3A) +#define ONA10IV_VOL_MINUS_22_125DB (0x3B) +#define ONA10IV_VOL_MINUS_22_500DB (0x3C) +#define ONA10IV_VOL_MINUS_22_875DB (0x3D) +#define ONA10IV_VOL_MINUS_23_250DB (0x3E) +#define ONA10IV_VOL_MINUS_23_625DB (0x3F) +#define ONA10IV_VOL_MINUS_24_000DB (0x40) +#define ONA10IV_VOL_MINUS_24_375DB (0x41) +#define ONA10IV_VOL_MINUS_24_750DB (0x42) +#define ONA10IV_VOL_MINUS_25_125DB (0x43) +#define ONA10IV_VOL_MINUS_25_500DB (0x44) +#define ONA10IV_VOL_MINUS_25_875DB (0x45) +#define ONA10IV_VOL_MINUS_26_250DB (0x46) +#define ONA10IV_VOL_MINUS_26_625DB (0x47) +#define ONA10IV_VOL_MINUS_27_000DB (0x48) +#define ONA10IV_VOL_MINUS_27_375DB (0x49) +#define ONA10IV_VOL_MINUS_27_750DB (0x4A) +#define ONA10IV_VOL_MINUS_28_125DB (0x4B) +#define ONA10IV_VOL_MINUS_28_500DB (0x4C) +#define ONA10IV_VOL_MINUS_28_875DB (0x4D) +#define ONA10IV_VOL_MINUS_29_250DB (0x4E) +#define ONA10IV_VOL_MINUS_29_625DB (0x4F) +#define ONA10IV_VOL_MINUS_30_000DB (0x50) +#define ONA10IV_VOL_MINUS_30_375DB (0x51) +#define ONA10IV_VOL_MINUS_30_750DB (0x52) +#define ONA10IV_VOL_MINUS_31_125DB (0x53) +#define ONA10IV_VOL_MINUS_31_500DB (0x54) +#define ONA10IV_VOL_MINUS_31_875DB (0x55) +#define ONA10IV_VOL_MINUS_32_250DB (0x56) +#define ONA10IV_VOL_MINUS_32_625DB (0x57) +#define ONA10IV_VOL_MINUS_33_000DB (0x58) +#define ONA10IV_VOL_MINUS_33_375DB (0x59) +#define ONA10IV_VOL_MINUS_33_750DB (0x5A) +#define ONA10IV_VOL_MINUS_34_125DB (0x5B) +#define ONA10IV_VOL_MINUS_34_500DB (0x5C) +#define ONA10IV_VOL_MINUS_34_875DB (0x5D) +#define ONA10IV_VOL_MINUS_35_250DB (0x5E) +#define ONA10IV_VOL_MINUS_35_625DB (0x5F) +#define ONA10IV_VOL_MINUS_36_000DB (0x60) +#define ONA10IV_VOL_MINUS_36_375DB (0x61) +#define ONA10IV_VOL_MINUS_36_750DB (0x62) +#define ONA10IV_VOL_MINUS_37_125DB (0x63) +#define ONA10IV_VOL_MINUS_37_500DB (0x64) +#define ONA10IV_VOL_MINUS_37_875DB (0x65) +#define ONA10IV_VOL_MINUS_38_250DB (0x66) +#define ONA10IV_VOL_MINUS_38_625DB (0x67) +#define ONA10IV_VOL_MINUS_39_000DB (0x68) +#define ONA10IV_VOL_MINUS_39_375DB (0x69) +#define ONA10IV_VOL_MINUS_39_750DB (0x6A) +#define ONA10IV_VOL_MINUS_40_125DB (0x6B) +#define ONA10IV_VOL_MINUS_40_500DB (0x6C) +#define ONA10IV_VOL_MINUS_40_875DB (0x6D) +#define ONA10IV_VOL_MINUS_41_250DB (0x6E) +#define ONA10IV_VOL_MINUS_41_625DB (0x6F) +#define ONA10IV_VOL_MINUS_42_000DB (0x70) +#define ONA10IV_VOL_MINUS_42_375DB (0x71) +#define ONA10IV_VOL_MINUS_42_750DB (0x72) +#define ONA10IV_VOL_MINUS_43_125DB (0x73) +#define ONA10IV_VOL_MINUS_43_500DB (0x74) +#define ONA10IV_VOL_MINUS_43_875DB (0x75) +#define ONA10IV_VOL_MINUS_44_250DB (0x76) +#define ONA10IV_VOL_MINUS_44_625DB (0x77) +#define ONA10IV_VOL_MINUS_45_000DB (0x78) +#define ONA10IV_VOL_MINUS_45_375DB (0x79) +#define ONA10IV_VOL_MINUS_45_750DB (0x7A) +#define ONA10IV_VOL_MINUS_46_125DB (0x7B) +#define ONA10IV_VOL_MINUS_46_500DB (0x7C) +#define ONA10IV_VOL_MINUS_46_875DB (0x7D) +#define ONA10IV_VOL_MINUS_47_250DB (0x7E) +#define ONA10IV_VOL_MINUS_47_625DB (0x7F) +#define ONA10IV_VOL_MINUS_48_000DB (0x80) +#define ONA10IV_VOL_MINUS_48_375DB (0x81) +#define ONA10IV_VOL_MINUS_48_750DB (0x82) +#define ONA10IV_VOL_MINUS_49_125DB (0x83) +#define ONA10IV_VOL_MINUS_49_500DB (0x84) +#define ONA10IV_VOL_MINUS_49_875DB (0x85) +#define ONA10IV_VOL_MINUS_50_250DB (0x86) +#define ONA10IV_VOL_MINUS_50_625DB (0x87) +#define ONA10IV_VOL_MINUS_51_000DB (0x88) +#define ONA10IV_VOL_MINUS_51_375DB (0x89) +#define ONA10IV_VOL_MINUS_51_750DB (0x8A) +#define ONA10IV_VOL_MINUS_52_125DB (0x8B) +#define ONA10IV_VOL_MINUS_52_500DB (0x8C) +#define ONA10IV_VOL_MINUS_52_875DB (0x8D) +#define ONA10IV_VOL_MINUS_53_250DB (0x8E) +#define ONA10IV_VOL_MINUS_53_625DB (0x8F) +#define ONA10IV_VOL_MINUS_54_000DB (0x90) +#define ONA10IV_VOL_MINUS_54_375DB (0x91) +#define ONA10IV_VOL_MINUS_54_750DB (0x92) +#define ONA10IV_VOL_MINUS_55_125DB (0x93) +#define ONA10IV_VOL_MINUS_55_500DB (0x94) +#define ONA10IV_VOL_MINUS_55_875DB (0x95) +#define ONA10IV_VOL_MINUS_56_250DB (0x96) +#define ONA10IV_VOL_MINUS_56_625DB (0x97) +#define ONA10IV_VOL_MINUS_57_000DB (0x98) +#define ONA10IV_VOL_MINUS_57_375DB (0x99) +#define ONA10IV_VOL_MINUS_57_750DB (0x9A) +#define ONA10IV_VOL_MINUS_58_125DB (0x9B) +#define ONA10IV_VOL_MINUS_58_500DB (0x9C) +#define ONA10IV_VOL_MINUS_58_875DB (0x9D) +#define ONA10IV_VOL_MINUS_59_250DB (0x9E) +#define ONA10IV_VOL_MINUS_59_625DB (0x9F) +#define ONA10IV_VOL_MINUS_60_000DB (0xA0) +#define ONA10IV_VOL_MINUS_60_375DB (0xA1) +#define ONA10IV_VOL_MINUS_60_750DB (0xA2) +#define ONA10IV_VOL_MINUS_61_125DB (0xA3) +#define ONA10IV_VOL_MINUS_61_500DB (0xA4) +#define ONA10IV_VOL_MINUS_61_875DB (0xA5) +#define ONA10IV_VOL_MINUS_62_250DB (0xA6) +#define ONA10IV_VOL_MINUS_62_625DB (0xA7) +#define ONA10IV_VOL_MINUS_63_000DB (0xA8) +#define ONA10IV_VOL_MINUS_63_375DB (0xA9) +#define ONA10IV_VOL_MINUS_63_750DB (0xAA) +#define ONA10IV_VOL_MINUS_64_125DB (0xAB) +#define ONA10IV_VOL_MINUS_64_500DB (0xAC) +#define ONA10IV_VOL_MINUS_64_875DB (0xAD) +#define ONA10IV_VOL_MINUS_65_250DB (0xAE) +#define ONA10IV_VOL_MINUS_65_625DB (0xAF) +#define ONA10IV_VOL_MINUS_66_000DB (0xB0) +#define ONA10IV_VOL_MINUS_66_375DB (0xB1) +#define ONA10IV_VOL_MINUS_66_750DB (0xB2) +#define ONA10IV_VOL_MINUS_67_125DB (0xB3) +#define ONA10IV_VOL_MINUS_67_500DB (0xB4) +#define ONA10IV_VOL_MINUS_67_875DB (0xB5) +#define ONA10IV_VOL_MINUS_68_250DB (0xB6) +#define ONA10IV_VOL_MINUS_68_625DB (0xB7) +#define ONA10IV_VOL_MINUS_69_000DB (0xB8) +#define ONA10IV_VOL_MINUS_69_375DB (0xB9) +#define ONA10IV_VOL_MINUS_69_750DB (0xBA) +#define ONA10IV_VOL_MINUS_70_125DB (0xBB) +#define ONA10IV_VOL_MINUS_70_500DB (0xBC) +#define ONA10IV_VOL_MINUS_70_875DB (0xBD) +#define ONA10IV_VOL_MINUS_71_250DB (0xBE) +#define ONA10IV_VOL_MINUS_71_625DB (0xBF) +#define ONA10IV_VOL_MINUS_72_000DB (0xC0) +#define ONA10IV_VOL_MINUS_72_375DB (0xC1) +#define ONA10IV_VOL_MINUS_72_750DB (0xC2) +#define ONA10IV_VOL_MINUS_73_125DB (0xC3) +#define ONA10IV_VOL_MINUS_73_500DB (0xC4) +#define ONA10IV_VOL_MINUS_73_875DB (0xC5) +#define ONA10IV_VOL_MINUS_74_250DB (0xC6) +#define ONA10IV_VOL_MINUS_74_625DB (0xC7) +#define ONA10IV_VOL_MINUS_75_000DB (0xC8) +#define ONA10IV_VOL_MINUS_75_375DB (0xC9) +#define ONA10IV_VOL_MINUS_75_750DB (0xCA) +#define ONA10IV_VOL_MINUS_76_125DB (0xCB) +#define ONA10IV_VOL_MINUS_76_500DB (0xCC) +#define ONA10IV_VOL_MINUS_76_875DB (0xCD) +#define ONA10IV_VOL_MINUS_77_250DB (0xCE) +#define ONA10IV_VOL_MINUS_77_625DB (0xCF) +#define ONA10IV_VOL_MINUS_78_000DB (0xD0) +#define ONA10IV_VOL_MINUS_78_375DB (0xD1) +#define ONA10IV_VOL_MINUS_78_750DB (0xD2) +#define ONA10IV_VOL_MINUS_79_125DB (0xD3) +#define ONA10IV_VOL_MINUS_79_500DB (0xD4) +#define ONA10IV_VOL_MINUS_79_875DB (0xD5) +#define ONA10IV_VOL_MINUS_80_250DB (0xD6) +#define ONA10IV_VOL_MINUS_80_625DB (0xD7) +#define ONA10IV_VOL_MINUS_81_000DB (0xD8) +#define ONA10IV_VOL_MINUS_81_375DB (0xD9) +#define ONA10IV_VOL_MINUS_81_750DB (0xDA) +#define ONA10IV_VOL_MINUS_82_125DB (0xDB) +#define ONA10IV_VOL_MINUS_82_500DB (0xDC) +#define ONA10IV_VOL_MINUS_82_875DB (0xDD) +#define ONA10IV_VOL_MINUS_83_250DB (0xDE) +#define ONA10IV_VOL_MINUS_83_625DB (0xDF) +#define ONA10IV_VOL_MINUS_84_000DB (0xE0) +#define ONA10IV_VOL_MINUS_84_375DB (0xE1) +#define ONA10IV_VOL_MINUS_84_750DB (0xE2) +#define ONA10IV_VOL_MINUS_85_125DB (0xE3) +#define ONA10IV_VOL_MINUS_85_500DB (0xE4) +#define ONA10IV_VOL_MINUS_85_875DB (0xE5) +#define ONA10IV_VOL_MINUS_86_250DB (0xE6) +#define ONA10IV_VOL_MINUS_86_625DB (0xE7) +#define ONA10IV_VOL_MINUS_87_000DB (0xE8) +#define ONA10IV_VOL_MINUS_87_375DB (0xE9) +#define ONA10IV_VOL_MINUS_87_750DB (0xEA) +#define ONA10IV_VOL_MINUS_88_125DB (0xEB) +#define ONA10IV_VOL_MINUS_88_500DB (0xEC) +#define ONA10IV_VOL_MINUS_88_875DB (0xED) +#define ONA10IV_VOL_MINUS_89_250DB (0xEE) +#define ONA10IV_VOL_MINUS_89_625DB (0xEF) +#define ONA10IV_VOL_MINUS_90_000DB (0xF0) +#define ONA10IV_VOL_MINUS_90_375DB (0xF1) +#define ONA10IV_VOL_MINUS_90_750DB (0xF2) +#define ONA10IV_VOL_MINUS_91_125DB (0xF3) +#define ONA10IV_VOL_MINUS_91_500DB (0xF4) +#define ONA10IV_VOL_MINUS_91_875DB (0xF5) +#define ONA10IV_VOL_MINUS_92_250DB (0xF6) +#define ONA10IV_VOL_MINUS_92_625DB (0xF7) +#define ONA10IV_VOL_MINUS_93_000DB (0xF8) +#define ONA10IV_VOL_MINUS_93_375DB (0xF9) +#define ONA10IV_VOL_MINUS_93_750DB (0xFA) +#define ONA10IV_VOL_MINUS_94_125DB (0xFB) +#define ONA10IV_VOL_MINUS_94_500DB (0xFC) +#define ONA10IV_VOL_MINUS_94_875DB (0xFD) +#define ONA10IV_VOL_MINUS_95_250DB (0xFE) +#define ONA10IV_VOL_AMP_MUTE (0xFF) +#define ONA10IV_VOL_AMP_MASK (0xFF) + +#define ONA10IV_REG_VOL_CTRL (0x0A) +#define ONA10IV_VOL_RAMP (0x01<<5) +#define ONA10IV_VOL_ZEROCROSS (0x00<<5) +#define ONA10IV_VOL_RAMP_0_01MSSTEP (0x00<<3) +#define ONA10IV_VOL_RAMP_0_10MSSTEP (0x01<<3) +#define ONA10IV_VOL_RAMP_0_50MSSTEP (0x02<<3) +#define ONA10IV_VOL_RAMP_1_00MSSTEP (0x03<<3) +#define ONA10IV_AVOL_UP_DISABLE (0x00<<2) +#define ONA10IV_AVOL_UP_RAMP (0x01<<2) +#define ONA10IV_AVOL_DN_DISABLE (0x00<<1) +#define ONA10IV_AVOL_DN_RAMP (0x01<<1) +#define ONA10IV_MUTE_NORMAL (0x00<<0) +#define ONA10IV_MUTE_AMPLIFIER_OUT (0x01<<0) + +#define ONA10IV_REG_GAIN_CTRL1 (0x0B) +#define ONA10IV_PCM_AMP_GAIN_PLUS_16DB (0x00<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_12DB (0x08<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_9DB (0x0E<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_6DB (0x14<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_3DB (0x1A<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_MASK (0x1F<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_MAX (0x20) + +#define ONA10IV_REG_GAIN_CTRL2 (0x0C) +#define ONA10IV_PDM_DAC_MAP_7_23DBV (0x00<<0) +#define ONA10IV_PDM_DAC_MAP_4_31DBV (0x01<<0) +#define ONA10IV_PDM_DAC_MAP_M0_13DBV (0x02<<0) +#define ONA10IV_PDM_DAC_MAP_M9_68DBV (0x03<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_16DB (0x00<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_12DB (0x08<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_9DB (0x0E<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_6DB (0x14<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_3DB (0x1A<<0) + +#define ONA10IV_REG_EMI_CTRL (0x0D) +#define ONA10IV_AMP_ERC_1_00VPNS (0x00<<3) +#define ONA10IV_AMP_ERC_0_75VPNS (0x01<<3) +#define ONA10IV_AMP_ERC_0_50VPNS (0x02<<3) +#define ONA10IV_AMP_ERC_3_50VPNS (0x03<<3) +#define ONA10IV_SS_MOD_5_3PERCENT (0x00<<0) +#define ONA10IV_SS_MOD_7_0PERCENT (0x01<<0) +#define ONA10IV_SS_MOD_10_6PERCENT (0x02<<0) +#define ONA10IV_SS_MOD_21_2PERCENT (0x03<<0) +#define ONA10IV_SS_MOD_0_0PERCENT (0x04<<0) +#define ONA10IV_SS_MOD_3_0PERCENT (0x05<<0) +#define ONA10IV_SS_MOD_3_6PERCENT (0x06<<0) +#define ONA10IV_SS_MOD_4_2PERCENT (0x07<<0) + +#define ONA10IV_REG_AGC_BATT (0x0E) +#define ONA10IV_BATT_ATH_9_763V (0x00<<0) +#define ONA10IV_BATT_ATH_9_835V (0x01<<0) +#define ONA10IV_BATT_ATH_9_907V (0x02<<0) +#define ONA10IV_BATT_ATH_9_979V (0x03<<0) +#define ONA10IV_BATT_ATH_10_510V (0x04<<0) +#define ONA10IV_BATT_ATH_10_123V (0x05<<0) +#define ONA10IV_BATT_ATH_10_195V (0x06<<0) +#define ONA10IV_BATT_ATH_10_267V (0x07<<0) +#define ONA10IV_BATT_ATH_10_339V (0x08<<0) +#define ONA10IV_BATT_ATH_10_411V (0x09<<0) +#define ONA10IV_BATT_ATH_10_483V (0x0A<<0) +#define ONA10IV_BATT_ATH_10_555V (0x0B<<0) +#define ONA10IV_BATT_ATH_10_627V (0x0C<<0) +#define ONA10IV_BATT_ATH_10_699V (0x0D<<0) +#define ONA10IV_BATT_ATH_10_771V (0x0E<<0) +#define ONA10IV_BATT_ATH_10_843V (0x0F<<0) +#define ONA10IV_BATT_ATH_10_915V (0x10<<0) +#define ONA10IV_BATT_ATH_10_987V (0x11<<0) +#define ONA10IV_BATT_ATH_11_590V (0x12<<0) +#define ONA10IV_BATT_ATH_11_131V (0x13<<0) +#define ONA10IV_BATT_ATH_11_203V (0x14<<0) +#define ONA10IV_BATT_ATH_11_275V (0x15<<0) +#define ONA10IV_BATT_ATH_11_347V (0x16<<0) +#define ONA10IV_BATT_ATH_11_419V (0x17<<0) +#define ONA10IV_BATT_ATH_11_491V (0x18<<0) +#define ONA10IV_BATT_ATH_11_563V (0x19<<0) +#define ONA10IV_BATT_ATH_11_635V (0x1A<<0) +#define ONA10IV_BATT_ATH_11_707V (0x1B<<0) +#define ONA10IV_BATT_ATH_11_779V (0x1C<<0) +#define ONA10IV_BATT_ATH_11_851V (0x1D<<0) +#define ONA10IV_BATT_ATH_11_923V (0x1E<<0) +#define ONA10IV_BATT_ATH_11_995V (0x1F<<0) + +#define ONA10IV_REG_AGC_CTRL1 (0x0F) +#define ONA10IV_AGC_HOLD_10MS (0x00<<4) +#define ONA10IV_AGC_HOLD_45MS (0x01<<4) +#define ONA10IV_AGC_HOLD_80MS (0x02<<4) +#define ONA10IV_AGC_HOLD_115MS (0x03<<4) +#define ONA10IV_AGC_HOLD_150MS (0x04<<4) +#define ONA10IV_AGC_HOLD_185MS (0x05<<4) +#define ONA10IV_AGC_HOLD_220MS (0x06<<4) +#define ONA10IV_AGC_HOLD_255MS (0x07<<4) +#define ONA10IV_AGC_HOLD_290MS (0x08<<4) +#define ONA10IV_AGC_HOLD_325MS (0x09<<4) +#define ONA10IV_AGC_HOLD_360MS (0x0A<<4) +#define ONA10IV_AGC_HOLD_395MS (0x0B<<4) +#define ONA10IV_AGC_HOLD_430MS (0x0C<<4) +#define ONA10IV_AGC_HOLD_465MS (0x0D<<4) +#define ONA10IV_AGC_HOLD_500MS (0x0E<<4) +#define ONA10IV_AGC_HOLD_INFINIT (0x0F<<4) +#define ONA10IV_AGC_ATTACK_10USPDB (0x00<<0) +#define ONA10IV_AGC_ATTACK_45USPDB (0x01<<0) +#define ONA10IV_AGC_ATTACK_80USPDB (0x02<<0) +#define ONA10IV_AGC_ATTACK_115USPDB (0x03<<0) +#define ONA10IV_AGC_ATTACK_150USPDB (0x04<<0) +#define ONA10IV_AGC_ATTACK_185USPDB (0x05<<0) +#define ONA10IV_AGC_ATTACK_220USPDB (0x06<<0) +#define ONA10IV_AGC_ATTACK_255USPDB (0x07<<0) +#define ONA10IV_AGC_ATTACK_290USPDB (0x08<<0) +#define ONA10IV_AGC_ATTACK_325USPDB (0x09<<0) +#define ONA10IV_AGC_ATTACK_360USPDB (0x0A<<0) +#define ONA10IV_AGC_ATTACK_395USPDB (0x0B<<0) +#define ONA10IV_AGC_ATTACK_430USPDB (0x0C<<0) +#define ONA10IV_AGC_ATTACK_465USPDB (0x0D<<0) +#define ONA10IV_AGC_ATTACK_500USPDB (0x0E<<0) +#define ONA10IV_AGC_ATTACK_535USPDB (0x0F<<0) + +#define ONA10IV_REG_AGC_CTRL2 (0x10) +#define ONA10IV_AGC_MAX_ATT_MINUS_9DB (0x00<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_8DB (0x01<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_7DB (0x02<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_6DB (0x03<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_5DB (0x04<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_4DB (0x05<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_3DB (0x06<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_2DB (0x07<<4) +#define ONA10IV_AGC_RELEASE_5MSPDB (0x00<<0) +#define ONA10IV_AGC_RELEASE_75MSPDB (0x01<<0) +#define ONA10IV_AGC_RELEASE_145MSPDB (0x02<<0) +#define ONA10IV_AGC_RELEASE_215MSPDB (0x03<<0) +#define ONA10IV_AGC_RELEASE_285MSPDB (0x04<<0) +#define ONA10IV_AGC_RELEASE_355MSPDB (0x05<<0) +#define ONA10IV_AGC_RELEASE_425MSPDB (0x06<<0) +#define ONA10IV_AGC_RELEASE_495MSPDB (0x07<<0) +#define ONA10IV_AGC_RELEASE_565MSPDB (0x08<<0) +#define ONA10IV_AGC_RELEASE_635MSPDB (0x09<<0) +#define ONA10IV_AGC_RELEASE_705MSPDB (0x0A<<0) +#define ONA10IV_AGC_RELEASE_775MSPDB (0x0B<<0) +#define ONA10IV_AGC_RELEASE_845MSPDB (0x0C<<0) +#define ONA10IV_AGC_RELEASE_915MSPDB (0x0D<<0) +#define ONA10IV_AGC_RELEASE_985MSPDB (0x0E<<0) +#define ONA10IV_AGC_RELEASE_1055MSPDB (0x0F<<0) + +#define ONA10IV_REG_AGC_CTRL3 (0x11) +#define ONA10IV_AGC_TIMEOUT_DISABLE (0x00<<0) +#define ONA10IV_AGC_TIMEOUT_565MS (0x01<<0) +#define ONA10IV_AGC_TIMEOUT_775MS (0x02<<0) +#define ONA10IV_AGC_TIMEOUT_985MS (0x03<<0) + +#define ONA10IV_REG_MAGC_CTRL1 (0x12) +#define ONA10IV_RX_SLOT8_IGNORES (0x00<<7) +#define ONA10IV_RX_SLOT8_LISTENS (0x01<<7) +#define ONA10IV_RX_SLOT7_IGNORES (0x00<<6) +#define ONA10IV_RX_SLOT7_LISTENS (0x01<<6) +#define ONA10IV_RX_SLOT6_IGNORES (0x00<<5) +#define ONA10IV_RX_SLOT6_LISTENS (0x01<<5) +#define ONA10IV_RX_SLOT5_IGNORES (0x00<<4) +#define ONA10IV_RX_SLOT5_LISTENS (0x01<<4) +#define ONA10IV_RX_SLOT4_IGNORES (0x00<<3) +#define ONA10IV_RX_SLOT4_LISTENS (0x01<<3) +#define ONA10IV_RX_SLOT3_IGNORES (0x00<<2) +#define ONA10IV_RX_SLOT3_LISTENS (0x01<<2) +#define ONA10IV_RX_SLOT2_IGNORES (0x00<<1) +#define ONA10IV_RX_SLOT2_LISTENS (0x01<<1) +#define ONA10IV_RX_SLOT1_IGNORES (0x00<<0) +#define ONA10IV_RX_SLOT1_LISTENS (0x01<<0) + +#define ONA10IV_REG_MAGC_CTRL2 (0x13) +#define ONA10IV_TX_SLOT8_HIZ (0x00<<7) +#define ONA10IV_TX_SLOT8_TRANSMIT (0x01<<7) +#define ONA10IV_TX_SLOT7_HIZ (0x00<<6) +#define ONA10IV_TX_SLOT7_TRANSMIT (0x01<<6) +#define ONA10IV_TX_SLOT6_HIZ (0x00<<5) +#define ONA10IV_TX_SLOT6_TRANSMIT (0x01<<5) +#define ONA10IV_TX_SLOT5_HIZ (0x00<<4) +#define ONA10IV_TX_SLOT5_TRANSMIT (0x01<<4) +#define ONA10IV_TX_SLOT4_HIZ (0x00<<3) +#define ONA10IV_TX_SLOT4_TRANSMIT (0x01<<3) +#define ONA10IV_TX_SLOT3_HIZ (0x00<<2) +#define ONA10IV_TX_SLOT3_TRANSMIT (0x01<<2) +#define ONA10IV_TX_SLOT2_HIZ (0x00<<1) +#define ONA10IV_TX_SLOT2_TRANSMIT (0x01<<1) +#define ONA10IV_TX_SLOT1_HIZ (0x00<<0) +#define ONA10IV_TX_SLOT1_TRANSMIT (0x01<<0) + +#define ONA10IV_REG_MAGC_CTRL3 (0x14) +#define ONA10IV_MAGC_ODRV_100P (0x00<<1) +#define ONA10IV_MAGC_ODRV_75P (0x01<<1) +#define ONA10IV_MAGC_ODRV_50P (0x02<<1) +#define ONA10IV_MAGC_ODRV_25P (0x03<<1) +#define ONA10IV_MAGC_EN_DISABLE (0x00<<0) +#define ONA10IV_MAGC_EN_ENABLE (0x01<<0) + +#define ONA10IV_REG_SENSE_CTRL (0x15) +#define ONA10IV_T_HOLD_10MS (0x00<<6) +#define ONA10IV_T_HOLD_45MS (0x01<<6) +#define ONA10IV_T_HOLD_80MS (0x02<<6) +#define ONA10IV_T_HOLD_115MS (0x03<<6) +#define ONA10IV_T_ATTACK_2MS_STEP (0x00<<4) +#define ONA10IV_T_ATTACK_4MS_STEP (0x01<<4) +#define ONA10IV_T_ATTACK_6MS_STEP (0x02<<4) +#define ONA10IV_T_ATTACK_8MS_STEP (0x03<<4) +#define ONA10IV_T_ATH_140CELSIUS (0x00<<2) +#define ONA10IV_T_ATH_130CELSIUS (0x01<<2) +#define ONA10IV_T_ATH_120CELSIUS (0x02<<2) +#define ONA10IV_T_ATH_110CELSIUS (0x03<<2) +#define ONA10IV_T_SAMP_40MS (0x00<<0) +#define ONA10IV_T_SAMP_20MS (0x01<<0) +#define ONA10IV_T_SAMP_10MS (0x02<<0) +#define ONA10IV_T_SAMP_CONSTANTLY (0x03<<0) + +#define ONA10IV_REG_T_SENSE_OUT1 (0x16) +#define ONA10IV_REG_T_SENSE_OUT2 (0x17) +#define ONA10IV_T_SENSE_OUT2_BIT (0x07) +#define ONA10IV_T_SENSE_OUT2_SIGN_BIT (0x04) + +#define ONA10IV_REG_DATAO_CTRL1 (0x18) +#define ONA10IV_HPF_BP_NORMAL (0x00<<4) +#define ONA10IV_HPF_BP_HIGHPASS (0x01<<4) +#define ONA10IV_PDM_OUT_FALLING (0x00<<2) +#define ONA10IV_PDM_OUT_RISING (0x01<<2) +#define ONA10IV_DATA_ODRV_100P (0x00<<0) +#define ONA10IV_DATA_ODRV_75P (0x01<<0) +#define ONA10IV_DATA_ODRV_50P (0x02<<0) +#define ONA10IV_DATA_ODRV_25P (0x03<<0) + +#define ONA10IV_REG_DATAO_CTRL2 (0x19) +#define ONA10IV_I_DATAO_TX_DISABLE (0x00<<7) +#define ONA10IV_I_DATAO_TX_ENABLE (0x01<<7) +#define ONA10IV_V_DATAO_TX_DISABLE (0x00<<6) +#define ONA10IV_V_DATAO_TX_ENABLE (0x01<<6) +#define ONA10IV_I_SLOT_1 (0x00<<3) +#define ONA10IV_I_SLOT_2 (0x01<<3) +#define ONA10IV_I_SLOT_3 (0x02<<3) +#define ONA10IV_I_SLOT_4 (0x03<<3) +#define ONA10IV_I_SLOT_5 (0x04<<3) +#define ONA10IV_I_SLOT_6 (0x05<<3) +#define ONA10IV_I_SLOT_7 (0x06<<3) +#define ONA10IV_I_SLOT_8 (0x07<<3) +#define ONA10IV_V_SLOT_1 (0x00<<0) +#define ONA10IV_V_SLOT_2 (0x01<<0) +#define ONA10IV_V_SLOT_3 (0x02<<0) +#define ONA10IV_V_SLOT_4 (0x03<<0) +#define ONA10IV_V_SLOT_5 (0x04<<0) +#define ONA10IV_V_SLOT_6 (0x05<<0) +#define ONA10IV_V_SLOT_7 (0x06<<0) +#define ONA10IV_V_SLOT_8 (0x07<<0) + +#define ONA10IV_REG_DATAO_CTRL3 (0x1A) +#define ONA10IV_T_DATAO_TX_DISABLE (0x00<<3) +#define ONA10IV_T_DATAO_TX_ENABLE (0x01<<3) +#define ONA10IV_T_SLOT_1 (0x00<<0) +#define ONA10IV_T_SLOT_2 (0x01<<0) +#define ONA10IV_T_SLOT_3 (0x02<<0) +#define ONA10IV_T_SLOT_4 (0x03<<0) +#define ONA10IV_T_SLOT_5 (0x04<<0) +#define ONA10IV_T_SLOT_6 (0x05<<0) +#define ONA10IV_T_SLOT_7 (0x06<<0) +#define ONA10IV_T_SLOT_8 (0x07<<0) +#endif // _ONA10IV__

2 1

[PATCH] ASoC: Intel: bytcr_rt5640: Fix HP Pavilion x2 Detachable quirks
by Hans de Goede 19 Nov '20

19 Nov '20

The HP Pavilion x2 Detachable line comes in many variants: 1. Bay Trail SoC + AXP288 PMIC, Micro-USB charging (10-k010nz, ...) DMI_SYS_VENDOR: "Hewlett-Packard" DMI_PRODUCT_NAME: "HP Pavilion x2 Detachable PC 10" DMI_BOARD_NAME: "8021" 2. Bay Trail SoC + AXP288 PMIC, Type-C charging (10-n000nd, 10-n010nl, ...) DMI_SYS_VENDOR: "Hewlett-Packard" DMI_PRODUCT_NAME: "HP Pavilion x2 Detachable" DMI_BOARD_NAME: "815D" 3. Cherry Trail SoC + AXP288 PMIC, Type-C charging (10-n101ng, ...) DMI_SYS_VENDOR: "HP" DMI_PRODUCT_NAME: "HP Pavilion x2 Detachable" DMI_BOARD_NAME: "813E" 4. Cherry Trail SoC + TI PMIC, Type-C charging (10-p002nd, 10-p018wm, ...) DMI_SYS_VENDOR: "HP" DMI_PRODUCT_NAME: "HP x2 Detachable 10-p0XX" DMI_BOARD_NAME: "827C" 5. Cherry Trail SoC + TI PMIC, Type-C charging (x2-210-g2, ...) DMI_SYS_VENDOR: "HP" DMI_PRODUCT_NAME: "HP x2 210 G2" DMI_BOARD_NAME: "82F4" Variant 1 needs the exact same quirk as variant 2, so relax the DMI check for the existing quirk a bit so that it matches both variant 1 and 2 (note the other variants will still not match). Variant 2 already has an existing quirk (which now also matches variant 1) Variant 3 uses a cx2072x codec, so is not applicable here. Variant 4 almost works with the defaults, but it also needs a quirk to fix jack-detection, add a new quirk for this. Variant 5 does use a RT5640 codec (based on old dmesg output), but was otherwise not tested, keep using the defaults for this variant. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1896924 Fixes: ec8e8418ff7d ("ASoC: Intel: bytcr_rt5640: Add quirks for various devices") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 9dadf6561444..f790514a147d 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -520,10 +520,10 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { .driver_data = (void *)(BYT_RT5640_IN1_MAP | BYT_RT5640_MCLK_EN), }, - { /* HP Pavilion x2 10-n000nd */ + { /* HP Pavilion x2 10-k0XX, 10-n0XX */ .matches = { - DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), - DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HP Pavilion x2 Detachable"), + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pavilion x2 Detachable"), }, .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | BYT_RT5640_JD_SRC_JD2_IN4N | @@ -532,6 +532,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* HP Pavilion x2 10-p0XX */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP x2 Detachable 10-p0XX"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.28.0

3 2

[PATCH 1/2] ASoC: SOF: relax PCM period and buffer size constraints
by Kai Vehmanen 19 Nov '20

19 Nov '20

Current SOF implementation limits period and buffer sizes to multiples of period_min. Period_min is defined in topology, but is in practise set to align with the SOF DSP timer tick (typically 1ms). While this approach helps user-space to avoid period sizes, which are not aligned to the DSP timer tick, it causes problems to applications which want to align data processing size to that of ALSA period size. One example is JACK audio server, which limits period sizes to power of two values. Other ALSA drivers where audio data transfer is driven by a timer tick, like USB, do not constraint period and buffer sizes to exact multiple of the timer tick. To align SOF to follow the same behaviour, drop the additional alignment constraints. As a side-effect, this patch can cause irregularity to period wakeup timing. This happens when application chooses settings which were previously forbidden. For example, if application configures period size to 2^14 bytes and audio config of S32_LE/2ch/48000Hz, one period represents 42.667ms of audio. Without this patch, this configuration is not allowed by SOF. With the patch applied, configuration is allowed but the wakeups are paced by the DSP timer tick, which is typically 1ms. Application will see period wakeups with a 42/43/42/43ms repeating pattern. Both approaches are valid within ALSA context, but relaxing the constraints is better aligned with existing applications and other ALSA drivers like USB audio. BugLink: https://github.com/thesofproject/linux/issues/2391 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/pcm.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 0a70e685f826..37d12162e448 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -478,17 +478,10 @@ static int sof_pcm_open(struct snd_soc_component *component, caps = &spcm->pcm.caps[substream->stream]; - /* set any runtime constraints based on topology */ - snd_pcm_hw_constraint_step(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_BUFFER_BYTES, - le32_to_cpu(caps->period_size_min)); - snd_pcm_hw_constraint_step(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_PERIOD_BYTES, - le32_to_cpu(caps->period_size_min)); - /* set runtime config */ runtime->hw.info = ops->hw_info; /* platform-specific */ + /* set any runtime constraints based on topology */ runtime->hw.formats = le64_to_cpu(caps->formats); runtime->hw.period_bytes_min = le32_to_cpu(caps->period_size_min); runtime->hw.period_bytes_max = le32_to_cpu(caps->period_size_max); -- 2.28.0

2 2

[PATCH v3 0/6] ASoC: codecs: add support for LPASS Codec macros
by Srinivas Kandagatla 19 Nov '20

19 Nov '20

This patchset adds support for two Codec Macro blocks( WSA and VA) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD Codecs to provide headphone/ear/lineout etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Thanks, srini -Changes since v2: - various unnecessary variable intializations removed, suggested by Pierre - fixed a static checker error - collected reviews for dt-bindings. - fixed licence headers as suggested by Pierre. Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add bindings for lpass wsa macro codec ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro ASoC: codecs: lpass-wsa-macro: add dapm widgets and route ASoC: qcom: dt-bindings: add bindings for lpass va macro codec ASoC: codecs: lpass-va-macro: Add support to VA Macro ASoC: codecs: lpass-va-macro: add dapm widgets and routes .../bindings/sound/qcom,lpass-va-macro.yaml | 67 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 69 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-va-macro.c | 1503 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 2464 +++++++++++++++++ sound/soc/codecs/lpass-wsa-macro.h | 17 + 7 files changed, 4132 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml create mode 100644 sound/soc/codecs/lpass-va-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.h -- 2.21.0

3 8

[PATCH v2] ASoC: fsl_sai: Correct the clock source for mclk0
by Shengjiu Wang 19 Nov '20

19 Nov '20

On VF610, mclk0 = bus_clk; On i.MX6SX/6UL/6ULL/7D, mclk0 = mclk1; On i.MX7ULP, mclk0 = bus_clk; On i.MX8QM/8QXP, mclk0 = bus_clk; On i.MX8MQ/8MN/8MM/8MP, mclk0 = bus_clk; So add variable mclk0_is_mclk1 in fsl_sai_soc_data to distinguish these platforms. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - rename mclk0_mclk1_match to mclk0_is_mclk1 - skip MCLK0 when it is same as MCLK1 on master clock scanning sound/soc/fsl/fsl_sai.c | 20 ++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 19 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 3e5c1eaccd5e..f3d3d20d35d7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -359,7 +359,14 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) if (sai->is_slave_mode) return 0; - for (id = 0; id < FSL_SAI_MCLK_MAX; id++) { + /* + * There is no point in polling MCLK0 if it is identical to MCLK1. + * And given that MQS use case has to use MCLK1 though two clocks + * are the same, we simply skip MCLK0 and start to find from MCLK1. + */ + id = sai->soc_data->mclk0_is_mclk1 ? 1 : 0; + + for (; id < FSL_SAI_MCLK_MAX; id++) { clk_rate = clk_get_rate(sai->mclk_clk[id]); if (!clk_rate) continue; @@ -1040,7 +1047,6 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - sai->mclk_clk[0] = sai->bus_clk; for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); @@ -1051,6 +1057,11 @@ static int fsl_sai_probe(struct platform_device *pdev) } } + if (sai->soc_data->mclk0_is_mclk1) + sai->mclk_clk[0] = sai->mclk_clk[1]; + else + sai->mclk_clk[0] = sai->bus_clk; + irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; @@ -1165,6 +1176,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_edma = false, .fifo_depth = 32, .reg_offset = 0, + .mclk0_is_mclk1 = false, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { @@ -1172,6 +1184,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_edma = false, .fifo_depth = 32, .reg_offset = 0, + .mclk0_is_mclk1 = true, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { @@ -1179,6 +1192,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_edma = false, .fifo_depth = 16, .reg_offset = 8, + .mclk0_is_mclk1 = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { @@ -1186,6 +1200,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_edma = false, .fifo_depth = 128, .reg_offset = 8, + .mclk0_is_mclk1 = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { @@ -1193,6 +1208,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_edma = true, .fifo_depth = 64, .reg_offset = 0, + .mclk0_is_mclk1 = false, }; static const struct of_device_id fsl_sai_ids[] = { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 4bbcd0dbe8f1..ff2619f1b214 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -219,6 +219,7 @@ struct fsl_sai_soc_data { bool use_imx_pcm; bool use_edma; + bool mclk0_is_mclk1; unsigned int fifo_depth; unsigned int reg_offset; }; -- 2.27.0

3 2

[PATCH v3] ASoC: sunxi: do not select COMMON_CLK to fix builds
by Krzysztof Kozlowski 19 Nov '20

19 Nov '20

COMMON_CLK is a user-selectable option with its own dependencies. The most important dependency is !HAVE_LEGACY_CLK. User-selectable drivers should not select COMMON_CLK because they will create a dependency cycle and build failures. For example on MIPS a configuration with COMMON_CLK (selected by SND_SUN8I_CODEC) and HAVE_LEGACY_CLK (selected by SOC_RT305X) is possible: WARNING: unmet direct dependencies detected for COMMON_CLK Depends on [n]: !HAVE_LEGACY_CLK [=y] Selected by [y]: - SND_SUN8I_CODEC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && (ARCH_SUNXI || COMPILE_TEST [=y]) && OF [=y] && (MACH_SUN8I || ARM64 && ARCH_SUNXI || COMPILE_TEST [=y]) /usr/bin/mips-linux-gnu-ld: drivers/clk/clk.o: in function `clk_set_rate': (.text+0xaeb4): multiple definition of `clk_set_rate'; arch/mips/ralink/clk.o:(.text+0x88): first defined here Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> Reviewed-by: Samuel Holland <samuel(a)sholland.org> Acked-by: Maxime Ripard <mripard(a)kernel.org> --- Changes since v2: 1. Fix title, 2. Add Maxime's ack. Changes since v1: 1. Do not select COMMON_CLK in arch/arm/mach-sunxi/Kconfig, 2. Add Samuel's review. --- sound/soc/sunxi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 69b9d8515335..ddcaaa98d3cb 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -14,7 +14,7 @@ config SND_SUN8I_CODEC tristate "Allwinner SUN8I audio codec" depends on OF depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select COMMON_CLK + depends on COMMON_CLK select REGMAP_MMIO help This option enables the digital part of the internal audio codec for -- 2.25.1

