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| 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 /libavformat/rtsp.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 'libavformat/rtsp.c')
| -rw-r--r-- | libavformat/rtsp.c | 12 |
1 files changed, 9 insertions, 3 deletions
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 18ec8a5edc..24be912045 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -228,7 +228,7 @@ static int sdp_parse_rtpmap(AVFormatContext *s, codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS; if (i > 0) { codec->sample_rate = i; - av_set_pts_info(st, 32, 1, codec->sample_rate); + avpriv_set_pts_info(st, 32, 1, codec->sample_rate); get_word_sep(buf, sizeof(buf), "/", &p); i = atoi(buf); if (i > 0) @@ -246,11 +246,14 @@ static int sdp_parse_rtpmap(AVFormatContext *s, case AVMEDIA_TYPE_VIDEO: av_log(s, AV_LOG_DEBUG, "video codec set to: %s\n", c_name); if (i > 0) - av_set_pts_info(st, 32, 1, i); + avpriv_set_pts_info(st, 32, 1, i); break; default: break; } + if (rtsp_st->dynamic_handler && rtsp_st->dynamic_handler->init) + rtsp_st->dynamic_handler->init(s, st->index, + rtsp_st->dynamic_protocol_context); return 0; } @@ -382,11 +385,14 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && st->codec->sample_rate > 0) - av_set_pts_info(st, 32, 1, st->codec->sample_rate); + avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); /* Even static payload types may need a custom depacketizer */ handler = ff_rtp_handler_find_by_id( rtsp_st->sdp_payload_type, st->codec->codec_type); init_rtp_handler(handler, rtsp_st, st->codec); + if (handler && handler->init) + handler->init(s, st->index, + rtsp_st->dynamic_protocol_context); } } /* put a default control url */ |