From 33c8516841ea4fa12fdb8961711bf95095c607ee Mon Sep 17 00:00:00 2001 From: Rander Wang Date: Fri, 25 Jun 2021 15:50:39 -0500 Subject: ASoC: Intel: boards: fix xrun issue on platform with max98373 MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit On TGL platform with max98373 codec the trigger start sequence is fe first, then codec component and sdw link is the last. Recently a delay was introduced in max98373 codec driver and this resulted to the start of sdw stream transmission was delayed and the data transmitted by fw can't be consumed by sdw controller, so xrun happened. Adding delay in trigger function is a bad idea. This patch enable spk pin in prepare function and disable it in hw_free to avoid xrun issue caused by delay in trigger. Fixes: 3a27875e91fb ("ASoC: max98373: Added 30ms turn on/off time delay") BugLink: https://github.com/thesofproject/sof/issues/4066 Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Rander Wang Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210625205042.65181-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_max98373.c | 81 ++++++++++++++++++++----------- 1 file changed, 53 insertions(+), 28 deletions(-) (limited to 'sound') diff --git a/sound/soc/intel/boards/sof_sdw_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c index 0e7ed906b341..25daef910aee 100644 --- a/sound/soc/intel/boards/sof_sdw_max98373.c +++ b/sound/soc/intel/boards/sof_sdw_max98373.c @@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd) +static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int ret; + int j; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - /* enable max98373 first */ - ret = max_98373_trigger(substream, cmd); - if (ret < 0) - break; - - ret = sdw_trigger(substream, cmd); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = sdw_trigger(substream, cmd); - if (ret < 0) - break; - - ret = max_98373_trigger(substream, cmd); - break; - default: - ret = -EINVAL; - break; + /* set spk pin by playback only */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + for_each_rtd_codec_dais(rtd, j, codec_dai) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(cpu_dai->component); + char pin_name[16]; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + if (enable) + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + else + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + + if (!ret) + snd_soc_dapm_sync(dapm); } - return ret; + return 0; +} + +static int mx8373_sdw_prepare(struct snd_pcm_substream *substream) +{ + int ret = 0; + + /* according to soc_pcm_prepare dai link prepare is called first */ + ret = sdw_prepare(substream); + if (ret < 0) + return ret; + + return mx8373_enable_spk_pin(substream, true); +} + +static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream) +{ + int ret = 0; + + /* according to soc_pcm_hw_free dai link free is called first */ + ret = sdw_hw_free(substream); + if (ret < 0) + return ret; + + return mx8373_enable_spk_pin(substream, false); } static const struct snd_soc_ops max_98373_sdw_ops = { .startup = sdw_startup, - .prepare = sdw_prepare, - .trigger = max98373_sdw_trigger, - .hw_free = sdw_hw_free, + .prepare = mx8373_sdw_prepare, + .trigger = sdw_trigger, + .hw_free = mx8373_sdw_hw_free, .shutdown = sdw_shutdown, }; -- cgit v1.2.3-70-g09d2 From 0c4f8fd3ed9cb27228497f0ae495ea6cef7017b1 Mon Sep 17 00:00:00 2001 From: Peter Robinson Date: Sun, 27 Jun 2021 11:59:55 +0100 Subject: ASoC: remove zte zx dangling kconfig In commit dc98f1d we removed the zte zx sound drivers but there was a dangling Kconfig left around for the codec so fix this. Fixes: dc98f1d655ca ("ASoC: remove zte zx drivers") Signed-off-by: Peter Robinson Cc: Arnd Bergmann Cc: Mark Brown Acked-by: Arnd Bergmann Link: https://lore.kernel.org/r/20210627105955.3410015-1-pbrobinson@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 ----- 1 file changed, 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3abdda48dc8e..bea7b47eddbe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1813,11 +1813,6 @@ config SND_SOC_ZL38060 which consists of a Digital Signal Processor (DSP), several Digital Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. -config SND_SOC_ZX_AUD96P22 - tristate "ZTE ZX AUD96P22 CODEC" - depends on I2C - select REGMAP_I2C - # Amp config SND_SOC_LM4857 tristate -- cgit v1.2.3-70-g09d2 From dd6fb8ff2210f74b056bf9234d0605e8c26a8ac0 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sat, 26 Jun 2021 16:59:39 +0100 Subject: ASoC: wm_adsp: Correct wm_coeff_tlv_get handling When wm_coeff_tlv_get was updated it was accidentally switch to the _raw version of the helper causing it to ignore the current DSP state it should be checking. Switch the code back to the correct helper so that users can't read the controls when they arn't available. Fixes: 