Heya!
As you might now I am currently working on something called "glitch-free playback model" for PulseAudio: http://0pointer.de/blog/projects/pulse-glitch-free.html
One of the central ideas is to schedule audio via timer interrupts, no longer via period-based interrupts. To acomplish this I need to be able to estimate in advance how long (in time) it will take until the playback buffer runs empty at a certain point. Then I can set myself a system timer to wakeup a certain margin earlier than this and fill the buffer up again. That way I minimize wakeups and should always have enough safety margin to never get an xrun.
Now, the way I implemented this right now is: in each iteration of my playback loop I first fill up the playback buffer completely if it is not fully filled up yet. After that I obviously can go to sleep for the time it takes to play buffer_size samples minus some safety margin, which i set to 20ms. Apparently this is actually not as obvious as I thought. This works fine on many soundcards but it apparently doesn't work on USB hardware. There, ALSA detects a buffer underrun much earlier than when my timer elapses. My loop wakes up due to activity on the ALSA fd, not because of the timer.
My code then goes and asks _avail_update() for the free space in the buffer and sees that the full buffer is available, thus assuming a xrun happened. This will cause PA to double the safety margin and do some other things to make sure an XRUN doesn't happen the next time again.
So, how do I properly estimate how long the filled up buffer will take to play? Or to word it differently: how do I properly estimate how much time will pass until an XRUN would occur?
[My interpretation of what's going on here with lots of guessing how things work is this: the buffer exposed by ALSA is merely a send buffer and just one of the buffers involved before the data actually reaches the speakers. ALSA just takes the data out of this buffer and sends it to the device via USB block-by-block. Thus the send buffer maintained by ALSA is much quicker empty than the audio data is actually played. And Boom! -- If that should actually be the case then I'd like to request an API that allows me to query an upper limit for how much buffering/latency actually happens after the audio data left the ALSA buffer. I could then use this an subtract it from my time to sleep.]
Thanks,
Lennart