pc/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 "pc/peerconnection.h"

#include <algorithm>
#include <limits>
#include <queue>
#include <set>
#include <utility>
#include <vector>

#include "absl/memory/memory.h"
#include "api/jsepicecandidate.h"
#include "api/jsepsessiondescription.h"
#include "api/mediaconstraintsinterface.h"
#include "api/mediastreamproxy.h"
#include "api/mediastreamtrackproxy.h"
#include "call/call.h"
#include "logging/rtc_event_log/icelogger.h"
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/sctp/sctptransport.h"
#include "pc/audiotrack.h"
#include "pc/channel.h"
#include "pc/channelmanager.h"
#include "pc/dtmfsender.h"
#include "pc/mediastream.h"
#include "pc/mediastreamobserver.h"
#include "pc/remoteaudiosource.h"
#include "pc/rtpmediautils.h"
#include "pc/rtpreceiver.h"
#include "pc/rtpsender.h"
#include "pc/sctputils.h"
#include "pc/sdputils.h"
#include "pc/streamcollection.h"
#include "pc/videocapturertracksource.h"
#include "pc/videotrack.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"

using cricket::ContentInfo;
using cricket::ContentInfos;
using cricket::MediaContentDescription;
using cricket::SessionDescription;
using cricket::MediaProtocolType;
using cricket::TransportInfo;

using cricket::LOCAL_PORT_TYPE;
using cricket::STUN_PORT_TYPE;
using cricket::RELAY_PORT_TYPE;
using cricket::PRFLX_PORT_TYPE;

namespace webrtc {

// Error messages
const char kBundleWithoutRtcpMux[] =
    "rtcp-mux must be enabled when BUNDLE "
    "is enabled.";
const char kInvalidCandidates[] = "Description contains invalid candidates.";
const char kInvalidSdp[] = "Invalid session description.";
const char kMlineMismatchInAnswer[] =
    "The order of m-lines in answer doesn't match order in offer. Rejecting "
    "answer.";
const char kMlineMismatchInSubsequentOffer[] =
    "The order of m-lines in subsequent offer doesn't match order from "
    "previous offer/answer.";
const char kSdpWithoutDtlsFingerprint[] =
    "Called with SDP without DTLS fingerprint.";
const char kSdpWithoutSdesCrypto[] = "Called with SDP without SDES crypto.";
const char kSdpWithoutIceUfragPwd[] =
    "Called with SDP without ice-ufrag and ice-pwd.";
const char kSessionError[] = "Session error code: ";
const char kSessionErrorDesc[] = "Session error description: ";
const char kDtlsSrtpSetupFailureRtp[] =
    "Couldn't set up DTLS-SRTP on RTP channel.";
const char kDtlsSrtpSetupFailureRtcp[] =
    "Couldn't set up DTLS-SRTP on RTCP channel.";

namespace {

static const char kDefaultStreamId[] = "default";
static const char kDefaultAudioSenderId[] = "defaulta0";
static const char kDefaultVideoSenderId[] = "defaultv0";

// The length of RTCP CNAMEs.
static const int kRtcpCnameLength = 16;

enum {
  MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
  MSG_SET_SESSIONDESCRIPTION_FAILED,
  MSG_CREATE_SESSIONDESCRIPTION_FAILED,
  MSG_GETSTATS,
  MSG_FREE_DATACHANNELS,
  MSG_REPORT_USAGE_PATTERN,
};

static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;

struct SetSessionDescriptionMsg : public rtc::MessageData {
  explicit SetSessionDescriptionMsg(
      webrtc::SetSessionDescriptionObserver* observer)
      : observer(observer) {}

  rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
  RTCError error;
};

struct CreateSessionDescriptionMsg : public rtc::MessageData {
  explicit CreateSessionDescriptionMsg(
      webrtc::CreateSessionDescriptionObserver* observer)
      : observer(observer) {}

  rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
  RTCError 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;
};

// 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->id()) != nullptr) {
    RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
                      << " is already added.";
    return false;
  }
  return true;
}

// 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 RtpTransceiverDirectionHasSend(desc->direction())
             ? desc->streams()
             : std::vector<cricket::StreamParams>();
}

bool IsValidOfferToReceiveMedia(int value) {
  typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
  return (value >= Options::kUndefined) &&
         (value <= Options::kMaxOfferToReceiveMedia);
}

// Add options to |[audio/video]_media_description_options| from |senders|.
void AddRtpSenderOptions(
    const std::vector<rtc::scoped_refptr<
        RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
    cricket::MediaDescriptionOptions* audio_media_description_options,
    cricket::MediaDescriptionOptions* video_media_description_options) {
  for (const auto& sender : senders) {
    if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
      if (audio_media_description_options) {
        audio_media_description_options->AddAudioSender(
            sender->id(), sender->internal()->stream_ids());
      }
    } else {
      RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO);
      if (video_media_description_options) {
        video_media_description_options->AddVideoSender(
            sender->id(), sender->internal()->stream_ids(), 1);
      }
    }
  }
}

// Add options to |session_options| from |rtp_data_channels|.
void AddRtpDataChannelOptions(
    const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
        rtp_data_channels,
    cricket::MediaDescriptionOptions* data_media_description_options) {
  if (!data_media_description_options) {
    return;
  }
  // 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) {
      // Legacy RTP data channels are signaled with the track/stream ID set to
      // the data channel's label.
      data_media_description_options->AddRtpDataChannel(channel->label(),
                                                        channel->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 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;
}

bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
  bool ok = error.ok();
  if (error_out) {
    *error_out = std::move(error);
  }
  return ok;
}

std::string GetSignalingStateString(
    PeerConnectionInterface::SignalingState state) {
  switch (state) {
    case PeerConnectionInterface::kStable:
      return "kStable";
    case PeerConnectionInterface::kHaveLocalOffer:
      return "kHaveLocalOffer";
    case PeerConnectionInterface::kHaveLocalPrAnswer:
      return "kHavePrAnswer";
    case PeerConnectionInterface::kHaveRemoteOffer:
      return "kHaveRemoteOffer";
    case PeerConnectionInterface::kHaveRemotePrAnswer:
      return "kHaveRemotePrAnswer";
    case PeerConnectionInterface::kClosed:
      return "kClosed";
  }
  RTC_NOTREACHED();
  return "";
}

IceCandidatePairType GetIceCandidatePairCounter(
    const cricket::Candidate& local,
    const cricket::Candidate& remote) {
  const auto& l = local.type();
  const auto& r = remote.type();
  const auto& host = LOCAL_PORT_TYPE;
  const auto& srflx = STUN_PORT_TYPE;
  const auto& relay = RELAY_PORT_TYPE;
  const auto& prflx = PRFLX_PORT_TYPE;
  if (l == host && r == host) {
    bool local_private = IPIsPrivate(local.address().ipaddr());
    bool remote_private = IPIsPrivate(remote.address().ipaddr());
    if (local_private) {
      if (remote_private) {
        return kIceCandidatePairHostPrivateHostPrivate;
      } else {
        return kIceCandidatePairHostPrivateHostPublic;
      }
    } else {
      if (remote_private) {
        return kIceCandidatePairHostPublicHostPrivate;
      } else {
        return kIceCandidatePairHostPublicHostPublic;
      }
    }
  }
  if (l == host && r == srflx)
    return kIceCandidatePairHostSrflx;
  if (l == host && r == relay)
    return kIceCandidatePairHostRelay;
  if (l == host && r == prflx)
    return kIceCandidatePairHostPrflx;
  if (l == srflx && r == host)
    return kIceCandidatePairSrflxHost;
  if (l == srflx && r == srflx)
    return kIceCandidatePairSrflxSrflx;
  if (l == srflx && r == relay)
    return kIceCandidatePairSrflxRelay;
  if (l == srflx && r == prflx)
    return kIceCandidatePairSrflxPrflx;
  if (l == relay && r == host)
    return kIceCandidatePairRelayHost;
  if (l == relay && r == srflx)
    return kIceCandidatePairRelaySrflx;
  if (l == relay && r == relay)
    return kIceCandidatePairRelayRelay;
  if (l == relay && r == prflx)
    return kIceCandidatePairRelayPrflx;
  if (l == prflx && r == host)
    return kIceCandidatePairPrflxHost;
  if (l == prflx && r == srflx)
    return kIceCandidatePairPrflxSrflx;
  if (l == prflx && r == relay)
    return kIceCandidatePairPrflxRelay;
  return kIceCandidatePairMax;
}

// Logic to decide if an m= section can be recycled. This means that the new
// m= section is not rejected, but the old local or remote m= section is
// rejected. |old_content_one| and |old_content_two| refer to the m= section
// of the old remote and old local descriptions in no particular order.
// We need to check both the old local and remote because either
// could be the most current from the latest negotation.
bool IsMediaSectionBeingRecycled(SdpType type,
                                 const ContentInfo& content,
                                 const ContentInfo* old_content_one,
                                 const ContentInfo* old_content_two) {
  return type == SdpType::kOffer && !content.rejected &&
         ((old_content_one && old_content_one->rejected) ||
          (old_content_two && old_content_two->rejected));
}

// Verify that the order of media sections in |new_desc| matches
// |current_desc|. The number of m= sections in |new_desc| should be no
// less than |current_desc|. In the case of checking an answer's
// |new_desc|, the |current_desc| is the last offer that was set as the
// local or remote. In the case of checking an offer's |new_desc| we
// check against the local and remote descriptions stored from the last
// negotiation, because either of these could be the most up to date for
// possible rejected m sections. These are the |current_desc| and
// |secondary_current_desc|.
bool MediaSectionsInSameOrder(const SessionDescription& current_desc,
                              const SessionDescription* secondary_current_desc,
                              const SessionDescription& new_desc,
                              const SdpType type) {
  if (current_desc.contents().size() > new_desc.contents().size()) {
    return false;
  }

  for (size_t i = 0; i < current_desc.contents().size(); ++i) {
    const cricket::ContentInfo* secondary_content_info = nullptr;
    if (secondary_current_desc &&
        i < secondary_current_desc->contents().size()) {
      secondary_content_info = &secondary_current_desc->contents()[i];
    }
    if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i],
                                    &current_desc.contents()[i],
                                    secondary_content_info)) {
      // For new offer descriptions, if the media section can be recycled, it's
      // valid for the MID and media type to change.
      continue;
    }
    if (new_desc.contents()[i].name != current_desc.contents()[i].name) {
      return false;
    }
    const MediaContentDescription* new_desc_mdesc =
        new_desc.contents()[i].media_description();
    const MediaContentDescription* current_desc_mdesc =
        current_desc.contents()[i].media_description();
    if (new_desc_mdesc->type() != current_desc_mdesc->type()) {
      return false;
    }
  }
  return true;
}

bool MediaSectionsHaveSameCount(const SessionDescription& desc1,
                                const SessionDescription& desc2) {
  return desc1.contents().size() == desc2.contents().size();
}

void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
                             cricket::MediaType media_type) {
  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
                            kEnumCounterKeyProtocolMax);
  static const std::map<std::pair<KeyExchangeProtocolType, cricket::MediaType>,
                        KeyExchangeProtocolMedia>
      proto_media_counter_map = {
          {{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_AUDIO},
           kEnumCounterKeyProtocolMediaTypeDtlsAudio},
          {{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_VIDEO},
           kEnumCounterKeyProtocolMediaTypeDtlsVideo},
          {{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_DATA},
           kEnumCounterKeyProtocolMediaTypeDtlsData},
          {{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_AUDIO},
           kEnumCounterKeyProtocolMediaTypeSdesAudio},
          {{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_VIDEO},
           kEnumCounterKeyProtocolMediaTypeSdesVideo},
          {{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_DATA},
           kEnumCounterKeyProtocolMediaTypeSdesData}};

  auto it = proto_media_counter_map.find({protocol_type, media_type});
  if (it != proto_media_counter_map.end()) {
    RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
                              it->second, kEnumCounterKeyProtocolMediaTypeMax);
  }
}

void NoteAddIceCandidateResult(int result) {
  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result,
                            kAddIceCandidateMax);
}