73ecf1a673d3 ("ASoC: wm_adsp: Correct cache handling of new kernel control API") Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20210626155941.12251-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 37aa020f23f6..59d876d36cfd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1213,7 +1213,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size); + ret = wm_coeff_read_ctrl(ctl, ctl->cache, size); if (!ret && copy_to_user(bytes, ctl->cache, size)) ret = -EFAULT; -- cgit v1.2.3-70-g09d2 From e588332271b9cde6252dac8973b77e580cd639bd Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Sat, 26 Jun 2021 16:59:40 +0100 Subject: ASoC: wm_adsp: Add CCM_CORE_RESET to Halo start core When starting the Halo core it is advised to also write the core reset bit, this ensures the part starts up in the appropriate state. Omitting this doesn't cause issues on most parts but cs40l25 requires it and it is advised on all Halo parts. Signed-off-by: Charles Keepax Link: https://lore.kernel.org/r/20210626155941.12251-2-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 59d876d36cfd..549d98241dae 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -282,6 +282,7 @@ /* * HALO_CCM_CORE_CONTROL */ +#define HALO_CORE_RESET 0x00000200 #define HALO_CORE_EN 0x00000001 /* @@ -3333,7 +3334,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp) { return regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, - HALO_CORE_EN, HALO_CORE_EN); + HALO_CORE_RESET | HALO_CORE_EN, + HALO_CORE_RESET | HALO_CORE_EN); } static void wm_halo_stop_core(struct wm_adsp *dsp) -- cgit v1.2.3-70-g09d2 From 2c70ff56e49ae219640689a0c86041c0f656046f Mon Sep 17 00:00:00 2001 From: Lucas Stach Date: Mon, 28 Jun 2021 23:04:58 +0200 Subject: ASoC: codecs: allow SSM2518 to be selected by the user Allow the Analog SSM2518 driver to be enabled without a large bunch of other drivers. Signed-off-by: Lucas Stach Link: https://lore.kernel.org/r/20210628210458.2508973-1-l.stach@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bea7b47eddbe..3a42c4611414 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305 high-efficiency mono Class-D audio power amplifiers. config SND_SOC_SSM2518 - tristate + tristate "Analog Devices SSM2518 Class-D Amplifier" depends on I2C config SND_SOC_SSM2602 -- cgit v1.2.3-70-g09d2 From 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf Mon Sep 17 00:00:00 2001 From: Kyle Russell Date: Mon, 21 Jun 2021 21:09:41 -0400 Subject: ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- cgit v1.2.3-70-g09d2 From 0dfc21c1a4cac321749a53c92da616d9546d00e3 Mon Sep 17 00:00:00 2001 From: Thierry Reding Date: Thu, 8 Jul 2021 12:34:31 +0200 Subject: ASoC: tegra: Use ADMAIF component for DMA allocations DMA memory is currently allocated for the soundcard device, which is a virtual device added for the sole purpose of "stitching" together the audio device. It is not a real device and therefore doesn't have a DMA mask or a description of the path to and from memory of accesses. Memory accesses really originate from the ADMA controller that provides the DMA channels used by the PCM component. However, since the DMA memory is allocated up-front and the DMA channels aren't known at that point, there is no way of knowing the DMA channel provider at allocation time. The next best physical device in the memory path is the ADMAIF. Use it as the device to allocate DMA memory to. iommus and interconnects device tree properties can thus be added to the ADMAIF device tree node to describe the memory access path for audio. Signed-off-by: Thierry Reding Link: https://lore.kernel.org/r/20210708103432.1690385-2-thierry.reding@gmail.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 573374b89b10..d3276b4595af 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(tegra_pcm_pointer); -static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, +static int tegra_pcm_preallocate_dma_buffer(struct device *dev, struct snd_pcm *pcm, int stream, size_t size) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - buf->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL); + buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) return -ENOMEM; buf->private_data = NULL; buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; + buf->dev.dev = dev; buf->bytes = size; return 0; @@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) if (!buf->area) return; - dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr); + dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; } -static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd, +static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd, size_t size) { - struct snd_card *card = rtd->card->snd_card; struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32)); if (ret < 0) return