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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-05-18 20:11:44 +0200 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-05-18 22:57:02 +0200 |
| commit | 6ba692f8a7110c3960edb4b8e7a6736ee7124e2e (patch) | |
| tree | 893a69eb97e2be8f7c5d44ff08894c8baff8ba57 /libavfilter/af_aresample.c | |
| parent | c0c2424f737fc5c7f0056751f5d8126b991bfe5d (diff) | |
af_aresample: fix rounding that led to sample accumulation in the buffers.
This fixes a regression that apparently was missed when switching to the
in af resampler
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavfilter/af_aresample.c')
| -rw-r--r-- | libavfilter/af_aresample.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c index a88768496d..91aee91af2 100644 --- a/libavfilter/af_aresample.c +++ b/libavfilter/af_aresample.c @@ -165,7 +165,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref { AResampleContext *aresample = inlink->dst->priv; const int n_in = insamplesref->audio->nb_samples; - int n_out = n_in * aresample->ratio; + int n_out = n_in * aresample->ratio + 1; AVFilterLink *const outlink = inlink->dst->outputs[0]; AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out); |