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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:10:38 +0200 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:40:12 +0200 |
| commit | f8911b987de4a84ff8ae92f41ff492ece4acadb9 (patch) | |
| tree | 0ebda51a6ba23d790da30a7168870928954da395 /libavfilter/buffersrc.c | |
| parent | bf5386385dc504a076453ad58f61f808677be747 (diff) | |
| parent | 5467742232c312b7d61dca7ac57447f728d8d6c9 (diff) | |
| download | ffmpeg-f8911b987de4a84ff8ae92f41ff492ece4acadb9.tar.gz | |
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/buffersrc.c')
| -rw-r--r-- | libavfilter/buffersrc.c | 5 |
1 files changed, 3 insertions, 2 deletions
diff --git a/libavfilter/buffersrc.c b/libavfilter/buffersrc.c index dd9eb39b59..2592cfb64a 100644 --- a/libavfilter/buffersrc.c +++ b/libavfilter/buffersrc.c @@ -408,6 +408,7 @@ static int request_frame(AVFilterLink *link) { BufferSourceContext *c = link->src->priv; AVFilterBufferRef *buf; + int ret = 0; if (!av_fifo_size(c->fifo)) { if (c->eof) @@ -424,7 +425,7 @@ static int request_frame(AVFilterLink *link) ff_end_frame(link); break; case AVMEDIA_TYPE_AUDIO: - ff_filter_samples(link, avfilter_ref_buffer(buf, ~0)); + ret = ff_filter_samples(link, avfilter_ref_buffer(buf, ~0)); break; default: return AVERROR(EINVAL); @@ -432,7 +433,7 @@ static int request_frame(AVFilterLink *link) avfilter_unref_buffer(buf); - return 0; + return ret; } static int poll_frame(AVFilterLink *link) |