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

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