SDL audio implementation (mk 2) #13
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The code was basically acting as if it was 16 times larger than specified, which would typically result in an entire DMA buffer being processed at a time instead of in (roughly) the right size chunks. It'll still deviate from Sound_BatchSize fairly frequently (mostly due to any remainder at the end of the current DMA buffer), but by feeding audio to the host more regularly it should help minimise latency and reduce the chance of starvation.
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Thanks, I've had the opportunity to test this out - it works nicely on newer machines, but on older ones the sound continues to break up quite a bit. It's still an improvement over PR #12 though, so it should be OK to merge once it's ready. Sorry for taking a while to get back to you on this. |
ccawley2011
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Oct 10, 2024
This introduces an alternate Sound_FudgeRate implementation that acts as a direct scale factor to the time to the next DMA event, so that it adjusts the rate correctly regardless of the batch size. The SDL sound code has been changed to use the new Sound_FudgeRate system, under the control of a PID controller which aims to keep an average of 0.1s of audio buffered at a time. The PID controller results in much better behaviour than the old control logic, and should work well across a wide range of Arc & host audio settings and performance levels. The 0.1s buffer level probably isn't the best for all situations (some machines may be able to use lower values, others may need it higher), so in future it may be worth having it auto-adjust based on if there have been any underflows recently. But for now it works well enough, and is less than the old code aimed for.
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This is #12 with additional changes to fix a couple of issues with the core sound code and significantly reduce the number of audio breakups.
A couple of potential future improvements: