Moving MediaStreamSignaling logic into PeerConnection.

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 11 matching lines @@
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include "talk/app/webrtc/peerconnection.h"
 
 #include <vector>
 
+#include "talk/app/webrtc/audiotrack.h"
 #include "talk/app/webrtc/dtmfsender.h"
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
+#include "talk/app/webrtc/mediastream.h"
+#include "talk/app/webrtc/mediastreamproxy.h"
+#include "talk/app/webrtc/mediastreamtrackproxy.h"
+#include "talk/app/webrtc/remoteaudiosource.h"
+#include "talk/app/webrtc/remotevideocapturer.h"
 #include "talk/app/webrtc/rtpreceiver.h"
 #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 "webrtc/p2p/client/basicportallocator.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
+#include "webrtc/base/stringutils.h"
 #include "webrtc/system_wrappers/interface/field_trial.h"
 
 namespace {
 
+using webrtc::DataChannel;
+using webrtc::MediaConstraintsInterface;
+using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
+using webrtc::StreamCollection;
+
+static const char kDefaultStreamLabel[] = "default";
+static const char kDefaultAudioTrackLabel[] = "defaulta0";
+static const char kDefaultVideoTrackLabel[] = "defaultv0";
 
 // The min number of tokens must present in Turn host uri.
 // e.g. user@turn.example.org
 static const size_t kTurnHostTokensNum = 2;
 // Number of tokens must be preset when TURN uri has transport param.
 static const size_t kTurnTransportTokensNum = 2;
 // The default stun port.
 static const int kDefaultStunPort = 3478;
 static const int kDefaultStunTlsPort = 5349;
 static const char kTransport[] = "transport";
 static const char kUdpTransportType[] = "udp";
 static const char kTcpTransportType[] = "tcp";
 
 // NOTE: Must be in the same order as the ServiceType enum.
 static const char* kValidIceServiceTypes[] = {
     "stun", "stuns", "turn", "turns", "invalid" };
 
 enum ServiceType {
   STUN,     // Indicates a STUN server.
   STUNS,    // Indicates a STUN server used with a TLS session.
   TURN,     // Indicates a TURN server
   TURNS,    // Indicates a TURN server used with a TLS session.
   INVALID,  // Unknown.
 };
 
 enum {
   MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
   MSG_SET_SESSIONDESCRIPTION_FAILED,
+  MSG_CREATE_SESSIONDESCRIPTION_FAILED,
   MSG_GETSTATS,
 };
 
 struct SetSessionDescriptionMsg : public rtc::MessageData {
   explicit SetSessionDescriptionMsg(
       webrtc::SetSessionDescriptionObserver* observer)
       : observer(observer) {
   }
 
   rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
   std::string error;
 };
 
+struct CreateSessionDescriptionMsg : public rtc::MessageData {
+  explicit CreateSessionDescriptionMsg(
+      webrtc::CreateSessionDescriptionObserver* observer)
+      : observer(observer) {}
+
+  rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
+  std::string error;
+};
+
 struct GetStatsMsg : public rtc::MessageData {
   GetStatsMsg(webrtc::StatsObserver* observer,
               webrtc::MediaStreamTrackInterface* track)
       : observer(observer), track(track) {
   }
   rtc::scoped_refptr<webrtc::StatsObserver> observer;
   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
 };
 
 // |in_str| should be of format
@@ skipping 203 matching lines @@
         return false;
       }
     } else {
       LOG(WARNING) << "Empty uri.";
     }
   }
   return true;
 }
 
 // Check if we can send |new_stream| on a PeerConnection.
-// Currently only one audio but multiple video track is supported per
-// PeerConnection.
 bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
                             webrtc::MediaStreamInterface* new_stream) {
-  if (!new_stream || !current_streams)
+  if (!new_stream || !current_streams) {
     return false;
-  if (current_streams->find(new_stream->label()) != NULL) {
+  }
+  if (current_streams->find(new_stream->label()) != nullptr) {
     LOG(LS_ERROR) << "MediaStream with label " << new_stream->label()
                   << " is already added.";
     return false;
   }
-
   return true;
 }
 
+bool MediaContentDirectionHasSend(cricket::MediaContentDirection dir) {
+  return dir == cricket::MD_SENDONLY || dir == cricket::MD_SENDRECV;
+}
+
+bool IsValidOfferToReceiveMedia(int value) {
+  typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
+  return (value >= Options::kUndefined) &&
+         (value <= Options::kMaxOfferToReceiveMedia);
+}
+
+// Add the stream and RTP data channel info to |session_options|.
+void SetStreams(cricket::MediaSessionOptions* session_options,
+                rtc::scoped_refptr<StreamCollection> streams,
+                const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
+                    rtp_data_channels) {
+  session_options->streams.clear();
+  if (streams != nullptr) {
+    for (size_t i = 0; i < streams->count(); ++i) {
+      MediaStreamInterface* stream = streams->at(i);
+      // For each audio track in the stream, add it to the MediaSessionOptions.
+      for (const auto& track : stream->GetAudioTracks()) {
+        session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, track->id(),
+                                       stream->label());
+      }
+      // For each video track in the stream, add it to the MediaSessionOptions.
+      for (const auto& track : stream->GetVideoTracks()) {
+        session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, track->id(),
+                                       stream->label());
+      }
+    }
+  }
+
+  // Check for data channels.
+  for (const auto& kv : rtp_data_channels) {
+    const DataChannel* channel = kv.second;
+    if (channel->state() == DataChannel::kConnecting ||
+        channel->state() == DataChannel::kOpen) {
+      // |streamid| and |sync_label| are both set to the DataChannel label
+      // here so they can be signaled the same way as MediaStreams and Tracks.
+      // For MediaStreams, the sync_label is the MediaStream label and the
+      // track label is the same as |streamid|.
+      const std::string& streamid = channel->label();
+      const std::string& sync_label = channel->label();
+      session_options->AddSendStream(cricket::MEDIA_TYPE_DATA, streamid,
+                                     sync_label);
+    }
+  }
+}
+
 }  // namespace
 
