On Sat, Nov 16, 2019 at 05:50:16PM -0700, Ethan Funk wrote: > A very big structural difference between the APIs is how "rendering" > sample buffers is accomplished. > ... > After years of tweaking the PID filter, I had it working very well, > with no cost (other than processor overhead of the re-sampling)
If I understand what you write correctly, it seems to me that zita-j2a and a2j are doing exactly the same thing. They contain the separate callback, the resampler, buffering, and the control loop. E.g. for an output you'd have buffer -> resampler -> hardware Your separate callback would read the buffer, run the control loop and the resampler, and write to the hardware. The input to the buffer is where you connect to the world controlled by the master card. If your master period size is N, you would write to that buffer N samples during each master card callback. In other words, the input to the buffer is used in exactly the same way as you would use zita-j2a's Jack ports. So I'd be surprised if you really need any fundamental restructuring. ... But what really surprises me is that you had to do the resampling and buffering explicitly as part of your application. Doesn't OSX provide this at the system level (IIRC, by creating an 'aggregated' sound card) ? Ciao, -- FA _______________________________________________ Linux-audio-dev mailing list [email protected] https://lists.linuxaudio.org/listinfo/linux-audio-dev
