Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/peerconnection.cc

-/*
- *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/api/peerconnection.h"
-
-#include <algorithm>
-#include <cctype>  // for isdigit
-#include <utility>
-#include <vector>
-
-#include "webrtc/api/audiotrack.h"
-#include "webrtc/api/dtmfsender.h"
-#include "webrtc/api/jsepicecandidate.h"
-#include "webrtc/api/jsepsessiondescription.h"
-#include "webrtc/api/mediaconstraintsinterface.h"
-#include "webrtc/api/mediastream.h"
-#include "webrtc/api/mediastreamobserver.h"
-#include "webrtc/api/mediastreamproxy.h"
-#include "webrtc/api/mediastreamtrackproxy.h"
-#include "webrtc/api/remoteaudiosource.h"
-#include "webrtc/api/rtpreceiver.h"
-#include "webrtc/api/rtpsender.h"
-#include "webrtc/api/streamcollection.h"
-#include "webrtc/api/videocapturertracksource.h"
-#include "webrtc/api/videotrack.h"
-#include "webrtc/base/arraysize.h"
-#include "webrtc/base/bind.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/stringencode.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/base/trace_event.h"
-#include "webrtc/call/call.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/media/sctp/sctptransport.h"
-#include "webrtc/pc/channelmanager.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
-
-namespace {
-
-using webrtc::DataChannel;
-using webrtc::MediaConstraintsInterface;
-using webrtc::MediaStreamInterface;
-using webrtc::PeerConnectionInterface;
-using webrtc::RTCError;
-using webrtc::RTCErrorType;
-using webrtc::RtpSenderInternal;
-using webrtc::RtpSenderInterface;
-using webrtc::RtpSenderProxy;
-using webrtc::RtpSenderProxyWithInternal;
-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"};
-
-// The length of RTCP CNAMEs.
-static const int kRtcpCnameLength = 16;
-
-// 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|.
-RTCErrorType 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_with_empty_tokens(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_with_empty_tokens(uri_transport_param, '=', &tokens);
-    if (tokens[0] != kTransport) {
-      LOG(LS_WARNING) << "Invalid transport parameter key.";
-      return RTCErrorType::SYNTAX_ERROR;
-    }
-    if (tokens.size() < 2 ||
-        !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 RTCErrorType::SYNTAX_ERROR;
-    }
-  }
-
-  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 RTCErrorType::SYNTAX_ERROR;
-  }
-
-  // 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 RTCErrorType::SYNTAX_ERROR;
-  }
-  if (tokens.size() == kTurnHostTokensNum) {
-    if (tokens[0].empty() || tokens[1].empty()) {
-      LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
-      return RTCErrorType::SYNTAX_ERROR;
-    }
-    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_TLS;
-  }
-
-  std::string address;
-  if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
-    LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
-    return RTCErrorType::SYNTAX_ERROR;
-  }
-
-  if (port <= 0 || port > 0xffff) {
-    LOG(WARNING) << "Invalid port: " << port;
-    return RTCErrorType::SYNTAX_ERROR;
-  }
-
-  switch (service_type) {
-    case STUN:
-    case STUNS:
-      stun_servers->insert(rtc::SocketAddress(address, port));
-      break;
-    case TURN:
-    case TURNS: {
-      if (username.empty() || server.password.empty()) {
-        // The WebRTC spec requires throwing an InvalidAccessError when username
-        // or credential are ommitted; this is the native equivalent.
-        return RTCErrorType::INVALID_PARAMETER;
-      }
-      cricket::RelayServerConfig config = cricket::RelayServerConfig(
-          address, port, username, server.password, turn_transport_type);
-      if (server.tls_cert_policy ==
-          PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck) {
-        config.tls_cert_policy =
-            cricket::TlsCertPolicy::TLS_CERT_POLICY_INSECURE_NO_CHECK;
-      }
-      turn_servers->push_back(config);
-      break;
-    }
-    default:
-      // We shouldn't get to this point with an invalid service_type, we should
-      // have returned an error already.
-      RTC_NOTREACHED() << "Unexpected service type";
-      return RTCErrorType::INTERNAL_ERROR;
-  }
-  return RTCErrorType::NONE;
-}
-
-// 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<
-        RtpSenderProxyWithInternal<RtpSenderInternal>>>& 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->internal()->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);
-    }
-  }
-}
-
-uint32_t ConvertIceTransportTypeToCandidateFilter(
-    PeerConnectionInterface::IceTransportsType type) {
-  switch (type) {
-    case PeerConnectionInterface::kNone:
-      return cricket::CF_NONE;
-    case PeerConnectionInterface::kRelay:
-      return cricket::CF_RELAY;
-    case PeerConnectionInterface::kNoHost:
-      return (cricket::CF_ALL & ~cricket::CF_HOST);
-    case PeerConnectionInterface::kAll:
-      return cricket::CF_ALL;
-    default:
-      RTC_NOTREACHED();
-  }
-  return cricket::CF_NONE;
-}
-
-// Helper method to set a voice/video channel on all applicable senders
-// and receivers when one is created/destroyed by WebRtcSession.
-//
-// Used by On(Voice|Video)Channel(Created|Destroyed)
-template <class SENDER,
-          class RECEIVER,
-          class CHANNEL,
-          class SENDERS,
-          class RECEIVERS>
-void SetChannelOnSendersAndReceivers(CHANNEL* channel,
-                                     SENDERS& senders,
-                                     RECEIVERS& receivers,
-                                     cricket::MediaType media_type) {
-  for (auto& sender : senders) {
-    if (sender->media_type() == media_type) {
-      static_cast<SENDER*>(sender->internal())->SetChannel(channel);
-    }
-  }
-  for (auto& receiver : receivers) {
-    if (receiver->media_type() == media_type) {
-      if (!channel) {
-        receiver->internal()->Stop();
-      }
-      static_cast<RECEIVER*>(receiver->internal())->SetChannel(channel);
-    }
-  }
-}
-
-// Helper to set an error and return from a method.
-bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
-  if (error) {
-    error->set_type(type);
-  }
-  return type == webrtc::RTCErrorType::NONE;
-}
-
-}  // namespace
-
-namespace webrtc {
-
-static const char* const kRTCErrorTypeNames[] = {
-    "NONE",
-    "UNSUPPORTED_PARAMETER",
-    "INVALID_PARAMETER",
-    "INVALID_RANGE",
-    "SYNTAX_ERROR",
-    "INVALID_STATE",
-    "INVALID_MODIFICATION",
-    "NETWORK_ERROR",
-    "INTERNAL_ERROR",
-};
-static_assert(static_cast<int>(RTCErrorType::INTERNAL_ERROR) ==
-                  (arraysize(kRTCErrorTypeNames) - 1),
-              "kRTCErrorTypeNames must have as many strings as RTCErrorType "
-              "has values.");
-
-std::ostream& operator<<(std::ostream& stream, RTCErrorType error) {
-  int index = static_cast<int>(error);
-  return stream << kRTCErrorTypeNames[index];
-}
-
-bool PeerConnectionInterface::RTCConfiguration::operator==(
-    const PeerConnectionInterface::RTCConfiguration& o) const {
-  // This static_assert prevents us from accidentally breaking operator==.
-  struct stuff_being_tested_for_equality {
-    IceTransportsType type;
-    IceServers servers;
-    BundlePolicy bundle_policy;
-    RtcpMuxPolicy rtcp_mux_policy;
-    TcpCandidatePolicy tcp_candidate_policy;
-    CandidateNetworkPolicy candidate_network_policy;
-    int audio_jitter_buffer_max_packets;
-    bool audio_jitter_buffer_fast_accelerate;
-    int ice_connection_receiving_timeout;
-    int ice_backup_candidate_pair_ping_interval;
-    ContinualGatheringPolicy continual_gathering_policy;
-    std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
-    bool prioritize_most_likely_ice_candidate_pairs;
-    struct cricket::MediaConfig media_config;
-    bool disable_ipv6;
-    bool enable_rtp_data_channel;
-    bool enable_quic;
-    rtc::Optional<int> screencast_min_bitrate;
-    rtc::Optional<bool> combined_audio_video_bwe;
-    rtc::Optional<bool> enable_dtls_srtp;
-    int ice_candidate_pool_size;
-    bool prune_turn_ports;
-    bool presume_writable_when_fully_relayed;
-    bool enable_ice_renomination;
-    bool redetermine_role_on_ice_restart;
-  };
-  static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
-                "Did you add something to RTCConfiguration and forget to "
-                "update operator==?");
-  return type == o.type && servers == o.servers &&
-         bundle_policy == o.bundle_policy &&
-         rtcp_mux_policy == o.rtcp_mux_policy &&
-         tcp_candidate_policy == o.tcp_candidate_policy &&
-         candidate_network_policy == o.candidate_network_policy &&
-         audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
-         audio_jitter_buffer_fast_accelerate ==
-             o.audio_jitter_buffer_fast_accelerate &&
-         ice_connection_receiving_timeout ==
-             o.ice_connection_receiving_timeout &&
-         ice_backup_candidate_pair_ping_interval ==
-             o.ice_backup_candidate_pair_ping_interval &&
-         continual_gathering_policy == o.continual_gathering_policy &&
-         certificates == o.certificates &&
-         prioritize_most_likely_ice_candidate_pairs ==
-             o.prioritize_most_likely_ice_candidate_pairs &&
-         media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
-         enable_rtp_data_channel == o.enable_rtp_data_channel &&
-         enable_quic == o.enable_quic &&
-         screencast_min_bitrate == o.screencast_min_bitrate &&
-         combined_audio_video_bwe == o.combined_audio_video_bwe &&
-         enable_dtls_srtp == o.enable_dtls_srtp &&
-         ice_candidate_pool_size == o.ice_candidate_pool_size &&
-         prune_turn_ports == o.prune_turn_ports &&
-         presume_writable_when_fully_relayed ==
-             o.presume_writable_when_fully_relayed &&
-         enable_ice_renomination == o.enable_ice_renomination &&
-         redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart;
-}
-
-bool PeerConnectionInterface::RTCConfiguration::operator!=(
-    const PeerConnectionInterface::RTCConfiguration& o) const {
-  return !(*this == o);
-}
-
-// Generate a RTCP CNAME when a PeerConnection is created.
-std::string GenerateRtcpCname() {
-  std::string cname;
-  if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
-    LOG(LS_ERROR) << "Failed to generate CNAME.";
-    RTC_NOTREACHED();
-  }
-  return cname;
-}
-
-bool ExtractMediaSessionOptions(
-    const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
-    bool is_offer,
-    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 constraints don't prevent us, we always accept video.
-  if (rtc_options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
-    session_options->recv_audio = (rtc_options.offer_to_receive_audio > 0);
-  } else {
-    session_options->recv_audio = true;
-  }
-  // For offers, we only offer video if we have it or it's forced by options.
-  // For answers, we will always accept video (if offered).