 namespace webrtc {
 
+// Factory class for creating remote MediaStreams and MediaStreamTracks.
+class RemoteMediaStreamFactory {
+ public:
+  explicit RemoteMediaStreamFactory(rtc::Thread* signaling_thread,
+                                    cricket::ChannelManager* channel_manager)
+      : signaling_thread_(signaling_thread),
+        channel_manager_(channel_manager) {}
+
+  rtc::scoped_refptr<MediaStreamInterface> CreateMediaStream(
+      const std::string& stream_label) {
+    return MediaStreamProxy::Create(signaling_thread_,
+                                    MediaStream::Create(stream_label));
+  }
+
+  AudioTrackInterface* AddAudioTrack(webrtc::MediaStreamInterface* stream,
+                                     const std::string& track_id) {
+    return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(
+        stream, track_id, RemoteAudioSource::Create().get());
+  }
+
+  VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream,
+                                     const std::string& track_id) {
+    return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>(
+        stream, track_id,
+        VideoSource::Create(channel_manager_, new RemoteVideoCapturer(),
+                            nullptr)
+            .get());
+  }
+
+ private:
+  template <typename TI, typename T, typename TP, typename S>
+  TI* AddTrack(MediaStreamInterface* stream,
+               const std::string& track_id,
+               S* source) {
+    rtc::scoped_refptr<TI> track(
+        TP::Create(signaling_thread_, T::Create(track_id, source)));
+    track->set_state(webrtc::MediaStreamTrackInterface::kLive);
+    if (stream->AddTrack(track)) {
+      return track;
+    }
+    return nullptr;
+  }
+
+  rtc::Thread* signaling_thread_;
+  cricket::ChannelManager* channel_manager_;
+};
+
+bool ConvertRtcOptionsForOffer(
+    const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
+    cricket::MediaSessionOptions* session_options) {
+  typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
+  if (!IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) ||
+      !IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video)) {
+    return false;
+  }
+
+  // According to the spec, offer to receive audio/video if the constraint is
+  // not set and there are send streams.
+  if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
+    session_options->recv_audio =
+        session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO);
+  } else {
+    session_options->recv_audio = (rtc_options.offer_to_receive_audio > 0);
+  }
+  if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
+    session_options->recv_video =
+        session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO);
+  } else {
+    session_options->recv_video = (rtc_options.offer_to_receive_video > 0);
+  }
+
+  session_options->vad_enabled = rtc_options.voice_activity_detection;
+  session_options->transport_options.ice_restart = rtc_options.ice_restart;
+  session_options->bundle_enabled =
+      rtc_options.use_rtp_mux &&
+      (session_options->has_audio() || session_options->has_video() ||
+       session_options->has_data());
+
+  return true;
+}
+
+bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
+                               cricket::MediaSessionOptions* session_options) {
+  bool value = false;
+  size_t mandatory_constraints_satisfied = 0;
+
+  // kOfferToReceiveAudio defaults to true according to spec.
+  if (!FindConstraint(constraints,
+                      MediaConstraintsInterface::kOfferToReceiveAudio, &value,
+                      &mandatory_constraints_satisfied) ||
+      value) {
+    session_options->recv_audio = true;
+  }
+
+  // kOfferToReceiveVideo defaults to false according to spec. But
+  // if it is an answer and video is offered, we should still accept video
+  // per default.
+  value = false;
+  if (!FindConstraint(constraints,
+                      MediaConstraintsInterface::kOfferToReceiveVideo, &value,
+                      &mandatory_constraints_satisfied) ||
+      value) {
+    session_options->recv_video = true;
+  }
+
+  if (FindConstraint(constraints,
+                     MediaConstraintsInterface::kVoiceActivityDetection, &value,
+                     &mandatory_constraints_satisfied)) {
+    session_options->vad_enabled = value;
+  }
+
+  if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
+                     &mandatory_constraints_satisfied)) {
+    session_options->bundle_enabled = value;
+  } else {
+    // kUseRtpMux defaults to true according to spec.
+    session_options->bundle_enabled = true;
+  }
+  session_options->bundle_enabled =
+      session_options->bundle_enabled &&
+      (session_options->has_audio() || session_options->has_video() ||
+       session_options->has_data());
+
+  if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
+                     &value, &mandatory_constraints_satisfied)) {
+    session_options->transport_options.ice_restart = value;
+  } else {
+    // kIceRestart defaults to false according to spec.
+    session_options->transport_options.ice_restart = false;
+  }
+
+  if (!constraints) {
+    return true;
+  }
+  return mandatory_constraints_satisfied == constraints->GetMandatory().size();
+}
+
 PeerConnection::PeerConnection(PeerConnectionFactory* factory)
     : factory_(factory),
       observer_(NULL),
       uma_observer_(NULL),
       signaling_state_(kStable),
       ice_state_(kIceNew),
       ice_connection_state_(kIceConnectionNew),
-      ice_gathering_state_(kIceGatheringNew) {
-}
+      ice_gathering_state_(kIceGatheringNew),
+      local_streams_(StreamCollection::Create()),
+      remote_streams_(StreamCollection::Create()) {}
 
 PeerConnection::~PeerConnection() {
   ASSERT(signaling_thread()->IsCurrent());
-  if (mediastream_signaling_) {
-    mediastream_signaling_->TearDown();
-  }

[Taylor Brandstetter, 2015/10/07 00:26:19]

The stuff previously done in mediastream_signaling->TearDown() will now happen
when WebRtcSession is destroyed and signals that its channels are closed.

   // 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();
   }
 }
 
 bool PeerConnection::Initialize(
     const PeerConnectionInterface::RTCConfiguration& configuration,
     const MediaConstraintsInterface* constraints,
     PortAllocatorFactoryInterface* allocator_factory,
     rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
     PeerConnectionObserver* observer) {
-  ASSERT(observer != NULL);
-  if (!observer)
+  ASSERT(observer != nullptr);
+  if (!observer) {
     return false;
+  }
   observer_ = observer;
 
   std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_config;
   std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_config;
   if (!ParseIceServers(configuration.servers, &stun_config, &turn_config)) {
     return false;
   }
   port_allocator_.reset(
       allocator_factory->CreatePortAllocator(stun_config, turn_config));
 
   // To handle both internal and externally created port allocator, we will
   // enable BUNDLE here.
   int portallocator_flags = port_allocator_->flags();
   portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
                          cricket::PORTALLOCATOR_ENABLE_IPV6;
   bool value;
   // If IPv6 flag was specified, we'll not override it by experiment.
-  if (FindConstraint(
-          constraints, MediaConstraintsInterface::kEnableIPv6, &value, NULL)) {
+  if (FindConstraint(constraints, MediaConstraintsInterface::kEnableIPv6,
+                     &value, nullptr)) {
     if (!value) {
       portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
     }
   } else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default") ==
              "Disabled") {
     portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
   }
 
   if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
     portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
     LOG(LS_INFO) << "TCP candidates are disabled.";
   }
 
   port_allocator_->set_flags(portallocator_flags);
   // No step delay is used while allocating ports.
   port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
 
