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authorMichael Niedermayer <michaelni@gmx.at>2012-12-13 18:17:48 +0100
committerMichael Niedermayer <michaelni@gmx.at>2012-12-13 18:17:48 +0100
commit616e6ffac694831148b7ec815c709564c01d62e5 (patch)
treeaf5e0a026dae8fa3a2f40b5cb8950a0d13809f96 /libavfilter/af_aresample.c
parent7726916c14e3385148b92dbd23a89948c2c29c09 (diff)
downloadffmpeg-616e6ffac694831148b7ec815c709564c01d62e5.tar.gz
aresample: increase n_out buffer size a bit to reduce delay in some corner cases
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavfilter/af_aresample.c')
-rw-r--r--libavfilter/af_aresample.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
index 2dcfd695b0..03c8f01ef5 100644
--- a/libavfilter/af_aresample.c
+++ b/libavfilter/af_aresample.c
@@ -174,7 +174,7 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
- int n_out = FFMAX(n_in * aresample->ratio * 2, 1);
+ int n_out = n_in * aresample->ratio * 2 + 256;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
int ret;