December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH - extplug refinement from hw_params 1/1] pcm: extplug: Update slave PCM if extplug slave params were changed
by twischer＠de.adit-jv.com 07 Dec '18

07 Dec '18

From: Timo Wischer <twischer(a)de.adit-jv.com> from extplug hw_params() callback function. This feature is for example required in the following use case: - A filter plugin supports only 1-8 channels. Therefore it calls snd_pcm_extplug_set_slave_param_minmax(SND_PCM_EXTPLUG_HW_CHANNELS, 1, 8) to provide the salve PCM parameters limited to 1-8 channels to the user application - The user application requests 2 channels. But in this case the slave PCM will be configured with 1 channel because it is the first valid configuration - To update the salve PCM snd_pcm_extplug_set_slave_param_minmax(SND_PCM_EXTPLUG_HW_CHANNELS, 2, 2) could be called from the extplug hw_params() callback function Without this patch the call to snd_pcm_extplug_set_slave_param_minmax() would not have any effect. With this patch the slave device will also be configured with 2 channels and no down mixing has to be done by the extplug. This new feature can also be used for some specific dependencies. For example a down mixing extplug supports only 2to1 and 4to2 channel down mixing. Therefore the salve PCM has to be opened with 2 channels if the user application requests 4 channels. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_extplug.c b/src/pcm/pcm_extplug.c index 1f887c5..44afadb 100644 --- a/src/pcm/pcm_extplug.c +++ b/src/pcm/pcm_extplug.c @@ -26,6 +26,7 @@ * */ +#include <stdbool.h> #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_extplug.h" @@ -43,6 +44,7 @@ typedef struct snd_pcm_extplug_priv { snd_pcm_extplug_t *data; struct snd_ext_parm params[SND_PCM_EXTPLUG_HW_PARAMS]; struct snd_ext_parm sparams[SND_PCM_EXTPLUG_HW_PARAMS]; + bool sparams_changed; } extplug_priv_t; static const int hw_params_type[SND_PCM_EXTPLUG_HW_PARAMS] = { @@ -60,18 +62,49 @@ static const unsigned int excl_parbits[SND_PCM_EXTPLUG_HW_PARAMS] = { SND_PCM_HW_PARBIT_FRAME_BITS), }; +static bool snd_ext_parm_equal(const struct snd_ext_parm * const a, + const struct snd_ext_parm * const b) +{ + if (a != b || a->active != b->active || a->num_list != b->num_list) + return false; + + if (a->num_list > 0) { + for (unsigned int i = 0; i++; i < a->num_list) { + if (a->list[i] != b->list[i]) + return false; + } + } else { + if (a->min != b->min || a->max != b->max) + return false; + } + + return true; +} + +static int snd_ext_parm_set(struct snd_ext_parm *parm, unsigned int min, + unsigned int max, unsigned int num_list, + unsigned int *list) +{ + const struct snd_ext_parm new_parm = { + .min = min, + .max = max, + .num_list = num_list, + .list = list, + .active = 1, + }; + const int changed = snd_ext_parm_equal(parm, &new_parm) ? 0 : 1; + + free(parm->list); + *parm = new_parm; + return changed; +} + /* * set min/max values for the given parameter */ int snd_ext_parm_set_minmax(struct snd_ext_parm *parm, unsigned int min, unsigned int max) { - parm->num_list = 0; - free(parm->list); - parm->list = NULL; - parm->min = min; - parm->max = max; - parm->active = 1; - return 0; + return snd_ext_parm_set(parm, min, max, 0, NULL); } /* @@ -92,11 +125,7 @@ int snd_ext_parm_set_list(struct snd_ext_parm *parm, unsigned int num_list, cons memcpy(new_list, list, sizeof(*new_list) * num_list); qsort(new_list, num_list, sizeof(*new_list), val_compar); - free(parm->list); - parm->num_list = num_list; - parm->list = new_list; - parm->active = 1; - return 0; + return snd_ext_parm_set(parm, 0, 0, num_list, new_list); } void snd_ext_parm_clear(struct snd_ext_parm *parm) @@ -291,29 +320,37 @@ static int snd_pcm_extplug_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params */ static int