-  if (rtc_options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
-    session_options->recv_video = (rtc_options.offer_to_receive_video > 0);
-  } else if (is_offer) {
-    session_options->recv_video = false;
-  } else {
-    session_options->recv_video = true;
-  }
-
-  session_options->vad_enabled = rtc_options.voice_activity_detection;
-  session_options->bundle_enabled = rtc_options.use_rtp_mux;
-  for (auto& kv : session_options->transport_options) {
-    kv.second.ice_restart = rtc_options.ice_restart;
-  }
-
-  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;
-  }
-
-  bool ice_restart = false;
-  if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
-                     &value, &mandatory_constraints_satisfied)) {
-    // kIceRestart defaults to false according to spec.
-    ice_restart = true;
-  }
-  for (auto& kv : session_options->transport_options) {
-    kv.second.ice_restart = ice_restart;
-  }
-
-  if (!constraints) {
-    return true;
-  }
-  return mandatory_constraints_satisfied == constraints->GetMandatory().size();
-}
-
-RTCErrorType 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 RTCErrorType::SYNTAX_ERROR;
-        }
-        RTCErrorType err =
-            ParseIceServerUrl(server, url, stun_servers, turn_servers);
-        if (err != RTCErrorType::NONE) {
-          return err;
-        }
-      }
-    } else if (!server.uri.empty()) {
-      // Fallback to old .uri if new .urls isn't present.
-      RTCErrorType err =
-          ParseIceServerUrl(server, server.uri, stun_servers, turn_servers);
-      if (err != RTCErrorType::NONE) {
-        return err;
-      }
-    } else {
-      LOG(LS_ERROR) << "Empty uri.";
-      return RTCErrorType::SYNTAX_ERROR;
-    }
-  }
-  // 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 RTCErrorType::NONE;
-}
-
-PeerConnection::PeerConnection(PeerConnectionFactory* factory)
-    : factory_(factory),
-      observer_(NULL),
-      uma_observer_(NULL),
-      signaling_state_(kStable),
-      ice_connection_state_(kIceConnectionNew),
-      ice_gathering_state_(kIceGatheringNew),
-      event_log_(RtcEventLog::Create()),
-      rtcp_cname_(GenerateRtcpCname()),
-      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->internal()->Stop();
-  }
-  for (const auto& receiver : receivers_) {
-    receiver->internal()->Stop();
-  }
-  // Destroy stats_ because it depends on session_.
-  stats_.reset(nullptr);
-  if (stats_collector_) {
-    stats_collector_->WaitForPendingRequest();
-    stats_collector_ = nullptr;
-  }
-  // Now destroy session_ before destroying other members,
-  // because its destruction fires signals (such as VoiceChannelDestroyed)
-  // which will trigger some final actions in PeerConnection...
-  session_.reset(nullptr);
-  // port_allocator_ lives on the network thread and should be destroyed there.
-  network_thread()->Invoke<void>(RTC_FROM_HERE,
-                                 [this] { port_allocator_.reset(nullptr); });
-}
-
-bool PeerConnection::Initialize(
-    const PeerConnectionInterface::RTCConfiguration& configuration,
-    std::unique_ptr<cricket::PortAllocator> allocator,
-    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
-    PeerConnectionObserver* observer) {
-  TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
-  if (!allocator) {
-    LOG(LS_ERROR) << "PeerConnection initialized without a PortAllocator? "
-                  << "This shouldn't happen if using PeerConnectionFactory.";
-    return false;
-  }
-  if (!observer) {
-    // TODO(deadbeef): Why do we do this?
-    LOG(LS_ERROR) << "PeerConnection initialized without a "
-                  << "PeerConnectionObserver";
-    return false;
-  }
-  observer_ = observer;
-  port_allocator_ = std::move(allocator);
-
-  // The port allocator lives on the network thread and should be initialized
-  // there.
-  if (!network_thread()->Invoke<bool>(
-          RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
-                                   this, configuration))) {
-    return false;
-  }
-
-  media_controller_.reset(factory_->CreateMediaController(
-      configuration.media_config, event_log_.get()));
-
-  session_.reset(new WebRtcSession(
-      media_controller_.get(), factory_->network_thread(),
-      factory_->worker_thread(), factory_->signaling_thread(),
-      port_allocator_.get(),
-      std::unique_ptr<cricket::TransportController>(
-          factory_->CreateTransportController(
-              port_allocator_.get(),
-              configuration.redetermine_role_on_ice_restart)),
-#ifdef HAVE_SCTP
-      std::unique_ptr<cricket::SctpTransportInternalFactory>(
-          new cricket::SctpTransportFactory(factory_->network_thread()))
-#else
-      nullptr
-#endif
-          ));
-
-  stats_.reset(new StatsCollector(this));
-  stats_collector_ = RTCStatsCollector::Create(this);
-
-  // Initialize the WebRtcSession. It creates transport channels etc.
-  if (!session_->Initialize(factory_->options(), std::move(cert_generator),
-                            configuration)) {
-    return false;
-  }
-
-  // Register PeerConnection as receiver of local ice candidates.
-  // All the callbacks will be posted to the application from PeerConnection.
