Add PeerConnection IsClosed check.

webrtc/api/peerconnection.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 572 matching lines @@
     return false;
   }
 
   media_controller_.reset(
       factory_->CreateMediaController(configuration.media_config));
 
   session_.reset(
       new WebRtcSession(media_controller_.get(), factory_->network_thread(),
                         factory_->worker_thread(), factory_->signaling_thread(),
                         port_allocator_.get()));
+
+  session_->SetTransportController(
+      factory_->CreateTransportController(port_allocator_.get()));
   stats_.reset(new StatsCollector(this));
 
   // Initialize the WebRtcSession. It creates transport channels etc.
   if (!session_->Initialize(factory_->options(), std::move(cert_generator),
                             configuration)) {
     return false;
   }
 
   // Register PeerConnection as receiver of local ice candidates.
   // All the callbacks will be posted to the application from PeerConnection.
@@ skipping 150 matching lines @@
   (*it)->Stop();
   senders_.erase(it);
 
   observer_->OnRenegotiationNeeded();
   return true;
 }
 
 rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
     AudioTrackInterface* track) {
   TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
+  if (IsClosed()) {
+    return nullptr;
+  }
   if (!track) {
     LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
     return NULL;
   }
   if (!local_streams_->FindAudioTrack(track->id())) {
     LOG(LS_ERROR) << "CreateDtmfSender is called with a non local audio track.";
     return NULL;
   }
 
   rtc::scoped_refptr<DtmfSenderInterface> sender(
       DtmfSender::Create(track, signaling_thread(), session_.get()));
   if (!sender.get()) {
     LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
     return NULL;
   }
   return DtmfSenderProxy::Create(signaling_thread(), sender.get());
 }
 
 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
     const std::string& kind,
     const std::string& stream_id) {
   TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
+  if (IsClosed()) {
+    return nullptr;
+  }
   rtc::scoped_refptr<RtpSenderInterface> new_sender;
   if (kind == MediaStreamTrackInterface::kAudioKind) {
     new_sender = RtpSenderProxy::Create(
         signaling_thread(), new AudioRtpSender(session_.get(), stats_.get()));
   } else if (kind == MediaStreamTrackInterface::kVideoKind) {
     new_sender = RtpSenderProxy::Create(signaling_thread(),
                                         new VideoRtpSender(session_.get()));
   } else {
     LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
     return new_sender;
@@ skipping 182 matching lines @@
     return;
   }
 
   session_->CreateAnswer(observer, session_options);
 }
 
 void PeerConnection::SetLocalDescription(
     SetSessionDescriptionObserver* observer,
     SessionDescriptionInterface* desc) {
   TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
+  if (IsClosed()) {
+    return;
+  }
   if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
     return;
   }
   if (!desc) {
     PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
     return;
   }
   // Update stats here so that we have the most recent stats for tracks and
   // streams that might be removed by updating the session description.
@@ skipping 58 matching lines @@
   // MaybeStartGathering needs to be called after posting
   // MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
   // before signaling that SetLocalDescription completed.
   session_->MaybeStartGathering();
 }
 
 void PeerConnection::SetRemoteDescription(
     SetSessionDescriptionObserver* observer,
     SessionDescriptionInterface* desc) {
   TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
+  if (IsClosed()) {
+    return;
+  }
   if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
     return;
   }
   if (!desc) {
     PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
     return;
   }
   // Update stats here so that we have the most recent stats for tracks and
   // streams that might be removed by updating the session description.
@@ skipping 100 matching lines @@
   }
 
   // TODO(deadbeef): Shouldn't have to hop to the worker thread twice...
   session_->SetIceConfig(session_->ParseIceConfig(configuration));
   return true;
 }
 
 bool PeerConnection::AddIceCandidate(
     const IceCandidateInterface* ice_candidate) {
   TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
+  if (IsClosed()) {
+    return false;
+  }
   return session_->ProcessIceMessage(ice_candidate);
 }
 
 bool PeerConnection::RemoveIceCandidates(
     const std::vector<cricket::Candidate>& candidates) {
   TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
   return session_->RemoveRemoteIceCandidates(candidates);
 }
 
 void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
@@ skipping 24 matching lines @@
 
 const SessionDescriptionInterface* PeerConnection::remote_description() const {
   return session_->remote_description();
 }
 
 void PeerConnection::Close() {
   TRACE_EVENT0("webrtc", "PeerConnection::Close");
   // Update stats here so that we have the most recent stats for tracks and
   // streams before the channels are closed.
   stats_->UpdateStats(kStatsOutputLevelStandard);
-
   session_->Close();
 }
 
