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Merge remote-tracking branch 'qatar/master'

Browse Source 
* 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>
This commit is contained in:
Michael Niedermayer
2011-12-01 02:44:19 +01:00
parent 0275b75a7e c8f0e88b20
commit 9d76cf0b18
146 changed files with 569 additions and 467 deletions
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6
libavformat/rtpenc.c
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@ -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;