-  session_->RegisterIceObserver(this);
-  session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);
-  session_->SignalVoiceChannelCreated.connect(
-      this, &PeerConnection::OnVoiceChannelCreated);
-  session_->SignalVoiceChannelDestroyed.connect(
-      this, &PeerConnection::OnVoiceChannelDestroyed);
-  session_->SignalVideoChannelCreated.connect(
-      this, &PeerConnection::OnVideoChannelCreated);
-  session_->SignalVideoChannelDestroyed.connect(
-      this, &PeerConnection::OnVideoChannelDestroyed);
-  session_->SignalDataChannelCreated.connect(
-      this, &PeerConnection::OnDataChannelCreated);
-  session_->SignalDataChannelDestroyed.connect(
-      this, &PeerConnection::OnDataChannelDestroyed);
-  session_->SignalDataChannelOpenMessage.connect(
-      this, &PeerConnection::OnDataChannelOpenMessage);
-
-  configuration_ = configuration;
-  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(std::unique_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 std::unique_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<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
-  if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
-    new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-        signaling_thread(),
-        new AudioRtpSender(static_cast<AudioTrackInterface*>(track),
-                           session_->voice_channel(), stats_.get()));
-    if (!streams.empty()) {
-      new_sender->internal()->set_stream_id(streams[0]->label());
-    }
-    const TrackInfo* track_info = FindTrackInfo(
-        local_audio_tracks_, new_sender->internal()->stream_id(), track->id());
-    if (track_info) {
-      new_sender->internal()->SetSsrc(track_info->ssrc);
-    }
-  } else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
-    new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-        signaling_thread(),
-        new VideoRtpSender(static_cast<VideoTrackInterface*>(track),
-                           session_->video_channel()));
-    if (!streams.empty()) {
-      new_sender->internal()->set_stream_id(streams[0]->label());
-    }
-    const TrackInfo* track_info = FindTrackInfo(
-        local_video_tracks_, new_sender->internal()->stream_id(), track->id());
-    if (track_info) {
-      new_sender->internal()->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)->internal()->Stop();
-  senders_.erase(it);
-
-  observer_->OnRenegotiationNeeded();
-  return true;
-}
-
-rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
-    AudioTrackInterface* track) {
-  TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
-  if (IsClosed()) {
-    return nullptr;
-  }
-  if (!track) {
-    LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
-    return NULL;
-  }
-  if (!local_streams_->FindAudioTrack(track->id())) {
-    LOG(LS_ERROR) << "CreateDtmfSender is called with a non local audio track.";
-    return NULL;
-  }
-
-  rtc::scoped_refptr<DtmfSenderInterface> sender(
-      DtmfSender::Create(track, signaling_thread(), session_.get()));
-  if (!sender.get()) {
-    LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
-    return NULL;
-  }
-  return DtmfSenderProxy::Create(signaling_thread(), sender.get());
-}
-
-rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
-    const std::string& kind,
-    const std::string& stream_id) {
-  TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
-  if (IsClosed()) {
-    return nullptr;
-  }
-  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
-  if (kind == MediaStreamTrackInterface::kAudioKind) {
-    new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-        signaling_thread(),
-        new AudioRtpSender(session_->voice_channel(), stats_.get()));
-  } else if (kind == MediaStreamTrackInterface::kVideoKind) {
-    new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-        signaling_thread(), new VideoRtpSender(session_->video_channel()));
-  } else {
-    LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
-    return new_sender;
-  }
-  if (!stream_id.empty()) {
-    new_sender->internal()->set_stream_id(stream_id);
-  }
-  senders_.push_back(new_sender);
-  return new_sender;
-}
-
-std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
-    const {
-  std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
-  for (const auto& sender : senders_) {
-    ret.push_back(sender.get());
-  }
-  return ret;
-}
-
-std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
-PeerConnection::GetReceivers() const {
-  std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
-  for (const auto& receiver : receivers_) {
-    ret.push_back(receiver.get());
-  }
-  return ret;
-}
-
-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);
-  // The StatsCollector is used to tell if a track is valid because it may
-  // remember tracks that the PeerConnection previously removed.
-  if (track && !stats_->IsValidTrack(track->id())) {
-    LOG(LS_WARNING) << "GetStats is called with an invalid track: "
-                    << track->id();
-    return false;
-  }
-  signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
-                           new GetStatsMsg(observer, track));
-  return true;
-}
-
-void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
-  RTC_DCHECK(stats_collector_);
-  stats_collector_->GetStatsReport(callback);
-}
-
-PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
-  return signaling_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");
-#ifdef HAVE_QUIC
-  if (session_->data_channel_type() == cricket::DCT_QUIC) {
-    // TODO(zhihuang): Handle case when config is NULL.
-    if (!config) {
-      LOG(LS_ERROR) << "Missing config for QUIC data channel.";
-      return nullptr;
-    }
-    // TODO(zhihuang): Allow unreliable or ordered QUIC data channels.
-    if (!config->reliable || config->ordered) {
-      LOG(LS_ERROR) << "QUIC data channel does not implement unreliable or "
-                       "ordered delivery.";
-      return nullptr;
-    }
-    return session_->quic_data_transport()->CreateDataChannel(label, config);
-  }
-#endif  // HAVE_QUIC
-
-  bool first_datachannel = !HasDataChannels();
-
-  std::unique_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, session_options);
-}
-
-void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
-                                  const RTCOfferAnswerOptions& options) {
-  TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
-  if (!VERIFY(observer != nullptr)) {
-    LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
-    return;
-  }
-
-  cricket::MediaSessionOptions session_options;
-  if (!GetOptionsForAnswer(options, &session_options)) {
-    std::string error = "CreateAnswer called with invalid options.";
-    LOG(LS_ERROR) << error;
-    PostCreateSessionDescriptionFailure(observer, error);
-    return;
-  }
-
-  session_->CreateAnswer(observer, session_options);
-}
-
-void PeerConnection::SetLocalDescription(
-    SetSessionDescriptionObserver* observer,
-    SessionDescriptionInterface* desc) {
-  TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
-  if (IsClosed()) {
-    return;
-  }
-  if (!VERIFY(observer != nullptr)) {
-    LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
-    return;
-  }
-  if (!desc) {
-    PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
-    return;
-  }
-  // Update stats here so that we have the most recent stats for tracks and
-  // streams that might be removed by updating the session description.
