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[client] audio: allow the audiodev to determine the start fill level

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This commit is contained in:
Geoffrey McRae
2022-01-19 01:52:19 +11:00
parent 46da447429
commit 04ae9217e8
4 changed files with 36 additions and 23 deletions
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15
client/src/audio.c
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@@ -167,7 +167,7 @@ void audio_playbackStart(int channels, int sampleRate, PSAudioFormat format,
playbackStopNL();
}
const int bufferFrames = sampleRate / 10;
const int bufferFrames = sampleRate;
audio.playback.buffer = ringbuffer_new(bufferFrames,
channels * sizeof(uint16_t));
@@ -247,17 +247,14 @@ void audio_playbackData(uint8_t * data, size_t size)
if (!STREAM_ACTIVE(audio.playback.state))
return;
const int frames = size / audio.playback.stride;
int frames = size / audio.playback.stride;
ringbuffer_append(audio.playback.buffer, data, frames);
// don't start playback until the buffer is sifficiently full to avoid
// glitches
if (audio.playback.state == STREAM_STATE_SETUP &&
ringbuffer_getCount(audio.playback.buffer) >=
ringbuffer_getLength(audio.playback.buffer) / 4)
if (audio.playback.state == STREAM_STATE_SETUP)
{
audio.playback.state = STREAM_STATE_RUN;
audio.audioDev->playback.start();
frames = ringbuffer_getLength(audio.playback.buffer);
if (audio.audioDev->playback.start(frames))
audio.playback.state = STREAM_STATE_RUN;
}
}