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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/af_aconvert.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_aconvert.c')
| -rw-r--r-- | libavfilter/af_aconvert.c | 6 |
1 files changed, 4 insertions, 2 deletions
diff --git a/libavfilter/af_aconvert.c b/libavfilter/af_aconvert.c index 51167f4327..dc9cf455a3 100644 --- a/libavfilter/af_aconvert.c +++ b/libavfilter/af_aconvert.c @@ -135,12 +135,13 @@ static int config_output(AVFilterLink *outlink) return 0; } -static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) +static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref) { AConvertContext *aconvert = inlink->dst->priv; const int n = insamplesref->audio->nb_samples; AVFilterLink *const outlink = inlink->dst->outputs[0]; AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n); + int ret; swr_convert(aconvert->swr, outsamplesref->data, n, (void *)insamplesref->data, n); @@ -148,8 +149,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref avfilter_copy_buffer_ref_props(outsamplesref, insamplesref); outsamplesref->audio->channel_layout = outlink->channel_layout; - ff_filter_samples(outlink, outsamplesref); + ret = ff_filter_samples(outlink, outsamplesref); avfilter_unref_buffer(insamplesref); + return ret; } AVFilter avfilter_af_aconvert = { |