ret; if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size); if (ret) goto err; } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size); if (ret) goto err_free_play; } @@ -284,7 +281,16 @@ err: int tegra_pcm_construct(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max); + struct device *dev = component->dev; + + /* + * Fallback for backwards-compatibility with older device trees that + * have the iommus property in the virtual, top-level "sound" node. + */ + if (!of_get_property(dev->of_node, "iommus", NULL)) + dev = rtd->card->snd_card->dev; + + return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max); } EXPORT_SYMBOL_GPL(tegra_pcm_construct); -- cgit v1.2.3-70-g09d2 From 2169d6a0f0721935410533281fc7e87e4e2322d1 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Thu, 8 Jul 2021 11:12:55 +0200 Subject: ASoC: tlv320aic32x4: Fix TAS2505 volume controls None of the TAS2505 outputs are stereo, do not pretend they are by implementing them using SOC*DOUBLE* macros referencing the same register twice, use SOC*SINGLE* instead. Fix volume ranges and mute control for the codec according to datasheet. Fixes: b4525b6196cd7 ("ASoC: tlv320aic32x4: add support for TAS2505") Signed-off-by: Marek Vasut Cc: Claudius Heine Cc: Mark Brown Link: https://lore.kernel.org/r/20210708091255.56502-1-marex@denx.de Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 27 +++++++++++++-------------- 1 file changed, 13 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c63b717040ed..dcd8aeb45cb3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0); /* -12dB min, 0.5dB steps */ static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0); - -static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0); +/* -6dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1); static const char * const lo_cm_text[] = { @@ -1063,21 +1063,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { }; static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = { - SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, - AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), + SOC_SINGLE_S8_TLV("PCM Playback Volume", + AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm), SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum), - SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0, - tlv_driver_gain), - SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 6, 0x01, 1), - SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + SOC_SINGLE_TLV("HP Driver Gain Volume", + AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1), - SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1, - 0, 0, 117, 1, tlv_spk_vol), - SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2, - 4, 5, 0, tlv_amp_vol), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", + TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), }; static const struct snd_kcontrol_new hp_output_mixer_controls[] = { -- cgit v1.2.3-70-g09d2 From 6c621b811f99feb3941f04b386795b45f47cd771 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Jul 2021 17:02:34 +0100 Subject: ASoC: tlv320aic31xx: Make regmap cache only on probe() Currently the tlv320aic31xx driver has regulator support but does not enable the regulators during probe, deferring this until something causes ASoC to make the card active. It does put the device into cache only mode but only when the component level probe is called, however if interrupts are in use the driver will access the regmap before then which if the regulators are not powered on would cause I/O problems. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20210707160234.16253-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 51870d50f419..b504d63385b3 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1604,6 +1604,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, ret); return ret; } + regcache_cache_only(aic31xx->regmap, true); + aic31xx->dev = &i2c->dev; aic31xx->irq = i2c->irq; -- cgit v1.2.3-70-g09d2 From c71f78a662611fe2c67f3155da19b0eff0f29762 Mon Sep 17 00:00:00 2001 From: Maxim Schwalm Date: Mon, 12 Jul 2021 03:50:11 +0300 Subject: ASoC: rt5631: Fix regcache sync errors on resume The ALC5631 does not like multi-write accesses, avoid them. This fixes: rt5631 4-001a: Unable to sync registers 0x3a-0x3c. -121 errors on resume from suspend (and all registers after the registers in the error not being synced). Inspired by commit 2d30e9494f1e ("ASoC: rt5651: Fix regcache sync errors on resume") from Hans de Geode, which fixed the same errors on ALC5651. Signed-off-by: Maxim Schwalm Link: https://lore.kernel.org/r/20210712005011.28536-1-digetx@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 3000bc128b5b..38356ea2bd6e 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = { .reg_defaults = rt5631_reg, .num_reg_defaults = ARRAY_SIZE(rt5631_reg), .