Move talk/app/webrtc to webrtc/api

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,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * 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 <algorithm>
-#include <cctype>  // for isdigit
-#include <utility>
-#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/mediastreamobserver.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/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/media/sctp/sctpdataengine.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;
-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";
-
-// NOTE: Must be in the same order as the ServiceType enum.
-static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
-
-// NOTE: A loop below assumes that the first value of this enum is 0 and all
-// other values are incremental.
-enum ServiceType {
-  STUN = 0,  // 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.
-};
-static_assert(INVALID == arraysize(kValidIceServiceTypes),
-              "kValidIceServiceTypes must have as many strings as ServiceType "
-              "has values.");
-
-enum {
-  MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
-  MSG_SET_SESSIONDESCRIPTION_FAILED,
-  MSG_CREATE_SESSIONDESCRIPTION_FAILED,
-  MSG_GETSTATS,
-  MSG_FREE_DATACHANNELS,
-};
-
-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
-// stunURI       = scheme ":" stun-host [ ":" stun-port ]
-// scheme        = "stun" / "stuns"
-// stun-host     = IP-literal / IPv4address / reg-name
-// stun-port     = *DIGIT
-//
-// draft-petithuguenin-behave-turn-uris-01
-// turnURI       = scheme ":" turn-host [ ":" turn-port ]
-// turn-host     = username@IP-literal / IPv4address / reg-name
-bool GetServiceTypeAndHostnameFromUri(const std::string& in_str,
-                                      ServiceType* service_type,
-                                      std::string* hostname) {
-  const std::string::size_type colonpos = in_str.find(':');
-  if (colonpos == std::string::npos) {
-    LOG(LS_WARNING) << "Missing ':' in ICE URI: " << in_str;
-    return false;
-  }
-  if ((colonpos + 1) == in_str.length()) {
-    LOG(LS_WARNING) << "Empty hostname in ICE URI: " << in_str;
-    return false;
-  }
-  *service_type = INVALID;
-  for (size_t i = 0; i < arraysize(kValidIceServiceTypes); ++i) {
-    if (in_str.compare(0, colonpos, kValidIceServiceTypes[i]) == 0) {
-      *service_type = static_cast<ServiceType>(i);
-      break;
-    }
-  }
-  if (*service_type == INVALID) {
-    return false;
-  }
-  *hostname = in_str.substr(colonpos + 1, std::string::npos);
-  return true;
-}
-
-bool ParsePort(const std::string& in_str, int* port) {
-  // Make sure port only contains digits. FromString doesn't check this.
-  for (const char& c : in_str) {
-    if (!std::isdigit(c)) {
-      return false;
-    }
-  }
-  return rtc::FromString(in_str, port);
-}
-
-// This method parses IPv6 and IPv4 literal strings, along with hostnames in
-// standard hostname:port format.
-// Consider following formats as correct.
-// |hostname:port|, |[IPV6 address]:port|, |IPv4 address|:port,
-// |hostname|, |[IPv6 address]|, |IPv4 address|.
-bool ParseHostnameAndPortFromString(const std::string& in_str,
-                                    std::string* host,
-                                    int* port) {
-  RTC_DCHECK(host->empty());
-  if (in_str.at(0) == '[') {
-    std::string::size_type closebracket = in_str.rfind(']');
-    if (closebracket != std::string::npos) {
-      std::string::size_type colonpos = in_str.find(':', closebracket);
-      if (std::string::npos != colonpos) {
-        if (!ParsePort(in_str.substr(closebracket + 2, std::string::npos),
-                       port)) {
-          return false;
-        }
-      }
-      *host = in_str.substr(1, closebracket - 1);
-    } else {
-      return false;
-    }
-  } else {
-    std::string::size_type colonpos = in_str.find(':');
-    if (std::string::npos != colonpos) {
-      if (!ParsePort(in_str.substr(colonpos + 1, std::string::npos), port)) {
-        return false;
-      }
-      *host = in_str.substr(0, colonpos);
-    } else {
-      *host = in_str;
-    }
-  }
-  return !host->empty();
-}
-
-// Adds a STUN or TURN server to the appropriate list,
-// by parsing |url| and using the username/password in |server|.
