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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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@@ -408,6 +408,7 @@ static int request_frame(AVFilterLink *link)
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{
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BufferSourceContext *c = link->src->priv;
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AVFilterBufferRef *buf;
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int ret = 0;
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if (!av_fifo_size(c->fifo)) {
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if (c->eof)
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@@ -424,7 +425,7 @@ static int request_frame(AVFilterLink *link)
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ff_end_frame(link);
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break;
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case AVMEDIA_TYPE_AUDIO:
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ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
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ret = ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
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break;
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default:
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return AVERROR(EINVAL);
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@@ -432,7 +433,7 @@ static int request_frame(AVFilterLink *link)
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avfilter_unref_buffer(buf);
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return 0;
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return ret;
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}
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static int poll_frame(AVFilterLink *link)
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