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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2025-11-23 21:54:53 +02:00

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>
This commit is contained in:
Michael Niedermayer
2012-07-09 22:10:38 +02:00
45 changed files with 648 additions and 339 deletions

View File

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