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Compute buffering and compensation without lock

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Once underflow has been read with a lock, the buffering and compensation
may be performed without shared variables.
This commit is contained in:
Romain Vimont 2023-03-10 22:19:28 +01:00
parent 0b8a5ca923
commit eca8766545
1 changed files with 21 additions and 17 deletions
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38
app/src/audio_player.c
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@ -180,6 +180,8 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
// Read with lock held, to be used after unlocking
bool played = ap->played;
uint32_t underflow = ap->underflow;
if (played) {
uint32_t max_buffered_samples = ap->target_buffering
+ 12 * SC_AUDIO_OUTPUT_BUFFER_MS * ap->sample_rate / 1000
@ -191,23 +193,8 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
" samples", skip_samples);
}
// Number of samples added (or removed, if negative) for compensation
int32_t instant_compensation =
(int32_t) samples_written - frame->nb_samples;
int32_t inserted_silence = (int32_t) ap->underflow;
// The compensation must apply instantly, it must not be smoothed
ap->avg_buffering.avg += instant_compensation + inserted_silence;
ap->underflow = 0; // reset
// However, the buffering level must be smoothed
sc_average_push(&ap->avg_buffering, buffered_samples);
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGD("[Audio] buffered_samples=%" PRIu32 " avg_buffering=%f",
buffered_samples, sc_average_get(&ap->avg_buffering));
#endif
// reset (the current value was copied to a local variable)
ap->underflow = 0;
} else {
// SDL playback not started yet, do not accumulate more than
// max_initial_buffering samples, this would cause unnecessary delay
@ -230,6 +217,23 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
SDL_UnlockAudioDevice(ap->device);
if (played) {
// Number of samples added (or removed, if negative) for compensation
int32_t instant_compensation =
(int32_t) samples_written - frame->nb_samples;
int32_t inserted_silence = (int32_t) underflow;
// The compensation must apply instantly, it must not be smoothed
ap->avg_buffering.avg += instant_compensation + inserted_silence;
// However, the buffering level must be smoothed
sc_average_push(&ap->avg_buffering, buffered_samples);
#ifndef SC_AUDIO_PLAYER_NDEBUG
LOGD("[Audio] buffered_samples=%" PRIu32 " avg_buffering=%f",
buffered_samples, sc_average_get(&ap->avg_buffering));
#endif
ap->samples_since_resync += samples_written;
if (ap->samples_since_resync >= ap->sample_rate) {
// Recompute compensation every second