[alsa-devel] ALSA throwing buffers away?
radivojejovanovic at gmail.com
Tue Nov 23 18:51:06 CET 2010
I have changed the pointer callback as you suggest.
Unfortunately, the result is still the same. Still the buffer filling is
skipping to fill all the data correctly. I am using Advanced Linux Sound
Architecture Driver Version 1.0.21.
On Tue, Nov 23, 2010 at 9:40 AM, Clemens Ladisch <clemens at ladisch.de> wrote:
> Radivoje Jovanovic wrote:
> > > Radivoje Jovanovic wrote:
> > > > I am developing driver for a really simple hardware. Hardware has
> codec that
> > > > supports mono/stereo and the amplifier that supports only one speaker
> so the
> > > > driver has to be mono driver and the codec is setup to manage mono
> data. I
> > > > have setup ALSA with following parameters:
> > > >
> > > > .info = (SNDRV_PCM_INFO_NONINTERLEAVED |
> > >
> > > Better use SNDRV_PCM_INFO_INTERLEAVED; this is the format used by
> > > all (stereo) sound cards, and so it is expected even for mono files
> > > (where there actually isn't any difference).
> > I am worried about this approach since if I want to play stereo sound I
> > would have to increase .channels_max = 2,
> No, ALSA can automatically convert the sample format.
> > Here is my pointer callback:
> > offset=READ_REG16(AUDIO_CONFIG_DMA_CUR_ADDR_HIGH))<<16) |
> > offset = offset - substream->runtime->dma_addr;
> This offset looks as if it is in bytes, but you have to return a value
> measured in frames. Use bytes_to_frames().
> > if (offset >= runtime->buffer_size)
> > offset = 0;
> Remove this check; the ALSA framework already checks this and outputs
> debugging info if this happens.
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