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authorStefano Sabatini <stefasab@gmail.com>2011-09-17 01:30:15 +0200
committerStefano Sabatini <stefasab@gmail.com>2011-10-16 00:52:23 +0200
commit22137bb5c25364a2f7406914ad5688f449718f21 (patch)
treef4489d28eeb18843681a46be8972bc8e8aa5ab30
parentcc6b8c4b612d239bef31a8115402b03453c2b4bc (diff)
downloadffmpeg-22137bb5c25364a2f7406914ad5688f449718f21.tar.gz
af_aresample: fix properties setting in the output buffer
In particular: set output timebase to 1/output_sample_rate, fix output PTS computation, and do not forget to copy properties values from the input buffer.
-rw-r--r--libavfilter/af_aresample.c10
1 files changed, 6 insertions, 4 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
index 9af50cf208..4cf6936ba8 100644
--- a/libavfilter/af_aresample.c
+++ b/libavfilter/af_aresample.c
@@ -83,6 +83,7 @@ static int config_output(AVFilterLink *outlink)
aresample->out_rate = outlink->sample_rate;
else
outlink->sample_rate = aresample->out_rate;
+ outlink->time_base = (AVRational) {1, aresample->out_rate};
//TODO: make the resampling parameters configurable
aresample->resample = av_resample_init(aresample->out_rate, inlink->sample_rate,
@@ -270,13 +271,14 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
aresample->outsamplesref =
avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, requested_out_nb_samples);
- avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
- aresample->outsamplesref->pts =
- insamplesref->pts / inlink->sample_rate * outlink->sample_rate;
- aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
outlink->out_buf = aresample->outsamplesref;
}
+ avfilter_copy_buffer_ref_props(aresample->outsamplesref, insamplesref);
+ aresample->outsamplesref->audio->sample_rate = outlink->sample_rate;
+ aresample->outsamplesref->pts =
+ av_rescale(outlink->sample_rate, insamplesref->pts, inlink->sample_rate);
+
/* av_resample() works with planar audio buffers */
if (!inlink->planar && nb_channels > 1) {
int16_t *out[8];