2 1

[PATCH v2] ASoC: Intel: catpt: select WANT_DEV_COREDUMP
by Gustaw Lewandowski 19 Nov '20

19 Nov '20

From: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> Select WANT_DEV_COREDUMP for catpt driver. Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> -- Changes in v2: - change should be added to catpt only --- sound/soc/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a5b446d5af19..52ae7c981f56 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -24,6 +24,7 @@ config SND_SOC_INTEL_CATPT depends on DMADEVICES && SND_DMA_SGBUF select DW_DMAC_CORE select SND_SOC_ACPI_INTEL_MATCH + select WANT_DEV_COREDUMP help Enable support for Intel(R) Haswell and Broadwell platforms with I2S codec present. This is a recommended option. -- 2.29.0

3 2

[PATCH] ASoC: pcm: call snd_soc_dapm_stream_stop() in soc_pcm_hw_clean
by Ranjani Sridharan 19 Nov '20

19 Nov '20

Currently, the SND_SOC_DAPM_STREAM_START event is sent during pcm_prepare() but the SND_SOC_DAPM_STREAM_STOP event is sent only in dpcm_fe_dai_shutdown() after soc_pcm_close(). This results in an imbalance between when the DAPM widgets receive the PRE/POST_PMU/PMD events. So call snd_soc_dapm_stream_stop() in soc_pcm_hw_clean() before the snd_soc_pcm_component_hw_free() to keep the stream_stop DAPM event balanced with the stream_start event in soc_pm_prepare(). Also, in order to prevent duplicate DAPM stream events, remove the call for DAPM STREAM_START event in dpcm_fe_dai_prepare() and the call for DAPM STREAM_STOP event in dpcm_fe_dai_shutdown(). Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-pcm.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aaab5cfe986b..5250124dc8f5 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -662,8 +662,6 @@ static int soc_pcm_clean(struct snd_pcm_substream *substream, int rollback) soc_pcm_components_close(substream, rollback); - if (!rollback) - snd_soc_dapm_stream_stop(rtd, substream->stream); mutex_unlock(&rtd->card->pcm_mutex); @@ -882,6 +880,9 @@ static int soc_pcm_hw_clean(struct snd_pcm_substream *substream, int rollback) snd_soc_dai_digital_mute(dai, 1, substream->stream); } + /* run the stream event */ + snd_soc_dapm_stream_stop(rtd, substream->stream); + /* free any machine hw params */ snd_soc_link_hw_free(substream, rollback); @@ -1859,9 +1860,6 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) /* now shutdown the frontend */ soc_pcm_close(substream); - /* run the stream event for each BE */ - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); - fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return 0; @@ -2363,8 +2361,6 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; } - /* run the stream event for each BE */ - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; out: -- 2.25.1

2 1

Better documentation with the examples
by GitHub issues - opened 19 Nov '20

19 Nov '20

alsa-project/tinycompress issue #8 was opened from Andres6936: I have compiled the program and when I run the `cplay` script I get an error: ```shellsession andres6936@Tierra ~/CLionProjects/tinycompress-master/src/utils $ cplay TestAudio.mp3 Error: Can't find sync word ``` I have already checked out different mp3 files and I keep getting the same error. I have also tried different options, but it is always the same error. Issue URL : https://github.com/alsa-project/tinycompress/issues/8 Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH] ALSA: mixart: Fix mutex deadlock
by Takashi Iwai 19 Nov '20

19 Nov '20

The code change for switching to non-atomic mode brought the unexpected mutex deadlock in get_msg(). It converted the spinlock with the existing mutex, but there were calls with the already holding the mutex. Since the only place that needs the extra lock is the code path from snd_mixart_send_msg(), remove the mutex lock in get_msg() and apply in the caller side for fixing the mutex deadlock. Fixes: 8d3a8b5cb57d ("ALSA: mixart: Use nonatomic PCM ops") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/mixart/mixart_core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 0bdd33b0af65..fb8895af0363 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -70,7 +70,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, unsigned int i; #endif - mutex_lock(&mgr->msg_lock); err = 0; /* copy message descriptor from miXart to driver */ @@ -119,8 +118,6 @@ static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, writel_be(headptr, MIXART_MEM(mgr, MSG_OUTBOUND_FREE_HEAD)); _clean_exit: - mutex_unlock(&mgr->msg_lock); - return err; } @@ -258,7 +255,9 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int resp.data = resp_data; resp.size = max_resp_size; + mutex_lock(&mgr->msg_lock); err = get_msg(mgr, &resp, msg_frame); + mutex_unlock(&mgr->msg_lock); if( request->message_id != resp.message_id ) dev_err(&mgr->pci->dev, "RESPONSE ERROR!\n"); -- 2.16.4

1 0

[PATCH] ALSA: hda/ca0132: Fix compile warning without PCI
by Takashi Iwai 19 Nov '20

19 Nov '20

CONFIG_PCI=n leads to a compile warning like: sound/pci/hda/patch_ca0132.c:8214:10: warning: no case matching constant switch condition '0' due to the missed handling of QUIRK_NONE in ca0132_mmio_init(). Fix it. Fixes: bf2aa9ccc8e5 ("ALSA: hda/ca0132 - Cleanup ca0132_mmio_init function.") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index e0c38f2735c6..d8370a417e3d 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -9183,6 +9183,8 @@ static void ca0132_mmio_init(struct hda_codec *codec) case QUIRK_AE5: ca0132_mmio_init_ae5(codec); break; + default: + break; } } -- 2.16.4

1 0

[PATCH] ASoC: fsl_sai: Correct the clock source for mclk0
by Shengjiu Wang 19 Nov '20

19 Nov '20

On VF610, mclk0 = bus_clk; On i.MX6SX/6UL/6ULL/7D, mclk0 = mclk1; On i.MX7ULP, mclk0 = bus_clk; On i.MX8QM/8QXP, mclk0 = bus_clk; On i.MX8MQ/8MN/8MM/8MP, mclk0 = bus_clk; So add variable mclk0_mclk1_match in fsl_sai_soc_data to distinguish these platforms. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 11 ++++++++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 3e5c1eaccd5e..479fd27ace35 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1040,7 +1040,6 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - sai->mclk_clk[0] = sai->bus_clk; for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); @@ -1051,6 +1050,11 @@ static int fsl_sai_probe(struct platform_device *pdev) } } + if (sai->soc_data->mclk0_mclk1_match) + sai->mclk_clk[0] = sai->mclk_clk[1]; + else + sai->mclk_clk[0] = sai->bus_clk; + irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; @@ -1165,6 +1169,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_edma = false, .fifo_depth = 32, .reg_offset = 0, + .mclk0_mclk1_match = false, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { @@ -1172,6 +1177,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_edma = false, .fifo_depth = 32, .reg_offset = 0, + .mclk0_mclk1_match = true, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { @@ -1179,6 +1185,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_edma = false, .fifo_depth = 16, .reg_offset = 8, + .mclk0_mclk1_match = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { @@ -1186,6 +1193,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_edma = false, .fifo_depth = 128, .reg_offset = 8, + .mclk0_mclk1_match = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { @@ -1193,6 +1201,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_edma = true, .fifo_depth = 64, .reg_offset = 0, + .mclk0_mclk1_match = false, }; static const struct of_device_id fsl_sai_ids[] = { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 4bbcd0dbe8f1..390a9ca3b531 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -219,6 +219,7 @@ struct fsl_sai_soc_data { bool use_imx_pcm; bool use_edma; + bool mclk0_mclk1_match; unsigned int fifo_depth; unsigned int reg_offset; }; -- 2.27.0

3 4

ESS Sabre DAC ES901x Codec drivers
by Matt Flax 19 Nov '20

19 Nov '20

Are there any ESS DAC codec drivers out there ? For example ES9012, ES9018 DACs. http://www.esstech.com/files/2816/0453/7571/ES9018S_Datasheet_v2.2.pdf Matt

1 0

[PATCH v2 00/12] ASoC: soc-component: add snd_soc_component_xxx()
by Kuninori Morimoto 18 Nov '20

18 Nov '20

Hi Mark These are v2 of snd_soc_component_compr_xxx() function patches. I think component related function should be implemented at soc-component.c, otherwise it is confusable to read. These are for it. v1 -> v2 - fixup function return timing on snd_soc_component_compr_get_params() snd_soc_component_compr_get_metadata() - use mutex at compr side on snd_soc_component_compr_get_caps() snd_soc_component_compr_get_codec_caps() snd_soc_component_compr_copy() Link: https://lore.kernel.org/r/878sb78ac4.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (12): ASoC: soc-component: add snd_soc_component_compr_open() ASoC: soc-component: add snd_soc_component_compr_free() ASoC: soc-component: add snd_soc_component_compr_trigger() ASoC: soc-component: add snd_soc_component_compr_set_params() ASoC: soc-component: add snd_soc_component_compr_get_params() ASoC: soc-component: add snd_soc_component_compr_get_caps() ASoC: soc-component: add snd_soc_component_compr_get_codec_caps() ASoC: soc-component: add snd_soc_component_compr_ack() ASoC: soc-component: add snd_soc_component_compr_pointer() ASoC: soc-component: add snd_soc_component_compr_copy() ASoC: soc-component: add snd_soc_component_compr_set_metadata() ASoC: soc-component: add snd_soc_component_compr_get_metadata() include/sound/soc-component.h | 22 +++ sound/soc/soc-component.c | 255 ++++++++++++++++++++++++++++++++ sound/soc/soc-compress.c | 263 ++++------------------------------ 3 files changed, 303 insertions(+), 237 deletions(-) -- 2.25.1

3 14

[PATCH] ASOC: Intel: kbl_rt5663_rt5514_max98927: Do not try to disable disabled clock
by Guenter Roeck 18 Nov '20

18 Nov '20

In kabylake_set_bias_level(), enabling mclk may fail if the clock has already been enabled by the firmware. Attempts to disable that clock later will fail with a warning backtrace. mclk already disabled WARNING: CPU: 2 PID: 108 at drivers/clk/clk.c:952 clk_core_disable+0x1b6/0x1cf ... Call Trace: clk_disable+0x2d/0x3a kabylake_set_bias_level+0x72/0xfd [snd_soc_kbl_rt5663_rt5514_max98927] snd_soc_card_set_bias_level+0x2b/0x6f snd_soc_dapm_set_bias_level+0xe1/0x209 dapm_pre_sequence_async+0x63/0x96 async_run_entry_fn+0x3d/0xd1 process_one_work+0x2a9/0x526 ... Only disable the clock if it has been enabled. Fixes: 15747a802075 ("ASoC: eve: implement set_bias_level function for rt5514") Cc: Brent Lu <brent.lu(a)intel.com> Cc: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 922cd0176e1f..f95546c184aa 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -700,6 +700,8 @@ static int kabylake_set_bias_level(struct snd_soc_card *card, switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_ON) { + if (!__clk_is_enabled(priv->mclk)) + return 0; dev_dbg(card->dev, "Disable mclk"); clk_disable_unprepare(priv->mclk); } else { -- 2.17.1

2 1

[PATCH -next] ASoC: samsung: fix platform_no_drv_owner.cocci warnings
by Zou Wei 18 Nov '20

18 Nov '20

./sound/soc/samsung/midas_wm1811.c:534:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/samsung/midas_wm1811.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index d03340c..1f9a553 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -531,7 +531,6 @@ static struct platform_driver midas_driver = { .driver = { .name = "midas-audio", .of_match_table = midas_of_match, - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, }, .probe = midas_probe, -- 2.6.2

3 2

[PATCH] dt-bindings: mediatek: mt8192: revert "add audio afe document"
by Jiaxin Yu 18 Nov '20

18 Nov '20

This reverts commit 1afc60e00de3abbb6c559da409c5c1bb8c1d98ec. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… These dependency are continuing to upstream, I will revert this patch first. After the dependencies are merged, I will send 'mt8192-afe-pcm.yaml' again. .../bindings/sound/mt8192-afe-pcm.yaml | 100 ------------------ 1 file changed, 100 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml deleted file mode 100644 index 0170c93802d29..0000000000000 --- a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml +++ /dev/null @@ -1,100 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# -$schema: http://devicetree.org/meta-schemas/core.yaml# - -title: Mediatek AFE PCM controller for mt8192 - -maintainers: - - Jiaxin Yu <jiaxin.yu(a)mediatek.com> - - Shane Chien <shane.chien(a)mediatek.com> - -properties: - compatible: - const: mediatek,mt8192-audio - - interrupts: - maxItems: 1 - - resets: - maxItems: 1 - - reset-names: - const: audiosys - - mediatek,apmixedsys: - $ref: "/schemas/types.yaml#/definitions/phandle" - description: The phandle of the mediatek apmixedsys controller - - mediatek,infracfg: - $ref: "/schemas/types.yaml#/definitions/phandle" - description: The phandle of the mediatek infracfg controller - - mediatek,topckgen: - $ref: "/schemas/types.yaml#/definitions/phandle" - description: The phandle of the mediatek topckgen controller - - power-domains: - maxItems: 1 - - clocks: - items: - - description: AFE clock - - description: ADDA DAC clock - - description: ADDA DAC pre-distortion clock - - description: audio infra sys clock - - description: audio infra 26M clock - - clock-names: - items: - - const: aud_afe_clk - - const: aud_dac_clk - - const: aud_dac_predis_clk - - const: aud_infra_clk - - const: aud_infra_26m_clk - -required: - - compatible - - interrupts - - resets - - reset-names - - mediatek,apmixedsys - - mediatek,infracfg - - mediatek,topckgen - - power-domains - - clocks - - clock-names - -additionalProperties: false - -examples: - - | - #include <dt-bindings/clock/mt8192-clk.h> - #include <dt-bindings/interrupt-controller/arm-gic.h> - #include <dt-bindings/interrupt-controller/irq.h> - #include <dt-bindings/power/mt8192-power.h> - #include <dt-bindings/reset-controller/mt8192-resets.h> - - afe: mt8192-afe-pcm { - compatible = "mediatek,mt8192-audio"; - interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; - resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; - reset-names = "audiosys"; - mediatek,apmixedsys = <&apmixedsys>; - mediatek,infracfg = <&infracfg>; - mediatek,topckgen = <&topckgen>; - power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; - clocks = <&audsys CLK_AUD_AFE>, - <&audsys CLK_AUD_DAC>, - <&audsys CLK_AUD_DAC_PREDIS>, - <&infracfg CLK_INFRA_AUDIO>, - <&infracfg CLK_INFRA_AUDIO_26M_B>; - clock-names = "aud_afe_clk", - "aud_dac_clk", - "aud_dac_predis_clk", - "aud_infra_clk", - "aud_infra_26m_clk"; - }; - -... -- 2.18.0

3 3

[PATCH] ASoC: mmp-sspa: clear transmit phase bit for non-stereo formats
by Kyle Russell 18 Nov '20

18 Nov '20

The transmit phase register value is never cleared during hw params. So once hw params sets this bit to handle a two channel format, it remains configured for dual-phase, which is not desirable for mono playback. Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> --- sound/soc/pxa/mmp-sspa.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 4255851c71c1..52d4d8ace1c3 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -239,6 +239,7 @@ static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + sspa_ctrl &= ~SSPA_CTL_XPH; if (dev->of_node || params_channels(params) == 2) sspa_ctrl |= SSPA_CTL_XPH; -- 2.25.1

2 1

[PATCH v2] ARM: sunxi: do not select COMMON_CLK to fix builds
by Krzysztof Kozlowski 18 Nov '20

18 Nov '20

COMMON_CLK is a user-selectable option with its own dependencies. The most important dependency is !HAVE_LEGACY_CLK. User-selectable drivers should not select COMMON_CLK because they will create a dependency cycle and build failures. For example on MIPS a configuration with COMMON_CLK (selected by SND_SUN8I_CODEC) and HAVE_LEGACY_CLK (selected by SOC_RT305X) is possible: WARNING: unmet direct dependencies detected for COMMON_CLK Depends on [n]: !HAVE_LEGACY_CLK [=y] Selected by [y]: - SND_SUN8I_CODEC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && (ARCH_SUNXI || COMPILE_TEST [=y]) && OF [=y] && (MACH_SUN8I || ARM64 && ARCH_SUNXI || COMPILE_TEST [=y]) /usr/bin/mips-linux-gnu-ld: drivers/clk/clk.o: in function `clk_set_rate': (.text+0xaeb4): multiple definition of `clk_set_rate'; arch/mips/ralink/clk.o:(.text+0x88): first defined here Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> Reviewed-by: Samuel Holland <samuel(a)sholland.org> --- Changes since v1: 1. Do not select COMMON_CLK in arch/arm/mach-sunxi/Kconfig, 2. Add Samuel's review. --- sound/soc/sunxi/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index 69b9d8515335..ddcaaa98d3cb 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -14,7 +14,7 @@ config SND_SUN8I_CODEC tristate "Allwinner SUN8I audio codec" depends on OF depends on MACH_SUN8I || (ARM64 && ARCH_SUNXI) || COMPILE_TEST - select COMMON_CLK + depends on COMMON_CLK select REGMAP_MMIO help This option enables the digital part of the internal audio codec for -- 2.25.1

2 2

[PATCH 0/2] slimbus: qcom-ngd: Add SSR and PDR support
by Srinivas Kandagatla 18 Nov '20

18 Nov '20

Qualcomm DSPs support SSR(Subsystem Restart) and PDR(Protection Domain Restart) which usually restart the associated services! Current code does only relies on QMI channel notifications, but there could be race conditions between QMI notifcations and actual PDR/SSR events. This patchset adds support to these two events to address those race conditions. Tested it on Dragon Board DB845c. Srinivas Kandagatla (2): slimbus: qcom-ngd-ctrl: add Sub System Restart support slimbus: qcom-ngd-ctrl: add Protection Domain Restart Support drivers/slimbus/Kconfig | 2 + drivers/slimbus/qcom-ngd-ctrl.c | 121 ++++++++++++++++++++++++++++++-- 2 files changed, 119 insertions(+), 4 deletions(-) -- 2.21.0

1 2

[bug report] ALSA: mixart: Use nonatomic PCM ops
by Dan Carpenter 18 Nov '20

18 Nov '20

[ This bug predates git but my email script just uses `git blame -w`. -dan ] Hello Takashi Iwai, The patch 8d3a8b5cb57d: "ALSA: mixart: Use nonatomic PCM ops" from Sep 10, 2014, leads to the following static checker warning: sound/pci/mixart/mixart_core.c:73 get_msg() warn: called with lock held. '&mgr->msg_lock' sound/pci/mixart/mixart_core.c 63 static int get_msg(struct mixart_mgr *mgr, struct mixart_msg *resp, 64 u32 msg_frame_address ) 65 { 66 u32 headptr; 67 u32 size; 68 int err; 69 #ifndef __BIG_ENDIAN 70 unsigned int i; 71 #endif 72 73 mutex_lock(&mgr->msg_lock); ^^^^^^^^^^^^^^^^^^^^^^^^^^ Smatch says that get_msg() is called with this lock already held. 74 err = 0; 75 76 /* copy message descriptor from miXart to driver */ 77 size = readl_be(MIXART_MEM(mgr, msg_frame_address)); /* size of descriptor + response */ 78 resp->message_id = readl_be(MIXART_MEM(mgr, msg_frame_address + 4)); /* dwMessageID */ [ snip ] 419 irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) 420 { 421 struct mixart_mgr *mgr = dev_id; 422 int err; 423 struct mixart_msg resp; 424 u32 msg; 425 426 mutex_lock(&mgr->lock); ^^^^^^^^^^^^^^^^^^^^^^ We are holding the lock 427 /* process interrupt */ 428 while (retrieve_msg_frame(mgr, &msg)) { 429 430 switch (msg & MSG_TYPE_MASK) { 431 case MSG_TYPE_COMMAND: 432 resp.message_id = 0; 433 resp.data = mixart_msg_data; 434 resp.size = sizeof(mixart_msg_data); 435 err = get_msg(mgr, &resp, msg & ~MSG_TYPE_MASK); ^^^^^^^^^^^ When get_msg() is called on this path. 436 if( err < 0 ) { 437 dev_err(&mgr->pci->dev, 438 "interrupt: error(%d) reading mf %x\n", 439 err, msg); 440 break; 441 } 442 443 if(resp.message_id == MSG_SERVICES_TIMER_NOTIFY) { regards, dan carpenter

1 0

snd_pcm_hw_params_set_period_size_near() bug???
by GitHub issues - edited 18 Nov '20

18 Nov '20

alsa-project/alsa-lib issue #99 was edited from YangangCao: it's my fault, no big deal Issue URL : https://github.com/alsa-project/alsa-lib/issues/99 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/3] Convert Tegra HDA doc to json-schema
by Sameer Pujar 18 Nov '20

18 Nov '20

Fix Tegra194 DT and convert Tegra HDA doc to YAML format. Ran binding validation checks to ensure there are no warnings/errors for HDA. Sameer Pujar (3): arm64: tegra: Fix Tegra194 HDA {clock,reset}-names ordering dt-bindings: tegra: Convert HDA doc to json-schema dt-bindings: tegra: Add missing HDA properties .../bindings/sound/nvidia,tegra30-hda.txt | 35 ------- .../bindings/sound/nvidia,tegra30-hda.yaml | 112 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra194.dtsi | 12 +-- 3 files changed, 118 insertions(+), 41 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml -- 2.7.4

1 3

[PATCH] ALSA: usb-audio: disable 96khz support for HUAWEI USB-C HEADSET
by Macpaul Lin 18 Nov '20

18 Nov '20

The HUAWEI USB-C headset (VID:0x12d1, PID:0x3a07) reported it supports 96khz. However there will be some random issue under 96khz. Not sure if there is any alternate setting could be applied. Hence 48khz is suggested to be applied at this moment. Signed-off-by: Macpaul Lin <macpaul.lin(a)mediatek.com> Signed-off-by: Eddie Hung <eddie.hung(a)mediatek.com> --- sound/usb/format.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index 1b28d01..6f6e79b 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -217,6 +217,11 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof (chip->usb_id == USB_ID(0x041e, 0x4064) || chip->usb_id == USB_ID(0x041e, 0x4068))) rate = 8000; + /* Huawei headset can't support 96kHz fully */ + if (rate == 96000 && + chip->usb_id == USB_ID(0x12d1, 0x3a07) && + le16_to_cpu(udev->descriptor.bcdDevice) == 0x49) + continue; fp->rate_table[fp->nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) -- 1.7.9.5

3 9

[PATCH] ASoC: intel: SND_SOC_INTEL_KEEMBAY should depend on ARCH_KEEMBAY
by Geert Uytterhoeven 18 Nov '20

18 Nov '20

The Intel Keem Bay audio module is only present on Intel Keem Bay SoCs. Hence add a dependency on ARCH_KEEMBAY, to prevent asking the user about this driver when configuring a kernel without Intel Keem Bay platform support. Fixes: c544912bcc2dc806 ("ASoC: Intel: Add makefiles and kconfig changes for KeemBay") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a5b446d5af19fc2d..c1bf69a0bcfe18c3 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -198,7 +198,7 @@ endif ## SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL config SND_SOC_INTEL_KEEMBAY tristate "Keembay Platforms" - depends on ARM64 || COMPILE_TEST + depends on ARCH_KEEMBAY || COMPILE_TEST depends on COMMON_CLK help If you have a Intel Keembay platform then enable this option -- 2.25.1

2 1

[PATCH] ASoc: adi: Kconfig: Remove depends on for ADI reference designs
by Alexandru Ardelean 18 Nov '20

18 Nov '20

From: Bogdan Togorean <bogdan.togorean(a)analog.com> Audio ADI reference designs are also used on some ZynqMP boards, and can also be used on Intel FPGA boards and also on some more complex FPGA combinations (FPGA cards connected through PCIe). This change removes the dependency on Microblaze and Zynq architectures to allow the usage of this driver for the systems described above. Signed-off-by: Bogdan Togorean <bogdan.togorean(a)analog.com> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/adi/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/adi/Kconfig b/sound/soc/adi/Kconfig index e321e3b672da..0236dc5b4e9f 100644 --- a/sound/soc/adi/Kconfig +++ b/sound/soc/adi/Kconfig @@ -1,7 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_SOC_ADI tristate "Audio support for Analog Devices reference designs" - depends on MICROBLAZE || ARCH_ZYNQ || COMPILE_TEST help Audio support for various reference designs by Analog Devices. -- 2.17.1

2 1

[PATCH] dt-bindings: adau1977: convert text binding to yaml format
by Alexandru Ardelean 18 Nov '20

18 Nov '20

This change converts the old device-tree binding for ADAU1977 from text format to the new yaml format. Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- .../bindings/sound/adi,adau1977.txt | 61 --------------- .../bindings/sound/adi,adau1977.yaml | 77 +++++++++++++++++++ 2 files changed, 77 insertions(+), 61 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.txt create mode 100644 Documentation/devicetree/bindings/sound/adi,adau1977.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.txt b/Documentation/devicetree/bindings/sound/adi,adau1977.txt deleted file mode 100644 index 37f8aad01203..000000000000 --- a/Documentation/devicetree/bindings/sound/adi,adau1977.txt +++ /dev/null @@ -1,61 +0,0 @@ -Analog Devices ADAU1977/ADAU1978/ADAU1979 - -Datasheets: -https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1977.pdf -https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1978.pdf -https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU1979.pdf - -This driver supports both the I2C and SPI bus. - -Required properties: - - compatible: Should contain one of the following: - "adi,adau1977" - "adi,adau1978" - "adi,adau1979" - - - AVDD-supply: analog power supply for the device, please consult - Documentation/devicetree/bindings/regulator/regulator.txt - -Optional properties: - - reset-gpios: the reset pin for the chip, for more details consult - Documentation/devicetree/bindings/gpio/gpio.txt - - - DVDD-supply: supply voltage for the digital core, please consult - Documentation/devicetree/bindings/regulator/regulator.txt - -- adi,micbias: configures the voltage setting for the MICBIAS pin. - Select 0/1/2/3/4/5/6/7/8 to specify MICBIAS voltage - 5V/5.5V/6V/6.5V/7V/7.5V/8V/8.5V/9V - If not specified the default value will be "7" meaning 8.5 Volts. - This property is only valid for the ADAU1977 - -For required properties on SPI, please consult -Documentation/devicetree/bindings/spi/spi-bus.txt - -Required properties on I2C: - - - reg: The i2c address. Value depends on the state of ADDR0 - and ADDR1, as wired in hardware. - -Examples: - - adau1977_spi: adau1977@0 { - compatible = "adi,adau1977"; - spi-max-frequency = <600000>; - - AVDD-supply = <&regulator>; - DVDD-supply = <&regulator_digital>; - - adi,micbias = <3>; - reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; - }; - - adau1977_i2c: adau1977@11 { - compatible = "adi,adau1977"; - reg = <0x11>; - - AVDD-supply = <&regulator>; - DVDD-supply = <&regulator_digital>; - - reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/adi,adau1977.yaml b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml new file mode 100644 index 000000000000..bc097b6b066e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau1977.yaml @@ -0,0 +1,77 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,adau1977.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices ADAU1977/ADAU1978/ADAU1979 Quad ADC with Diagnostics + +maintainers: + - Lars-Peter Clausen <lars(a)metafoo.de> + - Bogdan Togorean <bogdan.togorean(a)analog.com> + +description: | + Analog Devices ADAU1977 and similar quad ADC with Diagnostics + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU197… + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU197… + https://www.analog.com/media/en/technical-documentation/data-sheets/ADAU197… + +properties: + compatible: + enum: + - adi,adau1977 + - adi,adau1978 + - adi,adau1979 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + reset-gpios: + maxItems: 1 + + AVDD-supply: + description: Analog power support for the device. + + DVDD-supply: + description: Supply voltage for digital core. + + adi,micbias: + description: | + Configures the voltage setting for the MICBIAS pin. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3, 4, 5, 6, 7, 8] + default: 7 + +required: + - reg + - compatible + - AVDD-supply + +additionalProperties: false + +examples: + - | + adau1977_spi: adau1977@0 { + compatible = "adi,adau1977"; + reg = <0>; + spi-max-frequency = <600000>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + adi,micbias = <3>; + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; + - | + adau1977_i2c: adau1977@11 { + compatible = "adi,adau1977"; + reg = <0x11>; + + AVDD-supply = <&regulator>; + DVDD-supply = <&regulator_digital>; + + reset-gpios = <&gpio 10 GPIO_ACTIVE_LOW>; + }; -- 2.27.0

4 6

[PATCH] ASoC: Fix 7/8 spaces indentation in Kconfig
by Geert Uytterhoeven 18 Nov '20

18 Nov '20

Some entries used 7 or 8 spaces instead if a single TAB. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- sound/soc/codecs/Kconfig | 18 +++++++++--------- sound/soc/generic/Kconfig | 2 +- sound/soc/intel/boards/Kconfig | 2 +- sound/soc/meson/Kconfig | 2 +- sound/soc/pxa/Kconfig | 14 +++++++------- 5 files changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 34c6dd04b85a3c9e..384c5de0107e4077 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -517,7 +517,7 @@ config SND_SOC_AK5558 select REGMAP_I2C config SND_SOC_ALC5623 - tristate "Realtek ALC5623 CODEC" + tristate "Realtek ALC5623 CODEC" depends on I2C config SND_SOC_ALC5632 @@ -728,7 +728,7 @@ config SND_SOC_JZ4770_CODEC will be called snd-soc-jz4770-codec. config SND_SOC_L3 - tristate + tristate config SND_SOC_DA7210 tristate @@ -768,10 +768,10 @@ config SND_SOC_HDMI_CODEC select HDMI config SND_SOC_ES7134 - tristate "Everest Semi ES7134 CODEC" + tristate "Everest Semi ES7134 CODEC" config SND_SOC_ES7241 - tristate "Everest Semi ES7241 CODEC" + tristate "Everest Semi ES7241 CODEC" config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" @@ -970,10 +970,10 @@ config SND_SOC_PCM186X_SPI select REGMAP_SPI config SND_SOC_PCM3008 - tristate + tristate config SND_SOC_PCM3060 - tristate + tristate config SND_SOC_PCM3060_I2C tristate "Texas Instruments PCM3060 CODEC - I2C" @@ -1436,7 +1436,7 @@ config SND_SOC_UDA1334 rate) and mute. config SND_SOC_UDA134X - tristate + tristate config SND_SOC_UDA1380 tristate @@ -1761,8 +1761,8 @@ config SND_SOC_MT6660 Select M to build this as module. config SND_SOC_NAU8540 - tristate "Nuvoton Technology Corporation NAU85L40 CODEC" - depends on I2C + tristate "Nuvoton Technology Corporation NAU85L40 CODEC" + depends on I2C config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index a90c3b28bce5fcac..4cafcf0e2bbfd38d 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_SIMPLE_CARD_UTILS - tristate + tristate config SND_SIMPLE_CARD tristate "ASoC Simple sound card support" diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index c10c37803c670b6b..dddb672a6d553357 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -552,7 +552,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_RT715_SDCA_SDW select SND_SOC_RT5682_SDW select SND_SOC_DMIC - help + help Add support for Intel SoundWire-based platforms connected to MAX98373, RT700, RT711, RT1308 and RT715 If unsure select "N". diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 363dc3b1bbe47032..917fce7fa3cb7bde 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -98,7 +98,7 @@ config SND_MESON_AXG_PDM in the Amlogic AXG SoC family config SND_MESON_CARD_UTILS - tristate + tristate config SND_MESON_CODEC_GLUE tristate diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 0ac85eada75cb1b7..9d40e8a206d104aa 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -221,13 +221,13 @@ config SND_PXA2XX_SOC_MIOA701 MIO A701. config SND_PXA2XX_SOC_IMOTE2 - tristate "SoC Audio support for IMote 2" - depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C - select SND_PXA2XX_SOC_I2S - select SND_SOC_WM8940 - help - Say Y if you want to add support for SoC audio on the - IMote 2. + tristate "SoC Audio support for IMote 2" + depends on SND_PXA2XX_SOC && MACH_INTELMOTE2 && I2C + select SND_PXA2XX_SOC_I2S + select SND_SOC_WM8940 + help + Say Y if you want to add support for SoC audio on the + IMote 2. config SND_MMP_SOC_BROWNSTONE tristate "SoC Audio support for Marvell Brownstone" -- 2.25.1

2 1

[PATCH] ASoC: fsl: SND_SOC_FSL_AUD2HTX should depend on ARCH_MXC
by Geert Uytterhoeven 18 Nov '20

18 Nov '20

The Freescale/NXP AUDIO TO HDMI TX module is only present on NXP i.MX 8 Series SoCs. Hence add a dependency on ARCH_MXC, to prevent asking the user about this driver when configuring a kernel without i.MX 8 platform support. Fixes: 8a24c834c053ef1b ("ASoC: fsl_aud2htx: Add aud2htx module driver") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 52a5622150088b49..ad9f90d314b7a033 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -107,6 +107,7 @@ config SND_SOC_FSL_XCVR config SND_SOC_FSL_AUD2HTX tristate "AUDIO TO HDMI TX module support" + depends on CONFIG_ARCH_MXC || COMPILE_TEST help Say Y if you want to add AUDIO TO HDMI TX support for NXP. -- 2.25.1

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[PATCH v4 0/7] Enable rk3066a HDMI sound
by Johan Jonker 17 Nov '20

17 Nov '20

First fix some legacy things in clk-rk3188.c that was never updated, because probably nobody used rk3066a I2S before in the mainline kernel. Update the rk3066a HDMI documents with a #sound-dai-cells property. Include the code for sound in the HDMI driver. Add a simple-sound-card compatible node to rk3066a.dtsi, because I2S0 and HDMI TX are connected internally. And as last enable rk3066a HDMI sound in the rk3066a-mk808.dts file. Johan Jonker (6): clk: rockchip: add CLK_SET_RATE_PARENT to sclk for rk3066a i2s and uart clocks clk: rockchip: fix i2s gate bits on rk3066 and rk3188 dt-bindings: display: add #sound-dai-cells property to rockchip rk3066 hdmi ARM: dts: rockchip: rk3066a: add #sound-dai-cells to hdmi node ARM: dts: rockchip: add hdmi-sound node to rk3066a.dtsi ARM: dts: rockchip: enable hdmi_sound and i2s0 for rk3066a-mk808 Zheng Yang (1): drm: rockchip: add sound support to rk3066 hdmi driver .../display/rockchip/rockchip,rk3066-hdmi.yaml | 4 + arch/arm/boot/dts/rk3066a-mk808.dts | 8 + arch/arm/boot/dts/rk3066a.dtsi | 17 ++ drivers/clk/rockchip/clk-rk3188.c | 35 +-- drivers/gpu/drm/rockchip/Kconfig | 2 + drivers/gpu/drm/rockchip/rk3066_hdmi.c | 272 ++++++++++++++++++++- 6 files changed, 320 insertions(+), 18 deletions(-) -- 2.11.0

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[PATCH v4 00/10] Auxiliary bus implementation and SOF multi-client support
by Dave Ertman 17 Nov '20

17 Nov '20

Brief history of Auxiliary Bus ============================== The auxiliary bus code was originally submitted upstream as virtual bus, and was submitted through the netdev tree.  This process generated up to v4.  This discussion can be found here: https://lore.kernel.org/netdev/20191111192219.30259-1-jeffrey.t.kirsher@int… At this point, GregKH requested that we take the review and revision process to an internal mailing list and garner the buy-in of a respected kernel contributor. The auxiliary bus (then known as virtual bus) was originally submitted along with implementation code for the ice driver and irdma drive, causing the complication of also having dependencies in the rdma tree. This new submission is utilizing an auxiliary bus consumer in only the sound driver tree to create the initial implementation and a single user. Since implementation work has started on this patch set, there have been multiple inquiries about the time frame of its completion. It appears that there will be numerous consumers of this functionality. The process of internal review and implementation using the sound drivers generated 19 internal versions.  The changes, including the name change from virtual bus to auxiliary bus, from these versions can be summarized as the following: - Fixed compilation and checkpatch errors - Improved documentation to address the motivation for virtual bus. - Renamed virtual bus to auxiliary bus - increased maximum device name size - Correct order in Kconfig and Makefile - removed the mid-layer adev->release layer for device unregister - pushed adev->id management to parent driver - all error paths out of ancillary_device_register return error code - all error paths out of ancillary_device_register use put_device - added adev->name element - modname in register cannot be NULL - added KBUILD_MODNAME as prefix for match_name - push adev->id responsibility to registering driver - uevent now parses adev->dev name - match_id function now parses adev->dev name - changed drivers probe function to also take an ancillary_device_id param - split ancillary_device_register into device_initialize and device_add - adjusted what is done in device_initialize and device_add - change adev to ancildev and adrv to ancildrv - change adev to ancildev in documentation ========================== Introduces the auxiliary bus implementation along with the example usage in the Sound Open Firmware(SOF) audio driver. In some subsystems, the functionality of the core device (PCI/ACPI/other) may be too complex for a single device to be managed as a monolithic block or a part of the functionality might need to be exposed to a different subsystem. Splitting the functionality into smaller orthogonal devices makes it easier to manage data, power management and domain-specific communication with the hardware. Also, common auxiliary_device functionality across primary devices can be handled by a common auxiliary_device. A key requirement for such a split is that there is no dependency on a physical bus, device, register accesses or regmap support. These individual devices split from the core cannot live on the platform bus as they are not physical devices that are controlled by DT/ACPI. The same argument applies for not using MFD in this scenario as it relies on individual function devices being physical devices that are DT enumerated. An example for this kind of requirement is the audio subsystem where a single IP handles multiple entities such as HDMI, Soundwire, local devices such as mics/speakers etc. The split for the core's functionality can be arbitrary or be defined by the DSP firmware topology and include hooks for test/debug. This allows for the audio core device to be minimal and tightly coupled with handling the hardware-specific logic and communication. The auxiliary bus is intended to be minimal, generic and avoid domain-specific assumptions. Each auxiliary bus device represents a part of its parent functionality. The generic behavior can be extended and specialized as needed by encapsulating an auxiliary bus device within other domain-specific structures and the use of .ops callbacks. The SOF driver adopts the auxiliary bus for implementing the multi-client support. A client in the context of the SOF driver represents a part of the core device's functionality. It is not a physical device but rather an auxiliary device that needs to communicate with the DSP via IPCs. With multi-client support,the sound card can be separated into multiple orthogonal auxiliary devices for local devices (mic/speakers etc), HDMI, sensing, probes, debug etc. In this series, we demonstrate the usage of the auxiliary bus with the help of the IPC test client which is used for testing the serialization of IPCs when multiple clients talk to the DSP at the same time. v4 changes: clean up passive voice clarify reg/unreg path for devices explicitly call out GPL v2 v3 changes: rename to auxiliary bus move .c file to drivers/base/ split auxdev unregister flow into uninitialize and delete steps update kernel-doc on register functions for new unregister model update documentation with new name and unregister flow remove check for release in auxiliary bus, allow core to catch Change driver register so only probe and id_table mandatory Fix matching logic in auxillary_match_id to account for longer id->name utilize auxiliary_bus callbacks for probe, remove and shutdown add auxiliary_find_device function add code to ensure unique auxdrv name shorten initialize/uninitialize in function names to init/uninit simplify looping logic in match_id in drives/base/Kconfig changed from tristate to bool Modified signature SOF client register/unregister API Modified PM runtime enabling sequence in the SOF ipc test aux driver Removed driver.name from the aux driver Added Probes client aux driver and device registration in the SOF driver. This allows for enabling the probes functionality in the SOF firmware for audio data extraction from specific points in the audio pipeline. Without auxiliary bus, the implementation requires modifying existing 15+ machine drivers to add the probes DAI links to the sounds cards. Using the auxiliary bus allows for splitting the probes implementation from the SOF core making it easy to maintain. v2 changes: defined pr_fmt for kernel messages replaced WARN_ON calls in registration with pr_err calls adding kernel-doc function comments for device_initialize and device_add fix typo in documentation removed inaccurate line in documentation fixed formatting in drivers/bus/Makefile changed unwind path for sof_client_dev_alloc() improved comments for client list and mem freeing during client unreg removed debugfs entries in sof_ipc_test_client_drv during remove changed the signature of sof_debug_ipc_flood_test() fix a looping error in ancillary_match_id updated error value in sof_client_dev_register() mutex held while traversing client list when unregistering clients updated includes in sof-client.h Dave Ertman (1): Add auxiliary bus support Ranjani Sridharan (9): ASoC: SOF: Introduce descriptors for SOF client ASoC: SOF: Create client driver for IPC test ASoC: SOF: ops: Add ops for client registration ASoC: SOF: Intel: Define ops for client registration ASoC: SOF: Intel: Remove IPC flood test support in SOF core ASoC: SOF: sof-client: Add client APIs to access probes ops ASoC: SOF: compress: move and export sof_probe_compr_ops ASoC: SOF: Add new client driver for probes support ASoC: SOF: Intel: CNL: register probes client Documentation/driver-api/auxiliary_bus.rst | 234 +++++++++++ Documentation/driver-api/index.rst | 1 + drivers/base/Kconfig | 3 + drivers/base/Makefile | 1 + drivers/base/auxiliary.c | 268 ++++++++++++ include/linux/auxiliary_bus.h | 78 ++++ include/linux/mod_devicetable.h | 8 + scripts/mod/devicetable-offsets.c | 3 + scripts/mod/file2alias.c | 8 + sound/soc/sof/Kconfig | 47 ++- sound/soc/sof/Makefile | 10 +- sound/soc/sof/compress.c | 60 +-- sound/soc/sof/compress.h | 1 + sound/soc/sof/core.c | 18 +- sound/soc/sof/debug.c | 457 --------------------- sound/soc/sof/intel/Kconfig | 9 + sound/soc/sof/intel/Makefile | 3 + sound/soc/sof/intel/apl.c | 16 + sound/soc/sof/intel/bdw.c | 16 + sound/soc/sof/intel/byt.c | 20 + sound/soc/sof/intel/cnl.c | 32 ++ sound/soc/sof/intel/hda-dai.c | 27 -- sound/soc/sof/intel/hda.h | 6 - sound/soc/sof/intel/intel-client.c | 40 ++ sound/soc/sof/intel/intel-client.h | 26 ++ sound/soc/sof/ops.h | 14 + sound/soc/sof/pcm.c | 11 - sound/soc/sof/probe.c | 124 +++--- sound/soc/sof/probe.h | 41 +- sound/soc/sof/sof-client.c | 170 ++++++++ sound/soc/sof/sof-client.h | 91 ++++ sound/soc/sof/sof-ipc-test-client.c | 321 +++++++++++++++ sound/soc/sof/sof-priv.h | 23 +- sound/soc/sof/sof-probes-client.c | 414 +++++++++++++++++++ 34 files changed, 1975 insertions(+), 626 deletions(-) create mode 100644 Documentation/driver-api/auxiliary_bus.rst create mode 100644 drivers/base/auxiliary.c create mode 100644 include/linux/auxiliary_bus.h create mode 100644 sound/soc/sof/intel/intel-client.c create mode 100644 sound/soc/sof/intel/intel-client.h create mode 100644 sound/soc/sof/sof-client.c create mode 100644 sound/soc/sof/sof-client.h create mode 100644 sound/soc/sof/sof-ipc-test-client.c create mode 100644 sound/soc/sof/sof-probes-client.c -- 2.26.2

