Merge from Chromium at DEPS revision 291560

This commit was generated by merge_to_master.py.

Change-Id: I64ade7f09d2b962ca9149b1fa2abfbc91bfe0832

28 files changed, 469 insertions, 396 deletions

diff --git a/app/webrtc/mediaconstraintsinterface.cc b/app/webrtc/mediaconstraintsinterface.cc
index 74f6542..d59716e 100644
--- a/app/webrtc/mediaconstraintsinterface.cc
+++ b/app/webrtc/mediaconstraintsinterface.cc
@@ -93,8 +93,6 @@ const char MediaConstraintsInterface::kEnableDscp[] = "googDscp";
 const char MediaConstraintsInterface::kEnableIPv6[] = "googIPv6";
 const char MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate[] =
     "googSuspendBelowMinBitrate";
-const char MediaConstraintsInterface::kImprovedWifiBwe[] =
-    "googImprovedWifiBwe";
 const char MediaConstraintsInterface::kNumUnsignalledRecvStreams[] =
     "googNumUnsignalledRecvStreams";
 const char MediaConstraintsInterface::kScreencastMinBitrate[] =
@@ -119,7 +117,6 @@ const char MediaConstraintsInterface::kHighBitrate[] =
 const char MediaConstraintsInterface::kVeryHighBitrate[] =
     "googVeryHighBitrate";
 const char MediaConstraintsInterface::kPayloadPadding[] = "googPayloadPadding";
-const char MediaConstraintsInterface::kOpusFec[] = "googOpusFec";
 
 
 // Set |value| to the value associated with the first appearance of |key|, or
diff --git a/app/webrtc/mediaconstraintsinterface.h b/app/webrtc/mediaconstraintsinterface.h
index b3efbb3..d74ffab 100644
--- a/app/webrtc/mediaconstraintsinterface.h
+++ b/app/webrtc/mediaconstraintsinterface.h
@@ -115,7 +115,6 @@ class MediaConstraintsInterface {
   // Temporary constraint to enable suspend below min bitrate feature.
   static const char kEnableVideoSuspendBelowMinBitrate[];
       // googSuspendBelowMinBitrate
-  static const char kImprovedWifiBwe[];  // googImprovedWifiBwe
   static const char kNumUnsignalledRecvStreams[];
       // googNumUnsignalledRecvStreams
   static const char kScreencastMinBitrate[];  // googScreencastMinBitrate
@@ -133,8 +132,6 @@ class MediaConstraintsInterface {
   static const char kHighBitrate[];  // googHighBitrate
   static const char kVeryHighBitrate[];  // googVeryHighBitrate
   static const char kPayloadPadding[];  // googPayloadPadding
-  // kOpusFec controls whether we ask the other side to turn on FEC for Opus.
-  static const char kOpusFec[];  // googOpusFec
 
   // The prefix of internal-only constraints whose JS set values should be
   // stripped by Chrome before passed down to Libjingle.
diff --git a/app/webrtc/peerconnection.cc b/app/webrtc/peerconnection.cc
index 5b61048..201269a 100644
--- a/app/webrtc/peerconnection.cc
+++ b/app/webrtc/peerconnection.cc
@@ -85,11 +85,12 @@ struct SetSessionDescriptionMsg : public rtc::MessageData {
 };
 
 struct GetStatsMsg : public rtc::MessageData {
-  explicit GetStatsMsg(webrtc::StatsObserver* observer)
-      : observer(observer) {
+  GetStatsMsg(webrtc::StatsObserver* observer,
+              webrtc::MediaStreamTrackInterface* track)
+      : observer(observer), track(track) {
   }
-  webrtc::StatsReports reports;
   rtc::scoped_refptr<webrtc::StatsObserver> observer;
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
 };
 
 // |in_str| should be of format
@@ -446,17 +447,15 @@ rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
 bool PeerConnection::GetStats(StatsObserver* observer,
                               MediaStreamTrackInterface* track,
                               StatsOutputLevel level) {
+  ASSERT(signaling_thread()->IsCurrent());
   if (!VERIFY(observer != NULL)) {
     LOG(LS_ERROR) << "GetStats - observer is NULL.";
     return false;
   }
 
   stats_->UpdateStats(level);
-  rtc::scoped_ptr<GetStatsMsg> msg(new GetStatsMsg(observer));
-  if (!stats_->GetStats(track, &(msg->reports))) {
-    return false;
-  }
-  signaling_thread()->Post(this, MSG_GETSTATS, msg.release());
+  signaling_thread()->Post(this, MSG_GETSTATS,
+                           new GetStatsMsg(observer, track));
   return true;
 }
 
@@ -757,7 +756,9 @@ void PeerConnection::OnMessage(rtc::Message* msg) {
     }
     case MSG_GETSTATS: {
       GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
-      param->observer->OnComplete(param->reports);
+      StatsReports reports;
+      stats_->GetStats(param->track, &reports);
+      param->observer->OnComplete(reports);
       delete param;
       break;
     }
diff --git a/app/webrtc/peerconnectionfactory.cc b/app/webrtc/peerconnectionfactory.cc
index 81d864c..5dccba8 100644
--- a/app/webrtc/peerconnectionfactory.cc
+++ b/app/webrtc/peerconnectionfactory.cc
@@ -239,14 +239,14 @@ bool PeerConnectionFactory::Initialize_s() {
       new cricket::DummyDeviceManager());
   // TODO:  Need to make sure only one VoE is created inside
   // WebRtcMediaEngine.
-  cricket::WebRtcMediaEngine* webrtc_media_engine(
-      new cricket::WebRtcMediaEngine(default_adm_.get(),
-                                     NULL,  // No secondary adm.
-                                     video_encoder_factory_.get(),
-                                     video_decoder_factory_.get()));
+  cricket::MediaEngineInterface* media_engine(
+      cricket::WebRtcMediaEngineFactory::Create(default_adm_.get(),
+                                                NULL,  // No secondary adm.
+                                                video_encoder_factory_.get(),
+                                                video_decoder_factory_.get()));
 
   channel_manager_.reset(new cricket::ChannelManager(
-      webrtc_media_engine, device_manager, worker_thread_));
+      media_engine, device_manager, worker_thread_));
   channel_manager_->SetVideoRtxEnabled(true);
   if (!channel_manager_->Init()) {
     return false;
diff --git a/app/webrtc/peerconnectioninterface.h b/app/webrtc/peerconnectioninterface.h
index 59785e8..6ef4847 100644
--- a/app/webrtc/peerconnectioninterface.h
+++ b/app/webrtc/peerconnectioninterface.h
@@ -113,7 +113,18 @@ class StreamCollectionInterface : public rtc::RefCountInterface {
 
 class StatsObserver : public rtc::RefCountInterface {
  public:
-  virtual void OnComplete(const std::vector<StatsReport>& reports) = 0;
+  // TODO(tommi): Remove.
+  virtual void OnComplete(const std::vector<StatsReport>& reports) {}
+
+  // TODO(tommi): Make pure virtual and remove implementation.
+  virtual void OnComplete(const StatsReports& reports) {
+    std::vector<StatsReportCopyable> report_copies;
+    for (size_t i = 0; i < reports.size(); ++i)
+      report_copies.push_back(StatsReportCopyable(*reports[i]));
+    std::vector<StatsReport>* r =
+        reinterpret_cast<std::vector<StatsReport>*>(&report_copies);
+     OnComplete(*r);
+   }
 
  protected:
   virtual ~StatsObserver() {}
diff --git a/app/webrtc/statscollector.cc b/app/webrtc/statscollector.cc
index 9e2d75b..5271827 100644
--- a/app/webrtc/statscollector.cc
+++ b/app/webrtc/statscollector.cc
@@ -234,7 +234,6 @@ void StatsReport::ReplaceValue(StatsReport::StatsValueName name,
 }
 
 namespace {
-typedef std::map<std::string, StatsReport> StatsMap;
 
 double GetTimeNow() {
   return rtc::Timing::WallTimeNow() * rtc::kNumMillisecsPerSec;
@@ -294,17 +293,16 @@ bool ExtractValueFromReport(
   return false;
 }
 
-void AddTrackReport(StatsMap* reports, const std::string& track_id) {
+void AddTrackReport(StatsSet* reports, const std::string& track_id) {
   // Adds an empty track report.
-  StatsReport report;
-  report.type = StatsReport::kStatsReportTypeTrack;
-  report.id = StatsId(StatsReport::kStatsReportTypeTrack, track_id);
-  report.AddValue(StatsReport::kStatsValueNameTrackId, track_id);
-  (*reports)[report.id] = report;
+  StatsReport* report = reports->ReplaceOrAddNew(
+      StatsId(StatsReport::kStatsReportTypeTrack, track_id));
+  report->type = StatsReport::kStatsReportTypeTrack;
+  report->AddValue(StatsReport::kStatsValueNameTrackId, track_id);
 }
 
 template <class TrackVector>
-void CreateTrackReports(const TrackVector& tracks, StatsMap* reports) {
+void CreateTrackReports(const TrackVector& tracks, StatsSet* reports) {
   for (size_t j = 0; j < tracks.size(); ++j) {
     webrtc::MediaStreamTrackInterface* track = tracks[j];
     AddTrackReport(reports, track->id());
@@ -469,7 +467,7 @@ void ExtractStats(const cricket::BandwidthEstimationInfo& info,
                   double stats_gathering_started,
                   PeerConnectionInterface::StatsOutputLevel level,
                   StatsReport* report) {
-  report->id = StatsReport::kStatsReportVideoBweId;
+  ASSERT(report->id == StatsReport::kStatsReportVideoBweId);
   report->type = StatsReport::kStatsReportTypeBwe;
 
