[alsa-devel] ASoC: Choppy audio on OMAP using McBSP3 interface
Hello,
I wrote a ASoC driver for a TI DM3730-based custom board interfacing a TI TAS5709 codec over the McBSP2 interface. I'm using the Linux kernel 2.6.32. The driver works very well, no problems with the sound quality at all.
Unfortunately due to reasons alien to me I needed to move the driver to work using the McBSP3 interface instead. All I did was then to change the pointer in my machine code to point to the McBSP3 interface:
static struct snd_soc_dai_link bp_audio_dai = { .name = "tas5709", .stream_name = "tas5709", .cpu_dai = &omap_mcbsp_dai[2], <-- from [1] to [2] .codec_dai = &tas5709_dai, .init = bp_audio_init, .ops = &bp_audio_ops, };
and in my machine init code:
*(unsigned int *)bp_audio_dai.cpu_dai->private_data = 2;
With this change the audio driver works well when playing local audio files via the aplay command. But it fails miserably when playing streaming audio from a VoIP connection, the audio is very choppy as it happens when there is a large packet loss. However audio is mono 16 bits @8 KHz, not really very demanding.
I realize there is a big difference in the internal buffer size for the two interfaces, the McBSP2 having 10 times the buffer space of the McBSP3. I suspect this is somehow the source of my problems. I'm inclined to believe that there is some DMA-related optimization issue that has to be addressed somehow, but this is just a guess.
I fail to understand why is it that I can play local audio files but not streaming audio. What makes the difference? Any suggestions on how to address this issue?
Thank you,
On 05/31/2011 09:21 AM, Pedro Sanchez wrote:
Hello,
I wrote a ASoC driver for a TI DM3730-based custom board interfacing a TI TAS5709 codec over the McBSP2 interface. I'm using the Linux kernel 2.6.32. The driver works very well, no problems with the sound quality at all.
Unfortunately due to reasons alien to me I needed to move the driver to work using the McBSP3 interface instead. All I did was then to change the pointer in my machine code to point to the McBSP3 interface:
static struct snd_soc_dai_link bp_audio_dai = { .name = "tas5709", .stream_name = "tas5709", .cpu_dai =&omap_mcbsp_dai[2],<-- from [1] to [2] .codec_dai =&tas5709_dai, .init = bp_audio_init, .ops =&bp_audio_ops, };
and in my machine init code:
*(unsigned int *)bp_audio_dai.cpu_dai->private_data = 2;
With this change the audio driver works well when playing local audio files via the aplay command. But it fails miserably when playing streaming audio from a VoIP connection, the audio is very choppy as it happens when there is a large packet loss. However audio is mono 16 bits @8 KHz, not really very demanding.
I realize there is a big difference in the internal buffer size for the two interfaces, the McBSP2 having 10 times the buffer space of the McBSP3. I suspect this is somehow the source of my problems. I'm inclined to believe that there is some DMA-related optimization issue that has to be addressed somehow, but this is just a guess.
I fail to understand why is it that I can play local audio files but not streaming audio. What makes the difference? Any suggestions on how to address this issue?
Thank you,
Fixed!, For future reference here's the change:
--- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -1254,7 +1254,7 @@ static inline void __devexit omap_additional_remove(struct
static inline void __devinit omap34xx_device_init(struct omap_mcbsp *mcbsp) { - mcbsp->dma_op_mode = MCBSP_DMA_MODE_THRESHOLD; + mcbsp->dma_op_mode = MCBSP_DMA_MODE_FRAME; if (cpu_is_omap34xx()) { mcbsp->max_tx_thres = max_thres(mcbsp); mcbsp->max_rx_thres = max_thres(mcbsp);
This changes the default DMA operation to be frame-based as opposed to be triggered by a threshold (which is probably good just for McBSP2).
participants (1)
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Pedro Sanchez