6 19

[PATCH RESEND 0/3] alsa-lib/ASoC: use inclusive language for bclk/fsync/topology
by Pierre-Louis Bossart 17 Nov '20

17 Nov '20

This patchset is just a rebase of the code suggested in September [1], resent for convenience for comparison with kernel patches. The SOF (Sound Open Firmware) tree contains a lot of references in topology files to 'codec_slave'/'codec_master' terms, which in turn come from alsa-lib and ALSA/ASoC topology support at the kernel level. These terms are no longer compatible with the guidelines adopted by the kernel community [2], standard organizations, and need to change in backwards-compatible ways. The main/secondary terms typically suggested in guidelines don't mean anything for clocks, this patchset suggests instead the use of 'provider' and 'consumer' terms, with the 'codec' prefix kept to make it clear that the codec is the reference. The CM/CS suffixes are also replaced by CP/CC. It can be argued that the change of suffix is invasive, but finding a replacement that keeps the M and S shortcuts has proven difficult in quite a few contexts. The previous definitions are kept for backwards-compatibility so this change should not have any functional impact. It is suggested that new contributions only use the new terms but there is no requirement to transition immediately to the new definitions for existing code. Intel will however update all its past contributions related to bit clock/frame sync configurations immediately. This patchset contains the alsa-lib changes only, the kernel changes will be shared in a different patchset. The SOF firmware changes are already applied, only topology-related changes will be modified after alsa-lib changes. Feedback welcome ~Pierre [1] https://mailman.alsa-project.org/pipermail/alsa-devel/2020-September/174387… [2] https://lkml.org/lkml/2020/7/4/229 Changes since RFC: replaced 'follower' by 'consumer' as suggested by Jaroslav and Marc minor cleanups Pierre-Louis Bossart (3): topology: use inclusive language for bclk topology: use inclusive language for fsync topology: use inclusive language in documentation include/sound/uapi/asoc.h | 22 +++++++----- include/topology.h | 8 ++--- src/topology/pcm.c | 75 +++++++++++++++++++++++++++------------ 3 files changed, 71 insertions(+), 34 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: hdmi-codec: Get ELD in before reporting plugged event
by Cheng-Yi Chiang 17 Nov '20

17 Nov '20

In plugged callback, ELD should be updated from display driver so that user space can query information from ELD immediately after receiving jack plugged event. When jack is unplugged, clear ELD buffer so that user space does not get obsolete information of unplugged HDMI. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/codecs/hdmi-codec.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 403d4c6a49a8..f0e0e1836000 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -692,10 +692,16 @@ static void plugged_cb(struct device *dev, bool plugged) { struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); - if (plugged) + if (plugged) { + if (hcp->hcd.ops->get_eld) { + hcp->hcd.ops->get_eld(dev->parent, hcp->hcd.data, + hcp->eld, sizeof(hcp->eld)); + } hdmi_codec_jack_report(hcp, SND_JACK_LINEOUT); - else + } else { + memset(hcp->eld, 0, sizeof(hcp->eld)); hdmi_codec_jack_report(hcp, 0); + } } static int hdmi_codec_set_jack(struct snd_soc_component *component, -- 2.29.2.299.gdc1121823c-goog

1 0

alsa-gobject: various fixes
by GitHub pull_request - opened 17 Nov '20

17 Nov '20

alsa-project/alsa-gobject pull request #56 was opened from takaswie: This patchset includes various fixes toward v0.2.0 release. ``` Takashi Sakamoto (10): timer: device_id: fix function comment timer: user_instance: fix function comment rawmidi: stream_pair: update function comment for error reporting hwdep: query: fix usage of uninitialized value rawmidi: query: fix usage of uninitialized value ctl: query: fulfil comment about GError timer: query: fulfil comment about GError seq: query: fulfil comment about GError hwdep: query: fulfil comment about GError rawmidi: query: fulfil comment about GError src/ctl/query.c | 8 ++++---- src/hwdep/query.c | 10 +++++----- src/rawmidi/query.c | 12 ++++++------ src/rawmidi/stream-pair.c | 23 ++++++++++++++--------- src/seq/query.c | 26 +++++++++++++------------- src/timer/device-id.c | 2 +- src/timer/query.c | 14 +++++++------- src/timer/user-instance.c | 22 +++++++++++----------- 8 files changed, 61 insertions(+), 56 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/56 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/56.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] alsa-gobject: ctl: change prototype of function relevant to TLV
by GitHub pull_request - opened 17 Nov '20

17 Nov '20

alsa-project/alsa-gobject pull request #55 was opened from takaswie: In ALSA control interface, the data of TLV (Type-Length-Value) style is defined as array with unsigned int type of element. However, current implementation of ALSACtl.Card handles them as signed int type. This commit fixes the bug. This loses backward compatibility to v0.1.0 in a point of type of function argument. Request URL : https://github.com/alsa-project/alsa-gobject/pull/55 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/55.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ASoC: meson: fix COMPILE_TEST error
by Jerome Brunet 17 Nov '20

17 Nov '20

When compiled with CONFIG_HAVE_CLK, the kernel need to get provider for the clock API. This is usually selected by the platform and the sound drivers should not really care about this. However COMPILE_TEST is special and the platform required may not have been selected, leading to this type of error: > aiu-encoder-spdif.c:(.text+0x3a0): undefined reference to `clk_set_parent' Since we need a sane provider of the API with COMPILE_TEST, depends on COMMON_CLK. Fixes: 6dc4fa179fb8 ("ASoC: meson: add axg fifo base driver") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 363dc3b1bbe4..ce0cbdc69b2e 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -1,6 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only menu "ASoC support for Amlogic platforms" - depends on ARCH_MESON || COMPILE_TEST + depends on ARCH_MESON || (COMPILE_TEST && COMMON_CLK) config SND_MESON_AIU tristate "Amlogic AIU" -- 2.28.0

2 1

[PATCH] Asoc: qcom: lpass-platform: Fix memory leak
by Srinivasa Rao Mandadapu 17 Nov '20

17 Nov '20

lpass_pcm_data is not freed in error paths. Free it in error paths to avoid memory leak. Fixes: 022d00ee0b55 ("ASoC: lpass-platform: Fix broken pcm data usage") Signed-off-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-platform.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 36d1512..7a3fdf8 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -122,8 +122,10 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component, else dma_ch = 0; - if (dma_ch < 0) + if (dma_ch < 0) { + kfree(data); return dma_ch; + } if (cpu_dai->driver->id == LPASS_DP_RX) { map = drvdata->hdmiif_map; @@ -147,6 +149,7 @@ static int lpass_platform_pcmops_open(struct snd_soc_component *component, ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) { + kfree(data); dev_err(soc_runtime->dev, "setting constraints failed: %d\n", ret); return -EINVAL; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

[PATCH] Asoc: qcom: sc7180: Register shutdown handler for lpass platform
by Srinivasa Rao Mandadapu 17 Nov '20

17 Nov '20

From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Register shutdown handler to stop sc7180 lpass platform driver and to disable audio clocks. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 10 ++++++++++ sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 1 + 3 files changed, 12 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 1fd862e..d33eae6 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -914,5 +914,15 @@ int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev) } EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_remove); +void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev) +{ + struct lpass_data *drvdata = platform_get_drvdata(pdev); + + if (drvdata->variant->exit) + drvdata->variant->exit(pdev); + +} +EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_shutdown); + MODULE_DESCRIPTION("QTi LPASS CPU Driver"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index f619590..61b51b5 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -297,6 +297,7 @@ static struct platform_driver sc7180_lpass_cpu_platform_driver = { }, .probe = asoc_qcom_lpass_cpu_platform_probe, .remove = asoc_qcom_lpass_cpu_platform_remove, + .shutdown = asoc_qcom_lpass_cpu_platform_shutdown, }; module_platform_driver(sc7180_lpass_cpu_platform_driver); diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index b4830f3..32a68c4 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -255,6 +255,7 @@ struct lpass_variant { /* register the platform driver from the CPU DAI driver */ int asoc_qcom_lpass_platform_register(struct platform_device *); int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev); +void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[PATCH] ASoC: qcom: sc7180: initialize the "no_headphone" variable
by Dan Carpenter 17 Nov '20

17 Nov '20

The "no_headphone" variable is never set to "false" so it could be uninitialized. Fixes: e936619b7ce7 ("ASoC: qcom: sc7180: Modify machine driver for sound card") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- Apparently we recently turned off uninitialized variable warnings by default in commit 78a5255ffb6a ("Stop the ad-hoc games with -Wno-maybe-initialized") sound/soc/qcom/sc7180.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index e2e6567566af..de70fa792aea 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -316,7 +316,7 @@ static int sc7180_snd_platform_probe(struct platform_device *pdev) struct snd_soc_dai_link *link; int ret; int i; - bool no_headphone; + bool no_headphone = false; /* Allocate the private data */ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); -- 2.29.2

2 1

[PATCH] ASoC: SOF: Intel: initial support for Alderlake-S
by Kai Vehmanen 17 Nov '20

17 Nov '20

Add Kconfig entries, PCI ID and chip info for Alderlake-S product. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/soc/sof/intel/Kconfig | 16 ++++++++++++++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/tgl.c | 16 ++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 21 +++++++++++++++++++++ 4 files changed, 54 insertions(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index b233302f05e4..b607dba5fb2a 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -29,6 +29,7 @@ config SND_SOC_SOF_INTEL_PCI select SND_SOC_SOF_TIGERLAKE if SND_SOC_SOF_TIGERLAKE_SUPPORT select SND_SOC_SOF_ELKHARTLAKE if SND_SOC_SOF_ELKHARTLAKE_SUPPORT select SND_SOC_SOF_JASPERLAKE if SND_SOC_SOF_JASPERLAKE_SUPPORT + select SND_SOC_SOF_ALDERLAKE if SND_SOC_SOF_ALDERLAKE_SUPPORT help This option is not user-selectable but automagically handled by 'select' statements at a higher level. @@ -269,6 +270,21 @@ config SND_SOC_SOF_JASPERLAKE This option is not user-selectable but automagically handled by 'select' statements at a higher level. +config SND_SOC_SOF_ALDERLAKE_SUPPORT + bool "SOF support for Alderlake" + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Alderlake processors. + Say Y if you have such a device. + If unsure select "N". + +config SND_SOC_SOF_ALDERLAKE + tristate + select SND_SOC_SOF_HDA_COMMON + help + This option is not user-selectable but automagically handled by + 'select' statements at a higher level + config SND_SOC_SOF_HDA_COMMON tristate select SND_INTEL_DSP_CONFIG diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 1bc4dabdd394..75954e642c5e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -742,6 +742,7 @@ extern const struct sof_intel_dsp_desc tgl_chip_info; extern const struct sof_intel_dsp_desc tglh_chip_info; extern const struct sof_intel_dsp_desc ehl_chip_info; extern const struct sof_intel_dsp_desc jsl_chip_info; +extern const struct sof_intel_dsp_desc adls_chip_info; /* machine driver select */ void hda_machine_select(struct snd_sof_dev *sdev); diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index 0278b67de1ec..bdcc66d0df78 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -151,3 +151,19 @@ const struct sof_intel_dsp_desc tglh_chip_info = { .ssp_base_offset = CNL_SSP_BASE_OFFSET, }; EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); + +const struct sof_intel_dsp_desc adls_chip_info = { + /* Alderlake-S */ + .cores_num = 2, + .init_core_mask = BIT(0), + .host_managed_cores_mask = BIT(0), + .ipc_req = CNL_DSP_REG_HIPCIDR, + .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY, + .ipc_ack = CNL_DSP_REG_HIPCIDA, + .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE, + .ipc_ctl = CNL_DSP_REG_HIPCCTL, + .rom_init_timeout = 300, + .ssp_count = ICL_SSP_COUNT, + .ssp_base_offset = CNL_SSP_BASE_OFFSET, +}; +EXPORT_SYMBOL_NS(adls_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 8f62e3487dc1..f2a107481336 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -284,6 +284,23 @@ static const struct sof_dev_desc jsl_desc = { }; #endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) +static const struct sof_dev_desc adls_desc = { + .machines = snd_soc_acpi_intel_hda_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &adls_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .default_fw_filename = "sof-adl-s.ri", + .nocodec_tplg_filename = "sof-adl-nocodec.tplg", + .ops = &sof_tgl_ops, +}; +#endif + static const struct dev_pm_ops sof_pci_pm = { .prepare = snd_sof_prepare, .complete = snd_sof_complete, @@ -490,6 +507,10 @@ static const struct pci_device_id sof_pci_ids[] = { .driver_data = (unsigned long)&ehl_desc}, { PCI_DEVICE(0x8086, 0x4b58), .driver_data = (unsigned long)&ehl_desc}, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ALDERLAKE) + { PCI_DEVICE(0x8086, 0x7ad0), + .driver_data = (unsigned long)&adls_desc}, #endif { 0, } }; -- 2.28.0

2 1

[PATCH 0/5] ASoC: Intel: catpt: Offload fixes and code optimization
by Cezary Rojewski 17 Nov '20

17 Nov '20

First two of the series address bugs connected mainly to offload streams: - scenarios with very low buffer sizes: RESET_STREAM IPC timeouts - fix lp clock selection when switching between PAUSE <-> RESUME states: glitches on first offload when no additional stream is opened simultaneously Follow ups are: code reduction and optimization oriented patches. This has been foretold in: [PATCH v10 00/14] ASoC: Intel: Catpt - Lynx and Wildcat point https://www.spinics.net/lists/alsa-devel/msg116440.html Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. First, optimize applying of user settings - prevent redundand calls from happening - and then as mentioned above, streamline power on/off sequence for LPT and WPT. Cezary Rojewski (5): ASoC: Intel: catpt: Skip position update for unprepared streams ASoC: Intel: catpt: Correct clock selection for dai trigger ASoC: Intel: catpt: Optimize applying user settings ASoC: Intel: catpt: Streamline power routines across LPT and WPT ASoC: Intel: catpt: Cleanup after power routines streamlining sound/soc/intel/catpt/core.h | 10 ++- sound/soc/intel/catpt/device.c | 18 +++--- sound/soc/intel/catpt/dsp.c | 56 ++-------------- sound/soc/intel/catpt/pcm.c | 113 ++++++++++++++++----------------- 4 files changed, 74 insertions(+), 123 deletions(-) -- 2.17.1

2 6

[PATCH v3 0/4] ASoC: pcm512x: Patch series to set fmt from `set_fmt()`
by Kirill Marinushkin 17 Nov '20

17 Nov '20

Set format from `set_fmt()` func instead of `hw_params()`, plus supportive commits Kirill Marinushkin (4): ASoC: pcm512x: Fix not setting word length if DAIFMT_CBS_CFS ASoC: pcm512x: Rearrange operations in `hw_params()` ASoC: pcm512x: Move format check into `set_fmt()` ASoC: pcm512x: Add support for more data formats sound/soc/codecs/pcm512x.c | 134 ++++++++++++++++++++++++++++----------------- 1 file changed, 84 insertions(+), 50 deletions(-) -- 2.13.6

3 6

[PATCH 0/5] This patch series enables DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 17 Nov '20

17 Nov '20

v1: Initial patch version, which contains fix for S24_LE format and also enable DMA mode on Intel Keembay platform Michael Sit Wei Hong (5): ASoC: Intel: KMB: Fix S24_LE configuration dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: soc-generic-dmaengine-pcm: Add custom prepare and submit function ASoC: dmaengine_pcm: expose functions to header file for custom functions ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ include/sound/dmaengine_pcm.h | 21 ++ sound/core/pcm_dmaengine.c | 46 ++-- sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 208 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + sound/soc/soc-generic-dmaengine-pcm.c | 8 +- 7 files changed, 270 insertions(+), 38 deletions(-) -- 2.17.1

3 9

[PATCH] Asoc: qcom: lpass-sc7180: Add 32 bit format support for capture
by Srinivasa Rao Mandadapu 17 Nov '20

17 Nov '20

From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Add 32 bit format support for capture in lpass-sc7180 snd_soc_dai_driver capabilities. Need to add contstraints in machine driver so that only specific format allowed. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Manidadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-sc7180.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 61b51b5..3dc80fc 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -34,7 +34,8 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .capture = { .stream_name = "Primary Capture", - .formats = SNDRV_PCM_FMTBIT_S16, + .formats = SNDRV_PCM_FMTBIT_S16 | + SNDRV_PCM_FMTBIT_S32, .rates = SNDRV_PCM_RATE_48000, .rate_min = 48000, .rate_max = 48000, -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[PATCH] ALSA: hda: Add Alderlake-S PCI ID and HDMI codec vid
by Kai Vehmanen 16 Nov '20

16 Nov '20

Add HD Audio PCI ID and HDMI codec vendor ID for Intel Alder Lake. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 4 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d539f52009a1..6852668f1bcb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2506,6 +2506,9 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Alderlake-S */ + { PCI_DEVICE(0x8086, 0x7ad0), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Elkhart Lake */ { PCI_DEVICE(0x8086, 0x4b55), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ccd1df059654..b0068f8ca46d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4274,6 +4274,7 @@ HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), -- 2.28.0

2 1

[PATCH 0/2] alsa-gobject: rawmidi: fix to select substream
by GitHub pull_request - opened 16 Nov '20

16 Nov '20

alsa-project/alsa-gobject pull request #54 was opened from takaswie: This patchset fixes #53 . ALSARawmidi.StreamPair.open() has `subdevice_id` argument to select one of substreams supported by the Rawmidi device. However, this parameter is ignored and the first substream is always selected. In ALSA rawmidi core, private structure associated to file descriptor to control character device assists selection of rawmidi substream. It requires to keep the file descriptor until the selected substream is actually attached by open rawmidi character device. Current implementation of libalsarawmidi is to close the file descriptor before opening rawmidi character device. This is the cause of the issue. This commit fixes the bug by keeping the file descriptor till opening rawmidi character device. ``` Takashi Sakamoto (2): rawmidi: query: allow caller to take ownership of file descriptor for control character device rawmidi: stream_pair: fix to select substream src/rawmidi/privates.h | 2 +- src/rawmidi/query.c | 16 ++++++++++------ src/rawmidi/stream-pair.c | 4 +++- 3 files changed, 14 insertions(+), 8 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/54 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/54.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ALSA: usb-audio: Use ALC1220-VB-DT mapping for ASUS ROG Strix TRX40 mobo
by Kai-Heng Feng 16 Nov '20

16 Nov '20

ASUS ROG Strix also uses ALC1220-VB-DT, so adjust the mapping and add profile name to let userspace pick correct UCM profile. BugLink: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1031 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/card.c | 4 ++++ sound/usb/mixer_maps.c | 3 ++- 2 files changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index fa764b61fe9c..4457214a3ae6 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -379,6 +379,10 @@ static const struct usb_audio_device_name usb_audio_names[] = { DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"), + /* ASUS ROG Strix */ + PROFILE_NAME(0x0b05, 0x1917, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + /* Dell WD15 Dock */ PROFILE_NAME(0x0bda, 0x4014, "Dell", "WD15 Dock", "Dell-WD15-Dock"), /* Dell WD19 Dock */ diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index c369c81e74c4..a7212f16660e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -561,7 +561,8 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { }, { /* ASUS ROG Strix */ .id = USB_ID(0x0b05, 0x1917), - .map = asus_rog_map, + .map = trx40_mobo_map, + .connector_map = trx40_mobo_connector_map, }, { /* MSI TRX40 Creator */ .id = USB_ID(0x0db0, 0x0d64), -- 2.28.0

2 1

[sound:test/usb-hacks-wip 44/53] sound/usb/endpoint.c:1255:5: warning: no previous prototype for function 'snd_usb_sync_endpoint_start'
by kernel test robot 16 Nov '20

16 Nov '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/usb-hacks-wip head: 8d9dd7caccc6daebab8b2a1a7bbd09e605b83fba commit: ca31918f6843b5369d0c900c43bfe5f71441b5b2 [44/53] ALSA: usb-audio: Refactor endpoint management config: x86_64-randconfig-a001-20201115 (attached as .config) compiler: clang version 12.0.0 (https://github.com/llvm/llvm-project 9a85643cd357e412cff69067bb5c4840e228c2ab) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # install x86_64 cross compiling tool for clang build # apt-get install binutils-x86-64-linux-gnu # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound test/usb-hacks-wip git checkout ca31918f6843b5369d0c900c43bfe5f71441b5b2 # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross ARCH=x86_64 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): >> sound/usb/endpoint.c:1255:5: warning: no previous prototype for function 'snd_usb_sync_endpoint_start' [-Wmissing-prototypes] int snd_usb_sync_endpoint_start(struct snd_usb_endpoint *ep, ^ sound/usb/endpoint.c:1255:1: note: declare 'static' if the function is not intended to be used outside of this translation unit int snd_usb_sync_endpoint_start(struct snd_usb_endpoint *ep, ^ static >> sound/usb/endpoint.c:1298:6: warning: no previous prototype for function 'snd_usb_sync_endpoint_stop' [-Wmissing-prototypes] void snd_usb_sync_endpoint_stop(struct snd_usb_endpoint *ep) ^ sound/usb/endpoint.c:1298:1: note: declare 'static' if the function is not intended to be used outside of this translation unit void snd_usb_sync_endpoint_stop(struct snd_usb_endpoint *ep) ^ static 2 warnings generated. vim +/snd_usb_sync_endpoint_start +1255 sound/usb/endpoint.c 1254 > 1255 int snd_usb_sync_endpoint_start(struct snd_usb_endpoint *ep, 1256 struct snd_usb_endpoint *data_ep) 1257 { 1258 int err; 1259 1260 err = snd_usb_endpoint_start(ep); 1261 return err; 1262 } 1263 1264 /** 1265 * snd_usb_endpoint_stop: stop an snd_usb_endpoint 1266 * 1267 * @ep: the endpoint to stop (may be NULL) 1268 * 1269 * A call to this function will decrement the use count of the endpoint. 1270 * In case the last user has requested the endpoint stop, the URBs will 1271 * actually be deactivated. 1272 * 1273 * Must be balanced to calls of snd_usb_endpoint_start(). 1274 * 1275 * The caller needs to synchronize the pending stop operation via 1276 * snd_usb_endpoint_sync_pending_stop(). 1277 */ 1278 void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep) 1279 { 1280 if (!ep) 1281 return; 1282 1283 usb_audio_dbg(ep->chip, "Stopping %s EP 0x%x (count %d)\n", 1284 ep_type_name(ep->type), ep->ep_num, ep->use_count); 1285 1286 if (snd_BUG_ON(ep->use_count == 0)) 1287 return; 1288 1289 if (ep->sync_master) 1290 WRITE_ONCE(ep->sync_master->sync_slave, NULL); 1291 1292 if (--ep->use_count == 0) { 1293 deactivate_urbs(ep, false); 1294 set_bit(EP_FLAG_STOPPING, &ep->flags); 1295 } 1296 } 1297 > 1298 void snd_usb_sync_endpoint_stop(struct snd_usb_endpoint *ep) 1299 { 1300 snd_usb_endpoint_stop(ep); 1301 } 1302 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH 0/5] This patch series enables DMA mode on Intel Keem Bay platform
by Michael Sit Wei Hong 16 Nov '20

16 Nov '20

v1: Initial patch version, which contains fix for S24_LE format and also enable DMA mode on Intel Keembay platform Michael Sit Wei Hong (5): ASoC: Intel: KMB: Fix S24_LE configuration dt-bindings: sound: intel, keembay-i2s: Add info for device to use DMA ASoC: soc-generic-dmaengine-pcm: Add custom prepare and submit function ASoC: dmaengine_pcm: expose functions to header file for custom functions ASoC: Intel: KMB: Enable DMA transfer mode .../bindings/sound/intel,keembay-i2s.yaml | 14 ++ include/sound/dmaengine_pcm.h | 21 ++ sound/core/pcm_dmaengine.c | 46 ++-- sound/soc/intel/Kconfig | 2 + sound/soc/intel/keembay/kmb_platform.c | 208 ++++++++++++++++-- sound/soc/intel/keembay/kmb_platform.h | 9 + sound/soc/soc-generic-dmaengine-pcm.c | 8 +- 7 files changed, 270 insertions(+), 38 deletions(-) -- 2.17.1

1 5

rawmidi: substream 0 is always selected, ignoring parameter
by GitHub issues - opened 16 Nov '20

16 Nov '20

alsa-project/alsa-gobject issue #53 was opened from takaswie: This issue is confirmed in v0.1.0 release. `ALSARawmidi.StreamPair.open()` has `subdevice_id` to select one of substreams supported by the Rawmidi device. However, this parameter is ignored and the first substream is always selected. Currently, the procedure in `ALSARawmidi.StreamPair.open()` to open Rawmidi character device is: 1. open control character device 2. ioctl call with SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE 3. close control character device 4. open rawmidi character device The cause to ignore is step 3. After closing control character device, private structure assigned to file descriptor of control character device is dropped in kernel land. At step 4, the selection information is already lost and the first substream is assigned to file descriptor of rawmidi character device. The solution is to keep file descriptor of control character device when opening rawmidi character device, thus the procedure should be: 1. open control character device 2. ioctl call with SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE 3. open rawmidi character device 4. close control character device Issue URL : https://github.com/alsa-project/alsa-gobject/issues/53 Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 00/15] alsa-gobject: rawmidi: enhancement for error reporting
by GitHub pull_request - opened 16 Nov '20

16 Nov '20

alsa-project/alsa-gobject pull request #52 was opened from takaswie: The patchset is the part of work about #47. Current implementation uses library-wide error domain to report error. this is partly against rule of GError usage. Furthermore, the error delivers information just about errno and hard to know the cause of error. This patchset enhances error reporting. The library-wide error domain is obsoleted. A new error domain is added just for ALSARawmidi.StreamPair. The error domain delivers information about the cause of error ``` Takashi Sakamoto (15): timer: user_instance: fix code for GFileError seq: user_client: fix code for GFileError rawmidi: skip check of return value from g_malloc() rawmidi: check whether method argument for GError is available rawmidi: add checks for method arguments rawmidi: stream_pair: just return when character device is not opened rawmidi: query: simplify count check rawmidi: add GLib enumerations for error reporting rawmidi: stream-pair: add GQuark to report error for ALSARawmidi.StreamPair rawmidi: report error for ioctl rawmidi: report open/read/write/fcntl error rawmidi: report error for card disconnection state rawmidi: stream_pair: report error about write-only file descriptor rawmidi: query: use GFileError to report error rawmidi: obsolete library-wide GQuark for error reporting src/rawmidi/alsarawmidi-enum-types.h | 14 +++ src/rawmidi/alsarawmidi.map | 5 + src/rawmidi/privates.h | 6 - src/rawmidi/query.c | 117 ++++++++---------- src/rawmidi/stream-pair.c | 172 +++++++++++++++++++++------ src/rawmidi/stream-pair.h | 4 + src/seq/user-client.c | 6 +- src/timer/user-instance.c | 2 +- tests/alsarawmidi-enums | 6 + 9 files changed, 216 insertions(+), 116 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/52 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/52.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 1/2] ASoC: rt5682: Add a new property for the DMIC clock driving
by Oder Chiou 16 Nov '20

16 Nov '20

The patch adds a new property to set the DMIC clock driving. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5682.h | 1 + sound/soc/codecs/rt5682-i2c.c | 5 +++++ sound/soc/codecs/rt5682.c | 3 +++ sound/soc/codecs/rt5682.h | 14 ++++++++++++++ 4 files changed, 23 insertions(+) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index e1f790561ac1..3900a07e3935 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -40,6 +40,7 @@ struct rt5682_platform_data { unsigned int btndet_delay; unsigned int dmic_clk_rate; unsigned int dmic_delay; + bool dmic_clk_driving_high; const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 6b4e0eb30c89..37d13120f5ba 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -221,6 +221,11 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, case RT5682_DMIC1_CLK_GPIO3: /* share with BCLK2 */ regmap_update_bits(rt5682->regmap, RT5682_GPIO_CTRL_1, RT5682_GP3_PIN_MASK, RT5682_GP3_PIN_DMIC_CLK); + if (rt5682->pdata.dmic_clk_driving_high) + regmap_update_bits(rt5682->regmap, + RT5682_PAD_DRIVING_CTRL, + RT5682_PAD_DRV_GP3_MASK, + 2 << RT5682_PAD_DRV_GP3_SFT); break; default: diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index a9acce7b6cca..f299b30f4f59 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2989,6 +2989,9 @@ int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + rt5682->pdata.dmic_clk_driving_high = device_property_read_bool(dev, + "realtek,dmic-clk-driving-high"); + return 0; } EXPORT_SYMBOL_GPL(rt5682_parse_dt); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 354acd735ef4..99b85cfe6248 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1271,6 +1271,20 @@ #define RT5682_CP_CLK_HP_300KHZ (0x2 << 4) #define RT5682_CP_CLK_HP_600KHZ (0x3 << 4) +/* Pad Driving Control (0x0136) */ +#define RT5682_PAD_DRV_GP1_MASK (0x3 << 14) +#define RT5682_PAD_DRV_GP1_SFT 14 +#define RT5682_PAD_DRV_GP2_MASK (0x3 << 12) +#define RT5682_PAD_DRV_GP2_SFT 12 +#define RT5682_PAD_DRV_GP3_MASK (0x3 << 10) +#define RT5682_PAD_DRV_GP3_SFT 10 +#define RT5682_PAD_DRV_GP4_MASK (0x3 << 8) +#define RT5682_PAD_DRV_GP4_SFT 8 +#define RT5682_PAD_DRV_GP5_MASK (0x3 << 6) +#define RT5682_PAD_DRV_GP5_SFT 6 +#define RT5682_PAD_DRV_GP6_MASK (0x3 << 4) +#define RT5682_PAD_DRV_GP6_SFT 4 + /* Chopper and Clock control for DAC (0x013a)*/ #define RT5682_CKXEN_DAC1_MASK (0x1 << 13) #define RT5682_CKXEN_DAC1_SFT 13 -- 2.29.0

3 4

USB Audio: Xonar U7 MKII needs `iec958_device` entry
by GitHub issues - opened 15 Nov '20

15 Nov '20

alsa-project/alsa-lib issue #100 was opened from ndl: `USB-Audio.conf` includes V1 of Asus Xonar U7 card but not [MKII version](https://www.asus.com/ch-en/Sound-Cards/Xonar-U7-MKII/), the section `USB-Audio.pcm.iec958_device` should be extended to include the string "Xonar U7 MKII" 1 without this S/PDIF output doesn't work. Issue URL : https://github.com/alsa-project/alsa-lib/issues/100 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

fireface: FF800 periodic burst of Xruns
by David Runge 15 Nov '20

15 Nov '20

Hi! I'm currently testing the snd-fireface module on kernel > 5.9 with the RME Fireface 800 (FF800). Unfortunately, when running jackd using that device it generates large bursts of Xruns (~10000) quite periodically. /usr/bin/jackd -P80 -p 512 -d alsa -d hw:Fireface800 -n 3 -p 256 -r 48000 The issues do not seem to correlate with any load events on the system (and in between the Xrun events there are no Xruns even under load). I am not entirely sure how to further debug this. Any hints and suggestions would be very much welcome! Note: While using libffado for the device works it is not at all stable since kernel > 5.9 (sporadic rediscovery events of firewire-ohci, requiring the unplugging/replugging of the device's firewire cable and of course the restart of jackd). This is why I tried the snd-fireface module again, but it seems to suffer from a different set of issues ;-) Best, David -- https://sleepmap.de

1 0

[PATCH 0/9] alsa-gobject: hwdep: enhancement for error reporting
by GitHub pull_request - opened 15 Nov '20

15 Nov '20

alsa-project/alsa-gobject pull request #51 was opened from takaswie: The patchset is the part of work about #47. Current implementation uses library-wide error domain to report error. this is partly against rule of GError usage. Furthermore, the error delivers information just about errno and hard to know the cause of error. This patchset enhances error reporting. The library-wide error domain is obsoleted. ALSAHwdep library is small and includes global functions. GFileError is used for the functions to report error since It has no specific error domain in current design. ``` Takashi Sakamoto (9): seq: query: remove duplicated semicolon ctl: query: simplify count check timer: instance_param: simplify count check hwdep: skip check of return value from g_malloc() hwdep: check whether method argument for GError is available hwdep: add checks for method arguments hwdep: query: simplify count check hwdep: query: use GFileError to report error hwdep; obsolete library-wide GQuark for error reporting src/ctl/query.c | 9 ++-- src/hwdep/privates.h | 6 --- src/hwdep/query.c | 101 ++++++++++++++++-------------------- src/seq/query.c | 2 +- src/timer/instance-params.c | 5 +- 5 files changed, 50 insertions(+), 73 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/51 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/51.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH v3 00/10] Auxiliary bus implementation and SOF multi-client support
by Dave Ertman 15 Nov '20

15 Nov '20

Brief history of Auxiliary Bus ============================== The auxiliary bus code was originally submitted upstream as virtual bus, and was submitted through the netdev tree.  This process generated up to v4.  This discussion can be found here: https://lore.kernel.org/netdev/20191111192219.30259-1-jeffrey.t.kirsher@int… At this point, GregKH requested that we take the review and revision process to an internal mailing list and garner the buy-in of a respected kernel contributor. The auxiliary bus (then known as virtual bus) was originally submitted along with implementation code for the ice driver and irdma drive, causing the complication of also having dependencies in the rdma tree. This new submission is utilizing an auxiliary bus consumer in only the sound driver tree to create the initial implementation and a single user. Since implementation work has started on this patch set, there have been multiple inquiries about the time frame of its completion. It appears that there will be numerous consumers of this functionality. The process of internal review and implementation using the sound drivers generated 19 internal versions.  The changes, including the name change from virtual bus to auxiliary bus, from these versions can be summarized as the following: - Fixed compilation and checkpatch errors - Improved documentation to address the motivation for virtual bus. - Renamed virtual bus to auxiliary bus - increased maximum device name size - Correct order in Kconfig and Makefile - removed the mid-layer adev->release layer for device unregister - pushed adev->id management to parent driver - all error paths out of ancillary_device_register return error code - all error paths out of ancillary_device_register use put_device - added adev->name element - modname in register cannot be NULL - added KBUILD_MODNAME as prefix for match_name - push adev->id responsibility to registering driver - uevent now parses adev->dev name - match_id function now parses adev->dev name - changed drivers probe function to also take an ancillary_device_id param - split ancillary_device_register into device_initialize and device_add - adjusted what is done in device_initialize and device_add - change adev to ancildev and adrv to ancildrv - change adev to ancildev in documentation ========================== Introduces the auxiliary bus implementation along with the example usage in the Sound Open Firmware(SOF) audio driver. In some subsystems, the functionality of the core device (PCI/ACPI/other) may be too complex for a single device to be managed as a monolithic block or a part of the functionality might need to be exposed to a different subsystem. Splitting the functionality into smaller orthogonal devices makes it easier to manage data, power management and domain-specific communication with the hardware. Also, common auxiliary_device functionality across primary devices can be handled by a common auxiliary_device. A key requirement for such a split is that there is no dependency on a physical bus, device, register accesses or regmap support. These individual devices split from the core cannot live on the platform bus as they are not physical devices that are controlled by DT/ACPI. The same argument applies for not using MFD in this scenario as it relies on individual function devices being physical devices that are DT enumerated. An example for this kind of requirement is the audio subsystem where a single IP handles multiple entities such as HDMI, Soundwire, local devices such as mics/speakers etc. The split for the core's functionality can be arbitrary or be defined by the DSP firmware topology and include hooks for test/debug. This allows for the audio core device to be minimal and tightly coupled with handling the hardware-specific logic and communication. The auxiliary bus is intended to be minimal, generic and avoid domain-specific assumptions. Each auxiliary bus device represents a part of its parent functionality. The generic behavior can be extended and specialized as needed by encapsulating an auxiliary bus device within other domain-specific structures and the use of .ops callbacks. The SOF driver adopts the auxiliary bus for implementing the multi-client support. A client in the context of the SOF driver represents a part of the core device's functionality. It is not a physical device but rather an auxiliary device that needs to communicate with the DSP via IPCs. With multi-client support,the sound card can be separated into multiple orthogonal auxiliary devices for local devices (mic/speakers etc), HDMI, sensing, probes, debug etc. In this series, we demonstrate the usage of the auxiliary bus with the help of the IPC test client which is used for testing the serialization of IPCs when multiple clients talk to the DSP at the same time. v3 changes: rename to auxiliary bus move .c file to drivers/base/ split auxdev unregister flow into uninitialize and delete steps update kernel-doc on register functions for new unregister model update documentation with new name and unregister flow remove check for release in auxiliary bus, allow core to catch Change driver register so only probe and id_table mandatory Fix matching logic in auxillary_match_id to account for longer id->name utilize auxiliary_bus callbacks for probe, remove and shutdown add auxiliary_find_device function add code to ensure unique auxdrv name shorten initialize/uninitialize in function names to init/uninit simplify looping logic in match_id in drives/base/Kconfig changed from tristate to bool Modified signature SOF client register/unregister API Modified PM runtime enabling sequence in the SOF ipc test aux driver Removed driver.name from the aux driver Added Probes client aux driver and device registration in the SOF driver. This allows for enabling the probes functionality in the SOF firmware for audio data extraction from specific points in the audio pipeline. Without auxiliary bus, the implementation requires modifying existing 15+ machine drivers to add the probes DAI links to the sounds cards. Using the auxiliary bus allows for splitting the probes implementation from the SOF core making it easy to maintain. v2 changes: defined pr_fmt for kernel messages replaced WARN_ON calls in registration with pr_err calls adding kernel-doc function comments for device_initialize and device_add fix typo in documentation removed inaccurate line in documentation fixed formatting in drivers/bus/Makefile changed unwind path for sof_client_dev_alloc() improved comments for client list and mem freeing during client unreg removed debugfs entries in sof_ipc_test_client_drv during remove changed the signature of sof_debug_ipc_flood_test() fix a looping error in ancillary_match_id updated error value in sof_client_dev_register() mutex held while traversing client list when unregistering clients updated includes in sof-client.h Dave Ertman (1): Add auxiliary bus support Ranjani Sridharan (9): ASoC: SOF: Introduce descriptors for SOF client ASoC: SOF: Create client driver for IPC test ASoC: SOF: ops: Add ops for client registration ASoC: SOF: Intel: Define ops for client registration ASoC: SOF: Intel: Remove IPC flood test support in SOF core ASoC: SOF: sof-client: Add client APIs to access probes ops ASoC: SOF: compress: move and export sof_probe_compr_ops ASoC: SOF: Add new client driver for probes support ASoC: SOF: Intel: CNL: register probes client Documentation/driver-api/auxiliary_bus.rst | 228 ++++++++++ Documentation/driver-api/index.rst | 1 + drivers/base/Kconfig | 3 + drivers/base/Makefile | 1 + drivers/base/auxiliary.c | 267 ++++++++++++ include/linux/auxiliary_bus.h | 78 ++++ include/linux/mod_devicetable.h | 8 + scripts/mod/devicetable-offsets.c | 3 + scripts/mod/file2alias.c | 8 + sound/soc/sof/Kconfig | 47 ++- sound/soc/sof/Makefile | 10 +- sound/soc/sof/compress.c | 60 +-- sound/soc/sof/compress.h | 1 + sound/soc/sof/core.c | 18 +- sound/soc/sof/debug.c | 457 --------------------- sound/soc/sof/intel/Kconfig | 9 + sound/soc/sof/intel/Makefile | 3 + sound/soc/sof/intel/apl.c | 16 + sound/soc/sof/intel/bdw.c | 16 + sound/soc/sof/intel/byt.c | 20 + sound/soc/sof/intel/cnl.c | 32 ++ sound/soc/sof/intel/hda-dai.c | 27 -- sound/soc/sof/intel/hda.h | 6 - sound/soc/sof/intel/intel-client.c | 40 ++ sound/soc/sof/intel/intel-client.h | 26 ++ sound/soc/sof/ops.h | 14 + sound/soc/sof/pcm.c | 11 - sound/soc/sof/probe.c | 124 +++--- sound/soc/sof/probe.h | 41 +- sound/soc/sof/sof-client.c | 170 ++++++++ sound/soc/sof/sof-client.h | 91 ++++ sound/soc/sof/sof-ipc-test-client.c | 321 +++++++++++++++ sound/soc/sof/sof-priv.h | 23 +- sound/soc/sof/sof-probes-client.c | 414 +++++++++++++++++++ 34 files changed, 1968 insertions(+), 626 deletions(-) create mode 100644 Documentation/driver-api/auxiliary_bus.rst create mode 100644 drivers/base/auxiliary.c create mode 100644 include/linux/auxiliary_bus.h create mode 100644 sound/soc/sof/intel/intel-client.c create mode 100644 sound/soc/sof/intel/intel-client.h create mode 100644 sound/soc/sof/sof-client.c create mode 100644 sound/soc/sof/sof-client.h create mode 100644 sound/soc/sof/sof-ipc-test-client.c create mode 100644 sound/soc/sof/sof-probes-client.c -- 2.26.2