 void PeerConnection::OnSessionStateChange(WebRtcSession* /*session*/,
                                           WebRtcSession::State state) {
   switch (state) {
     case WebRtcSession::STATE_INIT:
       ChangeSignalingState(PeerConnectionInterface::kStable);
       break;
     case WebRtcSession::STATE_SENTOFFER:
@@ skipping 120 matching lines @@
   RTC_DCHECK(signaling_thread()->IsCurrent());
   if (IsClosed()) {
     return;
   }
   ice_gathering_state_ = new_state;
   observer_->OnIceGatheringChange(ice_gathering_state_);
 }
 
 void PeerConnection::OnIceCandidate(const IceCandidateInterface* candidate) {
   RTC_DCHECK(signaling_thread()->IsCurrent());
+  if (IsClosed()) {
+    return;
+  }
   observer_->OnIceCandidate(candidate);
 }
 
 void PeerConnection::OnIceCandidatesRemoved(
     const std::vector<cricket::Candidate>& candidates) {
   RTC_DCHECK(signaling_thread()->IsCurrent());
+  if (IsClosed()) {
+    return;
+  }
   observer_->OnIceCandidatesRemoved(candidates);
 }
 
 void PeerConnection::OnIceConnectionReceivingChange(bool receiving) {
   RTC_DCHECK(signaling_thread()->IsCurrent());
+  if (IsClosed()) {
+    return;
+  }
   observer_->OnIceConnectionReceivingChange(receiving);
 }
 
 void PeerConnection::ChangeSignalingState(
     PeerConnectionInterface::SignalingState signaling_state) {
   signaling_state_ = signaling_state;
   if (signaling_state == kClosed) {
     ice_connection_state_ = kIceConnectionClosed;
     observer_->OnIceConnectionChange(ice_connection_state_);
     if (ice_gathering_state_ != kIceGatheringComplete) {
       ice_gathering_state_ = kIceGatheringComplete;
       observer_->OnIceGatheringChange(ice_gathering_state_);
     }
   }
   observer_->OnSignalingChange(signaling_state_);
 }
 
 void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
                                        MediaStreamInterface* stream) {
+  if (IsClosed()) {
+    return;
+  }
   auto sender = FindSenderForTrack(track);
   if (sender != senders_.end()) {
     // We already have a sender for this track, so just change the stream_id
     // so that it's correct in the next call to CreateOffer.
     (*sender)->set_stream_id(stream->label());
     return;
   }
 
   // Normal case; we've never seen this track before.
   rtc::scoped_refptr<RtpSenderInterface> new_sender = RtpSenderProxy::Create(
@@ skipping 10 matching lines @@
       FindTrackInfo(local_audio_tracks_, stream->label(), track->id());
   if (track_info) {
     new_sender->SetSsrc(track_info->ssrc);
   }
 }
 
 // TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
 // indefinitely, when we have unified plan SDP.
 void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
                                          MediaStreamInterface* stream) {
+  if (IsClosed()) {
+    return;
+  }
   auto sender = FindSenderForTrack(track);
   if (sender == senders_.end()) {
     LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
                     << " doesn't exist.";
     return;
   }
   (*sender)->Stop();
   senders_.erase(sender);
 }
 
 void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
                                        MediaStreamInterface* stream) {
+  if (IsClosed()) {
+    return;
+  }
   auto sender = FindSenderForTrack(track);
   if (sender != senders_.end()) {
     // We already have a sender for this track, so just change the stream_id
     // so that it's correct in the next call to CreateOffer.
     (*sender)->set_stream_id(stream->label());
     return;
   }
 
   // Normal case; we've never seen this track before.
   rtc::scoped_refptr<RtpSenderInterface> new_sender = RtpSenderProxy::Create(
       signaling_thread(),
       new VideoRtpSender(track, stream->label(), session_.get()));
   senders_.push_back(new_sender);
   const TrackInfo* track_info =
       FindTrackInfo(local_video_tracks_, stream->label(), track->id());
   if (track_info) {
     new_sender->SetSsrc(track_info->ssrc);
   }
 }
 
 void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
                                          MediaStreamInterface* stream) {
+  if (IsClosed()) {
+    return;
+  }
   auto sender = FindSenderForTrack(track);
   if (sender == senders_.end()) {
     LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
                     << " doesn't exist.";
     return;
   }
   (*sender)->Stop();
   senders_.erase(sender);
 }
 
@@ skipping 682 matching lines @@
   port_allocator_->set_candidate_filter(
       ConvertIceTransportTypeToCandidateFilter(configuration.type));
   // Call this last since it may create pooled allocator sessions using the
   // candidate filter set above.
   port_allocator_->SetConfiguration(stun_servers, turn_servers,
                                     configuration.ice_candidate_pool_size);
   return true;
 }
 
 }  // namespace webrtc