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, }; static int rt5631_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3-70-g09d2 From aa21548e34c19c12e924c736f3fd9e6a4d0f5419 Mon Sep 17 00:00:00 2001 From: Sathya Prakash M R Date: Mon, 12 Jul 2021 15:16:20 -0500 Subject: ASoC: SOF: Intel: Update ADL descriptor to use ACPI power states The ADL descriptor was missing an ACPI power setting, causing the DSP to enter D3 even with a D0i1-compatible wake-on-voice/hotwording capture stream. Fixes: 4ad03f894b3c ('ASoC: SOF: Intel: Update ADL P to use its own descriptor') Reviewed-by: Ranjani Sridharan Signed-off-by: Sathya Prakash M R Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20210712201620.44311-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/pci-tgl.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index a00262184efa..d04ce84fe7cc 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = { static const struct sof_dev_desc adl_desc = { .machines = snd_soc_acpi_intel_adl_machines, .alt_machines = snd_soc_acpi_intel_adl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, -- cgit v1.2.3-70-g09d2 From 9431f8df233f808baa5fcc62b520cc6503fdf022 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Tue, 13 Jul 2021 15:04:17 +0100 Subject: ASoC: codecs: wcd938x: make sdw dependency explicit in Kconfig currenlty wcd938x has only soundwire interface and depends on symbols from wcd938x soundwire module, so make this dependency explicit in Kconfig Without this one of the randconfig endup setting CONFIG_SND_SOC_WCD938X=y CONFIG_SND_SOC_WCD938X_SDW=m resulting in some undefined reference to wcd938x_sdw* symbols. Reported-by: kernel test robot Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210713140417.23693-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3a42c4611414..032c87637f63 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1557,6 +1557,7 @@ config SND_SOC_WCD934X Qualcomm SoCs like SDM845. config SND_SOC_WCD938X + depends on SND_SOC_WCD938X_SDW tristate config SND_SOC_WCD938X_SDW -- cgit v1.2.3-70-g09d2 From f99986c0fcad8e1d7d842e9a636f55bcc6748da5 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Fri, 16 Jul 2021 11:57:35 +0100 Subject: ASoC: codecs: wcd938x: setup irq during component bind SoundWire registers are only accessable after sdw components are succesfully binded. Setup irqs at that point instead of doing at probe. Signed-off-by: Srinivas Kandagatla Link: https://lore.kernel.org/r/20210716105735.6073-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 78b76eceff8f..2fcc97370be2 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component) (WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0); } - ret = wcd938x_irq_init(wcd938x, component->dev); - if (ret) { - dev_err(component->dev, "%s: IRQ init failed: %d\n", - __func__, ret); - return ret; - } - wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, WCD938X_IRQ_HPHR_PDM_WD_INT); wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, @@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); if (!wcd938x->tx_sdw_dev) { dev_err(dev, "could not get txslave with matching of dev\n"); @@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev) return PTR_ERR(wcd938x->regmap); } + ret = wcd938x_irq_init(wcd938x, dev); + if (ret) { + dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret); + return ret; + } + + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; + wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; + ret = wcd938x_set_micbias_data(wcd938x); if (ret < 0) { dev_err(dev, "%s: bad micbias pdata\n", __func__); -- cgit v1.2.3-70-g09d2 From 59dd33f82dc0975c55d3d46801e7ca45532d7673 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 16 Jul 2021 18:00:12 +0530 Subject: ASoC: soc-pcm: add a flag to reverse the stop sequence On stream stop, currently CPU DAI stop sequence invoked first followed by DMA. For Few platforms, it is required to stop the DMA first before stopping CPU DAI. Introduced new flag in dai_link structure for reordering stop sequence. Based on flag check, ASoC core will re-order the stop sequence. Fixes: 4378f1fbe92405 ("ASoC: soc-pcm: Use different sequence for start/stop trigger") Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20210716123015.15697-1-vijendar.mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc.h | 6 ++++++ sound/soc/soc-pcm.c | 22 ++++++++++++++++------ 2 files changed, 22 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 675849d07284..8e6dd8a257c5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -712,6 +712,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* This flag will reorder stop sequence. By enabling this flag + * DMA controller stop sequence will be invoked first followed by + * CPU DAI driver stop sequence + */ + unsigned int stop_dma_first:1; + #ifdef CONFIG_SND_SOC_TOPOLOGY struct snd_soc_dobj dobj; /* For topology */ #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 46513bb97904..d1c570ca21ea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1015,6 +1015,7 @@ out: static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = -EINVAL, _ret = 0; int rollback = 0; @@ -1055,14 +1056,23 @@ start_err: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); - if (ret < 0) - break; + if (rtd->dai_link->stop_dma_first) { + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; - ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); - if (ret < 0) - break; + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } else { + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } ret = snd_soc_link_trigger(substream, cmd, rollback); break; } -- cgit v1.2.3-70-g09d2 From 7883490cba002121a5870e786a1dc0acce5e1caf Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 16 Jul 2021 18:00:13 +0530 Subject: ASoC: amd: reverse stop sequence for stoneyridge platform For Stoneyridge platform, it is required to invoke DMA driver stop first rather than invoking DWC I2S controller stop. Enable dai_link structure stop_dma_fist flag to reverse the stop sequence. Signed-off-by: Vijendar Mukunda Link: https://lore.kernel.org/r/20210716123015.15697-2-vijendar.mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 84e3906abd4f..9449fb40a956 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -576,6 +576,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_rt5682_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_play_ops, SND_SOC_DAILINK_REG(designware1, rt5682, platform), }, @@ -585,6 +586,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_cap_ops, SND_SOC_DAILINK_REG(designware2, rt5682, platform), }, @@ -594,6 +596,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -604,6 +607,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -614,6 +618,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, -- cgit v1.2.3-70-g09d2 From 6a503e1c455316fd0bfd8188c0a62cce7c5525ca Mon Sep 17 00:00:00 2001 From: Oder Chiou Date: Fri, 16 Jul 2021 16:58:53 +0800 Subject: ASoC: rt5682: Fix the issue of garbled recording after powerd_dbus_suspend While using the DMIC recording, the garbled data will be captured by the DMIC. It is caused by the critical power of PLL closed in the jack detect function. Signed-off-by: Oder Chiou Link: https://lore.kernel.org/r/20210716085853.20170-1-oder_chiou@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e4c91571abae..abcd6f483788 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) 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); - if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); - if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, -- cgit v1.2.3-70-g09d2 From 78d2a05ef22e7b5863b01e073dd6a06b3979bb00 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Sat, 17 Jul 2021 15:28:18 +0300 Subject: ASoC: ti: j721e-evm: Fix unbalanced domain activity tracking during startup In case of an error within j721e_audio_startup() the domain->active must be decremented to avoid unbalanced counter. Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20210717122820.1467-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown --- sound/soc/ti/j721e-evm.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index a7c0484d44ec..017c4ad11ca6 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream) j721e_rule_rate, &priv->rate_range, SNDRV_PCM_HW_PARAM_RATE, -1); - mutex_unlock(&priv->mutex); if (ret) - return ret; + goto out; /* Reset TDM slots to 32 */ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } - return 0; + if (ret == -ENOTSUPP) + ret = 0; +out: + if (ret) + domain->active--; + mutex_unlock(&priv->mutex); + + return ret; } static int j721e_audio_hw_params(struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From 82d28b67f780910f816fe1cfb0f676fc38c4cbb3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Sat, 17 Jul 2021 15:28:19 +0300 Subject: ASoC: ti: j721e-evm: Check for not initialized parent_clk_id During probe the parent_clk_id is set to -1 which should not be used to array index within hsdiv_rates[]. Signed-off-by: Peter Ujfalusi Link: https://lore.kernel.org/r/20210717122820.1467-3-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown --- sound/soc/ti/j721e-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 017c4ad11ca6..265bbc5a2f96 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv, return ret; } - if (priv->hsdiv_rates[domain->parent_clk_id] != scki) { + if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) { dev_dbg(priv->dev, "%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n", audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI", -- cgit v1.2.3-70-g09d2