-bool ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
-                       const std::string& url,
-                       cricket::ServerAddresses* stun_servers,
-                       std::vector<cricket::RelayServerConfig>* turn_servers) {
-  // draft-nandakumar-rtcweb-stun-uri-01
-  // stunURI       = scheme ":" stun-host [ ":" stun-port ]
-  // scheme        = "stun" / "stuns"
-  // stun-host     = IP-literal / IPv4address / reg-name
-  // stun-port     = *DIGIT
-
-  // draft-petithuguenin-behave-turn-uris-01
-  // turnURI       = scheme ":" turn-host [ ":" turn-port ]
-  //                 [ "?transport=" transport ]
-  // scheme        = "turn" / "turns"
-  // transport     = "udp" / "tcp" / transport-ext
-  // transport-ext = 1*unreserved
-  // turn-host     = IP-literal / IPv4address / reg-name
-  // turn-port     = *DIGIT
-  RTC_DCHECK(stun_servers != nullptr);
-  RTC_DCHECK(turn_servers != nullptr);
-  std::vector<std::string> tokens;
-  cricket::ProtocolType turn_transport_type = cricket::PROTO_UDP;
-  RTC_DCHECK(!url.empty());
-  rtc::tokenize(url, '?', &tokens);
-  std::string uri_without_transport = tokens[0];
-  // Let's look into transport= param, if it exists.
-  if (tokens.size() == kTurnTransportTokensNum) {  // ?transport= is present.
-    std::string uri_transport_param = tokens[1];
-    rtc::tokenize(uri_transport_param, '=', &tokens);
-    if (tokens[0] == kTransport) {
-      // As per above grammar transport param will be consist of lower case
-      // letters.
-      if (!cricket::StringToProto(tokens[1].c_str(), &turn_transport_type) ||
-          (turn_transport_type != cricket::PROTO_UDP &&
-           turn_transport_type != cricket::PROTO_TCP)) {
-        LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
-        return false;
-      }
-    }
-  }
-
-  std::string hoststring;
-  ServiceType service_type;
-  if (!GetServiceTypeAndHostnameFromUri(uri_without_transport,
-                                       &service_type,
-                                       &hoststring)) {
-    LOG(LS_WARNING) << "Invalid transport parameter in ICE URI: " << url;
-    return false;
-  }
-
-  // GetServiceTypeAndHostnameFromUri should never give an empty hoststring
-  RTC_DCHECK(!hoststring.empty());
-
-  // Let's break hostname.
-  tokens.clear();
-  rtc::tokenize_with_empty_tokens(hoststring, '@', &tokens);
-
-  std::string username(server.username);
-  if (tokens.size() > kTurnHostTokensNum) {
-    LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
-    return false;
-  }
-  if (tokens.size() == kTurnHostTokensNum) {
-    if (tokens[0].empty() || tokens[1].empty()) {
-      LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
-      return false;
-    }
-    username.assign(rtc::s_url_decode(tokens[0]));
-    hoststring = tokens[1];
-  } else {
-    hoststring = tokens[0];
-  }
-
-  int port = kDefaultStunPort;
-  if (service_type == TURNS) {
-    port = kDefaultStunTlsPort;
-    turn_transport_type = cricket::PROTO_TCP;
-  }
-
-  std::string address;
-  if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
-    LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
-    return false;
-  }
-
-  if (port <= 0 || port > 0xffff) {
-    LOG(WARNING) << "Invalid port: " << port;
-    return false;
-  }
-
-  switch (service_type) {
-    case STUN:
-    case STUNS:
-      stun_servers->insert(rtc::SocketAddress(address, port));
-      break;
-    case TURN:
-    case TURNS: {
-      bool secure = (service_type == TURNS);
-      turn_servers->push_back(
-          cricket::RelayServerConfig(address, port, username, server.password,
-                                     turn_transport_type, secure));
-      break;
-    }
-    case INVALID:
-    default:
-      LOG(WARNING) << "Configuration not supported: " << url;
-      return false;
-  }
-  return true;
-}
-
-// Check if we can send |new_stream| on a PeerConnection.
-bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
-                            webrtc::MediaStreamInterface* new_stream) {
-  if (!new_stream || !current_streams) {
-    return false;
-  }
-  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;
-}
-
-// If the direction is "recvonly" or "inactive", treat the description
-// as containing no streams.
-// See: https://code.google.com/p/webrtc/issues/detail?id=5054
-std::vector<cricket::StreamParams> GetActiveStreams(
-    const cricket::MediaContentDescription* desc) {
-  return MediaContentDirectionHasSend(desc->direction())
-             ? desc->streams()
-             : std::vector<cricket::StreamParams>();
-}
-
-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 AddSendStreams(
-    cricket::MediaSessionOptions* session_options,
-    const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
-    const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
-        rtp_data_channels) {
-  session_options->streams.clear();
-  for (const auto& sender : senders) {
-    session_options->AddSendStream(sender->media_type(), sender->id(),
-                                   sender->stream_id());
-  }
-
-  // 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(uint32_t ssrc,
-                                     AudioProviderInterface* provider,
-                                     webrtc::MediaStreamInterface* stream,
-                                     const std::string& track_id) {
-    return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(
-        stream, track_id, RemoteAudioSource::Create(ssrc, provider));
-  }
-
-  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, true)
-            .get());
-  }
-
- private:
-  template <typename TI, typename T, typename TP, typename S>
-  TI* AddTrack(MediaStreamInterface* stream,
-               const std::string& track_id,
-               const 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;
-  }
-
-  if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
-    session_options->recv_audio = (rtc_options.offer_to_receive_audio > 0);
-  }
-  if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
-    session_options->recv_video = (rtc_options.offer_to_receive_video > 0);
-  }
-
-  session_options->vad_enabled = rtc_options.voice_activity_detection;
-  session_options->audio_transport_options.ice_restart =
-      rtc_options.ice_restart;
-  session_options->video_transport_options.ice_restart =
-      rtc_options.ice_restart;
-  session_options->data_transport_options.ice_restart = rtc_options.ice_restart;
-  session_options->bundle_enabled = rtc_options.use_rtp_mux;
-
-  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;
-  }
-
-  if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
-                     &value, &mandatory_constraints_satisfied)) {
-    session_options->audio_transport_options.ice_restart = value;
-    session_options->video_transport_options.ice_restart = value;
-    session_options->data_transport_options.ice_restart = value;
-  } else {
-    // kIceRestart defaults to false according to spec.
-    session_options->audio_transport_options.ice_restart = false;
-    session_options->video_transport_options.ice_restart = false;
-    session_options->data_transport_options.ice_restart = false;
-  }
-
-  if (!constraints) {
-    return true;
-  }
-  return mandatory_constraints_satisfied == constraints->GetMandatory().size();
-}
-
-bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
-                     cricket::ServerAddresses* stun_servers,
-                     std::vector<cricket::RelayServerConfig>* turn_servers) {
-  for (const webrtc::PeerConnectionInterface::IceServer& server : servers) {
-    if (!server.urls.empty()) {
-      for (const std::string& url : server.urls) {
-        if (url.empty()) {
-          LOG(LS_ERROR) << "Empty uri.";
-          return false;
-        }
-        if (!ParseIceServerUrl(server, url, stun_servers, turn_servers)) {
-          return false;
-        }
-      }
-    } else if (!server.uri.empty()) {
-      // Fallback to old .uri if new .urls isn't present.
-      if (!ParseIceServerUrl(server, server.uri, stun_servers, turn_servers)) {
-        return false;
-      }
-    } else {
-      LOG(LS_ERROR) << "Empty uri.";
-      return false;
-    }
-  }
-  // Candidates must have unique priorities, so that connectivity checks
-  // are performed in a well-defined order.
-  int priority = static_cast<int>(turn_servers->size() - 1);
-  for (cricket::RelayServerConfig& turn_server : *turn_servers) {
-    // First in the list gets highest priority.
