diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-07-10 00:10:38 +0400 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-07-10 00:40:12 +0400 |
commit | f8911b987de4a84ff8ae92f41ff492ece4acadb9 (patch) | |
tree | 0ebda51a6ba23d790da30a7168870928954da395 /libavfilter/buffersink.c | |
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/buffersink.c')
-rw-r--r-- | libavfilter/buffersink.c | 8 |
1 files changed, 7 insertions, 1 deletions
diff --git a/libavfilter/buffersink.c b/libavfilter/buffersink.c index 642350080b..9e908adf6b 100644 --- a/libavfilter/buffersink.c +++ b/libavfilter/buffersink.c @@ -56,6 +56,12 @@ static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf) link->cur_buf = NULL; }; +static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf) +{ + start_frame(link, buf); + return 0; +} + int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf) { BufferSinkContext *s = ctx->priv; @@ -160,7 +166,7 @@ AVFilter avfilter_asink_abuffer = { .inputs = (AVFilterPad[]) {{ .name = "default", .type = AVMEDIA_TYPE_AUDIO, - .filter_samples = start_frame, + .filter_samples = filter_samples, .min_perms = AV_PERM_READ, .needs_fifo = 1 }, { .name = NULL }}, |