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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8
libavfilter/af_aresample.c
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@@ -168,13 +168,14 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
int n_out = n_in * aresample->ratio * 2 ;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
int ret;
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
@@ -193,15 +194,16 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
if (n_out <= 0) {
avfilter_unref_buffer(outsamplesref);
avfilter_unref_buffer(insamplesref);
return;
return 0;
}
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
ff_filter_samples(outlink, outsamplesref);
ret = ff_filter_samples(outlink, outsamplesref);
aresample->req_fullfilled= 1;
avfilter_unref_buffer(insamplesref);
return ret;
}
static int request_frame(AVFilterLink *outlink)