-    turn_server.priority = priority--;
-  }
-  return true;
-}
-
-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),
-      local_streams_(StreamCollection::Create()),
-      remote_streams_(StreamCollection::Create()) {}
-
-PeerConnection::~PeerConnection() {
-  TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  // 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,
-    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_ = std::move(allocator);
-
-  cricket::ServerAddresses stun_servers;
-  std::vector<cricket::RelayServerConfig> turn_servers;
-  if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
-    return false;
-  }
-  port_allocator_->SetIceServers(stun_servers, turn_servers);
-
-  // 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, 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);
-
-  media_controller_.reset(factory_->CreateMediaController());
-
-  remote_stream_factory_.reset(new RemoteMediaStreamFactory(
-      factory_->signaling_thread(), media_controller_->channel_manager()));
-
-  session_.reset(
-      new WebRtcSession(media_controller_.get(), factory_->signaling_thread(),
-                        factory_->worker_thread(), port_allocator_.get()));
-  stats_.reset(new StatsCollector(this));
-
-  // Initialize the WebRtcSession. It creates transport channels etc.
-  if (!session_->Initialize(factory_->options(), constraints,
-                            std::move(dtls_identity_store), 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_->SignalDataChannelOpenMessage.connect(
-      this, &PeerConnection::OnDataChannelOpenMessage);
-  return true;
-}
-
-rtc::scoped_refptr<StreamCollectionInterface>
-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);
-  MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
-  observer->SignalAudioTrackAdded.connect(this,
-                                          &PeerConnection::OnAudioTrackAdded);
-  observer->SignalAudioTrackRemoved.connect(
-      this, &PeerConnection::OnAudioTrackRemoved);
-  observer->SignalVideoTrackAdded.connect(this,
-                                          &PeerConnection::OnVideoTrackAdded);
-  observer->SignalVideoTrackRemoved.connect(
-      this, &PeerConnection::OnVideoTrackRemoved);
-  stream_observers_.push_back(rtc::scoped_ptr<MediaStreamObserver>(observer));
-
-  for (const auto& track : local_stream->GetAudioTracks()) {
-    OnAudioTrackAdded(track.get(), local_stream);
-  }
-  for (const auto& track : local_stream->GetVideoTracks()) {
-    OnVideoTrackAdded(track.get(), local_stream);
-  }
-
-  stats_->AddStream(local_stream);
-  observer_->OnRenegotiationNeeded();
-  return true;
-}
-
-void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
-  TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
-  for (const auto& track : local_stream->GetAudioTracks()) {
-    OnAudioTrackRemoved(track.get(), local_stream);
-  }
-  for (const auto& track : local_stream->GetVideoTracks()) {
-    OnVideoTrackRemoved(track.get(), local_stream);
-  }
-
-  local_streams_->RemoveStream(local_stream);
-  stream_observers_.erase(
-      std::remove_if(
-          stream_observers_.begin(), stream_observers_.end(),
-          [local_stream](const rtc::scoped_ptr<MediaStreamObserver>& observer) {
-            return observer->stream()->label().compare(local_stream->label()) ==
-                   0;
-          }),
-      stream_observers_.end());
-
-  if (IsClosed()) {
-    return;
-  }
-  observer_->OnRenegotiationNeeded();
-}
-
-rtc::scoped_refptr<RtpSenderInterface> PeerConnection::AddTrack(
-    MediaStreamTrackInterface* track,
-    std::vector<MediaStreamInterface*> streams) {
-  TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
-  if (IsClosed()) {
-    return nullptr;
-  }
-  if (streams.size() >= 2) {
-    LOG(LS_ERROR)
-        << "Adding a track with two streams is not currently supported.";
-    return nullptr;
-  }
-  // TODO(deadbeef): Support adding a track to two different senders.
-  if (FindSenderForTrack(track) != senders_.end()) {
-    LOG(LS_ERROR) << "Sender for track " << track->id() << " already exists.";
-    return nullptr;
-  }
-
-  // TODO(deadbeef): Support adding a track to multiple streams.
