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Use 20 ms as the default audio buffer size on Android
It turns out that Android will not pick a low latency buffer size by default.
(cherry picked from commit c381b2bdfa)
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@@ -321,11 +321,15 @@ static bool BuildAAudioStream(SDL_AudioDevice *device)
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ctx.AAudioStreamBuilder_setSampleRate(builder, device->spec.freq);
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ctx.AAudioStreamBuilder_setChannelCount(builder, device->spec.channels);
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// If no specific buffer size has been requested, the device will pick the optimal
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if(SDL_GetHint(SDL_HINT_AUDIO_DEVICE_SAMPLE_FRAMES)) {
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ctx.AAudioStreamBuilder_setBufferCapacityInFrames(builder, 2 * device->sample_frames); // AAudio requires that the buffer capacity is at least
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ctx.AAudioStreamBuilder_setFramesPerDataCallback(builder, device->sample_frames); // twice the size of the data callback buffer size
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}
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int32_t sample_frames;
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if (SDL_GetHint(SDL_HINT_AUDIO_DEVICE_SAMPLE_FRAMES)) {
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sample_frames = device->sample_frames;
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} else {
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// Use 20 ms for the default audio buffer size
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sample_frames = (device->spec.freq / 50);
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}
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ctx.AAudioStreamBuilder_setBufferCapacityInFrames(builder, 2 * sample_frames); // AAudio requires that the buffer capacity is at least
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ctx.AAudioStreamBuilder_setFramesPerDataCallback(builder, sample_frames); // twice the size of the data callback buffer size
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const aaudio_direction_t direction = (recording ? AAUDIO_DIRECTION_INPUT : AAUDIO_DIRECTION_OUTPUT);
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ctx.AAudioStreamBuilder_setDirection(builder, direction);
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