From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 16 Jan 2012 18:38:51 +0000 (+0000)
Subject: ASoC: Allow drivers to specify how many bits are significant on a DAI
X-Git-Tag: firefly_0821_release~3680^2~2691^2~52^2~72
X-Git-Url: http://demsky.eecs.uci.edu/git/?a=commitdiff_plain;h=58ba9b25454fe9b6ded804f69cb7ed4500b685fc;p=firefly-linux-kernel-4.4.55.git

ASoC: Allow drivers to specify how many bits are significant on a DAI

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>
Acked-by: Liam Girdwood <lrg@ti.com>
---

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 0992dff55959..55381fca6e0d 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 cdc860a5ff37..8bb17937d59a 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -62,6 +62,39 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+/*
+ * 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
@@ -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);