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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/audio.h | |
| parent | bf5386385dc504a076453ad58f61f808677be747 (diff) | |
| parent | 5467742232c312b7d61dca7ac57447f728d8d6c9 (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/audio.h')
| -rw-r--r-- | libavfilter/audio.h | 7 |
1 files changed, 5 insertions, 2 deletions
diff --git a/libavfilter/audio.h b/libavfilter/audio.h index cab1a6c722..a84c378ec8 100644 --- a/libavfilter/audio.h +++ b/libavfilter/audio.h @@ -70,14 +70,17 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms, * @param samplesref a reference to the buffer of audio samples being sent. The * receiving filter will free this reference when it no longer * needs it or pass it on to the next filter. + * + * @return >= 0 on success, a negative AVERROR on error. The receiving filter + * is responsible for unreferencing samplesref in case of error. */ -void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); +int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref); /** * Send a buffer of audio samples to the next link, without checking * min_samples. */ -void ff_filter_samples_framed(AVFilterLink *link, +int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref); #endif /* AVFILTER_AUDIO_H */ |