[alsa-devel] Please help in adding ams-delta support to ASoC
Peter Ujfalusi
peter.ujfalusi at nokia.com
Wed Jun 10 10:12:32 CEST 2009
On Tuesday 09 June 2009 18:17:42 ext Janusz Krzysztofik wrote:
> Peter Ujfalusi wrote:
> > I'm kind of bad at visualizing things, is it possible to put somewhere
> > the screenshoot of the scope showing at least one sample?
>
> Good idea. I'll take some screenshots for future reference and let you
> know when available.
>
> > I have been looking for clues around the net, and it seams that the codec
> > in question has stereo 16 bit format.
>
> From the very minimal mcbsp setup I get best audio experience with
> (using original omap-alsa based driver - see below), it looks like the
> codec speeks DSP (single phase), 16-bit mono @8kHz on output, but 8-bit
> stereo @8kHz on input. Capturing one channel only (the first one) I
> can't hear myself speaking, so audio from the microphone must be sent
> over the second input channel.
>
> +static struct omap_mcbsp_reg_cfg mcbsp_regs = {
> + .spcr2 = FREE | XINTM(3) | XRST,
> + .spcr1 = RINTM(3) | RRST,
> + .rcr1 = RFRLEN1(2 - 1) | RWDLEN1(OMAP_MCBSP_WORD_8),
> + .xcr1 = XFRLEN1(1 - 1) | XWDLEN1(OMAP_MCBSP_WORD_16),
> +};
>
>
> +static snd_pcm_hardware_t vc_snd_omap_alsa_playback = {
> + .info = (SNDRV_PCM_INFO_INTERLEAVED |
> SNDRV_PCM_INFO_BLOCK_TRANSFER |
> + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
> + .formats = (SNDRV_PCM_FMTBIT_S16_LE),
> + .rates = (SNDRV_PCM_RATE_8000 |
> + SNDRV_PCM_RATE_KNOT),
> + .rate_min = 8000,
> + .rate_max = 8000,
> + .channels_min = 1,
> + .channels_max = 1,
> + .buffer_bytes_max = 128 * 1024,
> + .period_bytes_min = 32,
> + .period_bytes_max = 8 * 1024,
> + .periods_min = 16,
> + .periods_max = 255,
> + .fifo_size = 0,
> +};
> +
> +static snd_pcm_hardware_t vc_snd_omap_alsa_capture = {
> + .info = (SNDRV_PCM_INFO_INTERLEAVED |
> SNDRV_PCM_INFO_BLOCK_TRANSFER |
> + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
> + .formats = (SNDRV_PCM_FMTBIT_U8),
> + .rates = (SNDRV_PCM_RATE_8000 |
> + SNDRV_PCM_RATE_KNOT),
> + .rate_min = 8000,
> + .rate_max = 8000,
> + .channels_min = 2,
> + .channels_max = 2,
> + .buffer_bytes_max = 128 * 1024,
> + .period_bytes_min = 32,
> + .period_bytes_max = 8 * 1024,
> + .periods_min = 16,
> + .periods_max = 255,
> + .fifo_size = 0,
> +};
>
> For other combinations of single/dual phase, sample size, mono/stereo,
> sound I get is much more distorted. Playing with polarisation and delays
> for getting still better experience remains on my todo list.
>
> BTW, I can't see any way of specifying a similiar mcbsp setup in new
> omap asoc framework.
Yeah, 8 bit is not supported - in fact only 16 bit is supported - with ASoC on
OMAP.
But you should be able to use the playback with the current ASoC.
>
> --------------
>
> > --- a/sound/soc/omap/omap-pcm.c
> > +++ b/sound/soc/omap/omap-pcm.c
> > @@ -193,11 +193,15 @@ static int omap_pcm_trigger(struct
> > snd_pcm_substream *substream, int cmd)
> > case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
> > prtd->period_index = 0;
> > omap_start_dma(prtd->dma_ch);
> > + printk("omap_pcm_trigger START: DMA pointer at 0x%08x\n",
> > + (unsigned)omap_get_dma_src_pos(prtd->dma_ch));
> > break;
> >
> > case SNDRV_PCM_TRIGGER_STOP:
> > case SNDRV_PCM_TRIGGER_SUSPEND:
> > case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
> > + printk("omap_pcm_trigger STOP: DMA pointer at 0x%08x\n",
> > + (unsigned)omap_get_dma_src_pos(prtd->dma_ch));
> > prtd->period_index = -1;
> > omap_stop_dma(prtd->dma_ch);
> > break;
> >
> >
> > Than start a playback, and stop it with CTRL+C, see if the two pointers
> > are different...
>
> Both playback and capture start with their own but always the same value
> (something like 0x1101a0d0 for playback, 0xe101a0d0 for capture), and
> always stop with this value unchanged.
Hmm, this means that the DMA was not running at all (since it has not been
progressed). So you might have the missing DMA interrupt problem, but first we
should get the DMA to actually run.
Looking at the dma code for clues...
Even bigger Hmmm...
For OMAP1 the DMA synchronization is not configured.
I just wonder how this supposed to be working on OMAP1 platforms.
Anyway, can you try the following patch (and please keep the previous printk
in the omap-pcm.c):
--- a/arch/arm/plat-omap/dma.c
+++ b/arch/arm/plat-omap/dma.c
@@ -266,6 +266,8 @@ void omap_set_dma_transfer_params(int lch, int data_type,
int elem_count,
ccr &= ~(1 << 5);
if (sync_mode == OMAP_DMA_SYNC_FRAME)
ccr |= 1 << 5;
+ if (dma_trigger)
+ ccr |= dma_trigger & 0x1f;
dma_write(ccr, CCR(lch));
ccr = dma_read(CCR2(lch));
in the if (cpu_class_is_omap1()) { path...
The interesting thing is the following:
the old ALSA implementation configured the DMA in this way:
omap_set_dma_transfer_params(channel, dt, cen, cfn, 0x00, 0, 0);
Meaning, no DMA synchronization, which is fair, since the DMA code for OMAP1
does not configured it anyway.
>
> > 2) I would I think try to port the 2.6.28 pure ALSA version to the head
> > of l- o, with minimal (only the needed) changes and see if it is still
> > working.
>
> Trying to use the new driver on the last l-o revision supporting both
> omap-alsa and omap asoc, I got a broken system that hanged up completely
> at sound device first access. The same for earlier and later l-o
> revisions I have ever tried, unlike mainline that at least does not
> hang. From all that I would conclude that porting the old driver could
> be just waste of time, as the problem is probably omap asoc related, not
> just omap, probably existed from the start of omap asoc life, and could
> be solved on any l-o or mainline revision, including those eariler l-o
> that supported both frameworks. But I can be wrong, of course.
There seams to be no major difference between the 2.6.28 and the latest
arch/arm/plat-omap/dma.c and mcbsp.c
I would expected that by just copying (I know that there are other things
involved) the old ALSA arm directory to the latest kernel and compiling it
should be working... It seams I was wrong.
--
Péter
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