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| author | Michael Niedermayer <michaelni@gmx.at> | 2011-12-01 05:44:19 +0400 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2011-12-01 05:54:24 +0400 |
| commit | 9d76cf0b18976487d71e39bbdc1b53755e366535 (patch) | |
| tree | d71801d63301c89e4c860eb2dee38b47348cd5b7 /libavformat/flic.c | |
| parent | 0275b75a7e705ef5a6bd6610f1450671f78000b6 (diff) | |
| parent | c8f0e88b205208da0e74f9345d4c4eb6d725774b (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/flic.c')
| -rw-r--r-- | libavformat/flic.c | 11 |
1 files changed, 6 insertions, 5 deletions
diff --git a/libavformat/flic.c b/libavformat/flic.c index f4c1a1b9fa..89652a20f6 100644 --- a/libavformat/flic.c +++ b/libavformat/flic.c @@ -34,6 +34,7 @@ #include "libavutil/intreadwrite.h" #include "libavutil/audioconvert.h" #include "avformat.h" +#include "internal.h" #define FLIC_FILE_MAGIC_1 0xAF11 #define FLIC_FILE_MAGIC_2 0xAF12 @@ -167,10 +168,10 @@ static int flic_read_header(AVFormatContext *s, /* Since the header information is incorrect we have to figure out the * framerate using block_align and the fact that the audio is 22050 Hz. * We usually have two cases: 2205 -> 10 fps and 1470 -> 15 fps */ - av_set_pts_info(st, 64, ast->codec->block_align, FLIC_TFTD_SAMPLE_RATE); - av_set_pts_info(ast, 64, 1, FLIC_TFTD_SAMPLE_RATE); + avpriv_set_pts_info(st, 64, ast->codec->block_align, FLIC_TFTD_SAMPLE_RATE); + avpriv_set_pts_info(ast, 64, 1, FLIC_TFTD_SAMPLE_RATE); } else if (AV_RL16(&header[0x10]) == FLIC_CHUNK_MAGIC_1) { - av_set_pts_info(st, 64, FLIC_MC_SPEED, 70); + avpriv_set_pts_info(st, 64, FLIC_MC_SPEED, 70); /* rewind the stream since the first chunk is at offset 12 */ avio_seek(pb, 12, SEEK_SET); @@ -182,10 +183,10 @@ static int flic_read_header(AVFormatContext *s, memcpy(st->codec->extradata, header, 12); } else if (magic_number == FLIC_FILE_MAGIC_1) { - av_set_pts_info(st, 64, speed, 70); + avpriv_set_pts_info(st, 64, speed, 70); } else if ((magic_number == FLIC_FILE_MAGIC_2) || (magic_number == FLIC_FILE_MAGIC_3)) { - av_set_pts_info(st, 64, speed, 1000); + avpriv_set_pts_info(st, 64, speed, 1000); } else { av_log(s, AV_LOG_INFO, "Invalid or unsupported magic chunk in file\n"); return AVERROR_INVALIDDATA; |