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| author | Michael Niedermayer <michaelni@gmx.at> | 2011-12-01 02:44:19 +0100 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-12-01 02:54:24 +0100 |
| commit | 9d76cf0b18976487d71e39bbdc1b53755e366535 (patch) | |
| tree | d71801d63301c89e4c860eb2dee38b47348cd5b7 /libavformat/rtpenc.c | |
| parent | 0275b75a7e705ef5a6bd6610f1450671f78000b6 (diff) | |
| parent | c8f0e88b205208da0e74f9345d4c4eb6d725774b (diff) | |
| download | ffmpeg-9d76cf0b18976487d71e39bbdc1b53755e366535.tar.gz | |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rtpenc.c')
| -rw-r--r-- | libavformat/rtpenc.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 83f728bc37..73ac76fae7 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -130,7 +130,7 @@ static int rtp_write_header(AVFormatContext *s1) } } - av_set_pts_info(st, 32, 1, 90000); + avpriv_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { case CODEC_ID_MP2: case CODEC_ID_MP3: @@ -166,7 +166,7 @@ static int rtp_write_header(AVFormatContext *s1) case CODEC_ID_ADPCM_G722: /* Due to a historical error, the clock rate for G722 in RTP is * 8000, even if the sample rate is 16000. See RFC 3551. */ - av_set_pts_info(st, 32, 1, 8000); + avpriv_set_pts_info(st, 32, 1, 8000); break; case CODEC_ID_AMR_NB: case CODEC_ID_AMR_WB: @@ -190,7 +190,7 @@ static int rtp_write_header(AVFormatContext *s1) default: defaultcase: if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - av_set_pts_info(st, 32, 1, st->codec->sample_rate); + avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); } s->buf_ptr = s->buf; break; |