Fix latency-related buffer sizing #17
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Turns out that eSpeak-NG (the main user of this lib) enforces a minimum buffer size of 60ms which is also the default size. This explains why smaller LATENCY values were inducing choppiness in the audio on some systems. Adjust the comment accordingly,. Also make sure computed buffer sizes don't land in the middle of a sample frame. Doing (samplerate * channels * LATENCY) / 1000 is wrong. Both ALSA and PulseAudio provide nice abstractions for buffer sizing so let's use them directly. In the ALSA case in particular, we want the period to be 60ms, not the whole buffer, so to interleave speech audio computation and audio playback.
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Turns out that eSpeak-NG (the main user of this lib) enforces a minimum
buffer size of 60ms which is also the default size. This explains why
smaller LATENCY values were inducing choppiness in the audio on some
systems. Adjust the comment accordingly,.
Also make sure computed buffer sizes don't land in the middle of a
sample frame. Doing (samplerate * channels * LATENCY) / 1000 is wrong.
Both ALSA and PulseAudio provide nice abstractions for buffer sizing
so let's use them directly. In the ALSA case in particular, we want the
period to be 60ms, not the whole buffer, so to interleave speech audio
computation and audio playback.