-  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_->GetSctpSslRole(&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(RTC_FROM_HERE, this,
-                           MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
-
-  // MaybeStartGathering needs to be called after posting
-  // MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
-  // before signaling that SetLocalDescription completed.
-  session_->MaybeStartGathering();
-}
-
-void PeerConnection::SetRemoteDescription(
-    SetSessionDescriptionObserver* observer,
-    SessionDescriptionInterface* desc) {
-  TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
-  if (IsClosed()) {
-    return;
-  }
-  if (!VERIFY(observer != nullptr)) {
-    LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
-    return;
-  }
-  if (!desc) {
-    PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
-    return;
-  }
-  // Update stats here so that we have the most recent stats for tracks and
-  // streams that might be removed by updating the session description.
-  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_->GetSctpSslRole(&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);
-    // Call both the raw pointer and scoped_refptr versions of the method
-    // for compatibility.
-    observer_->OnAddStream(new_stream);
-    observer_->OnAddStream(
-        rtc::scoped_refptr<MediaStreamInterface>(new_stream));
-  }
-
-  UpdateEndedRemoteMediaStreams();
-
-  SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
-  signaling_thread()->Post(RTC_FROM_HERE, this,
-                           MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
-}
-
-PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
-  return configuration_;
-}
-
-bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
-                                      RTCError* error) {
-  TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
-
-  if (session_->local_description() &&
-      configuration.ice_candidate_pool_size !=
-          configuration_.ice_candidate_pool_size) {
-    LOG(LS_ERROR) << "Can't change candidate pool size after calling "
-                     "SetLocalDescription.";
-    return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
-  }
-
-  // The simplest (and most future-compatible) way to tell if the config was
-  // modified in an invalid way is to copy each property we do support
-  // modifying, then use operator==. There are far more properties we don't
-  // support modifying than those we do, and more could be added.
-  RTCConfiguration modified_config = configuration_;
-  modified_config.servers = configuration.servers;
-  modified_config.type = configuration.type;
-  modified_config.ice_candidate_pool_size =
-      configuration.ice_candidate_pool_size;
-  modified_config.prune_turn_ports = configuration.prune_turn_ports;
-  if (configuration != modified_config) {
-    LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
-    return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
-  }
-
-  // Note that this isn't possible through chromium, since it's an unsigned
-  // short in WebIDL.
-  if (configuration.ice_candidate_pool_size < 0 ||
-      configuration.ice_candidate_pool_size > UINT16_MAX) {
-    return SafeSetError(RTCErrorType::INVALID_RANGE, error);
-  }
-
-  // Parse ICE servers before hopping to network thread.
-  cricket::ServerAddresses stun_servers;
-  std::vector<cricket::RelayServerConfig> turn_servers;
-  RTCErrorType parse_error =
-      ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
-  if (parse_error != RTCErrorType::NONE) {
-    return SafeSetError(parse_error, error);
-  }
-
-  // In theory this shouldn't fail.
-  if (!network_thread()->Invoke<bool>(
-          RTC_FROM_HERE,
-          rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
-                    stun_servers, turn_servers, modified_config.type,
-                    modified_config.ice_candidate_pool_size,
-                    modified_config.prune_turn_ports))) {
-    LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
-    return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
-  }
-
-  // As described in JSEP, calling setConfiguration with new ICE servers or
-  // candidate policy must set a "needs-ice-restart" bit so that the next offer
-  // triggers an ICE restart which will pick up the changes.
-  if (modified_config.servers != configuration_.servers ||
-      modified_config.type != configuration_.type ||
-      modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
-    session_->SetNeedsIceRestartFlag();
-  }
-  configuration_ = modified_config;
-  return SafeSetError(RTCErrorType::NONE, error);
-}
-
-bool PeerConnection::AddIceCandidate(
-    const IceCandidateInterface* ice_candidate) {
-  TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
-  if (IsClosed()) {
-    return false;
-  }
-  return session_->ProcessIceMessage(ice_candidate);
-}
-
-bool PeerConnection::RemoveIceCandidates(
-    const std::vector<cricket::Candidate>& candidates) {
-  TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
-  return session_->RemoveRemoteIceCandidates(candidates);
-}
-
-void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
-  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_->SetMetricsObserver(uma_observer_);
-    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);
-    }
-  }
-}
-
-bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
-                                      int64_t max_size_bytes) {
-  return factory_->worker_thread()->Invoke<bool>(
-      RTC_FROM_HERE, rtc::Bind(&PeerConnection::StartRtcEventLog_w, this, file,
-                               max_size_bytes));
-}
-
-void PeerConnection::StopRtcEventLog() {
-  factory_->worker_thread()->Invoke<void>(
-      RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
-}
-
-const SessionDescriptionInterface* PeerConnection::local_description() const {
-  return session_->local_description();
-}
-
-const SessionDescriptionInterface* PeerConnection::remote_description() const {
-  return session_->remote_description();
-}
-
-const SessionDescriptionInterface* PeerConnection::current_local_description()
-    const {
-  return session_->current_local_description();
-}
-
-const SessionDescriptionInterface* PeerConnection::current_remote_description()
-    const {
-  return session_->current_remote_description();
-}
-
-const SessionDescriptionInterface* PeerConnection::pending_local_description()
-    const {
-  return session_->pending_local_description();
-}
-
-const SessionDescriptionInterface* PeerConnection::pending_remote_description()
-    const {
-  return session_->pending_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_NOTREACHED() << "Not implemented";
-      break;
-  }
-}
-
-void PeerConnection::CreateAudioReceiver(MediaStreamInterface* stream,
-                                         const std::string& track_id,
-                                         uint32_t ssrc) {
-  rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
-      receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
-          signaling_thread(), new AudioRtpReceiver(stream, track_id, ssrc,
-                                                   session_->voice_channel()));
-
-  receivers_.push_back(receiver);
-  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
-  streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
-  observer_->OnAddTrack(receiver, streams);
-}
-
-void PeerConnection::CreateVideoReceiver(MediaStreamInterface* stream,
-                                         const std::string& track_id,
-                                         uint32_t ssrc) {
-  rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
-      receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
-          signaling_thread(),
-          new VideoRtpReceiver(stream, track_id, factory_->worker_thread(),
-                               ssrc, session_->video_channel()));
-  receivers_.push_back(receiver);
-  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
-  streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
-  observer_->OnAddTrack(receiver, streams);
-}
-
-// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
-// description.
