Define the audio output buffer in milliseconds
In theory, this buffer must be dimensioned for a target duration, so its size in bytes should depend on the sample rate.
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1 changed files with 5 additions and 5 deletions
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@ -13,7 +13,7 @@
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#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
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#define SC_AV_SAMPLE_FMT AV_SAMPLE_FMT_FLT
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#define SC_SDL_SAMPLE_FMT AUDIO_F32
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#define SC_SDL_SAMPLE_FMT AUDIO_F32
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#define SC_AUDIO_OUTPUT_BUFFER_SAMPLES 240 // 5ms at 48000Hz
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#define SC_AUDIO_OUTPUT_BUFFER_MS 5
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static inline uint32_t
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static inline uint32_t
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bytes_to_samples(struct sc_audio_player *ap, size_t bytes) {
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bytes_to_samples(struct sc_audio_player *ap, size_t bytes) {
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@ -202,7 +202,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
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bool played = ap->played;
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bool played = ap->played;
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if (played) {
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if (played) {
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uint32_t max_buffered_samples = ap->target_buffering
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uint32_t max_buffered_samples = ap->target_buffering
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+ 12 * SC_AUDIO_OUTPUT_BUFFER_SAMPLES
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+ 12 * SC_AUDIO_OUTPUT_BUFFER_MS * ap->sample_rate / 1000
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+ ap->target_buffering / 10;
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+ ap->target_buffering / 10;
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if (buffered_samples > max_buffered_samples) {
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if (buffered_samples > max_buffered_samples) {
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uint32_t skip_samples = buffered_samples - max_buffered_samples;
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uint32_t skip_samples = buffered_samples - max_buffered_samples;
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@ -231,7 +231,7 @@ sc_audio_player_frame_sink_push(struct sc_frame_sink *sink,
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// max_initial_buffering samples, this would cause unnecessary delay
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// max_initial_buffering samples, this would cause unnecessary delay
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// (and glitches to compensate) on start.
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// (and glitches to compensate) on start.
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uint32_t max_initial_buffering = ap->target_buffering
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uint32_t max_initial_buffering = ap->target_buffering
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+ 2 * SC_AUDIO_OUTPUT_BUFFER_SAMPLES;
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+ 2 * SC_AUDIO_OUTPUT_BUFFER_MS * ap->sample_rate / 1000;
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if (buffered_samples > max_initial_buffering) {
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if (buffered_samples > max_initial_buffering) {
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uint32_t skip_samples = buffered_samples - max_initial_buffering;
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uint32_t skip_samples = buffered_samples - max_initial_buffering;
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size_t skip_bytes = samples_to_bytes(ap, skip_samples);
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size_t skip_bytes = samples_to_bytes(ap, skip_samples);
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@ -298,7 +298,7 @@ sc_audio_player_frame_sink_open(struct sc_frame_sink *sink,
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.freq = ctx->sample_rate,
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.freq = ctx->sample_rate,
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.format = SC_SDL_SAMPLE_FMT,
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.format = SC_SDL_SAMPLE_FMT,
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.channels = nb_channels,
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.channels = nb_channels,
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.samples = SC_AUDIO_OUTPUT_BUFFER_SAMPLES,
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.samples = SC_AUDIO_OUTPUT_BUFFER_MS * ctx->sample_rate / 1000,
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.callback = sc_audio_player_sdl_callback,
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.callback = sc_audio_player_sdl_callback,
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.userdata = ap,
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.userdata = ap,
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};
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};
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