I ran into a couple of other issues which I am still chasing but thought it
was worth putting my "findings so far" out there.
1. Using the Alsa driver, a call to "getBufferOnSoundcard" returns zero
immediately after the first few calls to AddSamples. I think this is because
it only calls snd_pcm_delay if the state is running or draining, but the
state seems to stay as "PREPARED" for a while. I don't understand ALSA
enough to know what causes it to change from Prepared to Running, i.e. will
the first call to writei do this or does it need a minimum number of bytes?
2. Because of the asynchronous nature of the drivers, there can be a gap of
around 5ms to 10ms between my first call to AddSamples and the data being
sent to the soundcard. (i.e. it has to wake from a sleep and then the task
has to be scheduled by the kernel). Can you think of any way of getting rid
of this? Is it not possible to feed the soundcard from AddSamples directly
if the soundcard is empty?
Is the issue to sync the record and playback streams?
I wonder if we shouldn't just move the record code into the audio driver
and then we can use the various alsa features to ensure absolutely
perfect record and playback sync? This does mean adding a fairly
serious bunch of code, but it needs to go somewhere I guess...
I use Jack for my audio in and out, so I'm going to have to write record
routines for this anyway. Perhaps we should be designing the interface
in order to just add this stuff into the audio classes?
Ed W
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