On Sat, Sep 20, 2014 at 04:41:31PM +0100, Will Godfrey wrote: > While getting to grips with Yoshimi I set up a microsecond timer in the > function > that actually generates the data. To get a worst-case scenario I did this on > my > office machine that's not specially set up for audio. > > I configured jack for 512 frames/period, 2 periods/buffer and 48k, giving an > overall latency of 21.3mS > > Running a moderately complex 12 part tune, the data 'build' time was varying > between about 1.8mS and 6.1mS per buffer. It dropped to less than 1mS when > there > was no sound being produced. > > That was a lot more variation than I was expecting but considering the variety > of calls that were being made, dependent on which voices were sounding and > with > what effects, I don't know how this could be avoided.
If the load changes in only function of the number of active voices that is perfectly OK. > I did another check for continuous sounds, and under those circumstances the > time didn't vary significantly. That's a good sign. You should also test this with smaller period sizes. If all is OK the required calculation time should just decrease in proportion. If that is the case then the CPU load as seen by Jack should remain the same, apart from a small increase due to overhead (task switching etc.). The thing to be avoided is code that e.g. generates a heavy load every fourth period and does almost nothing in the three periods in between. Or a synth that generates a peak load whenever a voice is started and much less while the note lasts. IIRC yoshi/zyn use FFTs in some of the algorithms, so this sort of thing could happen. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow) _______________________________________________ Linux-audio-dev mailing list [email protected] http://lists.linuxaudio.org/listinfo/linux-audio-dev