snd_pcm_extplug_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) { - extplug_priv_t *ext = pcm->private_data; snd_pcm_t *slave = ext->plug.gen.slave; - int err = snd_pcm_hw_params_slave(pcm, params, - snd_pcm_extplug_hw_refine_cchange, - snd_pcm_extplug_hw_refine_sprepare, - snd_pcm_extplug_hw_refine_schange, - snd_pcm_generic_hw_params); - if (err < 0) - return err; - ext->data->slave_format = slave->format; - ext->data->slave_subformat = slave->subformat; - ext->data->slave_channels = slave->channels; - ext->data->rate = slave->rate; - INTERNAL(snd_pcm_hw_params_get_format)(params, &ext->data->format); - INTERNAL(snd_pcm_hw_params_get_subformat)(params, &ext->data->subformat); - INTERNAL(snd_pcm_hw_params_get_channels)(params, &ext->data->channels); - - if (ext->data->callback->hw_params) { - err = ext->data->callback->hw_params(ext->data, params); + + do { + ext->sparams_changed = false; + + int err = snd_pcm_hw_params_slave(pcm, params, + snd_pcm_extplug_hw_refine_cchange, + snd_pcm_extplug_hw_refine_sprepare, + snd_pcm_extplug_hw_refine_schange, + snd_pcm_generic_hw_params); if (err < 0) return err; - } + ext->data->slave_format = slave->format; + ext->data->slave_subformat = slave->subformat; + ext->data->slave_channels = slave->channels; + ext->data->rate = slave->rate; + INTERNAL(snd_pcm_hw_params_get_format)( + params, &ext->data->format); + INTERNAL(snd_pcm_hw_params_get_subformat)( + params, &ext->data->subformat); + INTERNAL(snd_pcm_hw_params_get_channels)( + params, &ext->data->channels); + + if (ext->data->callback->hw_params) { + err = ext->data->callback->hw_params(ext->data, params); + if (err < 0) + return err; + } + } while (ext->sparams_changed); + return 0; } @@ -411,6 +448,7 @@ static void clear_ext_params(extplug_priv_t *ext) int i; for (i = 0; i < SND_PCM_EXTPLUG_HW_PARAMS; i++) { snd_ext_parm_clear(&ext->params[i]); + ext->sparams_changed |= ext->sparams[i].active; snd_ext_parm_clear(&ext->sparams[i]); } } @@ -637,7 +675,9 @@ parameters are sample format and channels. To define the constraints of the slave PCM configuration, use either #snd_pcm_extplug_set_slave_param_minmax() and #snd_pcm_extplug_set_slave_param_list(). The arguments are as same -as former functions. +as former functions. Both functions can also be called from the hw_params +callback. This can be used to further limit the configuration of the slave +device depending on the configuration of the user device. To clear the parameter constraints, call #snd_pcm_extplug_params_reset() function. @@ -768,11 +808,17 @@ void snd_pcm_extplug_params_reset(snd_pcm_extplug_t *extplug) int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, unsigned int num_list, const unsigned int *list) { extplug_priv_t *ext = extplug->pcm->private_data; + int changed = 0; + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; } - return snd_ext_parm_set_list(&ext->sparams[type], num_list, list); + changed = snd_ext_parm_set_list(&ext->sparams[type], num_list, list); + if (changed > 0) + ext->sparams_changed = true; + + return changed; } /** @@ -790,6 +836,8 @@ int snd_pcm_extplug_set_slave_param_list(snd_pcm_extplug_t *extplug, int type, u int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, unsigned int min, unsigned int max) { extplug_priv_t *ext = extplug->pcm->private_data; + int changed = 0; + if (type < 0 || type >= SND_PCM_EXTPLUG_HW_PARAMS) { SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; @@ -798,7 +846,11 @@ int snd_pcm_extplug_set_slave_param_minmax(snd_pcm_extplug_t *extplug, int type, SNDERR("EXTPLUG: invalid parameter type %d", type); return -EINVAL; } - return snd_ext_parm_set_minmax(&ext->sparams[type], min, max); + changed = snd_ext_parm_set_minmax(&ext->sparams[type], min, max); + if (changed > 0) + ext->sparams_changed = true; + + return changed; } /** -- 2.7.4