   // Clear out stats from previous GatherStats calls if any.
@@ -583,9 +581,9 @@ void StatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track,
 
   // Create the kStatsReportTypeTrack report for the new track if there is no
   // report yet.
-  StatsMap::iterator it = reports_.find(
+  StatsReport* found = reports_.Find(
       StatsId(StatsReport::kStatsReportTypeTrack, audio_track->id()));
-  if (it == reports_.end())
+  if (!found)
     AddTrackReport(&reports_, audio_track->id());
 }
 
@@ -603,49 +601,46 @@ void StatsCollector::RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
   ASSERT(false);
 }
 
-bool StatsCollector::GetStats(MediaStreamTrackInterface* track,
+void StatsCollector::GetStats(MediaStreamTrackInterface* track,
                               StatsReports* reports) {
   ASSERT(reports != NULL);
-  reports->clear();
+  ASSERT(reports->empty());
 
-  StatsMap::iterator it;
   if (!track) {
-    for (it = reports_.begin(); it != reports_.end(); ++it) {
-      reports->push_back(it->second);
-    }
-    return true;
+    StatsSet::const_iterator it;
+    for (it = reports_.begin(); it != reports_.end(); ++it)
+      reports->push_back(&(*it));
+    return;
   }
 
-  it = reports_.find(StatsId(StatsReport::kStatsReportTypeSession,
-                             session_->id()));
-  if (it != reports_.end()) {
-    reports->push_back(it->second);
-  }
+  StatsReport* report =
+      reports_.Find(StatsId(StatsReport::kStatsReportTypeSession,
+                            session_->id()));
+  if (report)
+    reports->push_back(report);
 
-  it = reports_.find(StatsId(StatsReport::kStatsReportTypeTrack, track->id()));
+  report = reports_.Find(
+      StatsId(StatsReport::kStatsReportTypeTrack, track->id()));
 
-  if (it == reports_.end()) {
-    LOG(LS_WARNING) << "No StatsReport is available for "<< track->id();
-    return false;
-  }
+  if (!report)
+    return;
 
-  reports->push_back(it->second);
+  reports->push_back(report);
 
   std::string track_id;
-  for (it = reports_.begin(); it != reports_.end(); ++it) {
-    if (it->second.type != StatsReport::kStatsReportTypeSsrc) {
+  for (StatsSet::const_iterator it = reports_.begin(); it != reports_.end();
+       ++it) {
+    if (it->type != StatsReport::kStatsReportTypeSsrc)
       continue;
-    }
-    if (ExtractValueFromReport(it->second,
+
+    if (ExtractValueFromReport(*it,
                                StatsReport::kStatsValueNameTrackId,
                                &track_id)) {
       if (track_id == track->id()) {
-        reports->push_back(it->second);
+        reports->push_back(&(*it));
       }
     }
   }
-
-  return true;
 }
 
 void
@@ -672,13 +667,13 @@ StatsReport* StatsCollector::PrepareLocalReport(
     const std::string& transport_id,
     TrackDirection direction) {
   const std::string ssrc_id = rtc::ToString<uint32>(ssrc);
-  StatsMap::iterator it = reports_.find(StatsId(
-      StatsReport::kStatsReportTypeSsrc, ssrc_id, direction));
+  StatsReport* report = reports_.Find(
+      StatsId(StatsReport::kStatsReportTypeSsrc, ssrc_id, direction));
 
   // Use the ID of the track that is currently mapped to the SSRC, if any.
   std::string track_id;
   if (!GetTrackIdBySsrc(ssrc, &track_id, direction)) {
-    if (it == reports_.end()) {
+    if (!report) {
       // The ssrc is not used by any track or existing report, return NULL
       // in such case to indicate no report is prepared for the ssrc.
       return NULL;
@@ -686,13 +681,13 @@ StatsReport* StatsCollector::PrepareLocalReport(
 
     // The ssrc is not used by any existing track. Keeps the old track id
     // since we want to report the stats for inactive ssrc.
-    ExtractValueFromReport(it->second,
+    ExtractValueFromReport(*report,
                            StatsReport::kStatsValueNameTrackId,
                            &track_id);
   }
 
-  StatsReport* report = GetOrCreateReport(StatsReport::kStatsReportTypeSsrc,
-                                          ssrc_id, direction);
+  report = GetOrCreateReport(
+      StatsReport::kStatsReportTypeSsrc, ssrc_id, direction);
 
   // Clear out stats from previous GatherStats calls if any.
   // This is required since the report will be returned for the new values.
@@ -715,13 +710,13 @@ StatsReport* StatsCollector::PrepareRemoteReport(
     const std::string& transport_id,
     TrackDirection direction) {
   const std::string ssrc_id = rtc::ToString<uint32>(ssrc);
-  StatsMap::iterator it = reports_.find(StatsId(
-      StatsReport::kStatsReportTypeRemoteSsrc, ssrc_id, direction));
+  StatsReport* report = reports_.Find(
+      StatsId(StatsReport::kStatsReportTypeRemoteSsrc, ssrc_id, direction));
 
   // Use the ID of the track that is currently mapped to the SSRC, if any.
   std::string track_id;
   if (!GetTrackIdBySsrc(ssrc, &track_id, direction)) {
-    if (it == reports_.end()) {
+    if (!report) {
       // The ssrc is not used by any track or existing report, return NULL
       // in such case to indicate no report is prepared for the ssrc.
       return NULL;
@@ -729,12 +724,12 @@ StatsReport* StatsCollector::PrepareRemoteReport(
 
     // The ssrc is not used by any existing track. Keeps the old track id
     // since we want to report the stats for inactive ssrc.
-    ExtractValueFromReport(it->second,
+    ExtractValueFromReport(*report,
                            StatsReport::kStatsValueNameTrackId,
                            &track_id);
   }
 
-  StatsReport* report = GetOrCreateReport(
+  report = GetOrCreateReport(
       StatsReport::kStatsReportTypeRemoteSsrc, ssrc_id, direction);
 
   // Clear out stats from previous GatherStats calls if any.
@@ -778,18 +773,17 @@ std::string StatsCollector::AddOneCertificateReport(
   rtc::Base64::EncodeFromArray(
       der_buffer.data(), der_buffer.length(), &der_base64);
 
-  StatsReport report;
-  report.type = StatsReport::kStatsReportTypeCertificate;
-  report.id = StatsId(report.type, fingerprint);
-  report.timestamp = stats_gathering_started_;
-  report.AddValue(StatsReport::kStatsValueNameFingerprint, fingerprint);
-  report.AddValue(StatsReport::kStatsValueNameFingerprintAlgorithm,
-                  digest_algorithm);
-  report.AddValue(StatsReport::kStatsValueNameDer, der_base64);
+  StatsReport* report = reports_.ReplaceOrAddNew(
+      StatsId(StatsReport::kStatsReportTypeCertificate, fingerprint));
+  report->type = StatsReport::kStatsReportTypeCertificate;
+  report->timestamp = stats_gathering_started_;
+  report->AddValue(StatsReport::kStatsValueNameFingerprint, fingerprint);
+  report->AddValue(StatsReport::kStatsValueNameFingerprintAlgorithm,
+                   digest_algorithm);
+  report->AddValue(StatsReport::kStatsValueNameDer, der_base64);
   if (!issuer_id.empty())
-    report.AddValue(StatsReport::kStatsValueNameIssuerId, issuer_id);
-  reports_[report.id] = report;
-  return report.id;
+    report->AddValue(StatsReport::kStatsValueNameIssuerId, issuer_id);
+  return report->id;
 }
 
 std::string StatsCollector::AddCertificateReports(
@@ -819,15 +813,13 @@ std::string StatsCollector::AddCertificateReports(
 
 void StatsCollector::ExtractSessionInfo() {
   // Extract information from the base session.
-  StatsReport report;
-  report.id = StatsId(StatsReport::kStatsReportTypeSession, session_->id());
-  report.type = StatsReport::kStatsReportTypeSession;
-  report.timestamp = stats_gathering_started_;
-  report.values.clear();
-  report.AddBoolean(StatsReport::kStatsValueNameInitiator,
-                    session_->initiator());
-
-  reports_[report.id] = report;
+  StatsReport* report = reports_.ReplaceOrAddNew(
+      StatsId(StatsReport::kStatsReportTypeSession, session_->id()));
+  report->type = StatsReport::kStatsReportTypeSession;
+  report->timestamp = stats_gathering_started_;
+  report->values.clear();
+  report->AddBoolean(StatsReport::kStatsValueNameInitiator,
+                     session_->initiator());
 
