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 <perex@xxxxxxxx> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc. ------------------------------------------------------------------------------ Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day trial. Simplify your report design, integration and deployment - and focus on what you do best, core application coding. Discover what's new with Crystal Reports now. http://p.sf.net/sfu/bobj-july _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