9 33

[PATCH 00/15] alsa-gobject: seq: enhancement for error reporting
by GitHub pull_request - opened 15 Nov '20

15 Nov '20

alsa-project/alsa-gobject pull request #50 was opened from takaswie: The patchset is the part of work about #47. Current implementation uses library-wide error domain to report error. this is partly against rule of GError usage. Furthermore, the error delivers information just about errno and hard to know the cause of error. This patchset enhances error reporting. The library-wide error domain is obsoleted. A new error domain is added just for ALSASeq.UserClient. The error domain delivers information about the cause of error. ``` Takashi Sakamoto (15): seq: skip check of return value from g_malloc() seq: check whether method argument for GError is available seq: add checks for method arguments seq: user_client: just return when character device is not opened seq: add GLib enumeration to report type of error for ALSASeq.UserClient seq: user_client: add GQuark to report error for ALSASeq.UserClient seq: user_client: report error for ioctl seq: user_client: report open/write error seq: user_client: report error for port access permission seq: user_client: report error for queue access permission seq: user_client: update function comment for error reporting seq: query: use GFileError to report error seq: query: code refactoring to unify code to open file descriptor seq: query: code refactoring for error path seq: obsolete library-wide error quark src/seq/alsaseq-enum-types.h | 14 ++ src/seq/alsaseq.map | 5 + src/seq/client-info.c | 20 +- src/seq/event-cntr.c | 341 +++++++++++++++++---------------- src/seq/privates.h | 6 - src/seq/query.c | 356 +++++++++++++++-------------------- src/seq/queue-status.c | 4 + src/seq/queue-tempo.c | 4 + src/seq/queue-timer.c | 4 + src/seq/remove-filter.c | 42 ++--- src/seq/user-client.c | 275 ++++++++++++++++++--------- src/seq/user-client.h | 4 + tests/alsaseq-enums | 7 + 13 files changed, 566 insertions(+), 516 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/50 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/50.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[Question]: Mute all channels when starting X11
by GitHub issues - edited 14 Nov '20

14 Nov '20

alsa-project/alsa-utils issue #72 was edited from atreyasha: Hello, thank you for developing `alsa-utils`. I have a specific question pertaining to `amixer`. ## Current behaviour When starting `X11` and `i3` (my window manager), I would like to have all audio channels (speaker, headphones, etc.) muted. At the moment, I have the following command in my `i3` configuration file: ``` amixer -q sset Master mute ``` And as expected, the Master channel is indeed muted after starting `X11`. However, when I connect or disconnect my headphones, the volume automatically reverts back to the volume that was saved during the last boot. ## Expected behaviour I would like all volume channels to stay muted. So for example after starting `X11` and `i3`, I would like to see that both my speakers and headphones are muted, even when I plug/unplug the jack. Would you have any recommendations on how to do this? I was considering manually muting both headphones and speakers with `amixer` and then re-using the `alsa` state file during boots. But I am unsure because this seems like a hack'ish solution. ## System specifications OS/Kernel: Arch Linux, 5.9.8-arch1-1, x86_64 GNU/Linux alsa-utils: 1.2.4-2 (installed via `pacman` package manager) wm: `i3-gaps` Issue URL : https://github.com/alsa-project/alsa-utils/issues/72 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[Question]: Mute all channels when starting X11
by GitHub issues - edited 14 Nov '20

14 Nov '20

alsa-project/alsa-utils issue #72 was edited from atreyasha: Hello, thank you for developing `alsa-utils`. I have a specific question pertaining to `amixer`. ## Current behaviour When starting `X11` and `i3` (my window manager), I would like to have all audio channels (speaker, headphones, etc.) muted. At the moment, I have the following command in my `i3` configuration file: ``` amixer -q sset Master mute ``` And as expected, the Master channel is indeed muted after starting `X11`. However, when I connect or disconnect my headphones, the volume automatically reverts back to the volume that was saved during the last boot. ## Expected behaviour I would like all volume channels to stay muted. So for example after starting `X11` and `i3`, I would like to see that both my speakers and headphones are muted, even when I plug/unplug the jack. Would you have any recommendations on how to do this? I was considering manually muting both headphones and speakers with `amixer` and then re-using the `alsa` state file during boots. But I am unsure because this seems like a hack'ish solution. ## System specifications OS/Kernel: Linux lt410 5.9.8-arch1-1 #1 SMP PREEMPT Tue, 10 Nov 2020 22:44:11 +0000 x86_64 GNU/Linux alsa-utils: 1.2.4-2 (installed via `pacman` package manager) wm: `i3-gaps` Issue URL : https://github.com/alsa-project/alsa-utils/issues/72 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[Question]: Mute all channels at boot
by GitHub issues - opened 14 Nov '20

14 Nov '20

alsa-project/alsa-utils issue #72 was opened from atreyasha: Hello, thank you for developing `alsa-utils`. I have a specific question pertaining to `amixer`. ## Current behaviour When starting `X11` and `i3` (my window manager), I would like to have all audio channels (speaker, headphones, etc.) muted. At the moment, I have the following command in my `i3` configuration file: ``` amixer -q sset Master mute ``` And as expected, the Master channel is indeed muted after starting `X11`. However, when I connect or disconnect my headphones, the volume automatically reverts back to the volume that was saved during the last boot. ## Expected behaviour I would like all volume channels to stay muted. So for example after starting `X11` and `i3`, I would like to see that both my speakers and headphones are muted, even when I plug/unplug the jack. Would you have any recommendations on how to do this? I was considering manually muting both headphones and speakers with `amixer` and then re-using the `alsa` state file during boots. But I am unsure because this seems like a hack'ish solution. ## System specifications OS/Kernel: Linux lt410 5.9.8-arch1-1 #1 SMP PREEMPT Tue, 10 Nov 2020 22:44:11 +0000 x86_64 GNU/Linux alsa-utils: 1.2.4-2 (installed via `pacman` package manager) wm: `i3-gaps` Issue URL : https://github.com/alsa-project/alsa-utils/issues/72 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 00/16] alsa-gobject: timer: enhancement for error reporting
by GitHub pull_request - opened 14 Nov '20

14 Nov '20

alsa-project/alsa-gobject pull request #49 was opened from takaswie: The patchset is a part of work about #47. Current implementation uses library-wide error domain to report error. this is partly against rule of GError usage. Furthermore, the error delivers information just about errno and hard to know the cause of error. This patchset enhances error reporting. The library-wide error domain is obsoleted. A new error domain is added just for ALSATimer.UserInstance. The error domain delivers information about the cause of error. ``` Takashi Sakamoto (16): timer: user_instance: fix function comment timer: skip check of return value from g_malloc() timer: check whether method argument for GError is available timer: add checks for method arguments timer: instance-params: add checks for method arguments timer: user_instance: just return when character device is not opened timer: add GLib enumeration to report type of error for ALSATimer.UserInstance timer: user_instance: add GQuark to report error for ALSATimer.UserInstance timer: user_instance: report error due to ioctl failure timer: user_instance: report error due to open system call timer: user_instance: report error for timer instance not found timer: user_instance: report error for unattached timer instance timer: user_instance: report error for timer already attached timer: query: use GFileError to report error timer: query: code refactoring to unify open function timer: obsolete library-wide GQuark for error reporting src/timer/alsatimer-enum-types.h | 18 ++++ src/timer/alsatimer.map | 5 + src/timer/instance-params.c | 20 ++-- src/timer/instance-status.c | 3 +- src/timer/privates.h | 6 -- src/timer/query.c | 127 ++++++++++++------------ src/timer/user-instance.c | 165 +++++++++++++++++++++++-------- src/timer/user-instance.h | 4 + tests/alsatimer-enums | 8 ++ 9 files changed, 231 insertions(+), 125 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/49 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/49.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[sound:test/usb-hacks 31/48] sound/usb/endpoint.c:504:33: sparse: sparse: incorrect type in assignment (different base types)
by kernel test robot 13 Nov '20

13 Nov '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/usb-hacks head: 1f19fce1baf5f21347d1c9697ab9d48a5e25301f commit: c95d75571aefae9a8c06940486cf066a893aceb3 [31/48] ALSA: usb-audio: Create endpoint objects at parsing phase config: c6x-randconfig-s031-20201113 (attached as .config) compiler: c6x-elf-gcc (GCC) 9.3.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # apt-get install sparse # sparse version: v0.6.3-107-gaf3512a6-dirty # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound test/usb-hacks git checkout c95d75571aefae9a8c06940486cf066a893aceb3 # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' ARCH=c6x If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> "sparse warnings: (new ones prefixed by >>)" >> sound/usb/endpoint.c:504:33: sparse: sparse: incorrect type in assignment (different base types) @@ expected unsigned int syncmaxsize @@ got restricted __le16 [usertype] wMaxPacketSize @@ >> sound/usb/endpoint.c:504:33: sparse: expected unsigned int syncmaxsize >> sound/usb/endpoint.c:504:33: sparse: got restricted __le16 [usertype] wMaxPacketSize vim +504 sound/usb/endpoint.c 458 459 #define ep_type_name(type) \ 460 (type == SND_USB_ENDPOINT_TYPE_DATA ? "data" : "sync") 461 462 static int 463 add_endpoint(struct snd_usb_audio *chip, int ep_num, int type, 464 struct usb_endpoint_descriptor *desc) 465 { 466 struct snd_usb_endpoint *ep; 467 bool is_playback; 468 469 ep = snd_usb_get_endpoint(chip, ep_num); 470 if (ep) 471 return 0; 472 473 usb_audio_dbg(chip, "Creating new %s endpoint #%x\n", 474 ep_type_name(type), 475 ep_num); 476 ep = kzalloc(sizeof(*ep), GFP_KERNEL); 477 if (!ep) 478 return -ENOMEM; 479 480 ep->chip = chip; 481 spin_lock_init(&ep->lock); 482 ep->type = type; 483 ep->ep_num = ep_num; 484 INIT_LIST_HEAD(&ep->ready_playback_urbs); 485 486 is_playback = ((ep_num & USB_ENDPOINT_DIR_MASK) == USB_DIR_OUT); 487 ep_num &= USB_ENDPOINT_NUMBER_MASK; 488 if (is_playback) 489 ep->pipe = usb_sndisocpipe(chip->dev, ep_num); 490 else 491 ep->pipe = usb_rcvisocpipe(chip->dev, ep_num); 492 493 if (type == SND_USB_ENDPOINT_TYPE_SYNC) { 494 if (desc->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && 495 desc->bRefresh >= 1 && desc->bRefresh <= 9) 496 ep->syncinterval = desc->bRefresh; 497 else if (snd_usb_get_speed(chip->dev) == USB_SPEED_FULL) 498 ep->syncinterval = 1; 499 else if (desc->bInterval >= 1 && desc->bInterval <= 16) 500 ep->syncinterval = desc->bInterval - 1; 501 else 502 ep->syncinterval = 3; 503 > 504 ep->syncmaxsize = desc->wMaxPacketSize; 505 } 506 507 list_add_tail(&ep->list, &chip->ep_list); 508 return 0; 509 } 510 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH] ALSA: firewire: Clean up a locking issue in copy_resp_to_buf()
by Dan Carpenter 13 Nov '20

13 Nov '20

The spin_lock/unlock_irq() functions cannot be nested. The problem is that presumably we would want the IRQs to be re-enabled on the second call the spin_unlock_irq() but instead it will be enabled at the first call so IRQs will be enabled earlier than expected. In this situation the copy_resp_to_buf() function is only called from one function and it is called with IRQs disabled. We can just use the regular spin_lock/unlock() functions. Fixes: 555e8a8f7f14 ("ALSA: fireworks: Add command/response functionality into hwdep interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/fireworks/fireworks_transaction.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/fireworks/fireworks_transaction.c b/sound/firewire/fireworks/fireworks_transaction.c index 0f533f5bd960..9f8c53b39f95 100644 --- a/sound/firewire/fireworks/fireworks_transaction.c +++ b/sound/firewire/fireworks/fireworks_transaction.c @@ -123,7 +123,7 @@ copy_resp_to_buf(struct snd_efw *efw, void *data, size_t length, int *rcode) t = (struct snd_efw_transaction *)data; length = min_t(size_t, be32_to_cpu(t->length) * sizeof(u32), length); - spin_lock_irq(&efw->lock); + spin_lock(&efw->lock); if (efw->push_ptr < efw->pull_ptr) capacity = (unsigned int)(efw->pull_ptr - efw->push_ptr); @@ -190,7 +190,7 @@ handle_resp_for_user(struct fw_card *card, int generation, int source, copy_resp_to_buf(efw, data, length, rcode); end: - spin_unlock_irq(&instances_lock); + spin_unlock(&instances_lock); } static void -- 2.28.0

3 2

[PATCH 0/4] ASoC: use inclusive language for bclk/fsync/topology
by Pierre-Louis Bossart 13 Nov '20

13 Nov '20

The SOF (Sound Open Firmware) tree contains a lot of references in topology files to 'codec_slave'/'codec_master' terms, which in turn come from alsa-lib and ALSA/ASoC topology support at the kernel level. These terms are no longer compatible with the guidelines adopted by the kernel community [1], standard organizations, and need to change in backwards-compatible ways. The main/secondary terms typically suggested in guidelines don't mean anything for clocks, this patchset suggests instead the use of 'provider' and 'consumer' terms, with the 'codec' prefix kept to make it clear that the codec is the reference. The CM/CS suffixes are also replaced by CP/CC. It can be argued that the change of suffix is invasive, but finding a replacement that keeps the M and S shortcuts has proven difficult in quite a few contexts. The previous definitions are kept for backwards-compatibility so this change should not have any functional impact. It is suggested that new contributions only use the new terms but there is no requirement to transition immediately to the new definitions for existing code. Intel will however update all its past contributions related to bit clock/frame sync configurations immediately. This patchset contains the kernel changes only, the alsa-lib changes were shared separately. Feedback welcome ~Pierre [1] https://lkml.org/lkml/2020/7/4/229 Pierre-Louis Bossart (4): ASoC: topology: use inclusive language for bclk and fsync ASoC: SOF: use inclusive language for bclk and fsync ASoC: Intel: atom: use inclusive language for SSP bclk/fsync ASoC: Intel: keembay: use inclusive language for bclk and fsync include/sound/soc-dai.h | 32 +++++++++++++++--------- include/sound/sof/dai.h | 16 ++++++++---- include/uapi/sound/asoc.h | 22 ++++++++++------ sound/soc/intel/atom/sst-atom-controls.c | 12 ++++----- sound/soc/intel/atom/sst-atom-controls.h | 4 +-- sound/soc/intel/keembay/kmb_platform.c | 22 ++++++++-------- sound/soc/intel/keembay/kmb_platform.h | 8 +++--- sound/soc/soc-topology.c | 24 +++++++++--------- sound/soc/sof/topology.c | 18 ++++++------- 9 files changed, 89 insertions(+), 69 deletions(-) -- 2.25.1

2 5

[PATCH] ASoC: rt1015: increase the time to detect BCLK
by shumingf＠realtek.com 13 Nov '20

13 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> To meet the most platform, the detection time should be increased to avoid that the flushing DAC data fails. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1015.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 32d748cc5ae3..ac4c9f43b338 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -566,7 +566,7 @@ static void rt1015_flush_work(struct work_struct *work) struct rt1015_priv *rt1015 = container_of(work, struct rt1015_priv, flush_work.work); struct snd_soc_component *component = rt1015->component; - unsigned int val, i = 0, count = 20; + unsigned int val, i = 0, count = 200; while (i < count) { usleep_range(1000, 1500); -- 2.29.0

