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>
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@ -56,6 +56,12 @@ static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf)
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link->cur_buf = NULL;
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};
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static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
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{
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start_frame(link, buf);
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return 0;
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}
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int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
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{
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BufferSinkContext *s = ctx->priv;
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@ -160,7 +166,7 @@ AVFilter avfilter_asink_abuffer = {
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.inputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = start_frame,
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.filter_samples = filter_samples,
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.min_perms = AV_PERM_READ,
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.needs_fifo = 1 },
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{ .name = NULL }},
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