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authorStefan Agner <stefan.agner@toradex.com>2016-02-24 13:23:40 -0800
committerStefan Agner <stefan.agner@toradex.com>2016-02-25 15:13:59 -0800
commit0fe93c78e3975f1f69011b02a83deb496d5d9bb7 (patch)
tree67bf3c48b8b3c56ba2c4c0d1db5522231d82f6be /sound
parent7051acc9f1e1d7a8cba32b384f151d4b010fbaf1 (diff)
ASoC: fsl_sai_ac97: add audio capture support
This adds initial audio capture support. A capturing sampling rate between 8 to 48kHz is supported. Signed-off-by: Stefan Agner <stefan.agner@toradex.com>
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/fsl/fsl_sai_ac97.c61
1 files changed, 58 insertions, 3 deletions
diff --git a/sound/soc/fsl/fsl_sai_ac97.c b/sound/soc/fsl/fsl_sai_ac97.c
index 669107b8fc2b..8adaab3417e6 100644
--- a/sound/soc/fsl/fsl_sai_ac97.c
+++ b/sound/soc/fsl/fsl_sai_ac97.c
@@ -238,8 +238,58 @@ static void fsl_dma_tx_complete(void *arg)
static void fsl_dma_rx_complete(void *arg)
{
struct fsl_sai_ac97 *sai = arg;
+ struct ac97_rx *aclink;
+ struct imx_pcm_runtime_data *iprtd = sai->iprtd;
+ struct dma_tx_state state;
+ enum dma_status status;
+ int bufid;
+ int i;
async_tx_ack(sai->dma_rx_desc);
+
+ status = dmaengine_tx_status(sai->dma_rx_chan, sai->dma_rx_cookie, &state);
+
+ /* Calculate the id of the running buffer */
+ if (state.residue % SAI_AC97_RBUF_SIZE == 0)
+ bufid = 4 - (state.residue / SAI_AC97_RBUF_SIZE);
+ else
+ bufid = 3 - (state.residue / SAI_AC97_RBUF_SIZE);
+
+ /* Calculate the id of the last processed buffer */
+ bufid = (bufid + 3) % 4;
+
+ /* First frame of the just completed buffer... */
+ aclink = (struct ac97_rx *)(sai->rx_buf.area + (bufid * SAI_AC97_RBUF_SIZE));
+
+ if (iprtd != NULL && atomic_read(&iprtd->capturing))
+ {
+ struct snd_dma_buffer *buf = &iprtd->substream->dma_buffer;
+ u16 *ptr = (u16 *)buf->area;
+
+ /*
+ * Loop through all AC97 frames, but only some might have data:
+ * Depending on bit rate, the valid flag might not be set for
+ * all frames (see AC97 VBR specification)
+ */
+ for (i = 0; i < SAI_AC97_RBUF_FRAMES; i++, aclink++) {
+ if (!aclink->valid)
+ continue;
+
+ if (aclink->slot_valid & (1 << 9)) {
+ ptr[iprtd->offset / 2] = aclink->slots_data[0] >> 4;
+ iprtd->offset+=2;
+ }
+
+ if (aclink->slot_valid & (1 << 8)) {
+ ptr[iprtd->offset / 2] = aclink->slots_data[1] >> 4;
+ iprtd->offset+=2;
+ }
+
+ iprtd->offset %= (SAI_AC97_RBUF_FRAMES * 4 * SAI_AC97_RBUF_COUNT);
+ }
+
+ snd_pcm_period_elapsed(iprtd->substream);
+ }
}
static int vf610_sai_ac97_read_write(struct snd_ac97 *ac97, bool isread,
@@ -442,7 +492,7 @@ static struct snd_soc_dai_driver fsl_sai_ac97_dai = {
.stream_name = "PCM Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_48000,
+ .rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,
},
.ops = &fsl_sai_pcm_dai_ops,
@@ -584,8 +634,6 @@ static int snd_fsl_sai_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
atomic_set(&iprtd->playing, 0);
else
atomic_set(&iprtd->capturing, 0);
- if (!atomic_read(&iprtd->playing) &&
- !atomic_read(&iprtd->capturing))
break;
default:
@@ -711,6 +759,13 @@ static int fsl_sai_pcm_new(struct snd_soc_pcm_runtime *rtd)
return ret;
}
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = imx_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret)
+ return ret;
+ }
+
pr_debug("%s, %p\n", __func__, pcm);
return 0;