When testing out a game I'm making with Allegro 5, I couldn't help but notice that the timing on a rapidly-playing (about 5 times per second) sound seemed "off". What I mean by this is that the sound wasn't playing at a constant rhythm, but rather erratically changing the timing between each sound.

Back when I was working with SDL, there was an option you could set when creating initializing the audio that was essentially the size of a "chunk" of audio. This is a completely different thing than the frequency and its normal value was usually around 2048. Making this value a high number would reduce the instance of pops and crackles that might happen on slower computers, but it caused horrid timing issues like the one I described above. Likewise, lowering the value made crackling more likely to happen, but it almost completely fixed any timing issues I was having.

Is there a way to adjusting this sort of setting in Allegro 5? I read through the API but the only thing that looked promising were the 'fragment_count' and 'samples' arguments to al_create_audio_stream(). I don't think this is what I'm looking for, though, because I'm not sure you can even play sound effects to an audio stream.

al_create_audio_stream accepts a frag_samples parameter - this is essentially your buffer size, in number of samples per fragment (fragment_count) per audio stream.

It's not hard, just work. Also unsure how to expose it through the API but maybe that can come later.

From what I understand from the API guide, you can load and play a sound as an audio stream, but that's not the same thing as playing multiple sample instances through an already-running audio stream, and I don't think that's what they're for. I'm probably not going to solve any lag issues unless I can change the buffer size of the actual mixer.