diff --git a/talk/app/webrtc/audiotrack.h b/talk/app/webrtc/audiotrack.h
index fc09d2b9e..19db73f83 100644
--- a/talk/app/webrtc/audiotrack.h
+++ b/talk/app/webrtc/audiotrack.h
@@ -41,17 +41,21 @@ class AudioTrack : public MediaStreamTrack<AudioTrackInterface> {
   static talk_base::scoped_refptr<AudioTrack> Create(
       const std::string& id, AudioSourceInterface* source);
 
-  virtual AudioSourceInterface* GetSource() const {
+  // AudioTrackInterface implementation.
+  virtual AudioSourceInterface* GetSource() const OVERRIDE {
     return audio_source_.get();
   }
-
-  // This method is used for supporting multiple sources/sinks for AudioTracks.
-  virtual cricket::AudioRenderer* GetRenderer() {
+  // TODO(xians): Implement these methods.
+  virtual void AddSink(AudioTrackSinkInterface* sink) OVERRIDE {}
+  virtual void RemoveSink(AudioTrackSinkInterface* sink) OVERRIDE {}
+  virtual bool GetSignalLevel(int* level) OVERRIDE { return false; }
+  virtual AudioProcessorInterface* GetAudioProcessor() OVERRIDE { return NULL; }
+  virtual cricket::AudioRenderer* GetRenderer() OVERRIDE {
     return NULL;
   }
 
-  // Implement MediaStreamTrack
-  virtual std::string kind() const;
+  // MediaStreamTrack implementation.
+  virtual std::string kind() const OVERRIDE;
 
  protected:
   AudioTrack(const std::string& label, AudioSourceInterface* audio_source);
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index fa0572e38..acd5bf612 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -174,18 +174,55 @@ class AudioTrackSinkInterface {
   virtual ~AudioTrackSinkInterface() {}
 };
 
+// Interface of the audio processor used by the audio track to collect
+// statistics.
+class AudioProcessorInterface : public talk_base::RefCountInterface {
+ public:
+  struct AudioProcessorStats {
+    AudioProcessorStats() : typing_noise_detected(false),
+                            echo_return_loss(0),
+                            echo_return_loss_enhancement(0),
+                            echo_delay_median_ms(0),
+                            aec_quality_min(0.0),
+                            echo_delay_std_ms(0) {}
+    ~AudioProcessorStats() {}
+
+    bool typing_noise_detected;
+    int echo_return_loss;
+    int echo_return_loss_enhancement;
+    int echo_delay_median_ms;
+    float aec_quality_min;
+    int echo_delay_std_ms;
+  };
+
+  // Get audio processor statistics.
+  virtual void GetStats(AudioProcessorStats* stats) = 0;
+
+ protected:
+  virtual ~AudioProcessorInterface() {}
+};
+
 class AudioTrackInterface : public MediaStreamTrackInterface {
  public:
   // TODO(xians): Figure out if the following interface should be const or not.
   virtual AudioSourceInterface* GetSource() const =  0;
 
-  // Adds/Removes a sink that will receive the audio data from the track.
-  // TODO(xians): Make them pure virtual after Chrome implements these
-  // interfaces.
-  virtual void AddSink(AudioTrackSinkInterface* sink) {}
-  virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
+  // Add/Remove a sink that will receive the audio data from the track.
+  virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
+  virtual void RemoveSink(AudioTrackSinkInterface* sink) = 0;
 
-  // Gets a pointer to the audio renderer of this AudioTrack.
+  // Get the signal level from the audio track.
+  // Return true on success, otherwise false.
+  // TODO(xians): Change the interface to int GetSignalLevel() and pure virtual
+  // after Chrome has the correct implementation of the interface.
+  virtual bool GetSignalLevel(int* level) { return false; }
+
+  // Get the audio processor used by the audio track. Return NULL if the track
+  // does not have any processor.
+  // TODO(xians): Make the interface pure virtual.
+  virtual AudioProcessorInterface* GetAudioProcessor() { return NULL; }
+
+  // Get a pointer to the audio renderer of this AudioTrack.
   // The pointer is valid for the lifetime of this AudioTrack.
   // TODO(xians): Remove the following interface after Chrome switches to
   // AddSink() and RemoveSink() interfaces.
diff --git a/talk/app/webrtc/mediastreamsignaling.cc b/talk/app/webrtc/mediastreamsignaling.cc
index 14648eee4..e1b33f102 100644
--- a/talk/app/webrtc/mediastreamsignaling.cc
+++ b/talk/app/webrtc/mediastreamsignaling.cc
@@ -321,8 +321,29 @@ bool MediaStreamSignaling::AddLocalStream(MediaStreamInterface* local_stream) {
 
