boatbod
Member
I do not like using "hw:0,0" or "plughw:0,0" as audio parameters for the simple fact that it locks the hardware, which kills audio for everything else, and visa versa.
Having said that, I did some testing with those parameters, and discovered my hardware also does not support 8000 natively. The lowest sample rate my hardware supports is 41000. So I modified the source code to implement a "fractional resampler" between the volume scaler and the audio sink. This works of course, but it still does not solve the "aU" messages you see in the terminal, those can be ignored. The audio output however was just fine. I do not think this modification will solve your problem.
Under Ubuntu, I recommend you use "default" or don't specify anything and let alsa handle the resampling at the driver level.
The modification I had you make to the "config.conf" file should have taken care of the stuttering. If it didn't work, I don't know what else to tell you. You aren't running in a virtual machine are you?
No, not a virtual machine, just an older Core i5 2410M laptop with 8GB running Ubuntu Mate 14.04 that is dedicated to providing a broadcastify feed. The aU's are issued about 1 per second or so, and while the audio quality is generally ok, the glitches do cause key words/ phrases to go silent right when you don't want them to.
I've tried all sorts of values of nperiods & period_time but it doesn't make much if any discernible difference.
I did determine that if I disable PulseAudio I get no usable sound due to the mismatched sample rate. When I try a 'plughw' device, scope.py errors out attempting to set alsa time periods inside the audio alsa sink code. If I try a plain 'hw' or 'default' device , the sample rate mismatch prevents the audio doing more than the occasional squeak.
How can I try your rational resampler modification? Sounds promising.