-  mediastream_signaling_.reset(new MediaStreamSignaling(
-      factory_->signaling_thread(), this, factory_->channel_manager()));
+  remote_stream_factory_.reset(new RemoteMediaStreamFactory(
+      factory_->signaling_thread(), factory_->channel_manager()));
 
-  session_.reset(new WebRtcSession(factory_->channel_manager(),
-                                   factory_->signaling_thread(),
-                                   factory_->worker_thread(),
-                                   port_allocator_.get(),
-                                   mediastream_signaling_.get()));
-  stats_.reset(new StatsCollector(session_.get()));
+  session_.reset(new WebRtcSession(
+      factory_->channel_manager(), factory_->signaling_thread(),
+      factory_->worker_thread(), port_allocator_.get()));
+  stats_.reset(new StatsCollector(this, session_.get()));
 
   // Initialize the WebRtcSession. It creates transport channels etc.
   if (!session_->Initialize(factory_->options(), constraints,
-                            dtls_identity_store.Pass(), configuration))
+                            dtls_identity_store.Pass(), configuration)) {
     return false;
+  }
 
   // Register PeerConnection as receiver of local ice candidates.
   // All the callbacks will be posted to the application from PeerConnection.
   session_->RegisterIceObserver(this);
   session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
+  session_->SignalVoiceChannelDestroyed.connect(
+      this, &PeerConnection::OnVoiceChannelDestroyed);
+  session_->SignalVideoChannelDestroyed.connect(
+      this, &PeerConnection::OnVideoChannelDestroyed);
+  session_->SignalDataChannelCreated.connect(
+      this, &PeerConnection::OnDataChannelCreated);
+  session_->SignalDataChannelDestroyed.connect(
+      this, &PeerConnection::OnDataChannelDestroyed);
+  session_->SignalDataChannelCreationRequested.connect(
+      this, &PeerConnection::OnDataChannelCreationRequested);
   return true;
 }
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::local_streams() {
-  return mediastream_signaling_->local_streams();
+  return local_streams_;
 }
 
 rtc::scoped_refptr<StreamCollectionInterface>
 PeerConnection::remote_streams() {
-  return mediastream_signaling_->remote_streams();
+  return remote_streams_;
 }
 
 // TODO(deadbeef): Create RtpSenders immediately here, even if local
 // description hasn't yet been set.
 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
   if (IsClosed()) {
     return false;
   }
-  if (!CanAddLocalMediaStream(mediastream_signaling_->local_streams(),
-                              local_stream))
-    return false;
-
-  if (!mediastream_signaling_->AddLocalStream(local_stream)) {
+  if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
     return false;
   }
+
+  local_streams_->AddStream(local_stream);
+
+  // Find tracks that have already been configured in SDP. This can occur if a
+  // local session description that contains the MSID of these tracks is set
+  // before AddLocalStream is called. It can also occur if the local session
+  // description is not changed and RemoveLocalStream  is called and later
+  // AddLocalStream is called again with the same stream.
+  for (const auto& track : local_stream->GetAudioTracks()) {
+    const TrackInfo* track_info =
+        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      CreateAudioSender(local_stream, track.get(), track_info->ssrc);
+    }
+  }
+  for (const auto& track : local_stream->GetVideoTracks()) {
+    const TrackInfo* track_info =
+        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      CreateVideoSender(local_stream, track.get(), track_info->ssrc);
+    }
+  }
+
   stats_->AddStream(local_stream);
   observer_->OnRenegotiationNeeded();
   return true;
 }
 
+// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
+// indefinitely.
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
-  mediastream_signaling_->RemoveLocalStream(local_stream);
+  for (const auto& track : local_stream->GetAudioTracks()) {
+    const TrackInfo* track_info =
+        FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      DestroyAudioSender(local_stream, track.get(), track_info->ssrc);
+    }
+  }
+  for (const auto& track : local_stream->GetVideoTracks()) {
+    const TrackInfo* track_info =
+        FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
+    if (track_info) {
+      DestroyVideoSender(local_stream, track.get());
+    }
+  }
+
+  local_streams_->RemoveStream(local_stream);
+
   if (IsClosed()) {
     return;
   }
   observer_->OnRenegotiationNeeded();
 }
 
 rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
     AudioTrackInterface* track) {
   if (!track) {
     LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
     return NULL;
   }
-  if (!mediastream_signaling_->local_streams()->FindAudioTrack(track->id())) {
+  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;
   }
@@ skipping 49 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) {
-  bool first_datachannel = !mediastream_signaling_->HasDataChannels();
+  bool first_datachannel = !HasDataChannels();
 
   rtc::scoped_ptr<InternalDataChannelInit> internal_config;
   if (config) {
     internal_config.reset(new InternalDataChannelInit(*config));
   }
   rtc::scoped_refptr<DataChannelInterface> channel(
-      session_->CreateDataChannel(label, internal_config.get()));
-  if (!channel.get())
-    return NULL;
+      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) {
-  if (!VERIFY(observer != NULL)) {
+  if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
     return;
   }
   RTCOfferAnswerOptions options;
 
   bool value;
   size_t mandatory_constraints = 0;
 
   if (FindConstraint(constraints,
                      MediaConstraintsInterface::kOfferToReceiveAudio,
@@ skipping 30 matching lines @@
                      &value,
                      &mandatory_constraints)) {
     options.use_rtp_mux = value;
   }
 