2 1

[PATCH] ona10iv: add ona10iv smart PA kernel driver
by Jamie Meacham 13 Nov '20

13 Nov '20

From: Jamie Meacham <jamie.meacham(a)onsemi.com> add ona10iv smart PA kernel driver Signed-off-by: Jamie Meacham <jamie.meacham(a)onsemi.com> --- sound/soc/codecs/Kconfig | 8 sound/soc/codecs/Makefile | 2 sound/soc/codecs/ona10iv.c | 699 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/ona10iv.h | 632 +++++++++++++++++++++++++++++++ 4 files changed, 1341 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5791b7056..53aa98f5c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -136,6 +136,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_NAU8824 imply SND_SOC_NAU8825 imply SND_SOC_HDMI_CODEC + imply SND_SOC_ONA10IV imply SND_SOC_PCM1681 imply SND_SOC_PCM1789_I2C imply SND_SOC_PCM179X_I2C @@ -920,6 +921,13 @@ config SND_SOC_MSM8916_WCD_DIGITAL tristate "Qualcomm MSM8916 WCD DIGITAL Codec" select REGMAP_MMIO +config SND_SOC_ONA10IV + tristate "ON Semiconductor ONA10IV Speaker Amplifier" + depends on I2C + help + Adds support for the ON Semiconductor ONA10IV + speaker amplifier with I/V feedback + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 11ce98c25..c11aa20e7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -135,6 +135,7 @@ snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi-codec.o +snd-soc-ona10iv-objs := ona10iv.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1789-codec-objs := pcm1789.o snd-soc-pcm1789-i2c-objs := pcm1789-i2c.o @@ -444,6 +445,7 @@ obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_ONA10IV) += snd-soc-ona10iv.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o obj-$(CONFIG_SND_SOC_PCM1789_I2C) += snd-soc-pcm1789-i2c.o diff --git a/sound/soc/codecs/ona10iv.c b/sound/soc/codecs/ona10iv.c new file mode 100644 index 000000000..53db45adc --- /dev/null +++ b/sound/soc/codecs/ona10iv.c @@ -0,0 +1,699 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * ALSA SoC ON Semiconductor ONA10IV 16W Digital Input Mono Class-D + * Audio Amplifier with Speaker I/V Sense + * + * Copyright (C) 2020 ON Semiconductor - https://www.onsemi.com/ + * Author: Jamie Meacham <jamie.meacham(a)onsemi.com> + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/err.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "ona10iv.h" + +/////////////////////////////////////////////////////// +// Local function prototypes +/////////////////////////////////////////////////////// +static void param_errcheck(int retval, struct device *dev, + const char *errmsg, int param_val); + +/////////////////////////////////////////////////////// +// Local structure definitions +/////////////////////////////////////////////////////// +struct ona10iv_priv { + struct regmap *regmap; + struct snd_soc_component *component; + int volume; +}; + +/////////////////////////////////////////////////////// +// Function implementations +/////////////////////////////////////////////////////// +static void ona10iv_reset(struct ona10iv_priv *ona10iv) +{ + snd_soc_component_write(ona10iv->component, ONA10IV_REG_PWR_CTRL, + ONA10IV_RST); +} + +static int ona10iv_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + int from_state; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + dev_dbg(component->dev, "ONA10IV-->active\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY_RELEASE); + break; + + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + dev_dbg(component->dev, "ONA10IV-->standby\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); + break; + + case SND_SOC_BIAS_OFF: + dev_dbg(component->dev, "ONA10IV-->soft shut-down\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_SHUTDOWN | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); + + break; + default: + return -EINVAL; + } + + if (ret < 0) + return ret; + + return 0; +} + +#ifdef CONFIG_PM +static int ona10iv_codec_suspend(struct snd_soc_component *component) +{ + int from_state; + int ret; + + dev_dbg(component->dev, "ONA10IV-->suspend (standby)\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); // pause for volume ramp to complete + + if (ret < 0) + return ret; + + return 0; +} + +static int ona10iv_codec_resume(struct snd_soc_component *component) +{ + int from_state; + int ret; + + dev_dbg(component->dev, "ONA10IV-->resume (active)\n"); + from_state = snd_soc_component_get_bias_level(component); + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + + if (from_state == SND_SOC_BIAS_ON) + msleep(255); // pause for volume ramp to complete + + if (ret < 0) + return ret; + + return 0; +} +#else +#define ona10iv_codec_suspend NULL +#define ona10iv_codec_resume NULL +#endif + +static int ona10iv_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_dbg(component->dev, "ONA10IV-->post power-up\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY_RELEASE); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_dbg(component->dev, "ONA10IV-->pre power-down\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_NORMAL | ONA10IV_STBY); + msleep(255); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, "ONA10IV-->post power-down\n"); + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_PWR_CTRL, ONA10IV_PWR_STATE_MASK, + ONA10IV_SD_N_SHUTDOWN | ONA10IV_STBY_RELEASE); + break; + default: + dev_err(component->dev, "Not supported evevt\n"); + ret = -EINVAL; + } + + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", ONA10IV_REG_DATAO_CTRL2, 7, 1, 0); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", ONA10IV_REG_DATAO_CTRL2, 6, 1, 0); + +static const struct snd_soc_dapm_widget ona10iv_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ona10iv DAI", "DAI Playback", + 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SWITCH("ISENSE", ONA10IV_REG_DATAO_CTRL2, + 7, 0, &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", ONA10IV_REG_DATAO_CTRL2, + 7, 0, &vsense_switch), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, + 0, 0, ona10iv_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON"), + SND_SOC_DAPM_ADC("VMON", "DAI Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("IMON", "DAI Capture", SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route ona10iv_audio_map[] = { + {"DAC", NULL, "ona10iv DAI"}, + {"OUT", NULL, "DAC"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static int ona10iv_mute(struct snd_soc_dai *dai, int mute, int dir) +{ + int ret; + struct snd_soc_component *component = dai->component; + struct ona10iv_priv *ona10iv; + + ona10iv = snd_soc_component_get_drvdata(component); + + /* using "mute" bit can cause a pop. Instead store volume setting + * and set volume to minimum allowing a smooth ramp-down. + */ + + dev_dbg(component->dev, "mute dir: %d val: %d\n", dir, mute); + + if (dir != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + if (mute) { + ret = regmap_read(ona10iv->regmap, ONA10IV_REG_MAX_VOL, + &ona10iv->volume); + if (ret < 0) { + dev_err(component->dev, "Failed to read: %d\n", ret); + return ret; + } + ret = regmap_write(ona10iv->regmap, ONA10IV_REG_MAX_VOL, + ONA10IV_VOL_AMP_MUTE); + } else { + ret = regmap_write(ona10iv->regmap, ONA10IV_REG_MAX_VOL, + ona10iv->volume); + } + + return ret; +} + +static int ona10iv_set_bitwidth(struct ona10iv_priv *ona10iv, int format) +{ + struct snd_soc_component *component = ona10iv->component; + int slot_width, sample_width; + int ret = 0; + + dev_dbg(component->dev, + "ONA10IV-->set bitwidth for format: %d\n", format); + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + sample_width = ONA10IV_SAMPW_16; + slot_width = ONA10IV_SLOTW_16; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + sample_width = ONA10IV_SAMPW_24; + slot_width = ONA10IV_SLOTW_24; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + sample_width = ONA10IV_SAMPW_32; + slot_width = ONA10IV_SLOTW_32; + break; + default: + ret = -EINVAL; + } + param_errcheck(ret, component->dev, + "error no case match for format", format); + + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL2, ONA10IV_SAMPW_MASK, + sample_width); + param_errcheck(ret, component->dev, + "error writing sample width", sample_width); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL2, ONA10IV_SLOTW_MASK, + slot_width); + param_errcheck(ret, component->dev, + "error writing slot width", slot_width); + if (ret < 0) + return ret; + + return 0; +} + +static int ona10iv_set_samplerate(struct ona10iv_priv *ona10iv, + int samplerate) +{ + struct snd_soc_component *component = ona10iv->component; + int rate_val; + int ret = 0; + + dev_dbg(component->dev, "ONA10IV-->samplerate: %d\n", samplerate); + + switch (samplerate) { + case 16000: + rate_val = ONA10IV_FS_16KHZ; + break; + case 22050: + rate_val = ONA10IV_FS_22KHZ; + break; + case 32000: + rate_val = ONA10IV_FS_32KHZ; + break; + case 44100: + rate_val = ONA10IV_FS_44KHZ; + break; + case 48000: + rate_val = ONA10IV_FS_48KHZ; + break; + case 96000: + rate_val = ONA10IV_FS_96KHZ; + break; + default: + ret = -EINVAL; + } + param_errcheck(ret, component->dev, + "error no case match for rate", samplerate); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL1, ONA10IV_FS_MASK, + rate_val); + param_errcheck(ret, component->dev, + "error writing rate", rate_val); + if (ret < 0) + return ret; + + return 0; +} + +static int ona10iv_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct ona10iv_priv *ona10iv = + snd_soc_component_get_drvdata(component); + int ret; + + dev_dbg(component->dev, "ONA10IV-->setting hw params\n"); + + ret = ona10iv_set_bitwidth(ona10iv, params_format(params)); + param_errcheck(ret, component->dev, + "error setting bitwidth for format", + params_format(params)); + if (ret) + return ret; + + ret = ona10iv_set_samplerate(ona10iv, params_rate(params)); + param_errcheck(ret, component->dev, + "error setting rate", params_rate(params)); + if (ret) + return ret; + + return 0; +} + +static int ona10iv_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + int left_slot, right_slot; + int slot_val; + int i_enable, v_enable; + int i_slot, v_slot; + int reg_val; + int ret; + + dev_dbg(component->dev, + "set dai txm: %x rxm %x slots: %d width: %d\n", + tx_mask, rx_mask, slots, slot_width); + + if (tx_mask == 0 || slots != 2) + return -EINVAL; + + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + + + // right slot checked but not used. + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + switch (left_slot) { + case 1: + slot_val = ONA10IV_A_SLOT_1; + break; + case 2: + slot_val = ONA10IV_A_SLOT_2; + break; + } + + ret = snd_soc_component_update_bits(component, + ONA10IV_REG_DAI_CTRL3, ONA10IV_A_SLOT_MSK, + slot_val); + param_errcheck(ret, component->dev, + "error setting transmit slot", slot_val); + if (ret < 0) + return ret; + + i_slot = 0; + v_slot = 0; + if (rx_mask == 0) { + i_enable = 0; + v_enable = 0; + } else { + i_enable = 1; + i_slot = __ffs(rx_mask); + rx_mask &= ~(1 << i_slot); + if (rx_mask != 0) { + v_enable = 1; + v_slot = __ffs(rx_mask); + rx_mask &= ~(1 << v_slot); + } + } + + if (rx_mask != 0 || i_slot >= slots || v_slot >= slots) + return -EINVAL; + + if (i_enable) { + reg_val = ONA10IV_I_DATAO_TX_ENABLE; + switch (i_slot) { + case 1: + reg_val |= ONA10IV_I_SLOT_1; + break; + case 2: + reg_val |= ONA10IV_I_SLOT_2; + break; + } + ret = snd_soc_component_write(component, + ONA10IV_REG_DATAO_CTRL2, reg_val); + param_errcheck(ret, component->dev, + "error setting current sense slot and enable", + reg_val); + if (ret < 0) + return ret; + } + + if (v_enable) { + reg_val = ONA10IV_V_DATAO_TX_ENABLE; + switch (v_slot) { + case 1: + reg_val |= ONA10IV_V_SLOT_1; + break; + case 2: + reg_val |= ONA10IV_V_SLOT_2; + break; + } + ret = snd_soc_component_write(component, + ONA10IV_REG_DATAO_CTRL2, reg_val); + param_errcheck(ret, component->dev, + "error setting voltage sense slot and enable", + reg_val); + if (ret < 0) + return ret; + } + + return 0; +} + +static void param_errcheck(int retval, struct device *dev, + const char *errmsg, int param_val) +{ + if (retval < 0) + dev_dbg(dev, "Error %d: %s = %d\n", retval, errmsg, param_val); +} + +static struct snd_soc_dai_ops ona10iv_dai_ops = { + .mute_stream = ona10iv_mute, + .hw_params = ona10iv_hw_params, + .set_fmt = NULL, + .set_tdm_slot = ona10iv_set_dai_tdm_slot, + .startup = NULL, + .shutdown = NULL, + .prepare = NULL, +}; + +#define ONA10IV_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +#define ONA10IV_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +static struct snd_soc_dai_driver ona10iv_dai[] = { + { + .name = "ona10iv DAI", + .id = 0, + .playback = { + .stream_name = "DAI Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ONA10IV_RATES, + .formats = ONA10IV_FORMATS, + }, + .capture = { + .stream_name = "DAI Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ONA10IV_RATES, + .formats = ONA10IV_FORMATS, + }, + .ops = &ona10iv_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int ona10iv_codec_probe(struct snd_soc_component *component) +{ + struct ona10iv_priv *ona10iv = + snd_soc_component_get_drvdata(component); + + ona10iv->component = component; + + dev_dbg(component->dev, "ONA10IV-->device probe\n"); + + ona10iv_reset(ona10iv); + + return 0; +} + +static DECLARE_TLV_DB_MINMAX(ona10iv_digital_tlv, 0, 1600); +static DECLARE_TLV_DB_MINMAX(ona10iv_playback_volume, -9562, 0); + +static const struct snd_kcontrol_new ona10iv_snd_controls[] = { + SOC_SINGLE_TLV("Playback Volume", ONA10IV_REG_MAX_VOL, 0, 255, 1, + ona10iv_playback_volume), + SOC_SINGLE_TLV("Amp Gain", ONA10IV_REG_GAIN_CTRL1, 0, 31, 1, + ona10iv_digital_tlv), +}; + +static const struct snd_soc_component_driver ona10iv_component_driver = { + .probe = ona10iv_codec_probe, + .suspend = ona10iv_codec_suspend, + .resume = ona10iv_codec_resume, + .set_bias_level = ona10iv_set_bias_level, + .controls = ona10iv_snd_controls, + .num_controls = ARRAY_SIZE(ona10iv_snd_controls), + .dapm_widgets = ona10iv_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ona10iv_dapm_widgets), + .dapm_routes = ona10iv_audio_map, + .num_dapm_routes = ARRAY_SIZE(ona10iv_audio_map), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .use_pmdown_time = 1, +}; + +static const struct reg_default ona10iv_reg_defaults[] = { + { ONA10IV_REG_PWR_CTRL, 0x00 }, + { ONA10IV_REG_INT_MASK, 0x00 }, + { ONA10IV_REG_ERR_CTRL, 0x00 }, + { ONA10IV_REG_DAI_CTRL1, 0x06 }, + { ONA10IV_REG_DAI_CTRL2, 0x55 }, + { ONA10IV_REG_DAI_CTRL3, 0x00 }, + { ONA10IV_REG_MAX_VOL, 0x00 }, + { ONA10IV_REG_VOL_CTRL, 0x0E }, + { ONA10IV_REG_GAIN_CTRL1, 0x00 }, + { ONA10IV_REG_EMI_CTRL, 0x1C }, + { ONA10IV_REG_AGC_BATT, 0x00 }, + { ONA10IV_REG_AGC_CTRL1, 0x00 }, + { ONA10IV_REG_AGC_CTRL2, 0x00 }, + { ONA10IV_REG_AGC_CTRL3, 0x00 }, + { ONA10IV_REG_MAGC_CTRL1, 0x00 }, + { ONA10IV_REG_MAGC_CTRL2, 0x00 }, + { ONA10IV_REG_MAGC_CTRL3, 0x00 }, + { ONA10IV_REG_SENSE_CTRL, 0x00 }, + { ONA10IV_REG_DATAO_CTRL1, 0x00 }, + { ONA10IV_REG_DATAO_CTRL2, 0x08 }, + { ONA10IV_REG_DATAO_CTRL3, 0x02 }, +}; + +static bool ona10iv_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ONA10IV_REG_PWR_CTRL: /* reset bit is self clearing */ + case ONA10IV_REG_INT_FLAG: + case ONA10IV_REG_ERR_STAT: + case ONA10IV_REG_T_SENSE_OUT1: + case ONA10IV_REG_T_SENSE_OUT2:/* Sticky interrupt flags */ + return true; + } + + return false; +} + +static bool ona10iv_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ONA10IV_REG_DEVICE_ID: + case ONA10IV_REG_ERR_STAT: + case ONA10IV_REG_T_SENSE_OUT1: + case ONA10IV_REG_T_SENSE_OUT2: + return false; + } + + return true; +} + +static const struct regmap_config ona10iv_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = ona10iv_writeable, + .volatile_reg = ona10iv_volatile, + .reg_defaults = ona10iv_reg_defaults, + .max_register = ONA10IV_REG_DATAO_CTRL3, + .num_reg_defaults = ARRAY_SIZE(ona10iv_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static int ona10iv_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct ona10iv_priv *ona10iv; + int ret, reg; + + ona10iv = devm_kzalloc(&i2c->dev, sizeof(*ona10iv), GFP_KERNEL); + if (ona10iv == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, ona10iv); + + ona10iv->regmap = devm_regmap_init_i2c(i2c, &ona10iv_i2c_regmap); + if (IS_ERR(ona10iv->regmap)) { + ret = PTR_ERR(ona10iv->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + ret = regmap_read(ona10iv->regmap, ONA10IV_REG_DEVICE_ID, &reg); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read: %d\n", ret); + return ret; + } + dev_dbg(&i2c->dev, "device revision: %x\n", reg); + + ona10iv->volume = 0; + + ret = devm_snd_soc_register_component(&i2c->dev, + &ona10iv_component_driver, + ona10iv_dai, ARRAY_SIZE(ona10iv_dai)); + if (ret < 0) + dev_err(&i2c->dev, "Failed to register component: %d\n", ret); + return ret; +} + +static int ona10iv_i2c_remove(struct i2c_client *client) +{ + return 0; +} + +static const struct i2c_device_id ona10iv_i2c_id[] = { + { "ona10iv", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ona10iv_i2c_id); + +static const struct of_device_id ona10iv_of_match[] = { + { .compatible = "onnn,ona10iv" }, + {}, +}; +MODULE_DEVICE_TABLE(of, ona10iv_of_match); + +static struct i2c_driver ona10iv_i2c_driver = { + .driver = { + .name = "ona10iv", + .of_match_table = of_match_ptr(ona10iv_of_match), + }, + .probe = ona10iv_i2c_probe, + .remove = ona10iv_i2c_remove, + .id_table = ona10iv_i2c_id, +}; +module_i2c_driver(ona10iv_i2c_driver); + +MODULE_AUTHOR("Jamie Meacham <jamie.meacham(a)onsemi.com>"); +MODULE_DESCRIPTION("ONA10IV I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/ona10iv.h b/sound/soc/codecs/ona10iv.h new file mode 100644 index 000000000..2c46088ac --- /dev/null +++ b/sound/soc/codecs/ona10iv.h @@ -0,0 +1,632 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC ONA10IV codec driver + * + * Copyright (C) 2020 ON Semiconductor - https://www.onsemi.com/ + */ +#ifndef __ONA10IV__ +#define __ONA10IV__ + + +/** ONA10IV Register Map **/ +#define ONA10IV_REG_DEVICE_ID (0x00) + +#define ONA10IV_REG_PWR_CTRL (0x01) +#define ONA10IV_TFB_PD (0x01<<7) +#define ONA10IV_BATT_CFG (0x01<<6) +#define ONA10IV_AGC_PD (0x01<<5) +#define ONA10IV_IVSNS_PD (0x01<<4) +#define ONA10IV_AMP_PD (0x01<<3) +#define ONA10IV_SD_N_SHUTDOWN (0x00<<2) +#define ONA10IV_SD_N_NORMAL (0x01<<2) +#define ONA10IV_STBY_RELEASE (0x00<<1) +#define ONA10IV_STBY (0x01<<1) +#define ONA10IV_PWR_STATE_MASK (0x03<<1) +#define ONA10IV_RST_RELEASE (0x00<<0) +#define ONA10IV_RST (0x01<<0) + +#define ONA10IV_REG_INT_FLAG (0x02) +#define ONA10IV_TFB_I (0x01<<5) +#define ONA10IV_AGC_I (0x01<<4) +#define ONA10IV_CERR_I (0x01<<3) +#define ONA10IV_TERR_I (0x01<<2) +#define ONA10IV_IERR_I (0x01<<1) +#define ONA10IV_VERR_I (0x01<<0) + +#define ONA10IV_REG_INT_MASK (0x03) +#define ONA10IV_ALL_MASK (0x01<<6) +#define ONA10IV_TFB_M (0x01<<5) +#define ONA10IV_AGC_M (0x01<<4) +#define ONA10IV_CERR_M (0x01<<3) +#define ONA10IV_TERR_M (0x01<<2) +#define ONA10IV_IERR_M (0x01<<1) +#define ONA10IV_VERR_M (0x01<<0) + +#define ONA10IV_REG_ERR_STAT (0x04) +#define ONA10IV_TFB (0x01<<5) +#define ONA10IV_AGC (0x01<<4) +#define ONA10IV_CERR (0x01<<3) +#define ONA10IV_TERR (0x01<<2) +#define ONA10IV_IERR (0x01<<1) +#define ONA10IV_VERR (0x01<<0) + +#define ONA10IV_REG_ERR_CTRL (0x05) +#define ONA10IV_MRCV (0x01<<5) +#define ONA10IV_MAX_ARCV_ALWAYS (0x00<<3) +#define ONA10IV_MAX_ARCV_1_ATT (0x01<<3) +#define ONA10IV_MAX_ARCV_3_ATT (0x02<<3) +#define ONA10IV_MAX_ARCV_7_ATT (0x03<<3) +#define ONA10IV_ARCV_T (0x01<<2) +#define ONA10IV_ARCV_I (0x01<<1) +#define ONA10IV_ARCV_C (0x01<<0) + +#define ONA10IV_REG_DAI_CTRL1 (0x06) +#define ONA10IV_BEDGE_DAI_FALLING (0x00<<7) +#define ONA10IV_BEDGE_DAI_RISING (0x01<<7) +#define ONA10IV_FORMAT_I2S_DELAY (0x00<<6) +#define ONA10IV_FORMAT_NO_DELAY (0x01<<6) +#define ONA10IV_DAI_I2S (0x00<<4) +#define ONA10IV_DAI_TDM2 (0x01<<4) +#define ONA10IV_DAI_TDM4 (0x02<<4) +#define ONA10IV_DAI_TDM8 (0x03<<4) +#define ONA10IV_FS_16KHZ (0x02<<0) +#define ONA10IV_FS_22KHZ (0x03<<0) +#define ONA10IV_FS_32KHZ (0x04<<0) +#define ONA10IV_FS_44KHZ (0x05<<0) +#define ONA10IV_FS_48KHZ (0x06<<0) +#define ONA10IV_FS_96KHZ (0x08<<0) +#define ONA10IV_FS_MASK (0x0F<<0) + +#define ONA10IV_REG_DAI_CTRL2 (0x07) +#define ONA10IV_SAMPW_16 (0x00<<6) +#define ONA10IV_SAMPW_24 (0x01<<6) +#define ONA10IV_SAMPW_32 (0x02<<6) +#define ONA10IV_SAMPW_MASK (0x03<<6) +#define ONA10IV_SLOTW_16 (0x00<<4) +#define ONA10IV_SLOTW_24 (0x01<<4) +#define ONA10IV_SLOTW_32 (0x02<<4) +#define ONA10IV_SLOTW_MASK (0x03<<4) +#define ONA10IV_FRM_DLY_0 (0x00<<2) +#define ONA10IV_FRM_DLY_1 (0x01<<2) +#define ONA10IV_FRM_DLY_2 (0x02<<2) +#define ONA10IV_FRM_DLY_3 (0x03<<2) +#define ONA10IV_FRM_POL_NORMAL (0x00<<1) +#define ONA10IV_FRM_POL_INVERT (0x01<<1) +#define ONA10IV_FRCK_MODE_50 (0x00<<0) +#define ONA10IV_FRCK_MODE_PULSE (0x01<<0) + +#define ONA10IV_REG_DAI_CTRL3 (0x08) +#define ONA10IV_DAC_MODE_1 (0x00<<4) +#define ONA10IV_DAC_MODE_2 (0x01<<4) +#define ONA10IV_DAC_MODE_3 (0x02<<4) +#define ONA10IV_DAC_MODE_NOT_USED (0x03<<4) +#define ONA10IV_PDM_MODE_PCM (0x00<<3) +#define ONA10IV_PDM_MODE_PDM (0x01<<3) +#define ONA10IV_A_SLOT_1 (0x00<<0) +#define ONA10IV_A_SLOT_2 (0x01<<0) +#define ONA10IV_A_SLOT_3 (0x02<<0) +#define ONA10IV_A_SLOT_4 (0x03<<0) +#define ONA10IV_A_SLOT_5 (0x04<<0) +#define ONA10IV_A_SLOT_6 (0x05<<0) +#define ONA10IV_A_SLOT_7 (0x06<<0) +#define ONA10IV_A_SLOT_8 (0x07<<0) +#define ONA10IV_A_SLOT_MSK (0x07<<0) + +#define ONA10IV_REG_MAX_VOL (0x09) +#define ONA10IV_VOL_0DB (0x00) +#define ONA10IV_VOL_MINUS_0_375DB (0x01) +#define ONA10IV_VOL_MINUS_0_750DB (0x02) +#define ONA10IV_VOL_MINUS_1_125DB (0x03) +#define ONA10IV_VOL_MINUS_1_500DB (0x04) +#define ONA10IV_VOL_MINUS_1_875DB (0x05) +#define ONA10IV_VOL_MINUS_2_250DB (0x06) +#define ONA10IV_VOL_MINUS_2_625DB (0x07) +#define ONA10IV_VOL_MINUS_3_000DB (0x08) +#define ONA10IV_VOL_MINUS_3_375DB (0x09) +#define ONA10IV_VOL_MINUS_3_750DB (0x0A) +#define ONA10IV_VOL_MINUS_4_125DB (0x0B) +#define ONA10IV_VOL_MINUS_4_500DB (0x0C) +#define ONA10IV_VOL_MINUS_4_875DB (0x0D) +#define ONA10IV_VOL_MINUS_5_250DB (0x0E) +#define ONA10IV_VOL_MINUS_5_625DB (0x0F) +#define ONA10IV_VOL_MINUS_6_000DB (0x10) +#define ONA10IV_VOL_MINUS_6_375DB (0x11) +#define ONA10IV_VOL_MINUS_6_750DB (0x12) +#define ONA10IV_VOL_MINUS_7_125DB (0x13) +#define ONA10IV_VOL_MINUS_7_500DB (0x14) +#define ONA10IV_VOL_MINUS_7_875DB (0x15) +#define ONA10IV_VOL_MINUS_8_250DB (0x16) +#define ONA10IV_VOL_MINUS_8_625DB (0x17) +#define ONA10IV_VOL_MINUS_9_000DB (0x18) +#define ONA10IV_VOL_MINUS_9_375DB (0x19) +#define ONA10IV_VOL_MINUS_9_750DB (0x1A) +#define ONA10IV_VOL_MINUS_10_125DB (0x1B) +#define ONA10IV_VOL_MINUS_10_500DB (0x1C) +#define ONA10IV_VOL_MINUS_10_875DB (0x1D) +#define ONA10IV_VOL_MINUS_11_250DB (0x1E) +#define ONA10IV_VOL_MINUS_11_625DB (0x1F) +#define ONA10IV_VOL_MINUS_12_000DB (0x20) +#define ONA10IV_VOL_MINUS_12_375DB (0x21) +#define ONA10IV_VOL_MINUS_12_750DB (0x22) +#define ONA10IV_VOL_MINUS_13_125DB (0x23) +#define ONA10IV_VOL_MINUS_13_500DB (0x24) +#define ONA10IV_VOL_MINUS_13_875DB (0x25) +#define ONA10IV_VOL_MINUS_14_250DB (0x26) +#define ONA10IV_VOL_MINUS_14_625DB (0x27) +#define ONA10IV_VOL_MINUS_15_000DB (0x28) +#define ONA10IV_VOL_MINUS_15_375DB (0x29) +#define ONA10IV_VOL_MINUS_15_750DB (0x2A) +#define ONA10IV_VOL_MINUS_16_125DB (0x2B) +#define ONA10IV_VOL_MINUS_16_500DB (0x2C) +#define ONA10IV_VOL_MINUS_16_875DB (0x2D) +#define ONA10IV_VOL_MINUS_17_250DB (0x2E) +#define ONA10IV_VOL_MINUS_17_625DB (0x2F) +#define ONA10IV_VOL_MINUS_18_000DB (0x30) +#define ONA10IV_VOL_MINUS_18_375DB (0x31) +#define ONA10IV_VOL_MINUS_18_750DB (0x32) +#define ONA10IV_VOL_MINUS_19_125DB (0x33) +#define ONA10IV_VOL_MINUS_19_500DB (0x34) +#define ONA10IV_VOL_MINUS_19_875DB (0x35) +#define ONA10IV_VOL_MINUS_20_250DB (0x36) +#define ONA10IV_VOL_MINUS_20_625DB (0x37) +#define ONA10IV_VOL_MINUS_21_000DB (0x38) +#define ONA10IV_VOL_MINUS_21_375DB (0x39) +#define ONA10IV_VOL_MINUS_21_750DB (0x3A) +#define ONA10IV_VOL_MINUS_22_125DB (0x3B) +#define ONA10IV_VOL_MINUS_22_500DB (0x3C) +#define ONA10IV_VOL_MINUS_22_875DB (0x3D) +#define ONA10IV_VOL_MINUS_23_250DB (0x3E) +#define ONA10IV_VOL_MINUS_23_625DB (0x3F) +#define ONA10IV_VOL_MINUS_24_000DB (0x40) +#define ONA10IV_VOL_MINUS_24_375DB (0x41) +#define ONA10IV_VOL_MINUS_24_750DB (0x42) +#define ONA10IV_VOL_MINUS_25_125DB (0x43) +#define ONA10IV_VOL_MINUS_25_500DB (0x44) +#define ONA10IV_VOL_MINUS_25_875DB (0x45) +#define ONA10IV_VOL_MINUS_26_250DB (0x46) +#define ONA10IV_VOL_MINUS_26_625DB (0x47) +#define ONA10IV_VOL_MINUS_27_000DB (0x48) +#define ONA10IV_VOL_MINUS_27_375DB (0x49) +#define ONA10IV_VOL_MINUS_27_750DB (0x4A) +#define ONA10IV_VOL_MINUS_28_125DB (0x4B) +#define ONA10IV_VOL_MINUS_28_500DB (0x4C) +#define ONA10IV_VOL_MINUS_28_875DB (0x4D) +#define ONA10IV_VOL_MINUS_29_250DB (0x4E) +#define ONA10IV_VOL_MINUS_29_625DB (0x4F) +#define ONA10IV_VOL_MINUS_30_000DB (0x50) +#define ONA10IV_VOL_MINUS_30_375DB (0x51) +#define ONA10IV_VOL_MINUS_30_750DB (0x52) +#define ONA10IV_VOL_MINUS_31_125DB (0x53) +#define ONA10IV_VOL_MINUS_31_500DB (0x54) +#define ONA10IV_VOL_MINUS_31_875DB (0x55) +#define ONA10IV_VOL_MINUS_32_250DB (0x56) +#define ONA10IV_VOL_MINUS_32_625DB (0x57) +#define ONA10IV_VOL_MINUS_33_000DB (0x58) +#define ONA10IV_VOL_MINUS_33_375DB (0x59) +#define ONA10IV_VOL_MINUS_33_750DB (0x5A) +#define ONA10IV_VOL_MINUS_34_125DB (0x5B) +#define ONA10IV_VOL_MINUS_34_500DB (0x5C) +#define ONA10IV_VOL_MINUS_34_875DB (0x5D) +#define ONA10IV_VOL_MINUS_35_250DB (0x5E) +#define ONA10IV_VOL_MINUS_35_625DB (0x5F) +#define ONA10IV_VOL_MINUS_36_000DB (0x60) +#define ONA10IV_VOL_MINUS_36_375DB (0x61) +#define ONA10IV_VOL_MINUS_36_750DB (0x62) +#define ONA10IV_VOL_MINUS_37_125DB (0x63) +#define ONA10IV_VOL_MINUS_37_500DB (0x64) +#define ONA10IV_VOL_MINUS_37_875DB (0x65) +#define ONA10IV_VOL_MINUS_38_250DB (0x66) +#define ONA10IV_VOL_MINUS_38_625DB (0x67) +#define ONA10IV_VOL_MINUS_39_000DB (0x68) +#define ONA10IV_VOL_MINUS_39_375DB (0x69) +#define ONA10IV_VOL_MINUS_39_750DB (0x6A) +#define ONA10IV_VOL_MINUS_40_125DB (0x6B) +#define ONA10IV_VOL_MINUS_40_500DB (0x6C) +#define ONA10IV_VOL_MINUS_40_875DB (0x6D) +#define ONA10IV_VOL_MINUS_41_250DB (0x6E) +#define ONA10IV_VOL_MINUS_41_625DB (0x6F) +#define ONA10IV_VOL_MINUS_42_000DB (0x70) +#define ONA10IV_VOL_MINUS_42_375DB (0x71) +#define ONA10IV_VOL_MINUS_42_750DB (0x72) +#define ONA10IV_VOL_MINUS_43_125DB (0x73) +#define ONA10IV_VOL_MINUS_43_500DB (0x74) +#define ONA10IV_VOL_MINUS_43_875DB (0x75) +#define ONA10IV_VOL_MINUS_44_250DB (0x76) +#define ONA10IV_VOL_MINUS_44_625DB (0x77) +#define ONA10IV_VOL_MINUS_45_000DB (0x78) +#define ONA10IV_VOL_MINUS_45_375DB (0x79) +#define ONA10IV_VOL_MINUS_45_750DB (0x7A) +#define ONA10IV_VOL_MINUS_46_125DB (0x7B) +#define ONA10IV_VOL_MINUS_46_500DB (0x7C) +#define ONA10IV_VOL_MINUS_46_875DB (0x7D) +#define ONA10IV_VOL_MINUS_47_250DB (0x7E) +#define ONA10IV_VOL_MINUS_47_625DB (0x7F) +#define ONA10IV_VOL_MINUS_48_000DB (0x80) +#define ONA10IV_VOL_MINUS_48_375DB (0x81) +#define ONA10IV_VOL_MINUS_48_750DB (0x82) +#define ONA10IV_VOL_MINUS_49_125DB (0x83) +#define ONA10IV_VOL_MINUS_49_500DB (0x84) +#define ONA10IV_VOL_MINUS_49_875DB (0x85) +#define ONA10IV_VOL_MINUS_50_250DB (0x86) +#define ONA10IV_VOL_MINUS_50_625DB (0x87) +#define ONA10IV_VOL_MINUS_51_000DB (0x88) +#define ONA10IV_VOL_MINUS_51_375DB (0x89) +#define ONA10IV_VOL_MINUS_51_750DB (0x8A) +#define ONA10IV_VOL_MINUS_52_125DB (0x8B) +#define ONA10IV_VOL_MINUS_52_500DB (0x8C) +#define ONA10IV_VOL_MINUS_52_875DB (0x8D) +#define ONA10IV_VOL_MINUS_53_250DB (0x8E) +#define ONA10IV_VOL_MINUS_53_625DB (0x8F) +#define ONA10IV_VOL_MINUS_54_000DB (0x90) +#define ONA10IV_VOL_MINUS_54_375DB (0x91) +#define ONA10IV_VOL_MINUS_54_750DB (0x92) +#define ONA10IV_VOL_MINUS_55_125DB (0x93) +#define ONA10IV_VOL_MINUS_55_500DB (0x94) +#define ONA10IV_VOL_MINUS_55_875DB (0x95) +#define ONA10IV_VOL_MINUS_56_250DB (0x96) +#define ONA10IV_VOL_MINUS_56_625DB (0x97) +#define ONA10IV_VOL_MINUS_57_000DB (0x98) +#define ONA10IV_VOL_MINUS_57_375DB (0x99) +#define ONA10IV_VOL_MINUS_57_750DB (0x9A) +#define ONA10IV_VOL_MINUS_58_125DB (0x9B) +#define ONA10IV_VOL_MINUS_58_500DB (0x9C) +#define ONA10IV_VOL_MINUS_58_875DB (0x9D) +#define ONA10IV_VOL_MINUS_59_250DB (0x9E) +#define ONA10IV_VOL_MINUS_59_625DB (0x9F) +#define ONA10IV_VOL_MINUS_60_000DB (0xA0) +#define ONA10IV_VOL_MINUS_60_375DB (0xA1) +#define ONA10IV_VOL_MINUS_60_750DB (0xA2) +#define ONA10IV_VOL_MINUS_61_125DB (0xA3) +#define ONA10IV_VOL_MINUS_61_500DB (0xA4) +#define ONA10IV_VOL_MINUS_61_875DB (0xA5) +#define ONA10IV_VOL_MINUS_62_250DB (0xA6) +#define ONA10IV_VOL_MINUS_62_625DB (0xA7) +#define ONA10IV_VOL_MINUS_63_000DB (0xA8) +#define ONA10IV_VOL_MINUS_63_375DB (0xA9) +#define ONA10IV_VOL_MINUS_63_750DB (0xAA) +#define ONA10IV_VOL_MINUS_64_125DB (0xAB) +#define ONA10IV_VOL_MINUS_64_500DB (0xAC) +#define ONA10IV_VOL_MINUS_64_875DB (0xAD) +#define ONA10IV_VOL_MINUS_65_250DB (0xAE) +#define ONA10IV_VOL_MINUS_65_625DB (0xAF) +#define ONA10IV_VOL_MINUS_66_000DB (0xB0) +#define ONA10IV_VOL_MINUS_66_375DB (0xB1) +#define ONA10IV_VOL_MINUS_66_750DB (0xB2) +#define ONA10IV_VOL_MINUS_67_125DB (0xB3) +#define ONA10IV_VOL_MINUS_67_500DB (0xB4) +#define ONA10IV_VOL_MINUS_67_875DB (0xB5) +#define ONA10IV_VOL_MINUS_68_250DB (0xB6) +#define ONA10IV_VOL_MINUS_68_625DB (0xB7) +#define ONA10IV_VOL_MINUS_69_000DB (0xB8) +#define ONA10IV_VOL_MINUS_69_375DB (0xB9) +#define ONA10IV_VOL_MINUS_69_750DB (0xBA) +#define ONA10IV_VOL_MINUS_70_125DB (0xBB) +#define ONA10IV_VOL_MINUS_70_500DB (0xBC) +#define ONA10IV_VOL_MINUS_70_875DB (0xBD) +#define ONA10IV_VOL_MINUS_71_250DB (0xBE) +#define ONA10IV_VOL_MINUS_71_625DB (0xBF) +#define ONA10IV_VOL_MINUS_72_000DB (0xC0) +#define ONA10IV_VOL_MINUS_72_375DB (0xC1) +#define ONA10IV_VOL_MINUS_72_750DB (0xC2) +#define ONA10IV_VOL_MINUS_73_125DB (0xC3) +#define ONA10IV_VOL_MINUS_73_500DB (0xC4) +#define ONA10IV_VOL_MINUS_73_875DB (0xC5) +#define ONA10IV_VOL_MINUS_74_250DB (0xC6) +#define ONA10IV_VOL_MINUS_74_625DB (0xC7) +#define ONA10IV_VOL_MINUS_75_000DB (0xC8) +#define ONA10IV_VOL_MINUS_75_375DB (0xC9) +#define ONA10IV_VOL_MINUS_75_750DB (0xCA) +#define ONA10IV_VOL_MINUS_76_125DB (0xCB) +#define ONA10IV_VOL_MINUS_76_500DB (0xCC) +#define ONA10IV_VOL_MINUS_76_875DB (0xCD) +#define ONA10IV_VOL_MINUS_77_250DB (0xCE) +#define ONA10IV_VOL_MINUS_77_625DB (0xCF) +#define ONA10IV_VOL_MINUS_78_000DB (0xD0) +#define ONA10IV_VOL_MINUS_78_375DB (0xD1) +#define ONA10IV_VOL_MINUS_78_750DB (0xD2) +#define ONA10IV_VOL_MINUS_79_125DB (0xD3) +#define ONA10IV_VOL_MINUS_79_500DB (0xD4) +#define ONA10IV_VOL_MINUS_79_875DB (0xD5) +#define ONA10IV_VOL_MINUS_80_250DB (0xD6) +#define ONA10IV_VOL_MINUS_80_625DB (0xD7) +#define ONA10IV_VOL_MINUS_81_000DB (0xD8) +#define ONA10IV_VOL_MINUS_81_375DB (0xD9) +#define ONA10IV_VOL_MINUS_81_750DB (0xDA) +#define ONA10IV_VOL_MINUS_82_125DB (0xDB) +#define ONA10IV_VOL_MINUS_82_500DB (0xDC) +#define ONA10IV_VOL_MINUS_82_875DB (0xDD) +#define ONA10IV_VOL_MINUS_83_250DB (0xDE) +#define ONA10IV_VOL_MINUS_83_625DB (0xDF) +#define ONA10IV_VOL_MINUS_84_000DB (0xE0) +#define ONA10IV_VOL_MINUS_84_375DB (0xE1) +#define ONA10IV_VOL_MINUS_84_750DB (0xE2) +#define ONA10IV_VOL_MINUS_85_125DB (0xE3) +#define ONA10IV_VOL_MINUS_85_500DB (0xE4) +#define ONA10IV_VOL_MINUS_85_875DB (0xE5) +#define ONA10IV_VOL_MINUS_86_250DB (0xE6) +#define ONA10IV_VOL_MINUS_86_625DB (0xE7) +#define ONA10IV_VOL_MINUS_87_000DB (0xE8) +#define ONA10IV_VOL_MINUS_87_375DB (0xE9) +#define ONA10IV_VOL_MINUS_87_750DB (0xEA) +#define ONA10IV_VOL_MINUS_88_125DB (0xEB) +#define ONA10IV_VOL_MINUS_88_500DB (0xEC) +#define ONA10IV_VOL_MINUS_88_875DB (0xED) +#define ONA10IV_VOL_MINUS_89_250DB (0xEE) +#define ONA10IV_VOL_MINUS_89_625DB (0xEF) +#define ONA10IV_VOL_MINUS_90_000DB (0xF0) +#define ONA10IV_VOL_MINUS_90_375DB (0xF1) +#define ONA10IV_VOL_MINUS_90_750DB (0xF2) +#define ONA10IV_VOL_MINUS_91_125DB (0xF3) +#define ONA10IV_VOL_MINUS_91_500DB (0xF4) +#define ONA10IV_VOL_MINUS_91_875DB (0xF5) +#define ONA10IV_VOL_MINUS_92_250DB (0xF6) +#define ONA10IV_VOL_MINUS_92_625DB (0xF7) +#define ONA10IV_VOL_MINUS_93_000DB (0xF8) +#define ONA10IV_VOL_MINUS_93_375DB (0xF9) +#define ONA10IV_VOL_MINUS_93_750DB (0xFA) +#define ONA10IV_VOL_MINUS_94_125DB (0xFB) +#define ONA10IV_VOL_MINUS_94_500DB (0xFC) +#define ONA10IV_VOL_MINUS_94_875DB (0xFD) +#define ONA10IV_VOL_MINUS_95_250DB (0xFE) +#define ONA10IV_VOL_AMP_MUTE (0xFF) +#define ONA10IV_VOL_AMP_MASK (0xFF) + +#define ONA10IV_REG_VOL_CTRL (0x0A) +#define ONA10IV_VOL_RAMP (0x01<<5) +#define ONA10IV_VOL_ZEROCROSS (0x00<<5) +#define ONA10IV_VOL_RAMP_0_01MSSTEP (0x00<<3) +#define ONA10IV_VOL_RAMP_0_10MSSTEP (0x01<<3) +#define ONA10IV_VOL_RAMP_0_50MSSTEP (0x02<<3) +#define ONA10IV_VOL_RAMP_1_00MSSTEP (0x03<<3) +#define ONA10IV_AVOL_UP_DISABLE (0x00<<2) +#define ONA10IV_AVOL_UP_RAMP (0x01<<2) +#define ONA10IV_AVOL_DN_DISABLE (0x00<<1) +#define ONA10IV_AVOL_DN_RAMP (0x01<<1) +#define ONA10IV_MUTE_NORMAL (0x00<<0) +#define ONA10IV_MUTE_AMPLIFIER_OUT (0x01<<0) + +#define ONA10IV_REG_GAIN_CTRL1 (0x0B) +#define ONA10IV_PCM_AMP_GAIN_PLUS_16DB (0x00<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_12DB (0x08<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_9DB (0x0E<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_6DB (0x14<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_3DB (0x1A<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_MASK (0x1F<<0) +#define ONA10IV_PCM_AMP_GAIN_PLUS_MAX (0x20) + +#define ONA10IV_REG_GAIN_CTRL2 (0x0C) +#define ONA10IV_PDM_DAC_MAP_7_23DBV (0x00<<0) +#define ONA10IV_PDM_DAC_MAP_4_31DBV (0x01<<0) +#define ONA10IV_PDM_DAC_MAP_M0_13DBV (0x02<<0) +#define ONA10IV_PDM_DAC_MAP_M9_68DBV (0x03<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_16DB (0x00<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_12DB (0x08<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_9DB (0x0E<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_6DB (0x14<<0) +#define ONA10IV_PDM_AMP_GAIN_PLUS_3DB (0x1A<<0) + +#define ONA10IV_REG_EMI_CTRL (0x0D) +#define ONA10IV_AMP_ERC_1_00VPNS (0x00<<3) +#define ONA10IV_AMP_ERC_0_75VPNS (0x01<<3) +#define ONA10IV_AMP_ERC_0_50VPNS (0x02<<3) +#define ONA10IV_AMP_ERC_3_50VPNS (0x03<<3) +#define ONA10IV_SS_MOD_5_3PERCENT (0x00<<0) +#define ONA10IV_SS_MOD_7_0PERCENT (0x01<<0) +#define ONA10IV_SS_MOD_10_6PERCENT (0x02<<0) +#define ONA10IV_SS_MOD_21_2PERCENT (0x03<<0) +#define ONA10IV_SS_MOD_0_0PERCENT (0x04<<0) +#define ONA10IV_SS_MOD_3_0PERCENT (0x05<<0) +#define ONA10IV_SS_MOD_3_6PERCENT (0x06<<0) +#define ONA10IV_SS_MOD_4_2PERCENT (0x07<<0) + +#define ONA10IV_REG_AGC_BATT (0x0E) +#define ONA10IV_BATT_ATH_9_763V (0x00<<0) +#define ONA10IV_BATT_ATH_9_835V (0x01<<0) +#define ONA10IV_BATT_ATH_9_907V (0x02<<0) +#define ONA10IV_BATT_ATH_9_979V (0x03<<0) +#define ONA10IV_BATT_ATH_10_510V (0x04<<0) +#define ONA10IV_BATT_ATH_10_123V (0x05<<0) +#define ONA10IV_BATT_ATH_10_195V (0x06<<0) +#define ONA10IV_BATT_ATH_10_267V (0x07<<0) +#define ONA10IV_BATT_ATH_10_339V (0x08<<0) +#define ONA10IV_BATT_ATH_10_411V (0x09<<0) +#define ONA10IV_BATT_ATH_10_483V (0x0A<<0) +#define ONA10IV_BATT_ATH_10_555V (0x0B<<0) +#define ONA10IV_BATT_ATH_10_627V (0x0C<<0) +#define ONA10IV_BATT_ATH_10_699V (0x0D<<0) +#define ONA10IV_BATT_ATH_10_771V (0x0E<<0) +#define ONA10IV_BATT_ATH_10_843V (0x0F<<0) +#define ONA10IV_BATT_ATH_10_915V (0x10<<0) +#define ONA10IV_BATT_ATH_10_987V (0x11<<0) +#define ONA10IV_BATT_ATH_11_590V (0x12<<0) +#define ONA10IV_BATT_ATH_11_131V (0x13<<0) +#define ONA10IV_BATT_ATH_11_203V (0x14<<0) +#define ONA10IV_BATT_ATH_11_275V (0x15<<0) +#define ONA10IV_BATT_ATH_11_347V (0x16<<0) +#define ONA10IV_BATT_ATH_11_419V (0x17<<0) +#define ONA10IV_BATT_ATH_11_491V (0x18<<0) +#define ONA10IV_BATT_ATH_11_563V (0x19<<0) +#define ONA10IV_BATT_ATH_11_635V (0x1A<<0) +#define ONA10IV_BATT_ATH_11_707V (0x1B<<0) +#define ONA10IV_BATT_ATH_11_779V (0x1C<<0) +#define ONA10IV_BATT_ATH_11_851V (0x1D<<0) +#define ONA10IV_BATT_ATH_11_923V (0x1E<<0) +#define ONA10IV_BATT_ATH_11_995V (0x1F<<0) + +#define ONA10IV_REG_AGC_CTRL1 (0x0F) +#define ONA10IV_AGC_HOLD_10MS (0x00<<4) +#define ONA10IV_AGC_HOLD_45MS (0x01<<4) +#define ONA10IV_AGC_HOLD_80MS (0x02<<4) +#define ONA10IV_AGC_HOLD_115MS (0x03<<4) +#define ONA10IV_AGC_HOLD_150MS (0x04<<4) +#define ONA10IV_AGC_HOLD_185MS (0x05<<4) +#define ONA10IV_AGC_HOLD_220MS (0x06<<4) +#define ONA10IV_AGC_HOLD_255MS (0x07<<4) +#define ONA10IV_AGC_HOLD_290MS (0x08<<4) +#define ONA10IV_AGC_HOLD_325MS (0x09<<4) +#define ONA10IV_AGC_HOLD_360MS (0x0A<<4) +#define ONA10IV_AGC_HOLD_395MS (0x0B<<4) +#define ONA10IV_AGC_HOLD_430MS (0x0C<<4) +#define ONA10IV_AGC_HOLD_465MS (0x0D<<4) +#define ONA10IV_AGC_HOLD_500MS (0x0E<<4) +#define ONA10IV_AGC_HOLD_INFINIT (0x0F<<4) +#define ONA10IV_AGC_ATTACK_10USPDB (0x00<<0) +#define ONA10IV_AGC_ATTACK_45USPDB (0x01<<0) +#define ONA10IV_AGC_ATTACK_80USPDB (0x02<<0) +#define ONA10IV_AGC_ATTACK_115USPDB (0x03<<0) +#define ONA10IV_AGC_ATTACK_150USPDB (0x04<<0) +#define ONA10IV_AGC_ATTACK_185USPDB (0x05<<0) +#define ONA10IV_AGC_ATTACK_220USPDB (0x06<<0) +#define ONA10IV_AGC_ATTACK_255USPDB (0x07<<0) +#define ONA10IV_AGC_ATTACK_290USPDB (0x08<<0) +#define ONA10IV_AGC_ATTACK_325USPDB (0x09<<0) +#define ONA10IV_AGC_ATTACK_360USPDB (0x0A<<0) +#define ONA10IV_AGC_ATTACK_395USPDB (0x0B<<0) +#define ONA10IV_AGC_ATTACK_430USPDB (0x0C<<0) +#define ONA10IV_AGC_ATTACK_465USPDB (0x0D<<0) +#define ONA10IV_AGC_ATTACK_500USPDB (0x0E<<0) +#define ONA10IV_AGC_ATTACK_535USPDB (0x0F<<0) + +#define ONA10IV_REG_AGC_CTRL2 (0x10) +#define ONA10IV_AGC_MAX_ATT_MINUS_9DB (0x00<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_8DB (0x01<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_7DB (0x02<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_6DB (0x03<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_5DB (0x04<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_4DB (0x05<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_3DB (0x06<<4) +#define ONA10IV_AGC_MAX_ATT_MINUS_2DB (0x07<<4) +#define ONA10IV_AGC_RELEASE_5MSPDB (0x00<<0) +#define ONA10IV_AGC_RELEASE_75MSPDB (0x01<<0) +#define ONA10IV_AGC_RELEASE_145MSPDB (0x02<<0) +#define ONA10IV_AGC_RELEASE_215MSPDB (0x03<<0) +#define ONA10IV_AGC_RELEASE_285MSPDB (0x04<<0) +#define ONA10IV_AGC_RELEASE_355MSPDB (0x05<<0) +#define ONA10IV_AGC_RELEASE_425MSPDB (0x06<<0) +#define ONA10IV_AGC_RELEASE_495MSPDB (0x07<<0) +#define ONA10IV_AGC_RELEASE_565MSPDB (0x08<<0) +#define ONA10IV_AGC_RELEASE_635MSPDB (0x09<<0) +#define ONA10IV_AGC_RELEASE_705MSPDB (0x0A<<0) +#define ONA10IV_AGC_RELEASE_775MSPDB (0x0B<<0) +#define ONA10IV_AGC_RELEASE_845MSPDB (0x0C<<0) +#define ONA10IV_AGC_RELEASE_915MSPDB (0x0D<<0) +#define ONA10IV_AGC_RELEASE_985MSPDB (0x0E<<0) +#define ONA10IV_AGC_RELEASE_1055MSPDB (0x0F<<0) + +#define ONA10IV_REG_AGC_CTRL3 (0x11) +#define ONA10IV_AGC_TIMEOUT_DISABLE (0x00<<0) +#define ONA10IV_AGC_TIMEOUT_565MS (0x01<<0) +#define ONA10IV_AGC_TIMEOUT_775MS (0x02<<0) +#define ONA10IV_AGC_TIMEOUT_985MS (0x03<<0) + +#define ONA10IV_REG_MAGC_CTRL1 (0x12) +#define ONA10IV_RX_SLOT8_IGNORES (0x00<<7) +#define ONA10IV_RX_SLOT8_LISTENS (0x01<<7) +#define ONA10IV_RX_SLOT7_IGNORES (0x00<<6) +#define ONA10IV_RX_SLOT7_LISTENS (0x01<<6) +#define ONA10IV_RX_SLOT6_IGNORES (0x00<<5) +#define ONA10IV_RX_SLOT6_LISTENS (0x01<<5) +#define ONA10IV_RX_SLOT5_IGNORES (0x00<<4) +#define ONA10IV_RX_SLOT5_LISTENS (0x01<<4) +#define ONA10IV_RX_SLOT4_IGNORES (0x00<<3) +#define ONA10IV_RX_SLOT4_LISTENS (0x01<<3) +#define ONA10IV_RX_SLOT3_IGNORES (0x00<<2) +#define ONA10IV_RX_SLOT3_LISTENS (0x01<<2) +#define ONA10IV_RX_SLOT2_IGNORES (0x00<<1) +#define ONA10IV_RX_SLOT2_LISTENS (0x01<<1) +#define ONA10IV_RX_SLOT1_IGNORES (0x00<<0) +#define ONA10IV_RX_SLOT1_LISTENS (0x01<<0) + +#define ONA10IV_REG_MAGC_CTRL2 (0x13) +#define ONA10IV_TX_SLOT8_HIZ (0x00<<7) +#define ONA10IV_TX_SLOT8_TRANSMIT (0x01<<7) +#define ONA10IV_TX_SLOT7_HIZ (0x00<<6) +#define ONA10IV_TX_SLOT7_TRANSMIT (0x01<<6) +#define ONA10IV_TX_SLOT6_HIZ (0x00<<5) +#define ONA10IV_TX_SLOT6_TRANSMIT (0x01<<5) +#define ONA10IV_TX_SLOT5_HIZ (0x00<<4) +#define ONA10IV_TX_SLOT5_TRANSMIT (0x01<<4) +#define ONA10IV_TX_SLOT4_HIZ (0x00<<3) +#define ONA10IV_TX_SLOT4_TRANSMIT (0x01<<3) +#define ONA10IV_TX_SLOT3_HIZ (0x00<<2) +#define ONA10IV_TX_SLOT3_TRANSMIT (0x01<<2) +#define ONA10IV_TX_SLOT2_HIZ (0x00<<1) +#define ONA10IV_TX_SLOT2_TRANSMIT (0x01<<1) +#define ONA10IV_TX_SLOT1_HIZ (0x00<<0) +#define ONA10IV_TX_SLOT1_TRANSMIT (0x01<<0) + +#define ONA10IV_REG_MAGC_CTRL3 (0x14) +#define ONA10IV_MAGC_ODRV_100P (0x00<<1) +#define ONA10IV_MAGC_ODRV_75P (0x01<<1) +#define ONA10IV_MAGC_ODRV_50P (0x02<<1) +#define ONA10IV_MAGC_ODRV_25P (0x03<<1) +#define ONA10IV_MAGC_EN_DISABLE (0x00<<0) +#define ONA10IV_MAGC_EN_ENABLE (0x01<<0) + +#define ONA10IV_REG_SENSE_CTRL (0x15) +#define ONA10IV_T_HOLD_10MS (0x00<<6) +#define ONA10IV_T_HOLD_45MS (0x01<<6) +#define ONA10IV_T_HOLD_80MS (0x02<<6) +#define ONA10IV_T_HOLD_115MS (0x03<<6) +#define ONA10IV_T_ATTACK_2MS_STEP (0x00<<4) +#define ONA10IV_T_ATTACK_4MS_STEP (0x01<<4) +#define ONA10IV_T_ATTACK_6MS_STEP (0x02<<4) +#define ONA10IV_T_ATTACK_8MS_STEP (0x03<<4) +#define ONA10IV_T_ATH_140CELSIUS (0x00<<2) +#define ONA10IV_T_ATH_130CELSIUS (0x01<<2) +#define ONA10IV_T_ATH_120CELSIUS (0x02<<2) +#define ONA10IV_T_ATH_110CELSIUS (0x03<<2) +#define ONA10IV_T_SAMP_40MS (0x00<<0) +#define ONA10IV_T_SAMP_20MS (0x01<<0) +#define ONA10IV_T_SAMP_10MS (0x02<<0) +#define ONA10IV_T_SAMP_CONSTANTLY (0x03<<0) + +#define ONA10IV_REG_T_SENSE_OUT1 (0x16) +#define ONA10IV_REG_T_SENSE_OUT2 (0x17) +#define ONA10IV_T_SENSE_OUT2_BIT (0x07) +#define ONA10IV_T_SENSE_OUT2_SIGN_BIT (0x04) + +#define ONA10IV_REG_DATAO_CTRL1 (0x18) +#define ONA10IV_HPF_BP_NORMAL (0x00<<4) +#define ONA10IV_HPF_BP_HIGHPASS (0x01<<4) +#define ONA10IV_PDM_OUT_FALLING (0x00<<2) +#define ONA10IV_PDM_OUT_RISING (0x01<<2) +#define ONA10IV_DATA_ODRV_100P (0x00<<0) +#define ONA10IV_DATA_ODRV_75P (0x01<<0) +#define ONA10IV_DATA_ODRV_50P (0x02<<0) +#define ONA10IV_DATA_ODRV_25P (0x03<<0) + +#define ONA10IV_REG_DATAO_CTRL2 (0x19) +#define ONA10IV_I_DATAO_TX_DISABLE (0x00<<7) +#define ONA10IV_I_DATAO_TX_ENABLE (0x01<<7) +#define ONA10IV_V_DATAO_TX_DISABLE (0x00<<6) +#define ONA10IV_V_DATAO_TX_ENABLE (0x01<<6) +#define ONA10IV_I_SLOT_1 (0x00<<3) +#define ONA10IV_I_SLOT_2 (0x01<<3) +#define ONA10IV_I_SLOT_3 (0x02<<3) +#define ONA10IV_I_SLOT_4 (0x03<<3) +#define ONA10IV_I_SLOT_5 (0x04<<3) +#define ONA10IV_I_SLOT_6 (0x05<<3) +#define ONA10IV_I_SLOT_7 (0x06<<3) +#define ONA10IV_I_SLOT_8 (0x07<<3) +#define ONA10IV_V_SLOT_1 (0x00<<0) +#define ONA10IV_V_SLOT_2 (0x01<<0) +#define ONA10IV_V_SLOT_3 (0x02<<0) +#define ONA10IV_V_SLOT_4 (0x03<<0) +#define ONA10IV_V_SLOT_5 (0x04<<0) +#define ONA10IV_V_SLOT_6 (0x05<<0) +#define ONA10IV_V_SLOT_7 (0x06<<0) +#define ONA10IV_V_SLOT_8 (0x07<<0) + +#define ONA10IV_REG_DATAO_CTRL3 (0x1A) +#define ONA10IV_T_DATAO_TX_DISABLE (0x00<<3) +#define ONA10IV_T_DATAO_TX_ENABLE (0x01<<3) +#define ONA10IV_T_SLOT_1 (0x00<<0) +#define ONA10IV_T_SLOT_2 (0x01<<0) +#define ONA10IV_T_SLOT_3 (0x02<<0) +#define ONA10IV_T_SLOT_4 (0x03<<0) +#define ONA10IV_T_SLOT_5 (0x04<<0) +#define ONA10IV_T_SLOT_6 (0x05<<0) +#define ONA10IV_T_SLOT_7 (0x06<<0) +#define ONA10IV_T_SLOT_8 (0x07<<0) +#endif // _ONA10IV__

2 1

ctl: enhancement for error reporting
by GitHub pull_request - opened 13 Nov '20

13 Nov '20

alsa-project/alsa-gobject pull request #48 was opened from takaswie: Current implementation uses library-wide error domain to report error, however this is partly against rule of GError usage. Furthermore, the error delivers information just about errno and hard to know the cause of error. This patchset enhances error reporting. The library-wide error domain is obsoleted. A new error domain is added just for ALSACtl.Card. The error domain delivers information about the cause of error. Request URL : https://github.com/alsa-project/alsa-gobject/pull/48 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/48.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH] ALSA: ctl: fix error path at adding user-defined element set
by Takashi Sakamoto 13 Nov '20

13 Nov '20

When processing request to add/replace user-defined element set, check of given element identifier and decision of numeric identifier is done in "__snd_ctl_add_replace()" helper function. When the result of check is wrong, the helper function returns error code. The error code shall be returned to userspace application. Current implementation includes bug to return zero to userspace application regardless of the result. This commit fixes the bug. Cc: <stable(a)vger.kernel.org> Fixes: e1a7bfe38079 ("ALSA: control: Fix race between adding and removing a user element") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 421ddc7..f341fc4 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1539,7 +1539,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, unlock: up_write(&card->controls_rwsem); - return 0; + return err; } static int snd_ctl_elem_add_user(struct snd_ctl_file *file, -- 2.25.1

2 1

HP headset Mic detect issue
by Kailang 13 Nov '20

13 Nov '20

Hi Takashi, If it turn on power save mode, Headset Mic will lose after boot. Attach patch will solve it. Many Thanks. BR, Kailang

2 1

HP mute led support
by Kailang 13 Nov '20

13 Nov '20

Hi Takashi, HP machine have mute led support. Attach patch will enable it. Many Thanks. BR, Kailang

2 1

[PATCH 00/12] ASoC: soc-component: add snd_soc_component_xxx()
by Kuninori Morimoto 13 Nov '20

13 Nov '20

Hi Mark I think component related function should be implemented at soc-component.c, otherwise it is confusable to read. These are for it. Kuninori Morimoto (12): ASoC: soc-component: add snd_soc_component_compr_open() ASoC: soc-component: add snd_soc_component_compr_free() ASoC: soc-component: add snd_soc_component_compr_trigger() ASoC: soc-component: add snd_soc_component_compr_set_params() ASoC: soc-component: add snd_soc_component_compr_get_params() ASoC: soc-component: add snd_soc_component_compr_get_caps() ASoC: soc-component: add snd_soc_component_compr_get_codec_caps() ASoC: soc-component: add snd_soc_component_compr_ack() ASoC: soc-component: add snd_soc_component_compr_pointer() ASoC: soc-component: add snd_soc_component_compr_copy() ASoC: soc-component: add snd_soc_component_compr_set_metadata() ASoC: soc-component: add snd_soc_component_compr_get_metadata() include/sound/soc-component.h | 22 +++ sound/soc/soc-component.c | 244 ++++++++++++++++++++++++++++++++++ sound/soc/soc-compress.c | 223 ++++--------------------------- 3 files changed, 293 insertions(+), 196 deletions(-) -- 2.25.1

2 15

[PATCH 0/4] ASoC: SOF: Kconfig corrections
by Pierre-Louis Bossart 12 Nov '20

12 Nov '20

Three cosmetic patches and a fix for a randconfig issue. Pierre-Louis Bossart (1): ASoC: SOF: Intel: fix Kconfig dependency for SND_INTEL_DSP_CONFIG Randy Dunlap (3): ASoC: SOF: imx: fix Kconfig punctuation ASoC: SOF: Kconfig: fix Kconfig punctuation and wording ASoC: SOF: Intel: fix Kconfig punctuation and wording sound/soc/sof/Kconfig | 42 ++++++++++++------------ sound/soc/sof/imx/Kconfig | 10 +++--- sound/soc/sof/intel/Kconfig | 64 ++++++++++++++++++------------------- 3 files changed, 58 insertions(+), 58 deletions(-) -- 2.25.1

2 7

[PATCH v2 0/4] ASoC: SOF: Kconfig corrections
by Pierre-Louis Bossart 12 Nov '20

12 Nov '20

Three cosmetic patches and a fix for a randconfig issue. v2: rebase to remove git am issue (worked with --3way before) Pierre-Louis Bossart (1): ASoC: SOF: Intel: fix Kconfig dependency for SND_INTEL_DSP_CONFIG Randy Dunlap (3): ASoC: SOF: imx: fix Kconfig punctuation ASoC: SOF: Kconfig: fix Kconfig punctuation and wording ASoC: SOF: Intel: fix Kconfig punctuation and wording sound/soc/sof/Kconfig | 42 ++++++++++++------------ sound/soc/sof/imx/Kconfig | 10 +++--- sound/soc/sof/intel/Kconfig | 64 ++++++++++++++++++------------------- 3 files changed, 58 insertions(+), 58 deletions(-) -- 2.25.1

