March 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC:hdac_hda:use correct format to setup hda codec" to the asoc tree
by Mark Brown 11 Mar '19

11 Mar '19

The patch ASoC:hdac_hda:use correct format to setup hda codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 03d0aa4d4fddce4a5d865d819a4d98bfc3d451e6 Mon Sep 17 00:00:00 2001 From: Rander Wang <rander.wang(a)linux.intel.com> Date: Fri, 8 Mar 2019 16:38:58 +0800 Subject: [PATCH] ASoC:hdac_hda:use correct format to setup hda codec The current implementation of the hdac_hda codec results in zero-valued samples on capture and noise with headset playback when SOF is used on platforms with an on-board HDaudio codec. This is root-caused to SOF using be_hw_params_fixup, and the prepare() call using invalid runtime fields to determine the format. This patch moves the format handling to the hw_params() callback, as done already for hdac_hdmi, to make sure the fixed-up information is taken into account but keeps the codec initialization in prepare() as the stream_tag is only available at that time. Moving everything in the prepare() callback is possible but the code is less elegant so this two-step solution was chosen. The solution was tested with the SST driver with no regressions, and all the issues with SOF playback and capture are solved. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/hdac_hda.c | 53 +++++++++++++++++++++++++++---------- sound/soc/codecs/hdac_hda.h | 1 + 2 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index ffecdaaa8cf2..f889d94c8e3c 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -38,6 +38,9 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -50,6 +53,7 @@ static const struct snd_soc_dai_ops hdac_hda_dai_ops = { .startup = hdac_hda_dai_open, .shutdown = hdac_hda_dai_close, .prepare = hdac_hda_dai_prepare, + .hw_params = hdac_hda_dai_hw_params, .hw_free = hdac_hda_dai_hw_free, .set_tdm_slot = hdac_hda_dai_set_tdm_slot, }; @@ -139,6 +143,39 @@ static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int hdac_hda_dai_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 hdac_hda_priv *hda_pvt; + unsigned int format_val; + unsigned int maxbps; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + maxbps = dai->driver->playback.sig_bits; + else + maxbps = dai->driver->capture.sig_bits; + + hda_pvt = snd_soc_component_get_drvdata(component); + format_val = snd_hdac_calc_stream_format(params_rate(params), + params_channels(params), + params_format(params), + maxbps, + 0); + if (!format_val) { + dev_err(dai->dev, + "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + params_rate(params), params_channels(params), + params_format(params), maxbps); + + return -EINVAL; + } + + hda_pvt->pcm[dai->id].format_val[substream->stream] = format_val; + return 0; +} + static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,10 +199,9 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct snd_pcm_runtime *runtime = substream->runtime; struct hdac_device *hdev; - struct hda_pcm_stream *hda_stream; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; @@ -179,19 +215,8 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - format_val = snd_hdac_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hda_stream->maxbps, - 0); - if (!format_val) { - dev_err(&hdev->dev, - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); - return -EINVAL; - } - stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + format_val = hda_pvt->pcm[dai->id].format_val[substream->stream]; ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, stream, format_val, substream); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index e444ef593360..6b1bd4f428e7 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -8,6 +8,7 @@ struct hdac_hda_pcm { int stream_tag[2]; + unsigned int format_val[2]; }; struct hdac_hda_priv { -- 2.20.1

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[alsa-devel] [PATCH v3] ASoC: dpcm: prevent snd_soc_dpcm use after free
by KaiChieh Chuang 11 Mar '19

