diff options
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 /libavdevice/alsa-audio-dec.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 'libavdevice/alsa-audio-dec.c')
-rw-r--r-- | libavdevice/alsa-audio-dec.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/libavdevice/alsa-audio-dec.c b/libavdevice/alsa-audio-dec.c index bd00b1535b..bb9d233a4e 100644 --- a/libavdevice/alsa-audio-dec.c +++ b/libavdevice/alsa-audio-dec.c @@ -46,6 +46,7 @@ */ #include <alsa/asoundlib.h> +#include "libavformat/internal.h" #include "libavutil/opt.h" #include "libavutil/mathematics.h" @@ -80,7 +81,7 @@ static av_cold int audio_read_header(AVFormatContext *s1, st->codec->codec_id = codec_id; 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 */ o = 2 * M_PI * s->period_size / s->sample_rate * 1.5; // bandwidth: 1.5Hz s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate, sqrt(2 * o), o * o); |