-  rtc::scoped_refptr<RtpSenderInterface> new_sender;
-  if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
-    new_sender = RtpSenderProxy::Create(
-        signaling_thread(),
-        new AudioRtpSender(static_cast<AudioTrackInterface*>(track),
-                           session_.get(), stats_.get()));
-    if (!streams.empty()) {
-      new_sender->set_stream_id(streams[0]->label());
-    }
-    const TrackInfo* track_info = FindTrackInfo(
-        local_audio_tracks_, new_sender->stream_id(), track->id());
-    if (track_info) {
-      new_sender->SetSsrc(track_info->ssrc);
-    }
-  } else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
-    new_sender = RtpSenderProxy::Create(
-        signaling_thread(),
-        new VideoRtpSender(static_cast<VideoTrackInterface*>(track),
-                           session_.get()));
-    if (!streams.empty()) {
-      new_sender->set_stream_id(streams[0]->label());
-    }
-    const TrackInfo* track_info = FindTrackInfo(
-        local_video_tracks_, new_sender->stream_id(), track->id());
-    if (track_info) {
-      new_sender->SetSsrc(track_info->ssrc);
-    }
-  } else {
-    LOG(LS_ERROR) << "CreateSender called with invalid kind: " << track->kind();
-    return rtc::scoped_refptr<RtpSenderInterface>();
-  }
-
-  senders_.push_back(new_sender);
-  observer_->OnRenegotiationNeeded();
-  return new_sender;
-}
-
-bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
-  TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
-  if (IsClosed()) {
-    return false;
-  }
-
-  auto it = std::find(senders_.begin(), senders_.end(), sender);
-  if (it == senders_.end()) {
-    LOG(LS_ERROR) << "Couldn't find sender " << sender->id() << " to remove.";
-    return false;
-  }
-  (*it)->Stop();
-  senders_.erase(it);
-
-  observer_->OnRenegotiationNeeded();
-  return true;
-}
-
-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,
-    const std::string& stream_id) {
-  TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
-  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;
-  }
-  if (!stream_id.empty()) {
-    new_sender->set_stream_id(stream_id);
-  }
-  senders_.push_back(new_sender);
-  return new_sender;
-}
-
-std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
-    const {
-  return senders_;
-}
-
-std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
-PeerConnection::GetReceivers() const {
-  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;
-}
-
-PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
-  return signaling_state_;
-}
-
-PeerConnectionInterface::IceState PeerConnection::ice_state() {
-  return ice_state_;
-}
-
-PeerConnectionInterface::IceConnectionState
-PeerConnection::ice_connection_state() {
-  return ice_connection_state_;
-}
-
-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,
-                     MediaConstraintsInterface::kOfferToReceiveAudio,
-                     &value,
-                     &mandatory_constraints)) {
-    options.offer_to_receive_audio =
-        value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
-  }
-
-  if (FindConstraint(constraints,
-                     MediaConstraintsInterface::kOfferToReceiveVideo,
-                     &value,
-                     &mandatory_constraints)) {
-    options.offer_to_receive_video =
-        value ? RTCOfferAnswerOptions::kOfferToReceiveMediaTrue : 0;
-  }
-
-  if (FindConstraint(constraints,
-                     MediaConstraintsInterface::kVoiceActivityDetection,
-                     &value,
-                     &mandatory_constraints)) {
-    options.voice_activity_detection = value;
-  }
-
-  if (FindConstraint(constraints,
-                     MediaConstraintsInterface::kIceRestart,
-                     &value,
-                     &mandatory_constraints)) {
-    options.ice_restart = value;
-  }
-
-  if (FindConstraint(constraints,
-                     MediaConstraintsInterface::kUseRtpMux,
-                     &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.
-  stats_->UpdateStats(kStatsOutputLevelStandard);
-  std::string error;
-  if (!session_->SetLocalDescription(desc, &error)) {
-    PostSetSessionDescriptionFailure(observer, error);
-    return;
-  }
-
-  // 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(session_->data_channel(), &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) {
-    if (audio_content->rejected) {
-      RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
-    } else {
-      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) {
-    if (video_content->rejected) {
-      RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
-    } else {
-      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) {
-  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.
