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authorMichael Niedermayer <michaelni@gmx.at>2011-12-01 02:44:19 +0100
committerMichael Niedermayer <michaelni@gmx.at>2011-12-01 02:54:24 +0100
commit9d76cf0b18976487d71e39bbdc1b53755e366535 (patch)
treed71801d63301c89e4c860eb2dee38b47348cd5b7 /libavformat/rtpenc.c
parent0275b75a7e705ef5a6bd6610f1450671f78000b6 (diff)
parentc8f0e88b205208da0e74f9345d4c4eb6d725774b (diff)
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.c6
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;