-void PeerConnection::DestroyReceiver(const std::string& track_id) {
-  auto it = FindReceiverForTrack(track_id);
-  if (it == receivers_.end()) {
-    LOG(LS_WARNING) << "RtpReceiver for track with id " << track_id
-                    << " doesn't exist.";
-  } else {
-    (*it)->internal()->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());
-  if (IsClosed()) {
-    return;
-  }
-  observer_->OnIceCandidate(candidate);
-}
-
-void PeerConnection::OnIceCandidatesRemoved(
-    const std::vector<cricket::Candidate>& candidates) {
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  if (IsClosed()) {
-    return;
-  }
-  observer_->OnIceCandidatesRemoved(candidates);
-}
-
-void PeerConnection::OnIceConnectionReceivingChange(bool receiving) {
-  RTC_DCHECK(signaling_thread()->IsCurrent());
-  if (IsClosed()) {
-    return;
-  }
-  observer_->OnIceConnectionReceivingChange(receiving);
-}
-
-void PeerConnection::ChangeSignalingState(
-    PeerConnectionInterface::SignalingState signaling_state) {
-  signaling_state_ = signaling_state;
-  if (signaling_state == kClosed) {
-    ice_connection_state_ = kIceConnectionClosed;
-    observer_->OnIceConnectionChange(ice_connection_state_);
-    if (ice_gathering_state_ != kIceGatheringComplete) {
-      ice_gathering_state_ = kIceGatheringComplete;
-      observer_->OnIceGatheringChange(ice_gathering_state_);
-    }
-  }
-  observer_->OnSignalingChange(signaling_state_);
-}
-
-void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
-                                       MediaStreamInterface* stream) {
-  if (IsClosed()) {
-    return;
-  }
-  auto sender = FindSenderForTrack(track);
-  if (sender != senders_.end()) {
-    // We already have a sender for this track, so just change the stream_id
-    // so that it's correct in the next call to CreateOffer.
-    (*sender)->internal()->set_stream_id(stream->label());
-    return;
-  }
-
-  // Normal case; we've never seen this track before.
-  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
-      RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-          signaling_thread(),
-          new AudioRtpSender(track, stream->label(), session_->voice_channel(),
-                             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->internal()->SetSsrc(track_info->ssrc);
-  }
-}
-
-// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely, when we have unified plan SDP.
-void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
-                                         MediaStreamInterface* stream) {
-  if (IsClosed()) {
-    return;
-  }
-  auto sender = FindSenderForTrack(track);
-  if (sender == senders_.end()) {
-    LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
-                    << " doesn't exist.";
-    return;
-  }
-  (*sender)->internal()->Stop();
-  senders_.erase(sender);
-}
-
-void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
-                                       MediaStreamInterface* stream) {
-  if (IsClosed()) {
-    return;
-  }
-  auto sender = FindSenderForTrack(track);
-  if (sender != senders_.end()) {
-    // We already have a sender for this track, so just change the stream_id
-    // so that it's correct in the next call to CreateOffer.
-    (*sender)->internal()->set_stream_id(stream->label());
-    return;
-  }
-
-  // Normal case; we've never seen this track before.
-  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender =
-      RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
-          signaling_thread(), new VideoRtpSender(track, stream->label(),
-                                                 session_->video_channel()));
-  senders_.push_back(new_sender);
-  const TrackInfo* track_info =
-      FindTrackInfo(local_video_tracks_, stream->label(), track->id());
-  if (track_info) {
-    new_sender->internal()->SetSsrc(track_info->ssrc);
-  }
-}
-
-void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
-                                         MediaStreamInterface* stream) {
-  if (IsClosed()) {
-    return;
-  }
-  auto sender = FindSenderForTrack(track);
-  if (sender == senders_.end()) {
-    LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
-                    << " doesn't exist.";
-    return;
-  }
-  (*sender)->internal()->Stop();
-  senders_.erase(sender);
-}
-
-void PeerConnection::PostSetSessionDescriptionFailure(
-    SetSessionDescriptionObserver* observer,
-    const std::string& error) {
-  SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
-  msg->error = error;
-  signaling_thread()->Post(RTC_FROM_HERE, this,
-                           MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
-}
-
-void PeerConnection::PostCreateSessionDescriptionFailure(
-    CreateSessionDescriptionObserver* observer,
-    const std::string& error) {
-  CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
-  msg->error = error;
-  signaling_thread()->Post(RTC_FROM_HERE, this,
-                           MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
-}
-
-bool PeerConnection::GetOptionsForOffer(
-    const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
-    cricket::MediaSessionOptions* session_options) {
-  // TODO(deadbeef): Once we have transceivers, enumerate them here instead of
-  // ContentInfos.