2 7

[PATCH 1/6] ASoC: wm1133-ev1: Remove unused driver
by Fabio Estevam 12 Nov '20

12 Nov '20

Since commit c93197b0041d ("ARM: imx: Remove i.MX31 board files"), the MACH_MX31ADS_WM1133_EV1 non-DT platform is no longer supported, so get rid of its machine audio driver too. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/Kconfig | 11 -- sound/soc/fsl/Makefile | 2 - sound/soc/fsl/wm1133-ev1.c | 289 ------------------------------------- 3 files changed, 302 deletions(-) delete mode 100644 sound/soc/fsl/wm1133-ev1.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 52a562215008..3ad547193df8 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -239,17 +239,6 @@ config SND_SOC_IMX_SSI comment "SoC Audio support for Freescale i.MX boards:" -config SND_MXC_SOC_WM1133_EV1 - tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" - depends on MACH_MX31ADS_WM1133_EV1 - select SND_SOC_WM8350 - select SND_SOC_IMX_PCM_FIQ - select SND_SOC_IMX_AUDMUX - select SND_SOC_IMX_SSI - help - Enable support for audio on the i.MX31ADS with the WM1133-EV1 - PMIC board with WM8835x fitted. - config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 2181b7f9f677..8d3e40415050 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -65,7 +65,6 @@ obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o -snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-spdif-objs := imx-spdif.o @@ -75,7 +74,6 @@ snd-soc-imx-audmix-objs := imx-audmix.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o -obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c deleted file mode 100644 index 99611a037ada..000000000000 --- a/sound/soc/fsl/wm1133-ev1.c +++ /dev/null @@ -1,289 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0+ -// -// wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS -// -// Copyright (c) 2010 Wolfson Microelectronics plc -// Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> -// -// Based on an earlier driver for the same hardware by Liam Girdwood. - -#include <linux/platform_device.h> -#include <linux/clk.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/jack.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "imx-ssi.h" -#include "../codecs/wm8350.h" -#include "imx-audmux.h" - -/* There is a silicon mic on the board optionally connected via a solder pad - * SP1. Define this to enable it. - */ -#undef USE_SIMIC - -struct _wm8350_audio { - unsigned int channels; - snd_pcm_format_t format; - unsigned int rate; - unsigned int sysclk; - unsigned int bclkdiv; - unsigned int clkdiv; - unsigned int lr_rate; -}; - -/* in order of power consumption per rate (lowest first) */ -static const struct _wm8350_audio wm8350_audio[] = { - /* 16bit mono modes */ - {1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1, - WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,}, - - /* 16 bit stereo modes */ - {2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000, - WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000, - WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000, - WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600, - WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600, - WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - {2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200, - WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,}, - - /* 24bit stereo modes */ - {2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, - {2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, - {2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, - {2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200, - WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,}, -}; - -static int wm1133_ev1_hw_params(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 *codec_dai = asoc_rtd_to_codec(rtd, 0); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int i, found = 0; - snd_pcm_format_t format = params_format(params); - unsigned int rate = params_rate(params); - unsigned int channels = params_channels(params); - - /* find the correct audio parameters */ - for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) { - if (rate == wm8350_audio[i].rate && - format == wm8350_audio[i].format && - channels == wm8350_audio[i].channels) { - found = 1; - break; - } - } - if (!found) - return -EINVAL; - - /* codec FLL input is 14.75 MHz from MCLK */ - snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk); - - /* TODO: The SSI driver should figure this out for us */ - switch (channels) { - case 2: - snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0); - break; - case 1: - snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 1, 0); - break; - default: - return -EINVAL; - } - - /* set MCLK as the codec system clock for DAC and ADC */ - snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK, - wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN); - - /* set codec BCLK division for sample rate */ - snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV, - wm8350_audio[i].bclkdiv); - - /* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */ - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate); - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate); - - /* now configure DAC and ADC clocks */ - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv); - - snd_soc_dai_set_clkdiv(codec_dai, - WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv); - - return 0; -} - -static const struct snd_soc_ops wm1133_ev1_ops = { - .hw_params = wm1133_ev1_hw_params, -}; - -static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = { -#ifdef USE_SIMIC - SND_SOC_DAPM_MIC("SiMIC", NULL), -#endif - SND_SOC_DAPM_MIC("Mic1 Jack", NULL), - SND_SOC_DAPM_MIC("Mic2 Jack", NULL), - SND_SOC_DAPM_LINE("Line In Jack", NULL), - SND_SOC_DAPM_LINE("Line Out Jack", NULL), - SND_SOC_DAPM_HP("Headphone Jack", NULL), -}; - -/* imx32ads soc_card audio map */ -static const struct snd_soc_dapm_route wm1133_ev1_map[] = { - -#ifdef USE_SIMIC - /* SiMIC --> IN1LN (with automatic bias) via SP1 */ - { "IN1LN", NULL, "Mic Bias" }, - { "Mic Bias", NULL, "SiMIC" }, -#endif - - /* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */ - { "IN1LN", NULL, "Mic Bias" }, - { "IN1LP", NULL, "Mic1 Jack" }, - { "Mic Bias", NULL, "Mic1 Jack" }, - - /* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */ - { "IN1RN", NULL, "Mic Bias" }, - { "IN1RP", NULL, "Mic2 Jack" }, - { "Mic Bias", NULL, "Mic2 Jack" }, - - /* Line in Jack --> AUX (L+R) */ - { "IN3R", NULL, "Line In Jack" }, - { "IN3L", NULL, "Line In Jack" }, - - /* Out1 --> Headphone Jack */ - { "Headphone Jack", NULL, "OUT1R" }, - { "Headphone Jack", NULL, "OUT1L" }, - - /* Out1 --> Line Out Jack */ - { "Line Out Jack", NULL, "OUT2R" }, - { "Line Out Jack", NULL, "OUT2L" }, -}; - -static struct snd_soc_jack hp_jack; - -static struct snd_soc_jack_pin hp_jack_pins[] = { - { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE }, -}; - -static struct snd_soc_jack mic_jack; - -static struct snd_soc_jack_pin mic_jack_pins[] = { - { .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE }, - { .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE }, -}; - -static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; - - /* Headphone jack detection */ - snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE, - &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); - wm8350_hp_jack_detect(component, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE); - - /* Microphone jack detection */ - snd_soc_card_jack_new(rtd->card, "Microphone", - SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack, - mic_jack_pins, ARRAY_SIZE(mic_jack_pins)); - wm8350_mic_jack_detect(component, &mic_jack, SND_JACK_MICROPHONE, - SND_JACK_BTN_0); - - snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias"); - - return 0; -} - - -SND_SOC_DAILINK_DEFS(ev1, - DAILINK_COMP_ARRAY(COMP_CPU("imx-ssi.0")), - DAILINK_COMP_ARRAY(COMP_CODEC("wm8350-codec.0-0x1a", "wm8350-hifi")), - DAILINK_COMP_ARRAY(COMP_PLATFORM("imx-ssi.0"))); - -static struct snd_soc_dai_link wm1133_ev1_dai = { - .name = "WM1133-EV1", - .stream_name = "Audio", - .init = wm1133_ev1_init, - .ops = &wm1133_ev1_ops, - .symmetric_rates = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - SND_SOC_DAILINK_REG(ev1), -}; - -static struct snd_soc_card wm1133_ev1 = { - .name = "WM1133-EV1", - .owner = THIS_MODULE, - .dai_link = &wm1133_ev1_dai, - .num_links = 1, - - .dapm_widgets = wm1133_ev1_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets), - .dapm_routes = wm1133_ev1_map, - .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map), -}; - -static struct platform_device *wm1133_ev1_snd_device; - -static int __init wm1133_ev1_audio_init(void) -{ - int ret; - unsigned int ptcr, pdcr; - - /* SSI0 mastered by port 5 */ - ptcr = IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) | - IMX_AUDMUX_V2_PTCR_TCLKDIR | - IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); - pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5); - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr); - - ptcr = IMX_AUDMUX_V2_PTCR_SYN; - pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0); - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr); - - wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1); - if (!wm1133_ev1_snd_device) - return -ENOMEM; - - platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1); - ret = platform_device_add(wm1133_ev1_snd_device); - - if (ret) - platform_device_put(wm1133_ev1_snd_device); - - return ret; -} -module_init(wm1133_ev1_audio_init); - -static void __exit wm1133_ev1_audio_exit(void) -{ - platform_device_unregister(wm1133_ev1_snd_device); -} -module_exit(wm1133_ev1_audio_exit); - -MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); -MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS"); -MODULE_LICENSE("GPL"); -- 2.17.1

2 6

[PATCH] ASoC: Intel: Boards: tgl_max98373: add dpcm_capture flag for speaker_smart_amp
by Ranjani Sridharan 12 Nov '20

12 Nov '20

From: Dharageswari R <dharageswari.r(a)intel.com> Smart_amp_speaker device has the playback stream and capture stream associated to it. Hence add the dpcm_capture = 1 flag while dailink creation. This patches fixes: ERR kernel [timestamp] SSP1-Codec: ASoC: no backend capture stream Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Dharageswari R <dharageswari.r(a)intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index e7d9a82ca70d..891f908659f5 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -734,6 +734,8 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].num_codecs = ARRAY_SIZE(max_98373_components); links[id].init = max98373_spk_codec_init; links[id].ops = &max_98373_ops; + /* feedback stream */ + links[id].dpcm_capture = 1; } else if (sof_rt5682_quirk & SOF_MAX98360A_SPEAKER_AMP_PRESENT) { links[id].codecs = max98360a_component; -- 2.25.1

2 1

[PATCH 0/4] ASoC: soundwire: SDCA device quirk and cleanups
by Pierre-Louis Bossart 12 Nov '20

12 Nov '20

Add one quirk to handle new TigerLake-SDCA device and cleanup code. Pierre-Louis Bossart (4): ASoC: Intel: sof_sdw: add quirk for new TigerLake-SDCA device ASoC: codecs: rt*.c: remove useless pointer cast ASoC: rt711: remove warnings ASoC: codecs: max98373-sdw: align regmap use with other codecs sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/rt1308-sdw.c | 2 +- sound/soc/codecs/rt5682-sdw.c | 2 +- sound/soc/codecs/rt700.c | 2 +- sound/soc/codecs/rt711-sdw.c | 15 ++++++++------- sound/soc/codecs/rt711.c | 2 +- sound/soc/codecs/rt715.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 8 files changed, 24 insertions(+), 13 deletions(-) -- 2.25.1

2 5

[PATCH v2 0/2] Support SC7180 sound card variation without headset
by xuyuqing 12 Nov '20

12 Nov '20

Use compatible string to let machine driver handle the case where ALC5682 headset codec is replaced with ADAU7002 dmic codec xuyuqing (2): ASoC: google: dt-bindings: add new compatible for sc7180-coachz ASoC: qcom: sc7180: Add machine driver for sound card .../bindings/sound/google,sc7180-trogdor.yaml | 4 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7180.c | 86 ++++++++++++++++--- 3 files changed, 77 insertions(+), 14 deletions(-) -- 2.25.1

3 4

[PATCH] ASoC: SOF: loader: do not warn about unknown firmware headers
by Kai Vehmanen 12 Nov '20

12 Nov '20

The firmware extended data IPC and manifest structures are designed to be extendable without breaking the driver-firmware ABI. Driver should not raise a warning in case a new header type is detected at firmware boot. There are already checks for IPC ABI compatibility in snd_sof_ipc_valid() and if the versions are deemed compatible, extra fields in IPC messages should not trigger warnings. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Karol Trzciński <karolx.trzcinski(a)linux.intel.com> --- sound/soc/sof/loader.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index ba9ed66f98bc..ce68d708fc6f 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -124,7 +124,7 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) /* They are supported but we don't do anything here */ break; default: - dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", + dev_info(sdev->dev, "unknown ext header type %d size 0x%x\n", ext_hdr->type, ext_hdr->hdr.size); ret = 0; break; @@ -280,7 +280,7 @@ static int snd_sof_fw_ext_man_parse(struct snd_sof_dev *sdev, ret = ext_man_get_dbg_abi_info(sdev, elem_hdr); break; default: - dev_warn(sdev->dev, "warning: unknown sof_ext_man header type %d size 0x%X\n", + dev_info(sdev->dev, "unknown sof_ext_man header type %d size 0x%X\n", elem_hdr->type, elem_hdr->size); break; } -- 2.28.0

2 1

[PATCH v2 0/4] ASoC: Fix error handling in wm899x
by Zhang Qilong 12 Nov '20

12 Nov '20

The first patch fixed a wrong free in wm8997_probe. The remaining three patches fixed PM disable depth imbalance on error handling. Zhang Qilong (4): ASoC: arizona: Fix a wrong free in wm8997_probe ASoC: arizona: Fix PM disable depth imbalance on error ASoC: wm8994: Fix PM disable depth imbalance on error ASoC: wm8998: Fix PM disable depth imbalance on error sound/soc/codecs/wm8994.c | 6 +++++- sound/soc/codecs/wm8997.c | 8 ++++++-- sound/soc/codecs/wm8998.c | 4 +++- 3 files changed, 14 insertions(+), 4 deletions(-) -- 2.25.4

3 6

[PATCH 0/2] Refactor Audio Graph schema
by Sameer Pujar 12 Nov '20

12 Nov '20

There can be custom sound cards reusing most of the audio-graph implementation. To allow this refactor the audio-graph-card schema into following files. * audio-graph.yaml : defines all common bindings * audio-graph-card.yaml : define compatible property Custom sound cards can just reference 'audio-graph.yaml' and define its own compatible and specific properties. This depends on following series: http://patchwork.ozlabs.org/project/devicetree-bindings/patch/2020110220365… Sameer Pujar (2): ASoC: audio-graph-card: Refactor schema ASoC: renesas,rsnd: Update audio graph references .../bindings/sound/audio-graph-card.yaml | 102 +------------------- .../devicetree/bindings/sound/audio-graph.yaml | 107 +++++++++++++++++++++ .../devicetree/bindings/sound/renesas,rsnd.yaml | 6 +- 3 files changed, 112 insertions(+), 103 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-graph.yaml -- 2.7.4

3 6

[PATCH v2 0/2] ASoC: Fix error handling in arizona
by Zhang Qilong 12 Nov '20

12 Nov '20

The first patch fixed a wrong free in wm8997_probe. The other fixed PM disable depth imbalance on error handling. Change Log v2: - add collected review and ack. Zhang Qilong (2): ASoC: arizona: Fix a wrong free in wm8997_probe ASoC: arizona: Fix PM disable depth imbalance on error sound/soc/codecs/wm8997.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) -- 2.25.4

2 3

[PATCH] ASoC: pcm512x: Add support for data formats RJ and LJ
by Kirill Marinushkin 12 Nov '20

12 Nov '20

Currently, pcm512x driver supports only I2S data format. This commit adds RJ and LJ as well. I don't expect regression WRT existing sound cards, because: * default value in corresponding register of pcm512x codec is 0 == I2S * existing in-tree sound cards with pcm512x codec are configured for I2S * i don't see how existing off-tree sound cards with pcm512x codec could be configured differently - it would not work * tested explicitly, that there is no regression with Raspberry Pi + sound card `sound/soc/bcm/hifiberry_dacplus.c` Signed-off-by: Kirill Marinushkin <kmarinushkin(a)birdec.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Matthias Reichl <hias(a)horus.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/codecs/pcm512x.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 8153d3d01654..c6e975fb4a43 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1167,6 +1167,7 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); int alen; + int afmt; int gpio; int clock_output; int master_mode; @@ -1195,6 +1196,22 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + switch (pcm512x->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + afmt = PCM512x_AFMT_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + afmt = PCM512x_AFMT_RTJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + afmt = PCM512x_AFMT_LTJ; + break; + default: + dev_err(component->dev, "unsupported DAI format: 0x%x\n", + pcm512x->fmt); + return -EINVAL; + } + switch (pcm512x->fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: ret = regmap_update_bits(pcm512x->regmap, @@ -1236,6 +1253,13 @@ static int pcm512x_hw_params(struct snd_pcm_substream *substream, return ret; } + ret = regmap_update_bits(pcm512x->regmap, PCM512x_I2S_1, + PCM512x_AFMT, afmt); + if (ret != 0) { + dev_err(component->dev, "Failed to set data format: %d\n", ret); + return ret; + } + if (pcm512x->pll_out) { ret = regmap_write(pcm512x->regmap, PCM512x_FLEX_A, 0x11); if (ret != 0) { -- 2.13.6

2 6

Alsa Utils + Lib 1.2.2-1ubuntu2, 1.2.2-2.1ubuntu2.1 breaks sound on HP Envy x360 (Mint 20)
by GitHub issues - edited 11 Nov '20

11 Nov '20

alsa-project/alsa-utils issue #71 was edited from kingkode1337: After update, sound no longer works and does not pick up my sound card. ``` Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) HD Audio Controller Subsystem: Hewlett-Packard Company Family 17h (Models 10h-1fh) HD Audio Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel ``` It now shows as a Dummy Device, under the upgrade so I have abstained from doing so. Issue URL : https://github.com/alsa-project/alsa-utils/issues/71 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Alsa Utils + Lib 1.2.2-1ubuntu2, 1.2.2-2.1ubuntu2.1 breaks sound on HP Envy x360 (Mint 20)
by GitHub issues - opened 11 Nov '20

11 Nov '20

alsa-project/alsa-utils issue #71 was opened from kingkode1337: After update, sound no longer works and does not pick up my sound card. ``` Advanced Micro Devices, Inc. [AMD] Family 17h (Models 10h-1fh) HD Audio Controller Subsystem: Hewlett-Packard Company Family 17h (Models 10h-1fh) HD Audio Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel ``` It now shows as a Dummy Device, under the upgrade so I have abstained from doing so. Issue URL : https://github.com/alsa-project/alsa-utils/issues/71 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: tegra: Don't warn on probe deferral
by Jon Hunter 11 Nov '20

11 Nov '20

Deferred probe is an expected return value for snd_soc_register_card(). Given that the driver deals with it properly, there's no need to output a warning that may potentially confuse users. Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/tegra/tegra_alc5632.c | 4 ++-- sound/soc/tegra/tegra_max98090.c | 8 +++----- sound/soc/tegra/tegra_rt5640.c | 8 +++----- sound/soc/tegra/tegra_rt5677.c | 4 ++-- sound/soc/tegra/tegra_sgtl5000.c | 4 ++-- sound/soc/tegra/tegra_wm8753.c | 8 +++----- sound/soc/tegra/tegra_wm8903.c | 8 +++----- sound/soc/tegra/tegra_wm9712.c | 4 ++-- sound/soc/tegra/trimslice.c | 8 +++----- 9 files changed, 23 insertions(+), 33 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 8661877bf4c6..0a0efd24e4b0 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -203,8 +203,8 @@ static int tegra_alc5632_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto err_put_cpu_of_node; } diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 9d8e16473ab9..00c19704057b 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -247,11 +247,9 @@ static int tegra_max98090_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_rt5640.c b/sound/soc/tegra/tegra_rt5640.c index c73bd23b3d67..9afba37a3b08 100644 --- a/sound/soc/tegra/tegra_rt5640.c +++ b/sound/soc/tegra/tegra_rt5640.c @@ -193,11 +193,9 @@ static int tegra_rt5640_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_rt5677.c b/sound/soc/tegra/tegra_rt5677.c index 7504507dd8b8..d30f8b6deda4 100644 --- a/sound/soc/tegra/tegra_rt5677.c +++ b/sound/soc/tegra/tegra_rt5677.c @@ -268,8 +268,8 @@ static int tegra_rt5677_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto err_put_cpu_of_node; } diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index e1dc8e7d337a..885332170c77 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -154,8 +154,8 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto err_put_cpu_of_node; } diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index fa41fa366daf..efd793886689 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -156,11 +156,9 @@ static int tegra_wm8753_driver_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index ef6652aaac9b..e4863fa37b0c 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -352,11 +352,9 @@ static int tegra_wm8903_driver_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "devm_snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 726edfa21a29..4f09a178049d 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -117,8 +117,8 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); + dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); goto codec_unregister; } diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index baae4cce7fc6..6c1cc3d0ac33 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -144,11 +144,9 @@ static int tegra_snd_trimslice_probe(struct platform_device *pdev) return ret; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", - ret); - return ret; - } + if (ret) + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card failed\n"); return 0; } -- 2.25.1

2 1

[PATCH v2] ASoC: mediatek: mt8192: skip first time data at the beginning of DMIC recording
by Jiaxin Yu 11 Nov '20

11 Nov '20

We can choose to drop away any length of data from the beginning according to project needs. Some projects don't want to throw away any data, because they want to use recorded data to do echo cancellation, so they have to make sure that they are aligned with the reference data as much as possible. Or there are other algorithms in the upper layer to eliminate this noise. Or some projects want to eliminate this noise form the kernel layer. However, the minimum recommended value is 50ms to skip pop noise. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- This seems like something that would apply equally to all DMICs so should be done at a more general level, so we implement it in the DMIC driver. sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 +++- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index dc628fb6c1025..31f1431bf5792 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -175,6 +175,7 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 select SND_SOC_MT6359 select SND_SOC_RT1015 select SND_SOC_RT5682_I2C + select SND_SOC_DMIC help This adds ASoC driver for Mediatek MT8192 boards with the MT6359 RT1015 RT5682 audio codec. diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 41d8a8567bb38..b7f42a530d069 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -540,7 +540,9 @@ SND_SOC_DAILINK_DEFS(playback_hdmi, SND_SOC_DAILINK_DEFS(primary_codec, DAILINK_COMP_ARRAY(COMP_CPU("ADDA")), DAILINK_COMP_ARRAY(COMP_CODEC("mt6359-sound", - "mt6359-snd-codec-aif1")), + "mt6359-snd-codec-aif1"), + COMP_CODEC("dmic-codec", + "dmic-hifi")), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(primary_codec_ch34, -- 2.18.0

2 1

[PATCH 0/3] Fix PM disable depth imbalance on error
by Zhang Qilong 11 Nov '20

11 Nov '20

The pm_runtime_enable will increase power disable depth. Thus a pairing decrement is needed on the error handling path to keep it balanced according to context. This series of patches fixed it. Zhang Qilong (3): ASoC: wm8994: Fix PM disable depth imbalance on error ASoC: wm8997: Fix PM disable depth imbalance on error ASoC: wm8998: Fix PM disable depth imbalance on error sound/soc/codecs/wm8994.c | 6 +++++- sound/soc/codecs/wm8997.c | 9 +++++++-- sound/soc/codecs/wm8998.c | 4 +++- 3 files changed, 15 insertions(+), 4 deletions(-) -- 2.25.4

4 8

Re: 回覆: CLEVO uses the model name and SSID list of Realtek audio CODEC ALC293 (ALC1220)
by Takashi Iwai 11 Nov '20

11 Nov '20

On Wed, 11 Nov 2020 02:28:50 +0100, Pshou wrote: > > Hi Alsa-devel ML: > > Can you help me add some source code to patch_realtek.c? > > Add some CLEVO SSID in the alc293 (alc1220). > > Best regards > Pshou Applied now with minor coding style fixes. Thanks. Takashi > > -----原始郵件----- > 寄件者: Takashi Iwai <tiwai(a)suse.de> > 寄件日期: 2020年11月10日 15:02 > 收件者: Pshou <pshou(a)realtek.com> > 副本: Kailang <kailang(a)realtek.com> > 主旨: Re: CLEVO uses the model name and SSID list of Realtek audio CODEC ALC293 (ALC1220) > > On Tue, 10 Nov 2020 03:03:38 +0100, > Pshou wrote: > > > > Hi Takashi: > > > > Can you help me add some source code to patch_realtek.c? > > > > Add some CLEVO SSID in the alc293 (alc1220). > > > > Best regards > > > > pshou > > Could you just submit to alsa-devel ML and Cc to me? > A patch has to be submitted to the ML for public review. > > > thanks, > > Takashi > > ------Please consider the environment before printing this e-mail. > [2 0001-Add-alc293-and-alc1220-in-clove.patch <application/octet-stream (base64)>] >

1 0

[PATCH] ASoC: fsl_xcvr: fix break condition
by Viorel Suman (OSS) 10 Nov '20

10 Nov '20

From: Viorel Suman <viorel.suman(a)nxp.com> The break condition copied by mistake as same as loop condition in the previous version, but must be the opposite. So fix it. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index c055179e6d11..2a28810d0e29 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -247,7 +247,7 @@ static int fsl_xcvr_ai_write(struct fsl_xcvr *xcvr, u8 reg, u32 data, bool phy) regmap_write(xcvr->regmap, FSL_XCVR_PHY_AI_CTRL_TOG, idx); ret = regmap_read_poll_timeout(xcvr->regmap, FSL_XCVR_PHY_AI_CTRL, val, - (val & idx) != ((val & tidx) >> 1), + (val & idx) == ((val & tidx) >> 1), 10, 10000); if (ret) dev_err(dev, "AI timeout: failed to set %s reg 0x%02x=0x%08x\n", -- 2.26.2

2 1

[PATCH v5 0/7] Audio Graph Updates
by Sameer Pujar 10 Nov '20

10 Nov '20

This series is a prepraration for using generic graph driver for Tegra210 audio. Tegra audio graph series will be sent in a separate series because it has some dependency over other series for documentation work. This series can focus on the generic ASoC driver updates. Below are the summary of changes done. * Support multiple instances of a component. For example there can be multiple I2S devices which can use the same component driver. * Support open platforms with empty Codec endpoint. Customers can plug their own HW and can populate codec endpoint. * In a component model there can be many components which can be connected together. In such cases Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Expose structures or helpers to be re-used by other similar graph drivers. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Changelog ========= v4 -> v5 -------- * No changes in the core/audio-graph driver patches which are part of the current series. * Dropped Tegra audio patches and doc patches. This will be sent separately once the doc depdendencies are resolved. v3 -> v4 -------- * Added new patches to convert graph.txt and audio-graph-card.txt to corresponding json-schema files. Later these references are used in Tegra audio graph schema. * AHUB component binding docs are updated to reflect the usage of ports/port/endpoint * More common stuff is moved into graph_parse_of() and this is used by both generic and Tegra audio graph. * DT binding for Tegra audio graph is updated to included "ports { }" * As per the suggestion 'void *data' member is dropped from 'asoc_simple_priv' and instead container method is used to maintain required custom data internal to Tegra audio graph. v2 -> v3 -------- * Dropped new compatible addition in generic graph driver after reviewing it with Morimoto-san. Instead added Tegra audio graph driver and new compatibles are added in the same. * Added new patches to expose new members for customization in audio graph driver. * Added new patch for Tegra audio graph driver and related documentation. * Minor change in below commit where mutex version of helper is used "ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM" * DT binding is updated to use the newly exposed compatibles * No changes in other patches v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. Sameer Pujar (7): ASoC: soc-core: Fix component name_prefix parsing ASoC: soc-pcm: Get all BEs along DAPM path ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: audio-graph: Support empty Codec endpoint ASoC: audio-graph: Expose new members for asoc_simple_priv ASoC: audio-graph: Expose helpers from audio graph include/sound/graph_card.h | 16 ++++ include/sound/simple_card_utils.h | 3 + include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 175 ++++++++++++++++++++++++----------- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- 6 files changed, 143 insertions(+), 58 deletions(-) create mode 100644 include/sound/graph_card.h -- 2.7.4

3 10

[PATCH] ASoC: qcom: common: Fix refcounting in qcom_snd_parse_of()
by Dan Carpenter 10 Nov '20

10 Nov '20

There are two issues in this function. 1) We can't drop the refrences on "cpu", "codec" and "platform" before we take the reference. This doesn't cause a problem on the first iteration because those pointers start as NULL so the of_node_put() is a no-op. But on the subsequent iterations, it will lead to a use after free. 2) If the devm_kzalloc() allocation failed then the code returned directly instead of cleaning up. Fixes: c1e6414cdc37 ("ASoC: qcom: common: Fix refcount imbalance on error") Fixes: 1e36ea360ab9 ("ASoC: qcom: common: use modern dai_link style") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/qcom/common.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 54660f126d09..09af00700700 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -58,7 +58,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); if (!dlc) { ret = -ENOMEM; - goto err; + goto err_put_np; } link->cpus = &dlc[0]; @@ -70,7 +70,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) ret = of_property_read_string(np, "link-name", &link->name); if (ret) { dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; + goto err_put_np; } cpu = of_get_child_by_name(np, "cpu"); @@ -130,8 +130,10 @@ int qcom_snd_parse_of(struct snd_soc_card *card) } else { /* DPCM frontend */ dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL); - if (!dlc) - return -ENOMEM; + if (!dlc) { + ret = -ENOMEM; + goto err; + } link->codecs = dlc; link->num_codecs = 1; @@ -158,10 +160,11 @@ int qcom_snd_parse_of(struct snd_soc_card *card) return 0; err: - of_node_put(np); of_node_put(cpu); of_node_put(codec); of_node_put(platform); +err_put_np: + of_node_put(np); return ret; } EXPORT_SYMBOL(qcom_snd_parse_of); -- 2.28.0

2 1

[PATCH v6 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 10 Nov '20

10 Nov '20

Note: - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> v6: - Documentation: Addressed suggestions from Rob Herring. - Define "maxItems: 1" in dmic-gpios property. - Only keep one example and add dmic-gpios property in. v5: - Machine driver: - Fix a format string warning (Reported-by: kernel test robot <lkp(a)intel.com>). detailed info at https://lore.kernel.org/patchwork/patch/1331087/ v4: - Machine driver: Addressed suggestions from Tzung-Bi. - Remove redundant judgments in dmic_set() and dmic_get(). - Remove 1 level indent of judgment of IS_ERR(data->dmic_sel). v3: - Machine driver: Addressed suggestions from Tzung-Bi. - Move variables "dmic_switch" and "dmic_sel" into struct sc7180_snd_data. - Remove redundant judgments in dmic_set(). v2: - Documentation: Modify the dimc-gpios property description and examples. - Machine driver: - Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible - See gpio property and use anadditional control. Thanks for the review! Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/google,sc7180-trogdor.yaml | 8 ++- sound/soc/qcom/sc7180.c | 61 +++++++++++++++++++ 2 files changed, 68 insertions(+), 1 deletion(-) -- 2.25.1

4 8

[PATCH v1 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 10 Nov '20

10 Nov '20

Note: - This patch depends on this patch series ASoC: qcom: dt-bindings: Add sc7180 machine bindings https://patchwork.kernel.org/patch/11773221/ ASoC: qcom: sc7180: Add machine driver for sound card registration https://patchwork.kernel.org/patch/11773223/ - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> Thanks for the review! Ajye Huang (2): ASoC: qcom: dt-bindings: Modify sc7180 machine bindings ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/qcom,sc7180.yaml | 64 ++++++++++++++++++- sound/soc/qcom/sc7180.c | 60 +++++++++++++++++ 2 files changed, 123 insertions(+), 1 deletion(-) -- 2.25.1

4 8

[PATCH v2 0/2] ASoC: soc-compress: tidyup STREAM vs COMPRESS
by Kuninori Morimoto 10 Nov '20

10 Nov '20

Hi Mark These are v2 of tidyup patch-set for SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx. soc-compress is using both SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx, but mixed use. This is confusable, but no problem. Because these are defined as UAPI and are using same value. This patch-set make sure these are same value. v1 -> v2 - checks COMPRESS vs PCM_STREAM by using BUILD_BUG_ON() Link: https://lore.kernel.org/r/87wnzcfnkk.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (2): ASoC: soc-compress: tidyup STREAM vs COMPRESS ASoC: soc-compress: assume SNDRV_PCM_STREAM_xxx and SND_COMPRESS_xxx are same sound/soc/soc-compress.c | 63 +++++++++++++++------------------------- 1 file changed, 23 insertions(+), 40 deletions(-) -- 2.25.1

2 7

[PATCH] ASoC: nau8315: add codec driver
by David Lin 10 Nov '20

10 Nov '20

Add codec driver for Nuvoton NAU8315. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8315.txt | 18 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8315.c | 166 ++++++++++++++++++ 4 files changed, 191 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8315.txt create mode 100644 sound/soc/codecs/nau8315.c diff --git a/Documentation/devicetree/bindings/sound/nau8315.txt b/Documentation/devicetree/bindings/sound/nau8315.txt new file mode 100644 index 000000000000..6eaec46f384c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8315.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8315 Mono Class-D Amplifier + +Required properties: +- compatible : "nuvoton,nau8315" + +Optional properties: +- enable-gpios : GPIO specifier for the chip's device enable input(EN) pin. + If this option is not specified then driver does not manage + the pin state (e.g. chip is always on). + +Example: + +#include <dt-bindings/gpio/gpio.h> + +nau8315 { + compatible = "nuvoton,nau8315"; + enable-gpios = <&gpio1 5 GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 5791b7056af6..2b6eceb3573c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -130,6 +130,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MT6358 imply SND_SOC_MT6359 imply SND_SOC_MT6660 + imply SND_SOC_NAU8315 imply SND_SOC_NAU8540 imply SND_SOC_NAU8810 imply SND_SOC_NAU8822 @@ -1760,6 +1761,10 @@ config SND_SOC_MT6660 Select N if you don't have MT6660 on board. Select M to build this as module. +config SND_SOC_NAU8315 + tristate "Nuvoton Technology Corporation NAU8315 CODEC" + depends on GPIOLIB + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 11ce98c25d6c..144d9256d4d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -129,6 +129,7 @@ snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o snd-soc-mt6359-objs := mt6359.o snd-soc-mt6660-objs := mt6660.o +snd-soc-nau8315-objs := nau8315.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -438,6 +439,7 @@ obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o obj-$(CONFIG_SND_SOC_MT6359) += snd-soc-mt6359.o obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o +obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/nau8315.c b/sound/soc/codecs/nau8315.c new file mode 100644 index 000000000000..e6bc5c0a5036 --- /dev/null +++ b/sound/soc/codecs/nau8315.c @@ -0,0 +1,166 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// nau8315.c -- NAU8315 ALSA SoC Audio Amplifier Driver +// +// Copyright 2020 Nuvoton Technology Crop. +// +// Author: David Lin <ctlin0(a)nuvoton.com> +// +// Based on MAX98357A.c + +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/err.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/soc-dapm.h> + +struct nau8315_priv { + struct gpio_desc *enable; + int enpin_switch; +}; + +static int nau8315_daiops_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (!nau8315->enable) + return 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (nau8315->enpin_switch) { + gpiod_set_value(nau8315->enable, 1); + dev_dbg(component->dev, "set enable to 1"); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + gpiod_set_value(nau8315->enable, 0); + dev_dbg(component->dev, "set enable to 0"); + break; + } + + return 0; +} + +static int nau8315_enpin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8315_priv *nau8315 = + snd_soc_component_get_drvdata(component); + + if (event & SND_SOC_DAPM_POST_PMU) + nau8315->enpin_switch = 1; + else if (event & SND_SOC_DAPM_POST_PMD) + nau8315->enpin_switch = 0; + + return 0; +} + +static const struct snd_soc_dapm_widget nau8315_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("Speaker"), + SND_SOC_DAPM_OUT_DRV_E("EN_Pin", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8315_enpin_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route nau8315_dapm_routes[] = { + {"EN_Pin", NULL, "HiFi Playback"}, + {"Speaker", NULL, "EN_Pin"}, +}; + +static const struct snd_soc_component_driver nau8315_component_driver = { + .dapm_widgets = nau8315_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8315_dapm_widgets), + .dapm_routes = nau8315_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8315_dapm_routes), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops nau8315_dai_ops = { + .trigger = nau8315_daiops_trigger, +}; + +#define NAU8315_RATES SNDRV_PCM_RATE_8000_96000 +#define NAU8315_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE) + +static struct snd_soc_dai_driver nau8315_dai_driver = { + .name = "nau8315-hifi", + .playback = { + .stream_name = "HiFi Playback", + .formats = NAU8315_FORMATS, + .rates = NAU8315_RATES, + .channels_min = 1, + .channels_max = 2, + }, + .ops = &nau8315_dai_ops, +}; + +static int nau8315_platform_probe(struct platform_device *pdev) +{ + struct nau8315_priv *nau8315; + + nau8315 = devm_kzalloc(&pdev->dev, sizeof(*nau8315), GFP_KERNEL); + if (!nau8315) + return -ENOMEM; + + nau8315->enable = devm_gpiod_get_optional(&pdev->dev, + "enable", GPIOD_OUT_LOW); + if (IS_ERR(nau8315->enable)) + return PTR_ERR(nau8315->enable); + + dev_set_drvdata(&pdev->dev, nau8315); + + return devm_snd_soc_register_component(&pdev->dev, + &nau8315_component_driver, + &nau8315_dai_driver, 1); +} + +#ifdef CONFIG_OF +static const struct of_device_id nau8315_device_id[] = { + { .compatible = "nuvoton,nau8315" }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8315_device_id); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8315_acpi_match[] = { + { "NVTN2010", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8315_acpi_match); +#endif + +static struct platform_driver nau8315_platform_driver = { + .driver = { + .name = "nau8315", + .of_match_table = of_match_ptr(nau8315_device_id), + .acpi_match_table = ACPI_PTR(nau8315_acpi_match), + }, + .probe = nau8315_platform_probe, +}; +module_platform_driver(nau8315_platform_driver); + +MODULE_DESCRIPTION("ASoC NAU8315 Mono Class-D Amplifier Driver"); +MODULE_AUTHOR("David Lin <ctlin0(a)nuvoton.com>"); +MODULE_LICENSE("GPL v2"); -- 2.25.1

5 11

[PATCH v4 0/9] ASoC: Mediatek: Add support for MT8192 SoC
by Jiaxin Yu 10 Nov '20

10 Nov '20

This series of patches adds support for Mediatek AFE for MT8192 SoC. At the same time, the calibration function of MT6359 is completed with real machine driver. The patch is based on broonie tree "for-next" branch. Change since v3: - use normal conditional statements to improve legiblity in [v3,3/9] - remove mtk_i2s_hd_en_event as there's trace in the core - impove mt8192_i2s_enum and mt8192_adda_enum Change since v2: - split the dai driver files as a separate patch - fix dt-bindings to GPL-2.0-only License - remove unnecessary preperty descriptions such as 'maxItems' Change since v1: - fixed some typos related to dt-bindings in [v1,3/5] and [v1,5/5] - add vendor prefix to the properties, such as "mediatek,apmixedsys" - add a dependency description to indicate the required header files Jiaxin Yu (9): ASoC: mediatek: mt6359: add the calibration functions ASoC: mediatek: mt8192: add platform driver ASoC: mediatek: mt8192: support i2s in platform driver ASoC: mediatek: mt8192: support adda in platform driver ASoC: mediatek: mt8192: support pcm in platform driver ASoC: mediatek: mt8192: support tdm in platform driver dt-bindings: mediatek: mt8192: add audio afe document ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682 dt-bindings: mediatek: mt8192: add mt8192-mt6358-rt1015-rt5682 document .../bindings/sound/mt8192-afe-pcm.yaml | 100 + .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 42 + sound/soc/codecs/mt6359.c | 110 + sound/soc/codecs/mt6359.h | 7 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 13 +- sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8192/Makefile | 16 + sound/soc/mediatek/mt8192/mt8192-afe-clk.c | 669 ++++ sound/soc/mediatek/mt8192/mt8192-afe-clk.h | 244 ++ sound/soc/mediatek/mt8192/mt8192-afe-common.h | 170 + .../soc/mediatek/mt8192/mt8192-afe-control.c | 163 + sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 306 ++ sound/soc/mediatek/mt8192/mt8192-afe-gpio.h | 19 + sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2389 +++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 1471 ++++++++ sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 2110 +++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-pcm.c | 409 +++ sound/soc/mediatek/mt8192/mt8192-dai-tdm.c | 778 ++++ .../mediatek/mt8192/mt8192-interconnection.h | 65 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 1058 ++++++ sound/soc/mediatek/mt8192/mt8192-reg.h | 3131 +++++++++++++++++ 23 files changed, 13291 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8192-mt6359-rt1015-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8192/Makefile create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-clk.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-common.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-control.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-adda.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-interconnection.h create mode 100644 sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c create mode 100644 sound/soc/mediatek/mt8192/mt8192-reg.h -- 2.18.0

4 12

[PATCH v3 0/9] slab: provide and use krealloc_array()
by Bartosz Golaszewski 10 Nov '20

10 Nov '20

From: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Andy brought to my attention the fact that users allocating an array of equally sized elements should check if the size multiplication doesn't overflow. This is why we have helpers like kmalloc_array(). However we don't have krealloc_array() equivalent and there are many users who do their own multiplication when calling krealloc() for arrays. This series provides krealloc_array() and uses it in a couple places. A separate series will follow adding devm_krealloc_array() which is needed in the xilinx adc driver. v1 -> v2: - added a kernel doc for krealloc_array() - mentioned krealloc et al in the docs - collected review tags v2 -> v3: - add a patch improving krealloc()'s kerneldoc - fix a typo - improve .rst doc - tweak line breaks Bartosz Golaszewski (9): mm: slab: clarify krealloc()'s behavior with __GFP_ZERO mm: slab: provide krealloc_array() ALSA: pcm: use krealloc_array() vhost: vringh: use krealloc_array() pinctrl: use krealloc_array() edac: ghes: use krealloc_array() drm: atomic: use krealloc_array() hwtracing: intel: use krealloc_array() dma-buf: use krealloc_array() Documentation/core-api/memory-allocation.rst | 4 ++++ drivers/dma-buf/sync_file.c | 3 +-- drivers/edac/ghes_edac.c | 4 ++-- drivers/gpu/drm/drm_atomic.c | 3 ++- drivers/hwtracing/intel_th/msu.c | 2 +- drivers/pinctrl/pinctrl-utils.c | 2 +- drivers/vhost/vringh.c | 3 ++- include/linux/slab.h | 18 ++++++++++++++++++ mm/slab_common.c | 6 +++--- sound/core/pcm_lib.c | 4 ++-- 10 files changed, 36 insertions(+), 13 deletions(-) -- 2.29.1

2 10

Subject: [RFC] clang tooling cleanups
by trix＠redhat.com 10 Nov '20

10 Nov '20

This rfc will describe An upcoming treewide cleanup. How clang tooling was used to programatically do the clean up. Solicit opinions on how to generally use clang tooling. The clang warning -Wextra-semi-stmt produces about 10k warnings. Reviewing these, a subset of semicolon after a switch looks safe to fix all the time. An example problem void foo(int a) { switch(a) { case 1: ... }; <--- extra semicolon } Treewide, there are about 100 problems in 50 files for x86_64 allyesconfig. These fixes will be the upcoming cleanup. clang already supports fixing this problem. Add to your command line clang -c -Wextra-semi-stmt -Xclang -fixit foo.c foo.c:8:3: warning: empty expression statement has no effect; remove unnecessary ';' to silence this warning [-Wextra-semi-stmt] }; ^ foo.c:8:3: note: FIX-IT applied suggested code changes 1 warning generated. The big problem is using this treewide is it will fix all 10k problems. 10k changes to analyze and upstream is not practical. Another problem is the generic fixer only removes the semicolon. So empty lines with some tabs need to be manually cleaned. What is needed is a more precise fixer. Enter clang-tidy. https://clang.llvm.org/extra/clang-tidy/ Already part of the static checker infrastructure, invoke on the clang build with make clang-tidy It is only a matter of coding up a specific checker for the cleanup. Upstream this is review is happening here https://reviews.llvm.org/D90180 The development of a checker/fixer is Start with a reproducer void foo (int a) { switch (a) {}; } Generate the abstract syntax tree (AST) clang -Xclang -ast-dump foo.c `-FunctionDecl |-ParmVarDecl `-CompoundStmt |-SwitchStmt | |-ImplicitCastExpr | | `-DeclRefExpr | `-CompoundStmt `-NullStmt Write a matcher to get you most of the way void SwitchSemiCheck::registerMatchers(MatchFinder *Finder) { Finder->addMatcher( compoundStmt(has(switchStmt().bind("switch"))).bind("comp"), this); } The 'bind' method is important, it allows a string to be associated with a node in the AST. In this case these are `-FunctionDecl |-ParmVarDecl `-CompoundStmt <-------- comp |-SwitchStmt <-------- switch | |-ImplicitCastExpr | | `-DeclRefExpr | `-CompoundStmt `-NullStmt When a match is made the 'check' method will be called. void SwitchSemiCheck::check(const MatchFinder::MatchResult &Result) { auto *C = Result.Nodes.getNodeAs<CompoundStmt>("comp"); auto *S = Result.Nodes.getNodeAs<SwitchStmt>("switch"); This is where the string in the bind calls are changed to nodes `-FunctionDecl |-ParmVarDecl `-CompoundStmt <-------- comp, C |-SwitchStmt <-------- switch, S | |-ImplicitCastExpr | | `-DeclRefExpr | `-CompoundStmt `-NullStmt <---------- looking for N And then more logic to find the NullStmt auto Current = C->body_begin(); auto Next = Current; Next++; while (Next != C->body_end()) { if (*Current == S) { if (const auto *N = dyn_cast<NullStmt>(*Next)) { When it is found, a warning is printed and a FixItHint is proposed. auto H = FixItHint::CreateReplacement( SourceRange(S->getBody()->getEndLoc(), N->getSemiLoc()), "}"); diag(N->getSemiLoc(), "unneeded semicolon") << H; This fixit replaces from the end of switch to the semicolon with a '}'. Because the end of the switch is '}' this has the effect of removing all the whitespace as well as the semicolon. Because of the checker's placement in clang-tidy existing linuxkernel checkers, all that was needed to fix the tree was to add a '-fix'to the build's clang-tidy call. I am looking for opinions on what we want to do specifically with cleanups and generally about other source-to-source programmatic changes to the code base. For cleanups, I think we need a new toplevel target clang-tidy-fix And an explicit list of fixers that have a very high (100%?) fix rate. Ideally a bot should make the changes, but a bot could also nag folks. Is there interest in a bot making the changes? Does one already exist? The general source-to-source is a bit blue sky. Ex/ could automagicly refactor api, outline similar cut-n-pasted functions etc. Anything on someone's wishlist you want to try out ? Signed-off-by: Tom Rix <trix(a)redhat.com>