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[alsa-devel] Applied "ASoC: max9867: Remove useless assignment" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Remove useless assignment 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 8efc1afd7e0570d9bb37f5127f0742304a8ad2fc Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:18:17 +0100 Subject: [PATCH] ASoC: max9867: Remove useless assignment ret is assigned later, no need to initialize it. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 1cda54b59854..1b155f40b106 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -476,7 +476,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max9867_priv *max9867; - int ret = 0, reg; + int ret, reg; max9867 = devm_kzalloc(&i2c->dev, sizeof(*max9867), GFP_KERNEL); if (!max9867) @@ -497,10 +497,8 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "device revision: %x\n", reg); ret = devm_snd_soc_register_component(&i2c->dev, &max9867_component, max9867_dai, ARRAY_SIZE(max9867_dai)); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret); - return ret; - } return ret; } -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2 0/5] max9867 driver update
by Ladislav Michl 07 Dec '18

07 Dec '18

Hi Mark, this is yet another round of driver update. Tested on AT91SAM9G20 based board with codec in master mode. All implemeted controls do work now. Comments from previous serie taken into account. Ladislav Michl (5): ASoC: max9867: Remove useless assignment ASoC: max9867: Fix power management ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix signal paths ASoC: max9867: Add copyright and module author

2 9

[alsa-devel] Applied "ASoC: max9867: Calculate LRCLK divider" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Calculate LRCLK divider 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 715ee1916b95bfd0773487397a0354ad9a4660e6 Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:04 +0100 Subject: [PATCH] ASoC: max9867: Calculate LRCLK divider Drop "Common NI Values Table" and calculate LRCLK divider, then add allowed rate constraints based on master clock frequency. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 207 ++++++++++++++++++------------------- sound/soc/codecs/max9867.h | 31 ++---- 2 files changed, 110 insertions(+), 128 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index c454de434d76..faec094161c1 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -126,124 +126,106 @@ static const struct snd_soc_dapm_route max9867_audio_map[] = { {"LINE_IN", NULL, "Right Line"}, }; -enum rates { - pcm_rate_8, pcm_rate_16, pcm_rate_24, - pcm_rate_32, pcm_rate_44, - pcm_rate_48, max_pcm_rate, +static const unsigned int max9867_rates_44k1[] = { + 11025, 22050, 44100, }; -static const struct ni_div_rates { - u32 mclk; - u16 ni[max_pcm_rate]; -} ni_div[] = { - {11289600, {0x116A, 0x22D4, 0x343F, 0x45A9, 0x6000, 0x687D} }, - {12000000, {0x1062, 0x20C5, 0x3127, 0x4189, 0x5A51, 0x624E} }, - {12288000, {0x1000, 0x2000, 0x3000, 0x4000, 0x5833, 0x6000} }, - {13000000, {0x0F20, 0x1E3F, 0x2D5F, 0x3C7F, 0x535F, 0x5ABE} }, - {19200000, {0x0A3D, 0x147B, 0x1EB8, 0x28F6, 0x3873, 0x3D71} }, - {24000000, {0x1062, 0x20C5, 0x1893, 0x4189, 0x5A51, 0x624E} }, - {26000000, {0x0F20, 0x1E3F, 0x16AF, 0x3C7F, 0x535F, 0x5ABE} }, - {27000000, {0x0E90, 0x1D21, 0x15D8, 0x3A41, 0x5048, 0x5762} }, +static const struct snd_pcm_hw_constraint_list max9867_constraints_44k1 = { + .list = max9867_rates_44k1, + .count = ARRAY_SIZE(max9867_rates_44k1), }; -static inline int get_ni_value(int mclk, int rate) +static const unsigned int max9867_rates_48k[] = { + 8000, 16000, 32000, 48000, +}; + +static const struct snd_pcm_hw_constraint_list max9867_constraints_48k = { + .list = max9867_rates_48k, + .count = ARRAY_SIZE(max9867_rates_48k), +}; + +struct max9867_priv { + struct regmap *regmap; + const struct snd_pcm_hw_constraint_list *constraints; + unsigned int sysclk, pclk; + bool master, dsp_a; +}; + +static int max9867_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { - int i, ret = 0; + struct max9867_priv *max9867 = + snd_soc_component_get_drvdata(dai->component); - /* find the closest rate index*/ - for (i = 0; i < ARRAY_SIZE(ni_div); i++) { - if (ni_div[i].mclk >= mclk) - break; - } - if (i == ARRAY_SIZE(ni_div)) - return -EINVAL; + if (max9867->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, max9867->constraints); - switch (rate) { - case 8000: - return ni_div[i].ni[pcm_rate_8]; - case 16000: - return ni_div[i].ni[pcm_rate_16]; - case 32000: - return ni_div[i].ni[pcm_rate_32]; - case 44100: - return ni_div[i].ni[pcm_rate_44]; - case 48000: - return ni_div[i].ni[pcm_rate_48]; - default: - pr_err("%s wrong rate %d\n", __func__, rate); - ret = -EINVAL; - } - return ret; + return 0; } static int max9867_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + int value; + unsigned long int rate, ratio; struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - unsigned int ni_h, ni_l; - int value; + unsigned int ni = DIV_ROUND_CLOSEST_ULL(96ULL * 0x10000 * params_rate(params), + max9867->pclk); - value = get_ni_value(max9867->sysclk, params_rate(params)); - if (value < 0) - return value; - - ni_h = (0xFF00 & value) >> 8; - ni_l = 0x00FF & value; /* set up the ni value */ regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_NI_HIGH_MASK, ni_h); + MAX9867_NI_HIGH_MASK, (0xFF00 & ni) >> 8); regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_NI_LOW_MASK, ni_l); - if (!max9867->master) { - /* - * digital pll locks on to any externally supplied LRCLK signal - * and also enable rapid lock mode. - */ - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, - MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); - regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, - MAX9867_PLL, MAX9867_PLL); - } else { - unsigned long int bclk_rate, pclk_bclk_ratio; - int bclk_value; - - bclk_rate = params_rate(params) * 2 * params_width(params); - pclk_bclk_ratio = max9867->pclk/bclk_rate; - switch (params_width(params)) { - case 8: - case 16: - switch (pclk_bclk_ratio) { - case 2: - bclk_value = MAX9867_IFC1B_PCLK_2; - break; - case 4: - bclk_value = MAX9867_IFC1B_PCLK_4; - break; + MAX9867_NI_LOW_MASK, 0x00FF & ni); + if (max9867->master) { + if (max9867->dsp_a) { + value = MAX9867_IFC1B_48X; + } else { + rate = params_rate(params) * 2 * params_width(params); + ratio = max9867->pclk / rate; + switch (params_width(params)) { case 8: - bclk_value = MAX9867_IFC1B_PCLK_8; - break; case 16: - bclk_value = MAX9867_IFC1B_PCLK_16; + switch (ratio) { + case 2: + value = MAX9867_IFC1B_PCLK_2; + break; + case 4: + value = MAX9867_IFC1B_PCLK_4; + break; + case 8: + value = MAX9867_IFC1B_PCLK_8; + break; + case 16: + value = MAX9867_IFC1B_PCLK_16; + break; + default: + return -EINVAL; + } + break; + case 24: + value = MAX9867_IFC1B_48X; + break; + case 32: + value = MAX9867_IFC1B_64X; break; default: - dev_err(component->dev, - "unsupported sampling rate\n"); return -EINVAL; } - break; - case 24: - bclk_value = MAX9867_IFC1B_24BIT; - break; - case 32: - bclk_value = MAX9867_IFC1B_32BIT; - break; - default: - dev_err(component->dev, "unsupported sampling rate\n"); - return -EINVAL; } regmap_update_bits(max9867->regmap, MAX9867_IFC1B, - MAX9867_IFC1B_BCLK_MASK, bclk_value); + MAX9867_IFC1B_BCLK_MASK, value); + } else { + /* + * digital pll locks on to any externally supplied LRCLK signal + * and also enable rapid lock mode. + */ + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKLOW, + MAX9867_RAPID_LOCK, MAX9867_RAPID_LOCK); + regmap_update_bits(max9867->regmap, MAX9867_AUDIOCLKHIGH, + MAX9867_PLL, MAX9867_PLL); } return 0; } @@ -285,8 +267,16 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, freq); return -EINVAL; } - value = value << MAX9867_PSCLK_SHIFT; + if (freq % 48000 == 0) + max9867->constraints = &max9867_constraints_48k; + else if (freq % 44100 == 0) + max9867->constraints = &max9867_constraints_44k1; + else + dev_warn(component->dev, + "Unable to set exact rate with %uHz clock frequency\n", + freq); max9867->sysclk = freq; + value = value << MAX9867_PSCLK_SHIFT; /* exact integer mode is not supported */ value &= ~MAX9867_FREQ_MASK; regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, @@ -299,16 +289,17 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - u8 iface1A = 0, iface1B = 0; + u8 iface1A, iface1B; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - max9867->master = 1; - iface1A |= MAX9867_MASTER; + max9867->master = true; + iface1A = MAX9867_MASTER; + iface1B = MAX9867_IFC1B_48X; break; case SND_SOC_DAIFMT_CBS_CFS: - max9867->master = 0; - iface1A &= ~MAX9867_MASTER; + max9867->master = false; + iface1A = iface1B = 0; break; default: return -EINVAL; @@ -316,9 +307,11 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + max9867->dsp_a = false; iface1A |= MAX9867_I2S_DLY; break; case SND_SOC_DAIFMT_DSP_A: + max9867->dsp_a = true; iface1A |= MAX9867_TDM_MODE | MAX9867_SDOUT_HIZ; break; default: @@ -344,20 +337,18 @@ static int max9867_dai_set_fmt(struct snd_soc_dai *codec_dai, regmap_write(max9867->regmap, MAX9867_IFC1A, iface1A); regmap_write(max9867->regmap, MAX9867_IFC1B, iface1B); + return 0; } static const struct snd_soc_dai_ops max9867_dai_ops = { - .set_fmt = max9867_dai_set_fmt, .set_sysclk = max9867_set_dai_sysclk, + .set_fmt = max9867_dai_set_fmt, .digital_mute = max9867_mute, - .hw_params = max9867_dai_hw_params, + .startup = max9867_startup, + .hw_params = max9867_dai_hw_params, }; -#define MAX9867_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -#define MAX9867_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) - static struct snd_soc_dai_driver max9867_dai[] = { { .name = "max9867-aif1", @@ -365,15 +356,15 @@ static struct snd_soc_dai_driver max9867_dai[] = { .stream_name = "HiFi Playback", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "HiFi Capture", .channels_min = 2, .channels_max = 2, - .rates = MAX9867_RATES, - .formats = MAX9867_FORMATS, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &max9867_dai_ops, .symmetric_rates = 1, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index d9170850c96e..7f037ab701a5 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -26,13 +26,11 @@ #define MAX9867_PSCLK_10_20 0x1 #define MAX9867_PSCLK_20_40 0x2 #define MAX9867_PSCLK_40_60 0x3 -#define MAX9867_AUDIOCLKHIGH 0x06 -#define MAX9867_NI_HIGH_WIDTH 0x7 -#define MAX9867_NI_HIGH_MASK 0x7F -#define MAX9867_NI_LOW_MASK 0x7F -#define MAX9867_NI_LOW_SHIFT 0x1 -#define MAX9867_PLL (1<<7) -#define MAX9867_AUDIOCLKLOW 0x07 +#define MAX9867_AUDIOCLKHIGH 0x06 +#define MAX9867_NI_HIGH_MASK 0x7F +#define MAX9867_NI_LOW_MASK 0xFE +#define MAX9867_PLL (1<<7) +#define MAX9867_AUDIOCLKLOW 0x07 #define MAX9867_RAPID_LOCK 0x01 #define MAX9867_IFC1A 0x08 #define MAX9867_MASTER (1<<7) @@ -43,12 +41,12 @@ #define MAX9867_BCI_MODE (1<<5) #define MAX9867_IFC1B 0x09 #define MAX9867_IFC1B_BCLK_MASK 7 -#define MAX9867_IFC1B_32BIT 0x01 -#define MAX9867_IFC1B_24BIT 0x02 -#define MAX9867_IFC1B_PCLK_2 4 -#define MAX9867_IFC1B_PCLK_4 5 -#define MAX9867_IFC1B_PCLK_8 6 -#define MAX9867_IFC1B_PCLK_16 7 +#define MAX9867_IFC1B_64X 0x01 +#define MAX9867_IFC1B_48X 0x02 +#define MAX9867_IFC1B_PCLK_2 0x04 +#define MAX9867_IFC1B_PCLK_4 0x05 +#define MAX9867_IFC1B_PCLK_8 0x06 +#define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a #define MAX9867_DACGAIN 0x0b #define MAX9867_DACLEVEL 0x0c @@ -72,11 +70,4 @@ #define MAX9867_CACHEREGNUM 10 -/* codec private data */ -struct max9867_priv { - struct regmap *regmap; - unsigned int sysclk; - unsigned int pclk; - unsigned int master; -}; #endif -- 2.19.0.rc2