// Checks that each non-rejected content has SDES crypto keys or a DTLS
// fingerprint, unless it's in a BUNDLE group, in which case only the
// BUNDLE-tag section (first media section/description in the BUNDLE group)
// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
// by Channel's |srtp_required| check.
RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
  const cricket::ContentGroup* bundle =
      desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
  for (const cricket::ContentInfo& content_info : desc->contents()) {
    if (content_info.rejected) {
      continue;
    }
    // Note what media is used with each crypto protocol, for all sections.
    NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
                                         : webrtc::kEnumCounterKeyProtocolSdes,
                            content_info.media_description()->type());
    const std::string& mid = content_info.name;
    if (bundle && bundle->HasContentName(mid) &&
        mid != *(bundle->FirstContentName())) {
      // This isn't the first media section in the BUNDLE group, so it's not
      // required to have crypto attributes, since only the crypto attributes
      // from the first section actually get used.
      continue;
    }

    // If the content isn't rejected or bundled into another m= section, crypto
    // must be present.
    const MediaContentDescription* media = content_info.media_description();
    const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
    if (!media || !tinfo) {
      // Something is not right.
      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
    }
    if (dtls_enabled) {
      if (!tinfo->description.identity_fingerprint) {
        RTC_LOG(LS_WARNING)
            << "Session description must have DTLS fingerprint if "
               "DTLS enabled.";
        return RTCError(RTCErrorType::INVALID_PARAMETER,
                        kSdpWithoutDtlsFingerprint);
      }
    } else {
      if (media->cryptos().empty()) {
        RTC_LOG(LS_WARNING)
            << "Session description must have SDES when DTLS disabled.";
        return RTCError(RTCErrorType::INVALID_PARAMETER, kSdpWithoutSdesCrypto);
      }
    }
  }
  return RTCError::OK();
}

// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
// media section/description in the BUNDLE group) needs a ufrag and pwd.
bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
  const cricket::ContentGroup* bundle =
      desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
  for (const cricket::ContentInfo& content_info : desc->contents()) {
    if (content_info.rejected) {
      continue;
    }
    const std::string& mid = content_info.name;
    if (bundle && bundle->HasContentName(mid) &&
        mid != *(bundle->FirstContentName())) {
      // This isn't the first media section in the BUNDLE group, so it's not
      // required to have ufrag/password, since only the ufrag/password from
      // the first section actually get used.
      continue;
    }

    // If the content isn't rejected or bundled into another m= section,
    // ice-ufrag and ice-pwd must be present.
    const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
    if (!tinfo) {
      // Something is not right.
      RTC_LOG(LS_ERROR) << kInvalidSdp;
      return false;
    }
    if (tinfo->description.ice_ufrag.empty() ||
        tinfo->description.ice_pwd.empty()) {
      RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
      return false;
    }
  }
  return true;
}

bool GetTrackIdBySsrc(const SessionDescription* session_description,
                      uint32_t ssrc,
                      std::string* track_id) {
  RTC_DCHECK(track_id != NULL);

  const cricket::AudioContentDescription* audio_desc =
      cricket::GetFirstAudioContentDescription(session_description);
  if (audio_desc) {
    const auto* found = cricket::GetStreamBySsrc(audio_desc->streams(), ssrc);
    if (found) {
      *track_id = found->id;
      return true;
    }
  }

  const cricket::VideoContentDescription* video_desc =
      cricket::GetFirstVideoContentDescription(session_description);
  if (video_desc) {
    const auto* found = cricket::GetStreamBySsrc(video_desc->streams(), ssrc);
    if (found) {
      *track_id = found->id;
      return true;
    }
  }
  return false;
}

// Get the SCTP port out of a SessionDescription.
// Return -1 if not found.
int GetSctpPort(const SessionDescription* session_description) {
  const cricket::DataContentDescription* data_desc =
      GetFirstDataContentDescription(session_description);
  RTC_DCHECK(data_desc);
  if (!data_desc) {
    return -1;
  }
  std::string value;
  cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
                                   cricket::kGoogleSctpDataCodecName);
  for (const cricket::DataCodec& codec : data_desc->codecs()) {
    if (!codec.Matches(match_pattern)) {
      continue;
    }
    if (codec.GetParam(cricket::kCodecParamPort, &value)) {
      return rtc::FromString<int>(value);
    }
  }
  return -1;
}

// Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
                              const SessionDescriptionInterface* new_desc,
                              const std::string& content_name) {
  if (!old_desc) {
    return false;
  }
  const SessionDescription* new_sd = new_desc->description();
  const SessionDescription* old_sd = old_desc->description();
  const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
  if (!cinfo || cinfo->rejected) {
    return false;
  }
  // If the content isn't rejected, check if ufrag and password has changed.
  const cricket::TransportDescription* new_transport_desc =
      new_sd->GetTransportDescriptionByName(content_name);
  const cricket::TransportDescription* old_transport_desc =
      old_sd->GetTransportDescriptionByName(content_name);
  if (!new_transport_desc || !old_transport_desc) {
    // No transport description exists. This is not an ICE restart.
    return false;
  }
  if (cricket::IceCredentialsChanged(
          old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
          new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
    RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
                     << ".";
    return true;
  }
  return false;
}

// Generates a string error message for SetLocalDescription/SetRemoteDescription
// from an RTCError.
std::string GetSetDescriptionErrorMessage(cricket::ContentSource source,
                                          SdpType type,
                                          const RTCError& error) {
  std::ostringstream oss;
  oss << "Failed to set " << (source == cricket::CS_LOCAL ? "local" : "remote")
      << " " << SdpTypeToString(type) << " sdp: " << error.message();
  return oss.str();
}

std::string GetStreamIdsString(rtc::ArrayView<const std::string> stream_ids) {
  std::string output = "streams=[";
  const char* separator = "";
  for (const auto& stream_id : stream_ids) {
    output.append(separator).append(stream_id);
    separator = ", ";
  }
  output.append("]");
  return output;
}

absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
    int rtc_configuration_parameter) {
  if (rtc_configuration_parameter ==
      webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
    return absl::nullopt;
  }
  return rtc_configuration_parameter;
}

}  // namespace

// Upon completion, posts a task to execute the callback of the
// SetSessionDescriptionObserver asynchronously on the same thread. At this
// point, the state of the peer connection might no longer reflect the effects
// of the SetRemoteDescription operation, as the peer connection could have been
// modified during the post.
// TODO(hbos): Remove this class once we remove the version of
// PeerConnectionInterface::SetRemoteDescription() that takes a
// SetSessionDescriptionObserver as an argument.
class PeerConnection::SetRemoteDescriptionObserverAdapter
    : public rtc::RefCountedObject<SetRemoteDescriptionObserverInterface> {
 public:
  SetRemoteDescriptionObserverAdapter(
      rtc::scoped_refptr<PeerConnection> pc,
      rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper)
      : pc_(std::move(pc)), wrapper_(std::move(wrapper)) {}

  // SetRemoteDescriptionObserverInterface implementation.
  void OnSetRemoteDescriptionComplete(RTCError error) override {
    if (error.ok())
      pc_->PostSetSessionDescriptionSuccess(wrapper_);
    else
      pc_->PostSetSessionDescriptionFailure(wrapper_, std::move(error));
  }

 private:
  rtc::scoped_refptr<PeerConnection> pc_;
  rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper_;
};

bool PeerConnectionInterface::RTCConfiguration::operator==(
    const PeerConnectionInterface::RTCConfiguration& o) const {
  // This static_assert prevents us from accidentally breaking operator==.
  // Note: Order matters! Fields must be ordered the same as RTCConfiguration.
  struct stuff_being_tested_for_equality {
    IceServers servers;
    IceTransportsType type;
    BundlePolicy bundle_policy;
    RtcpMuxPolicy rtcp_mux_policy;
    std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
    int ice_candidate_pool_size;
    bool disable_ipv6;
    bool disable_ipv6_on_wifi;
    int max_ipv6_networks;
    bool disable_link_local_networks;
    bool enable_rtp_data_channel;
    absl::optional<int> screencast_min_bitrate;
    absl::optional<bool> combined_audio_video_bwe;
    absl::optional<bool> enable_dtls_srtp;
    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;
    bool prioritize_most_likely_ice_candidate_pairs;
    struct cricket::MediaConfig media_config;
    bool prune_turn_ports;
    bool presume_writable_when_fully_relayed;
    bool enable_ice_renomination;
    bool redetermine_role_on_ice_restart;
    absl::optional<int> ice_check_interval_strong_connectivity;
    absl::optional<int> ice_check_interval_weak_connectivity;
    absl::optional<int> ice_check_min_interval;
    absl::optional<int> ice_unwritable_timeout;
    absl::optional<int> ice_unwritable_min_checks;
    absl::optional<int> stun_candidate_keepalive_interval;
    absl::optional<rtc::IntervalRange> ice_regather_interval_range;
    webrtc::TurnCustomizer* turn_customizer;
    SdpSemantics sdp_semantics;
    absl::optional<rtc::AdapterType> network_preference;
    bool active_reset_srtp_params;
  };
  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 &&
         disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
         max_ipv6_networks == o.max_ipv6_networks &&
         disable_link_local_networks == o.disable_link_local_networks &&
         enable_rtp_data_channel == o.enable_rtp_data_channel &&
         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 &&
         ice_check_interval_strong_connectivity ==
             o.ice_check_interval_strong_connectivity &&
         ice_check_interval_weak_connectivity ==
             o.ice_check_interval_weak_connectivity &&
         ice_check_min_interval == o.ice_check_min_interval &&
         ice_unwritable_timeout == o.ice_unwritable_timeout &&
         ice_unwritable_min_checks == o.ice_unwritable_min_checks &&
         stun_candidate_keepalive_interval ==
             o.stun_candidate_keepalive_interval &&
         ice_regather_interval_range == o.ice_regather_interval_range &&
         turn_customizer == o.turn_customizer &&
         sdp_semantics == o.sdp_semantics &&
         network_preference == o.network_preference &&
         active_reset_srtp_params == o.active_reset_srtp_params;
}