11 Mar '19

The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Use irq spin lock version, since the lock may be used in interrupts. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 1 + sound/soc/soc-pcm.c | 40 +++++++++++++++++++++++++++++++++------- 3 files changed, 36 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index eb7db60..1e2be35 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1083,6 +1083,8 @@ struct snd_soc_card { struct mutex mutex; struct mutex dapm_mutex; + spinlock_t dpcm_lock; + bool instantiated; bool topology_shortname_created; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 5a5764d..d887576 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2820,6 +2820,7 @@ int snd_soc_register_card(struct snd_soc_card *card) card->instantiated = 0; mutex_init(&card->mutex); mutex_init(&card->dapm_mutex); + spin_lock_init(&card->dpcm_lock); return snd_soc_bind_card(card); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a5b40e8..30df83a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1213,6 +1213,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { struct snd_soc_dpcm *dpcm; + unsigned long flags; /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1228,8 +1229,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->fe = fe; be->dpcm[stream].runtime = fe->dpcm[stream].runtime; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, @@ -1275,6 +1278,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; + unsigned long flags; for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", @@ -1294,8 +1298,10 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) #ifdef CONFIG_DEBUG_FS debugfs_remove(dpcm->debugfs_state); #endif + 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); } } @@ -1547,10 +1553,13 @@ int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; + unsigned long flags; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) dpcm->be->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); } static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, @@ -2571,6 +2580,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_soc_dpcm *dpcm; enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret; + unsigned long flags; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); @@ -2640,11 +2650,13 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) dpcm_be_dai_shutdown(fe, stream); disconnect: /* disconnect any non started BEs */ + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); return ret; } @@ -3220,7 +3232,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + unsigned long flags; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3229,12 +3244,15 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, state = dpcm->fe->dpcm[stream].state; if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) - return 0; + state == SND_SOC_DPCM_STATE_SUSPEND) { + ret = 0; + break; + } } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to free/stop this BE DAI */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); @@ -3247,7 +3265,10 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; int state; + int ret = 1; + unsigned long flags; + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { if (dpcm->fe == fe) @@ -3257,12 +3278,15 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, if (state == SND_SOC_DPCM_STATE_START || state == SND_SOC_DPCM_STATE_PAUSED || state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) - return 0; + state == SND_SOC_DPCM_STATE_PREPARE) { + ret = 0; + break; + } } + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); /* it's safe to change hw_params */ - return 1; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); @@ -3301,6 +3325,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_hw_params *params = &fe->dpcm[stream].hw_params; struct snd_soc_dpcm *dpcm; ssize_t offset = 0; + unsigned long flags; /* FE state */ offset += snprintf(buf + offset, size - offset, @@ -3328,6 +3353,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, goto out; } + spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_be(fe, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; params = &dpcm->hw_params; @@ -3348,7 +3374,7 @@ static ssize_t dpcm_show_state(struct snd_soc_pcm_runtime *fe, params_channels(params), params_rate(params)); } - + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); out: return offset; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8824: fix the issue of the widget with prefix name
by John Hsu 11 Mar '19

11 Mar '19

The driver has two issues when machine add prefix name for codec. (1)The stream name of DAI can't find the AIF widgets. (2)The drivr can enable/disalbe the MICBIAS and SAR widgets. The patch will fix these issues caused by prefixed name added. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 46 +++++++++++++++++++++++++++++++------- 1 file changed, 38 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 87ed3dc496dc..5ab05e75edea 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -681,8 +681,8 @@ static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADCR", NULL, NAU8824_REG_ANALOG_ADC_2, NAU8824_ADCR_EN_SFT, 0), - SND_SOC_DAPM_AIF_OUT("AIFTX", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_IN("AIFRX", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DACL", NULL, NAU8824_REG_RDAC, NAU8824_DACL_EN_SFT, 0), @@ -831,6 +831,36 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } +static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) +{ + struct snd_soc_dapm_context *dapm = nau8824->dapm; + const char *prefix = dapm->component->name_prefix; + char prefixed_pin[80]; + + if (prefix) { + snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", + prefix, pin); + snd_soc_dapm_disable_pin(dapm, prefixed_pin); + } else { + snd_soc_dapm_disable_pin(dapm, pin); + } +} + +static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) +{ + struct snd_soc_dapm_context *dapm = nau8824->dapm; + const char *prefix = dapm->component->name_prefix; + char prefixed_pin[80]; + + if (prefix) { + snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", + prefix, pin); + snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); + } else { + snd_soc_dapm_force_enable_pin(dapm, pin); + } +} + static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -839,8 +869,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -870,8 +900,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_force_enable_pin(dapm, "SAR"); + nau8824_dapm_enable_pin(nau8824, "MICBIAS"); + nau8824_dapm_enable_pin(nau8824, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -882,8 +912,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - snd_soc_dapm_disable_pin(dapm, "SAR"); - snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + nau8824_dapm_disable_pin(nau8824, "SAR"); + nau8824_dapm_disable_pin(nau8824, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; -- 2.18.0