1 0

[alsa-devel] Applied "ASoC: max9867: Fix signal paths" to the asoc tree
by Mark Brown 07 Dec '18

07 Dec '18

The patch ASoC: max9867: Fix signal paths 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 bc2610a6c4c92f0bf6bbf4c6641407536f30daaf Mon Sep 17 00:00:00 2001 From: Ladislav Michl <ladis(a)linux-mips.org> Date: Tue, 4 Dec 2018 19:21:33 +0100 Subject: [PATCH] ASoC: max9867: Fix signal paths Sound capture and line bypass currently do not work as well as some mixer controls. Fix that by building proper audio paths and adjusting volume controls to match datasheet. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max9867.c | 188 ++++++++++++++++++++----------------- sound/soc/codecs/max9867.h | 8 +- 2 files changed, 102 insertions(+), 94 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index faec094161c1..7a08e2c70e17 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -23,107 +23,124 @@ static const char *const max9867_spmode[] = { "Stereo Single", "Mono Single", "Stereo Single Fast", "Mono Single Fast" }; -static const char *const max9867_sidetone_text[] = { - "None", "Left", "Right", "LeftRight", "LeftRightDiv2", -}; static const char *const max9867_filter_text[] = {"IIR", "FIR"}; static SOC_ENUM_SINGLE_DECL(max9867_filter, MAX9867_CODECFLTR, 7, max9867_filter_text); static SOC_ENUM_SINGLE_DECL(max9867_spkmode, MAX9867_MODECONFIG, 0, max9867_spmode); -static SOC_ENUM_SINGLE_DECL(max9867_sidetone, MAX9867_DACGAIN, 6, - max9867_sidetone_text); -static DECLARE_TLV_DB_SCALE(max9860_capture_tlv, -600, 200, 0); -static DECLARE_TLV_DB_SCALE(max9860_mic_tlv, 2000, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_left_tlv, -1200, 100, 1); -static DECLARE_TLV_DB_SCALE(max9860_adc_right_tlv, -1200, 100, 1); -static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max98088_micboost_tlv, - 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0), - 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_master_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-8600, 200, 1), + 3, 17, TLV_DB_SCALE_ITEM(-7800, 400, 0), + 18, 25, TLV_DB_SCALE_ITEM(-2000, 200, 0), + 26, 34, TLV_DB_SCALE_ITEM( -500, 100, 0), + 35, 40, TLV_DB_SCALE_ITEM( 350, 50, 0), +); +static DECLARE_TLV_DB_SCALE(max9867_mic_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_line_tlv, -600, 200, 0); +static DECLARE_TLV_DB_SCALE(max9867_adc_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dac_tlv, -1500, 100, 0); +static DECLARE_TLV_DB_SCALE(max9867_dacboost_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(max9867_micboost_tlv, + 0, 2, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + 3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0), ); static const struct snd_kcontrol_new max9867_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", MAX9867_LEFTVOL, - MAX9867_RIGHTVOL, 0, 63, 1), - SOC_DOUBLE_R_TLV("Capture Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, - 0, 15, 1, max9860_capture_tlv), - SOC_DOUBLE_R_TLV("Mic Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 0, 31, 1, max9860_mic_tlv), - SOC_DOUBLE_R_TLV("Mic Boost Volume", MAX9867_LEFTMICGAIN, - MAX9867_RIGHTMICGAIN, 5, 3, 0, max98088_micboost_tlv), - SOC_ENUM("Digital Sidetone Src", max9867_sidetone), - SOC_SINGLE("Sidetone Volume", MAX9867_DACGAIN, 0, 31, 1), - SOC_SINGLE("DAC Volume", MAX9867_DACLEVEL, 4, 3, 0), - SOC_SINGLE("DAC Attenuation", MAX9867_DACLEVEL, 0, 15, 1), - SOC_SINGLE_TLV("ADC Left Volume", MAX9867_ADCLEVEL, - 4, 15, 1, max9860_adc_left_tlv), - SOC_SINGLE_TLV("ADC Right Volume", MAX9867_ADCLEVEL, - 0, 15, 1, max9860_adc_right_tlv), + SOC_DOUBLE_R_TLV("Master Playback Volume", MAX9867_LEFTVOL, + MAX9867_RIGHTVOL, 0, 41, 1, max9867_master_tlv), + SOC_DOUBLE_R_TLV("Line Capture Volume", MAX9867_LEFTLINELVL, + MAX9867_RIGHTLINELVL, 