[alsa-devel] Trying to understand alsa

[Andrew Eikum]

On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote:
> I have an application that works using 512 sample packets of 22050Hz 16
> bit mono audio.  The 'receiver' takes many audio streams from a network
> via UDP, at the moment it pipes them into pulse.
> 
> Can alsa buffer audio. At the moment every time I and set an audio
> buffer size I get a negative response from
> snd_pcm_hw_params_set_buffer_size .  I'm somewhat confused about the
> units alsa uses ...
> 

You don't want to over-specify your requirements. You require a buffer
size of "at least" 3 * 512 frames. So use set_buffer_size_min().
Otherwise ALSA will try to set exactly that buffer size, which can
fail.

Check the function signatures for units. Notice that
set_buffer_size*() all take snd_pcm_uframes_t, that is, the number of
frames you want to store. In ALSA terms, a "frame" is a set of a
single sample for every channel. Since you have mono audio, a frame
and a sample are actually the same unit (for 16 bit stereo audio,
1 frame = 2 samples = 32 bits).

> What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x
> 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K
> Byte, 512 samples per buffer) in for playback.  The packets are arriving
> at roughly the correct rate, I just need a buffer to  iron out any
> jitter in network transmit, do I have to do this myself ?
> 
> Can somebody help by telling me which numbers I push into which places
> to make it work ?
> 
> At the moment I get i keep getting a broken pipe, if I underrun how can
> I make it just wait for me ?
> 

If a packet arrives very late (and one will, eventually), you will
underrun.  That's unavoidable. You can check for SND_PCM_STATE_XRUN
from snd_pcm_state().  It's undocumented, but you need to call
snd_pcm_avail_update() first to get an accurate reading from
snd_pcm_state(). When an underrun occurs, recover with
snd_pcm_recover() and then start writing data again.

If a packet arrives early, you'll need to check that the ALSA buffer
isn't full (see snd_pcm_avail_update()), and store it within your
application to write later if it is full.

There's some mostly-accurate information here:
http://0pointer.de/blog/projects/guide-to-sound-apis.html
As with all ALSA documentation, it is confusing and often incorrect,
but it's probably the most helpful document I've found.

Good luck!

Andrew