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