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author | Michael Niedermayer <michaelni@gmx.at> | 2011-12-01 02:44:19 +0100 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-12-01 02:54:24 +0100 |
commit | 9d76cf0b18976487d71e39bbdc1b53755e366535 (patch) | |
tree | d71801d63301c89e4c860eb2dee38b47348cd5b7 /libavdevice/oss_audio.c | |
parent | 0275b75a7e705ef5a6bd6610f1450671f78000b6 (diff) | |
parent | c8f0e88b205208da0e74f9345d4c4eb6d725774b (diff) | |
download | ffmpeg-streaming-9d76cf0b18976487d71e39bbdc1b53755e366535.zip ffmpeg-streaming-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 'libavdevice/oss_audio.c')
-rw-r--r-- | libavdevice/oss_audio.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c index 4432376..e3ab926 100644 --- a/libavdevice/oss_audio.c +++ b/libavdevice/oss_audio.c @@ -40,6 +40,7 @@ #include "libavutil/opt.h" #include "libavcodec/avcodec.h" #include "avdevice.h" +#include "libavformat/internal.h" #define AUDIO_BLOCK_SIZE 4096 @@ -225,7 +226,7 @@ static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) st->codec->sample_rate = s->sample_rate; st->codec->channels = s->channels; - av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ + avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ return 0; } |