September 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: cs42l42: Use two thresholds and increased wait time for manual type detection
by Vitaly Rodionov 27 Sep '21

27 Sep '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Some headsets require very different comparator thresholds for type detection, as well as longer settling times. In order to detect a larger number of headsets, use 2 thresholds to give maximum coverage (1.25V and 1.75V), as well as a longer settling time of 100ms. This will not affect default audotodetect mode and applies to manual mode type detection only. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 84 +++++++++++++++++++++++++++----------- sound/soc/codecs/cs42l42.h | 5 +++ 2 files changed, 66 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 78f8c89410b6..3f202acd4a8e 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1048,7 +1048,8 @@ static struct snd_soc_dai_driver cs42l42_dai = { static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) { unsigned int hs_det_status; - unsigned int hs_det_comp; + unsigned int hs_det_comp1; + unsigned int hs_det_comp2; unsigned int hs_det_sw; /* Set hs detect to manual, active mode */ @@ -1063,23 +1064,40 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) (0 << CS42L42_HSBIAS_REF_SHIFT) | (0 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + /* Configure HS DET comparator reference levels. */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL1, + CS42L42_HSDET_COMP1_LVL_MASK | + CS42L42_HSDET_COMP2_LVL_MASK, + (CS42L42_HSDET_COMP1_LVL_VAL << CS42L42_HSDET_COMP1_LVL_SHIFT) | + (CS42L42_HSDET_COMP2_LVL_VAL << CS42L42_HSDET_COMP2_LVL_SHIFT)); + /* Open the SW_HSB_HS3 switch and close SW_HSB_HS4 for a Type 1 headset. */ regmap_write(cs42l42->regmap, CS42L42_HS_SWITCH_CTL, CS42L42_HSDET_SW_COMP1); + msleep(100); + regmap_read(cs42l42->regmap, CS42L42_HS_DET_STATUS, &hs_det_status); - hs_det_comp = (hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> + hs_det_comp1 = (hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> CS42L42_HSDET_COMP1_OUT_SHIFT; + hs_det_comp2 = (hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> + CS42L42_HSDET_COMP2_OUT_SHIFT; /* Close the SW_HSB_HS3 switch for a Type 2 headset. */ regmap_write(cs42l42->regmap, CS42L42_HS_SWITCH_CTL, CS42L42_HSDET_SW_COMP2); + msleep(100); + regmap_read(cs42l42->regmap, CS42L42_HS_DET_STATUS, &hs_det_status); - hs_det_comp |= ((hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> + hs_det_comp1 |= ((hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> + CS42L42_HSDET_COMP1_OUT_SHIFT) << 1; + hs_det_comp2 |= ((hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> CS42L42_HSDET_COMP2_OUT_SHIFT) << 1; - switch (hs_det_comp) { + /* Use Comparator 1 with 1.25V Threshold. */ + switch (hs_det_comp1) { case CS42L42_HSDET_COMP_TYPE1: cs42l42->hs_type = CS42L42_PLUG_CTIA; hs_det_sw = CS42L42_HSDET_SW_TYPE1; @@ -1088,14 +1106,26 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = CS42L42_PLUG_OMTP; hs_det_sw = CS42L42_HSDET_SW_TYPE2; break; - case CS42L42_HSDET_COMP_TYPE3: - cs42l42->hs_type = CS42L42_PLUG_HEADPHONE; - hs_det_sw = CS42L42_HSDET_SW_TYPE3; - break; default: - cs42l42->hs_type = CS42L42_PLUG_INVALID; - hs_det_sw = CS42L42_HSDET_SW_TYPE4; - break; + /* Fallback to Comparator 2 with 1.75V Threshold. */ + switch (hs_det_comp2) { + case CS42L42_HSDET_COMP_TYPE1: + cs42l42->hs_type = CS42L42_PLUG_CTIA; + hs_det_sw = CS42L42_HSDET_SW_TYPE1; + break; + case CS42L42_HSDET_COMP_TYPE2: + cs42l42->hs_type = CS42L42_PLUG_OMTP; + hs_det_sw = CS42L42_HSDET_SW_TYPE2; + break; + case CS42L42_HSDET_COMP_TYPE3: + cs42l42->hs_type = CS42L42_PLUG_HEADPHONE; + hs_det_sw = CS42L42_HSDET_SW_TYPE3; + break; + default: + cs42l42->hs_type = CS42L42_PLUG_INVALID; + hs_det_sw = CS42L42_HSDET_SW_TYPE4; + break; + } } /* Set Switches */ @@ -1112,6 +1142,14 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) (0 << CS42L42_HSDET_SET_SHIFT) | (0 << CS42L42_HSBIAS_REF_SHIFT) | (0 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + + /* Configure HS DET comparator reference levels. */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL1, + CS42L42_HSDET_COMP1_LVL_MASK | + CS42L42_HSDET_COMP2_LVL_MASK, + (CS42L42_HSDET_COMP1_LVL_DEFAULT << CS42L42_HSDET_COMP1_LVL_SHIFT) | + (CS42L42_HSDET_COMP2_LVL_DEFAULT << CS42L42_HSDET_COMP2_LVL_SHIFT)); } static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) @@ -1134,6 +1172,18 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = (hs_det_status & CS42L42_HSDET_TYPE_MASK) >> CS42L42_HSDET_TYPE_SHIFT; + /* Set hs detect to automatic, disabled mode */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL2, + CS42L42_HSDET_CTRL_MASK | + CS42L42_HSDET_SET_MASK | + CS42L42_HSBIAS_REF_MASK | + CS42L42_HSDET_AUTO_TIME_MASK, + (2 << CS42L42_HSDET_CTRL_SHIFT) | + (2 << CS42L42_HSDET_SET_SHIFT) | + (0 << CS42L42_HSBIAS_REF_SHIFT) | + (3 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + /* Run Manual detection if auto detect has not found a headset. * We Re-Run with Manual Detection if the original detection was invalid or headphones, * to ensure that a headset mic is detected in all cases. @@ -1142,18 +1192,6 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type == CS42L42_PLUG_HEADPHONE) { dev_dbg(cs42l42->component->dev, "Running Manual Detection Fallback\n"); cs42l42_manual_hs_type_detect(cs42l42); - } else { - /* Set hs detect to automatic, disabled mode */ - regmap_update_bits(cs42l42->regmap, - CS42L42_HSDET_CTL2, - CS42L42_HSDET_CTRL_MASK | - CS42L42_HSDET_SET_MASK | - CS42L42_HSBIAS_REF_MASK | - CS42L42_HSDET_AUTO_TIME_MASK, - (2 << CS42L42_HSDET_CTRL_SHIFT) | - (2 << CS42L42_HSDET_SET_SHIFT) | - (0 << CS42L42_HSBIAS_REF_SHIFT) | - (3 << CS42L42_HSDET_AUTO_TIME_SHIFT)); } /* Set up button detection */ diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 2aeabba73e05..0704c902475f 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -188,6 +188,11 @@ #define CS42L42_HSDET_COMP2_LVL_SHIFT 4 #define CS42L42_HSDET_COMP2_LVL_MASK (15 << CS42L42_HSDET_COMP2_LVL_SHIFT) +#define CS42L42_HSDET_COMP1_LVL_VAL 12 /* 1.25V Comparator */ +#define CS42L42_HSDET_COMP2_LVL_VAL 2 /* 1.75V Comparator */ +#define CS42L42_HSDET_COMP1_LVL_DEFAULT 7 /* 1V Comparator */ +#define CS42L42_HSDET_COMP2_LVL_DEFAULT 7 /* 2V Comparator */ + #define CS42L42_HSDET_CTL2 (CS42L42_PAGE_11 + 0x20) #define CS42L42_HSDET_AUTO_TIME_SHIFT 0 #define CS42L42_HSDET_AUTO_TIME_MASK (3 << CS42L42_HSDET_AUTO_TIME_SHIFT) -- 2.25.1