4 3

[PATCH] ALSA: firewire: fix comparison to bool warning
by xiakaixu1987＠gmail.com 10 Nov '20

10 Nov '20

From: Kaixu Xia <kaixuxia(a)tencent.com> Fix the following coccicheck warning: ./sound/firewire/amdtp-stream.h:273:6-19: WARNING: Comparison to bool Reported-by: Tosk Robot <tencent_os_robot(a)tencent.com> Signed-off-by: Kaixu Xia <kaixuxia(a)tencent.com> --- sound/firewire/amdtp-stream.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 2ceb57d1d58e..a3daa1f2c1c4 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -270,7 +270,7 @@ static inline bool amdtp_stream_wait_callback(struct amdtp_stream *s, unsigned int timeout) { return wait_event_timeout(s->callback_wait, - s->callbacked == true, + s->callbacked, msecs_to_jiffies(timeout)) > 0; } -- 2.20.0

3 2

[sound:test/usb-hacks 14/20] sound/usb/pcm.c:579:35: warning: variable 'altsd' set but not used
by kernel test robot 10 Nov '20

10 Nov '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/usb-hacks head: 2adedc89c82670019ee3b016d2a2237176768e3a commit: 7502ae0cddc65ab020cd3c3dc4085f3e70f70f21 [14/20] ALSA: usb-audio: Track implicit fb sync endpoint in audioformat list config: mips-randconfig-r025-20201110 (attached as .config) compiler: mips-linux-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound test/usb-hacks git checkout 7502ae0cddc65ab020cd3c3dc4085f3e70f70f21 # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=mips If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): sound/usb/pcm.c: In function 'set_format': >> sound/usb/pcm.c:579:35: warning: variable 'altsd' set but not used [-Wunused-but-set-variable] 579 | struct usb_interface_descriptor *altsd; | ^~~~~ vim +/altsd +579 sound/usb/pcm.c 71bb64c56d787a2 Eldad Zack 2013-08-03 571 71bb64c56d787a2 Eldad Zack 2013-08-03 572 /* 71bb64c56d787a2 Eldad Zack 2013-08-03 573 * find a matching format and set up the interface 71bb64c56d787a2 Eldad Zack 2013-08-03 574 */ 71bb64c56d787a2 Eldad Zack 2013-08-03 575 static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) 71bb64c56d787a2 Eldad Zack 2013-08-03 576 { 71bb64c56d787a2 Eldad Zack 2013-08-03 577 struct usb_device *dev = subs->dev; 71bb64c56d787a2 Eldad Zack 2013-08-03 578 struct usb_host_interface *alts; 71bb64c56d787a2 Eldad Zack 2013-08-03 @579 struct usb_interface_descriptor *altsd; 71bb64c56d787a2 Eldad Zack 2013-08-03 580 struct usb_interface *iface; 71bb64c56d787a2 Eldad Zack 2013-08-03 581 int err; 71bb64c56d787a2 Eldad Zack 2013-08-03 582 71bb64c56d787a2 Eldad Zack 2013-08-03 583 iface = usb_ifnum_to_if(dev, fmt->iface); 71bb64c56d787a2 Eldad Zack 2013-08-03 584 if (WARN_ON(!iface)) 71bb64c56d787a2 Eldad Zack 2013-08-03 585 return -EINVAL; b099b9693d23d03 Takashi Iwai 2018-05-02 586 alts = usb_altnum_to_altsetting(iface, fmt->altsetting); 0141254b0a74b37 Johan Hovold 2019-12-20 587 if (WARN_ON(!alts)) 71bb64c56d787a2 Eldad Zack 2013-08-03 588 return -EINVAL; 0141254b0a74b37 Johan Hovold 2019-12-20 589 altsd = get_iface_desc(alts); 71bb64c56d787a2 Eldad Zack 2013-08-03 590 92adc96f8eecd95 Hui Wang 2019-12-18 591 if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt) 71bb64c56d787a2 Eldad Zack 2013-08-03 592 return 0; 71bb64c56d787a2 Eldad Zack 2013-08-03 593 71bb64c56d787a2 Eldad Zack 2013-08-03 594 /* close the old interface */ 92adc96f8eecd95 Hui Wang 2019-12-18 595 if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) { 8a463225b110474 Takashi Iwai 2018-05-02 596 if (!subs->stream->chip->keep_iface) { 71bb64c56d787a2 Eldad Zack 2013-08-03 597 err = usb_set_interface(subs->dev, subs->interface, 0); 71bb64c56d787a2 Eldad Zack 2013-08-03 598 if (err < 0) { 0ba41d917eeb87f Takashi Iwai 2014-02-26 599 dev_err(&dev->dev, 0ba41d917eeb87f Takashi Iwai 2014-02-26 600 "%d:%d: return to setting 0 failed (%d)\n", 0ba41d917eeb87f Takashi Iwai 2014-02-26 601 fmt->iface, fmt->altsetting, err); 71bb64c56d787a2 Eldad Zack 2013-08-03 602 return -EIO; 71bb64c56d787a2 Eldad Zack 2013-08-03 603 } 8a463225b110474 Takashi Iwai 2018-05-02 604 } 71bb64c56d787a2 Eldad Zack 2013-08-03 605 subs->interface = -1; 71bb64c56d787a2 Eldad Zack 2013-08-03 606 subs->altset_idx = 0; 71bb64c56d787a2 Eldad Zack 2013-08-03 607 } 71bb64c56d787a2 Eldad Zack 2013-08-03 608 92adc96f8eecd95 Hui Wang 2019-12-18 609 if (subs->need_setup_fmt) 92adc96f8eecd95 Hui Wang 2019-12-18 610 subs->need_setup_fmt = false; 92adc96f8eecd95 Hui Wang 2019-12-18 611 71bb64c56d787a2 Eldad Zack 2013-08-03 612 /* set interface */ b099b9693d23d03 Takashi Iwai 2018-05-02 613 if (iface->cur_altsetting != alts) { 6874daad4b0fbed Jurgen Kramer 2014-11-28 614 err = snd_usb_select_mode_quirk(subs, fmt); 6874daad4b0fbed Jurgen Kramer 2014-11-28 615 if (err < 0) 6874daad4b0fbed Jurgen Kramer 2014-11-28 616 return -EIO; 6874daad4b0fbed Jurgen Kramer 2014-11-28 617 71bb64c56d787a2 Eldad Zack 2013-08-03 618 err = usb_set_interface(dev, fmt->iface, fmt->altsetting); 71bb64c56d787a2 Eldad Zack 2013-08-03 619 if (err < 0) { 0ba41d917eeb87f Takashi Iwai 2014-02-26 620 dev_err(&dev->dev, 0ba41d917eeb87f Takashi Iwai 2014-02-26 621 "%d:%d: usb_set_interface failed (%d)\n", 0ba41d917eeb87f Takashi Iwai 2014-02-26 622 fmt->iface, fmt->altsetting, err); 71bb64c56d787a2 Eldad Zack 2013-08-03 623 return -EIO; 71bb64c56d787a2 Eldad Zack 2013-08-03 624 } 0ba41d917eeb87f Takashi Iwai 2014-02-26 625 dev_dbg(&dev->dev, "setting usb interface %d:%d\n", 71bb64c56d787a2 Eldad Zack 2013-08-03 626 fmt->iface, fmt->altsetting); 71bb64c56d787a2 Eldad Zack 2013-08-03 627 snd_usb_set_interface_quirk(dev); 71bb64c56d787a2 Eldad Zack 2013-08-03 628 } 71bb64c56d787a2 Eldad Zack 2013-08-03 629 b099b9693d23d03 Takashi Iwai 2018-05-02 630 subs->interface = fmt->iface; b099b9693d23d03 Takashi Iwai 2018-05-02 631 subs->altset_idx = fmt->altset_idx; 71bb64c56d787a2 Eldad Zack 2013-08-03 632 subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, 71bb64c56d787a2 Eldad Zack 2013-08-03 633 alts, fmt->endpoint, subs->direction, 71bb64c56d787a2 Eldad Zack 2013-08-03 634 SND_USB_ENDPOINT_TYPE_DATA); 71bb64c56d787a2 Eldad Zack 2013-08-03 635 71bb64c56d787a2 Eldad Zack 2013-08-03 636 if (!subs->data_endpoint) 71bb64c56d787a2 Eldad Zack 2013-08-03 637 return -EINVAL; 71bb64c56d787a2 Eldad Zack 2013-08-03 638 7502ae0cddc65ab Takashi Iwai 2020-11-05 639 err = set_sync_endpoint(subs, fmt); 71bb64c56d787a2 Eldad Zack 2013-08-03 640 if (err < 0) 71bb64c56d787a2 Eldad Zack 2013-08-03 641 return err; 71bb64c56d787a2 Eldad Zack 2013-08-03 642 d133f2c22e9cb7b Eldad Zack 2013-08-03 643 err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt); d133f2c22e9cb7b Eldad Zack 2013-08-03 644 if (err < 0) e5779998bf8b70e Daniel Mack 2010-03-04 645 return err; e5779998bf8b70e Daniel Mack 2010-03-04 646 e5779998bf8b70e Daniel Mack 2010-03-04 647 subs->cur_audiofmt = fmt; e5779998bf8b70e Daniel Mack 2010-03-04 648 e5779998bf8b70e Daniel Mack 2010-03-04 649 snd_usb_set_format_quirk(subs, fmt); e5779998bf8b70e Daniel Mack 2010-03-04 650 e5779998bf8b70e Daniel Mack 2010-03-04 651 return 0; e5779998bf8b70e Daniel Mack 2010-03-04 652 } e5779998bf8b70e Daniel Mack 2010-03-04 653 :::::: The code at line 579 was first introduced by commit :::::: 71bb64c56d787a221752b1de034fe8c07c737f5c ALSA: usb-audio: separate sync endpoint setting from set_format :::::: TO: Eldad Zack <eldad(a)fogrefinery.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[sound:test/usb-hacks 14/20] sound/usb/pcm.c:579:35: warning: variable 'altsd' set but not used
by kernel test robot 10 Nov '20

10 Nov '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/usb-hacks head: 2adedc89c82670019ee3b016d2a2237176768e3a commit: 7502ae0cddc65ab020cd3c3dc4085f3e70f70f21 [14/20] ALSA: usb-audio: Track implicit fb sync endpoint in audioformat list config: i386-allyesconfig (attached as .config) compiler: gcc-9 (Debian 9.3.0-15) 9.3.0 reproduce (this is a W=1 build): # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound test/usb-hacks git checkout 7502ae0cddc65ab020cd3c3dc4085f3e70f70f21 # save the attached .config to linux build tree make W=1 ARCH=i386 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): sound/usb/pcm.c: In function 'set_format': >> sound/usb/pcm.c:579:35: warning: variable 'altsd' set but not used [-Wunused-but-set-variable] 579 | struct usb_interface_descriptor *altsd; | ^~~~~ vim +/altsd +579 sound/usb/pcm.c 71bb64c56d787a Eldad Zack 2013-08-03 571 71bb64c56d787a Eldad Zack 2013-08-03 572 /* 71bb64c56d787a Eldad Zack 2013-08-03 573 * find a matching format and set up the interface 71bb64c56d787a Eldad Zack 2013-08-03 574 */ 71bb64c56d787a Eldad Zack 2013-08-03 575 static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) 71bb64c56d787a Eldad Zack 2013-08-03 576 { 71bb64c56d787a Eldad Zack 2013-08-03 577 struct usb_device *dev = subs->dev; 71bb64c56d787a Eldad Zack 2013-08-03 578 struct usb_host_interface *alts; 71bb64c56d787a Eldad Zack 2013-08-03 @579 struct usb_interface_descriptor *altsd; 71bb64c56d787a Eldad Zack 2013-08-03 580 struct usb_interface *iface; 71bb64c56d787a Eldad Zack 2013-08-03 581 int err; 71bb64c56d787a Eldad Zack 2013-08-03 582 71bb64c56d787a Eldad Zack 2013-08-03 583 iface = usb_ifnum_to_if(dev, fmt->iface); 71bb64c56d787a Eldad Zack 2013-08-03 584 if (WARN_ON(!iface)) 71bb64c56d787a Eldad Zack 2013-08-03 585 return -EINVAL; b099b9693d23d0 Takashi Iwai 2018-05-02 586 alts = usb_altnum_to_altsetting(iface, fmt->altsetting); 0141254b0a74b3 Johan Hovold 2019-12-20 587 if (WARN_ON(!alts)) 71bb64c56d787a Eldad Zack 2013-08-03 588 return -EINVAL; 0141254b0a74b3 Johan Hovold 2019-12-20 589 altsd = get_iface_desc(alts); 71bb64c56d787a Eldad Zack 2013-08-03 590 92adc96f8eecd9 Hui Wang 2019-12-18 591 if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt) 71bb64c56d787a Eldad Zack 2013-08-03 592 return 0; 71bb64c56d787a Eldad Zack 2013-08-03 593 71bb64c56d787a Eldad Zack 2013-08-03 594 /* close the old interface */ 92adc96f8eecd9 Hui Wang 2019-12-18 595 if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) { 8a463225b11047 Takashi Iwai 2018-05-02 596 if (!subs->stream->chip->keep_iface) { 71bb64c56d787a Eldad Zack 2013-08-03 597 err = usb_set_interface(subs->dev, subs->interface, 0); 71bb64c56d787a Eldad Zack 2013-08-03 598 if (err < 0) { 0ba41d917eeb87 Takashi Iwai 2014-02-26 599 dev_err(&dev->dev, 0ba41d917eeb87 Takashi Iwai 2014-02-26 600 "%d:%d: return to setting 0 failed (%d)\n", 0ba41d917eeb87 Takashi Iwai 2014-02-26 601 fmt->iface, fmt->altsetting, err); 71bb64c56d787a Eldad Zack 2013-08-03 602 return -EIO; 71bb64c56d787a Eldad Zack 2013-08-03 603 } 8a463225b11047 Takashi Iwai 2018-05-02 604 } 71bb64c56d787a Eldad Zack 2013-08-03 605 subs->interface = -1; 71bb64c56d787a Eldad Zack 2013-08-03 606 subs->altset_idx = 0; 71bb64c56d787a Eldad Zack 2013-08-03 607 } 71bb64c56d787a Eldad Zack 2013-08-03 608 92adc96f8eecd9 Hui Wang 2019-12-18 609 if (subs->need_setup_fmt) 92adc96f8eecd9 Hui Wang 2019-12-18 610 subs->need_setup_fmt = false; 92adc96f8eecd9 Hui Wang 2019-12-18 611 71bb64c56d787a Eldad Zack 2013-08-03 612 /* set interface */ b099b9693d23d0 Takashi Iwai 2018-05-02 613 if (iface->cur_altsetting != alts) { 6874daad4b0fbe Jurgen Kramer 2014-11-28 614 err = snd_usb_select_mode_quirk(subs, fmt); 6874daad4b0fbe Jurgen Kramer 2014-11-28 615 if (err < 0) 6874daad4b0fbe Jurgen Kramer 2014-11-28 616 return -EIO; 6874daad4b0fbe Jurgen Kramer 2014-11-28 617 71bb64c56d787a Eldad Zack 2013-08-03 618 err = usb_set_interface(dev, fmt->iface, fmt->altsetting); 71bb64c56d787a Eldad Zack 2013-08-03 619 if (err < 0) { 0ba41d917eeb87 Takashi Iwai 2014-02-26 620 dev_err(&dev->dev, 0ba41d917eeb87 Takashi Iwai 2014-02-26 621 "%d:%d: usb_set_interface failed (%d)\n", 0ba41d917eeb87 Takashi Iwai 2014-02-26 622 fmt->iface, fmt->altsetting, err); 71bb64c56d787a Eldad Zack 2013-08-03 623 return -EIO; 71bb64c56d787a Eldad Zack 2013-08-03 624 } 0ba41d917eeb87 Takashi Iwai 2014-02-26 625 dev_dbg(&dev->dev, "setting usb interface %d:%d\n", 71bb64c56d787a Eldad Zack 2013-08-03 626 fmt->iface, fmt->altsetting); 71bb64c56d787a Eldad Zack 2013-08-03 627 snd_usb_set_interface_quirk(dev); 71bb64c56d787a Eldad Zack 2013-08-03 628 } 71bb64c56d787a Eldad Zack 2013-08-03 629 b099b9693d23d0 Takashi Iwai 2018-05-02 630 subs->interface = fmt->iface; b099b9693d23d0 Takashi Iwai 2018-05-02 631 subs->altset_idx = fmt->altset_idx; 71bb64c56d787a Eldad Zack 2013-08-03 632 subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, 71bb64c56d787a Eldad Zack 2013-08-03 633 alts, fmt->endpoint, subs->direction, 71bb64c56d787a Eldad Zack 2013-08-03 634 SND_USB_ENDPOINT_TYPE_DATA); 71bb64c56d787a Eldad Zack 2013-08-03 635 71bb64c56d787a Eldad Zack 2013-08-03 636 if (!subs->data_endpoint) 71bb64c56d787a Eldad Zack 2013-08-03 637 return -EINVAL; 71bb64c56d787a Eldad Zack 2013-08-03 638 7502ae0cddc65a Takashi Iwai 2020-11-05 639 err = set_sync_endpoint(subs, fmt); 71bb64c56d787a Eldad Zack 2013-08-03 640 if (err < 0) 71bb64c56d787a Eldad Zack 2013-08-03 641 return err; 71bb64c56d787a Eldad Zack 2013-08-03 642 d133f2c22e9cb7 Eldad Zack 2013-08-03 643 err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt); d133f2c22e9cb7 Eldad Zack 2013-08-03 644 if (err < 0) e5779998bf8b70 Daniel Mack 2010-03-04 645 return err; e5779998bf8b70 Daniel Mack 2010-03-04 646 e5779998bf8b70 Daniel Mack 2010-03-04 647 subs->cur_audiofmt = fmt; e5779998bf8b70 Daniel Mack 2010-03-04 648 e5779998bf8b70 Daniel Mack 2010-03-04 649 snd_usb_set_format_quirk(subs, fmt); e5779998bf8b70 Daniel Mack 2010-03-04 650 e5779998bf8b70 Daniel Mack 2010-03-04 651 return 0; e5779998bf8b70 Daniel Mack 2010-03-04 652 } e5779998bf8b70 Daniel Mack 2010-03-04 653 :::::: The code at line 579 was first introduced by commit :::::: 71bb64c56d787a221752b1de034fe8c07c737f5c ALSA: usb-audio: separate sync endpoint setting from set_format :::::: TO: Eldad Zack <eldad(a)fogrefinery.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[PATCH AUTOSEL 4.19 02/21] ALSA: hda: Reinstate runtime_allow() for all hda controllers
by Sasha Levin 10 Nov '20

10 Nov '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit 9fc149c3bce7bdbb94948a8e6bd025e3b3538603 ] The broken jack detection should be fixed by commit a6e7d0a4bdb0 ("ALSA: hda: fix jack detection with Realtek codecs when in D3"), let's try enabling runtime PM by default again. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20201027130038.16463-4-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d43245937db7e..8e1eb5f243a27 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2478,6 +2478,7 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); + pm_runtime_allow(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.27.0

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[PATCH AUTOSEL 5.4 01/42] ASoC: qcom: sdm845: set driver name correctly
by Sasha Levin 10 Nov '20

10 Nov '20

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 3f48b6eba15ea342ef4cb420b580f5ed6605669f ] With the current state of code, we would endup with something like below in /proc/asound/cards for 2 machines based on this driver. Machine 1: 0 [DB845c ]: DB845c - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: Lenovo-YOGA-C63 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 This is not very UCM friendly both w.r.t to common up configs and card identification, and UCM2 became totally not usefull with just one ucm sdm845.conf for two machines which have different setups w.r.t HDMI and other dais. Reasons for such thing is partly because Qualcomm machine drivers never cared to set driver_name. This patch sets up driver name for the this driver to sort out the UCM integration issues! after this patch contents of /proc/asound/cards: Machine 1: 0 [DB845c ]: sdm845 - DB845c DB845c Machine 2: 0 [LenovoYOGAC6301]: sdm845 - Lenovo-YOGA-C630-13Q50 LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216 with this its possible to align with what UCM2 expects and we can have sdm845/DB845.conf sdm845/LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216.conf ... for board variants. This should scale much better! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Link: https://lore.kernel.org/r/20201023095849.22894-1-srinivas.kandagatla@linaro… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/sdm845.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 7e6c41e63d8e1..23e1de61e92e4 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -16,6 +16,7 @@ #include "qdsp6/q6afe.h" #include "../codecs/rt5663.h" +#define DRIVER_NAME "sdm845" #define DEFAULT_SAMPLE_RATE_48K 48000 #define DEFAULT_MCLK_RATE 24576000 #define TDM_BCLK_RATE 6144000 @@ -407,6 +408,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->driver_name = DRIVER_NAME; card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; -- 2.27.0

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[PATCH AUTOSEL 5.9 01/55] ASoC: mediatek: mt8183-da7219: fix DAPM paths for rt1015
by Sasha Levin 10 Nov '20

10 Nov '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit eb5a558705c7f63d06b4ddd072898b1ca894e053 ] RT1015's output widget name is "SPO" instead of "Speaker". Fixes it to use the correct names. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20201019044724.1601476-1-tzungbi@google.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 31 +++++++++++++++---- 1 file changed, 25 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index a6c690c5308d3..58b76e985f7f3 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -624,15 +624,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = { }, }; +static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const +struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL", + "aud_tdm_out_on", "aud_tdm_out_off"), +}; + +static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = { + {"Left Spk", NULL, "Left SPO"}, + {"Right Spk", NULL, "Right SPO"}, + {"I2S Playback", NULL, "TDM_OUT_PINCTRL"}, +}; + static struct snd_soc_card mt8183_da7219_rt1015_card = { .name = "mt8183_da7219_rt1015", .owner = THIS_MODULE, - .controls = mt8183_da7219_max98357_snd_controls, - .num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls), - .dapm_widgets = mt8183_da7219_max98357_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets), - .dapm_routes = mt8183_da7219_max98357_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes), + .controls = mt8183_da7219_rt1015_snd_controls, + .num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls), + .dapm_widgets = mt8183_da7219_rt1015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets), + .dapm_routes = mt8183_da7219_rt1015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes), .dai_link = mt8183_da7219_dai_links, .num_links = ARRAY_SIZE(mt8183_da7219_dai_links), .aux_dev = &mt8183_da7219_max98357_headset_dev, -- 2.27.0

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ASoC: Question regarding device-tree multi-lane I2S for Allwinner SoC
by Clément Péron 09 Nov '20

09 Nov '20

Hi device-tree and sound Maintainers, I have a question regarding multi-lane i2S representation. On the Allwinner SoC the I2S/PCM Interface can handle up to four lanes as output or input. For each lane we can enable up to sixteen slots. And for each output slot we can choose which slot to map. The only representation I found is for Amlogic device-tree they did the following sound node : /* 8ch hdmi interface */ dai-link-7 { sound-dai = <&tdmif_b>; dai-format = "i2s"; dai-tdm-slot-tx-mask-0 = <1 1>; dai-tdm-slot-tx-mask-1 = <1 1>; dai-tdm-slot-tx-mask-2 = <1 1>; dai-tdm-slot-tx-mask-3 = <1 1>; mclk-fs = <256>; codec { sound-dai = <&tohdmitx TOHDMITX_I2S_IN_B>; }; }; This kind of representation gives the information that 2 slots should be enabled as TX per lane but don't give which slot to map. I was thinking about a representation per lane but maybe it's a bit complicated ? dai-format = "dsp_a"; dai-tdm-slot-width = <32>; // Lane 0 : Output 8 channels 0-7 using TDM dai-tdm-slot-tx-mask-0 = <1 1 1 1 1 1 1 1>; dai-tdm-slot-chmap-0 = <0 1 2 3 4 5 6 7>; // Lane 1 : Output 3 channels 5-7 using TDM dai-tdm-slot-tx-mask-1 = <1 1 1>; dai-tdm-slot-chmap-1 = <5 6 7>; I will only start to support HDMI, so 4 I2S lanes for now it should look like this dai-format = "i2s"; dai-tdm-slot-width = <32>; frame-inversion; // Lane 0 : Output channels 0,1 dai-tdm-slot-tx-mask-0 = <1 1>; dai-tdm-slot-chmap-0 = <0 1>; // Lane 1: Output channels 2,3 dai-tdm-slot-tx-mask-1 = <1 1>; dai-tdm-slot-chmap-1 = <2 3>; // Lane 2: Output channels 4,5 dai-tdm-slot-tx-mask-2 = <1 1>; dai-tdm-slot-chmap-2 = <4 5>; // Lane 3: Output channels 6,7 dai-tdm-slot-tx-mask-3 = <1 1>; dai-tdm-slot-chmap-3 = <6 7>; What do you think? Do you have any remark / idea about this ? Thanks for your help Clement

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USB Audio device: Shure MV7 not recognized
by GitHub issues - edited 09 Nov '20

09 Nov '20

alsa-project/alsa-lib issue #98 was edited from vchuravy: Hello! i hope this is the right venue for reporting not working audio devices. ``` > alsa-capabilities 9: `USB Audio' Digital USB Audio Class audio output on device Shure MV7 -hardware address: hw:3,0 (or hw:MV7,0) -device label: Shure MV7 -formats: (n/a) -usb audio class: UAC2 (isochronous asynchronous) -streamfile: /proc/asound/card3/stream0 -character device: /dev/snd/pcmC3D0p -monitor file: /proc/asound/card3/pcm0p/sub0/hw_params ``` `alsa-info.sh`: http://alsa-project.org/db/?f=7bfa1d9ac09aac53a52834e95b833d299dfe3896 `pulseaudio -vvv` https://gist.github.com/vchuravy/f5f8ee972d5e2179387c8c53423379f6 During the pulseaudio session I plugged the device in and then unplugged it. I am happy to dig in, but I would need some guidance on where to start. As a side-note the microphone seems to work on Android so I am wondering if they are using a different driver. The device has a head-phone jack with feedback from the mic, and without pulseaudio running I hear the microphone inputs. So I was wondering if this was in particular a pulseaudio issue. ``` > arecord -l **** List of CAPTURE Hardware Devices **** card 1: Generic [HD-Audio Generic], device 0: ALCS1200A Analog [ALCS1200A Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 2: ALCS1200A Alt Analog [ALCS1200A Alt Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: C925e [Logitech Webcam C925e], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: MV7 [Shure MV7], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 > arecord -d 3 -f U8 sample.mp3 ALSA lib pcm_dsnoop.c:638:(snd_pcm_dsnoop_open) unable to open slave arecord: main:830: audio open error: No such file or directory ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/98 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

USB Audio device: Shure MV7 not recognized
by GitHub issues - edited 09 Nov '20

09 Nov '20

alsa-project/alsa-lib issue #98 was edited from vchuravy: Hello! i hope this is the right venue for reporting not working audio devices. ``` > alsa-capabilities 9: `USB Audio' Digital USB Audio Class audio output on device Shure MV7 -hardware address: hw:3,0 (or hw:MV7,0) -device label: Shure MV7 -formats: (n/a) -usb audio class: UAC2 (isochronous asynchronous) -streamfile: /proc/asound/card3/stream0 -character device: /dev/snd/pcmC3D0p -monitor file: /proc/asound/card3/pcm0p/sub0/hw_params ``` `alsa-info.sh`: http://alsa-project.org/db/?f=7bfa1d9ac09aac53a52834e95b833d299dfe3896 `pulseaudio -vvv` https://gist.github.com/vchuravy/f5f8ee972d5e2179387c8c53423379f6 During the pulseaudio session I plugged the device in and then unplugged it. I am happy to dig in, but I would need some guidance on where to start. As a side-note the microphone seems to work on Android so I am wondering if they are using a different driver. Issue URL : https://github.com/alsa-project/alsa-lib/issues/98 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

USB Audio device: Shure MV7 not recognized
by GitHub issues - opened 09 Nov '20

09 Nov '20

alsa-project/alsa-lib issue #98 was opened from vchuravy: Hello! i hope this is the right venue for reporting not working audio devices. ``` > alsa-capabilities 9: `USB Audio' Digital USB Audio Class audio output on device Shure MV7 -hardware address: hw:3,0 (or hw:MV7,0) -device label: Shure MV7 -formats: (n/a) -usb audio class: UAC2 (isochronous asynchronous) -streamfile: /proc/asound/card3/stream0 -character device: /dev/snd/pcmC3D0p -monitor file: /proc/asound/card3/pcm0p/sub0/hw_params ``` `alsa-info.sh`: http://alsa-project.org/db/?f=1fb51d03454930fce4d6164323cfb7b3c7c88dda `pulseaudio -vvv` https://gist.github.com/vchuravy/f5f8ee972d5e2179387c8c53423379f6 During the pulseaudio session I plugged the device in and then unplugged it. I am happy to dig in, but I would need some guidance on where to start. As a side-note the microphone seems to work on Android so I am wondering if they are using a different driver. Issue URL : https://github.com/alsa-project/alsa-lib/issues/98 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: qcom: sc7180: Add missing PM ops
by Cheng-Yi Chiang 09 Nov '20

09 Nov '20

Use PM ops snd_soc_pm_ops to handle suspend/resume like other machine drivers. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sc7180.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index b391f64c3a80..42e366ecc689 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -258,6 +258,7 @@ static struct platform_driver sc7180_snd_driver = { .driver = { .name = "msm-snd-sc7180", .of_match_table = sc7180_snd_device_id, + .pm = &snd_soc_pm_ops, }, }; module_platform_driver(sc7180_snd_driver); -- 2.29.2.222.g5d2a92d10f8-goog

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[PATCH 0/4] ASoC: ti: davinci-mcasp: Handle incomplete DT node gracefully
by Peter Ujfalusi 09 Nov '20

09 Nov '20

Hi, The series is inspired by the effort to standardize TI's arm64 dtsi files to keep all nodes in 'okay' state and let the board dts files disable not needed peripherals (and not properly configured): https://lore.kernel.org/lkml/20201104224356.18040-1-nm@ti.com/ In the unlikely (or likely?) event when the dts misses to disable the McASP node which is not configured we currenly and luckily just manage to not crash as we had fixup code in place for legacy pdata boots. This however prints out a message which does not really help to identify the issue. This series will reduce some of the noise during boot (first patch) then adds the needed changes to handle the variations of 'okay': A - have all required DT properties for audio B - gpiochip is enabled A && !B -> everything is OK for audio, no gpiochip registered A && B -> everything is OK for audio, gpiochip is registered !A && B -> audio is not OK, gpiochip is registered. dev_dbg() for audio and do not register SOC DAI and PCM !A && !B -> audio is not OK, no gpiochip. dev_err() and fail the probe Regards, Peter --- Peter Ujfalusi (4): ASoC: ti: davinci-mcasp: Use platform_get_irq_byname_optional ASoC: ti: davinci-mcasp: Remove legacy dma_request parsing ASoC: ti: davinci-mcasp: Simplify the configuration parameter handling ASoC: ti: davinci-mcasp: Handle missing required DT properties sound/soc/ti/davinci-mcasp.c | 294 ++++++++++++++--------------------- 1 file changed, 119 insertions(+), 175 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[PATCH] ASoC: rt1015: modification for calibration to get better performance
by jack.yu＠realtek.com 09 Nov '20

09 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Modification for calibration to get better performance. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 101 ++++++++++++++++++-------------------- 1 file changed, 48 insertions(+), 53 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 52787fbd522f..32d748cc5ae3 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -497,18 +497,40 @@ static void rt1015_calibrate(struct rt1015_priv *rt1015) snd_soc_dapm_mutex_lock(&component->dapm); regcache_cache_bypass(regmap, true); - regmap_write(regmap, RT1015_PWR1, 0xd7df); - regmap_write(regmap, RT1015_PWR4, 0x00b2); - regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2008); + regmap_write(regmap, RT1015_PWR9, 0xAA60); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0089); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008A); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008C); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008D); + regmap_write(regmap, RT1015_PWR4, 0x80B2); + regmap_write(regmap, RT1015_CLASSD_SEQ, 0x5797); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2100); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0100); + regmap_write(regmap, RT1015_PWR5, 0x2175); + regmap_write(regmap, RT1015_MIXER1, 0x005D); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); + regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12F7); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x1205); + msleep(200); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2000); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0180); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x00A1); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x0A05); + msleep(200); + regmap_write(regmap, RT1015_PWR4, 0x00B2); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2028); regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); - regmap_write(regmap, RT1015_GAT_BOOST, 0x0efe); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000d); - regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000e); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a00); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a01); - regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a05); - msleep(500); - regmap_write(regmap, RT1015_PWR1, 0x0); + regmap_write(regmap, RT1015_PWR5, 0x0175); + regmap_write(regmap, RT1015_CLSD_INTERNAL1, 0x1721); + regmap_write(regmap, RT1015_CLASSD_SEQ, 0x570E); + regmap_write(regmap, RT1015_MIXER1, 0x203D); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5A01); + regmap_write(regmap, RT1015_CLSD_INTERNAL2, 0x12FF); + regmap_write(regmap, RT1015_GAT_BOOST, 0x0eFE); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x008E); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x0088); + regmap_write(regmap, RT1015_SYS_RST1, 0x05F5); + regmap_write(regmap, RT1015_SYS_RST2, 0x0b9a); regcache_cache_bypass(regmap, false); regcache_mark_dirty(regmap); @@ -603,6 +625,8 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b0a); snd_soc_component_write(component, RT1015_GAT_BOOST, 0xacfe); snd_soc_component_write(component, @@ -610,10 +634,14 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, snd_soc_component_write(component, RT1015_GAT_BOOST, 0xecfe); } else { + snd_soc_component_write(component, + 0x032d, 0xaa60); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f7); snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x026e); + RT1015_SYS_RST2, 0x0b0a); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x008e); } break; @@ -627,11 +655,17 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, RT1015_PWR9, 0xa800); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f5); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b9a); } else { snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x0268); + 0x032d, 0xaa60); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x0088); snd_soc_component_write(component, RT1015_SYS_RST1, 0x05f5); + snd_soc_component_write(component, + RT1015_SYS_RST2, 0x0b9a); } rt1015->dac_is_used = 0; @@ -664,38 +698,12 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, } static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { - SND_SOC_DAPM_SUPPLY("LDO2", RT1015_PWR1, RT1015_PWR_LDO2_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("INT RC CLK", RT1015_PWR1, RT1015_PWR_INTCLK_BIT, - 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("ISENSE", RT1015_PWR1, RT1015_PWR_ISENSE_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("VSENSE", RT1015_PWR1, RT1015_PWR_VSENSE_BIT, 0, - NULL, 0), SND_SOC_DAPM_SUPPLY("PLL", RT1015_PWR1, RT1015_PWR_PLL_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("BG1 BG2", RT1015_PWR1, RT1015_PWR_BG_1_2_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("MBIAS BG", RT1015_PWR1, RT1015_PWR_MBIAS_BG_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("VBAT", RT1015_PWR1, RT1015_PWR_VBAT_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("MBIAS", RT1015_PWR1, RT1015_PWR_MBIAS_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("ADCV", RT1015_PWR1, RT1015_PWR_ADCV_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("MIXERV", RT1015_PWR1, RT1015_PWR_MIXERV_BIT, 0, - NULL, 0), - SND_SOC_DAPM_SUPPLY("SUMV", RT1015_PWR1, RT1015_PWR_SUMV_BIT, 0, NULL, - 0), - SND_SOC_DAPM_SUPPLY("VREFLV", RT1015_PWR1, RT1015_PWR_VREFLV_BIT, 0, - NULL, 0), - SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC_E("DAC", NULL, RT1015_PWR1, RT1015_PWR_DAC_BIT, 0, + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, rt1015_amp_drv_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_OUTPUT("SPO"), @@ -703,19 +711,7 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { static const struct snd_soc_dapm_route rt1015_dapm_routes[] = { { "DAC", NULL, "AIFRX" }, - { "DAC", NULL, "LDO2" }, { "DAC", NULL, "PLL", rt1015_is_sys_clk_from_pll}, - { "DAC", NULL, "INT RC CLK" }, - { "DAC", NULL, "ISENSE" }, - { "DAC", NULL, "VSENSE" }, - { "DAC", NULL, "BG1 BG2" }, - { "DAC", NULL, "MBIAS BG" }, - { "DAC", NULL, "VBAT" }, - { "DAC", NULL, "MBIAS" }, - { "DAC", NULL, "ADCV" }, - { "DAC", NULL, "MIXERV" }, - { "DAC", NULL, "SUMV" }, - { "DAC", NULL, "VREFLV" }, { "Amp Drv", NULL, "DAC" }, { "SPO", NULL, "Amp Drv" }, }; @@ -1058,7 +1054,6 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; rt1015->cali_done = 0; - snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); -- 2.29.0

2 1

[PATCH] ASoC: fsl_aud2htx: Remove dev_err() usage after platform_get_irq()
by Shengjiu Wang 09 Nov '20

09 Nov '20

platform_get_irq() would print error message internally, so dev_err() after platform_get_irq() is not needed Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_aud2htx.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 124aeb70f24e..4091ccc7c3e9 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -211,11 +211,8 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) } irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "no irq for node %s\n", - dev_name(&pdev->dev)); + if (irq < 0) return irq; - } ret = devm_request_irq(&pdev->dev, irq, fsl_aud2htx_isr, 0, dev_name(&pdev->dev), aud2htx); -- 2.27.0

3 2

[PATCH v1 0/2] Support SC7180 sound card variation without headset
by xuyuqing 09 Nov '20

09 Nov '20

This series is based on patch [v5,2/2] ASoC: qcom: sc7180: Modify machine driver for 2mic Use compatible string to let machine driver handle the case where ALC5682 headset codec is replaced with ADAU7002 dmic codec xuyuqing (2): ASoC: google: dt-bindings: add new compatible for sc7180-coachz ASoC: qcom: sc7180: Modify machine driver for adau7002 .../bindings/sound/google,sc7180-trogdor.yaml | 4 +- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7180.c | 119 ++++++++++++++++-- 3 files changed, 112 insertions(+), 12 deletions(-) -- 2.25.1

2 3

[PATCH] ASoC: Intel: Skylake: Add alternative topology binary name
by Mateusz Gorski 09 Nov '20

09 Nov '20

[ Upstream commit 1b290ef023b3eeb4f4688b582fecb773915ef937 ] Add alternative topology binary file name based on used machine driver and fallback to use this name after failed attempt to load topology file with name based on NHLT. This change addresses multiple issues with current mechanism, for example - there are devices without NHLT table, and that currently results in tplg_name being empty. Signed-off-by: Mateusz Gorski <mateusz.gorski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200427132727.24942-2-mateusz.gorski@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- This functionality is merged on upstream kernel and widely used. Merging it to LTS kernel would improve the user experience and resolve some of the problems regarding topology naming that the users are facing. sound/soc/intel/skylake/skl-topology.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 69cd7a81bf2a..4b114ece58c6 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -14,6 +14,7 @@ #include <linux/uuid.h> #include <sound/intel-nhlt.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-topology.h> #include <uapi/sound/snd_sst_tokens.h> #include <uapi/sound/skl-tplg-interface.h> @@ -3565,8 +3566,20 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_info(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", - skl->tplg_name, ret); + char alt_tplg_name[64]; + + snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", + skl->mach->drv_name); + dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", + skl->tplg_name, ret, alt_tplg_name); + + ret = request_firmware(&fw, alt_tplg_name, bus->dev); + if (!ret) + goto component_load; + + dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", + alt_tplg_name, ret); + ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", @@ -3575,6 +3588,8 @@ int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) } } +component_load: + /* * The complete tplg for SKL is loaded as index 0, we don't use * any other index -- 2.17.1