-  stats_->UpdateStats(kStatsOutputLevelStandard);
-  std::string error;
-  if (!session_->SetRemoteDescription(desc, &error)) {
-    PostSetSessionDescriptionFailure(observer, error);
-    return;
-  }
-
-  // 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(session_->data_channel(), &role)) {
-    AllocateSctpSids(role);
-  }
-
-  const cricket::SessionDescription* remote_desc = desc->description();
-  const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
-  const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
-  const cricket::AudioContentDescription* audio_desc =
-      GetFirstAudioContentDescription(remote_desc);
-  const cricket::VideoContentDescription* video_desc =
-      GetFirstVideoContentDescription(remote_desc);
-  const cricket::DataContentDescription* data_desc =
-      GetFirstDataContentDescription(remote_desc);
-
-  // Check if the descriptions include streams, just in case the peer supports
-  // MSID, but doesn't indicate so with "a=msid-semantic".
-  if (remote_desc->msid_supported() ||
-      (audio_desc && !audio_desc->streams().empty()) ||
-      (video_desc && !video_desc->streams().empty())) {
-    remote_peer_supports_msid_ = true;
-  }
-
-  // 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.
-  if (audio_content) {
-    if (audio_content->rejected) {
-      RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
-    } else {
-      bool default_audio_track_needed =
-          !remote_peer_supports_msid_ &&
-          MediaContentDirectionHasSend(audio_desc->direction());
-      UpdateRemoteStreamsList(GetActiveStreams(audio_desc),
-                              default_audio_track_needed, audio_desc->type(),
-                              new_streams);
-    }
-  }
-
-  // Find all video rtp streams and create corresponding remote VideoTracks
-  // and MediaStreams.
-  if (video_content) {
-    if (video_content->rejected) {
-      RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
-    } else {
-      bool default_video_track_needed =
-          !remote_peer_supports_msid_ &&
-          MediaContentDirectionHasSend(video_desc->direction());
-      UpdateRemoteStreamsList(GetActiveStreams(video_desc),
-                              default_video_track_needed, video_desc->type(),
-                              new_streams);
-    }
-  }
-
-  // Update the DataChannels with the information from the remote peer.
-  if (data_desc) {
-    if (rtc::starts_with(data_desc->protocol().data(),
-                         cricket::kMediaProtocolRtpPrefix)) {
-      UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
-    }
-  }
-
-  // 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);
-}
-
-bool PeerConnection::SetConfiguration(const RTCConfiguration& config) {
-  TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
-  if (port_allocator_) {
-    cricket::ServerAddresses stun_servers;
-    std::vector<cricket::RelayServerConfig> turn_servers;
-    if (!ParseIceServers(config.servers, &stun_servers, &turn_servers)) {
-      return false;
-    }
-    port_allocator_->SetIceServers(stun_servers, turn_servers);
-  }
-  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) {
-    case WebRtcSession::STATE_INIT:
-      ChangeSignalingState(PeerConnectionInterface::kStable);
-      break;
-    case WebRtcSession::STATE_SENTOFFER:
-      ChangeSignalingState(PeerConnectionInterface::kHaveLocalOffer);
-      break;
-    case WebRtcSession::STATE_SENTPRANSWER:
-      ChangeSignalingState(PeerConnectionInterface::kHaveLocalPrAnswer);
-      break;
-    case WebRtcSession::STATE_RECEIVEDOFFER:
-      ChangeSignalingState(PeerConnectionInterface::kHaveRemoteOffer);
-      break;
-    case WebRtcSession::STATE_RECEIVEDPRANSWER:
-      ChangeSignalingState(PeerConnectionInterface::kHaveRemotePrAnswer);
-      break;
-    case WebRtcSession::STATE_INPROGRESS:
-      ChangeSignalingState(PeerConnectionInterface::kStable);
-      break;
-    case WebRtcSession::STATE_CLOSED:
-      ChangeSignalingState(PeerConnectionInterface::kClosed);
-      break;
-    default:
-      break;
-  }
-}
-
-void PeerConnection::OnMessage(rtc::Message* msg) {
-  switch (msg->message_id) {
-    case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
-      SetSessionDescriptionMsg* param =
-          static_cast<SetSessionDescriptionMsg*>(msg->pdata);
-      param->observer->OnSuccess();
-      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;
-    }
-    case MSG_FREE_DATACHANNELS: {
-      sctp_data_channels_to_free_.clear();
-      break;
-    }
-    default:
-      RTC_DCHECK(false && "Not implemented");
-      break;
-  }
-}
-
-void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
-                                         AudioTrackInterface* audio_track,
-                                         uint32_t ssrc) {
-  receivers_.push_back(RtpReceiverProxy::Create(
-      signaling_thread(),
-      new AudioRtpReceiver(audio_track, ssrc, session_.get())));
-}
-
-void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
-                                         VideoTrackInterface* video_track,
-                                         uint32_t ssrc) {
-  receivers_.push_back(RtpReceiverProxy::Create(
-      signaling_thread(),
-      new VideoRtpReceiver(video_track, ssrc, session_.get())));
-}
-
-// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
-// description.