   CreateOffer(observer, options);
 }
 
 void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
                                  const RTCOfferAnswerOptions& options) {
-  if (!VERIFY(observer != NULL)) {
+  if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
     return;
   }
-  session_->CreateOffer(observer, options);
+
+  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) {
-  if (!VERIFY(observer != NULL)) {
+  if (!VERIFY(observer != nullptr)) {
     LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
     return;
   }
-  session_->CreateAnswer(observer, constraints);
+
+  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) {
-  if (!VERIFY(observer != NULL)) {
+  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.
   stats_->UpdateStats(kStatsOutputLevelStandard);
   std::string error;
   if (!session_->SetLocalDescription(desc, &error)) {
     PostSetSessionDescriptionFailure(observer, error);
     return;
   }
-  SetSessionDescriptionMsg* msg =  new SetSessionDescriptionMsg(observer);
+
+  // If setting the description decided our SSL role, allocate any necessary
+  // SCTP sids.
+  rtc::SSLRole role;
+  if (session_->data_channel_type() == cricket::DCT_SCTP &&
+      session_->GetSslRole(&role)) {
+    AllocateSctpSids(role);
+  }
+
+  // Update state and SSRC of local MediaStreams and DataChannels based on the
+  // local session description.
+  const cricket::ContentInfo* audio_content =
+      GetFirstAudioContent(desc->description());
+  if (audio_content) {
+    const cricket::AudioContentDescription* audio_desc =
+        static_cast<const cricket::AudioContentDescription*>(
+            audio_content->description);
+    UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
+  }
+
+  const cricket::ContentInfo* video_content =
+      GetFirstVideoContent(desc->description());
+  if (video_content) {
+    const cricket::VideoContentDescription* video_desc =
+        static_cast<const cricket::VideoContentDescription*>(
+            video_content->description);
+    UpdateLocalTracks(video_desc->streams(), video_desc->type());
+  }
+
+  const cricket::ContentInfo* data_content =
+      GetFirstDataContent(desc->description());
+  if (data_content) {
+    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);
+
   // 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) {
-  if (!VERIFY(observer != NULL)) {
+  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.
   stats_->UpdateStats(kStatsOutputLevelStandard);
   std::string error;
   if (!session_->SetRemoteDescription(desc, &error)) {
     PostSetSessionDescriptionFailure(observer, error);
     return;
   }
-  SetSessionDescriptionMsg* msg  = new SetSessionDescriptionMsg(observer);
+
+  // If setting the description decided our SSL role, allocate any necessary
+  // SCTP sids.
+  rtc::SSLRole role;
+  if (session_->data_channel_type() == cricket::DCT_SCTP &&
+      session_->GetSslRole(&role)) {
+    AllocateSctpSids(role);
+  }
+
+  const cricket::SessionDescription* remote_desc = desc->description();
+
+  // We wait to signal new streams until we finish processing the description,
+  // since only at that point will new streams have all their tracks.
+  rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
+
+  // Find all audio rtp streams and create corresponding remote AudioTracks
+  // and MediaStreams.
+  const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
+  if (audio_content) {
+    const cricket::AudioContentDescription* desc =
+        static_cast<const cricket::AudioContentDescription*>(
+            audio_content->description);
+    UpdateRemoteStreamsList(desc->streams(), desc->type(), new_streams);
+    remote_info_.default_audio_track_needed =
+        MediaContentDirectionHasSend(desc->direction()) &&
+        desc->streams().empty();
+  }
+
+  // Find all video rtp streams and create corresponding remote VideoTracks
+  // and MediaStreams.
+  const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
+  if (video_content) {
+    const cricket::VideoContentDescription* desc =
+        static_cast<const cricket::VideoContentDescription*>(
+            video_content->description);
+    UpdateRemoteStreamsList(desc->streams(), desc->type(), new_streams);
+    remote_info_.default_video_track_needed =
+        MediaContentDirectionHasSend(desc->direction()) &&
+        desc->streams().empty();
+  }
+
+  // Update the DataChannels with the information from the remote peer.
+  const cricket::ContentInfo* data_content = GetFirstDataContent(remote_desc);
+  if (data_content) {
+    const cricket::DataContentDescription* data_desc =
+        static_cast<const cricket::DataContentDescription*>(
+            data_content->description);
+    if (rtc::starts_with(data_desc->protocol().data(),
+                         cricket::kMediaProtocolRtpPrefix)) {
+      UpdateRemoteRtpDataChannels(data_desc->streams());
+    }
+  }
+
+  // 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);
+  }
+
+  // Find removed MediaStreams.
+  if (remote_info_.IsDefaultMediaStreamNeeded() &&
+      remote_streams_->find(kDefaultStreamLabel) != nullptr) {
+    // The default media stream already exists. No need to do anything.
+  } else {
+    UpdateEndedRemoteMediaStreams();
+    remote_info_.msid_supported |= remote_streams_->count() > 0;
+  }
+  MaybeCreateDefaultStream();
+
+  SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
   signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
 }
 
-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);
-}
-
 bool PeerConnection::SetConfiguration(const RTCConfiguration& config) {
   if (port_allocator_) {
     std::vector<PortAllocatorFactoryInterface::StunConfiguration> stuns;
     std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turns;
     if (!ParseIceServers(config.servers, &stuns, &turns)) {
       return false;
     }
 
     std::vector<rtc::SocketAddress> stun_hosts;
     typedef std::vector<StunConfiguration>::const_iterator StunIt;
@@ skipping 111 matching lines @@
       delete param;
       break;
     }
     case MSG_SET_SESSIONDESCRIPTION_FAILED: {
       SetSessionDescriptionMsg* param =
           static_cast<SetSessionDescriptionMsg*>(msg->pdata);
       param->observer->OnFailure(param->error);
       delete param;
       break;
     }
+    case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
+      CreateSessionDescriptionMsg* param =
+          static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
+      param->observer->OnFailure(param->error);
+      delete param;
+      break;
+    }
     case MSG_GETSTATS: {
       GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
       StatsReports reports;
       stats_->GetStats(param->track, &reports);
       param->observer->OnComplete(reports);
       delete param;
       break;
     }
     default:
       ASSERT(false && "Not implemented");
       break;
   }
 }
 
-void PeerConnection::OnAddRemoteStream(MediaStreamInterface* stream) {
-  stats_->AddStream(stream);
-  observer_->OnAddStream(stream);
-}
-
-void PeerConnection::OnRemoveRemoteStream(MediaStreamInterface* stream) {
-  observer_->OnRemoveStream(stream);
-}
-
-void PeerConnection::OnAddDataChannel(DataChannelInterface* data_channel) {
-  observer_->OnDataChannel(DataChannelProxy::Create(signaling_thread(),
-                                                    data_channel));
-}
-
-void PeerConnection::OnAddRemoteAudioTrack(MediaStreamInterface* stream,
-                                           AudioTrackInterface* audio_track,
-                                           uint32 ssrc) {
+void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
+                                         AudioTrackInterface* audio_track,
+                                         uint32 ssrc) {
   receivers_.push_back(new AudioRtpReceiver(audio_track, ssrc, session_.get()));
 }
 