4 6

[PATCH] ALSA: usb-audio: Fix coding style problems
by Atul 08 Nov '20

08 Nov '20

From: Atul Gopinathan <leoatul12(a)gmail.com> Hi, This patch fixes some coding style warnings and errors that were generated using scripts/checkpatch.pl. Signed-off-by: Atul Gopinathan <leoatul12(a)gmail.com> --- sound/usb/pcm.c | 16 ++++++++-------- sound/usb/quirks.c | 21 +++++++++++---------- 2 files changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a860303cc522..ed5c77ce8e14 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -104,7 +104,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs) continue; } attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; - if (! found) { + if (!found) { found = fp; cur_attr = attr; continue; @@ -304,7 +304,7 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, if (altsd->bAlternateSetting != altsetting || altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC || (altsd->bInterfaceSubClass != 2 && - altsd->bInterfaceProtocol != 2 ) || + altsd->bInterfaceProtocol != 2) || altsd->bNumEndpoints < 1) return -ENOENT; epd = get_endpoint(*alts, 0); @@ -482,8 +482,9 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress - because there is a version of the SB Audigy 2 NX firmware lacking - the audio fields in the endpoint descriptors */ + * because there is a version of the SB Audigy 2 NX firmware lacking + * the audio fields in the endpoint descriptors + */ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0)) { @@ -911,7 +912,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) struct usb_interface *iface; int ret; - if (! subs->cur_audiofmt) { + if (!subs->cur_audiofmt) { dev_err(&subs->dev->dev, "no format is specified!\n"); return -ENXIO; } @@ -973,9 +974,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) return ret; } -static const struct snd_pcm_hardware snd_usb_hardware = -{ - .info = SNDRV_PCM_INFO_MMAP | +static const struct snd_pcm_hardware snd_usb_hardware = { + .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_INTERLEAVED | diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index c989ad8052ae..b6f3345db93f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -398,7 +398,7 @@ static int create_autodetect_quirks(struct snd_usb_audio *chip, } /* - * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. + * Create a stream for an Edirol UA-700/UA-25/UA-4FX interface. * The only way to detect the sample rate is by looking at wMaxPacketSize. */ static int create_uaxx_quirk(struct snd_usb_audio *chip, @@ -593,7 +593,7 @@ static int snd_usb_extigy_boot_quirk(struct usb_device *dev, struct usb_interfac le16_to_cpu(get_cfg_desc(config)->wTotalLength) == EXTIGY_FIRMWARE_SIZE_NEW) { dev_dbg(&dev->dev, "sending Extigy boot sequence...\n"); /* Send message to force it to reconnect with full interface. */ - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev,0), + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x10, 0x43, 0x0001, 0x000a, NULL, 0); if (err < 0) dev_dbg(&dev->dev, "error sending boot message: %d\n", err); @@ -646,8 +646,9 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) "error usb_driver_set_configuration: %d\n", err); /* Always return an error, so that we stop creating a device - that will just be destroyed and recreated with a new - configuration */ + * that will just be destroyed and recreated with a new + * configuration + */ return -ENODEV; } else dev_info(&dev->dev, "Fast Track Pro config OK\n"); @@ -895,9 +896,10 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) usb_reset_device(dev); /* return -EAGAIN, so the creation of an audio interface for this - * temporary device is aborted. The device will reconnect with a - * new product ID */ - return -EAGAIN; + * temporary device is aborted. The device will reconnect with a + * new product ID + */ + return -EAGAIN; } static void mbox2_setup_48_24_magic(struct usb_device *dev) @@ -919,7 +921,6 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev) 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003); - return; } /* Digidesign Mbox 2 needs to load firmware onboard @@ -1830,8 +1831,8 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, /* doesn't set the sample rate attribute, but supports it */ fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; - case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ - case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ + case USB_ID(0x0763, 0x2001): /* M-Audio Quattro USB */ + case USB_ID(0x0763, 0x2012): /* M-Audio Fast Track Pro USB */ case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is an older model 77d:223) */ -- 2.25.1

3 2

ThinkPad support headset button
by Kailang 07 Nov '20

07 Nov '20

Hi Takashi, Two ThinkPad want to support headset button. Attach was support patch. Many Thanks. BR, Kailang

2 1

Re: [Sound-open-firmware] linux-next: Tree for Nov 6 (sound/soc/sof/sof-pci-dev.o)
by Pierre-Louis Bossart 07 Nov '20

07 Nov '20

> on x86_64: > > ld: sound/soc/sof/sof-pci-dev.o: in function `sof_pci_probe': > sof-pci-dev.c:(.text+0x5c): undefined reference to `snd_intel_dsp_driver_probe' > > > Full randconfig file is attached. Nice catch, thanks Randy! Looks like we put the select SND_INTEL_DSP_CONFIG in the wrong place, it's not dependent on the HDaudio link being selected. Clearly a bug we've had for a while. The diff below makes the error go away but I have to run it past folks who are already enjoying their week-end. Will follow-up next week with a proper fix. diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index a066e08860cb..5bfc2f8b13b9 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -271,6 +271,7 @@ config SND_SOC_SOF_JASPERLAKE config SND_SOC_SOF_HDA_COMMON tristate + select SND_INTEL_DSP_CONFIG select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_HDA_LINK_BASELINE help @@ -330,7 +331,6 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level

2 1

[PATCH v3 1/2] ASoC: dt-bindings: fsl_aud2htx: Add binding doc for aud2htx module
by Shengjiu Wang 06 Nov '20

06 Nov '20

AUD2HTX (Audio Subsystem TO HDMI TX Subsystem) is a new IP module found on i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3: - Add additionalProperties changes in v2: - fix indentation issue - remove nodename .../bindings/sound/fsl,aud2htx.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml new file mode 100644 index 000000000000..aa4be7170717 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,aud2htx.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,aud2htx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP Audio Subsystem to HDMI RTX Subsystem Controller + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + const: fsl,imx8mp-aud2htx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: bus + + dmas: + items: + - description: DMA controller phandle and request line for TX + + dma-names: + items: + - const: tx + + power-domains: + maxItems: 1 + +required: + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mp-clock.h> + + aud2htx: aud2htx@30cb0000 { + compatible = "fsl,imx8mp-aud2htx"; + reg = <0x30cb0000 0x10000>; + interrupts = <GIC_SPI 130 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&audiomix_clk IMX8MP_CLK_AUDIOMIX_AUD2HTX_IPG>; + clock-names = "bus"; + dmas = <&sdma2 26 2 0>; + dma-names = "tx"; + power-domains = <&audiomix_pd>; + }; -- 2.27.0

5 8

[PATCH 0/7] PinePhone BT audio bringup
by Samuel Holland 06 Nov '20

06 Nov '20

This series makes use of the additional DAIs recently added to the sun8i-codec driver to add hardware routing for BT SCO (headset) audio on the PinePhone. The BT audio connection is represented by the "dummy" bt-sco codec. The connection to the Quectel EG-25G modem via AIF2 works as well, but I do not include it here because there is no appropriate codec driver in tree. We have been using an out-of-tree "dummy" codec driver similar to bt-sco, and I'm not sure if such a driver would be desired upstream. The modem has a similar amount of configurability as the rtl8723cs BT chip. For the BT chip, the DAI format and PCM parameters are set in a configuration firmware file and loaded at driver load time. For the modem, the DAI format and PCM parameters are set by (and can be queried from) an AT command. However, this AT command requires a modem restart to take effect, so the parameters cannot feasibly be changed at runtime. With a dummy driver, we pick some "standard" set of PCM parameters, e.g. 16 bit mono at 8 or 16 kHz, and assume the hardware agrees. Similarly, we assume the DAI format pre-programmed in the remote hardware matches what is configured in the DAI link (in this case, in the device tree). Is that the right way to model this? Or does the modem (and maybe even the BT chip) deserve a more featureful driver that reads and/or programs the format and params at probe time? Because of those unanswered questions, I'm sending BT SCO support first/only. Regards, Samuel Arnaud Ferraris (1): arm64: dts: allwinner: pinephone: Set audio card name Samuel Holland (6): ASoC: dt-bindings: sun8i-codec: Increase #sound-dai-cells ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Add pinmux nodes for AIF2/AIF3 arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: pinephone: Add support for Bluetooth audio .../sound/allwinner,sun8i-a33-codec.yaml | 2 +- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 25 +++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 37 ++++++++++++++----- 4 files changed, 55 insertions(+), 13 deletions(-) -- 2.26.2

3 9

[PATCH v3] ASoC: rt1015: add delay to fix pop noise from speaker
by jack.yu＠realtek.com 06 Nov '20

06 Nov '20

From: Jack Yu <jack.yu(a)realtek.com> Add delay to fix pop noise from speaker. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1015.txt | 6 ++++++ include/sound/rt1015.h | 15 ++++++++++++++ sound/soc/codecs/rt1015.c | 20 +++++++++++++++++++ sound/soc/codecs/rt1015.h | 2 ++ 4 files changed, 43 insertions(+) create mode 100644 include/sound/rt1015.h diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt index fcfd02d8d32f..e498966d436f 100644 --- a/Documentation/devicetree/bindings/sound/rt1015.txt +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -8,10 +8,16 @@ Required properties: - reg : The I2C address of the device. +Optional properties: + +- realtek,power-up-delay-ms + Set a delay time for flush work to be completed, + this value is adjustable depending on platform. Example: rt1015: codec@28 { compatible = "realtek,rt1015"; reg = <0x28>; + realtek,power-up-delay-ms = <50>; }; diff --git a/include/sound/rt1015.h b/include/sound/rt1015.h new file mode 100644 index 000000000000..70a7538d4c89 --- /dev/null +++ b/include/sound/rt1015.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * linux/sound/rt1015.h -- Platform data for RT1015 + * + * Copyright 2020 Realtek Microelectronics + */ + +#ifndef __LINUX_SND_RT1015_H +#define __LINUX_SND_RT1015_H + +struct rt1015_platform_data { + unsigned int power_up_delay_ms; +}; + +#endif diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..967193518349 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -27,10 +27,15 @@ #include <sound/soc-dapm.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <sound/rt1015.h> #include "rl6231.h" #include "rt1015.h" +static const struct rt1015_platform_data i2s_default_platform_data = { + .power_up_delay_ms = 50, +}; + static const struct reg_default rt1015_reg[] = { { 0x0000, 0x0000 }, { 0x0004, 0xa000 }, @@ -650,6 +655,7 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: if (rt1015->hw_config == RT1015_HW_28) schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); + msleep(rt1015->pdata.power_up_delay_ms); break; default: break; @@ -1067,9 +1073,16 @@ static struct acpi_device_id rt1015_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt1015_acpi_match); #endif +static void rt1015_parse_dt(struct rt1015_priv *rt1015, struct device *dev) +{ + device_property_read_u32(dev, "realtek,power-up-delay-ms", + &rt1015->pdata.power_up_delay_ms); +} + static int rt1015_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { + struct rt1015_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt1015_priv *rt1015; int ret; unsigned int val; @@ -1081,6 +1094,13 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1015); + rt1015->pdata = i2s_default_platform_data; + + if (pdata) + rt1015->pdata = *pdata; + else + rt1015_parse_dt(rt1015, &i2c->dev); + rt1015->regmap = devm_regmap_init_i2c(i2c, &rt1015_regmap); if (IS_ERR(rt1015->regmap)) { ret = PTR_ERR(rt1015->regmap); diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..15cadb361ec3 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -12,6 +12,7 @@ #ifndef __RT1015_H__ #define __RT1015_H__ +#include <sound/rt1015.h> #define RT1015_DEVICE_ID_VAL 0x1011 #define RT1015_DEVICE_ID_VAL2 0x1015 @@ -380,6 +381,7 @@ enum { struct rt1015_priv { struct snd_soc_component *component; + struct rt1015_platform_data pdata; struct regmap *regmap; int sysclk; int sysclk_src; -- 2.28.0

3 2

[PATCH v6 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 06 Nov '20

06 Nov '20

Hi Rob, Mark Cc: Sameer These are v6 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v5 -> v6 - add Acked-by/Reviewed-by on each patch - remove "audio-graph-card,xxx" property - remove un-needed if "ok" v4 -> v5 - move "endpoint" under "port" v3 -> v4 - fixuped yamllint warnings/errors v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 153 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 451 +++++++++++++++ 4 files changed, 604 insertions(+), 857 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

3 5

[PATCH mlx5-next v1 00/11] Convert mlx5 to use auxiliary bus
by Leon Romanovsky 06 Nov '20

06 Nov '20

From: Leon Romanovsky <leonro(a)nvidia.com> Changelog: v1: * Renamed _mlx5_rescan_driver to be mlx5_rescan_driver_locked like in other parts of the mlx5 driver. * Renamed MLX5_INTERFACE_PROTOCOL_VDPA to tbe MLX5_INTERFACE_PROTOCOL_VNET as a preparation to coming series from Eli C. * Some small naming renames in mlx5_vdpa. * Refactored adev index code to make Parav's SF series to apply more easily. * Fixed devlink reload bug that caused to lost TCP connection. v0: https://lore.kernel.org/lkml/20201026111849.1035786-1-leon@kernel.org/ -------------------------------------------------------------- Hi, This patch set converts mlx5 driver to use auxiliary bus [1]. In this series, we are connecting three subsystems (VDPA, netdev and RDMA) through mlx5_core PCI driver. That driver is responsible to create proper devices based on supported firmware. First four patches are preparitions and fixes that were spotted during code development, rest is the conversion itself. Thanks [1] https://lore.kernel.org/lkml/20201023003338.1285642-1-david.m.ertman@intel.… Leon Romanovsky (11): net/mlx5: Don't skip vport check net/mlx5: Properly convey driver version to firmware net/mlx5_core: Clean driver version and name vdpa/mlx5: Make hardware definitions visible to all mlx5 devices net/mlx5: Register mlx5 devices to auxiliary virtual bus vdpa/mlx5: Connect mlx5_vdpa to auxiliary bus net/mlx5e: Connect ethernet part to auxiliary bus RDMA/mlx5: Convert mlx5_ib to use auxiliary bus net/mlx5: Delete custom device management logic net/mlx5: Simplify eswitch mode check RDMA/mlx5: Remove IB representors dead code drivers/infiniband/hw/mlx5/counters.c | 7 - drivers/infiniband/hw/mlx5/ib_rep.c | 113 ++-- drivers/infiniband/hw/mlx5/ib_rep.h | 45 +- drivers/infiniband/hw/mlx5/main.c | 148 +++-- drivers/infiniband/hw/mlx5/mlx5_ib.h | 4 +- .../net/ethernet/mellanox/mlx5/core/Kconfig | 1 + drivers/net/ethernet/mellanox/mlx5/core/dev.c | 567 ++++++++++++------ .../net/ethernet/mellanox/mlx5/core/devlink.c | 4 +- .../ethernet/mellanox/mlx5/core/en_ethtool.c | 4 +- .../net/ethernet/mellanox/mlx5/core/en_main.c | 135 ++--- .../net/ethernet/mellanox/mlx5/core/en_rep.c | 42 +- .../net/ethernet/mellanox/mlx5/core/en_rep.h | 6 +- .../net/ethernet/mellanox/mlx5/core/en_tc.c | 8 +- .../mellanox/mlx5/core/esw/devlink_port.c | 2 +- .../net/ethernet/mellanox/mlx5/core/eswitch.c | 28 +- .../mellanox/mlx5/core/eswitch_offloads.c | 6 + .../mellanox/mlx5/core/ipoib/ethtool.c | 2 +- drivers/net/ethernet/mellanox/mlx5/core/lag.c | 58 +- .../net/ethernet/mellanox/mlx5/core/main.c | 50 +- .../ethernet/mellanox/mlx5/core/mlx5_core.h | 33 +- drivers/vdpa/mlx5/Makefile | 2 +- drivers/vdpa/mlx5/net/main.c | 76 --- drivers/vdpa/mlx5/net/mlx5_vnet.c | 53 +- drivers/vdpa/mlx5/net/mlx5_vnet.h | 24 - include/linux/mlx5/driver.h | 34 +- include/linux/mlx5/eswitch.h | 8 +- .../linux/mlx5/mlx5_ifc_vdpa.h | 6 +- 27 files changed, 818 insertions(+), 648 deletions(-) delete mode 100644 drivers/vdpa/mlx5/net/main.c delete mode 100644 drivers/vdpa/mlx5/net/mlx5_vnet.h rename drivers/vdpa/mlx5/core/mlx5_vdpa_ifc.h => include/linux/mlx5/mlx5_ifc_vdpa.h (97%) -- 2.28.0

8 28

[PATCH v5 0/2] Modify documentation and machine driver for SC7180 sound card
by Ajye Huang 06 Nov '20

06 Nov '20

Note: - The patch is made by the collaboration of Ajye Huang <ajye_huang(a)compal.corp-partner.google.com> Cheng-Yi Chiang <cychiang(a)chromium.org> v5: - Machine driver: - Fix a format string warning (Reported-by: kernel test robot <lkp(a)intel.com>). detailed info at https://lore.kernel.org/patchwork/patch/1331087/ v4: - Machine driver: Addressed suggestions from Tzung-Bi. - Remove redundant judgments in dmic_set() and dmic_get(). - Remove 1 level indent of judgment of IS_ERR(data->dmic_sel). v3: - Machine driver: Addressed suggestions from Tzung-Bi. - move variables "dmic_switch" and "dmic_sel" into struct sc7180_snd_data. - Remove redundant judgments in dmic_set(). v2: - Documentation: Modify the dimc-gpios property description and examples. - Machine driver: - Remove "qcom,sc7180-sndcard-rt5682-m98357-2mic" compatible - See gpio property and use anadditional control. Thanks for the review! Ajye Huang (2): ASoC: google: dt-bindings: modify machine bindings for two MICs case ASoC: qcom: sc7180: Modify machine driver for 2mic .../bindings/sound/google,sc7180-trogdor.yaml | 58 ++++++++++++++++++ sound/soc/qcom/sc7180.c | 61 +++++++++++++++++++ 2 files changed, 119 insertions(+) -- 2.25.1

3 5

[PATCH AUTOSEL 5.9 039/147] ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work
by Sasha Levin 05 Nov '20

05 Nov '20

From: Rander Wang <rander.wang(a)intel.com> [ Upstream commit 6c63c954e1c52f1262f986f36d95f557c6f8fa94 ] When hda_codec_probe() doesn't initialize audio component, we disable the codec and keep going. However,the resources are not released. The child_count of SOF device is increased in snd_hdac_ext_bus_device_init but is not decrease in error case, so SOF can't get suspended. snd_hdac_ext_bus_device_exit will be invoked in HDA framework if it gets a error. Now copy this behavior to release resources and decrease SOF device child_count to release SOF device. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20200825235040.1586478-3-ranjani.sridharan@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 2c5c451fa19d7..c475955c6eeba 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -151,7 +151,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev->bus->audio_component) { dev_dbg(sdev->dev, "iDisp hw present but no driver\n"); - return -ENOENT; + goto error; } hda_priv->need_display_power = true; } @@ -174,7 +174,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, * other return codes without modification */ if (ret == 0) - ret = -ENOENT; + goto error; } return ret; -- 2.25.1

3 3

[GIT PULL] ASoC fixes for v5.10-rc2
by Mark Brown 05 Nov '20

05 Nov '20

The following changes since commit 3650b228f83adda7e5ee532e2b90429c03f7b9ec: Linux 5.10-rc1 (2020-10-25 15:14:11 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.10-rc2 for you to fetch changes up to f9d7c6eb23f7e55e7a0ca5451da06909bdfdd0e4: ASoC: mchp-spdiftx: Do not set Validity bit(s) (2020-11-04 17:50:22 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.10 A batch of driver specific fixes that have come up since the merge window, nothing particularly major here but all good to have. ---------------------------------------------------------------- Bard Liao (1): ASoC: SOF: loader: handle all SOF_IPC_EXT types Cezary Rojewski (3): ASoC: Intel: atom: Remove duplicate kconfigs ASoC: Intel: catpt: Wake up device before configuring SSP port ASoC: Intel: catpt: Relax clock selection conditions Codrin Ciubotariu (1): ASoC: mchp-spdiftx: Do not set Validity bit(s) Mark Brown (1): Merge tag 'v5.10-rc1' into asoc-5.10 Olivier Moysan (1): ASoC: cs42l51: manage mclk shutdown delay Srinivas Kandagatla (4): ASoC: codecs: wsa881x: add missing stream rates and format ASoC: qcom: sdm845: set driver name correctly ASoC: codecs: wcd934x: Set digital gain range correctly ASoC: codecs: wcd9335: Set digital gain range correctly Tomasz Figa (1): ASoC: Intel: kbl_rt5663_max98927: Fix kabylake_ssp_fixup function Tzung-Bi Shih (1): ASoC: mediatek: mt8183-da7219: fix DAPM paths for rt1015 V Sujith Kumar Reddy (2): ASoC: qcom: lpass-sc7180: Fix MI2S bitwidth field bit positions ASoC: qcom: lpass-cpu: Fix clock disable failure sound/soc/atmel/mchp-spdiftx.c | 1 - sound/soc/codecs/cs42l51.c | 22 +++++++++++- sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd934x.c | 2 +- sound/soc/codecs/wsa881x.c | 2 ++ sound/soc/intel/Kconfig | 18 ---------- sound/soc/intel/atom/Makefile | 2 +- sound/soc/intel/atom/sst/Makefile | 6 ++-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 39 +++++++++++++++++----- sound/soc/intel/catpt/dsp.c | 9 +++-- sound/soc/intel/catpt/pcm.c | 10 ++++++ sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 31 +++++++++++++---- sound/soc/qcom/lpass-cpu.c | 14 +++++--- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 2 ++ sound/soc/sof/loader.c | 5 +++ 16 files changed, 119 insertions(+), 48 deletions(-)

2 1

[PATCH] ASoC: q6afe-clocks: fix warning on symbol scope
by Srinivas Kandagatla 05 Nov '20

05 Nov '20

This patch fixes below warning when module is compiled with W=1 C=1 sound/soc/qcom/qdsp6/q6afe-clocks.c:122:18: warning: symbol 'q6afe_clks' was not declared. Should it be static? Fixes: 520a1c396d196 ("ASoC: q6afe-clocks: add q6afe clock controller") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-clocks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-clocks.c b/sound/soc/qcom/qdsp6/q6afe-clocks.c index 2efc2eaa0424..87e4633afe2c 100644 --- a/sound/soc/qcom/qdsp6/q6afe-clocks.c +++ b/sound/soc/qcom/qdsp6/q6afe-clocks.c @@ -119,7 +119,7 @@ static const struct clk_ops clk_vote_q6afe_ops = { .unprepare = clk_unvote_q6afe_block, }; -struct q6afe_clk *q6afe_clks[Q6AFE_MAX_CLK_ID] = { +static struct q6afe_clk *q6afe_clks[Q6AFE_MAX_CLK_ID] = { [LPASS_CLK_ID_PRI_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_IBIT), [LPASS_CLK_ID_PRI_MI2S_EBIT] = Q6AFE_CLK(LPASS_CLK_ID_PRI_MI2S_EBIT), [LPASS_CLK_ID_SEC_MI2S_IBIT] = Q6AFE_CLK(LPASS_CLK_ID_SEC_MI2S_IBIT), -- 2.21.0

2 1

[PATCH] ASoc: qcom: lpass-cpu: fix warning on symbol scope
by Srinivas Kandagatla 05 Nov '20

05 Nov '20

This patch fixes below warning when module is compiled with W=1 C=1 lpass-cpu.c:677:22: warning: symbol 'lpass_hdmi_regmap_config' was not declared. Should it be static? Fixes: 7cb37b7bd0d3 ("ASoC: qcom: Add support for lpass hdmi driver") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 9d17c87445a9..1fd862e3fd8a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -674,7 +674,7 @@ static bool lpass_hdmi_regmap_volatile(struct device *dev, unsigned int reg) return false; } -struct regmap_config lpass_hdmi_regmap_config = { +static struct regmap_config lpass_hdmi_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, -- 2.21.0

2 1

[PATCH] ASoC: samsung: i2s: Remove redundant null check before clk_disable_unprepare
by Xu Wang 05 Nov '20

05 Nov '20

Because clk_disable_unprepare() already checked NULL clock parameter, so the additional check is unnecessary, just remove it. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/samsung/i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index df53d4ea808f..36969f0a3f9a 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1212,8 +1212,7 @@ static int i2s_runtime_suspend(struct device *dev) priv->suspend_i2scon = readl(priv->addr + I2SCON); priv->suspend_i2spsr = readl(priv->addr + I2SPSR); - if (priv->op_clk) - clk_disable_unprepare(priv->op_clk); + clk_disable_unprepare(priv->op_clk); clk_disable_unprepare(priv->clk); return 0; -- 2.17.1

2 1

[PATCH] ASoC: pcm: DRAIN support reactivation
by Cezary Rojewski 05 Nov '20

05 Nov '20

soc-pcm's dpcm_fe_dai_do_trigger() supported DRAIN commnad up to kernel v5.4 where explicit switch(cmd) has been introduced which takes into account all SNDRV_PCM_TRIGGER_xxx but SNDRV_PCM_TRIGGER_DRAIN. Update switch statement to reactive support for it. As DRAIN is somewhat unique by lacking negative/stop counterpart, bring behaviour of dpcm_fe_dai_do_trigger() for said command back to its pre-v5.4 state by adding it to START/RESUME/PAUSE_RELEASE group. Fixes: acbf27746ecf ("ASoC: pcm: update FE/BE trigger order based on the command") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/soc-pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index dcab9527ba3d..91bf33958159 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2231,6 +2231,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, true); break; case SNDRV_PCM_TRIGGER_STOP: @@ -2248,6 +2249,7 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_DRAIN: ret = dpcm_dai_trigger_fe_be(substream, cmd, false); break; case SNDRV_PCM_TRIGGER_STOP: -- 2.17.1

3 2

[PATCH -next v2] ASoC: mediatek: mt8192: Make some symbols static
by Zou Wei 05 Nov '20

05 Nov '20

Fix the following sparse warnings: ./mt8192-dai-i2s.c:2040:5: warning: symbol 'mt8192_dai_i2s_get_share' was not declared. Should it be static? ./mt8192-dai-i2s.c:2060:5: warning: symbol 'mt8192_dai_i2s_set_priv' was not declared. Should it be static? ./mt8192-afe-gpio.c:15:16: warning: symbol 'aud_pinctrl' was not declared. Should it be static? ./mt8192-afe-pcm.c:70:5: warning: symbol 'mt8192_get_memif_pbuf_size' was not declared. Should it be static? ./mt8192-afe-pcm.c:2137:39: warning: symbol 'mt8192_afe_component' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 2 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 4 ++-- sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c index ea00088..4208820 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c @@ -12,7 +12,7 @@ #include "mt8192-afe-common.h" #include "mt8192-afe-gpio.h" -struct pinctrl *aud_pinctrl; +static struct pinctrl *aud_pinctrl; enum mt8192_afe_gpio { MT8192_AFE_GPIO_DAT_MISO_OFF, diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index 4a4729f..e7fec2d 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -67,7 +67,7 @@ static int mt8192_irq_fs(struct snd_pcm_substream *substream, unsigned int rate) return mt8192_general_rate_transform(afe->dev, rate); } -int mt8192_get_memif_pbuf_size(struct snd_pcm_substream *substream) +static int mt8192_get_memif_pbuf_size(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -2134,7 +2134,7 @@ static int mt8192_afe_component_probe(struct snd_soc_component *component) return mtk_afe_add_sub_dai_control(component); } -const struct snd_soc_component_driver mt8192_afe_component = { +static const struct snd_soc_component_driver mt8192_afe_component = { .name = AFE_PCM_NAME, .probe = mt8192_afe_component_probe, .pointer = mtk_afe_pcm_pointer, diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c index 53c560e..5b29340 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c @@ -2037,7 +2037,7 @@ static const struct mtk_afe_i2s_priv mt8192_i2s_priv[DAI_I2S_NUM] = { }, }; -int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) { struct mt8192_afe_private *afe_priv = afe->platform_priv; const struct device_node *of_node = afe->dev->of_node; @@ -2057,7 +2057,7 @@ int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) return 0; } -int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) { int i; int ret; -- 2.6.2

3 2

[PATCH] ASoC: mediatek: mt8192: Fix build failure
by Sudip Mukherjee 05 Nov '20

05 Nov '20

A build of arm64 allmodconfig with next-20201105 fails with the error: ERROR: modpost: "mt8192_afe_gpio_request" undefined! ERROR: modpost: "mt8192_afe_gpio_init" undefined! Export the symbols so that mt8192-mt6359-rt1015-rt5682.ko finds it. Signed-off-by: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> --- build log at: https://travis-ci.com/github/sudipm-mukherjee/linux-test/jobs/428486008 sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c index ea000888c9e8..fbbe9ed9adb3 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c @@ -160,6 +160,7 @@ int mt8192_afe_gpio_init(struct device *dev) return 0; } +EXPORT_SYMBOL(mt8192_afe_gpio_init); static int mt8192_afe_gpio_adda_dl(struct device *dev, bool enable) { @@ -304,3 +305,4 @@ int mt8192_afe_gpio_request(struct device *dev, bool enable, return 0; } +EXPORT_SYMBOL(mt8192_afe_gpio_request); -- 2.11.0

2 1

[PATCH] ASoC: pxa: pxa-ssp: Remove redundant null check before clk_prepare_enable/clk_disable_unprepare
by Xu Wang 05 Nov '20

05 Nov '20

ecause clk_prepare_enable() and clk_disable_unprepare() already checked NULL clock parameter, so the additional checks are unnecessary, just remove them. Signed-off-by: Xu Wang <vulab(a)iscas.ac.cn> --- sound/soc/pxa/pxa-ssp.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index c4e7307a4437..b941adcbb8f9 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -99,8 +99,7 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, pxa_ssp_disable(ssp); } - if (priv->extclk) - clk_prepare_enable(priv->extclk); + clk_prepare_enable(priv->extclk); dma = kzalloc(sizeof(struct snd_dmaengine_dai_dma_data), GFP_KERNEL); if (!dma) @@ -124,8 +123,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(ssp->clk); } - if (priv->extclk) - clk_disable_unprepare(priv->extclk); + clk_disable_unprepare(priv->extclk); kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); -- 2.17.1

2 1

[PATCH -next] ASoC: mediatek: mt8192: Make some symbols static
by Zou Wei 05 Nov '20

05 Nov '20

Fix the following sparse warnings: ./mt8192-dai-i2s.c:2040:5: warning: symbol 'mt8192_dai_i2s_get_share' was not declared. Should it be static? ./mt8192-dai-i2s.c:2060:5: warning: symbol 'mt8192_dai_i2s_set_priv' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c index 53c560e..5b29340 100644 --- a/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c +++ b/sound/soc/mediatek/mt8192/mt8192-dai-i2s.c @@ -2037,7 +2037,7 @@ static const struct mtk_afe_i2s_priv mt8192_i2s_priv[DAI_I2S_NUM] = { }, }; -int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) { struct mt8192_afe_private *afe_priv = afe->platform_priv; const struct device_node *of_node = afe->dev->of_node; @@ -2057,7 +2057,7 @@ int mt8192_dai_i2s_get_share(struct mtk_base_afe *afe) return 0; } -int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) +static int mt8192_dai_i2s_set_priv(struct mtk_base_afe *afe) { int i; int ret; -- 2.6.2

1 0

[PATCH] sound: soc: mediatek: mt8192: fix modpost ERROR
by Anders Roxell 05 Nov '20

05 Nov '20

When enabling CONFIG_SND_SOC_MT8192_MT6359_RT1015_RT5682=m the following error shows up: ERROR: modpost: "mt8192_afe_gpio_request" [sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.ko] undefined! ERROR: modpost: "mt8192_afe_gpio_init" [sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.ko] undefined! Fix the link error by export the symbols mt8192_afe_gpio_init and mt8192_afe_gpio_request. Fixes: 18b13ff23fab ("ASoC: mediatek: mt8192: add machine driver with mt6359, rt1015 and rt5682") Signed-off-by: Anders Roxell <anders.roxell(a)linaro.org> --- sound/soc/mediatek/mt8192/mt8192-afe-gpio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c index ea000888c9e8..0286c95ed2b8 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-gpio.c @@ -160,6 +160,7 @@ int mt8192_afe_gpio_init(struct device *dev) return 0; } +EXPORT_SYMBOL_GPL(mt8192_afe_gpio_init); static int mt8192_afe_gpio_adda_dl(struct device *dev, bool enable) { @@ -304,3 +305,4 @@ int mt8192_afe_gpio_request(struct device *dev, bool enable, return 0; } +EXPORT_SYMBOL_GPL(mt8192_afe_gpio_request); -- 2.28.0

1 0

[PATCH 0/8] slab: provide and use krealloc_array()
by Bartosz Golaszewski 05 Nov '20

05 Nov '20

From: Bartosz Golaszewski <bgolaszewski(a)baylibre.com> Andy brought to my attention the fact that users allocating an array of equally sized elements should check if the size multiplication doesn't overflow. This is why we have helpers like kmalloc_array(). However we don't have krealloc_array() equivalent and there are many users who do their own multiplication when calling krealloc() for arrays. This series provides krealloc_array() and uses it in a couple places. A separate series will follow adding devm_krealloc_array() which is needed in the xilinx adc driver. Bartosz Golaszewski (8): mm: slab: provide krealloc_array() ALSA: pcm: use krealloc_array() vhost: vringh: use krealloc_array() pinctrl: use krealloc_array() edac: ghes: use krealloc_array() drm: atomic: use krealloc_array() hwtracing: intel: use krealloc_array() dma-buf: use krealloc_array() drivers/dma-buf/sync_file.c | 4 ++-- drivers/edac/ghes_edac.c | 4 ++-- drivers/gpu/drm/drm_atomic.c | 3 ++- drivers/hwtracing/intel_th/msu.c | 2 +- drivers/pinctrl/pinctrl-utils.c | 2 +- drivers/vhost/vringh.c | 3 ++- include/linux/slab.h | 11 +++++++++++ sound/core/pcm_lib.c | 4 ++-- 8 files changed, 23 insertions(+), 10 deletions(-) -- 2.29.1

11 20

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for Qu-16
by Geoffrey D. Bennett 05 Nov '20

05 Nov '20

This patch fixes audio distortion on playback for the Allen&Heath Qu-16. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee8..c4915de 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -336,6 +336,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ + case USB_ID(0x22f0, 0x0006): /* Allen&Heath Qu-16 */ ep = 0x81; ifnum = 3; goto add_sync_ep_from_ifnum;

2 1

[PATCH] ALSA: usb-audio: Add implicit feedback quirk for MODX
by Geoffrey D. Bennett 05 Nov '20

05 Nov '20

This patch fixes audio distortion on playback for the Yamaha MODX. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> Tested-by: Frank Slotta <frank.slotta(a)posteo.de> --- pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b401ee894e1b..22fbd1f48b3e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -345,6 +345,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2466, 0x8003): /* Fractal Audio Axe-Fx II */ + case USB_ID(0x0499, 0x172a): /* Yamaha MODX */ ep = 0x86; ifnum = 2; goto add_sync_ep_from_ifnum;

2 3

[PATCH v2 0/6] ASoC: codecs: add support for LPASS Codec macros
by Srinivas Kandagatla 04 Nov '20

04 Nov '20

This patchset adds support for two Codec Macro blocks( WSA and VA) available in Qualcomm LPASS (Low Power Audio SubSystem). There are WSA, VA, TX and RX Macros on LPASS IP, each of the Macro block has specific connectivity like WSA Macros are intended to connect to WSA Smart speaker codecs via SoundWire. VA Macro is intended for DMICs, and TX/RX for Analog codecs via SoundWire like other WCD Codecs to provide headphone/ear/lineout etc .. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset has been tested on support to Qualcomm Robotics RB5 Development Kit based on QRB5165 Robotics SoC. This board has 2 WSA881X smart speakers with onboard DMIC connected to internal LPASS codec via WSA and VA macros respectively. Thanks, srini -Changes since v1: - updated code to fix cppcheck errors suggested by Pierre - fixed various coding style and variable initialization suggested by Pierre. - yaml bindings clock names replaced with constants. Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add bindings for lpass wsa macro codec ASoC: codecs: lpass-wsa-macro: Add support to WSA Macro ASoC: codecs: lpass-wsa-macro: add dapm widgets and route ASoC: qcom: dt-bindings: add bindings for lpass va macro codec ASoC: codecs: lpass-va-macro: Add support to VA Macro ASoC: codecs: lpass-va-macro: add dapm widgets and routes .../bindings/sound/qcom,lpass-va-macro.yaml | 67 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 69 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/lpass-va-macro.c | 1500 ++++++++++ sound/soc/codecs/lpass-wsa-macro.c | 2472 +++++++++++++++++ sound/soc/codecs/lpass-wsa-macro.h | 15 + 7 files changed, 4135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-va-macro.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml create mode 100644 sound/soc/codecs/lpass-va-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.c create mode 100644 sound/soc/codecs/lpass-wsa-macro.h -- 2.21.0

3 15

[PATCH v5 0/3] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 04 Nov '20

04 Nov '20

Hi Rob, Mark Cc: Sameer These are v5 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v4 -> v5 - move "endpoint" under "port" v3 -> v4 - fixuped yamllint warnings/errors v2 -> v3 - fixuped Rob's pointed points v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (3): dt-bindings: ASoC: audio-graph-card: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: switch to yaml base Documentation dt-bindings: ASoC: renesas,rsnd: Add r8a77961 support .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 162 ++++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 457 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 620 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

3 7

[PATCH v3] ASoC: rt1015: support TDM slot configuration
by shumingf＠realtek.com 04 Nov '20

04 Nov '20

From: Shuming Fan <shumingf(a)realtek.com> Add TDM slot callback function to support TDM configuration Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1015.c | 101 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1015.h | 42 ++++++++++++++++ 2 files changed, 143 insertions(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 25fe2ddedd54..a9cd6ad0bf5a 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -944,6 +944,106 @@ static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) return 0; } +static int rt1015_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit; + + switch (slots) { + case 2: + val |= RT1015_I2S_TX_2CH; + break; + case 4: + val |= RT1015_I2S_TX_4CH; + break; + case 6: + val |= RT1015_I2S_TX_6CH; + break; + case 8: + val |= RT1015_I2S_TX_8CH; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + switch (slot_width) { + case 16: + val |= RT1015_I2S_CH_TX_LEN_16B; + break; + case 20: + val |= RT1015_I2S_CH_TX_LEN_20B; + break; + case 24: + val |= RT1015_I2S_CH_TX_LEN_24B; + break; + case 32: + val |= RT1015_I2S_CH_TX_LEN_32B; + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + if (rx_slotnum != 1) { + ret = -EINVAL; + dev_err(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + /* This is an assumption that the system sends stereo audio to the amplifier typically. + * And the stereo audio is placed in slot 0/2/4/6 as the starting slot. + * The users could select the channel from L/R/L+R by "Mono LR Select" control. + */ + first_bit = __ffs(rx_mask); + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1015_TDM1_4, + RT1015_TDM_I2S_TX_L_DAC1_1_MASK | + RT1015_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1015_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1015_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + if (tx_slotnum) { + ret = -EINVAL; + dev_err(component->dev, "doesn't need to support tx slots\n"); + goto _set_tdm_err_; + } + + snd_soc_component_update_bits(component, RT1015_TDM1_1, + RT1015_I2S_CH_TX_MASK | RT1015_I2S_CH_RX_MASK | + RT1015_I2S_CH_TX_LEN_MASK | RT1015_I2S_CH_RX_LEN_MASK, val); + +_set_tdm_err_: + return ret; +} + static int rt1015_probe(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = @@ -975,6 +1075,7 @@ static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, .set_bclk_ratio = rt1015_set_bclk_ratio, + .set_tdm_slot = rt1015_set_tdm_slot, }; static struct snd_soc_dai_driver rt1015_dai[] = { diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index d3fdd30aca6d..ad8274c80990 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -213,6 +213,12 @@ #define RT1015_ID_VERA 0x0 #define RT1015_ID_VERB 0x1 +/* 0x00f2 */ +#define RT1015_MONO_LR_SEL_MASK (0x3 << 4) +#define RT1015_MONO_L_CHANNEL (0x0 << 4) +#define RT1015_MONO_R_CHANNEL (0x1 << 4) +#define RT1015_MONO_LR_MIX_CHANNEL (0x2 << 4) + /* 0x0102 */ #define RT1015_DAC_VOL_MASK (0x7f << 9) #define RT1015_DAC_VOL_SFT 9 @@ -275,6 +281,42 @@ #define RT1015_TDM_INV_BCLK_MASK (0x1 << 15) #define RT1015_TDM_INV_BCLK_SFT 15 #define RT1015_TDM_INV_BCLK (0x1 << 15) +#define RT1015_I2S_CH_TX_MASK (0x3 << 10) +#define RT1015_I2S_CH_TX_SFT 10 +#define RT1015_I2S_TX_2CH (0x0 << 10) +#define RT1015_I2S_TX_4CH (0x1 << 10) +#define RT1015_I2S_TX_6CH (0x2 << 10) +#define RT1015_I2S_TX_8CH (0x3 << 10) +#define RT1015_I2S_CH_RX_MASK (0x3 << 8) +#define RT1015_I2S_CH_RX_SFT 8 +#define RT1015_I2S_RX_2CH (0x0 << 8) +#define RT1015_I2S_RX_4CH (0x1 << 8) +#define RT1015_I2S_RX_6CH (0x2 << 8) +#define RT1015_I2S_RX_8CH (0x3 << 8) +#define RT1015_I2S_LR_CH_SEL_MASK (0x1 << 7) +#define RT1015_I2S_LR_CH_SEL_SFT 7 +#define RT1015_I2S_LEFT_CH_SEL (0x0 << 7) +#define RT1015_I2S_RIGHT_CH_SEL (0x1 << 7) +#define RT1015_I2S_CH_TX_LEN_MASK (0x7 << 4) +#define RT1015_I2S_CH_TX_LEN_SFT 4 +#define RT1015_I2S_CH_TX_LEN_16B (0x0 << 4) +#define RT1015_I2S_CH_TX_LEN_20B (0x1 << 4) +#define RT1015_I2S_CH_TX_LEN_24B (0x2 << 4) +#define RT1015_I2S_CH_TX_LEN_32B (0x3 << 4) +#define RT1015_I2S_CH_TX_LEN_8B (0x4 << 4) +#define RT1015_I2S_CH_RX_LEN_MASK (0x7 << 0) +#define RT1015_I2S_CH_RX_LEN_SFT 0 +#define RT1015_I2S_CH_RX_LEN_16B (0x0 << 0) +#define RT1015_I2S_CH_RX_LEN_20B (0x1 << 0) +#define RT1015_I2S_CH_RX_LEN_24B (0x2 << 0) +#define RT1015_I2S_CH_RX_LEN_32B (0x3 << 0) +#define RT1015_I2S_CH_RX_LEN_8B (0x4 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1015_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1015_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1015_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1015_TDM_I2S_TX_R_DAC1_1_SFT 8 /* 0x0330 */ #define RT1015_ABST_AUTO_EN_MASK (0x1 << 13) -- 2.28.0

3 2

[PATCH v3 0/7] ASoC: topology: Change to resource managed memory
by Amadeusz Sławiński 04 Nov '20

04 Nov '20

Almost all other allocations in ASoC API are resource managed, the only exception is soc-topology.c. This patchset clean ups few unnecessary functions in preparation for change and then changes to devm_ functions for allocation. Amadeusz Sławiński (6): ASoC: topology: Remove unused functions from topology API ASoC: topology: Remove multistep topology loading ASoC: topology: Unify all device references ASoC: topology: Change allocations to resource managed ASoC: topology: Remove empty functions ASoC: topology: Simplify remove_widget function include/sound/soc-topology.h | 13 +- sound/soc/intel/skylake/skl-topology.c | 9 +- sound/soc/soc-topology.c | 302 +++++-------------------- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/topology.c | 4 +- 5 files changed, 57 insertions(+), 273 deletions(-) -- 2.25.1

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