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[alsa-devel] [PATCH] ASoC: stm32: i2s: fix registers declaration in regmap
by Olivier Moysan 11 Mar '19

11 Mar '19

- Declare SR as volatile, as it is changed by hardware. - Remove TXDR from readable and volatile register list, as it is intended for write accesses only. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index c18e068c1a0d..952634258b02 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -278,7 +278,6 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg) case STM32_I2S_CFG2_REG: case STM32_I2S_IER_REG: case STM32_I2S_SR_REG: - case STM32_I2S_TXDR_REG: case STM32_I2S_RXDR_REG: case STM32_I2S_CGFR_REG: return true; @@ -290,7 +289,7 @@ static bool stm32_i2s_readable_reg(struct device *dev, unsigned int reg) static bool stm32_i2s_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case STM32_I2S_TXDR_REG: + case STM32_I2S_SR_REG: case STM32_I2S_RXDR_REG: return true; default: -- 2.7.4

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[alsa-devel] Applied "ASoC: rsnd: src: fix compiler warnings" to the asoc tree
by Mark Brown 11 Mar '19

11 Mar '19

The patch ASoC: rsnd: src: fix compiler warnings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 399706df420ea8bc8b31283a919e6475f737d0ea Mon Sep 17 00:00:00 2001 From: Jiada Wang <jiada_wang(a)mentor.com> Date: Thu, 7 Mar 2019 15:15:54 +0900 Subject: [PATCH] ASoC: rsnd: src: fix compiler warnings compiler complains about following declarations sound/soc/sh/rcar/src.c:174:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern1[] = { ^~~~~ sound/soc/sh/rcar/src.c:183:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsdsr_table_pattern2[] = { ^~~~~ sound/soc/sh/rcar/src.c:192:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 bsisr_table[] = { ^~~~~ sound/soc/sh/rcar/src.c:201:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan288888[] = { ^~~~~ sound/soc/sh/rcar/src.c:210:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan244888[] = { ^~~~~ sound/soc/sh/rcar/src.c:219:1: warning: 'static' is not at beginning of declaration [-Wold-style-declaration] const static u32 chan222222[] = { ^~~~~ This patch moves the 'static' keyword to the front of the declaration to fix the compiler warnings Fixes: linux-next commit 7674bec4fc09 ("ASoC: rsnd: update BSDSR/BSDISR handling") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/src.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 45096a66c074..585ffba0244b 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -134,7 +134,7 @@ unsigned int rsnd_src_get_rate(struct rsnd_priv *priv, return rate; } -const static u32 bsdsr_table_pattern1[] = { +static const u32 bsdsr_table_pattern1[] = { 0x01800000, /* 6 - 1/6 */ 0x01000000, /* 6 - 1/4 */ 0x00c00000, /* 6 - 1/3 */ @@ -143,7 +143,7 @@ const static u32 bsdsr_table_pattern1[] = { 0x00400000, /* 6 - 1 */ }; -const static u32 bsdsr_table_pattern2[] = { +static const u32 bsdsr_table_pattern2[] = { 0x02400000, /* 6 - 1/6 */ 0x01800000, /* 6 - 1/4 */ 0x01200000, /* 6 - 1/3 */ @@ -152,7 +152,7 @@ const static u32 bsdsr_table_pattern2[] = { 0x00600000, /* 6 - 1 */ }; -const static u32 bsisr_table[] = { +static const u32 bsisr_table[] = { 0x00100060, /* 6 - 1/6 */ 0x00100040, /* 6 - 1/4 */ 0x00100030, /* 6 - 1/3 */ @@ -161,7 +161,7 @@ const static u32 bsisr_table[] = { 0x00100020, /* 6 - 1 */ }; -const static u32 chan288888[] = { +static const u32 chan288888[] = { 0x00000006, /* 1 to 2 */ 0x000001fe, /* 1 to 8 */ 0x000001fe, /* 1 to 8 */ @@ -170,7 +170,7 @@ const static u32 chan288888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan244888[] = { +static const u32 chan244888[] = { 0x00000006, /* 1 to 2 */ 0x0000001e, /* 1 to 4 */ 0x0000001e, /* 1 to 4 */ @@ -179,7 +179,7 @@ const static u32 chan244888[] = { 0x000001fe, /* 1 to 8 */ }; -const static u32 chan222222[] = { +static const u32 chan222222[] = { 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ 0x00000006, /* 1 to 2 */ -- 2.20.1