-void PeerConnection::OnAddRemoteVideoTrack(MediaStreamInterface* stream,
-                                           VideoTrackInterface* video_track,
-                                           uint32 ssrc) {
+void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
+                                         VideoTrackInterface* video_track,
+                                         uint32 ssrc) {
   receivers_.push_back(new VideoRtpReceiver(video_track, ssrc, session_.get()));
 }
 
 // TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
 // description.
-void PeerConnection::OnRemoveRemoteAudioTrack(
-    MediaStreamInterface* stream,
-    AudioTrackInterface* audio_track) {
+void PeerConnection::DestroyAudioReceiver(MediaStreamInterface* stream,
+                                          AudioTrackInterface* audio_track) {
   auto it = FindReceiverForTrack(audio_track);
   if (it == receivers_.end()) {
     LOG(LS_WARNING) << "RtpReceiver for track with id " << audio_track->id()
                     << " doesn't exist.";
   } else {
     (*it)->Stop();
     receivers_.erase(it);
   }
 }
 
-void PeerConnection::OnRemoveRemoteVideoTrack(
-    MediaStreamInterface* stream,
-    VideoTrackInterface* video_track) {
+void PeerConnection::DestroyVideoReceiver(MediaStreamInterface* stream,
+                                          VideoTrackInterface* video_track) {
   auto it = FindReceiverForTrack(video_track);
   if (it == receivers_.end()) {
     LOG(LS_WARNING) << "RtpReceiver for track with id " << video_track->id()
                     << " doesn't exist.";
   } else {
     (*it)->Stop();
     receivers_.erase(it);
   }
 }
 
-void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream,
-                                          AudioTrackInterface* audio_track,
-                                          uint32 ssrc) {
+void PeerConnection::CreateAudioSender(MediaStreamInterface* stream,
+                                       AudioTrackInterface* audio_track,
+                                       uint32 ssrc) {
   senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
   stats_->AddLocalAudioTrack(audio_track, ssrc);
 }
 
-void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream,
-                                          VideoTrackInterface* video_track,
-                                          uint32 ssrc) {
+void PeerConnection::CreateVideoSender(MediaStreamInterface* stream,
+                                       VideoTrackInterface* video_track,
+                                       uint32 ssrc) {
   senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
 }
 
 // TODO(deadbeef): Keep RtpSenders around even if track goes away in local
 // description.
-void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
-                                             AudioTrackInterface* audio_track,
-                                             uint32 ssrc) {
+void PeerConnection::DestroyAudioSender(MediaStreamInterface* stream,
+                                        AudioTrackInterface* audio_track,
+                                        uint32 ssrc) {
   auto it = FindSenderForTrack(audio_track);
   if (it == senders_.end()) {
     LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
                     << " doesn't exist.";
     return;
   } else {
     (*it)->Stop();
     senders_.erase(it);
   }
   stats_->RemoveLocalAudioTrack(audio_track, ssrc);
 }
 
-void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
-                                             VideoTrackInterface* video_track) {
+void PeerConnection::DestroyVideoSender(MediaStreamInterface* stream,
+                                        VideoTrackInterface* video_track) {
   auto it = FindSenderForTrack(video_track);
   if (it == senders_.end()) {
     LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id()
                     << " doesn't exist.";
     return;
   } else {
     (*it)->Stop();
     senders_.erase(it);
   }
 }
 
-void PeerConnection::OnRemoveLocalStream(MediaStreamInterface* stream) {
-}
-
 void PeerConnection::OnIceConnectionChange(
     PeerConnectionInterface::IceConnectionState new_state) {
   ASSERT(signaling_thread()->IsCurrent());
   // After transitioning to "closed", ignore any additional states from
   // WebRtcSession (such as "disconnected").
-  if (ice_connection_state_ == kIceConnectionClosed) {
+  if (IsClosed()) {
     return;
   }
   ice_connection_state_ = new_state;
   observer_->OnIceConnectionChange(ice_connection_state_);
 }
 
 void PeerConnection::OnIceGatheringChange(
     PeerConnectionInterface::IceGatheringState new_state) {
   ASSERT(signaling_thread()->IsCurrent());
   if (IsClosed()) {
@@ skipping 26 matching lines @@
     observer_->OnIceConnectionChange(ice_connection_state_);
     if (ice_gathering_state_ != kIceGatheringComplete) {
       ice_gathering_state_ = kIceGatheringComplete;
       observer_->OnIceGatheringChange(ice_gathering_state_);
     }
   }
   observer_->OnSignalingChange(signaling_state_);
   observer_->OnStateChange(PeerConnectionObserver::kSignalingState);
 }
 