   cricket::SessionStats stats;
   if (session_->GetStats(&stats)) {
@@ -865,61 +857,58 @@ void StatsCollector::ExtractSessionInfo() {
                = transport_iter->second.channel_stats.begin();
            channel_iter != transport_iter->second.channel_stats.end();
            ++channel_iter) {
-        StatsReport channel_report;
         std::ostringstream ostc;
         ostc << "Channel-" << transport_iter->second.content_name
              << "-" << channel_iter->component;
-        channel_report.id = ostc.str();
-        channel_report.type = StatsReport::kStatsReportTypeComponent;
-        channel_report.timestamp = stats_gathering_started_;
-        channel_report.AddValue(StatsReport::kStatsValueNameComponent,
-                                channel_iter->component);
+        StatsReport* channel_report = reports_.ReplaceOrAddNew(ostc.str());
+        channel_report->type = StatsReport::kStatsReportTypeComponent;
+        channel_report->timestamp = stats_gathering_started_;
+        channel_report->AddValue(StatsReport::kStatsValueNameComponent,
+                                 channel_iter->component);
         if (!local_cert_report_id.empty())
-          channel_report.AddValue(
+          channel_report->AddValue(
               StatsReport::kStatsValueNameLocalCertificateId,
               local_cert_report_id);
         if (!remote_cert_report_id.empty())
-          channel_report.AddValue(
+          channel_report->AddValue(
               StatsReport::kStatsValueNameRemoteCertificateId,
               remote_cert_report_id);
-        reports_[channel_report.id] = channel_report;
         for (size_t i = 0;
              i < channel_iter->connection_infos.size();
              ++i) {
-          StatsReport report;
-          const cricket::ConnectionInfo& info
-              = channel_iter->connection_infos[i];
           std::ostringstream ost;
           ost << "Conn-" << transport_iter->first << "-"
               << channel_iter->component << "-" << i;
-          report.id = ost.str();
-          report.type = StatsReport::kStatsReportTypeCandidatePair;
-          report.timestamp = stats_gathering_started_;
+          StatsReport* report = reports_.ReplaceOrAddNew(ost.str());
+          report->type = StatsReport::kStatsReportTypeCandidatePair;
+          report->timestamp = stats_gathering_started_;
           // Link from connection to its containing channel.
-          report.AddValue(StatsReport::kStatsValueNameChannelId,
-                          channel_report.id);
-          report.AddValue(StatsReport::kStatsValueNameBytesSent,
-                          info.sent_total_bytes);
-          report.AddValue(StatsReport::kStatsValueNameBytesReceived,
-                          info.recv_total_bytes);
-          report.AddBoolean(StatsReport::kStatsValueNameWritable,
-                            info.writable);
-          report.AddBoolean(StatsReport::kStatsValueNameReadable,
-                            info.readable);
-          report.AddBoolean(StatsReport::kStatsValueNameActiveConnection,
-                            info.best_connection);
-          report.AddValue(StatsReport::kStatsValueNameLocalAddress,
-                          info.local_candidate.address().ToString());
-          report.AddValue(StatsReport::kStatsValueNameRemoteAddress,
-                          info.remote_candidate.address().ToString());
-          report.AddValue(StatsReport::kStatsValueNameRtt, info.rtt);
-          report.AddValue(StatsReport::kStatsValueNameTransportType,
-                          info.local_candidate.protocol());
-          report.AddValue(StatsReport::kStatsValueNameLocalCandidateType,
-                          info.local_candidate.type());
-          report.AddValue(StatsReport::kStatsValueNameRemoteCandidateType,
-                          info.remote_candidate.type());
-          reports_[report.id] = report;
+          report->AddValue(StatsReport::kStatsValueNameChannelId,
+                           channel_report->id);
+
+          const cricket::ConnectionInfo& info =
+              channel_iter->connection_infos[i];
+          report->AddValue(StatsReport::kStatsValueNameBytesSent,
+                           info.sent_total_bytes);
+          report->AddValue(StatsReport::kStatsValueNameBytesReceived,
+                           info.recv_total_bytes);
+          report->AddBoolean(StatsReport::kStatsValueNameWritable,
+                             info.writable);
+          report->AddBoolean(StatsReport::kStatsValueNameReadable,
+                             info.readable);
+          report->AddBoolean(StatsReport::kStatsValueNameActiveConnection,
+                             info.best_connection);
+          report->AddValue(StatsReport::kStatsValueNameLocalAddress,
+                           info.local_candidate.address().ToString());
+          report->AddValue(StatsReport::kStatsValueNameRemoteAddress,
+                           info.remote_candidate.address().ToString());
+          report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt);
+          report->AddValue(StatsReport::kStatsValueNameTransportType,
+                           info.local_candidate.protocol());
+          report->AddValue(StatsReport::kStatsValueNameLocalCandidateType,
+                           info.local_candidate.type());
+          report->AddValue(StatsReport::kStatsValueNameRemoteCandidateType,
+                           info.remote_candidate.type());
         }
       }
     }
@@ -976,7 +965,8 @@ void StatsCollector::ExtractVideoInfo(
   if (video_info.bw_estimations.size() != 1) {
     LOG(LS_ERROR) << "BWEs count: " << video_info.bw_estimations.size();
   } else {
-    StatsReport* report = &reports_[StatsReport::kStatsReportVideoBweId];
+    StatsReport* report =
+        reports_.FindOrAddNew(StatsReport::kStatsReportVideoBweId);
     ExtractStats(
         video_info.bw_estimations[0], stats_gathering_started_, level, report);
   }
@@ -987,13 +977,7 @@ StatsReport* StatsCollector::GetReport(const std::string& type,
                                        TrackDirection direction) {
   ASSERT(type == StatsReport::kStatsReportTypeSsrc ||
          type == StatsReport::kStatsReportTypeRemoteSsrc);
-  std::string statsid = StatsId(type, id, direction);
-  StatsReport* report = NULL;
-  std::map<std::string, StatsReport>::iterator it = reports_.find(statsid);
-  if (it != reports_.end())
-    report = &(it->second);
-
-  return report;
+  return reports_.Find(StatsId(type, id, direction));
 }
 
 StatsReport* StatsCollector::GetOrCreateReport(const std::string& type,
@@ -1004,8 +988,8 @@ StatsReport* StatsCollector::GetOrCreateReport(const std::string& type,
   StatsReport* report = GetReport(type, id, direction);
   if (report == NULL) {
     std::string statsid = StatsId(type, id, direction);
-    report = &reports_[statsid];  // Create new element.
-    report->id = statsid;
+    report = reports_.FindOrAddNew(statsid);
+    ASSERT(report->id == statsid);
     report->type = type;
   }
 
diff --git a/app/webrtc/statscollector.h b/app/webrtc/statscollector.h
index a039813..db2d788 100644
--- a/app/webrtc/statscollector.h
+++ b/app/webrtc/statscollector.h
@@ -72,7 +72,11 @@ class StatsCollector {
   // be called before this function to get the most recent stats. |selector| is
   // a track label or empty string. The most recent reports are stored in
   // |reports|.
-  bool GetStats(MediaStreamTrackInterface* track,
+  // TODO(tommi): Change this contract to accept a callback object instead
+  // of filling in |reports|.  As is, there's a requirement that the caller
+  // uses |reports| immediately without allowing any async activity on
+  // the thread (message handling etc) and then discard the results.
+  void GetStats(MediaStreamTrackInterface* track,
                 StatsReports* reports);
 
   // Prepare an SSRC report for the given ssrc. Used internally
@@ -121,7 +125,7 @@ class StatsCollector {
                         TrackDirection direction);
 
   // A map from the report id to the report.
-  std::map<std::string, StatsReport> reports_;
+  StatsSet reports_;
   // Raw pointer to the session the statistics are gathered from.
   WebRtcSession* const session_;
   double stats_gathering_started_;
diff --git a/app/webrtc/statscollector_unittest.cc b/app/webrtc/statscollector_unittest.cc
index f264545..2f4c130 100644
--- a/app/webrtc/statscollector_unittest.cc
+++ b/app/webrtc/statscollector_unittest.cc
@@ -168,10 +168,10 @@ std::string ExtractStatsValue(const std::string& type,
     return kNoReports;
   }
   for (size_t i = 0; i < reports.size(); ++i) {
-    if (reports[i].type != type)
+    if (reports[i]->type != type)
       continue;
     std::string ret;
-    if (GetValue(&reports[i], name, &ret)) {
+    if (GetValue(reports[i], name, &ret)) {
       return ret;
     }
   }
@@ -184,10 +184,10 @@ std::string ExtractStatsValue(const std::string& type,
 const StatsReport* FindNthReportByType(
     const StatsReports& reports, const std::string& type, int n) {
   for (size_t i = 0; i < reports.size(); ++i) {
-    if (reports[i].type == type) {
+    if (reports[i]->type == type) {
       n--;
       if (n == 0)
-        return &reports[i];
+        return reports[i];
     }
   }
   return NULL;
@@ -196,8 +196,8 @@ const StatsReport* FindNthReportByType(
 const StatsReport* FindReportById(const StatsReports& reports,
                                   const std::string& id) {
   for (size_t i = 0; i < reports.size(); ++i) {
-    if (reports[i].id == id) {
-      return &reports[i];
+    if (reports[i]->id == id) {
+      return reports[i];
     }
   }
   return NULL;
@@ -433,7 +433,7 @@ void InitVoiceReceiverInfo(cricket::VoiceReceiverInfo* voice_receiver_info) {
 class StatsCollectorTest : public testing::Test {
  protected:
   StatsCollectorTest()
-    : media_engine_(new cricket::FakeMediaEngine),
+    : media_engine_(new cricket::FakeMediaEngine()),
       channel_manager_(
           new cricket::ChannelManager(media_engine_,
                                       new cricket::FakeDeviceManager(),
@@ -443,6 +443,8 @@ class StatsCollectorTest : public testing::Test {
     EXPECT_CALL(session_, GetStats(_)).WillRepeatedly(Return(false));
   }
 
