diff options
| -rw-r--r-- | libavformat/rtpdec.c | 5 | ||||
| -rw-r--r-- | libavformat/rtsp.c | 10 |
2 files changed, 8 insertions, 7 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index b7bdc0b..77b59a3 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -393,7 +393,6 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r return NULL; } } else { - av_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { case CODEC_ID_MPEG1VIDEO: case CODEC_ID_MPEG2VIDEO: @@ -405,16 +404,12 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r st->need_parsing = AVSTREAM_PARSE_FULL; break; case CODEC_ID_ADPCM_G722: - av_set_pts_info(st, 32, 1, st->codec->sample_rate); /* According to RFC 3551, the stream clock rate is 8000 * even if the sample rate is 16000. */ if (st->codec->sample_rate == 8000) st->codec->sample_rate = 16000; break; default: - if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - av_set_pts_info(st, 32, 1, st->codec->sample_rate); - } break; } } diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index e4269de..1dd166e 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -135,9 +135,10 @@ static void init_rtp_handler(RTPDynamicProtocolHandler *handler, /* parse the rtpmap description: <codec_name>/<clock_rate>[/<other params>] */ static int sdp_parse_rtpmap(AVFormatContext *s, - AVCodecContext *codec, RTSPStream *rtsp_st, + AVStream *st, RTSPStream *rtsp_st, int payload_type, const char *p) { + AVCodecContext *codec = st->codec; char buf[256]; int i; AVCodec *c; @@ -181,6 +182,7 @@ 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); get_word_sep(buf, sizeof(buf), "/", &p); i = atoi(buf); if (i > 0) @@ -197,6 +199,8 @@ static int sdp_parse_rtpmap(AVFormatContext *s, break; 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); break; default: break; @@ -329,6 +333,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, RTPDynamicProtocolHandler *handler; /* if standard payload type, we can find the codec right now */ ff_rtp_get_codec_info(st->codec, rtsp_st->sdp_payload_type); + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) + av_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); @@ -371,7 +377,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1, payload_type = atoi(buf1); st = s->streams[s->nb_streams - 1]; rtsp_st = st->priv_data; - sdp_parse_rtpmap(s, st->codec, rtsp_st, payload_type, p); + sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p); } else if (av_strstart(p, "fmtp:", &p) || av_strstart(p, "framesize:", &p)) { /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */ |
