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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-03-21 00:15:18 +0100 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-03-21 01:33:53 +0100 |
| commit | 0ebd83617fe008b7e9766f659cc3d9618b2d80d2 (patch) | |
| tree | 23bc388bf6b66cf58d7a90c0d2529e53ed984561 /libavformat/rtpenc.c | |
| parent | 745a33a44318ad6d6f74835a417397cdd9dda9a9 (diff) | |
| parent | c9594fe0fb6dd123fa25cb27fe5bc976ff3a9051 (diff) | |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rtpenc.c')
| -rw-r--r-- | libavformat/rtpenc.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 49da08b4e1..d7b6b3f124 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -126,7 +126,7 @@ static int rtp_write_header(AVFormatContext *s1) s->max_payload_size = s1->packet_size - 12; s->max_frames_per_packet = 0; - if (s1->max_delay) { + if (s1->max_delay > 0) { if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { int frame_size = av_get_audio_frame_duration(st->codec, 0); if (!frame_size) |