+  ~StatsCollectorTest() {}
+
   // This creates a standard setup with a transport called "trspname"
   // having one transport channel
   // and the specified virtual connection name.
@@ -786,7 +788,7 @@ TEST_F(StatsCollectorTest, TrackObjectExistsWithoutUpdateStats) {
   stats.GetStats(NULL, &reports);
   EXPECT_EQ((size_t)1, reports.size());
   EXPECT_EQ(std::string(StatsReport::kStatsReportTypeTrack),
-            reports[0].type);
+            reports[0]->type);
 
   std::string trackValue =
       ExtractStatsValue(StatsReport::kStatsReportTypeTrack,
@@ -832,6 +834,7 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
   EXPECT_TRUE(track_report);
 
   // Get report for the specific |track|.
+  reports.clear();
   stats.GetStats(track_, &reports);
   // |reports| should contain at least one session report, one track report,
   // and one ssrc report.
@@ -1396,6 +1399,7 @@ TEST_F(StatsCollectorTest, LocalAndRemoteTracksWithSameSsrc) {
   VerifyVoiceSenderInfoReport(track_report, voice_sender_info);
 
   // Get stats for the remote track.
+  reports.clear();
   stats.GetStats(remote_track.get(), &reports);
   track_report = FindNthReportByType(reports,
                                      StatsReport::kStatsReportTypeSsrc, 1);
@@ -1452,6 +1456,7 @@ TEST_F(StatsCollectorTest, TwoLocalTracksWithSameSsrc) {
   cricket::VoiceSenderInfo new_voice_sender_info;
   InitVoiceSenderInfo(&new_voice_sender_info);
   cricket::VoiceMediaInfo new_stats_read;
+  reports.clear();
   SetupAndVerifyAudioTrackStats(
       new_audio_track.get(), stream_.get(), &stats, &voice_channel, kVcName,
       media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports);
diff --git a/app/webrtc/statstypes.h b/app/webrtc/statstypes.h
index 2b1317a..8eae1ad 100644
--- a/app/webrtc/statstypes.h
+++ b/app/webrtc/statstypes.h
@@ -31,6 +31,7 @@
 #ifndef TALK_APP_WEBRTC_STATSTYPES_H_
 #define TALK_APP_WEBRTC_STATSTYPES_H_
 
+#include <set>
 #include <string>
 #include <vector>
 
@@ -39,14 +40,49 @@
 
 namespace webrtc {
 
+// TODO(tommi): Move all the implementation that's in this file and
+// statscollector.cc related to these types, into a new, statstypes.cc file.
+
 class StatsReport {
  public:
-  StatsReport() : timestamp(0) { }
+  // TODO(tommi): Remove this ctor.
+  StatsReport() : timestamp(0) {}
+
+  // TODO(tommi): Make protected and disallow copy completely once not needed.
+  StatsReport(const StatsReport& src)
+    : id(src.id),
+      type(src.type),
+      timestamp(src.timestamp),
+      values(src.values) {}
+
+  // TODO(tommi): Make this copy constructor protected.
+  StatsReport& operator=(const StatsReport& src) {
+    ASSERT(id == src.id);
+    type = src.type;
+    timestamp = src.timestamp;
+    values = src.values;
+    return *this;
+  }
+
+  // Constructor is protected to force use of StatsSet.
+  // TODO(tommi): Make this ctor protected.
+  explicit StatsReport(const std::string& id) : id(id), timestamp(0) {}
+
+  // Operators provided for STL container/algorithm support.
+  bool operator<(const StatsReport& other) const { return id < other.id; }
+  bool operator==(const StatsReport& other) const { return id == other.id; }
+  // Special support for being able to use std::find on a container
+  // without requiring a new StatsReport instance.
+  bool operator==(const std::string& other_id) const { return id == other_id; }
 
   // TODO(tommi): Change this to be an enum type that holds all the
   // kStatsValueName constants.
   typedef const char* StatsValueName;
 
+  // The unique identifier for this object.
+  // This is used as a key for this report in ordered containers,
+  // so it must never be changed.
+  // TODO(tommi): Make this member variable const.
   std::string id;  // See below for contents.
   std::string type;  // See below for contents.
 
@@ -239,7 +275,70 @@ class StatsReport {
   static const char kStatsValueNameDecodingPLCCNG[];
 };
 
-typedef std::vector<StatsReport> StatsReports;
+// This class is provided for the cases where we need to keep
+// snapshots of reports around.  This is an edge case.
+// TODO(tommi): Move into the private section of StatsSet.
+class StatsReportCopyable : public StatsReport {
+ public:
+  StatsReportCopyable(const std::string& id) : StatsReport(id) {}
+  explicit StatsReportCopyable(const StatsReport& src)
+      : StatsReport(src) {}
+
+  using StatsReport::operator=;
+};
+
+// Typedef for an array of const StatsReport pointers.
+// Ownership of the pointers held by this implementation is assumed to lie
+// elsewhere and lifetime guarantees are made by the implementation that uses
+// this type.  In the StatsCollector, object ownership lies with the StatsSet
+// class.
+typedef std::vector<const StatsReport*> StatsReports;
+
+// A map from the report id to the report.
+// This class wraps an STL container and provides a limited set of
+// functionality in order to keep things simple.
+// TODO(tommi): Use a thread checker here (currently not in libjingle).
+class StatsSet {
+ public:
+  StatsSet() {}
+  ~StatsSet() {}
+
+  typedef std::set<StatsReportCopyable> Container;
+  typedef Container::iterator iterator;
+  typedef Container::const_iterator const_iterator;
+
+  const_iterator begin() const { return list_.begin(); }
+  const_iterator end() const { return list_.end(); }
+
+  // Creates a new report object with |id| that does not already
+  // exist in the list of reports.
+  StatsReport* InsertNew(const std::string& id) {
+    ASSERT(Find(id) == NULL);
+    const StatsReport* ret = &(*list_.insert(StatsReportCopyable(id)).first);
+    return const_cast<StatsReport*>(ret);
+  }
+
+  StatsReport* FindOrAddNew(const std::string& id) {
+    StatsReport* ret = Find(id);
+    return ret ? ret : InsertNew(id);
+  }
+
+  StatsReport* ReplaceOrAddNew(const std::string& id) {
+    list_.erase(id);
+    return InsertNew(id);
+  }
+
+  // Looks for a report with the given |id|.  If one is not found, NULL
+  // will be returned.
+  StatsReport* Find(const std::string& id) {
+    const_iterator it = std::find(begin(), end(), id);
+    return it == end() ? NULL :
+        const_cast<StatsReport*>(static_cast<const StatsReport*>(&(*it)));
+  }
+
+ private:
+  Container list_;
+};
 
 }  // namespace webrtc
 
diff --git a/app/webrtc/test/mockpeerconnectionobservers.h b/app/webrtc/test/mockpeerconnectionobservers.h
index 884c7a8..174b80b 100644
--- a/app/webrtc/test/mockpeerconnectionobservers.h
+++ b/app/webrtc/test/mockpeerconnectionobservers.h
@@ -120,9 +120,13 @@ class MockStatsObserver : public webrtc::StatsObserver {
   MockStatsObserver()
       : called_(false) {}
   virtual ~MockStatsObserver() {}
-  virtual void OnComplete(const std::vector<webrtc::StatsReport>& reports) {
+  virtual void OnComplete(const StatsReports& reports) {
     called_ = true;
-    reports_ = reports;
+    reports_.clear();
+    reports_.reserve(reports.size());
+    StatsReports::const_iterator it;
+    for (it = reports.begin(); it != reports.end(); ++it)
+      reports_.push_back(StatsReportCopyable(*(*it)));
   }
 
   bool called() const { return called_; }
@@ -148,7 +152,7 @@ class MockStatsObserver : public webrtc::StatsObserver {
   }
 
  private:
-  int GetSsrcStatsValue(const std::string name) {
+  int GetSsrcStatsValue(StatsReport::StatsValueName name) {
     if (reports_.empty()) {
       return 0;
     }
@@ -167,7 +171,7 @@ class MockStatsObserver : public webrtc::StatsObserver {
   }
 
   bool called_;
-  std::vector<webrtc::StatsReport> reports_;
+  std::vector<StatsReportCopyable> reports_;
 };
 
 }  // namespace webrtc
diff --git a/app/webrtc/webrtcsession.cc b/app/webrtc/webrtcsession.cc
index 1f14ff7..6d57afb 100644
--- a/app/webrtc/webrtcsession.cc
+++ b/app/webrtc/webrtcsession.cc
@@ -579,18 +579,6 @@ bool WebRtcSession::Initialize(
       MediaConstraintsInterface::kPayloadPadding,
       &video_options_.use_payload_padding);
 
