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确保完全忽略webrtc客户端不支持的rtp

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This commit is contained in:
ziyue 2021-10-15 17:37:51 +08:00
parent 82b78f4544
commit efa839c60b
1 changed files with 6 additions and 6 deletions
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12
webrtc/WebRtcPlayer.cpp
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@ -33,6 +33,12 @@ void WebRtcPlayer::onStartWebRTC() {
CHECK(_play_src);
WebRtcTransportImp::onStartWebRTC();
if (canSendRtp()) {
//确保该rtp codec类型对方支持
memset(_can_send_rtp, 0, sizeof(_can_send_rtp));
for (auto &m : _answer_sdp->media) {
_can_send_rtp[m.type] = m.direction == RtpDirection::sendonly || m.direction == RtpDirection::sendrecv;
}
_play_src->pause(false);
_reader = _play_src->getRing()->attach(getPoller(), true);
weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
@ -53,12 +59,6 @@ void WebRtcPlayer::onStartWebRTC() {
}
strongSelf->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
});
//确保该rtp codec类型对方支持
memset(_can_send_rtp, 0, sizeof(_can_send_rtp));
for (auto &m : _answer_sdp->media) {
_can_send_rtp[m.type] = m.direction == RtpDirection::sendonly || m.direction == RtpDirection::sendrecv;
}
}
//使用完毕后，释放强引用，这样确保推流器断开后能及时注销媒体
_play_src = nullptr;