From ca6e2ce08679c094878d7f39a0349a7db1d13675 Mon Sep 17 00:00:00 2001 From: Eero Nurkkala Date: Thu, 20 Aug 2009 16:18:24 +0300 Subject: ASoC: Always syncronize audio transfers on frames 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 Acked-by: Jarkko Nikula Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcbsp.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index a7b09614734d..6e855080e6ea 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; -- cgit v1.2.1