-  // Find improved wifi bwe constraint.
-  if (FindConstraint(
-        constraints,
-        MediaConstraintsInterface::kImprovedWifiBwe,
-        &value,
-        NULL)) {
-    video_options_.use_improved_wifi_bandwidth_estimator.Set(value);
-  } else {
-    // Enable by default if the constraint is not set.
-    video_options_.use_improved_wifi_bandwidth_estimator.Set(true);
-  }
-
   SetOptionFromOptionalConstraint(constraints,
       MediaConstraintsInterface::kNumUnsignalledRecvStreams,
       &video_options_.unsignalled_recv_stream_limit);
@@ -622,10 +610,6 @@ bool WebRtcSession::Initialize(
         cricket::VideoOptions::HIGH);
   }
 
-  SetOptionFromOptionalConstraint(constraints,
-      MediaConstraintsInterface::kOpusFec,
-      &audio_options_.opus_fec);
-
   const cricket::VideoCodec default_codec(
       JsepSessionDescription::kDefaultVideoCodecId,
       JsepSessionDescription::kDefaultVideoCodecName,
diff --git a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
index ef97cda..213da7b 100644
--- a/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
+++ b/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
@@ -200,19 +200,49 @@ public class AppRTCDemoActivity extends Activity
   private void updateHUD(StatsReport[] reports) {
     StringBuilder builder = new StringBuilder();
     for (StatsReport report : reports) {
-      if (!report.id.equals("bweforvideo")) {
+      // bweforvideo to show statistics for video Bandwidth Estimation,
+      // which is global per-session.
+      if (report.id.equals("bweforvideo")) {
+        for (StatsReport.Value value : report.values) {
+          String name = value.name.replace("goog", "")
+              .replace("Available", "").replace("Bandwidth", "")
+              .replace("Bitrate", "").replace("Enc", "");
+
+          builder.append(name).append("=").append(value.value)
+              .append(" ");
+        }
+        builder.append("\n");
+      } else if (report.type.equals("googCandidatePair")) {
+        String activeConnectionStats = getActiveConnectionStats(report);
+        if (activeConnectionStats == null) {
+          continue;
+        }
+        builder.append(activeConnectionStats);
+      } else {
         continue;
       }
-      for (StatsReport.Value value : report.values) {
-        String name = value.name.replace("goog", "").replace("Available", "")
-            .replace("Bandwidth", "").replace("Bitrate", "").replace("Enc", "");
-        builder.append(name).append("=").append(value.value).append(" ");
-      }
       builder.append("\n");
     }
     hudView.setText(builder.toString() + hudView.getText());
   }
 
+  // Return the active connection stats else return null
+  private String getActiveConnectionStats(StatsReport report) {
+    StringBuilder activeConnectionbuilder = new StringBuilder();
+    // googCandidatePair to show information about the active
+    // connection.
+    for (StatsReport.Value value : report.values) {
+      if (value.name.equals("googActiveConnection")
+          && value.value.equals("false")) {
+        return null;
+      }
+      String name = value.name.replace("goog", "");
+      activeConnectionbuilder.append(name).append("=")
+          .append(value.value).append("\n");
+    }
+    return activeConnectionbuilder.toString();
+  }
+
   @Override
   public void onPause() {
     super.onPause();
diff --git a/examples/call/callclient.cc b/examples/call/callclient.cc
index 188f6c4..2c8a6bc 100644
--- a/examples/call/callclient.cc
+++ b/examples/call/callclient.cc
@@ -1285,8 +1285,9 @@ void CallClient::OnMucStatusUpdate(const buzz::Jid& jid,
     return;
   }
 
-  if (!status.available()) {
-    // Remove them from the room.
+  if (status.available()) {
+    muc->members()[status.jid().resource()] = status;
+  } else {
     muc->members().erase(status.jid().resource());
   }
 }
diff --git a/libjingle.gyp b/libjingle.gyp
index 6064277..f4f9bf9 100755
--- a/libjingle.gyp
+++ b/libjingle.gyp
@@ -531,6 +531,7 @@
         'media/webrtc/webrtcexport.h',
         'media/webrtc/webrtcmediaengine.cc',
         'media/webrtc/webrtcmediaengine.h',
+        'media/webrtc/webrtcmediaengine.cc',
         'media/webrtc/webrtcpassthroughrender.cc',
         'media/webrtc/webrtcpassthroughrender.h',
         'media/webrtc/webrtctexturevideoframe.cc',
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 3759acc..62d6b61 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -182,7 +182,6 @@ struct AudioOptions {
     recording_sample_rate.SetFrom(change.recording_sample_rate);
     playout_sample_rate.SetFrom(change.playout_sample_rate);
     dscp.SetFrom(change.dscp);
-    opus_fec.SetFrom(change.opus_fec);
   }
 
   bool operator==(const AudioOptions& o) const {
@@ -208,8 +207,7 @@ struct AudioOptions {
         rx_agc_limiter == o.rx_agc_limiter &&
         recording_sample_rate == o.recording_sample_rate &&
         playout_sample_rate == o.playout_sample_rate &&
-        dscp == o.dscp &&
-        opus_fec == o.opus_fec;
+        dscp == o.dscp;
   }
 
   std::string ToString() const {
@@ -240,7 +238,6 @@ struct AudioOptions {
     ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
     ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
     ost << ToStringIfSet("dscp", dscp);
-    ost << ToStringIfSet("opus_fec", opus_fec);
     ost << "}";
     return ost.str();
   }
@@ -278,8 +275,6 @@ struct AudioOptions {
   Settable<uint32> playout_sample_rate;
   // Set DSCP value for packet sent from audio channel.
   Settable<bool> dscp;
-  // Set Opus FEC
-  Settable<bool> opus_fec;
 };
 
 // Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
@@ -334,8 +329,6 @@ struct VideoOptions {
     unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
     use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
     screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
-    use_improved_wifi_bandwidth_estimator.SetFrom(
-        change.use_improved_wifi_bandwidth_estimator);
     use_payload_padding.SetFrom(change.use_payload_padding);
   }
 
@@ -372,8 +365,6 @@ struct VideoOptions {
         unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
         use_simulcast_adapter == o.use_simulcast_adapter &&
         screencast_min_bitrate == o.screencast_min_bitrate &&
-        use_improved_wifi_bandwidth_estimator ==
-            o.use_improved_wifi_bandwidth_estimator &&
         use_payload_padding == o.use_payload_padding;
   }
 
@@ -414,8 +405,6 @@ struct VideoOptions {
                          unsignalled_recv_stream_limit);
     ost << ToStringIfSet("use simulcast adapter", use_simulcast_adapter);
     ost << ToStringIfSet("screencast min bitrate", screencast_min_bitrate);
-    ost << ToStringIfSet("improved wifi bwe",
-                         use_improved_wifi_bandwidth_estimator);
     ost << ToStringIfSet("payload padding", use_payload_padding);
     ost << "}";
     return ost.str();
@@ -488,8 +477,6 @@ struct VideoOptions {
   Settable<bool> use_simulcast_adapter;
   // Force screencast to use a minimum bitrate
   Settable<int> screencast_min_bitrate;
-  // Enable improved bandwidth estiamtor on wifi.
-  Settable<bool> use_improved_wifi_bandwidth_estimator;
   // Enable payload padding.
   Settable<bool> use_payload_padding;
 };
diff --git a/media/base/mediaengine.cc b/media/base/mediaengine.cc
index 289f229..d38a312 100644
--- a/media/base/mediaengine.cc
+++ b/media/base/mediaengine.cc
@@ -27,90 +27,40 @@
 
 #include "talk/media/base/mediaengine.h"
 
-namespace cricket {
-const int MediaEngineInterface::kDefaultAudioDelayOffset = 0;
-}
-
 #if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
 
-#if defined(HAVE_LINPHONE)
-#include "talk/media/other/linphonemediaengine.h"
-#endif  // HAVE_LINPHONE
-#if defined(HAVE_WEBRTC_VOICE)
-#include "talk/media/webrtc/webrtcvoiceengine.h"
-#endif  // HAVE_WEBRTC_VOICE
-#if defined(HAVE_WEBRTC_VIDEO)
-#include "talk/media/webrtc/webrtcvideoengine.h"
-#endif  // HAVE_WEBRTC_VIDEO
-#if defined(HAVE_LMI)
-#include "talk/media/base/hybridvideoengine.h"
-#include "talk/media/lmi/lmimediaengine.h"
-#endif  // HAVE_LMI
+#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO)
+#include "talk/media/webrtc/webrtcmediaengine.h"
+#endif  // HAVE_WEBRTC_VOICE && HAVE_WEBRTC_VIDEO
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif  // HAVE_CONFIG
 
 namespace cricket {
-#if defined(HAVE_WEBRTC_VOICE)
-#define AUDIO_ENG_NAME WebRtcVoiceEngine
-#else
-#define AUDIO_ENG_NAME NullVoiceEngine
-#endif
-
-#if defined(HAVE_WEBRTC_VIDEO)
-#if !defined(HAVE_LMI)
-template<>
-CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine>::
-    CompositeMediaEngine() {
-  video_.SetVoiceEngine(&voice_);
-}
-#define VIDEO_ENG_NAME WebRtcVideoEngine
-#else
-// If we have both WebRtcVideoEngine and LmiVideoEngine, enable dual-stack.
-// This small class here allows us to hook the WebRtcVideoChannel up to
-// the capturer owned by the LMI engine, without infecting the rest of the
-// HybridVideoEngine classes with this abstraction violation.
-class WebRtcLmiHybridVideoEngine
-    : public HybridVideoEngine<WebRtcVideoEngine, LmiVideoEngine> {
- public:
-  void SetVoiceEngine(WebRtcVoiceEngine* engine) {
-    video1_.SetVoiceEngine(engine);
-  }
-};
-template<>
-CompositeMediaEngine<WebRtcVoiceEngine, WebRtcLmiHybridVideoEngine>::
-    CompositeMediaEngine() {
-  video_.SetVoiceEngine(&voice_);
-}
-#define VIDEO_ENG_NAME WebRtcLmiHybridVideoEngine
-#endif
-#elif defined(HAVE_LMI)
-#define VIDEO_ENG_NAME LmiVideoEngine
-#else
-#define VIDEO_ENG_NAME NullVideoEngine
-#endif
 