-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::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::OnIceConnectionChange(
-    PeerConnectionInterface::IceConnectionState new_state) {
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  // After transitioning to "closed", ignore any additional states from
-  // WebRtcSession (such as "disconnected").
-  if (IsClosed()) {
-    return;
-  }
-  ice_connection_state_ = new_state;
-  observer_->OnIceConnectionChange(ice_connection_state_);
-}
-
-void PeerConnection::OnIceGatheringChange(
-    PeerConnectionInterface::IceGatheringState new_state) {
-  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());
-  observer_->OnIceCandidate(candidate);
-}
-
-void PeerConnection::OnIceConnectionReceivingChange(bool receiving) {
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  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) {
-  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 AudioRtpSender(track, stream->label(), session_.get(), stats_.get()));
-  senders_.push_back(new_sender);
-  // If the sender has already been configured in SDP, we call SetSsrc,
-  // which will connect the sender to the underlying transport. This can
-  // occur if a local session description that contains the ID of the sender
-  // is set before AddStream is called. It can also occur if the local
-  // session description is not changed and RemoveStream is called, and
-  // later AddStream is called again with the same stream.
-  const TrackInfo* track_info =
-      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) {
-  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) {
-  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) {
-  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::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) {
-  if (!ConvertRtcOptionsForOffer(rtc_options, session_options)) {
-    return false;
-  }
-
-  AddSendStreams(session_options, senders_, rtp_data_channels_);
-  // Offer to receive audio/video if the constraint is not set and there are
-  // send streams, or we're currently receiving.
-  if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
-    session_options->recv_audio =
-        session_options->HasSendMediaStream(cricket::MEDIA_TYPE_AUDIO) ||
-        !remote_audio_tracks_.empty();
-  }
-  if (rtc_options.offer_to_receive_video == RTCOfferAnswerOptions::kUndefined) {
-    session_options->recv_video =
-        session_options->HasSendMediaStream(cricket::MEDIA_TYPE_VIDEO) ||
-        !remote_video_tracks_.empty();
-  }
-  session_options->bundle_enabled =
-      session_options->bundle_enabled &&
-      (session_options->has_audio() || session_options->has_video() ||
-       session_options->has_data());
-
-  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) {
-  session_options->recv_audio = false;
-  session_options->recv_video = false;
-  if (!ParseConstraintsForAnswer(constraints, session_options)) {
-    return false;
-  }
-
-  AddSendStreams(session_options, senders_, rtp_data_channels_);
-  session_options->bundle_enabled =
-      session_options->bundle_enabled &&
-      (session_options->has_audio() || session_options->has_video() ||
-       session_options->has_data());
-
-  // 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::RemoveTracks(cricket::MediaType media_type) {
-  UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
-  UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
-                          media_type, nullptr);
-}
-
-void PeerConnection::UpdateRemoteStreamsList(
-    const cricket::StreamParamsVec& streams,
-    bool default_track_needed,
-    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);
-    bool track_exists = params && params->id == info.track_id;
-    // If this is a default track, and we still need it, don't remove it.
-    if ((info.stream_label == kDefaultStreamLabel && default_track_needed) ||
-        track_exists) {
-      ++track_it;
-    } else {
-      OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
-      track_it = current_tracks->erase(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_t 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);
-    }
-  }
-
-  // Add default track if necessary.
