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https://github.com/mpv-player/mpv
synced 2024-12-25 16:33:02 +00:00
af_lavrresample: slightly better computation of total delay
On libavresample, don't ignore the buffered output data. On libswresample, don't round the total buffer size to the input samplerate.
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@ -91,9 +91,10 @@ struct af_resample {
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};
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#if HAVE_LIBAVRESAMPLE
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static int get_delay(struct af_resample *s)
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static double get_delay(struct af_resample *s)
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{
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return avresample_get_delay(s->avrctx);
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return avresample_get_delay(s->avrctx) / (double)s->ctx.in_rate +
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avresample_available(s->avrctx) / (double)s->ctx.out_rate;
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}
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static void drop_all_output(struct af_resample *s)
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{
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@ -104,9 +105,10 @@ static int get_out_samples(struct af_resample *s, int in_samples)
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return avresample_get_out_samples(s->avrctx, in_samples);
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}
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#else
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static int get_delay(struct af_resample *s)
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static double get_delay(struct af_resample *s)
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{
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return swr_get_delay(s->avrctx, s->ctx.in_rate);
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int64_t base = s->ctx.in_rate * (int64_t)s->ctx.out_rate;
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return swr_get_delay(s->avrctx, base) / (double)base;
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}
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static void drop_all_output(struct af_resample *s)
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{
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@ -419,8 +421,6 @@ static int filter(struct af_instance *af, struct mp_audio *in)
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if (in)
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mp_audio_copy_attributes(out, in);
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af->delay = get_delay(s) / (double)s->ctx.in_rate;
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if (out->samples) {
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out->samples = resample_frame(s->avrctx, out, in);
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if (out->samples < 0)
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@ -450,6 +450,9 @@ static int filter(struct af_instance *af, struct mp_audio *in)
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} else {
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talloc_free(out);
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}
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af->delay = get_delay(s);
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return 0;
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error:
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talloc_free(in);
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