-  if (session_->local_description()) {
-    for (const cricket::ContentInfo& content :
-         session_->local_description()->description()->contents()) {
-      session_options->transport_options[content.name] =
-          cricket::TransportOptions();
-    }
-  }
-  session_options->enable_ice_renomination =
-      configuration_.enable_ice_renomination;
-
-  if (!ExtractMediaSessionOptions(rtc_options, true, 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();
-  }
-
-  // Intentionally unset the data channel type for RTP data channel with the
-  // second condition. Otherwise the RTP data channels would be successfully
-  // negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
-  // when building with chromium. We want to leave RTP data channels broken, so
-  // people won't try to use them.
-  if (HasDataChannels() && session_->data_channel_type() != cricket::DCT_RTP) {
-    session_options->data_channel_type = session_->data_channel_type();
-  }
-
-  session_options->bundle_enabled =
-      session_options->bundle_enabled &&
-      (session_options->has_audio() || session_options->has_video() ||
-       session_options->has_data());
-
-  session_options->rtcp_cname = rtcp_cname_;
-  session_options->crypto_options = factory_->options().crypto_options;
-  return true;
-}
-
-void PeerConnection::InitializeOptionsForAnswer(
-    cricket::MediaSessionOptions* session_options) {
-  session_options->recv_audio = false;
-  session_options->recv_video = false;
-  session_options->enable_ice_renomination =
-      configuration_.enable_ice_renomination;
-}
-
-void PeerConnection::FinishOptionsForAnswer(
-    cricket::MediaSessionOptions* session_options) {
-  // TODO(deadbeef): Once we have transceivers, enumerate them here instead of
-  // ContentInfos.
-  if (session_->remote_description()) {
-    // Initialize the transport_options map.
-    for (const cricket::ContentInfo& content :
-         session_->remote_description()->description()->contents()) {
-      session_options->transport_options[content.name] =
-          cricket::TransportOptions();
-    }
-  }
-  AddSendStreams(session_options, senders_, rtp_data_channels_);
-  // 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.
-  // Intentionally unset the data channel type for RTP data channel. Otherwise
-  // the RTP data channels would be successfully negotiated by default and the
-  // unit tests in WebRtcDataBrowserTest will fail when building with chromium.
-  // We want to leave RTP data channels broken, so people won't try to use them.
-  if (session_->data_channel_type() != cricket::DCT_RTP) {
-    session_options->data_channel_type = session_->data_channel_type();
-  }
-  session_options->bundle_enabled =
-      session_options->bundle_enabled &&
-      (session_options->has_audio() || session_options->has_video() ||
-       session_options->has_data());
-
-  session_options->crypto_options = factory_->options().crypto_options;
-}
-
-bool PeerConnection::GetOptionsForAnswer(
-    const MediaConstraintsInterface* constraints,
-    cricket::MediaSessionOptions* session_options) {
-  InitializeOptionsForAnswer(session_options);
-  if (!ParseConstraintsForAnswer(constraints, session_options)) {
-    return false;
-  }
-  session_options->rtcp_cname = rtcp_cname_;
-
-  FinishOptionsForAnswer(session_options);
-  return true;
-}
-
-bool PeerConnection::GetOptionsForAnswer(
-    const RTCOfferAnswerOptions& options,
-    cricket::MediaSessionOptions* session_options) {
-  InitializeOptionsForAnswer(session_options);
-  if (!ExtractMediaSessionOptions(options, false, session_options)) {
-    return false;
-  }
-  session_options->rtcp_cname = rtcp_cname_;
-
-  FinishOptionsForAnswer(session_options);
-  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 = MediaStreamProxy::Create(rtc::Thread::Current(),
-                                        MediaStream::Create(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 = MediaStreamProxy::Create(
-          rtc::Thread::Current(), MediaStream::Create(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) {
-    CreateAudioReceiver(stream, track_id, ssrc);
-  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
-    CreateVideoReceiver(stream, track_id, ssrc);
-  } else {
-    RTC_NOTREACHED() << "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) {
-    // When the MediaEngine audio channel is destroyed, the RemoteAudioSource
-    // will be notified which will end the AudioRtpReceiver::track().
-    DestroyReceiver(track_id);
-    rtc::scoped_refptr<AudioTrackInterface> audio_track =
-        stream->FindAudioTrack(track_id);
-    if (audio_track) {
-      stream->RemoveTrack(audio_track);
-    }
-  } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
-    // Stopping or destroying a VideoRtpReceiver will end the
-    // VideoRtpReceiver::track().
-    DestroyReceiver(track_id);
-    rtc::scoped_refptr<VideoTrackInterface> video_track =
-        stream->FindVideoTrack(track_id);
-    if (video_track) {
-      // There's no guarantee the track is still available, e.g. the track may
-      // have been removed from the stream by an application.
-      stream->RemoveTrack(video_track);
-    }
-  } else {
-    RTC_NOTREACHED() << "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 (auto& stream : streams_to_remove) {
-    remote_streams_->RemoveStream(stream);
-    // Call both the raw pointer and scoped_refptr versions of the method
-    // for compatibility.
-    observer_->OnRemoveStream(stream.get());
-    observer_->OnRemoveStream(std::move(stream));
-  }
-}
-
-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) {
-  RtpSenderInternal* 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) {
-  RtpSenderInternal* 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);
-  rtc::scoped_refptr<DataChannelInterface> proxy_channel =
-      DataChannelProxy::Create(signaling_thread(), channel);
-  // Call both the raw pointer and scoped_refptr versions of the method
-  // for compatibility.
-  observer_->OnDataChannel(proxy_channel.get());
-  observer_->OnDataChannel(std::move(proxy_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_->GetSctpSslRole(&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);
-  }
-
-  SignalDataChannelCreated(channel.get());
-  return channel;
-}
-
-bool PeerConnection::HasDataChannels() const {
-#ifdef HAVE_QUIC
-  return !rtp_data_channels_.empty() || !sctp_data_channels_.empty() ||
-         (session_->quic_data_transport() &&
-          session_->quic_data_transport()->HasDataChannels());
-#else
-  return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
-#endif  // HAVE_QUIC
-}
-
-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(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
-                               nullptr);
-      return;
-    }
-  }
-}
-
-void PeerConnection::OnVoiceChannelCreated() {
-  SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver>(
-      session_->voice_channel(), senders_, receivers_,
-      cricket::MEDIA_TYPE_AUDIO);
-}
-
-void PeerConnection::OnVoiceChannelDestroyed() {
-  SetChannelOnSendersAndReceivers<AudioRtpSender, AudioRtpReceiver,
-                                  cricket::VoiceChannel>(
-      nullptr, senders_, receivers_, cricket::MEDIA_TYPE_AUDIO);
-}
-
-void PeerConnection::OnVideoChannelCreated() {
-  SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver>(
-      session_->video_channel(), senders_, receivers_,
-      cricket::MEDIA_TYPE_VIDEO);
-}
-
-void PeerConnection::OnVideoChannelDestroyed() {
-  SetChannelOnSendersAndReceivers<VideoRtpSender, VideoRtpReceiver,
-                                  cricket::VideoChannel>(
-      nullptr, senders_, receivers_, 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;
-  }
-
-  rtc::scoped_refptr<DataChannelInterface> proxy_channel =
-      DataChannelProxy::Create(signaling_thread(), channel);
-  // Call both the raw pointer and scoped_refptr versions of the method
-  // for compatibility.
-  observer_->OnDataChannel(proxy_channel.get());
-  observer_->OnDataChannel(std::move(proxy_channel));
-}
-
-RtpSenderInternal* PeerConnection::FindSenderById(const std::string& id) {
-  auto it = std::find_if(
-      senders_.begin(), senders_.end(),
-      [id](const rtc::scoped_refptr<
-           RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
-        return sender->id() == id;
-      });
-  return it != senders_.end() ? (*it)->internal() : nullptr;
-}
-
-std::vector<
-    rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
-PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
-  return std::find_if(
-      senders_.begin(), senders_.end(),
-      [track](const rtc::scoped_refptr<
-              RtpSenderProxyWithInternal<RtpSenderInternal>>& sender) {
-        return sender->track() == track;
-      });
-}
-
-std::vector<rtc::scoped_refptr<
-    RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
-PeerConnection::FindReceiverForTrack(const std::string& track_id) {
-  return std::find_if(
-      receivers_.begin(), receivers_.end(),
-      [track_id](const rtc::scoped_refptr<
-                 RtpReceiverProxyWithInternal<RtpReceiverInternal>>& receiver) {
-        return receiver->id() == track_id;
-      });
-}
-
-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;
-}
-
-bool PeerConnection::InitializePortAllocator_n(
-    const RTCConfiguration& configuration) {
-  cricket::ServerAddresses stun_servers;
-  std::vector<cricket::RelayServerConfig> turn_servers;
-  if (ParseIceServers(configuration.servers, &stun_servers, &turn_servers) !=
-      RTCErrorType::NONE) {
-    return false;
-  }
-
-  port_allocator_->Initialize();
-
-  // 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;
-  // If the disable-IPv6 flag was specified, we'll not override it
-  // by experiment.
-  if (configuration.disable_ipv6) {
-    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.";
-  }
-
-  if (configuration.candidate_network_policy ==
-      kCandidateNetworkPolicyLowCost) {
-    portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
-    LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
-  }
-
-  port_allocator_->set_flags(portallocator_flags);
-  // No step delay is used while allocating ports.
-  port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
-  port_allocator_->set_candidate_filter(
-      ConvertIceTransportTypeToCandidateFilter(configuration.type));
-
-  // Call this last since it may create pooled allocator sessions using the
-  // properties set above.
-  port_allocator_->SetConfiguration(stun_servers, turn_servers,
-                                    configuration.ice_candidate_pool_size,
-                                    configuration.prune_turn_ports);
-  return true;
-}
-
-bool PeerConnection::ReconfigurePortAllocator_n(
-    const cricket::ServerAddresses& stun_servers,
-    const std::vector<cricket::RelayServerConfig>& turn_servers,
-    IceTransportsType type,
-    int candidate_pool_size,
-    bool prune_turn_ports) {
-  port_allocator_->set_candidate_filter(
-      ConvertIceTransportTypeToCandidateFilter(type));
-  // Call this last since it may create pooled allocator sessions using the
-  // candidate filter set above.
-  return port_allocator_->SetConfiguration(
-      stun_servers, turn_servers, candidate_pool_size, prune_turn_ports);
-}
-
-bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
-                                        int64_t max_size_bytes) {
-  return event_log_->StartLogging(file, max_size_bytes);
-}
-
-void PeerConnection::StopRtcEventLog_w() {
-  event_log_->StopLogging();
-}
-}  // namespace webrtc