-  if (default_track_needed) {
-    rtc::scoped_refptr<MediaStreamInterface> default_stream =
-        remote_streams_->find(kDefaultStreamLabel);
-    if (!default_stream) {
-      // Create the new default MediaStream.
-      default_stream =
-          remote_stream_factory_->CreateMediaStream(kDefaultStreamLabel);
-      remote_streams_->AddStream(default_stream);
-      new_streams->AddStream(default_stream);
-    }
-    std::string default_track_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
-                                       ? kDefaultAudioTrackLabel
-                                       : kDefaultVideoTrackLabel;
-    const TrackInfo* default_track_info =
-        FindTrackInfo(*current_tracks, kDefaultStreamLabel, default_track_id);
-    if (!default_track_info) {
-      current_tracks->push_back(
-          TrackInfo(kDefaultStreamLabel, default_track_id, 0));
-      OnRemoteTrackSeen(kDefaultStreamLabel, default_track_id, 0, media_type);
-    }
-  }
-}
-
-void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
-                                       const std::string& track_id,
-                                       uint32_t 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(
-        ssrc, session_.get(), 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::EndRemoteTracks(cricket::MediaType media_type) {
-  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_t 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_t ssrc,
-                                      cricket::MediaType media_type) {
-  RtpSenderInterface* sender = FindSenderById(track_id);
-  if (!sender) {
-    LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
-                    << " has been configured in the local description.";
-    return;
-  }
-
-  if (sender->media_type() != media_type) {
-    LOG(LS_WARNING) << "An RtpSender has been configured in the local"
-                    << " description with an unexpected media type.";
-    return;
-  }
-
-  sender->set_stream_id(stream_label);
-  sender->SetSsrc(ssrc);
-}
-
-void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
-                                         const std::string& track_id,
-                                         uint32_t ssrc,
-                                         cricket::MediaType media_type) {
-  RtpSenderInterface* sender = FindSenderById(track_id);
-  if (!sender) {
-    // This is the normal case. I.e., RemoveStream has been called and the
-    // SessionDescriptions has been renegotiated.
-    return;
-  }
-
-  // A sender has been removed from the SessionDescription but it's still
-  // associated with the PeerConnection. This only occurs if the SDP doesn't
-  // match with the calls to CreateSender, AddStream and RemoveStream.
-  if (sender->media_type() != media_type) {
-    LOG(LS_WARNING) << "An RtpSender has been configured in the local"
-                    << " description with an unexpected media type.";
-    return;
-  }
-
-  sender->SetSsrc(0);
-}
-
-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) {
-      rtp_data_channels_.erase(it);
-      it = rtp_data_channels_.begin();
-    } else {
-      ++it;
-    }
-  }
-}
-
-void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
-                                                uint32_t 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(session_->data_channel(), &role)) &&
-          !sid_allocator_.AllocateSid(role, &new_config.id)) {
-        LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
-        return nullptr;
-      }
-    } else if (!sid_allocator_.ReserveSid(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_.ReleaseSid(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_.AllocateSid(role, &sid)) {
-        LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
-        continue;
-      }
-      channel->SetSctpSid(sid);
-    }
-  }
-}
-
-void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  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);
-      return;
-    }
-  }
-}
-
-void PeerConnection::OnVoiceChannelDestroyed() {
-  EndRemoteTracks(cricket::MEDIA_TYPE_AUDIO);
-}
-
-void PeerConnection::OnVideoChannelDestroyed() {
-  EndRemoteTracks(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->OnTransportChannelDestroyed();
-  }
-
-  std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
-  temp_sctp_dcs.swap(sctp_data_channels_);
-  for (const auto& channel : temp_sctp_dcs) {
-    channel->OnTransportChannelDestroyed();
-  }
-}
-
-void PeerConnection::OnDataChannelOpenMessage(
-    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));
-}
-
-RtpSenderInterface* PeerConnection::FindSenderById(const std::string& id) {
-  auto it =
-      std::find_if(senders_.begin(), senders_.end(),
-                   [id](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
-                     return sender->id() == id;
-                   });
-  return it != senders_.end() ? it->get() : nullptr;
-}
-
-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