+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);
+}
+
+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);
+}
+
+bool PeerConnection::GetOptionsForOffer(
+    const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
+    cricket::MediaSessionOptions* session_options) {
+  SetStreams(session_options, local_streams_, rtp_data_channels_);
+
+  if (!ConvertRtcOptionsForOffer(rtc_options, session_options)) {
+    return false;
+  }
+
+  if (session_->data_channel_type() == cricket::DCT_SCTP && HasDataChannels()) {
+    session_options->data_channel_type = cricket::DCT_SCTP;
+  }
+  return true;
+}
+
+bool PeerConnection::GetOptionsForAnswer(
+    const MediaConstraintsInterface* constraints,
+    cricket::MediaSessionOptions* session_options) {
+  SetStreams(session_options, local_streams_, rtp_data_channels_);
+  session_options->recv_audio = false;
+  session_options->recv_video = false;
+
+  if (!ParseConstraintsForAnswer(constraints, session_options)) {
+    return false;
+  }
+
+  // RTP data channel is handled in MediaSessionOptions::AddStream. SCTP streams
+  // are not signaled in the SDP so does not go through that path and must be
+  // handled here.
+  if (session_->data_channel_type() == cricket::DCT_SCTP) {
+    session_options->data_channel_type = cricket::DCT_SCTP;
+  }
+  return true;
+}
+
+void PeerConnection::UpdateRemoteStreamsList(
+    const cricket::StreamParamsVec& streams,
+    cricket::MediaType media_type,
+    StreamCollection* new_streams) {
+  TrackInfos* current_tracks = GetRemoteTracks(media_type);
+
+  // Find removed tracks. I.e., tracks where the track id or ssrc don't match
+  // the
+  // new StreamParam.
+  auto track_it = current_tracks->begin();
+  while (track_it != current_tracks->end()) {
+    const TrackInfo& info = *track_it;
+    const cricket::StreamParams* params =
+        cricket::GetStreamBySsrc(streams, info.ssrc);
+    if (!params || params->id != info.track_id) {
+      OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
+      track_it = current_tracks->erase(track_it);
+    } else {
+      ++track_it;
+    }
+  }
+
+  // Find new and active tracks.
+  for (const cricket::StreamParams& params : streams) {
+    // The sync_label is the MediaStream label and the |stream.id| is the
+    // track id.
+    const std::string& stream_label = params.sync_label;
+    const std::string& track_id = params.id;
+    uint32 ssrc = params.first_ssrc();
+
+    rtc::scoped_refptr<MediaStreamInterface> stream =
+        remote_streams_->find(stream_label);
+    if (!stream) {
+      // This is a new MediaStream. Create a new remote MediaStream.
+      stream = remote_stream_factory_->CreateMediaStream(stream_label);
+      remote_streams_->AddStream(stream);
+      new_streams->AddStream(stream);
+    }
+
+    const TrackInfo* track_info =
+        FindTrackInfo(*current_tracks, stream_label, track_id);
+    if (!track_info) {
+      current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
+      OnRemoteTrackSeen(stream_label, track_id, ssrc, media_type);
+    }
+  }
+}
+
+void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
+                                       const std::string& track_id,
+                                       uint32 ssrc,
+                                       cricket::MediaType media_type) {
+  MediaStreamInterface* stream = remote_streams_->find(stream_label);
+
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    AudioTrackInterface* audio_track =
+        remote_stream_factory_->AddAudioTrack(stream, track_id);
+    CreateAudioReceiver(stream, audio_track, ssrc);
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    VideoTrackInterface* video_track =
+        remote_stream_factory_->AddVideoTrack(stream, track_id);
+    CreateVideoReceiver(stream, video_track, ssrc);
+  } else {
+    RTC_DCHECK(false && "Invalid media type");
+  }
+}
+
+void PeerConnection::OnRemoteTrackRemoved(const std::string& stream_label,
+                                          const std::string& track_id,
+                                          cricket::MediaType media_type) {
+  MediaStreamInterface* stream = remote_streams_->find(stream_label);
+
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    rtc::scoped_refptr<AudioTrackInterface> audio_track =
+        stream->FindAudioTrack(track_id);
+    if (audio_track) {
+      audio_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      stream->RemoveTrack(audio_track);
+      DestroyAudioReceiver(stream, audio_track);
+    }
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    rtc::scoped_refptr<VideoTrackInterface> video_track =
+        stream->FindVideoTrack(track_id);
+    if (video_track) {
+      video_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      stream->RemoveTrack(video_track);
+      DestroyVideoReceiver(stream, video_track);
+    }
+  } else {
+    ASSERT(false && "Invalid media type");
+  }
+}
+
+void PeerConnection::UpdateEndedRemoteMediaStreams() {
+  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
+  for (size_t i = 0; i < remote_streams_->count(); ++i) {
+    MediaStreamInterface* stream = remote_streams_->at(i);
+    if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
+      streams_to_remove.push_back(stream);
+    }
+  }
+
+  for (const auto& stream : streams_to_remove) {
+    remote_streams_->RemoveStream(stream);
+    observer_->OnRemoveStream(stream);
+  }
+}
+
+void PeerConnection::MaybeCreateDefaultStream() {
+  if (!remote_info_.IsDefaultMediaStreamNeeded()) {
+    return;
+  }
+
+  bool default_created = false;
+
+  rtc::scoped_refptr<MediaStreamInterface> default_remote_stream =
+      remote_streams_->find(kDefaultStreamLabel);
+  if (default_remote_stream == nullptr) {
+    default_created = true;
+    default_remote_stream =
+        remote_stream_factory_->CreateMediaStream(kDefaultStreamLabel);
+    remote_streams_->AddStream(default_remote_stream);
+  }
+  if (remote_info_.default_audio_track_needed &&
+      default_remote_stream->GetAudioTracks().size() == 0) {
+    remote_audio_tracks_.push_back(
+        TrackInfo(kDefaultStreamLabel, kDefaultAudioTrackLabel, 0));
+    OnRemoteTrackSeen(kDefaultStreamLabel, kDefaultAudioTrackLabel, 0,
+                      cricket::MEDIA_TYPE_AUDIO);
+  }
+  if (remote_info_.default_video_track_needed &&
+      default_remote_stream->GetVideoTracks().size() == 0) {
+    remote_video_tracks_.push_back(
+        TrackInfo(kDefaultStreamLabel, kDefaultVideoTrackLabel, 0));
+    OnRemoteTrackSeen(kDefaultStreamLabel, kDefaultVideoTrackLabel, 0,
+                      cricket::MEDIA_TYPE_VIDEO);
+  }
+  if (default_created) {
+    stats_->AddStream(default_remote_stream);
+    observer_->OnAddStream(default_remote_stream);
+  }
+}
+
+void PeerConnection::RejectRemoteTracks(cricket::MediaType media_type) {

[pthatcher1, 2015/10/08 05:23:54]

Should this be called EndRemoteTracks?  All it does is set the state to ended.

[Taylor Brandstetter, 2015/10/09 19:54:09]

I agree; done. Funny enough, I think we already made a similar change in the
"video disable/reenable" CL that was rolled back.