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)) {
    RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
    RTC_NOTREACHED();
  }
  return cname;
}

bool ValidateOfferAnswerOptions(
    const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
  return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
         IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
}

// From |rtc_options|, fill parts of |session_options| shared by all generated
// m= sections (in other words, nothing that involves a map/array).
void ExtractSharedMediaSessionOptions(
    const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
    cricket::MediaSessionOptions* session_options) {
  session_options->vad_enabled = rtc_options.voice_activity_detection;
  session_options->bundle_enabled = rtc_options.use_rtp_mux;
}

PeerConnection::PeerConnection(PeerConnectionFactory* factory,
                               std::unique_ptr<RtcEventLog> event_log,
                               std::unique_ptr<Call> call)
    : factory_(factory),
      event_log_(std::move(event_log)),
      rtcp_cname_(GenerateRtcpCname()),
      local_streams_(StreamCollection::Create()),
      remote_streams_(StreamCollection::Create()),
      call_(std::move(call)) {}

PeerConnection::~PeerConnection() {
  TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
  RTC_DCHECK_RUN_ON(signaling_thread());

  // Need to stop transceivers before destroying the stats collector because
  // AudioRtpSender has a reference to the StatsCollector it will update when
  // stopping.
  for (auto transceiver : transceivers_) {
    transceiver->Stop();
  }

  stats_.reset(nullptr);
  if (stats_collector_) {
    stats_collector_->WaitForPendingRequest();
    stats_collector_ = nullptr;
  }

  // Don't destroy BaseChannels until after stats has been cleaned up so that
  // the last stats request can still read from the channels.
  DestroyAllChannels();

  RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";

  webrtc_session_desc_factory_.reset();
  sctp_invoker_.reset();
  sctp_factory_.reset();
  transport_controller_.reset();

  // port_allocator_ lives on the network thread and should be destroyed there.
  network_thread()->Invoke<void>(RTC_FROM_HERE,
                                 [this] { port_allocator_.reset(); });
  // call_ and event_log_ must be destroyed on the worker thread.
  worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
    call_.reset();
    // The event log must outlive call (and any other object that uses it).
    event_log_.reset();
  });
}

void PeerConnection::DestroyAllChannels() {
  // Destroy video channels first since they may have a pointer to a voice
  // channel.
  for (auto transceiver : transceivers_) {
    if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
      DestroyTransceiverChannel(transceiver);
    }
  }
  for (auto transceiver : transceivers_) {
    if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
      DestroyTransceiverChannel(transceiver);
    }
  }
  DestroyDataChannel();
}

bool PeerConnection::Initialize(
    const PeerConnectionInterface::RTCConfiguration& configuration,
    PeerConnectionDependencies dependencies) {
  TRACE_EVENT0("webrtc", "PeerConnection::Initialize");

  RTCError config_error = ValidateConfiguration(configuration);
  if (!config_error.ok()) {
    RTC_LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
    return false;
  }

  if (!dependencies.allocator) {
    RTC_LOG(LS_ERROR)
        << "PeerConnection initialized without a PortAllocator? "
           "This shouldn't happen if using PeerConnectionFactory.";
    return false;
  }

  if (!dependencies.observer) {
    // TODO(deadbeef): Why do we do this?
    RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
                         "PeerConnectionObserver";
    return false;
  }

  observer_ = dependencies.observer;
  async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
  port_allocator_ = std::move(dependencies.allocator);
  tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);

  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 false;
  }

  // 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,
                    stun_servers, turn_servers, configuration))) {
    return false;
  }
  // If initialization was successful, note if STUN or TURN servers
  // were supplied.
  if (!stun_servers.empty()) {
    NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
  }
  if (!turn_servers.empty()) {
    NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
  }

  // Send information about IPv4/IPv6 status.
  PeerConnectionAddressFamilyCounter address_family;
  if (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6) {
    address_family = kPeerConnection_IPv6;
  } else {
    address_family = kPeerConnection_IPv4;
  }
  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
                            kPeerConnectionAddressFamilyCounter_Max);

  const PeerConnectionFactoryInterface::Options& options = factory_->options();

  // RFC 3264: The numeric value of the session id and version in the
  // o line MUST be representable with a "64 bit signed integer".
  // Due to this constraint session id |session_id_| is max limited to
  // LLONG_MAX.
  session_id_ = rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX);
  JsepTransportController::Config config;
  config.redetermine_role_on_ice_restart =
      configuration.redetermine_role_on_ice_restart;
  config.ssl_max_version = factory_->options().ssl_max_version;
  config.disable_encryption = options.disable_encryption;
  config.bundle_policy = configuration.bundle_policy;
  config.rtcp_mux_policy = configuration.rtcp_mux_policy;
  config.crypto_options = options.crypto_options;
  config.transport_observer = this;
  config.event_log = event_log_.get();
#if defined(ENABLE_EXTERNAL_AUTH)
  config.enable_external_auth = true;
#endif
  config.active_reset_srtp_params = configuration.active_reset_srtp_params;
  transport_controller_.reset(new JsepTransportController(
      signaling_thread(), network_thread(), port_allocator_.get(),
      async_resolver_factory_.get(), config));
  transport_controller_->SignalIceConnectionState.connect(
      this, &PeerConnection::OnTransportControllerConnectionState);
  transport_controller_->SignalIceGatheringState.connect(
      this, &PeerConnection::OnTransportControllerGatheringState);
  transport_controller_->SignalIceCandidatesGathered.connect(
      this, &PeerConnection::OnTransportControllerCandidatesGathered);
  transport_controller_->SignalIceCandidatesRemoved.connect(
      this, &PeerConnection::OnTransportControllerCandidatesRemoved);
  transport_controller_->SignalDtlsHandshakeError.connect(
      this, &PeerConnection::OnTransportControllerDtlsHandshakeError);

  sctp_factory_ = factory_->CreateSctpTransportInternalFactory();

  stats_.reset(new StatsCollector(this));
  stats_collector_ = RTCStatsCollector::Create(this);

  configuration_ = configuration;

  // Obtain a certificate from RTCConfiguration if any were provided (optional).
  rtc::scoped_refptr<rtc::RTCCertificate> certificate;
  if (!configuration.certificates.empty()) {
    // TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
    // just picking the first one. The decision should be made based on the DTLS
    // handshake. The DTLS negotiations need to know about all certificates.
    certificate = configuration.certificates[0];
  }

  transport_controller_->SetIceConfig(ParseIceConfig(configuration));

  if (options.disable_encryption) {
    dtls_enabled_ = false;
  } else {
    // Enable DTLS by default if we have an identity store or a certificate.
    dtls_enabled_ = (dependencies.cert_generator || certificate);
    // |configuration| can override the default |dtls_enabled_| value.
    if (configuration.enable_dtls_srtp) {
      dtls_enabled_ = *(configuration.enable_dtls_srtp);
    }
  }

  // Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
  // It takes precendence over the disable_sctp_data_channels
  // PeerConnectionFactoryInterface::Options.
  if (configuration.enable_rtp_data_channel) {
    data_channel_type_ = cricket::DCT_RTP;
  } else {
    // DTLS has to be enabled to use SCTP.
    if (!options.disable_sctp_data_channels && dtls_enabled_) {
      data_channel_type_ = cricket::DCT_SCTP;
    }
  }

  video_options_.screencast_min_bitrate_kbps =
      configuration.screencast_min_bitrate;
  audio_options_.combined_audio_video_bwe =
      configuration.combined_audio_video_bwe;

  audio_options_.audio_jitter_buffer_max_packets =
      configuration.audio_jitter_buffer_max_packets;

  audio_options_.audio_jitter_buffer_fast_accelerate =
      configuration.audio_jitter_buffer_fast_accelerate;

  // Whether the certificate generator/certificate is null or not determines
  // what PeerConnectionDescriptionFactory will do, so make sure that we give it
  // the right instructions by clearing the variables if needed.
  if (!dtls_enabled_) {
    dependencies.cert_generator.reset();
    certificate = nullptr;
  } else if (certificate) {
    // Favor generated certificate over the certificate generator.
    dependencies.cert_generator.reset();
  }

  webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
      signaling_thread(), channel_manager(), this, session_id(),
      std::move(dependencies.cert_generator), certificate));
  webrtc_session_desc_factory_->SignalCertificateReady.connect(
      this, &PeerConnection::OnCertificateReady);

  if (options.disable_encryption) {
    webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
  }

  webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
      options.crypto_options.enable_encrypted_rtp_header_extensions);

  // Add default audio/video transceivers for Plan B SDP.
  if (!IsUnifiedPlan()) {
    transceivers_.push_back(
        RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
            signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
    transceivers_.push_back(
        RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
            signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
  }
  int delay_ms =
      return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
  signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
                                  MSG_REPORT_USAGE_PATTERN, nullptr);
  return true;
}