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[PATCH] ASoC: ad193x: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ad193x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ad193x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 278a55af158b..30b98b4267e1 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -243,22 +243,22 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, if (fmt & SND_SOC_DAIFMT_DSP_A) dac_fmt ^= AD193X_DAC_LEFT_HIGH; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: adc_fmt |= AD193X_ADC_LCR_MASTER; adc_fmt |= AD193X_ADC_BCLK_MASTER; dac_fmt |= AD193X_DAC_LCR_MASTER; dac_fmt |= AD193X_DAC_BCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */ + case SND_SOC_DAIFMT_CBC_CFP: adc_fmt |= AD193X_ADC_LCR_MASTER; dac_fmt |= AD193X_DAC_LCR_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + case SND_SOC_DAIFMT_CBP_CFC: adc_fmt |= AD193X_ADC_BCLK_MASTER; dac_fmt |= AD193X_DAC_BCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */ + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

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[PATCH] ASoC: meson: aiu: Fix spelling mistake "Unsupport" -> "Unsupported"
by Colin King 27 Sep '21

27 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err error message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/meson/aiu-encoder-spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-encoder-spdif.c b/sound/soc/meson/aiu-encoder-spdif.c index de850913975f..97da60db2c4d 100644 --- a/sound/soc/meson/aiu-encoder-spdif.c +++ b/sound/soc/meson/aiu-encoder-spdif.c @@ -113,7 +113,7 @@ static int aiu_encoder_spdif_hw_params(struct snd_pcm_substream *substream, val |= AIU_958_MISC_MODE_32BITS; break; default: - dev_err(dai->dev, "Unsupport physical width\n"); + dev_err(dai->dev, "Unsupported physical width\n"); return -EINVAL; } -- 2.32.0

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[PATCH] ASoC: samsung: add missing "fallthrough;"
by Arnd Bergmann 27 Sep '21

27 Sep '21

From: Arnd Bergmann <arnd(a)arndb.de> With gcc, we get a warning in this file: In file included from include/linux/io.h:13, from sound/soc/samsung/s3c-i2s-v2.c:16: sound/soc/samsung/s3c-i2s-v2.c: In function 's3c2412_i2s_trigger': arch/arm/include/asm/io.h:92:22: error: this statement may fall through [-Werror=implicit-fallthrough=] #define __raw_writel __raw_writel ^ arch/arm/include/asm/io.h:299:29: note: in expansion of macro '__raw_writel' #define writel_relaxed(v,c) __raw_writel((__force u32) cpu_to_le32(v),c) ^~~~~~~~~~~~ arch/arm/include/asm/io.h:307:36: note: in expansion of macro 'writel_relaxed' #define writel(v,c) ({ __iowmb(); writel_relaxed(v,c); }) ^~~~~~~~~~~~~~ sound/soc/samsung/s3c-i2s-v2.c:398:3: note: in expansion of macro 'writel' writel(0x0, i2s->regs + S3C2412_IISFIC); ^~~~~~ sound/soc/samsung/s3c-i2s-v2.c:400:2: note: here case SNDRV_PCM_TRIGGER_RESUME: ^~~~ >From all I can tell, this was indeed meant to fall through, so add "fallthrough;" statement to avoid the warning. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/s3c-i2s-v2.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index e9481187a08c..de66cc422e6e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -397,6 +397,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, /* clear again, just in case */ writel(0x0, i2s->regs + S3C2412_IISFIC); + fallthrough; + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!i2s->master) { -- 2.29.2

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[PATCH] ASoC: alc5623: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the alc5623 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/alc5623.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 54f489837162..b10357a6d655 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -641,12 +641,12 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; u16 iface = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface = ALC5623_DAI_SDP_MASTER_MODE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: iface = ALC5623_DAI_SDP_SLAVE_MODE; break; default: -- 2.20.1

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[PATCH] ASoC: alc5632: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the alc5632 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/alc5632.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 79813882a955..6d7af3736a91 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -815,12 +815,12 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; u16 iface = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface = ALC5632_DAI_SDP_MASTER_MODE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: iface = ALC5632_DAI_SDP_SLAVE_MODE; break; default: -- 2.20.1

