On Tue, 24 Nov 2009, Chuck Hallenbeck wrote:
> Hi list,
>
> My colleague Curtis and I are still unable to capture audio from a Delta66
> for streaming through icecast without unacceptable dropouts of samples,
> resulting in choppy sound and speech difficult to follow. Clemens Lavisch
> called our attention to a too small buffer size in our asound.conf file,
> where the capture devices are defined, but every effort to increase it
> reveals that it is capped at 5461 bytes. We can lower it, but cannot
> increase it. There are examples on the net everywhere of higher values,
> but our efforts cannot increase it beyond 5461. If anyone can suggest
> where we might look for a solution to this problem, We would be grateful.
The value 5461 comes from this formula:
262144 (256KB audio buffer) / (12 (channels) * 4 (bytes per sample))
The hardware you're using can handle only 256KB ring buffer giving approx.
247msec audio buffer at 22050Hz rate. This time should be enough.
I think that your system is not tuned in respect of real-time
responses. Check process-scheduler settings, disk I/O usage and
run all audio tasks with highest realtime priority. Ideally, the machine
should not do any other tasks.
Jaroslav
-----
Jaroslav Kysela <[email protected]>
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.
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