 MediaEngineFactory::MediaEngineCreateFunction
     MediaEngineFactory::create_function_ = NULL;
+
 MediaEngineFactory::MediaEngineCreateFunction
     MediaEngineFactory::SetCreateFunction(MediaEngineCreateFunction function) {
   MediaEngineCreateFunction old_function = create_function_;
   create_function_ = function;
   return old_function;
-};
+}
 
+// TODO(pthatcher): Remove this method and require all the users of
+// media engines to choose one explictly, or at least remove the
+// default behavior and require calling SetCreateFunction explictly.
 MediaEngineInterface* MediaEngineFactory::Create() {
   if (create_function_) {
     return create_function_();
-  } else {
-#if defined(HAVE_LINPHONE)
-    return new LinphoneMediaEngine("", "");
-#elif defined(AUDIO_ENG_NAME) && defined(VIDEO_ENG_NAME)
-    return new CompositeMediaEngine<AUDIO_ENG_NAME, VIDEO_ENG_NAME>();
-#else
-    return new NullMediaEngine();
-#endif
   }
+#if defined(HAVE_WEBRTC_VOICE) && defined(HAVE_WEBRTC_VIDEO)
+  return WebRtcMediaEngineFactory::Create();
+#else
+  return new NullMediaEngine();
+#endif  // HAVE_WEBRTC_VIDEO && HAVE_WEBRTC_VOICE
 }
 
 };  // namespace cricket
diff --git a/media/webrtc/fakewebrtcvideoengine.h b/media/webrtc/fakewebrtcvideoengine.h
index 3a9e411..5cba380 100644
--- a/media/webrtc/fakewebrtcvideoengine.h
+++ b/media/webrtc/fakewebrtcvideoengine.h
@@ -908,6 +908,9 @@ class FakeWebRtcVideoEngine
   WEBRTC_STUB(SetMTU, (int, unsigned int));
   WEBRTC_STUB(ReceivedBWEPacket, (const int, int64_t, int,
       const webrtc::RTPHeader&));
+  virtual bool SetBandwidthEstimationConfig(int, const webrtc::Config&) {
+    return true;
+  }
 
   // webrtc::ViERender
   WEBRTC_STUB(RegisterVideoRenderModule, (webrtc::VideoRender&));
diff --git a/media/webrtc/webrtcmediaengine.cc b/media/webrtc/webrtcmediaengine.cc
index 445564c..252b4e6 100644
--- a/media/webrtc/webrtcmediaengine.cc
+++ b/media/webrtc/webrtcmediaengine.cc
@@ -25,8 +25,53 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+#if defined(LIBPEERCONNECTION_LIB) || \
+    defined(LIBPEERCONNECTION_IMPLEMENTATION)
+
 #include "talk/media/webrtc/webrtcmediaengine.h"
+#include "talk/media/webrtc/webrtcvideoengine.h"
+#ifdef WEBRTC_CHROMIUM_BUILD
+#include "talk/media/webrtc/webrtcvideoengine2.h"
+#endif
+#include "talk/media/webrtc/webrtcvoiceengine.h"
+#ifdef WEBRTC_CHROMIUM_BUILD
 #include "webrtc/system_wrappers/interface/field_trial.h"
+#endif
+
+namespace cricket {
+
+class WebRtcMediaEngine :
+      public CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine> {
+ public:
+  WebRtcMediaEngine() {}
+  WebRtcMediaEngine(webrtc::AudioDeviceModule* adm,
+      webrtc::AudioDeviceModule* adm_sc,
+      WebRtcVideoEncoderFactory* encoder_factory,
+      WebRtcVideoDecoderFactory* decoder_factory) {
+    voice_.SetAudioDeviceModule(adm, adm_sc);
+    video_.SetVoiceEngine(&voice_);
+    video_.EnableTimedRender();
+    video_.SetExternalEncoderFactory(encoder_factory);
+    video_.SetExternalDecoderFactory(decoder_factory);
+  }
+};
+
+#ifdef WEBRTC_CHROMIUM_BUILD
+class WebRtcMediaEngine2 :
+      public CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine2> {
+ public:
+  WebRtcMediaEngine2(webrtc::AudioDeviceModule* adm,
+                     webrtc::AudioDeviceModule* adm_sc,
+                     WebRtcVideoEncoderFactory* encoder_factory,
+                     WebRtcVideoDecoderFactory* decoder_factory) {
+    voice_.SetAudioDeviceModule(adm, adm_sc);
+    video_.SetVoiceEngine(&voice_);
+    video_.EnableTimedRender();
+  }
+};
+#endif  // WEBRTC_CHROMIUM_BUILD
+
+}  // namespace cricket
 
 WRME_EXPORT
 cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
@@ -40,19 +85,35 @@ cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
         adm, adm_sc, encoder_factory, decoder_factory);
   }
 #endif // WEBRTC_CHROMIUM_BUILD
+  // This is just to get a diff to run pulse.
   return new cricket::WebRtcMediaEngine(
       adm, adm_sc, encoder_factory, decoder_factory);
 }
 
 WRME_EXPORT
 void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
-#ifdef WEBRTC_CHROMIUM_BUILD
-  if (webrtc::field_trial::FindFullName("WebRTC-NewVideoAPI") == "Enabled") {
-    delete static_cast<cricket::WebRtcMediaEngine2*>(media_engine);
-  } else {
-#endif // WEBRTC_CHROMIUM_BUILD
-    delete static_cast<cricket::WebRtcMediaEngine*>(media_engine);
-#ifdef WEBRTC_CHROMIUM_BUILD
-  }
-#endif // WEBRTC_CHROMIUM_BUILD
+  delete media_engine;
+}
+
+namespace cricket {
+
+// Used by ChannelManager when no media engine is passed in to it
+// explicitly (acts as a default).
+MediaEngineInterface* WebRtcMediaEngineFactory::Create() {
+  return new cricket::WebRtcMediaEngine();
 }
+
+// Used by PeerConnectionFactory to create a media engine passed into
+// ChannelManager.
+MediaEngineInterface* WebRtcMediaEngineFactory::Create(
+    webrtc::AudioDeviceModule* adm,
+    webrtc::AudioDeviceModule* adm_sc,
+    WebRtcVideoEncoderFactory* encoder_factory,
+    WebRtcVideoDecoderFactory* decoder_factory) {
+  return CreateWebRtcMediaEngine(adm, adm_sc, encoder_factory, decoder_factory);
+}
+
+}  // namespace cricket
+
+#endif  // defined(LIBPEERCONNECTION_LIB) ||
+        // defined(LIBPEERCONNECTION_IMPLEMENTATION)
diff --git a/media/webrtc/webrtcmediaengine.h b/media/webrtc/webrtcmediaengine.h
index 701417c..b906f5d 100644
--- a/media/webrtc/webrtcmediaengine.h
+++ b/media/webrtc/webrtcmediaengine.h
@@ -40,7 +40,6 @@ class WebRtcVideoDecoderFactory;
 class WebRtcVideoEncoderFactory;
 }
 
-
 #if !defined(LIBPEERCONNECTION_LIB) && \
     !defined(LIBPEERCONNECTION_IMPLEMENTATION)
 
@@ -53,19 +52,49 @@ cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
 WRME_EXPORT
 void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine);
 
+#endif  // !defined(LIBPEERCONNECTION_LIB) &&
+        // !defined(LIBPEERCONNECTION_IMPLEMENTATION)
+
 namespace cricket {
 
-class WebRtcMediaEngine : public cricket::MediaEngineInterface {
+class WebRtcMediaEngineFactory {
  public:
-  WebRtcMediaEngine(
+#if !defined(LIBPEERCONNECTION_LIB) && \
+    !defined(LIBPEERCONNECTION_IMPLEMENTATION)
+// A bare Create() isn't supported when using the delegating media
+// engine.
+#else
+  static MediaEngineInterface* Create();
+#endif  // !defined(LIBPEERCONNECTION_LIB) &&
+        // !defined(LIBPEERCONNECTION_IMPLEMENTATION)
+  static MediaEngineInterface* Create(
       webrtc::AudioDeviceModule* adm,
       webrtc::AudioDeviceModule* adm_sc,
-      cricket::WebRtcVideoEncoderFactory* encoder_factory,
-      cricket::WebRtcVideoDecoderFactory* decoder_factory)
+      WebRtcVideoEncoderFactory* encoder_factory,
+      WebRtcVideoDecoderFactory* decoder_factory);
+};
+
+}  // namespace cricket
+
+
+#if !defined(LIBPEERCONNECTION_LIB) && \
+    !defined(LIBPEERCONNECTION_IMPLEMENTATION)
+
+namespace cricket {
+
+// TODO(pthacther): Move this code into webrtcmediaengine.cc once
+// Chrome compiles it.  Right now it relies on only the .h file.
+class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
+ public:
+  DelegatingWebRtcMediaEngine(
+      webrtc::AudioDeviceModule* adm,
+      webrtc::AudioDeviceModule* adm_sc,
+      WebRtcVideoEncoderFactory* encoder_factory,
+      WebRtcVideoDecoderFactory* decoder_factory)
       : delegate_(CreateWebRtcMediaEngine(
           adm, adm_sc, encoder_factory, decoder_factory)) {
   }
-  virtual ~WebRtcMediaEngine() {
+  virtual ~DelegatingWebRtcMediaEngine() {
     DestroyWebRtcMediaEngine(delegate_);
   }
   virtual bool Init(rtc::Thread* worker_thread) OVERRIDE {
@@ -171,49 +200,16 @@ class WebRtcMediaEngine : public cricket::MediaEngineInterface {
   cricket::MediaEngineInterface* delegate_;
 };
 