1 1

[PATCH] ASoC: ak5558: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak5558 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak5558.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 37d4600b6f2c..c94cfde3e4a8 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -198,13 +198,13 @@ static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; u8 format; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFC: default: dev_err(dai->dev, "Clock mode unsupported"); return -EINVAL; -- 2.20.1

1 1

[PATCH] ASoC: cs42l42: Add HP Volume Scale control
by Vitaly Rodionov 27 Sep '21

27 Sep '21

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> cs42l42 has a configurable scaling of the maximum volume. Add an ALSA control for this. Note that the datasheet name is "full scale volume" but this conflicts with ALSA naming convention so the control is named "HP Volume Scale". Before this change the FULL_SCALE_VOLUME was set based on the value in RLA_STAT, but as there isn't any load detection result this always set the scaling to -6dB instead of the default 0dB. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d5e1e5228b5f..78f8c89410b6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -425,6 +425,14 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); +static const char * const cs42l42_full_scale_vol_text[] = { + "0dB", "-6dB" +}; + +static SOC_ENUM_SINGLE_DECL(cs42l42_full_scale_vol_enum, CS42L42_HP_CTL, + CS42L42_HP_FULL_SCALE_VOL_SHIFT, + cs42l42_full_scale_vol_text); + static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, @@ -450,6 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DACB_INV_SHIFT, true, false), SOC_SINGLE("DAC HPF Switch", CS42L42_DAC_CTL2, CS42L42_DAC_HPF_EN_SHIFT, true, false), + SOC_ENUM("HP Volume Scale", cs42l42_full_scale_vol_enum), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, 0x3f, 1, mixer_tlv) @@ -922,7 +931,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +1001,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } -- 2.25.1

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Broken Pipe with SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Ryan McClue 27 Sep '21

27 Sep '21

Hello again. I'm using alsa-kernel. My configuration: SNDRV_PCM_HW_PARAM_ACCESS = SNDRV_PCM_ACCESS_RW_INTERLEAVED SNDRV_PCM_HW_PARAM_FORMAT = SNDRV_PCM_FORMAT_S16_LE SNDRV_PCM_HW_PARAM_SUBFORMAT = SNDRV_PCM_SUBFORMAT_STD SNDRV_PCM_HW_PARAM_CHANNELS = 2 SNDRV_PCM_HW_PARAM_RATE = 48000 I have a vsynced game loop that is running at 60fps: > int num_base_samples = 48000 * (1 / 60); > int num_samples = num_base_samples * 2; > int16_t buffer[num_samples] = {}; > > while (true) { > int16_t *samples = buffer; > for (int sample_i = 0; sample_i < num_base_samples; sample_i++) { > *samples++ = 0x33; > *samples++ = 0x33; > } > > struct snd_xferi transfer = {}; > transfer.buf = buffer; > transfer.frames = num_base_samples; > ioctl(fd, SNDRV_PCM_IOCTL_WRITEI_FRAMES, &transfer); > > // ioctl(fd, SNDRV_PCM_IOCTL_DRAIN); --> NECESSARY??? > // ioctl(fd, SNDRV_PCM_IOCTL_PREPARE); --> NECESSARY??? > } On the first iteration of SNDRV_PCM_IOCTL_WRITEI_FRAMES I get no error. All subsequent iterations, I get Broken Pipe error. So, to counteract this at the end of each frame I call SNDRV_PCM_IOCTL_DRAIN and SNDRV_PCM_IOCTL_PREPARE. This removes the Broken Pipe error however slows the program down by half and no sound is heard. How best to solve this issue? Thanks again -- Ryan McClue, Sydney

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[PATCH v2 00/12] Add support for on demand pipeline setup/destroy
by Daniel Baluta 27 Sep '21

27 Sep '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries implements the new feature to setup/teardown pipeline as needed when a PCM is open/closed. This only only support single cores, with support for multicore arriving with a future patchseries. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 Changes since v1: - added my own Signed-off-by tag. Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 176 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1034 insertions(+), 538 deletions(-) -- 2.27.0

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