[alsa-devel] Writing a DMA-less PCM audio driver

[David Jander]

Hi all,

I am writing a PCM audio driver for a piece of (embedded) hardware that has no
DMA, only a FIFO capable of holding 2048 samples. I was trying to use a
kthread with high realtime (SCHED_FIFO) priority to keep the FIFO filled, and
sleep with schedule_timeout(x), where "x" is depending on the current FIFO fill
level. The same thread also calls snd_pcm_period_elapsed() on every completed
period. It seems to work, but I get sporadic audio skips and I am trying to
figure out where they come from. Most of the time schedule_timeout(1) takes
almost exactly 1ms (1 jiffy at HZ=1000) to complete, but sometimes it takes up
to 30ms. It seems related to file-IO (via NFS) happening on the system, but
the effect is far bigger if the program that is playing the audio (mplayer)
itself is producing file-IO. When using a large cache parameter on mplayer
(effectively preloading the entire MP3 file into RAM), the problem is almost
gone.
Before deciding whether I should debug the network driver to see if it could
produce such tremendous amounts of latency, is there a better way to solve this
problem? Are there any other DMA-less audio drivers I could look at as an
example?
Any suggestion is welcome.

Best regards,

-- 
David Jander
Protonic Holland.