0, 15, 1, max9867_line_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 0, 20, 1, max9867_mic_tlv), + SOC_DOUBLE_R_TLV("Mic Boost Capture Volume", MAX9867_LEFTMICGAIN, + MAX9867_RIGHTMICGAIN, 5, 4, 0, max9867_micboost_tlv), + SOC_SINGLE("Digital Sidetone Volume", MAX9867_SIDETONE, 0, 31, 1), + SOC_SINGLE_TLV("Digital Playback Volume", MAX9867_DACLEVEL, 0, 15, 1, + max9867_dac_tlv), + SOC_SINGLE_TLV("Digital Boost Playback Volume", MAX9867_DACLEVEL, 4, 3, 0, + max9867_dacboost_tlv), + SOC_DOUBLE_TLV("Digital Capture Volume", MAX9867_ADCLEVEL, 0, 4, 15, 1, + max9867_adc_tlv), SOC_ENUM("Speaker Mode", max9867_spkmode), SOC_SINGLE("Volume Smoothing Switch", MAX9867_MODECONFIG, 6, 1, 0), - SOC_SINGLE("ZCD Switch", MAX9867_MODECONFIG, 5, 1, 0), + SOC_SINGLE("Line ZC Switch", MAX9867_MODECONFIG, 5, 1, 0), SOC_ENUM("DSP Filter", max9867_filter), }; -static const char *const max9867_mux[] = {"None", "Mic", "Line", "Mic_Line"}; +/* Input mixer */ +static const struct snd_kcontrol_new max9867_input_mixer_controls[] = { + SOC_DAPM_DOUBLE("Line Capture Switch", MAX9867_INPUTCONFIG, 7, 5, 1, 0), + SOC_DAPM_DOUBLE("Mic Capture Switch", MAX9867_INPUTCONFIG, 6, 4, 1, 0), +}; + +/* Output mixer */ +static const struct snd_kcontrol_new max9867_output_mixer_controls[] = { + SOC_DAPM_DOUBLE_R("Line Bypass Switch", + MAX9867_LEFTLINELVL, MAX9867_RIGHTLINELVL, 6, 1, 1), +}; -static SOC_ENUM_SINGLE_DECL(max9867_mux_enum, - MAX9867_INPUTCONFIG, MAX9867_INPUT_SHIFT, - max9867_mux); +/* Sidetone mixer */ +static const struct snd_kcontrol_new max9867_sidetone_mixer_controls[] = { + SOC_DAPM_DOUBLE("Sidetone Switch", MAX9867_SIDETONE, 6, 7, 1, 0), +}; -static const struct snd_kcontrol_new max9867_dapm_mux_controls = - SOC_DAPM_ENUM("Route", max9867_mux_enum); +/* Line out switch */ +static const struct snd_kcontrol_new max9867_line_out_control = + SOC_DAPM_DOUBLE_R("Switch", + MAX9867_LEFTVOL, MAX9867_RIGHTVOL, 6, 1, 1); -static const struct snd_kcontrol_new max9867_left_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 6, 1, 0); -static const struct snd_kcontrol_new max9867_right_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_PWRMAN, 5, 1, 0); -static const struct snd_kcontrol_new max9867_line_dapm_control = - SOC_DAPM_SINGLE("Switch", MAX9867_LEFTLINELVL, 6, 1, 1); static const struct snd_soc_dapm_widget max9867_dapm_widgets[] = { - SND_SOC_DAPM_AIF_IN("DAI_OUT", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("Left DAC", NULL, MAX9867_PWRMAN, 3, 0), - SND_SOC_DAPM_DAC("Right DAC", NULL, MAX9867_PWRMAN, 2, 0), - SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_OUTPUT("HPOUT"), - - SND_SOC_DAPM_AIF_IN("DAI_IN", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture", MAX9867_PWRMAN, 1, 0), - SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture", MAX9867_PWRMAN, 0, 0), - SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, - &max9867_dapm_mux_controls), - - SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("Left Line", MAX9867_LEFTLINELVL, 6, 1, - &max9867_left_dapm_control), - SND_SOC_DAPM_SWITCH("Right Line", MAX9867_RIGTHLINELVL, 6, 1, - &max9867_right_dapm_control), - SND_SOC_DAPM_SWITCH("Line Mixer", SND_SOC_NOPM, 0, 0, - &max9867_line_dapm_control), - SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("LINL"), + SND_SOC_DAPM_INPUT("LINR"), + + SND_SOC_DAPM_PGA("Left Line Input", MAX9867_PWRMAN, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Line Input", MAX9867_PWRMAN, 5, 0, NULL, 0), + SND_SOC_DAPM_MIXER_NAMED_CTL("Input Mixer", SND_SOC_NOPM, 0, 0, + max9867_input_mixer_controls, + ARRAY_SIZE(max9867_input_mixer_controls)), + SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", MAX9867_PWRMAN, 1, 0), + SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", MAX9867_PWRMAN, 