[alsa-devel] [PATCHv5 18/20] ASoC: Always syncronize audio transfers on frames

[Eduardo Valentin]

From: Eero Nurkkala <ext-eero.nurkkala at nokia.com>

All these steps are required for ASoC to behave correctly.
rccr and xccr are format dependent, for example TDM audio
has different values than I2S or DSP_A. Also the
omap_mcbsp_xmit_enable and/or omap_mcbsp_recv_enable must
be called right after the DMA has started.
This provides no longer L and R channels switching at random.

Signed-off-by: Eero Nurkkala <ext-eero.nurkkala at nokia.com>
---
 sound/soc/omap/omap-mcbsp.c |    9 +++++++++
 1 files changed, 9 insertions(+), 0 deletions(-)

diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index a7b0961..6e85508 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -191,6 +191,11 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		mcbsp_data->active++;
 		omap_mcbsp_start(mcbsp_data->bus_id, play, !play);
+		/* Make sure data transfer is frame synchronized */
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			omap_mcbsp_xmit_enable(mcbsp_data->bus_id, 1);
+		else
+			omap_mcbsp_recv_enable(mcbsp_data->bus_id, 1);
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -336,11 +341,15 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* 1-bit data delay */
 		regs->rcr2	|= RDATDLY(1);
 		regs->xcr2	|= XDATDLY(1);
+		regs->rccr	|= RFULL_CYCLE | RDMAEN | RDISABLE;
+		regs->xccr	|= (DXENDLY(1) | XDMAEN | XDISABLE);
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* 1-bit data delay */
 		regs->rcr2      |= RDATDLY(1);
 		regs->xcr2      |= XDATDLY(1);
+		regs->rccr	|= RFULL_CYCLE | RDMAEN | RDISABLE;
+		regs->xccr	|= (DXENDLY(1) | XDMAEN | XDISABLE);
 		/* Invert FS polarity configuration */
 		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
 		break;
-- 
1.6.2.GIT

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