Well, so far no-one has been able to sufficiently answer my inquiry about how proper default ALSA sound capture should be coded. I've done enough research to know that my program is getting an underrun condition on the read. Fine. My question remains... When I set the hardware parameters, who should I need to set the period and/or buffer size/time? Isn't the default acceptable? If not, why not and how do I code around it? I've seen "simple capture examples" that just set the period size to "32" and leave everything else untouched. Is that a good default for all soundcards? If not, how do I discover what that default should be? Here's my angle ... using OSS I could ask the driver to set its own block size for best performance and I would just ask for that size and use it. Never any problem. ALSA doesn't seem to have that concept as far as I can tell. So, since all I want to do is capture audio from any soundcard I choose and record it to a file (I don't care much about perfect latency behavior) what the heck should a generic ALSA capture program do? Paul Braman ------------------------------------------------------------------------------ Start uncovering the many advantages of virtual appliances and start using them to simplify application deployment and accelerate your shift to cloud computing. http://p.sf.net/sfu/novell-sfdev2dev _______________________________________________ Alsa-user mailing list Alsa-user@xxxxxxxxxxxxxxxxxxxxx https://lists.sourceforge.net/lists/listinfo/alsa-user