 void MediaStreamSignaling::RemoveLocalStream(
     MediaStreamInterface* local_stream) {
-  local_streams_->RemoveStream(local_stream);
+  AudioTrackVector audio_tracks = local_stream->GetAudioTracks();
+  for (AudioTrackVector::const_iterator it = audio_tracks.begin();
+       it != audio_tracks.end(); ++it) {
+    const TrackInfo* track_info = FindTrackInfo(local_audio_tracks_,
+                                                local_stream->label(),
+                                                (*it)->id());
+    if (track_info) {
+      stream_observer_->OnRemoveLocalAudioTrack(local_stream, *it,
+                                                track_info->ssrc);
+    }
+  }
+  VideoTrackVector video_tracks = local_stream->GetVideoTracks();
+  for (VideoTrackVector::const_iterator it = video_tracks.begin();
+       it != video_tracks.end(); ++it) {
+    const TrackInfo* track_info = FindTrackInfo(local_video_tracks_,
+                                                local_stream->label(),
+                                                (*it)->id());
+    if (track_info) {
+      stream_observer_->OnRemoveLocalVideoTrack(local_stream, *it);
+    }
+  }
 
+  local_streams_->RemoveStream(local_stream);
   stream_observer_->OnRemoveLocalStream(local_stream);
 }
 
@@ -725,7 +746,8 @@ void MediaStreamSignaling::UpdateLocalTracks(
     cricket::StreamParams params;
     if (!cricket::GetStreamBySsrc(streams, info.ssrc, &params) ||
         params.id != info.track_id || params.sync_label != info.stream_label) {
-      OnLocalTrackRemoved(info.stream_label, info.track_id, media_type);
+      OnLocalTrackRemoved(info.stream_label, info.track_id, info.ssrc,
+                          media_type);
       track_it = current_tracks->erase(track_it);
     } else {
       ++track_it;
@@ -787,6 +809,7 @@ void MediaStreamSignaling::OnLocalTrackSeen(
 void MediaStreamSignaling::OnLocalTrackRemoved(
     const std::string& stream_label,
     const std::string& track_id,
+    uint32 ssrc,
     cricket::MediaType media_type) {
   MediaStreamInterface* stream = local_streams_->find(stream_label);
   if (!stream) {
@@ -803,7 +826,7 @@ void MediaStreamSignaling::OnLocalTrackRemoved(
     if (!audio_track) {
       return;
     }
-    stream_observer_->OnRemoveLocalAudioTrack(stream, audio_track);
+    stream_observer_->OnRemoveLocalAudioTrack(stream, audio_track, ssrc);
   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
     VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
     if (!video_track) {
diff --git a/talk/app/webrtc/mediastreamsignaling.h b/talk/app/webrtc/mediastreamsignaling.h
index c730b468d..dc83b47da 100644
--- a/talk/app/webrtc/mediastreamsignaling.h
+++ b/talk/app/webrtc/mediastreamsignaling.h
@@ -92,7 +92,8 @@ class MediaStreamSignalingObserver {
 
   // Triggered when the local SessionDescription has removed an audio track.
   virtual void OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
-                                       AudioTrackInterface* audio_track) = 0;
+                                       AudioTrackInterface* audio_track,
+                                       uint32 ssrc) = 0;
 