+  TrackInfos* current_tracks = GetRemoteTracks(media_type);
+  for (TrackInfos::iterator track_it = current_tracks->begin();
+       track_it != current_tracks->end(); ++track_it) {
+    const TrackInfo& info = *track_it;
+    MediaStreamInterface* stream = remote_streams_->find(info.stream_label);
+    if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+      AudioTrackInterface* track = stream->FindAudioTrack(info.track_id);
+      // There's no guarantee the track is still available, e.g. the track may
+      // have been removed from the stream by javascript.
+      if (track) {
+        track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      }
+    }
+    if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+      VideoTrackInterface* track = stream->FindVideoTrack(info.track_id);
+      // There's no guarantee the track is still available, e.g. the track may
+      // have been removed from the stream by javascript.
+      if (track) {
+        track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      }
+    }
+  }
+}
+
+void PeerConnection::UpdateLocalTracks(
+    const std::vector<cricket::StreamParams>& streams,
+    cricket::MediaType media_type) {
+  TrackInfos* current_tracks = GetLocalTracks(media_type);
+
+  // Find removed tracks. I.e., tracks where the track id, stream label or ssrc
+  // don't match the new StreamParam.
+  TrackInfos::iterator track_it = current_tracks->begin();
+  while (track_it != current_tracks->end()) {
+    const TrackInfo& info = *track_it;
+    const cricket::StreamParams* params =
+        cricket::GetStreamBySsrc(streams, info.ssrc);
+    if (!params || params->id != info.track_id ||
+        params->sync_label != info.stream_label) {
+      OnLocalTrackRemoved(info.stream_label, info.track_id, info.ssrc,
+                          media_type);
+      track_it = current_tracks->erase(track_it);
+    } else {
+      ++track_it;
+    }
+  }
+
+  // Find new and active tracks.
+  for (const cricket::StreamParams& params : streams) {
+    // The sync_label is the MediaStream label and the |stream.id| is the
+    // track id.
+    const std::string& stream_label = params.sync_label;
+    const std::string& track_id = params.id;
+    uint32 ssrc = params.first_ssrc();
+    const TrackInfo* track_info =
+        FindTrackInfo(*current_tracks, stream_label, track_id);
+    if (!track_info) {
+      current_tracks->push_back(TrackInfo(stream_label, track_id, ssrc));
+      OnLocalTrackSeen(stream_label, track_id, params.first_ssrc(), media_type);
+    }
+  }
+}
+
+void PeerConnection::OnLocalTrackSeen(const std::string& stream_label,
+                                      const std::string& track_id,
+                                      uint32 ssrc,
+                                      cricket::MediaType media_type) {
+  MediaStreamInterface* stream = local_streams_->find(stream_label);
+  if (!stream) {
+    LOG(LS_WARNING) << "An unknown local MediaStream with label "
+                    << stream_label << " has been configured.";
+    return;
+  }
+
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+    if (!audio_track) {
+      LOG(LS_WARNING) << "An unknown local AudioTrack with id , " << track_id
+                      << " has been configured.";
+      return;
+    }
+    CreateAudioSender(stream, audio_track, ssrc);
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+    if (!video_track) {
+      LOG(LS_WARNING) << "An unknown local VideoTrack with id , " << track_id
+                      << " has been configured.";
+      return;
+    }
+    CreateVideoSender(stream, video_track, ssrc);
+  } else {
+    RTC_DCHECK(false && "Invalid media type");
+  }
+}
+
+void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
+                                         const std::string& track_id,
+                                         uint32 ssrc,
+                                         cricket::MediaType media_type) {
+  MediaStreamInterface* stream = local_streams_->find(stream_label);
+  if (!stream) {
+    // This is the normal case. I.e., RemoveLocalStream has been called and the
+    // SessionDescriptions has been renegotiated.
+    return;
+  }
+  // A track has been removed from the SessionDescription but the MediaStream
+  // is still associated with PeerConnection. This only occurs if the SDP
+  // doesn't match with the calls to AddLocalStream and RemoveLocalStream.
+  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+    AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
+    if (!audio_track) {
+      return;
+    }
+    DestroyAudioSender(stream, audio_track, ssrc);
+  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+    VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
+    if (!video_track) {
+      return;
+    }
+    DestroyVideoSender(stream, video_track);
+  } else {
+    RTC_DCHECK(false && "Invalid media type.");
+  }
+}
+
+void PeerConnection::UpdateLocalRtpDataChannels(
+    const cricket::StreamParamsVec& streams) {
+  std::vector<std::string> existing_channels;
+
+  // Find new and active data channels.
+  for (const cricket::StreamParams& params : streams) {
+    // |it->sync_label| is actually the data channel label. The reason is that
+    // we use the same naming of data channels as we do for
+    // MediaStreams and Tracks.
+    // For MediaStreams, the sync_label is the MediaStream label and the
+    // track label is the same as |streamid|.
+    const std::string& channel_label = params.sync_label;
+    auto data_channel_it = rtp_data_channels_.find(channel_label);
+    if (!VERIFY(data_channel_it != rtp_data_channels_.end())) {
+      continue;
+    }
+    // Set the SSRC the data channel should use for sending.
+    data_channel_it->second->SetSendSsrc(params.first_ssrc());
+    existing_channels.push_back(data_channel_it->first);
+  }
+
+  UpdateClosingRtpDataChannels(existing_channels, true);
+}
+
+void PeerConnection::UpdateRemoteRtpDataChannels(
+    const cricket::StreamParamsVec& streams) {
+  std::vector<std::string> existing_channels;
+
+  // Find new and active data channels.
+  for (const cricket::StreamParams& params : streams) {
+    // The data channel label is either the mslabel or the SSRC if the mslabel
+    // does not exist. Ex a=ssrc:444330170 mslabel:test1.
+    std::string label = params.sync_label.empty()
+                            ? rtc::ToString(params.first_ssrc())
+                            : params.sync_label;
+    auto data_channel_it = rtp_data_channels_.find(label);
+    if (data_channel_it == rtp_data_channels_.end()) {
+      // This is a new data channel.
+      CreateRemoteRtpDataChannel(label, params.first_ssrc());
+    } else {
+      data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
+    }
+    existing_channels.push_back(label);
+  }
+
+  UpdateClosingRtpDataChannels(existing_channels, false);
+}
+
+void PeerConnection::UpdateClosingRtpDataChannels(
+    const std::vector<std::string>& active_channels,
+    bool is_local_update) {
+  auto it = rtp_data_channels_.begin();
+  while (it != rtp_data_channels_.end()) {
+    DataChannel* data_channel = it->second;
+    if (std::find(active_channels.begin(), active_channels.end(),
+                  data_channel->label()) != active_channels.end()) {
+      ++it;
+      continue;
+    }
+
+    if (is_local_update) {
+      data_channel->SetSendSsrc(0);
+    } else {
+      data_channel->RemotePeerRequestClose();
+    }
+
+    if (data_channel->state() == DataChannel::kClosed) {
+      it = rtp_data_channels_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+
+void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
+                                                uint32 remote_ssrc) {
+  rtc::scoped_refptr<DataChannel> channel(
+      InternalCreateDataChannel(label, nullptr));
+  if (!channel.get()) {
+    LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
+                    << "CreateDataChannel failed.";
+    return;
+  }
+  channel->SetReceiveSsrc(remote_ssrc);
+  observer_->OnDataChannel(
+      DataChannelProxy::Create(signaling_thread(), channel));
+}
+
+rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
+    const std::string& label,
+    const InternalDataChannelInit* config) {
+  if (IsClosed()) {
+    return nullptr;
+  }
+  if (session_->data_channel_type() == cricket::DCT_NONE) {
+    LOG(LS_ERROR)
+        << "InternalCreateDataChannel: Data is not supported in this call.";
+    return nullptr;
+  }
+  InternalDataChannelInit new_config =
+      config ? (*config) : InternalDataChannelInit();
+  if (session_->data_channel_type() == cricket::DCT_SCTP) {
+    if (new_config.id < 0) {
+      rtc::SSLRole role;
+      if (session_->GetSslRole(&role) &&
+          !sid_allocator_.AllocateSctpSid(role, &new_config.id)) {
+        LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
+        return nullptr;
+      }
+    } else if (!sid_allocator_.ReserveSctpSid(new_config.id)) {
+      LOG(LS_ERROR) << "Failed to create a SCTP data channel "
+                    << "because the id is already in use or out of range.";
+      return nullptr;
+    }
+  }
+
+  rtc::scoped_refptr<DataChannel> channel(DataChannel::Create(
+      session_.get(), session_->data_channel_type(), label, new_config));
+  if (!channel) {
+    sid_allocator_.ReleaseSctpSid(new_config.id);
+    return nullptr;
+  }
+
+  if (channel->data_channel_type() == cricket::DCT_RTP) {
+    if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
+      LOG(LS_ERROR) << "DataChannel with label " << channel->label()
+                    << " already exists.";
+      return nullptr;
+    }
+    rtp_data_channels_[channel->label()] = channel;
+  } else {
+    RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
+    sctp_data_channels_.push_back(channel);
+    channel->SignalClosed.connect(this,
+                                  &PeerConnection::OnSctpDataChannelClosed);
+  }
+
+  return channel;
+}
+
+bool PeerConnection::HasDataChannels() const {
+  return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
+}
+
+void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
+  for (const auto& channel : sctp_data_channels_) {
+    if (channel->id() < 0) {
+      int sid;
+      if (!sid_allocator_.AllocateSctpSid(role, &sid)) {
+        LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
+        continue;
+      }
+      channel->SetSctpSid(sid);
+    }
+  }
+}
+
+void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
+  for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
+       ++it) {
+    if (it->get() == channel) {
+      int sid = channel->id();
+      RTC_DCHECK(sid >= 0);
+      sid_allocator_.ReleaseSctpSid(sid);
+      sctp_data_channels_.erase(it);
+      return;
+    }
+  }
+}
+
+void PeerConnection::OnVoiceChannelDestroyed() {
+  RejectRemoteTracks(cricket::MEDIA_TYPE_AUDIO);
+}
+
+void PeerConnection::OnVideoChannelDestroyed() {
+  RejectRemoteTracks(cricket::MEDIA_TYPE_VIDEO);
+}
+
+void PeerConnection::OnDataChannelCreated() {
+  for (const auto& channel : sctp_data_channels_) {
+    channel->OnTransportChannelCreated();
+  }
+}
+
+void PeerConnection::OnDataChannelDestroyed() {
+  // Use a temporary copy of the RTP/SCTP DataChannel list because the
+  // DataChannel may callback to us and try to modify the list.
+  std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
+  temp_rtp_dcs.swap(rtp_data_channels_);
+  for (const auto& kv : temp_rtp_dcs) {
+    kv.second->OnDataEngineClose();

