Add tracing to public PeerConnection methods.

talk/app/webrtc/peerconnection.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 33 matching lines @@
 #include "talk/app/webrtc/rtpsender.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/media/sctp/sctpdataengine.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/base/trace_event.h"
 #include "webrtc/p2p/client/basicportallocator.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace {
 
 using webrtc::DataChannel;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
 using webrtc::RtpSenderInterface;
@@ skipping 524 matching lines @@
       observer_(NULL),
       uma_observer_(NULL),
       signaling_state_(kStable),
       ice_state_(kIceNew),
       ice_connection_state_(kIceConnectionNew),
       ice_gathering_state_(kIceGatheringNew),
       local_streams_(StreamCollection::Create()),
       remote_streams_(StreamCollection::Create()) {}
 
 PeerConnection::~PeerConnection() {
+  TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
   RTC_DCHECK(signaling_thread()->IsCurrent());
   // Finish any pending deletions.
   signaling_thread()->Clear(this, MSG_DELETE, nullptr);
   // Need to detach RTP senders/receivers from WebRtcSession,
   // since it's about to be destroyed.
   for (const auto& sender : senders_) {
     sender->Stop();
   }
   for (const auto& receiver : receivers_) {
     receiver->Stop();
@@ skipping 23 matching lines @@
   return Initialize(configuration, constraints, allocator.Pass(),
                     dtls_identity_store.Pass(), observer);
 }
 
 bool PeerConnection::Initialize(
     const PeerConnectionInterface::RTCConfiguration& configuration,
     const MediaConstraintsInterface* constraints,
     rtc::scoped_ptr<cricket::PortAllocator> allocator,
     rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     PeerConnectionObserver* observer) {
+  TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
   RTC_DCHECK(observer != nullptr);
   if (!observer) {
     return false;
   }
   observer_ = observer;
 
   port_allocator_ = allocator.Pass();
 
   std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_config;
   std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_config;
@@ skipping 70 matching lines @@
 PeerConnection::local_streams() {
   return local_streams_;
 }
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::remote_streams() {
   return remote_streams_;
 }
 
 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
+  TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
   if (IsClosed()) {
     return false;
   }
   if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
     return false;
   }
 
   local_streams_->AddStream(local_stream);
 
   for (const auto& track : local_stream->GetAudioTracks()) {
@@ skipping 40 matching lines @@
   }
 
   stats_->AddStream(local_stream);
   observer_->OnRenegotiationNeeded();
   return true;
 }
 
 // TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
 // indefinitely, when we have unified plan SDP.
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
+  TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
   for (const auto& track : local_stream->GetAudioTracks()) {
     auto sender = FindSenderForTrack(track.get());
     if (sender == senders_.end()) {
       LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
                       << " doesn't exist.";
       continue;
     }
     (*sender)->Stop();
     senders_.erase(sender);
   }
@@ skipping 11 matching lines @@
   local_streams_->RemoveStream(local_stream);
 
   if (IsClosed()) {
     return;
   }
   observer_->OnRenegotiationNeeded();
 }
 
 rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
     AudioTrackInterface* track) {
+  TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
   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) {
+  TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
   RtpSenderInterface* new_sender;
   if (kind == MediaStreamTrackInterface::kAudioKind) {
     new_sender = new AudioRtpSender(session_.get(), stats_.get());
   } else if (kind == MediaStreamTrackInterface::kVideoKind) {
     new_sender = new VideoRtpSender(session_.get());
   } else {
     LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
     return rtc::scoped_refptr<RtpSenderInterface>();
   }
   senders_.push_back(new_sender);
@@ skipping 15 matching lines @@
   for (const auto& receiver : receivers_) {
     receivers.push_back(
         RtpReceiverProxy::Create(signaling_thread(), receiver.get()));
   }
   return receivers;
 }
 
 bool PeerConnection::GetStats(StatsObserver* observer,
                               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,
                            new GetStatsMsg(observer, track));
   return true;
@@ skipping 14 matching lines @@
 
 PeerConnectionInterface::IceGatheringState
 PeerConnection::ice_gathering_state() {
   return ice_gathering_state_;
 }
 
 rtc::scoped_refptr<DataChannelInterface>
 PeerConnection::CreateDataChannel(
     const std::string& label,
     const DataChannelInit* config) {
+  TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
   bool first_datachannel = !HasDataChannels();
 
   rtc::scoped_ptr<InternalDataChannelInit> internal_config;
   if (config) {
     internal_config.reset(new InternalDataChannelInit(*config));
   }
   rtc::scoped_refptr<DataChannelInterface> channel(
       InternalCreateDataChannel(label, internal_config.get()));
   if (!channel.get()) {
     return nullptr;
   }
 
   // Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
   // the first SCTP DataChannel.
   if (session_->data_channel_type() == cricket::DCT_RTP || first_datachannel) {
     observer_->OnRenegotiationNeeded();
   }
 
   return DataChannelProxy::Create(signaling_thread(), channel.get());
 }
 
 void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
                                  const MediaConstraintsInterface* constraints) {
+  TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
   if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
     return;
   }
   RTCOfferAnswerOptions options;
 
   bool value;
   size_t mandatory_constraints = 0;
 
   if (FindConstraint(constraints,
@@ skipping 31 matching lines @@
                      &value,
                      &mandatory_constraints)) {
     options.use_rtp_mux = value;
   }
 
   CreateOffer(observer, options);
 }
 
 void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
                                  const RTCOfferAnswerOptions& options) {
+  TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
   if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
     return;
   }
 
   cricket::MediaSessionOptions session_options;
   if (!GetOptionsForOffer(options, &session_options)) {
     std::string error = "CreateOffer called with invalid options.";
     LOG(LS_ERROR) << error;
     PostCreateSessionDescriptionFailure(observer, error);
     return;
   }
 
   session_->CreateOffer(observer, options, session_options);
 }
 
 void PeerConnection::CreateAnswer(
     CreateSessionDescriptionObserver* observer,
     const MediaConstraintsInterface* constraints) {
+  TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
   if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
     return;
   }
 
   cricket::MediaSessionOptions session_options;
   if (!GetOptionsForAnswer(constraints, &session_options)) {
     std::string error = "CreateAnswer called with invalid constraints.";
     LOG(LS_ERROR) << error;
     PostCreateSessionDescriptionFailure(observer, error);
     return;
   }
 
   session_->CreateAnswer(observer, constraints, session_options);
 }
 
 void PeerConnection::SetLocalDescription(
     SetSessionDescriptionObserver* observer,
     SessionDescriptionInterface* desc) {
+  TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
   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 57 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 (!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 81 matching lines @@
     UpdateEndedRemoteMediaStreams();
     remote_info_.msid_supported |= remote_streams_->count() > 0;
   }
   MaybeCreateDefaultStream();
 
   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
   signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
 }
 
 bool PeerConnection::SetConfiguration(const RTCConfiguration& config) {
+  TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
   if (port_allocator_) {
     std::vector<PortAllocatorFactoryInterface::StunConfiguration> stuns;
     std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turns;
     if (!ParseIceServers(config.servers, &stuns, &turns)) {
       return false;
     }
 
     cricket::ServerAddresses cricket_stuns;
     std::vector<cricket::RelayServerConfig> cricket_turns;
     ConvertToCricketIceServers(stuns, turns, &cricket_stuns, &cricket_turns);
     port_allocator_->SetIceServers(cricket_stuns, cricket_turns);
   }
   session_->SetIceConfig(session_->ParseIceConfig(config));
   return session_->SetIceTransports(config.type);
 }
 
 bool PeerConnection::AddIceCandidate(
     const IceCandidateInterface* ice_candidate) {
+  TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
   return session_->ProcessIceMessage(ice_candidate);
 }
 
 void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
+  TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
   uma_observer_ = observer;
 
   if (session_) {
     session_->set_metrics_observer(uma_observer_);
   }
 
   // Send information about IPv4/IPv6 status.
   if (uma_observer_ && port_allocator_) {
     if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
       uma_observer_->IncrementEnumCounter(
           kEnumCounterAddressFamily, kPeerConnection_IPv6,
           kPeerConnectionAddressFamilyCounter_Max);
     } else {
       uma_observer_->IncrementEnumCounter(
           kEnumCounterAddressFamily, kPeerConnection_IPv4,
           kPeerConnectionAddressFamilyCounter_Max);
     }
   }
 }
 
 const SessionDescriptionInterface* PeerConnection::local_description() const {
   return session_->local_description();
 }
 
 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) {
@@ skipping 759 matching lines @@
 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
   for (const auto& channel : sctp_data_channels_) {
     if (channel->id() == sid) {
       return channel;
     }
   }
   return nullptr;
 }
 
 }  // namespace webrtc