   // Triggered when the local SessionDescription has removed a video track.
   virtual void OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
@@ -354,6 +355,7 @@ class MediaStreamSignaling {
   // MediaStreamTrack in a MediaStream in |local_streams_|.
   void OnLocalTrackRemoved(const std::string& stream_label,
                            const std::string& track_id,
+                           uint32 ssrc,
                            cricket::MediaType media_type);
 
   void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams);
diff --git a/talk/app/webrtc/mediastreamsignaling_unittest.cc b/talk/app/webrtc/mediastreamsignaling_unittest.cc
index 49625ef52..03452738b 100644
--- a/talk/app/webrtc/mediastreamsignaling_unittest.cc
+++ b/talk/app/webrtc/mediastreamsignaling_unittest.cc
@@ -298,7 +298,8 @@ class MockSignalingObserver : public webrtc::MediaStreamSignalingObserver {
   }
 
   virtual void OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
-                                       AudioTrackInterface* audio_track) {
+                                       AudioTrackInterface* audio_track,
+                                       uint32 ssrc) {
     RemoveTrack(&local_audio_tracks_, stream, audio_track);
   }
 
diff --git a/talk/app/webrtc/mediastreamtrackproxy.h b/talk/app/webrtc/mediastreamtrackproxy.h
index 7c622e733..f12522e55 100644
--- a/talk/app/webrtc/mediastreamtrackproxy.h
+++ b/talk/app/webrtc/mediastreamtrackproxy.h
@@ -42,6 +42,10 @@ BEGIN_PROXY_MAP(AudioTrack)
   PROXY_CONSTMETHOD0(TrackState, state)
   PROXY_CONSTMETHOD0(bool, enabled)
   PROXY_CONSTMETHOD0(AudioSourceInterface*, GetSource)
+  PROXY_METHOD1(void, AddSink, AudioTrackSinkInterface*)
+  PROXY_METHOD1(void, RemoveSink, AudioTrackSinkInterface*)
+  PROXY_METHOD1(bool, GetSignalLevel, int*)
+  PROXY_METHOD0(AudioProcessorInterface*, GetAudioProcessor)
   PROXY_METHOD0(cricket::AudioRenderer*, GetRenderer)
 
   PROXY_METHOD1(bool, set_enabled, bool)
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 72f37174b..af33d02dd 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -706,6 +706,7 @@ void PeerConnection::OnAddLocalAudioTrack(MediaStreamInterface* stream,
                                           AudioTrackInterface* audio_track,
                                           uint32 ssrc) {
   stream_handler_container_->AddLocalAudioTrack(stream, audio_track, ssrc);
+  stats_.AddLocalAudioTrack(audio_track, ssrc);
 }
 void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream,
                                           VideoTrackInterface* video_track,
@@ -714,8 +715,10 @@ void PeerConnection::OnAddLocalVideoTrack(MediaStreamInterface* stream,
 }
 
 void PeerConnection::OnRemoveLocalAudioTrack(MediaStreamInterface* stream,
-                                             AudioTrackInterface* audio_track) {
+                                             AudioTrackInterface* audio_track,
+                                             uint32 ssrc) {
   stream_handler_container_->RemoveLocalTrack(stream, audio_track);
+  stats_.RemoveLocalAudioTrack(audio_track, ssrc);
 }
 
 void PeerConnection::OnRemoveLocalVideoTrack(MediaStreamInterface* stream,
diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h
index 70155d992..b4c45c316 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -136,7 +136,8 @@ class PeerConnection : public PeerConnectionInterface,
                                     uint32 ssrc) OVERRIDE;
   virtual void OnRemoveLocalAudioTrack(
       MediaStreamInterface* stream,
-      AudioTrackInterface* audio_track) OVERRIDE;
+      AudioTrackInterface* audio_track,
+      uint32 ssrc) OVERRIDE;
   virtual void OnRemoveLocalVideoTrack(
       MediaStreamInterface* stream,
       VideoTrackInterface* video_track) OVERRIDE;
diff --git a/talk/app/webrtc/statscollector.cc b/talk/app/webrtc/statscollector.cc
index 4d5b32418..6af1e16ad 100644
--- a/talk/app/webrtc/statscollector.cc
+++ b/talk/app/webrtc/statscollector.cc
@@ -201,6 +201,19 @@ void StatsReport::AddBoolean(const std::string& name, bool value) {
   AddValue(name, value ? "true" : "false");
 }
 
