diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:10:38 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-07-09 22:40:12 +0200 |
commit | f8911b987de4a84ff8ae92f41ff492ece4acadb9 (patch) | |
tree | 0ebda51a6ba23d790da30a7168870928954da395 /libavfilter/af_amerge.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/af_amerge.c')
-rw-r--r-- | libavfilter/af_amerge.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c index 802188d028..660ae5dc9d 100644 --- a/libavfilter/af_amerge.c +++ b/libavfilter/af_amerge.c @@ -212,7 +212,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[], } } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) { AVFilterContext *ctx = inlink->dst; AMergeContext *am = ctx->priv; @@ -232,7 +232,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) for (i = 1; i < am->nb_inputs; i++) nb_samples = FFMIN(nb_samples, am->in[i].nb_samples); if (!nb_samples) - return; + return 0; outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples); outs = outbuf->data[0]; @@ -285,7 +285,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples) } } } - ff_filter_samples(ctx->outputs[0], outbuf); + return ff_filter_samples(ctx->outputs[0], outbuf); } static av_cold int init(AVFilterContext *ctx, const char *args) |