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Define the audio output buffer in milliseconds

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In theory, this buffer must be dimensioned for a target duration, so its
size in bytes should depend on the sample rate.
This commit is contained in:
Romain Vimont 2023-03-11 09:31:32 +01:00
parent 238ab872ba
commit bb509d9317
1 changed files with 5 additions and 5 deletions
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10
app/src/audio_player.c
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@ -13,7 +13,7 @@
#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
#define SC_SDL_SAMPLE_FMT AUDIO_F32
#define SC_AUDIO_OUTPUT_BUFFER_SAMPLES 240 // 5ms at 48000Hz
#define SC_AUDIO_OUTPUT_BUFFER_MS 5
static inline uint32_t
bytes_to_samples(struct sc_audio_player *ap, size_t bytes) {
@ -202,8 +202,8 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
bool played = ap->played;
if (played) {
uint32_t max_buffered_samples = ap->target_buffering
+ 12 * SC_AUDIO_OUTPUT_BUFFER_SAMPLES
+ ap->target_buffering / 10;
+ 12 * SC_AUDIO_OUTPUT_BUFFER_MS * ap->sample_rate / 1000
+ ap->target_buffering / 10;
if (buffered_samples > max_buffered_samples) {
uint32_t skip_samples = buffered_samples - max_buffered_samples;
size_t skip_bytes = samples_to_bytes(ap, skip_samples);
@ -231,7 +231,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
// max_initial_buffering samples, this would cause unnecessary delay
// (and glitches to compensate) on start.
uint32_t max_initial_buffering = ap->target_buffering
+ 2 * SC_AUDIO_OUTPUT_BUFFER_SAMPLES;
+ 2 * SC_AUDIO_OUTPUT_BUFFER_MS * ap->sample_rate / 1000;
if (buffered_samples > max_initial_buffering) {
uint32_t skip_samples = buffered_samples - max_initial_buffering;
size_t skip_bytes = samples_to_bytes(ap, skip_samples);
@ -298,7 +298,7 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
.freq = ctx->sample_rate,
.format = SC_SDL_SAMPLE_FMT,
.channels = nb_channels,
.samples = SC_AUDIO_OUTPUT_BUFFER_SAMPLES,
.samples = SC_AUDIO_OUTPUT_BUFFER_MS * ctx->sample_rate / 1000,
.callback = sc_audio_player_sdl_callback,
.userdata = ap,
};