From: Mark Brown broonie@opensource.wolfsonmicro.com

Most devices accept data in formats that don't correspond directly to their internal format. ALSA allows us to set a msbits constraint which tells userspace about this in case it finds it useful (for example, in order to avoid wasting effort dithering bits that will be ignored when raising the sample size of data) so provide a mechanism for drivers to specify the number of bits that are actually significant on a DAI and add the appropriate constraints along with all the others.

This is done slightly awkwardly as the constraint is specified per sample size - we loop over every possible sample size, including ones that the device doesn't support and including ones that have fewer bits than are actually used, but this is harmless as the upper layers do the right thing in these cases.

Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 36 ++++++++++++++++++++++++++++++++++++ 2 files changed, 37 insertions(+), 0 deletions(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h index 0992dff..55381fc 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -505,6 +505,7 @@ struct snd_soc_pcm_stream { unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ + unsigned int sig_bits; /* number of bits of content */ };

/* SoC audio ops */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cdc860a..8bb1793 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -63,6 +63,39 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, }

/* + * List of sample sizes that might go over the bus for parameter + * application. There ought to be a wildcard sample size for things + * like the DAC/ADC resolution to use but there isn't right now. + */ +static int sample_sizes[] = { + 8, 16, 24, 32, +}; + +static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret, i, bits; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + bits = dai->driver->playback.sig_bits; + else + bits = dai->driver->capture.sig_bits; + + if (!bits) + return; + + for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { + ret = snd_pcm_hw_constraint_msbits(substream->runtime, + 0, sample_sizes[i], + bits); + if (ret != 0) + dev_warn(dai->dev, + "Failed to set MSB %d/%d: %d\n", + bits, sample_sizes[i], ret); + } +} + +/* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls * startup for the cpu DAI, platform, machine and codec DAI. @@ -187,6 +220,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; }

+ soc_pcm_apply_msb(substream, codec_dai); + soc_pcm_apply_msb(substream, cpu_dai); + /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active) { ret = soc_pcm_apply_symmetry(substream, cpu_dai);

In order to avoid confusing the applications with msbits bigger than the selected sample size apply the msbits constraint only to sample seze bigger than the requested msbits.

Signed-off-by: Peter Ujfalusi peter.ujfalusi@ti.com --- sound/soc/soc-pcm.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8bb1793..e5e804d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -68,7 +68,7 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, * like the DAC/ADC resolution to use but there isn't right now. */ static int sample_sizes[] = { - 8, 16, 24, 32, + 32, 24, 16, 8, };

static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, @@ -85,6 +85,8 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, return;

for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { + if (sample_sizes[i] <= bits) + break; ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0, sample_sizes[i], bits);

Core can set the msbits constraint in behalf of the dai.

Signed-off-by: Peter Ujfalusi peter.ujfalusi@ti.com --- sound/soc/omap/omap-dmic.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-)

diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 0855c1c..4dcb5a7 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -113,12 +113,10 @@ static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,

mutex_lock(&dmic->mutex);

- if (!dai->active) { - snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + if (!dai->active) dmic->active = 1; - } else { + else ret = -EBUSY; - }

mutex_unlock(&dmic->mutex);

@@ -445,6 +443,7 @@ static struct snd_soc_dai_driver omap_dmic_dai = { .channels_max = 6, .rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, }, .ops = &omap_dmic_dai_ops, };

Core can set the msbits constraint in behalf of the dai.

Signed-off-by: Peter Ujfalusi peter.ujfalusi@ti.com --- sound/soc/codecs/tlv320dac33.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-)

diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f0aad26..21ccf0a 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -806,8 +806,6 @@ static int dac33_startup(struct snd_pcm_substream *substream, /* Stream started, save the substream pointer */ dac33->substream = substream;

- snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); - return 0; }

@@ -1516,6 +1514,7 @@ static struct snd_soc_dai_driver dac33_dai = { .channels_max = 2, .rates = DAC33_RATES, .formats = DAC33_FORMATS,}, + .sig_bits = 24, .ops = &dac33_dai_ops, };