external/srs
1
0
Fork 0
mirror of https://github.com/ossrs/srs.git synced 2025-03-09 15:49:59 +00:00
Code Issues Releases Wiki Activity

RTC: Generate timestamp only when transcode opus to aac.

Browse source
This commit is contained in:
winlin 2020-06-16 13:39:02 +08:00
parent 2c1bd6da3e
commit cf738754ae
4 changed files with 6 additions and 7 deletions
Download patch file Download diff file Expand all files Collapse all files

6
trunk/src/app/srs_app_rtc_conn.cpp
View file

@ -494,11 +494,8 @@ SrsRtcPlayer::SrsRtcPlayer(SrsRtcSession* s, int parent_cid)
session_ = s;
audio_timestamp = 0;
audio_sequence = 0;
video_sequence = 0;
mw_msgs = 0;
realtime = true;
@ -711,15 +708,12 @@ srs_error_t SrsRtcPlayer::send_packets(SrsRtcSource* source, const vector<SrsRtp
if (pkt->is_audio()) {
info.nn_audios++;
pkt->header.set_timestamp(audio_timestamp);
pkt->header.set_sequence(audio_sequence++);
pkt->header.set_ssrc(audio_ssrc);
pkt->header.set_payload_type(audio_payload_type);
// TODO: FIXME: Padding audio to the max payload in RTP packets.
// TODO: FIXME: Why 960? Need Refactoring?
audio_timestamp += 960;
continue;
}