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authorMichael Niedermayer <michaelni@gmx.at>2012-07-09 22:10:38 +0200
committerMichael Niedermayer <michaelni@gmx.at>2012-07-09 22:40:12 +0200
commitf8911b987de4a84ff8ae92f41ff492ece4acadb9 (patch)
tree0ebda51a6ba23d790da30a7168870928954da395 /libavfilter/af_pan.c
parentbf5386385dc504a076453ad58f61f808677be747 (diff)
parent5467742232c312b7d61dca7ac57447f728d8d6c9 (diff)
Merge remote-tracking branch 'qatar/master'
* qatar/master: mss3: use standard zigzag table mss3: split DSP functions that are used in MTS2(MSS4) into separate file motion-test: do not use getopt() tcp: add initial timeout limit for incoming connections configure: Change the rdtsc check to a linker check avconv: propagate fatal errors from lavfi. lavfi: add error handling to filter_samples(). fate-run: make avconv() properly deal with multiple inputs. asplit: don't leak the input buffer. af_resample: fix request_frame() behavior. af_asyncts: fix request_frame() behavior. libx264: support aspect ratio switching matroskadec: honor error_recognition when encountering unknown elements. lavr: resampling: add support for s32p, fltp, and dblp internal sample formats lavr: resampling: add filter type and Kaiser window beta to AVOptions lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format lavr: mix: validate internal sample format in ff_audio_mix_init() Conflicts: ffmpeg.c ffplay.c libavcodec/libx264.c libavfilter/audio.c libavfilter/split.c libavformat/tcp.c tests/fate-run.sh Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavfilter/af_pan.c')
-rw-r--r--libavfilter/af_pan.c6
1 files changed, 4 insertions, 2 deletions
diff --git a/libavfilter/af_pan.c b/libavfilter/af_pan.c
index f451e0034c..bb96ad0511 100644
--- a/libavfilter/af_pan.c
+++ b/libavfilter/af_pan.c
@@ -343,8 +343,9 @@ static int config_props(AVFilterLink *link)
return 0;
}
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
+ int ret;
int n = insamples->audio->nb_samples;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
@@ -354,8 +355,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
avfilter_copy_buffer_ref_props(outsamples, insamples);
outsamples->audio->channel_layout = outlink->channel_layout;
- ff_filter_samples(outlink, outsamples);
+ ret = ff_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
+ return ret;
}
static av_cold void uninit(AVFilterContext *ctx)