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[alsa-devel] [PATCH v11 0/5] Media Device Allocator API
by Shuah Khan 11 Mar '19

11 Mar '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. - This patch series in rested on 5.0-rc7 and addresses comments on v10 series from Hans Verkuil. Fixed problems found in resource sharing logic in au0828 adding a patch 5 to this series. - I am sharing the test plan used for testing resource sharing which could serve as a regression test plan. Test results can be found at: https://docs.google.com/document/d/1XMs3HYgLiHby6xVIvIxv75KSXAAN3F4uUpw2uLu… - v10 was tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. - v9 was tested on 4.20-rc6. - Tested sharing resources with kaffeine, vlc, xawtv, tvtime, and arecord. When analog is streaming, digital and audio user-space applications detect that the tuner is busy and exit. When digital is streaming, analog and audio applications detect that the tuner is busy and exit. When arecord is owns the tuner, digital and analog detect that the tuner is busy and exit. - Tested media device allocator API with bind/unbind testing on snd-usb-audio and au0828 drivers to make sure /dev/mediaX is released only when the last driver is unbound. - Addressed review comments from Hans on the RFC v8 (rebased on 4.19) - Updated change log to describe the use-case more clearly. - No changes to 0001,0002 code since the v7 referenced below. - 0003 is a new patch to enable ALSA defines that have been disabled for kernel between 4.9 and 4.19. - Minor merge conflict resolution in 0004. - Added SPDX to new files. Changes since v10: - Patch 1: Fixed SPDX tag and removed redundant IS_ENABLED(CONFIG_USB) around media_device_usb_allocate() - Sakari Ailus's review comment. - Patch 2 and 3: No changes - Patch 4: Fixed SPDX tag - Sakari Ailus's review comment. - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - Patch 5: This is a new patch added to fix resource sharing inconsistencies and problem found during testing using Han's tests. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (5): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies Documentation/media/kapi/mc-core.rst | 41 ++++ drivers/media/Makefile | 4 + drivers/media/media-dev-allocator.c | 142 +++++++++++ drivers/media/usb/au0828/au0828-core.c | 190 ++++++++++---- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 53 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 ++ sound/usb/card.h | 3 + sound/usb/media.c | 327 +++++++++++++++++++++++++ sound/usb/media.h | 74 ++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 18 files changed, 865 insertions(+), 61 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.17.1

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[alsa-devel] [PATCH v2] ASoC: fsl: sai: Fix underrrun for playback stream start
by Daniel Baluta 11 Mar '19

11 Mar '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> For [dir=playback, cmd = PCM_TRIGGER_START], fsl_sai_trigger does the following: * Enable DMA request * Enable Transmitter Now, because DMA doesn't copy data to TDR fast enough we hit TX FIFO underrun error: [ 37.175974] fsl-sai 308b0000.sai: isr: Transmit underrun detected The explanation for this can be found in SAI documentation for example chapter 16.16.3.5.2 FIFO pointers of the i.MX8DXP reference manual [1] " If the Transmit FIFO is empty, then to avoid a FIFO underrun, the Transmit Data Register must be written at least 3 bit clocks before the start of the next unmasked word. Before enabling the transmitter, the Transmit FIFO should be initialized with data (since after the transmitter is enabled, the transmitter will start a new frame, and if no data is in the FIFO, then the transmitter will immediately give an error). " [1] https://www.nxp.com/docs/en/reference-manual/IMX8DQXPRM.pdf Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v1: - as per Fabio's request clarify the issue and the solution - move words writing before enabling DMA request sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 2d376e3a3c80..a739baa99c1a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -522,7 +522,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int channels = substream->runtime->channels; u32 xcsr, count = 100; + int i; /* * Asynchronous mode: Clear SYNC for both Tx and Rx. @@ -542,6 +544,11 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + for (i = 0; tx && i < channels; i++) + regmap_write(sai->regmap, FSL_SAI_TDR, 0x0); + if (tx) + udelay(10); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); -- 2.17.1

