ASoC: add start treshold to prevent underflows in some cases

This is useful for apps that don't rely on audio blocking and
feed fixed amount of samples over time period (like emulators).

diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index acd68ef..b411eb4 100644 (file)
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -168,6 +168,31 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 
        omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
 
+       /*
+        * To handle realtime streaming sources properly, we need to be sure
+        * we have at least 2 periods of size larger than FIFO in the buffer
+        * to start playing without underflowing. This is because DMA is always
+        * ahead of playback by amount close to FIFO size and needs to have
+        * next period ready when previous one finishes transfering to FIFO.
+        */
+       if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
+           runtime->periods > 1) {
+               struct snd_soc_pcm_runtime *rtd = substream->private_data;
+               int *mcbsp_bus_id = rtd->dai->cpu_dai->private_data;
+               int fifo_samples = (*mcbsp_bus_id == 1) ? 1024 : 256;
+               int period_samples, period_frames;
+               
+               period_samples = bytes_to_samples(runtime,
+                               snd_pcm_lib_period_bytes(substream));
+               if (period_samples < fifo_samples + runtime->channels)
+                       period_samples = fifo_samples + runtime->channels;
+               period_frames = bytes_to_frames(runtime,
+                               samples_to_bytes(runtime, period_samples)) * 2;
+
+               if (runtime->start_threshold < period_frames)
+                       runtime->start_threshold = period_frames;
+       }
+
        return 0;
 }