-}  // namespace cricket
-#else
-
-#include "talk/media/webrtc/webrtcvideoengine.h"
-#ifdef WEBRTC_CHROMIUM_BUILD
-#include "talk/media/webrtc/webrtcvideoengine2.h"
-#endif
-#include "talk/media/webrtc/webrtcvoiceengine.h"
-
-namespace cricket {
-typedef CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine>
-        WebRtcCompositeMediaEngine;
-
-class WebRtcMediaEngine : public WebRtcCompositeMediaEngine {
- public:
-  WebRtcMediaEngine(webrtc::AudioDeviceModule* adm,
-      webrtc::AudioDeviceModule* adm_sc,
-      WebRtcVideoEncoderFactory* encoder_factory,
-      WebRtcVideoDecoderFactory* decoder_factory) {
-    voice_.SetAudioDeviceModule(adm, adm_sc);
-    video_.SetVoiceEngine(&voice_);
-    video_.EnableTimedRender();
-    video_.SetExternalEncoderFactory(encoder_factory);
-    video_.SetExternalDecoderFactory(decoder_factory);
-  }
-};
-
-#ifdef WEBRTC_CHROMIUM_BUILD
-typedef CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine2>
-        WebRtcCompositeMediaEngine2;
-
-class WebRtcMediaEngine2 : public WebRtcCompositeMediaEngine2 {
- public:
-  WebRtcMediaEngine2(webrtc::AudioDeviceModule* adm,
-                     webrtc::AudioDeviceModule* adm_sc,
-                     WebRtcVideoEncoderFactory* encoder_factory,
-                     WebRtcVideoDecoderFactory* decoder_factory) {
-    voice_.SetAudioDeviceModule(adm, adm_sc);
-    video_.SetVoiceEngine(&voice_);
-    video_.EnableTimedRender();
-  }
-};
-#endif
+// Used by PeerConnectionFactory to create a media engine passed into
+// ChannelManager.
+MediaEngineInterface* WebRtcMediaEngineFactory::Create(
+    webrtc::AudioDeviceModule* adm,
+    webrtc::AudioDeviceModule* adm_sc,
+    WebRtcVideoEncoderFactory* encoder_factory,
+    WebRtcVideoDecoderFactory* decoder_factory) {
+  return new cricket::DelegatingWebRtcMediaEngine(
+      adm, adm_sc, encoder_factory, decoder_factory);
+}
 
 }  // namespace cricket
 
diff --git a/media/webrtc/webrtcvideoengine.cc b/media/webrtc/webrtcvideoengine.cc
index 6ba48a7..83b1177 100644
--- a/media/webrtc/webrtcvideoengine.cc
+++ b/media/webrtc/webrtcvideoengine.cc
@@ -2958,11 +2958,6 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
     conference_mode_turned_off = true;
   }
 
-  bool improved_wifi_bwe_changed =
-      options.use_improved_wifi_bandwidth_estimator.IsSet() &&
-      options_.use_improved_wifi_bandwidth_estimator !=
-          options.use_improved_wifi_bandwidth_estimator;
-
 #ifdef USE_WEBRTC_DEV_BRANCH
   bool payload_padding_changed = options.use_payload_padding.IsSet() &&
       options_.use_payload_padding != options.use_payload_padding;
@@ -3070,18 +3065,6 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
       LOG(LS_WARNING) << "Cannot disable video suspension once it is enabled";
     }
   }
-  if (improved_wifi_bwe_changed) {
-    LOG(LS_INFO) << "Improved WIFI BWE called.";
-    webrtc::Config config;
-    config.Set(new webrtc::AimdRemoteRateControl(
-        options_.use_improved_wifi_bandwidth_estimator
-          .GetWithDefaultIfUnset(false)));
-    for (SendChannelMap::iterator it = send_channels_.begin();
-            it != send_channels_.end(); ++it) {
-      engine()->vie()->network()->SetBandwidthEstimationConfig(
-          it->second->channel_id(), config);
-    }
-  }
 #ifdef USE_WEBRTC_DEV_BRANCH
   if (payload_padding_changed) {
     LOG(LS_INFO) << "Payload-based padding called.";
@@ -3602,6 +3585,16 @@ bool WebRtcVideoMediaChannel::ConfigureSending(int channel_id,
     return false;
   }
 
+  // Enable improved WiFi Bandwidth Estimation
+  {
+    webrtc::Config config;
+    config.Set(new webrtc::AimdRemoteRateControl(true));
+    if (!engine()->vie()->network()->SetBandwidthEstimationConfig(channel_id,
+                                                                  config)) {
+      return false;
+    }
+  }
+
   send_channels_[local_ssrc_key] = send_channel.release();
 
   return true;
diff --git a/media/webrtc/webrtcvideoengine2.cc b/media/webrtc/webrtcvideoengine2.cc
index 5bb9db4..ea53596 100644
--- a/media/webrtc/webrtcvideoengine2.cc
+++ b/media/webrtc/webrtcvideoengine2.cc
@@ -59,17 +59,14 @@ static const int kDefaultRtcpReceiverReportSsrc = 1;
 
 struct VideoCodecPref {
   int payload_type;
+  int width;
+  int height;
   const char* name;
   int rtx_payload_type;
-} kDefaultVideoCodecPref = {100, kVp8CodecName, 96};
+} kDefaultVideoCodecPref = {100, 640, 400, kVp8CodecName, 96};
 
-VideoCodecPref kRedPref = {116, kRedCodecName, -1};
-VideoCodecPref kUlpfecPref = {117, kUlpfecCodecName, -1};
-
-// The formats are sorted by the descending order of width. We use the order to
-// find the next format for CPU and bandwidth adaptation.
-const VideoFormatPod kDefaultMaxVideoFormat = {
-    640, 400, FPS_TO_INTERVAL(kDefaultFramerate), FOURCC_ANY};
+VideoCodecPref kRedPref = {116, -1, -1, kRedCodecName, -1};
+VideoCodecPref kUlpfecPref = {117, -1, -1, kUlpfecCodecName, -1};
 
 static bool FindFirstMatchingCodec(const std::vector<VideoCodec>& codecs,
                                    const VideoCodec& requested_codec,
@@ -107,8 +104,8 @@ static bool IsRembEnabled(const VideoCodec& codec) {
 static VideoCodec DefaultVideoCodec() {
   VideoCodec default_codec(kDefaultVideoCodecPref.payload_type,
                            kDefaultVideoCodecPref.name,
-                           kDefaultMaxVideoFormat.width,
-                           kDefaultMaxVideoFormat.height,
+                           kDefaultVideoCodecPref.width,
+                           kDefaultVideoCodecPref.height,
                            kDefaultFramerate,
                            0);
   AddDefaultFeedbackParams(&default_codec);
@@ -285,13 +282,21 @@ void DefaultUnsignalledSsrcHandler::SetDefaultRenderer(
   }
 }
 
-WebRtcVideoEngine2::WebRtcVideoEngine2() {
+WebRtcVideoEngine2::WebRtcVideoEngine2()
+    : default_codec_format_(kDefaultVideoCodecPref.width,
+                            kDefaultVideoCodecPref.height,
+                            FPS_TO_INTERVAL(kDefaultFramerate),
+                            FOURCC_ANY) {
   // Construct without a factory or voice engine.
   Construct(NULL, NULL, new rtc::CpuMonitor(NULL));
 }
 
 WebRtcVideoEngine2::WebRtcVideoEngine2(
-    WebRtcVideoChannelFactory* channel_factory) {
+    WebRtcVideoChannelFactory* channel_factory)
+    : default_codec_format_(kDefaultVideoCodecPref.width,
+                            kDefaultVideoCodecPref.height,
+                            FPS_TO_INTERVAL(kDefaultFramerate),
+                            FOURCC_ANY) {
   // Construct without a voice engine.
   Construct(channel_factory, NULL, new rtc::CpuMonitor(NULL));
 }
@@ -308,7 +313,6 @@ void WebRtcVideoEngine2::Construct(WebRtcVideoChannelFactory* channel_factory,
   channel_factory_ = channel_factory;
 
   video_codecs_ = DefaultVideoCodecs();
-  default_codec_format_ = VideoFormat(kDefaultMaxVideoFormat);
 
   rtp_header_extensions_.push_back(
       RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
diff --git a/media/webrtc/webrtcvoiceengine.cc b/media/webrtc/webrtcvoiceengine.cc
index 81a6cdc..5cf53e4 100644
--- a/media/webrtc/webrtcvoiceengine.cc
+++ b/media/webrtc/webrtcvoiceengine.cc
@@ -237,7 +237,6 @@ static AudioOptions GetDefaultEngineOptions() {
   options.experimental_aec.Set(false);
   options.experimental_ns.Set(false);
   options.aec_dump.Set(false);
-  options.opus_fec.Set(false);
   return options;
 }
 
