Adding more detail to MessageQueue::Dispatch logging.

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 542 matching lines @@
   for (const auto& receiver : receivers_) {
     receiver->Stop();
   }
   // Destroy stats_ because it depends on session_.
   stats_.reset(nullptr);
   // Now destroy session_ before destroying other members,
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   session_.reset(nullptr);
   // port_allocator_ lives on the worker thread and should be destroyed there.
-  worker_thread()->Invoke<void>([this] { port_allocator_.reset(nullptr); });
+  worker_thread()->Invoke<void>(FROM_HERE,
+                                [this] { port_allocator_.reset(nullptr); });
 }
 
 bool PeerConnection::Initialize(
     const PeerConnectionInterface::RTCConfiguration& configuration,
     std::unique_ptr<cricket::PortAllocator> allocator,
     std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     PeerConnectionObserver* observer) {
   TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
   RTC_DCHECK(observer != nullptr);
   if (!observer) {
     return false;
   }
   observer_ = observer;
 
   port_allocator_ = std::move(allocator);
 
   // The port allocator lives on the worker thread and should be initialized
   // there.
-  if (!worker_thread()->Invoke<bool>(rtc::Bind(
-          &PeerConnection::InitializePortAllocator_w, this, configuration))) {
+  if (!worker_thread()->Invoke<bool>(
+          FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_w, this,
+                               configuration))) {
     return false;
   }
 
   media_controller_.reset(
       factory_->CreateMediaController(configuration.media_config));
 
   session_.reset(
       new WebRtcSession(media_controller_.get(), factory_->signaling_thread(),
                         factory_->worker_thread(), port_allocator_.get()));
   stats_.reset(new StatsCollector(this));
@@ skipping 220 matching lines @@
                               MediaStreamTrackInterface* track,
                               StatsOutputLevel level) {
   TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
   RTC_DCHECK(signaling_thread()->IsCurrent());
   if (!VERIFY(observer != NULL)) {
     LOG(LS_ERROR) << "GetStats - observer is NULL.";
     return false;
   }
 
   stats_->UpdateStats(level);
-  signaling_thread()->Post(this, MSG_GETSTATS,
+  signaling_thread()->Post(FROM_HERE, this, MSG_GETSTATS,
                            new GetStatsMsg(observer, track));
   return true;
 }
 
 PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
   return signaling_state_;
 }
 
 PeerConnectionInterface::IceState PeerConnection::ice_state() {
   return ice_state_;
@@ skipping 208 matching lines @@
     const cricket::DataContentDescription* data_desc =
         static_cast<const cricket::DataContentDescription*>(
             data_content->description);
     if (rtc::starts_with(data_desc->protocol().data(),
                          cricket::kMediaProtocolRtpPrefix)) {
       UpdateLocalRtpDataChannels(data_desc->streams());
     }
   }
 
   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
-  signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
+  signaling_thread()->Post(FROM_HERE, this, MSG_SET_SESSIONDESCRIPTION_SUCCESS,
+                           msg);
 
   // 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) {
@@ skipping 86 matching lines @@
   // Iterate new_streams and notify the observer about new MediaStreams.
   for (size_t i = 0; i < new_streams->count(); ++i) {
     MediaStreamInterface* new_stream = new_streams->at(i);
     stats_->AddStream(new_stream);
     observer_->OnAddStream(new_stream);
   }
 
   UpdateEndedRemoteMediaStreams();
 
   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
-  signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
+  signaling_thread()->Post(FROM_HERE, this, MSG_SET_SESSIONDESCRIPTION_SUCCESS,
+                           msg);
 }
 
 bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration) {
   TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
   if (port_allocator_) {
     if (!worker_thread()->Invoke<bool>(
-            rtc::Bind(&PeerConnection::ReconfigurePortAllocator_w, this,
-                      configuration))) {
+            FROM_HERE, rtc::Bind(&PeerConnection::ReconfigurePortAllocator_w,
+                                 this, configuration))) {
       return false;
     }
   }
 
   // TODO(deadbeef): Shouldn't have to hop to the worker thread twice...
   session_->SetIceConfig(session_->ParseIceConfig(configuration));
   return true;
 }
 
 bool PeerConnection::AddIceCandidate(
@@ skipping 286 matching lines @@
   }
   (*sender)->Stop();
   senders_.erase(sender);
 }
 
 void PeerConnection::PostSetSessionDescriptionFailure(
     SetSessionDescriptionObserver* observer,
     const std::string& error) {
   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
   msg->error = error;
-  signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
+  signaling_thread()->Post(FROM_HERE, this, MSG_SET_SESSIONDESCRIPTION_FAILED,
+                           msg);
 }
 
 void PeerConnection::PostCreateSessionDescriptionFailure(
     CreateSessionDescriptionObserver* observer,
     const std::string& error) {
   CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
   msg->error = error;
-  signaling_thread()->Post(this, MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
+  signaling_thread()->Post(FROM_HERE, this,
+                           MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
 }
 
 bool PeerConnection::GetOptionsForOffer(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
     cricket::MediaSessionOptions* session_options) {
   // TODO(deadbeef): Once we have transceivers, enumerate them here instead of
   // ContentInfos.
   if (session_->local_description()) {
     for (const cricket::ContentInfo& content :
          session_->local_description()->description()->contents()) {
@@ skipping 471 matching lines @@
   for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
        ++it) {
     if (it->get() == channel) {
       if (channel->id() >= 0) {
         sid_allocator_.ReleaseSid(channel->id());
       }
       // Since this method is triggered by a signal from the DataChannel,
       // we can't free it directly here; we need to free it asynchronously.
       sctp_data_channels_to_free_.push_back(*it);
       sctp_data_channels_.erase(it);
-      signaling_thread()->Post(this, MSG_FREE_DATACHANNELS, nullptr);
+      signaling_thread()->Post(FROM_HERE, this, MSG_FREE_DATACHANNELS, nullptr);
       return;
     }
   }
 }
 
 void PeerConnection::OnVoiceChannelDestroyed() {
   StopReceivers(cricket::MEDIA_TYPE_AUDIO);
 }
 
 void PeerConnection::OnVideoChannelDestroyed() {
@@ skipping 151 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