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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.
This commit is contained in:
wm4 2015-06-04 20:08:12 +02:00
parent 935997d4d6
commit fd96bddca9
1 changed files with 9 additions and 6 deletions
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15
audio/filter/af_lavrresample.c
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@ -91,9 +91,10 @@ struct af_resample {
};
#if HAVE_LIBAVRESAMPLE
static int get_delay(struct af_resample *s)
static double get_delay(struct af_resample *s)
{
return avresample_get_delay(s->avrctx);
return avresample_get_delay(s->avrctx) / (double)s->ctx.in_rate +
avresample_available(s->avrctx) / (double)s->ctx.out_rate;
}
static void drop_all_output(struct af_resample *s)
{
@ -104,9 +105,10 @@ static int get_out_samples(struct af_resample *s, int in_samples)
return avresample_get_out_samples(s->avrctx, in_samples);
}
#else
static int get_delay(struct af_resample *s)
static double get_delay(struct af_resample *s)
{
return swr_get_delay(s->avrctx, s->ctx.in_rate);
int64_t base = s->ctx.in_rate * (int64_t)s->ctx.out_rate;
return swr_get_delay(s->avrctx, base) / (double)base;
}
static void drop_all_output(struct af_resample *s)
{
@ -419,8 +421,6 @@ static int filter(struct af_instance *af, struct mp_audio *in)
if (in)
mp_audio_copy_attributes(out, in);
af->delay = get_delay(s) / (double)s->ctx.in_rate;
if (out->samples) {
out->samples = resample_frame(s->avrctx, out, in);
if (out->samples < 0)
@ -450,6 +450,9 @@ static int filter(struct af_instance *af, struct mp_audio *in)
} else {
talloc_free(out);
}
af->delay = get_delay(s);
return 0;
error:
talloc_free(in);