0, 0), + + SND_SOC_DAPM_MIXER("Digital", SND_SOC_NOPM, 0, 0, + max9867_sidetone_mixer_controls, + ARRAY_SIZE(max9867_sidetone_mixer_controls)), + SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", SND_SOC_NOPM, 0, 0, + max9867_output_mixer_controls, + ARRAY_SIZE(max9867_output_mixer_controls)), + SND_SOC_DAPM_DAC("DACL", "HiFi Playback", MAX9867_PWRMAN, 3, 0), + SND_SOC_DAPM_DAC("DACR", "HiFi Playback", MAX9867_PWRMAN, 2, 0), + SND_SOC_DAPM_SWITCH("Master Playback", SND_SOC_NOPM, 0, 0, + &max9867_line_out_control), + SND_SOC_DAPM_OUTPUT("LOUT"), + SND_SOC_DAPM_OUTPUT("ROUT"), }; static const struct snd_soc_dapm_route max9867_audio_map[] = { - {"Left DAC", NULL, "DAI_OUT"}, - {"Right DAC", NULL, "DAI_OUT"}, - {"Output Mixer", NULL, "Left DAC"}, - {"Output Mixer", NULL, "Right DAC"}, - {"HPOUT", NULL, "Output Mixer"}, - - {"Left ADC", NULL, "DAI_IN"}, - {"Right ADC", NULL, "DAI_IN"}, - {"Input Mixer", NULL, "Left ADC"}, - {"Input Mixer", NULL, "Right ADC"}, - {"Input Mux", "Line", "Input Mixer"}, - {"Input Mux", "Mic", "Input Mixer"}, - {"Input Mux", "Mic_Line", "Input Mixer"}, - {"Right Line", "Switch", "Input Mux"}, - {"Left Line", "Switch", "Input Mux"}, - {"LINE_IN", NULL, "Left Line"}, - {"LINE_IN", NULL, "Right Line"}, + {"Left Line Input", NULL, "LINL"}, + {"Right Line Input", NULL, "LINR"}, + {"Input Mixer", "Mic Capture Switch", "MICL"}, + {"Input Mixer", "Mic Capture Switch", "MICR"}, + {"Input Mixer", "Line Capture Switch", "Left Line Input"}, + {"Input Mixer", "Line Capture Switch", "Right Line Input"}, + {"ADCL", NULL, "Input Mixer"}, + {"ADCR", NULL, "Input Mixer"}, + + {"Digital", "Sidetone Switch", "ADCL"}, + {"Digital", "Sidetone Switch", "ADCR"}, + {"DACL", NULL, "Digital"}, + {"DACR", NULL, "Digital"}, + + {"Output Mixer", "Line Bypass Switch", "Left Line Input"}, + {"Output Mixer", "Line Bypass Switch", "Right Line Input"}, + {"Output Mixer", NULL, "DACL"}, + {"Output Mixer", NULL, "DACR"}, + {"Master Playback", "Switch", "Output Mixer"}, + {"LOUT", NULL, "Master Playback"}, + {"ROUT", NULL, "Master Playback"}, }; static const unsigned int max9867_rates_44k1[] = { @@ -235,13 +252,8 @@ static int max9867_mute(struct snd_soc_dai *dai, int mute) struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); - if (mute) - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, MAX9867_DAC_MUTE_MASK); - else - regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, - MAX9867_DAC_MUTE_MASK, 0); - return 0; + return regmap_update_bits(max9867->regmap, MAX9867_DACLEVEL, + 1 << 6, !!mute << 6); } static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -464,8 +476,8 @@ static const struct reg_default max9867_reg[] = { { 0x0B, 0x00 }, { 0x0C, 0x00 }, { 0x0D, 0x00 }, - { 0x0E, 0x00 }, - { 0x0F, 0x00 }, + { 0x0E, 0x40 }, + { 0x0F, 0x40 }, { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h index 7f037ab701a5..2277798291a1 100644 --- a/sound/soc/codecs/max9867.h +++ b/sound/soc/codecs/max9867.h @@ -48,20 +48,16 @@ #define MAX9867_IFC1B_PCLK_8 0x06 #define MAX9867_IFC1B_PCLK_16 0x07 #define MAX9867_CODECFLTR 0x0a -#define MAX9867_DACGAIN 0x0b +#define MAX9867_SIDETONE 0x0b #define MAX9867_DACLEVEL 0x0c -#define MAX9867_DAC_MUTE_SHIFT 0x6 -#define MAX9867_DAC_MUTE_WIDTH 0x1 -#define MAX9867_DAC_MUTE_MASK (0x1<<MAX9867_DAC_MUTE_SHIFT) #define MAX9867_ADCLEVEL 0x0d #define MAX9867_LEFTLINELVL 0x0e -#define MAX9867_RIGTHLINELVL 0x0f +#define MAX9867_RIGHTLINELVL 0x0f #define MAX9867_LEFTVOL 0x10 #define MAX9867_RIGHTVOL 0x11 #define MAX9867_LEFTMICGAIN 0x12 #define MAX9867_RIGHTMICGAIN 0x13 #define MAX9867_INPUTCONFIG 0x14 -#define MAX9867_INPUT_SHIFT 0x6 #define MAX9867_MICCONFIG 0x15 #define MAX9867_MODECONFIG 0x16 #define MAX9867_PWRMAN 0x17 -- 2.19.0.rc2