RTCError PeerConnection::ValidateConfiguration(
    const RTCConfiguration& config) const {
  if (config.ice_regather_interval_range &&
      config.continual_gathering_policy == GATHER_ONCE) {
    return RTCError(RTCErrorType::INVALID_PARAMETER,
                    "ice_regather_interval_range specified but continual "
                    "gathering policy is GATHER_ONCE");
  }
  auto result =
      cricket::P2PTransportChannel::ValidateIceConfig(ParseIceConfig(config));
  return result;
}

rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
  RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
                                 "Plan SdpSemantics. Please use GetSenders "
                                 "instead.";
  return local_streams_;
}

rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
  RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
                                 "Plan SdpSemantics. Please use GetReceivers "
                                 "instead.";
  return remote_streams_;
}

bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
  RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
                                 "SdpSemantics. Please use AddTrack instead.";
  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()) {
    AddAudioTrack(track.get(), local_stream);
  }
  for (const auto& track : local_stream->GetVideoTracks()) {
    AddVideoTrack(track.get(), local_stream);
  }

  stats_->AddStream(local_stream);
  Observer()->OnRenegotiationNeeded();
  return true;
}

void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
  RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
                                 "Plan SdpSemantics. Please use RemoveTrack "
                                 "instead.";
  TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
  if (!IsClosed()) {
    for (const auto& track : local_stream->GetAudioTracks()) {
      RemoveAudioTrack(track.get(), local_stream);
    }
    for (const auto& track : local_stream->GetVideoTracks()) {
      RemoveVideoTrack(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()->id().compare(local_stream->id()) == 0;
          }),
      stream_observers_.end());

  if (IsClosed()) {
    return;
  }
  Observer()->OnRenegotiationNeeded();
}

RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
    rtc::scoped_refptr<MediaStreamTrackInterface> track,
    const std::vector<std::string>& stream_ids) {
  TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
  if (!track) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
  }
  if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
        track->kind() == MediaStreamTrackInterface::kVideoKind)) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
                         "Track has invalid kind: " + track->kind());
  }
  if (IsClosed()) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
                         "PeerConnection is closed.");
  }
  if (FindSenderForTrack(track)) {
    LOG_AND_RETURN_ERROR(
        RTCErrorType::INVALID_PARAMETER,
        "Sender already exists for track " + track->id() + ".");
  }
  auto sender_or_error =
      (IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids)
                       : AddTrackPlanB(track, stream_ids));
  if (sender_or_error.ok()) {
    Observer()->OnRenegotiationNeeded();
    stats_->AddTrack(track);
  }
  return sender_or_error;
}

RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackPlanB(
    rtc::scoped_refptr<MediaStreamTrackInterface> track,
    const std::vector<std::string>& stream_ids) {
  if (stream_ids.size() > 1u) {
    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
                         "AddTrack with more than one stream is not "
                         "supported with Plan B semantics.");
  }
  std::vector<std::string> adjusted_stream_ids = stream_ids;
  if (adjusted_stream_ids.empty()) {
    adjusted_stream_ids.push_back(rtc::CreateRandomUuid());
  }
  cricket::MediaType media_type =
      (track->kind() == MediaStreamTrackInterface::kAudioKind
           ? cricket::MEDIA_TYPE_AUDIO
           : cricket::MEDIA_TYPE_VIDEO);
  auto new_sender =
      CreateSender(media_type, track->id(), track, adjusted_stream_ids);
  if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
    new_sender->internal()->SetVoiceMediaChannel(voice_media_channel());
    GetAudioTransceiver()->internal()->AddSender(new_sender);
    const RtpSenderInfo* sender_info =
        FindSenderInfo(local_audio_sender_infos_,
                       new_sender->internal()->stream_ids()[0], track->id());
    if (sender_info) {
      new_sender->internal()->SetSsrc(sender_info->first_ssrc);
    }
  } else {
    RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
    new_sender->internal()->SetVideoMediaChannel(video_media_channel());
    GetVideoTransceiver()->internal()->AddSender(new_sender);
    const RtpSenderInfo* sender_info =
        FindSenderInfo(local_video_sender_infos_,
                       new_sender->internal()->stream_ids()[0], track->id());
    if (sender_info) {
      new_sender->internal()->SetSsrc(sender_info->first_ssrc);
    }
  }
  return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
}

RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackUnifiedPlan(
    rtc::scoped_refptr<MediaStreamTrackInterface> track,
    const std::vector<std::string>& stream_ids) {
  auto transceiver = FindFirstTransceiverForAddedTrack(track);
  if (transceiver) {
    RTC_LOG(LS_INFO) << "Reusing an existing "
                     << cricket::MediaTypeToString(transceiver->media_type())
                     << " transceiver for AddTrack.";
    if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
      transceiver->internal()->set_direction(
          RtpTransceiverDirection::kSendRecv);
    } else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
      transceiver->internal()->set_direction(
          RtpTransceiverDirection::kSendOnly);
    }
    transceiver->sender()->SetTrack(track);
    transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
  } else {
    cricket::MediaType media_type =
        (track->kind() == MediaStreamTrackInterface::kAudioKind
             ? cricket::MEDIA_TYPE_AUDIO
             : cricket::MEDIA_TYPE_VIDEO);
    RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
                     << " transceiver in response to a call to AddTrack.";
    std::string sender_id = track->id();
    // Avoid creating a sender with an existing ID by generating a random ID.
    // This can happen if this is the second time AddTrack has created a sender
    // for this track.
    if (FindSenderById(sender_id)) {
      sender_id = rtc::CreateRandomUuid();
    }
    auto sender = CreateSender(media_type, sender_id, track, stream_ids);
    auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
    transceiver = CreateAndAddTransceiver(sender, receiver);
    transceiver->internal()->set_created_by_addtrack(true);
    transceiver->internal()->set_direction(RtpTransceiverDirection::kSendRecv);
  }
  return transceiver->sender();
}

rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindFirstTransceiverForAddedTrack(
    rtc::scoped_refptr<MediaStreamTrackInterface> track) {
  RTC_DCHECK(track);
  for (auto transceiver : transceivers_) {
    if (!transceiver->sender()->track() &&
        cricket::MediaTypeToString(transceiver->media_type()) ==
            track->kind() &&
        !transceiver->internal()->has_ever_been_used_to_send() &&
        !transceiver->stopped()) {
      return transceiver;
    }
  }
  return nullptr;
}

bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
  TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
  return RemoveTrackNew(sender).ok();
}

RTCError PeerConnection::RemoveTrackNew(
    rtc::scoped_refptr<RtpSenderInterface> sender) {
  if (!sender) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
  }
  if (IsClosed()) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
                         "PeerConnection is closed.");
  }
  if (IsUnifiedPlan()) {
    auto transceiver = FindTransceiverBySender(sender);
    if (!transceiver || !sender->track()) {
      return RTCError::OK();
    }
    sender->SetTrack(nullptr);
    if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
      transceiver->internal()->set_direction(
          RtpTransceiverDirection::kRecvOnly);
    } else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
      transceiver->internal()->set_direction(
          RtpTransceiverDirection::kInactive);
    }
  } else {
    bool removed;
    if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
      removed = GetAudioTransceiver()->internal()->RemoveSender(sender);
    } else {
      RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
      removed = GetVideoTransceiver()->internal()->RemoveSender(sender);
    }
    if (!removed) {
      LOG_AND_RETURN_ERROR(
          RTCErrorType::INVALID_PARAMETER,
          "Couldn't find sender " + sender->id() + " to remove.");
    }
  }
  Observer()->OnRenegotiationNeeded();
  return RTCError::OK();
}

rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindTransceiverBySender(
    rtc::scoped_refptr<RtpSenderInterface> sender) {
  for (auto transceiver : transceivers_) {
    if (transceiver->sender() == sender) {
      return transceiver;
    }
  }
  return nullptr;
}

RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
    rtc::scoped_refptr<MediaStreamTrackInterface> track) {
  return AddTransceiver(track, RtpTransceiverInit());
}

RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
    rtc::scoped_refptr<MediaStreamTrackInterface> track,
    const RtpTransceiverInit& init) {
  RTC_CHECK(IsUnifiedPlan())
      << "AddTransceiver is only available with Unified Plan SdpSemantics";
  if (!track) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
  }
  cricket::MediaType media_type;
  if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
    media_type = cricket::MEDIA_TYPE_AUDIO;
  } else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
    media_type = cricket::MEDIA_TYPE_VIDEO;
  } else {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
                         "Track kind is not audio or video");
  }
  return AddTransceiver(media_type, track, init);
}

RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
  return AddTransceiver(media_type, RtpTransceiverInit());
}

RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type,
                               const RtpTransceiverInit& init) {
  RTC_CHECK(IsUnifiedPlan())
      << "AddTransceiver is only available with Unified Plan SdpSemantics";
  if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
        media_type == cricket::MEDIA_TYPE_VIDEO)) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
                         "media type is not audio or video");
  }
  return AddTransceiver(media_type, nullptr, init);
}

RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
    cricket::MediaType media_type,
    rtc::scoped_refptr<MediaStreamTrackInterface> track,
    const RtpTransceiverInit& init,
    bool fire_callback) {
  RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
              media_type == cricket::MEDIA_TYPE_VIDEO));
  if (track) {
    RTC_DCHECK_EQ(media_type,
                  (track->kind() == MediaStreamTrackInterface::kAudioKind
                       ? cricket::MEDIA_TYPE_AUDIO
                       : cricket::MEDIA_TYPE_VIDEO));
  }

  // TODO(bugs.webrtc.org/7600): Verify init.

  RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
                   << " transceiver in response to a call to AddTransceiver.";
  // Set the sender ID equal to the track ID if the track is specified unless
  // that sender ID is already in use.
  std::string sender_id =
      (track && !FindSenderById(track->id()) ? track->id()
                                             : rtc::CreateRandomUuid());
  auto sender = CreateSender(media_type, sender_id, track, init.stream_ids);
  auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
  auto transceiver = CreateAndAddTransceiver(sender, receiver);
  transceiver->internal()->set_direction(init.direction);

  if (fire_callback) {
    Observer()->OnRenegotiationNeeded();
  }

  return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
}

rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::CreateSender(
    cricket::MediaType media_type,
    const std::string& id,
    rtc::scoped_refptr<MediaStreamTrackInterface> track,
    const std::vector<std::string>& stream_ids) {
  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
    RTC_DCHECK(!track ||
               (track->kind() == MediaStreamTrackInterface::kAudioKind));
    sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
        signaling_thread(),
        new AudioRtpSender(worker_thread(), id, stats_.get()));
    NoteUsageEvent(UsageEvent::AUDIO_ADDED);
  } else {
    RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
    RTC_DCHECK(!track ||
               (track->kind() == MediaStreamTrackInterface::kVideoKind));
    sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
        signaling_thread(), new VideoRtpSender(worker_thread(), id));
    NoteUsageEvent(UsageEvent::VIDEO_ADDED);
  }
  bool set_track_succeeded = sender->SetTrack(track);
  RTC_DCHECK(set_track_succeeded);
  sender->internal()->set_stream_ids(stream_ids);
  return sender;
}

rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
PeerConnection::CreateReceiver(cricket::MediaType media_type,
                               const std::string& receiver_id) {
  rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
      receiver;
  if (media_type == cricket::MEDIA_TYPE_AUDIO) {
    receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
        signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
                                                 std::vector<std::string>({})));
    NoteUsageEvent(UsageEvent::AUDIO_ADDED);
  } else {
    RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
    receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
        signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
                                                 std::vector<std::string>({})));
    NoteUsageEvent(UsageEvent::VIDEO_ADDED);
  }
  return receiver;
}

rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::CreateAndAddTransceiver(
    rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
    rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
        receiver) {
  // Ensure that the new sender does not have an ID that is already in use by
  // another sender.
  // Allow receiver IDs to conflict since those come from remote SDP (which
  // could be invalid, but should not cause a crash).
  RTC_DCHECK(!FindSenderById(sender->id()));
  auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
      signaling_thread(), new RtpTransceiver(sender, receiver));
  transceivers_.push_back(transceiver);
  transceiver->internal()->SignalNegotiationNeeded.connect(
      this, &PeerConnection::OnNegotiationNeeded);
  return transceiver;
}

void PeerConnection::OnNegotiationNeeded() {
  RTC_DCHECK_RUN_ON(signaling_thread());
  RTC_DCHECK(!IsClosed());
  Observer()->OnRenegotiationNeeded();
}

rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
    const std::string& kind,
    const std::string& stream_id) {
  RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
                                 "Plan SdpSemantics. Please use AddTransceiver "
                                 "instead.";
  TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
  if (IsClosed()) {
    return nullptr;
  }