[pthatcher1, 2015/10/08 05:23:54]

Can we rename these so they are consistent with one another? 
OnTransportChannelCreated/OnDataEngineClose is a weird combo.  Looking at the
implementations of those methods, Activate/Close might be a good name pair.

[Taylor Brandstetter, 2015/10/09 19:54:09]

Well, since the cricket::DataChannel actually IS being created (constructed) and
destroyed (destructed), I think Create/Destroy makes as much sense. And since
OnDataEngineClose is the only odd man out, I changed it to
OnTransportChannelDestroyed.

[pthatcher1, 2015/10/09 21:12:26]

Sounds good.

+  }
+
+  std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
+  temp_sctp_dcs.swap(sctp_data_channels_);
+  for (const auto& channel : temp_sctp_dcs) {
+    channel->OnDataEngineClose();
+  }
+}
+
+void PeerConnection::OnDataChannelCreationRequested(
+    const std::string& label,
+    const InternalDataChannelInit& config) {
+  rtc::scoped_refptr<DataChannel> channel(
+      InternalCreateDataChannel(label, &config));
+  if (!channel.get()) {
+    LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
+    return;
+  }
+
+  observer_->OnDataChannel(
+      DataChannelProxy::Create(signaling_thread(), channel));
+}
+
 std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
 PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
   return std::find_if(
       senders_.begin(), senders_.end(),
       [track](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
         return sender->track() == track;
       });
 }
 
 std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
 PeerConnection::FindReceiverForTrack(MediaStreamTrackInterface* track) {
   return std::find_if(
       receivers_.begin(), receivers_.end(),
       [track](const rtc::scoped_refptr<RtpReceiverInterface>& receiver) {
         return receiver->track() == track;
       });
 }
 
+PeerConnection::TrackInfos* PeerConnection::GetRemoteTracks(
+    cricket::MediaType media_type) {
+  RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
+             media_type == cricket::MEDIA_TYPE_VIDEO);
+  return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &remote_audio_tracks_
+                                                   : &remote_video_tracks_;
+}
+
+PeerConnection::TrackInfos* PeerConnection::GetLocalTracks(
+    cricket::MediaType media_type) {
+  RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
+             media_type == cricket::MEDIA_TYPE_VIDEO);
+  return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_tracks_
+                                                   : &local_video_tracks_;
+}
+
+const PeerConnection::TrackInfo* PeerConnection::FindTrackInfo(
+    const PeerConnection::TrackInfos& infos,
+    const std::string& stream_label,
+    const std::string track_id) const {
+  for (const TrackInfo& track_info : infos) {
+    if (track_info.stream_label == stream_label &&
+        track_info.track_id == track_id) {
+      return &track_info;
+    }
+  }
+  return nullptr;
+}
+
+DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
+  for (const auto& channel : sctp_data_channels_) {
+    if (channel->id() == sid) {
+      return channel;
+    }
+  }
+  return nullptr;
+}
+
 }  // namespace webrtc