1 0

[alsa-devel] [PATCH v2] ASoC: max9867: Add copyright and module author
by Ladislav Michl 07 Dec '18

07 Dec '18

Driver rewritten, assign copyright notice and change module author as original one remains silent and I want to be notified about bugs. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: -v2: convert entire comment block to C++ style sound/soc/codecs/max9867.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 7a08e2c70e17..8600c5439e1e 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -1,12 +1,10 @@ -/* - * max9867.c -- max9867 ALSA SoC Audio driver - * - * Copyright 2013-15 Maxim Integrated Products - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// MAX9867 ALSA SoC codec driver +// +// Copyright 2013-2015 Maxim Integrated Products +// Copyright 2018 Ladislav Michl <ladis(a)linux-mips.org> +// #include <linux/delay.h> #include <linux/i2c.h> @@ -551,6 +549,6 @@ static struct i2c_driver max9867_i2c_driver = { module_i2c_driver(max9867_i2c_driver); -MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); -MODULE_DESCRIPTION("ALSA SoC MAX9867 driver"); +MODULE_AUTHOR("Ladislav Michl <ladis(a)linux-mips.org>"); +MODULE_DESCRIPTION("ASoC MAX9867 driver"); MODULE_LICENSE("GPL"); -- 2.20.0.rc2

2 1

Re: [alsa-devel] HDA headset button support
by Takashi Iwai 07 Dec '18

07 Dec '18

On Wed, 28 Nov 2018 13:03:58 +0100, Kailang wrote: > > Hi Takashi, > > >I'd rather like to extend snd_hda_jack_add_kctl() with the key > >support (maybe adding a new function). You can pass there the > >combination of SND_JACK_BTN_X with KEY_XXX in an array. > > Do you mean create new function like below? > snd_hda_jack_add_key_kctl() > > It couldn't use snd_hda_jack_add_kctl() directly. > Because it had report pin state and get type from NID. > > type = get_input_jack_type(codec, nid); > ..... > ..... > state = snd_hda_jack_detect(codec, nid); > snd_jack_report(jack->jack, state ? jack->type : 0); > > So, to create a new function in hda_jack.c was better. Well, this seems more deeper than I thought. Below is my quick attempt to add the infrastructure. This adds more flexibility but still easiness for other possible button implementations, hopefully. Could you check the series below? thanks, Takashi

2 10

[alsa-devel] [PATCH 0/4] alsa-utils: axfer: add help texts
by GitHub pull_request - opened 07 Dec '18

07 Dec '18

alsa-project/alsa-utils pull request #7 was opened from takaswie: This patchset adds help texts for command entry, list and transfer subcommands. In this time, contents for each backend of transfer subcommand is not added yet. Unfortunately, execution of aliases (arecord/aplay) can print help text for command text only. This bug will be fixed in future commits. Additionally, due to program design, execution of the aliases cannot print help text of list subcommand. This is unavoidable loss of backward compatibility. ``` Takashi Sakamoto (4): axfer: print help text of command entry axfer: print help text of list subcommand axfer: print help text of transfer subcommand just for common options axfer: enable each backend to print own help axfer/main.c | 12 +++++++++++- axfer/subcmd-list.c | 9 ++++++++- axfer/xfer-libasound.c | 6 ++++++ axfer/xfer-libffado.c | 6 ++++++ axfer/xfer-options.c | 35 +++++++++++++++++++++++++++++++++++ axfer/xfer.c | 5 +++++ axfer/xfer.h | 2 ++ 7 files changed, 73 insertions(+), 2 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/7 Patch URL : https://github.com/alsa-project/alsa-utils/pull/7.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] ALC700 headphone no sound issue
by Kailang 07 Dec '18

07 Dec '18

Hi Takashi, In Intel ENG board with ALC700. It had headphone no sound issue. If it plugged headphone or headset into the jack, then do the reboot, it will have a chance to cause headphone no sound. It just need to run the headphone mode procedure after boot time. The issue will be fixed. It also suitable for ALC234 ALC274 and ALC294. BR, Kailang

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[alsa-devel] [GIT PULL] soundwire updates
by Vinod Koul 07 Dec '18

07 Dec '18

Hey Greg, Please pull to receive a single commit for soundwire tree for this cycle. Am off next week so sending this bit early The following changes since commit 651022382c7f8da46cb4872a545ee1da6d097d2a: Linux 4.20-rc1 (2018-11-04 15:37:52 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-4.21-rc1 for you to fetch changes up to b1635596860dd3229645e41276f6c83ca788e4c1: soundwire: intel: constify snd_soc_dai_ops structures (2018-11-12 15:36:06 +0530) ---------------------------------------------------------------- soundwire updates for 4.21-rc1 A single change: - constify snd_soc_dai_ops struct by Julia ---------------------------------------------------------------- Julia Lawall (1): soundwire: intel: constify snd_soc_dai_ops structures drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- ~Vinod

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