  // Internally we need to have one stream with Plan B semantics, so we
  // generate a random stream ID if not specified.
  std::vector<std::string> stream_ids;
  if (stream_id.empty()) {
    stream_ids.push_back(rtc::CreateRandomUuid());
    RTC_LOG(LS_INFO)
        << "No stream_id specified for sender. Generated stream ID: "
        << stream_ids[0];
  } else {
    stream_ids.push_back(stream_id);
  }

  // TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
  rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
  if (kind == MediaStreamTrackInterface::kAudioKind) {
    auto* audio_sender = new AudioRtpSender(
        worker_thread(), rtc::CreateRandomUuid(), stats_.get());
    audio_sender->SetVoiceMediaChannel(voice_media_channel());
    new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
        signaling_thread(), audio_sender);
    GetAudioTransceiver()->internal()->AddSender(new_sender);
  } else if (kind == MediaStreamTrackInterface::kVideoKind) {
    auto* video_sender =
        new VideoRtpSender(worker_thread(), rtc::CreateRandomUuid());
    video_sender->SetVideoMediaChannel(video_media_channel());
    new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
        signaling_thread(), video_sender);
    GetVideoTransceiver()->internal()->AddSender(new_sender);
  } else {
    RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
    return nullptr;
  }
  new_sender->internal()->set_stream_ids(stream_ids);

  return new_sender;
}

std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
    const {
  std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
  for (auto sender : GetSendersInternal()) {
    ret.push_back(sender);
  }
  return ret;
}

std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
PeerConnection::GetSendersInternal() const {
  std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
      all_senders;
  for (auto transceiver : transceivers_) {
    auto senders = transceiver->internal()->senders();
    all_senders.insert(all_senders.end(), senders.begin(), senders.end());
  }
  return all_senders;
}

std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
  std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
  for (const auto& receiver : GetReceiversInternal()) {
    ret.push_back(receiver);
  }
  return ret;
}

std::vector<
    rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
PeerConnection::GetReceiversInternal() const {
  std::vector<
      rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
      all_receivers;
  for (auto transceiver : transceivers_) {
    auto receivers = transceiver->internal()->receivers();
    all_receivers.insert(all_receivers.end(), receivers.begin(),
                         receivers.end());
  }
  return all_receivers;
}

std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::GetTransceivers() const {
  RTC_CHECK(IsUnifiedPlan())
      << "GetTransceivers is only supported with Unified Plan SdpSemantics.";
  std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
  for (auto transceiver : transceivers_) {
    all_transceivers.push_back(transceiver);
  }
  return all_transceivers;
}

bool PeerConnection::GetStats(StatsObserver* observer,
                              MediaStreamTrackInterface* track,
                              StatsOutputLevel level) {
  TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
  RTC_DCHECK(signaling_thread()->IsCurrent());
  if (!observer) {
    RTC_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())) {
    RTC_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) {
  TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
  RTC_DCHECK(stats_collector_);
  RTC_DCHECK(callback);
  stats_collector_->GetStatsReport(callback);
}

void PeerConnection::GetStats(
    rtc::scoped_refptr<RtpSenderInterface> selector,
    rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
  TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
  RTC_DCHECK(callback);
  RTC_DCHECK(stats_collector_);
  rtc::scoped_refptr<RtpSenderInternal> internal_sender;
  if (selector) {
    for (const auto& proxy_transceiver : transceivers_) {
      for (const auto& proxy_sender :
           proxy_transceiver->internal()->senders()) {
        if (proxy_sender == selector) {
          internal_sender = proxy_sender->internal();
          break;
        }
      }
      if (internal_sender)
        break;
    }
  }
  // If there is no |internal_sender| then |selector| is either null or does not
  // belong to the PeerConnection (in Plan B, senders can be removed from the
  // PeerConnection). This means that "all the stats objects representing the
  // selector" is an empty set. Invoking GetStatsReport() with a null selector
  // produces an empty stats report.
  stats_collector_->GetStatsReport(internal_sender, callback);
}

void PeerConnection::GetStats(
    rtc::scoped_refptr<RtpReceiverInterface> selector,
    rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
  TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
  RTC_DCHECK(callback);
  RTC_DCHECK(stats_collector_);
  rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
  if (selector) {
    for (const auto& proxy_transceiver : transceivers_) {
      for (const auto& proxy_receiver :
           proxy_transceiver->internal()->receivers()) {
        if (proxy_receiver == selector) {
          internal_receiver = proxy_receiver->internal();
          break;
        }
      }
      if (internal_receiver)
        break;
    }
  }
  // If there is no |internal_receiver| then |selector| is either null or does
  // not belong to the PeerConnection (in Plan B, receivers can be removed from
  // the PeerConnection). This means that "all the stats objects representing
  // the selector" is an empty set. Invoking GetStatsReport() with a null
  // selector produces an empty stats report.
  stats_collector_->GetStatsReport(internal_receiver, 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");

  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 (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
    Observer()->OnRenegotiationNeeded();
  }
  NoteUsageEvent(UsageEvent::DATA_ADDED);
  return DataChannelProxy::Create(signaling_thread(), channel.get());
}

void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
                                 const RTCOfferAnswerOptions& options) {
  TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");

  if (!observer) {
    RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
    return;
  }

  if (IsClosed()) {
    std::string error = "CreateOffer called when PeerConnection is closed.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailure(
        observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
    return;
  }

  if (!ValidateOfferAnswerOptions(options)) {
    std::string error = "CreateOffer called with invalid options.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailure(
        observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
    return;
  }

  // Legacy handling for offer_to_receive_audio and offer_to_receive_video.
  // Specified in WebRTC section 4.4.3.2 "Legacy configuration extensions".
  if (IsUnifiedPlan()) {
    RTCError error = HandleLegacyOfferOptions(options);
    if (!error.ok()) {
      PostCreateSessionDescriptionFailure(observer, std::move(error));
      return;
    }
  }

  cricket::MediaSessionOptions session_options;
  GetOptionsForOffer(options, &session_options);
  webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
}

RTCError PeerConnection::HandleLegacyOfferOptions(
    const RTCOfferAnswerOptions& options) {
  RTC_DCHECK(IsUnifiedPlan());

  if (options.offer_to_receive_audio == 0) {
    RemoveRecvDirectionFromReceivingTransceiversOfType(
        cricket::MEDIA_TYPE_AUDIO);
  } else if (options.offer_to_receive_audio == 1) {
    AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
  } else if (options.offer_to_receive_audio > 1) {
    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
                         "offer_to_receive_audio > 1 is not supported.");
  }

  if (options.offer_to_receive_video == 0) {
    RemoveRecvDirectionFromReceivingTransceiversOfType(
        cricket::MEDIA_TYPE_VIDEO);
  } else if (options.offer_to_receive_video == 1) {
    AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
  } else if (options.offer_to_receive_video > 1) {
    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
                         "offer_to_receive_video > 1 is not supported.");
  }

  return RTCError::OK();
}

void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
    cricket::MediaType media_type) {
  for (auto transceiver : GetReceivingTransceiversOfType(media_type)) {
    RtpTransceiverDirection new_direction =
        RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
    if (new_direction != transceiver->direction()) {
      RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
                       << " transceiver (MID="
                       << transceiver->mid().value_or("<not set>") << ") from "
                       << RtpTransceiverDirectionToString(
                              transceiver->direction())
                       << " to "
                       << RtpTransceiverDirectionToString(new_direction)
                       << " since CreateOffer specified offer_to_receive=0";
      transceiver->internal()->set_direction(new_direction);
    }
  }
}

void PeerConnection::AddUpToOneReceivingTransceiverOfType(
    cricket::MediaType media_type) {
  if (GetReceivingTransceiversOfType(media_type).empty()) {
    RTC_LOG(LS_INFO)
        << "Adding one recvonly " << cricket::MediaTypeToString(media_type)
        << " transceiver since CreateOffer specified offer_to_receive=1";
    RtpTransceiverInit init;
    init.direction = RtpTransceiverDirection::kRecvOnly;
    AddTransceiver(media_type, nullptr, init, /*fire_callback=*/false);
  }
}

std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
  std::vector<
      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
      receiving_transceivers;
  for (auto transceiver : transceivers_) {
    if (!transceiver->stopped() && transceiver->media_type() == media_type &&
        RtpTransceiverDirectionHasRecv(transceiver->direction())) {
      receiving_transceivers.push_back(transceiver);
    }
  }
  return receiving_transceivers;
}

void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
                                  const RTCOfferAnswerOptions& options) {
  TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
  if (!observer) {
    RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
    return;
  }

  if (!(signaling_state_ == kHaveRemoteOffer ||
        signaling_state_ == kHaveLocalPrAnswer)) {
    std::string error =
        "PeerConnection cannot create an answer in a state other than "
        "have-remote-offer or have-local-pranswer.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailure(
        observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
    return;
  }

  // The remote description should be set if we're in the right state.
  RTC_DCHECK(remote_description());

  if (IsUnifiedPlan()) {
    if (options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
                             "supported with Unified Plan semantics. Use the "
                             "RtpTransceiver API instead.";
    }
    if (options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
                             "supported with Unified Plan semantics. Use the "
                             "RtpTransceiver API instead.";
    }
  }

  cricket::MediaSessionOptions session_options;
  GetOptionsForAnswer(options, &session_options);

  webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
}

void PeerConnection::SetLocalDescription(
    SetSessionDescriptionObserver* observer,
    SessionDescriptionInterface* desc_ptr) {
  TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");

  // The SetLocalDescription contract is that we take ownership of the session
  // description regardless of the outcome, so wrap it in a unique_ptr right
  // away. Ideally, SetLocalDescription's signature will be changed to take the
  // description as a unique_ptr argument to formalize this agreement.
  std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr);

  if (!observer) {
    RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
    return;
  }

  if (!desc) {
    PostSetSessionDescriptionFailure(
        observer,
        RTCError(RTCErrorType::INTERNAL_ERROR, "SessionDescription is NULL."));
    return;
  }

