generic-library/ffmpeg
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Merge remote-tracking branch 'qatar/master'

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* 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>
This commit is contained in:
Michael Niedermayer
2012-07-09 22:10:38 +02:00
parent bf5386385d 5467742232
commit f8911b987d
45 changed files with 648 additions and 339 deletions
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5
libavfilter/af_asetnsamples.c
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@@ -131,7 +131,7 @@ static int push_samples(AVFilterLink *outlink)
return nb_out_samples;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
ASNSContext *asns = ctx->priv;
@@ -145,7 +145,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR,
"Stretching audio fifo failed, discarded %d samples\n", nb_samples);
return;
return -1;
}
}
av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
@@ -155,6 +155,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
push_samples(outlink);
return 0;
}
static int request_frame(AVFilterLink *outlink)