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[alsa-devel] [PATCH] ASoC: fsl: sai: Reduce underrun when stream starts
by Daniel Baluta 08 Mar '19

08 Mar '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> Write initial words into SAI FIFO to reduce the underrun error. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9e0872f73d..99edbba4e8fa 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -503,7 +503,9 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int channels = substream->runtime->channels; u32 xcsr, count = 100; + int i; /* * Asynchronous mode: Clear SYNC for both Tx and Rx. @@ -526,6 +528,11 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); + for (i = 0; tx && i < channels; i++) + regmap_write(sai->regmap, FSL_SAI_TDR, 0x0); + if (tx) + udelay(10); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); regmap_update_bits(sai->regmap, FSL_SAI_TCSR, -- 2.17.1

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[alsa-devel] [PATCH] ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function
by mac.chiang＠intel.com 08 Mar '19

08 Mar '19

From: Mac Chiang <mac.chiang(a)intel.com> s0ix entry failure is root cause by default speaker/vi sense path activate all the time that result in the audio IPs are not power gated(checked by pch_ip_power_gating_status). Adding the run time speaker control ensure playback widgets are going to free and off eventually. With this func, counter works in s0ix subsequently (checked by slp_s0_residency_usc). Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Harshapriya.n <harshapriya.n(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 50 ++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 723a493..e01ca9a 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -195,8 +195,58 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return 0; } +static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int j, ret; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + const char *name = codec_dai->component->name; + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + char pin_name[20]; + + if (strcmp(name, MAXIM_DEV0_NAME) && + strcmp(name, MAXIM_DEV1_NAME)) + continue; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to enable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to disable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + } + } + + return 0; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, + .trigger = kabylake_ssp0_trigger, }; static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, -- 2.7.4

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[alsa-devel] [PATCH Resend] ASoC: rt5682: Check JD status when system resume
by shumingf＠realtek.com 08 Mar '19

08 Mar '19

From: Shuming Fan <shumingf(a)realtek.com> The IRQ function may not work when system suspend. We remove snd_soc_dapm_force_enable_pin function call to make sure the bias off when idle and run into suspend/resume function. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 9d5acd2d04ab..b7e2e9937e6d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -910,13 +910,20 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); unsigned int val, count; if (jack_insert) { - snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, RT5682_PWR_VREF2); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + usleep_range(15000, 20000); + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, RT5682_PWR_CBJ); + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_HIGH); @@ -944,8 +951,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_dapm_disable_pin(dapm, "CBJ Power"); - snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2, 0); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, 0); rt5682->jack_type = 0; } @@ -1591,8 +1600,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, - rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1627,9 +1634,6 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("CBJ Power", RT5682_PWR_ANLG_3, - RT5682_PWR_CBJ_BIT, 0, NULL, 0), - /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5682_rec1_l_mix, ARRAY_SIZE(rt5682_rec1_l_mix)), @@ -1792,17 +1796,13 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*Vref*/ {"MICBIAS1", NULL, "Vref1"}, - {"MICBIAS1", NULL, "Vref2"}, {"MICBIAS2", NULL, "Vref1"}, - {"MICBIAS2", NULL, "Vref2"}, {"CLKDET SYS", NULL, "CLKDET"}, {"IN1P", NULL, "LDO2"}, {"BST1 CBJ", NULL, "IN1P"}, - {"BST1 CBJ", NULL, "CBJ Power"}, - {"CBJ Power", NULL, "Vref2"}, {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, {"RECMIX1L", NULL, "RECMIX1L Power"}, @@ -1912,9 +1912,7 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"HP Amp", NULL, "Capless"}, {"HP Amp", NULL, "Charge Pump"}, {"HP Amp", NULL, "CLKDET SYS"}, - {"HP Amp", NULL, "CBJ Power"}, {"HP Amp", NULL, "Vref1"}, - {"HP Amp", NULL, "Vref2"}, {"HPOL Playback", "Switch", "HP Amp"}, {"HPOR Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HPOL Playback"}, @@ -2363,6 +2361,8 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_cache_only(rt5682->regmap, false); regcache_sync(rt5682->regmap); + rt5682_irq(0, rt5682); + return 0; } #else -- 2.21.0

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