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Add --audio-output-buffer

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On some systems, the SDL audio callback is not called frequently enough
(for example it requests 5ms of samples every 10ms), because the output
buffer is too small.

By default, we want to use a small value (5ms) to minimize latency and
buffer underrun, but if it does not work well, users need a way to
increase it.

Refs #3793 <https://github.com/Genymobile/scrcpy/issues/3793>
This commit is contained in:
Romain Vimont
2023-03-13 09:23:02 +01:00
parent 4755b97908
commit 39544f34b4
10 changed files with 86 additions and 21 deletions
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43
app/src/audio_player.c
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@@ -59,8 +59,6 @@
#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
#define SC_SDL_SAMPLE_FMT AUDIO_F32
#define SC_AUDIO_OUTPUT_BUFFER_MS 5
#define TO_BYTES(SAMPLES) sc_audiobuf_to_bytes(&ap->buf, (SAMPLES))
#define TO_SAMPLES(BYTES) sc_audiobuf_to_samples(&ap->buf, (BYTES))
@@ -230,8 +228,8 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
if (played) {
uint32_t max_buffered_samples = ap->target_buffering
+ 12 * SC_AUDIO_OUTPUT_BUFFER_MS * ap->sample_rate / 1000
+ ap->target_buffering / 10;
+ 12 * ap->output_buffer
+ ap->target_buffering / 10;
if (buffered_samples > max_buffered_samples) {
uint32_t skip_samples = buffered_samples - max_buffered_samples;
sc_audiobuf_skip(&ap->buf, skip_samples);
@@ -246,7 +244,7 @@ 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_MS * ap->sample_rate / 1000;
+ 2 * ap->output_buffer;
if (buffered_samples > max_initial_buffering) {
uint32_t skip_samples = buffered_samples - max_initial_buffering;
sc_audiobuf_skip(&ap->buf, skip_samples);
@@ -333,11 +331,28 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
unsigned nb_channels = tmp;
#endif
assert(ctx->sample_rate > 0);
assert(!av_sample_fmt_is_planar(SC_AV_SAMPLE_FMT));
int out_bytes_per_sample = av_get_bytes_per_sample(SC_AV_SAMPLE_FMT);
assert(out_bytes_per_sample > 0);
ap->sample_rate = ctx->sample_rate;
ap->nb_channels = nb_channels;
ap->out_bytes_per_sample = out_bytes_per_sample;
ap->target_buffering = ap->target_buffering_delay * ap->sample_rate
/ SC_TICK_FREQ;
uint64_t aout_samples = ap->output_buffer_duration * ap->sample_rate
/ SC_TICK_FREQ;
assert(aout_samples <= 0xFFFF);
ap->output_buffer = (uint16_t) aout_samples;
SDL_AudioSpec desired = {
.freq = ctx->sample_rate,
.format = SC_SDL_SAMPLE_FMT,
.channels = nb_channels,
.samples = SC_AUDIO_OUTPUT_BUFFER_MS * ctx->sample_rate / 1000,
.samples = aout_samples,
.callback = sc_audio_player_sdl_callback,
.userdata = ap,
};
@@ -356,11 +371,6 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
}
ap->swr_ctx = swr_ctx;
assert(ctx->sample_rate > 0);
assert(!av_sample_fmt_is_planar(SC_AV_SAMPLE_FMT));
int out_bytes_per_sample = av_get_bytes_per_sample(SC_AV_SAMPLE_FMT);
assert(out_bytes_per_sample > 0);
#ifdef SCRCPY_LAVU_HAS_CHLAYOUT
av_opt_set_chlayout(swr_ctx, "in_chlayout", &ctx->ch_layout, 0);
av_opt_set_chlayout(swr_ctx, "out_chlayout", &ctx->ch_layout, 0);
@@ -383,13 +393,6 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
goto error_free_swr_ctx;
}
ap->sample_rate = ctx->sample_rate;
ap->nb_channels = nb_channels;
ap->out_bytes_per_sample = out_bytes_per_sample;
ap->target_buffering = ap->target_buffering_delay * ap->sample_rate
/ SC_TICK_FREQ;
// Use a ring-buffer of the target buffering size plus 1 second between the
// producer and the consumer. It's too big on purpose, to guarantee that
// the producer and the consumer will be able to access it in parallel
@@ -458,8 +461,10 @@ sc_audio_player_frame_sink_close(struct sc_frame_sink *sink) {
}
void
sc_audio_player_init(struct sc_audio_player *ap, sc_tick target_buffering) {
sc_audio_player_init(struct sc_audio_player *ap, sc_tick target_buffering,
sc_tick output_buffer_duration) {
ap->target_buffering_delay = target_buffering;
ap->output_buffer_duration = output_buffer_duration;
static const struct sc_frame_sink_ops ops = {
.open = sc_audio_player_frame_sink_open,