  // If a session error has occurred the PeerConnection is in a possibly
  // inconsistent state so fail right away.
  if (session_error() != SessionError::kNone) {
    std::string error_message = GetSessionErrorMsg();
    RTC_LOG(LS_ERROR) << "SetLocalDescription: " << error_message;
    PostSetSessionDescriptionFailure(
        observer,
        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
    return;
  }

  RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
  if (!error.ok()) {
    std::string error_message = GetSetDescriptionErrorMessage(
        cricket::CS_LOCAL, desc->GetType(), error);
    RTC_LOG(LS_ERROR) << error_message;
    PostSetSessionDescriptionFailure(
        observer,
        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
    return;
  }

  // Grab the description type before moving ownership to ApplyLocalDescription,
  // which may destroy it before returning.
  const SdpType type = desc->GetType();

  error = ApplyLocalDescription(std::move(desc));
  // |desc| may be destroyed at this point.

  if (!error.ok()) {
    // If ApplyLocalDescription fails, the PeerConnection could be in an
    // inconsistent state, so act conservatively here and set the session error
    // so that future calls to SetLocalDescription/SetRemoteDescription fail.
    SetSessionError(SessionError::kContent, error.message());
    std::string error_message =
        GetSetDescriptionErrorMessage(cricket::CS_LOCAL, type, error);
    RTC_LOG(LS_ERROR) << error_message;
    PostSetSessionDescriptionFailure(
        observer,
        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
    return;
  }
  RTC_DCHECK(local_description());

  PostSetSessionDescriptionSuccess(observer);

  // MaybeStartGathering needs to be called after posting
  // MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
  // before signaling that SetLocalDescription completed.
  transport_controller_->MaybeStartGathering();

  if (local_description()->GetType() == SdpType::kAnswer) {
    // TODO(deadbeef): We already had to hop to the network thread for
    // MaybeStartGathering...
    network_thread()->Invoke<void>(
        RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
                                 port_allocator_.get()));
    // Make UMA notes about what was agreed to.
    ReportNegotiatedSdpSemantics(*local_description());
  }
  NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED);
}

RTCError PeerConnection::ApplyLocalDescription(
    std::unique_ptr<SessionDescriptionInterface> desc) {
  RTC_DCHECK_RUN_ON(signaling_thread());
  RTC_DCHECK(desc);

  // 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);

  // Take a reference to the old local description since it's used below to
  // compare against the new local description. When setting the new local
  // description, grab ownership of the replaced session description in case it
  // is the same as |old_local_description|, to keep it alive for the duration
  // of the method.
  const SessionDescriptionInterface* old_local_description =
      local_description();
  std::unique_ptr<SessionDescriptionInterface> replaced_local_description;
  SdpType type = desc->GetType();
  if (type == SdpType::kAnswer) {
    replaced_local_description = pending_local_description_
                                     ? std::move(pending_local_description_)
                                     : std::move(current_local_description_);
    current_local_description_ = std::move(desc);
    pending_local_description_ = nullptr;
    current_remote_description_ = std::move(pending_remote_description_);
  } else {
    replaced_local_description = std::move(pending_local_description_);
    pending_local_description_ = std::move(desc);
  }
  // The session description to apply now must be accessed by
  // |local_description()|.
  RTC_DCHECK(local_description());

  RTCError error = PushdownTransportDescription(cricket::CS_LOCAL, type);
  if (!error.ok()) {
    return error;
  }

  if (IsUnifiedPlan()) {
    RTCError error = UpdateTransceiversAndDataChannels(
        cricket::CS_LOCAL, *local_description(), old_local_description,
        remote_description());
    if (!error.ok()) {
      return error;
    }
    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
    std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
    for (auto transceiver : transceivers_) {
      const ContentInfo* content =
          FindMediaSectionForTransceiver(transceiver, local_description());
      if (!content) {
        continue;
      }
      const MediaContentDescription* media_desc = content->media_description();
      // 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
      // the following steps:
      if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
        // 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
        // transceiver's [[FiredDirection]] slot is either "sendrecv" or
        // "recvonly", process the removal of a remote track for the media
        // description, given transceiver, removeList, and muteTracks.
        if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
            (transceiver->internal()->fired_direction() &&
             RtpTransceiverDirectionHasRecv(
                 *transceiver->internal()->fired_direction()))) {
          ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
                                      &removed_streams);
        }
        // 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
        // [[FiredDirection]] slots to direction.
        transceiver->internal()->set_current_direction(media_desc->direction());
        transceiver->internal()->set_fired_direction(media_desc->direction());
      }
    }
    auto observer = Observer();
    for (auto transceiver : remove_list) {
      observer->OnRemoveTrack(transceiver->receiver());
    }
    for (auto stream : removed_streams) {
      observer->OnRemoveStream(stream);
    }
  } else {
    // Media channels will be created only when offer is set. These may use new
    // transports just created by PushdownTransportDescription.
    if (type == SdpType::kOffer) {
      // TODO(bugs.webrtc.org/4676) - Handle CreateChannel failure, as new local
      // description is applied. Restore back to old description.
      RTCError error = CreateChannels(*local_description()->description());
      if (!error.ok()) {
        return error;
      }
    }
    // Remove unused channels if MediaContentDescription is rejected.
    RemoveUnusedChannels(local_description()->description());
  }

  error = UpdateSessionState(type, cricket::CS_LOCAL,
                             local_description()->description());
  if (!error.ok()) {
    return error;
  }

  if (remote_description()) {
    // Now that we have a local description, we can push down remote candidates.
    UseCandidatesInSessionDescription(remote_description());
  }

  pending_ice_restarts_.clear();
  if (session_error() != SessionError::kNone) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
  }

  // If setting the description decided our SSL role, allocate any necessary
  // SCTP sids.
  rtc::SSLRole role;
  if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
    AllocateSctpSids(role);
  }

  if (IsUnifiedPlan()) {
    for (auto transceiver : transceivers_) {
      const ContentInfo* content =
          FindMediaSectionForTransceiver(transceiver, local_description());
      if (!content) {
        continue;
      }
      const auto& streams = content->media_description()->streams();
      if (!content->rejected && !streams.empty()) {
        transceiver->internal()->sender_internal()->set_stream_ids(
            streams[0].stream_ids());
        transceiver->internal()->sender_internal()->SetSsrc(
            streams[0].first_ssrc());
      } else {
        // 0 is a special value meaning "this sender has no associated send
        // stream". Need to call this so the sender won't attempt to configure
        // a no longer existing stream and run into DCHECKs in the lower
        // layers.
        transceiver->internal()->sender_internal()->SetSsrc(0);
      }
    }
  } else {
    // Plan B semantics.

    // Update state and SSRC of local MediaStreams and DataChannels based on the
    // local session description.
    const cricket::ContentInfo* audio_content =
        GetFirstAudioContent(local_description()->description());
    if (audio_content) {
      if (audio_content->rejected) {
        RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
      } else {
        const cricket::AudioContentDescription* audio_desc =
            audio_content->media_description()->as_audio();
        UpdateLocalSenders(audio_desc->streams(), audio_desc->type());
      }
    }

    const cricket::ContentInfo* video_content =
        GetFirstVideoContent(local_description()->description());
    if (video_content) {
      if (video_content->rejected) {
        RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
      } else {
        const cricket::VideoContentDescription* video_desc =
            video_content->media_description()->as_video();
        UpdateLocalSenders(video_desc->streams(), video_desc->type());
      }
    }
  }

  const cricket::ContentInfo* data_content =
      GetFirstDataContent(local_description()->description());
  if (data_content) {
    const cricket::DataContentDescription* data_desc =
        data_content->media_description()->as_data();
    if (rtc::starts_with(data_desc->protocol().data(),
                         cricket::kMediaProtocolRtpPrefix)) {
      UpdateLocalRtpDataChannels(data_desc->streams());
    }
  }

  return RTCError::OK();
}

void PeerConnection::SetRemoteDescription(
    SetSessionDescriptionObserver* observer,
    SessionDescriptionInterface* desc) {
  SetRemoteDescription(
      std::unique_ptr<SessionDescriptionInterface>(desc),
      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>(
          new SetRemoteDescriptionObserverAdapter(this, observer)));
}

void PeerConnection::SetRemoteDescription(
    std::unique_ptr<SessionDescriptionInterface> desc,
    rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
  TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");

  if (!observer) {
    RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
    return;
  }

  if (!desc) {
    observer->OnSetRemoteDescriptionComplete(RTCError(
        RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL."));
    return;
  }

  // If a session error has occurred the PeerConnection is in a possibly
  // inconsistent state so fail right away.
  if (session_error() != SessionError::kNone) {
    std::string error_message = GetSessionErrorMsg();
    RTC_LOG(LS_ERROR) << "SetRemoteDescription: " << error_message;
    observer->OnSetRemoteDescriptionComplete(
        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
    return;
  }

  if (desc->GetType() == SdpType::kOffer) {
    // Report to UMA the format of the received offer.
    ReportSdpFormatReceived(*desc);
  }

  RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
  if (!error.ok()) {
    std::string error_message = GetSetDescriptionErrorMessage(
        cricket::CS_REMOTE, desc->GetType(), error);
    RTC_LOG(LS_ERROR) << error_message;
    observer->OnSetRemoteDescriptionComplete(
        RTCError(error.type(), std::move(error_message)));
    return;
  }

  // Grab the description type before moving ownership to
  // ApplyRemoteDescription, which may destroy it before returning.
  const SdpType type = desc->GetType();

  error = ApplyRemoteDescription(std::move(desc));
  // |desc| may be destroyed at this point.

  if (!error.ok()) {
    // If ApplyRemoteDescription fails, the PeerConnection could be in an
    // inconsistent state, so act conservatively here and set the session error
    // so that future calls to SetLocalDescription/SetRemoteDescription fail.
    SetSessionError(SessionError::kContent, error.message());
    std::string error_message =
        GetSetDescriptionErrorMessage(cricket::CS_REMOTE, type, error);
    RTC_LOG(LS_ERROR) << error_message;
    observer->OnSetRemoteDescriptionComplete(
        RTCError(error.type(), std::move(error_message)));
    return;
  }
  RTC_DCHECK(remote_description());

  if (type == SdpType::kAnswer) {
    // TODO(deadbeef): We already had to hop to the network thread for
    // MaybeStartGathering...
    network_thread()->Invoke<void>(
        RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
                                 port_allocator_.get()));
    // Make UMA notes about what was agreed to.
    ReportNegotiatedSdpSemantics(*remote_description());
  }

  observer->OnSetRemoteDescriptionComplete(RTCError::OK());
  NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED);
}

