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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/mm.c
parent0275b75a7e705ef5a6bd6610f1450671f78000b6 (diff)
parentc8f0e88b205208da0e74f9345d4c4eb6d725774b (diff)
downloadffmpeg-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/mm.c')
-rw-r--r--libavformat/mm.c5
1 files changed, 3 insertions, 2 deletions
diff --git a/libavformat/mm.c b/libavformat/mm.c
index 3ad47418fc..8bb933babe 100644
--- a/libavformat/mm.c
+++ b/libavformat/mm.c
@@ -33,6 +33,7 @@
#include "libavutil/intreadwrite.h"
#include "avformat.h"
+#include "internal.h"
#define MM_PREAMBLE_SIZE 6
@@ -113,7 +114,7 @@ static int read_header(AVFormatContext *s,
st->codec->codec_tag = 0; /* no fourcc */
st->codec->width = width;
st->codec->height = height;
- av_set_pts_info(st, 64, 1, frame_rate);
+ avpriv_set_pts_info(st, 64, 1, frame_rate);
/* audio stream */
if (length == MM_HEADER_LEN_AV) {
@@ -125,7 +126,7 @@ static int read_header(AVFormatContext *s,
st->codec->codec_id = CODEC_ID_PCM_U8;
st->codec->channels = 1;
st->codec->sample_rate = 8000;
- av_set_pts_info(st, 64, 1, 8000); /* 8000 hz */
+ avpriv_set_pts_info(st, 64, 1, 8000); /* 8000 hz */
}
mm->audio_pts = 0;