@@ -430,15 +429,6 @@ static bool IsOpusFecEnabled(const AudioCodec& codec) {
   return codec.GetParam(kCodecParamUseInbandFec, &value) && value == 1;
 }
 
-// Set params[kCodecParamUseInbandFec]. Caller should make sure codec is Opus.
-static void SetOpusFec(AudioCodec* codec, bool opus_fec) {
-  if (opus_fec) {
-    codec->SetParam(kCodecParamUseInbandFec, 1);
-  } else {
-    codec->RemoveParam(kCodecParamUseInbandFec);
-  }
-}
-
 void WebRtcVoiceEngine::ConstructCodecs() {
   LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
   int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
@@ -483,7 +473,6 @@ void WebRtcVoiceEngine::ConstructCodecs() {
           }
           // TODO(hellner): Add ptime, sprop-stereo, stereo and useinbandfec
           // when they can be set to values other than the default.
-          SetOpusFec(&codec, false);
         }
         codecs_.push_back(codec);
       } else {
@@ -916,16 +905,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     }
   }
 
-  bool opus_fec;
-  if (options.opus_fec.Get(&opus_fec)) {
-    LOG(LS_INFO) << "Opus FEC is enabled? " << opus_fec;
-    for (std::vector<AudioCodec>::iterator it = codecs_.begin();
-        it != codecs_.end(); ++it) {
-      if (IsOpus(*it))
-        SetOpusFec(&(*it), opus_fec);
-    }
-  }
-
   return true;
 }
 
diff --git a/media/webrtc/webrtcvoiceengine_unittest.cc b/media/webrtc/webrtcvoiceengine_unittest.cc
index 1798d1d..89d4c4d 100644
--- a/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -1230,31 +1230,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecIsacWithParamNoFec) {
 }
 #endif  // USE_WEBRTC_DEV_BRANCH
 
-// Test AudioOptions controls whether opus FEC is supported in codec list.
-TEST_F(WebRtcVoiceEngineTestFake, OpusFecViaOptions) {
-  EXPECT_TRUE(SetupEngine());
-  std::vector<cricket::AudioCodec> codecs = engine_.codecs();
-  int value;
-  for (std::vector<cricket::AudioCodec>::const_iterator it = codecs.begin();
-      it != codecs.end(); ++it) {
-    if (_stricmp(it->name.c_str(), cricket::kOpusCodecName) == 0) {
-      EXPECT_FALSE(it->GetParam(cricket::kCodecParamUseInbandFec, &value));
-    }
-  }
-
-  cricket::AudioOptions options;
-  options.opus_fec.Set(true);
-  EXPECT_TRUE(engine_.SetOptions(options));
-  codecs = engine_.codecs();
-  for (std::vector<cricket::AudioCodec>::const_iterator it = codecs.begin();
-      it != codecs.end(); ++it) {
-    if (_stricmp(it->name.c_str(), cricket::kOpusCodecName) == 0) {
-      EXPECT_TRUE(it->GetParam(cricket::kCodecParamUseInbandFec, &value));
-      EXPECT_EQ(1, value);
-    }
-  }
-}
-
 // Test that we can apply CELT with stereo mode but fail with mono mode.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCelt) {
   EXPECT_TRUE(SetupEngine());
diff --git a/p2p/base/dtlstransportchannel.cc b/p2p/base/dtlstransportchannel.cc
index c41d383..e0f4141 100644
--- a/p2p/base/dtlstransportchannel.cc
+++ b/p2p/base/dtlstransportchannel.cc
@@ -298,6 +298,11 @@ bool DtlsTransportChannelWrapper::SetSrtpCiphers(
   if (srtp_ciphers_ == ciphers)
     return true;
 
+  if (dtls_state_ == STATE_STARTED) {
+    LOG(LS_WARNING) << "Ignoring new SRTP ciphers while DTLS is negotiating";
+    return true;
+  }
+
   if (dtls_state_ == STATE_OPEN) {
     // We don't support DTLS renegotiation currently. If new set of srtp ciphers
     // are different than what's being used currently, we will not use it.
diff --git a/p2p/base/turnport_unittest.cc b/p2p/base/turnport_unittest.cc
index 4582dc0..44dc64f 100644
--- a/p2p/base/turnport_unittest.cc
+++ b/p2p/base/turnport_unittest.cc
@@ -378,7 +378,7 @@ TEST_F(TurnPortTest, TestTurnTcpAllocate) {
 
 // Testing turn port will attempt to create TCP socket on address resolution
 // failure.
-TEST_F(TurnPortTest, TestTurnTcpOnAddressResolveFailure) {
+TEST_F(TurnPortTest, DISABLED_TestTurnTcpOnAddressResolveFailure) {
   turn_server_.AddInternalSocket(kTurnTcpIntAddr, cricket::PROTO_TCP);
   CreateTurnPort(kTurnUsername, kTurnPassword, cricket::ProtocolAddress(
       rtc::SocketAddress("www.webrtc-blah-blah.com", 3478),
@@ -393,7 +393,7 @@ TEST_F(TurnPortTest, TestTurnTcpOnAddressResolveFailure) {
 
 // In case of UDP on address resolve failure, TurnPort will not create socket
 // and return allocate failure.
-TEST_F(TurnPortTest, TestTurnUdpOnAdressResolveFailure) {
+TEST_F(TurnPortTest, DISABLED_TestTurnUdpOnAdressResolveFailure) {
   CreateTurnPort(kTurnUsername, kTurnPassword, cricket::ProtocolAddress(
       rtc::SocketAddress("www.webrtc-blah-blah.com", 3478),
       cricket::PROTO_UDP));
diff --git a/session/media/channelmanager.cc b/session/media/channelmanager.cc
index 8ceb148..a1cdcc0 100644
--- a/session/media/channelmanager.cc
+++ b/session/media/channelmanager.cc
@@ -120,7 +120,7 @@ void ChannelManager::Construct(MediaEngineInterface* me,
   audio_options_ = media_engine_->GetAudioOptions();
   audio_in_device_ = DeviceManagerInterface::kDefaultDeviceName;
   audio_out_device_ = DeviceManagerInterface::kDefaultDeviceName;
-  audio_delay_offset_ = MediaEngineInterface::kDefaultAudioDelayOffset;
+  audio_delay_offset_ = kDefaultAudioDelayOffset;
   audio_output_volume_ = kNotSetOutputVolume;
   local_renderer_ = NULL;
   capturing_ = false;
diff --git a/session/media/channelmanager.h b/session/media/channelmanager.h
index e13ae12..f7a2890 100644
--- a/session/media/channelmanager.h
+++ b/session/media/channelmanager.h
@@ -42,6 +42,8 @@
 
 namespace cricket {
 
+const int kDefaultAudioDelayOffset = 0;
+
 class Soundclip;
 class VideoProcessor;
 class VoiceChannel;
diff --git a/session/media/channelmanager_unittest.cc b/session/media/channelmanager_unittest.cc
index 3896112..f234732 100644
--- a/session/media/channelmanager_unittest.cc
+++ b/session/media/channelmanager_unittest.cc
@@ -267,7 +267,7 @@ TEST_F(ChannelManagerTest, SetAudioOptionsBeforeInit) {
   EXPECT_EQ(options, set_options);
   // At this point, the media engine should also be initialized.
   EXPECT_EQ(options, fme_->audio_options());
-  EXPECT_EQ(cricket::MediaEngineInterface::kDefaultAudioDelayOffset,
+  EXPECT_EQ(cricket::kDefaultAudioDelayOffset,
             fme_->audio_delay_offset());
 }
 
@@ -294,7 +294,7 @@ TEST_F(ChannelManagerTest, SetAudioOptions) {
             fme_->audio_in_device());
   EXPECT_EQ(std::string(cricket::DeviceManagerInterface::kDefaultDeviceName),
             fme_->audio_out_device());
-  EXPECT_EQ(cricket::MediaEngineInterface::kDefaultAudioDelayOffset,
+  EXPECT_EQ(cricket::kDefaultAudioDelayOffset,
             fme_->audio_delay_offset());
   // Test setting specific values.
   AudioOptions options;
@@ -306,7 +306,7 @@ TEST_F(ChannelManagerTest, SetAudioOptions) {
   EXPECT_TRUE(
       fme_->audio_options().auto_gain_control.Get(&auto_gain_control));
   EXPECT_TRUE(auto_gain_control);
-  EXPECT_EQ(cricket::MediaEngineInterface::kDefaultAudioDelayOffset,
+  EXPECT_EQ(cricket::kDefaultAudioDelayOffset,
             fme_->audio_delay_offset());
   // Test setting bad values.
   EXPECT_FALSE(cm_->SetAudioOptions("audio-in9", "audio-out2", options));