RTCError PeerConnection::ApplyRemoteDescription(
    std::unique_ptr<SessionDescriptionInterface> desc) {
  RTC_DCHECK_RUN_ON(signaling_thread());
  RTC_DCHECK(desc);

  // 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);

  // Take a reference to the old remote description since it's used below to
  // compare against the new remote description. When setting the new remote
  // description, grab ownership of the replaced session description in case it
  // is the same as |old_remote_description|, to keep it alive for the duration
  // of the method.
  const SessionDescriptionInterface* old_remote_description =
      remote_description();
  std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
  SdpType type = desc->GetType();
  if (type == SdpType::kAnswer) {
    replaced_remote_description = pending_remote_description_
                                      ? std::move(pending_remote_description_)
                                      : std::move(current_remote_description_);
    current_remote_description_ = std::move(desc);
    pending_remote_description_ = nullptr;
    current_local_description_ = std::move(pending_local_description_);
  } else {
    replaced_remote_description = std::move(pending_remote_description_);
    pending_remote_description_ = std::move(desc);
  }
  // The session description to apply now must be accessed by
  // |remote_description()|.
  RTC_DCHECK(remote_description());

  RTCError error = PushdownTransportDescription(cricket::CS_REMOTE, type);
  if (!error.ok()) {
    return error;
  }
  // Transport and Media channels will be created only when offer is set.
  if (IsUnifiedPlan()) {
    RTCError error = UpdateTransceiversAndDataChannels(
        cricket::CS_REMOTE, *remote_description(), local_description(),
        old_remote_description);
    if (!error.ok()) {
      return error;
    }
  } else {
    // Media channels will be created only when offer is set. These may use new
    // transports just created by PushdownTransportDescription.
    if (type == SdpType::kOffer) {
      // TODO(mallinath) - Handle CreateChannel failure, as new local
      // description is applied. Restore back to old description.
      RTCError error = CreateChannels(*remote_description()->description());
      if (!error.ok()) {
        return error;
      }
    }
    // Remove unused channels if MediaContentDescription is rejected.
    RemoveUnusedChannels(remote_description()->description());
  }

  // NOTE: Candidates allocation will be initiated only when
  // SetLocalDescription is called.
  error = UpdateSessionState(type, cricket::CS_REMOTE,
                             remote_description()->description());
  if (!error.ok()) {
    return error;
  }

  if (local_description() &&
      !UseCandidatesInSessionDescription(remote_description())) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidCandidates);
  }

  if (old_remote_description) {
    for (const cricket::ContentInfo& content :
         old_remote_description->description()->contents()) {
      // Check if this new SessionDescription contains new ICE ufrag and
      // password that indicates the remote peer requests an ICE restart.
      // TODO(deadbeef): When we start storing both the current and pending
      // remote description, this should reset pending_ice_restarts and compare
      // against the current description.
      if (CheckForRemoteIceRestart(old_remote_description, remote_description(),
                                   content.name)) {
        if (type == SdpType::kOffer) {
          pending_ice_restarts_.insert(content.name);
        }
      } else {
        // We retain all received candidates only if ICE is not restarted.
        // When ICE is restarted, all previous candidates belong to an old
        // generation and should not be kept.
        // TODO(deadbeef): This goes against the W3C spec which says the remote
        // description should only contain candidates from the last set remote
        // description plus any candidates added since then. We should remove
        // this once we're sure it won't break anything.
        WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
            old_remote_description, content.name, mutable_remote_description());
      }
    }
  }

  if (session_error() != SessionError::kNone) {
    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
  }

  // Set the the ICE connection state to connecting since the connection may
  // become writable with peer reflexive candidates before any remote candidate
  // is signaled.
  // TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
  // is to have a new signal the indicates a change in checking state from the
  // transport and expose a new checking() member from transport that can be
  // read to determine the current checking state. The existing SignalConnecting
  // actually means "gathering candidates", so cannot be be used here.
  if (remote_description()->GetType() != SdpType::kOffer &&
      remote_description()->number_of_mediasections() > 0u &&
      ice_connection_state() == PeerConnectionInterface::kIceConnectionNew) {
    SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
  }

  // If setting the description decided our SSL role, allocate any necessary
  // SCTP sids.
  rtc::SSLRole role;
  if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
    AllocateSctpSids(role);
  }

  if (IsUnifiedPlan()) {
    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
        now_receiving_transceivers;
    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
    std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
    std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
    for (auto transceiver : transceivers_) {
      const ContentInfo* content =
          FindMediaSectionForTransceiver(transceiver, remote_description());
      if (!content) {
        continue;
      }
      const MediaContentDescription* media_desc = content->media_description();
      RtpTransceiverDirection local_direction =
          RtpTransceiverDirectionReversed(media_desc->direction());
      // From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6 "Set
      // the RTCSessionDescription: If direction is sendrecv or recvonly, and
      // transceiver's current direction is neither sendrecv nor recvonly,
      // process the addition of a remote track for the media description.
      std::vector<std::string> stream_ids;
      if (!media_desc->streams().empty()) {
        // The remote description has signaled the stream IDs.
        stream_ids = media_desc->streams()[0].stream_ids();
      }
      if (RtpTransceiverDirectionHasRecv(local_direction) &&
          (!transceiver->fired_direction() ||
           !RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
        RTC_LOG(LS_INFO) << "Processing the addition of a new track for MID="
                         << content->name << " (added to "
                         << GetStreamIdsString(stream_ids) << ").";

        std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams;
        for (const std::string& stream_id : stream_ids) {
          rtc::scoped_refptr<MediaStreamInterface> stream =
              remote_streams_->find(stream_id);
          if (!stream) {
            stream = MediaStreamProxy::Create(rtc::Thread::Current(),
                                              MediaStream::Create(stream_id));
            remote_streams_->AddStream(stream);
            added_streams.push_back(stream);
          }
          media_streams.push_back(stream);
        }
        // This will add the remote track to the streams.
        // TODO(hbos): When we remove remote_streams(), use set_stream_ids()
        // instead. https://crbug.com/webrtc/9480
        transceiver->internal()->receiver_internal()->SetStreams(media_streams);
        now_receiving_transceivers.push_back(transceiver);
      }
      // 2.2.8.1.7: If direction is "sendonly" or "inactive", and transceiver's
      // [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the
      // removal of a remote track for the media description, given transceiver,
      // removeList, and muteTracks.
      if (!RtpTransceiverDirectionHasRecv(local_direction) &&
          (transceiver->fired_direction() &&
           RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
        ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
                                    &removed_streams);
      }
      // 2.2.8.1.8: Set transceiver's [[FiredDirection]] slot to direction.
      transceiver->internal()->set_fired_direction(local_direction);
      // 2.2.8.1.9: If description is of type "answer" or "pranswer", then run
      // the following steps:
      if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
        // 2.2.8.1.9.1: Set transceiver's [[CurrentDirection]] slot to
        // direction.
        transceiver->internal()->set_current_direction(local_direction);
      }
      // 2.2.8.1.10: If the media description is rejected, and transceiver is
      // not already stopped, stop the RTCRtpTransceiver transceiver.
      if (content->rejected && !transceiver->stopped()) {
        RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->name
                         << " since the media section was rejected.";
        transceiver->Stop();
      }
      if (!content->rejected &&
          RtpTransceiverDirectionHasRecv(local_direction)) {
        // Set ssrc to 0 in the case of an unsignalled ssrc.
        uint32_t ssrc = 0;
        if (!media_desc->streams().empty() &&
            media_desc->streams()[0].has_ssrcs()) {
          ssrc = media_desc->streams()[0].first_ssrc();
        }
        transceiver->internal()->receiver_internal()->SetupMediaChannel(ssrc);
      }
    }
    // Once all processing has finished, fire off callbacks.
    auto observer = Observer();
    for (auto transceiver : now_receiving_transceivers) {
      stats_->AddTrack(transceiver->receiver()->track());
      observer->OnTrack(transceiver);
      observer->OnAddTrack(transceiver->receiver(),
                           transceiver->receiver()->streams());
    }
    for (auto stream : added_streams) {
      observer->OnAddStream(stream);
    }
    for (auto transceiver : remove_list) {
      observer->OnRemoveTrack(transceiver->receiver());
    }
    for (auto stream : removed_streams) {
      observer->OnRemoveStream(stream);
    }
  }

  const cricket::ContentInfo* audio_content =
      GetFirstAudioContent(remote_description()->description());
  const cricket::ContentInfo* video_content =
      GetFirstVideoContent(remote_description()->description());
  const cricket::AudioContentDescription* audio_desc =
      GetFirstAudioContentDescription(remote_description()->description());
  const cricket::VideoContentDescription* video_desc =
      GetFirstVideoContentDescription(remote_description()->description());
  const cricket::DataContentDescription* data_desc =
      GetFirstDataContentDescription(remote_description()->description());

  // Check if the descriptions include streams, just in case the peer supports
  // MSID, but doesn't indicate so with "a=msid-semantic".
  if (remote_description()->description()->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());

  if (!IsUnifiedPlan()) {
    // TODO(steveanton): When removing RTP senders/receivers in response to a
    // rejected media section, there is some cleanup logic that expects the
    // voice/ video channel to still be set. But in this method the voice/video
    // channel would have been destroyed by the SetRemoteDescription caller
    // above so the cleanup that relies on them fails to run. The RemoveSenders
    // calls should be moved to right before the DestroyChannel calls to fix
    // this.

