修复无人观看主动关闭webrtc推流线程安全的问题

This commit is contained in:
ziyue 2021-09-15 14:56:58 +08:00
parent 3c4285a056
commit 704ea59502
2 changed files with 10 additions and 3 deletions

View File

@ -75,7 +75,6 @@ void WebRtcSession::onError(const SockException &err) {
//在udp链接迁移时新的WebRtcSession对象将接管WebRtcTransport对象的生命周期
//本WebRtcSession对象将在超时后自动销毁
WarnP(this) << err.what();
_transport = nullptr;
}
void WebRtcSession::onManager() {

View File

@ -489,7 +489,7 @@ void WebRtcTransportImp::onStartWebRTC() {
if (!strongSelf) {
return;
}
strongSelf->onShutdown(SockException(Err_eof, "rtsp ring buffer detached"));
strongSelf->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
});
RtcSession rtsp_send_sdp;
@ -956,6 +956,8 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx
void WebRtcTransportImp::onShutdown(const SockException &ex){
WarnL << ex.what();
unrefSelf();
//触发发送dtls close通知
WebRtcTransport::onDestory();
auto session = _session.lock();
if (session) {
session->shutdown(ex);
@ -970,7 +972,13 @@ bool WebRtcTransportImp::close(MediaSource &sender, bool force) {
return false;
}
string err = StrPrinter << "close media:" << sender.getSchema() << "/" << sender.getVhost() << "/" << sender.getApp() << "/" << sender.getId() << " " << force;
onShutdown(SockException(Err_shutdown,err));
weak_ptr<WebRtcTransportImp> weak_self = static_pointer_cast<WebRtcTransportImp>(shared_from_this());
getPoller()->async([weak_self, err]() {
auto strong_self = weak_self.lock();
if (strong_self) {
strong_self->onShutdown(SockException(Err_shutdown, err));
}
});
return true;
}