    // Find all audio rtp streams and create corresponding remote AudioTracks
    // and MediaStreams.
    if (audio_content) {
      if (audio_content->rejected) {
        RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
      } else {
        bool default_audio_track_needed =
            !remote_peer_supports_msid_ &&
            RtpTransceiverDirectionHasSend(audio_desc->direction());
        UpdateRemoteSendersList(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) {
        RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
      } else {
        bool default_video_track_needed =
            !remote_peer_supports_msid_ &&
            RtpTransceiverDirectionHasSend(video_desc->direction());
        UpdateRemoteSendersList(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.
    auto observer = Observer();
    for (size_t i = 0; i < new_streams->count(); ++i) {
      MediaStreamInterface* new_stream = new_streams->at(i);
      stats_->AddStream(new_stream);
      observer->OnAddStream(
          rtc::scoped_refptr<MediaStreamInterface>(new_stream));
    }

    UpdateEndedRemoteMediaStreams();
  }

  return RTCError::OK();
}

void PeerConnection::ProcessRemovalOfRemoteTrack(
    rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
        transceiver,
    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
    std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
  RTC_DCHECK(transceiver->mid());
  RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
                   << *transceiver->mid();
  std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams =
      transceiver->internal()->receiver_internal()->streams();
  // This will remove the remote track from the streams.
  transceiver->internal()->receiver_internal()->set_stream_ids({});
  remove_list->push_back(transceiver);
  // Remove any streams that no longer have tracks.
  // TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead
  // of streams, see if the stream was removed by checking if this was the
  // last receiver with that stream ID.
  for (auto stream : media_streams) {
    if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
      remote_streams_->RemoveStream(stream);
      removed_streams->push_back(stream);
    }
  }
}

RTCError PeerConnection::UpdateTransceiversAndDataChannels(
    cricket::ContentSource source,
    const SessionDescriptionInterface& new_session,
    const SessionDescriptionInterface* old_local_description,
    const SessionDescriptionInterface* old_remote_description) {
  RTC_DCHECK(IsUnifiedPlan());

  const cricket::ContentGroup* bundle_group = nullptr;
  if (new_session.GetType() == SdpType::kOffer) {
    auto bundle_group_or_error =
        GetEarlyBundleGroup(*new_session.description());
    if (!bundle_group_or_error.ok()) {
      return bundle_group_or_error.MoveError();
    }
    bundle_group = bundle_group_or_error.MoveValue();
  }

  const ContentInfos& new_contents = new_session.description()->contents();
  for (size_t i = 0; i < new_contents.size(); ++i) {
    const cricket::ContentInfo& new_content = new_contents[i];
    cricket::MediaType media_type = new_content.media_description()->type();
    seen_mids_.insert(new_content.name);
    if (media_type == cricket::MEDIA_TYPE_AUDIO ||
        media_type == cricket::MEDIA_TYPE_VIDEO) {
      const cricket::ContentInfo* old_local_content = nullptr;
      if (old_local_description &&
          i < old_local_description->description()->contents().size()) {
        old_local_content =
            &old_local_description->description()->contents()[i];
      }
      const cricket::ContentInfo* old_remote_content = nullptr;
      if (old_remote_description &&
          i < old_remote_description->description()->contents().size()) {
        old_remote_content =
            &old_remote_description->description()->contents()[i];
      }
      auto transceiver_or_error =
          AssociateTransceiver(source, new_session.GetType(), i, new_content,
                               old_local_content, old_remote_content);
      if (!transceiver_or_error.ok()) {
        return transceiver_or_error.MoveError();
      }
      auto transceiver = transceiver_or_error.MoveValue();
      RTCError error =
          UpdateTransceiverChannel(transceiver, new_content, bundle_group);
      if (!error.ok()) {
        return error;
      }
    } else if (media_type == cricket::MEDIA_TYPE_DATA) {
      if (GetDataMid() && new_content.name != *GetDataMid()) {
        // Ignore all but the first data section.
        RTC_LOG(LS_INFO) << "Ignoring data media section with MID="
                         << new_content.name;
        continue;
      }
      RTCError error = UpdateDataChannel(source, new_content, bundle_group);
      if (!error.ok()) {
        return error;
      }
    } else {
      LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
                           "Unknown section type.");
    }
  }

  return RTCError::OK();
}

RTCError PeerConnection::UpdateTransceiverChannel(
    rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
        transceiver,
    const cricket::ContentInfo& content,
    const cricket::ContentGroup* bundle_group) {
  RTC_DCHECK(IsUnifiedPlan());
  RTC_DCHECK(transceiver);
  cricket::BaseChannel* channel = transceiver->internal()->channel();
  if (content.rejected) {
    if (channel) {
      transceiver->internal()->SetChannel(nullptr);
      DestroyBaseChannel(channel);
    }
  } else {
    if (!channel) {
      if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
        channel = CreateVoiceChannel(content.name);
      } else {
        RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
        channel = CreateVideoChannel(content.name);
      }
      if (!channel) {
        LOG_AND_RETURN_ERROR(
            RTCErrorType::INTERNAL_ERROR,
            "Failed to create channel for mid=" + content.name);
      }
      transceiver->internal()->SetChannel(channel);
    }
  }
  return RTCError::OK();
}

RTCError PeerConnection::UpdateDataChannel(
    cricket::ContentSource source,
    const cricket::ContentInfo& content,
    const cricket::ContentGroup* bundle_group) {
  if (data_channel_type_ == cricket::DCT_NONE) {
    // If data channels are disabled, ignore this media section. CreateAnswer
    // will take care of rejecting it.
    return RTCError::OK();
  }
  if (content.rejected) {
    DestroyDataChannel();
  } else {
    if (!rtp_data_channel_ && !sctp_transport_) {
      if (!CreateDataChannel(content.name)) {
        LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
                             "Failed to create data channel.");
      }
    }
    if (source == cricket::CS_REMOTE) {
      const MediaContentDescription* data_desc = content.media_description();
      if (data_desc && cricket::IsRtpProtocol(data_desc->protocol())) {
        UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
      }
    }
  }
  return RTCError::OK();
}

RTCErrorOr<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
PeerConnection::AssociateTransceiver(cricket::ContentSource source,
                                     SdpType type,
                                     size_t mline_index,
                                     const ContentInfo& content,
                                     const ContentInfo* old_local_content,
                                     const ContentInfo* old_remote_content) {
  RTC_DCHECK(IsUnifiedPlan());
  // If this is an offer then the m= section might be recycled. If the m=
  // section is being recycled (defined as: rejected in the current local or
  // remote description and not rejected in new description), dissociate the
  // currently associated RtpTransceiver by setting its mid property to null,
  // and discard the mapping between the transceiver and its m= section index.
  if (IsMediaSectionBeingRecycled(type, content, old_local_content,
                                  old_remote_content)) {
    // We want to dissociate the transceiver that has the rejected mid.
    const std::string& old_mid =
        (old_local_content && old_local_content->rejected)
            ? old_local_content->name
            : old_remote_content->name;
    auto old_transceiver = GetAssociatedTransceiver(old_mid);
    if (old_transceiver) {
      RTC_LOG(LS_INFO) << "Dissociating transceiver for MID=" << old_mid
                       << " since the media section is being recycled.";
      old_transceiver->internal()->set_mid(absl::nullopt);
      old_transceiver->internal()->set_mline_index(absl::nullopt);
    }
  }
  const MediaContentDescription* media_desc = content.media_description();
  auto transceiver = GetAssociatedTransceiver(content.name);
  if (source == cricket::CS_LOCAL) {
    // Find the RtpTransceiver that corresponds to this m= section, using the
    // mapping between transceivers and m= section indices established when
    // creating the offer.
    if (!transceiver) {
      transceiver = GetTransceiverByMLineIndex(mline_index);
    }
    if (!transceiver) {
      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
                           "Unknown transceiver");
    }
  } else {
    RTC_DCHECK_EQ(source, cricket::CS_REMOTE);
    // If the m= section is sendrecv or recvonly, and there are RtpTransceivers
    // of the same type...
    if (!transceiver &&
        RtpTransceiverDirectionHasRecv(media_desc->direction())) {
      transceiver = FindAvailableTransceiverToReceive(media_desc->type());
    }
    // If no RtpTransceiver was found in the previous step, create one with a
    // recvonly direction.
    if (!transceiver) {
      RTC_LOG(LS_INFO) << "Adding "
                       << cricket::MediaTypeToString(media_desc->type())
                       << " transceiver for MID=" << content.name
                       << " at i=" << mline_index
                       << " in response to the remote description.";
      std::string sender_id = rtc::CreateRandomUuid();
      auto sender = CreateSender(media_desc->type(), sender_id, nullptr, {});
      std::string receiver_id;
      if (!media_desc->streams().empty()) {
        receiver_id = media_desc->streams()[0].id;
      } else {
        receiver_id = rtc::CreateRandomUuid();
      }
      auto receiver = CreateReceiver(media_desc->type(), receiver_id);
      transceiver = CreateAndAddTransceiver(sender, receiver);
      transceiver->internal()->set_direction(
          RtpTransceiverDirection::kRecvOnly);
    }
  }
  RTC_DCHECK(transceiver);
  if (transceiver->media_type() != media_desc->type()) {
    LOG_AND_RETURN_ERROR(
        RTCErrorType::INVALID_PARAMETER,
        "Transceiver type does not match media description type.");
  }
  // Associate the found or created RtpTransceiver with the m= section by
  // setting the value of the RtpTransceiver's mid property to the MID of the m=
  // section, and establish a mapping between the transceiver and the index of
  // the m= section.
  transceiver->internal()->set_mid(content.name);
  transceiver->internal()->set_mline_index(mline_index);
  return std::move(transceiver);
}

rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetAssociatedTransceiver(const std::string& mid) const {
  RTC_DCHECK(IsUnifiedPlan());
  for (auto transceiver : transceivers_) {
    if (transceiver->mid() == mid) {
      return transceiver;
    }
  }
  return nullptr;
}

rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetTransceiverByMLineIndex(size_t mline_index) const {
  RTC_DCHECK(IsUnifiedPlan());
  for (auto transceiver : transceivers_) {
    if (transceiver->internal()->mline_index() == mline_index) {
      return transceiver;
    }
  }
  return nullptr;
}

rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindAvailableTransceiverToReceive(
    cricket::MediaType media_type) const {
  RTC_DCHECK(IsUnifiedPlan());
  // From JSEP section 5.10 (Applying a Remote Description):
  // If the m= section is sendrecv or recvonly, and there are RtpTransceivers of
  // the same type that were added to the PeerConnection by addTrack and are not
  // associated with any m= section and are not stopped, find the first such
  // RtpTransceiver.
  for (auto transceiver : transceivers_) {
    if (transceiver->media_type() == media_type &&
        transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
        !transceiver->stopped()) {
      return transceiver;
    }
  }
  return nullptr;
}

const cricket::ContentInfo* PeerConnection::FindMediaSectionForTransceiver(
    rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
        transceiver,
    const SessionDescriptionInterface* sdesc) const {
  RTC_DCHECK(transceiver);
  RTC_DCHECK(sdesc);
  if (IsUnifiedPlan()) {
    if (!transceiver->internal()->mid()) {
      // This transceiver is not associated with a media section yet.
      return nullptr;
    }
    return sdesc->description()->GetContentByName(
        *transceiver->internal()->mid());
  } else {
    // Plan B only allows at most one audio and one video section, so use the
    // first media section of that type.
    return cricket::GetFirstMediaContent(sdesc->description()->contents(),
                                         transceiver->media_type());
  }
}

PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
  return configuration_;
}

bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
                                      RTCError* error) {
  TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
  if (IsClosed()) {
    RTC_LOG(LS_ERROR) << "SetConfiguration: PeerConnection is closed.";
    return SafeSetError(RTCErrorType::INVALID_STATE, error);
  }

  // According to JSEP, after setLocalDescription, changing the candidate pool
  // size is not allowed, and changing the set of ICE servers will not result
  // in new candidates being gathered.
  if (local_description()