+void StatsReport::ReplaceValue(const std::string& name,
+                               const std::string& value) {
+  for (Values::iterator it = values.begin(); it != values.end(); ++it) {
+    if ((*it).name == name) {
+      it->value = value;
+      return;
+    }
+  }
+  // It is not reachable here, add an ASSERT to make sure the overwriting is
+  // always a success.
+  ASSERT(false);
+}
+
 namespace {
 typedef std::map<std::string, StatsReport> StatsMap;
 
@@ -458,6 +471,32 @@ void StatsCollector::AddStream(MediaStreamInterface* stream) {
                                        &reports_);
 }
 
+void StatsCollector::AddLocalAudioTrack(AudioTrackInterface* audio_track,
+                                        uint32 ssrc) {
+  ASSERT(audio_track != NULL);
+#ifdef _DEBUG
+  for (LocalAudioTrackVector::iterator it = local_audio_tracks_.begin();
+       it != local_audio_tracks_.end(); ++it) {
+    ASSERT(it->first != audio_track || it->second != ssrc);
+  }
+#endif
+  local_audio_tracks_.push_back(std::make_pair(audio_track, ssrc));
+}
+
+void StatsCollector::RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
+                                           uint32 ssrc) {
+  ASSERT(audio_track != NULL);
+  for (LocalAudioTrackVector::iterator it = local_audio_tracks_.begin();
+       it != local_audio_tracks_.end(); ++it) {
+    if (it->first == audio_track && it->second == ssrc) {
+      local_audio_tracks_.erase(it);
+      return;
+    }
+  }
+
+  ASSERT(false);
+}
+
 bool StatsCollector::GetStats(MediaStreamTrackInterface* track,
                               StatsReports* reports) {
   ASSERT(reports != NULL);
@@ -784,6 +823,8 @@ void StatsCollector::ExtractVoiceInfo() {
   }
   ExtractStatsFromList(voice_info.receivers, transport_id, this);
   ExtractStatsFromList(voice_info.senders, transport_id, this);
+
+  UpdateStatsFromExistingLocalAudioTracks();
 }
 
 void StatsCollector::ExtractVideoInfo(
@@ -840,19 +881,86 @@ bool StatsCollector::GetTransportIdFromProxy(const std::string& proxy,
   return true;
 }
 
-StatsReport* StatsCollector::GetOrCreateReport(const std::string& type,
-                                               const std::string& id) {
+StatsReport* StatsCollector::GetReport(const std::string& type,
+                                       const std::string& id) {
   std::string statsid = StatsId(type, id);
   StatsReport* report = NULL;
   std::map<std::string, StatsReport>::iterator it = reports_.find(statsid);
-  if (it == reports_.end()) {
-    report = &reports_[statsid];  // Create new element.
-    report->id = statsid;
-    report->type = type;
-  } else {
+  if (it != reports_.end())
     report = &(it->second);
-  }
+
   return report;
 }
 
+StatsReport* StatsCollector::GetOrCreateReport(const std::string& type,
+                                               const std::string& id) {
+  StatsReport* report = GetReport(type, id);
+  if (report == NULL) {
+    std::string statsid = StatsId(type, id);
+    report = &reports_[statsid];  // Create new element.
+    report->id = statsid;
+    report->type = type;
+  }
+
+  return report;
+}
+
+void StatsCollector::UpdateStatsFromExistingLocalAudioTracks() {
+  // Loop through the existing local audio tracks.
+  for (LocalAudioTrackVector::const_iterator it = local_audio_tracks_.begin();
+       it != local_audio_tracks_.end(); ++it) {
+    AudioTrackInterface* track = it->first;
+    uint32 ssrc = it->second;
+    std::string ssrc_id = talk_base::ToString<uint32>(ssrc);
+    StatsReport* report = GetReport(StatsReport::kStatsReportTypeSsrc,
+                                    ssrc_id);
+    ASSERT(report != NULL);
+
+    // The same ssrc can be used by both local and remote audio tracks.
+    std::string track_id;
+    if (!ExtractValueFromReport(*report,
+                                StatsReport::kStatsValueNameTrackId,
+                                &track_id) ||
+        track_id != track->id()) {
+      continue;
+    }
+
+    UpdateReportFromAudioTrack(track, report);
+  }
+}
+
+void StatsCollector::UpdateReportFromAudioTrack(AudioTrackInterface* track,
+                                                StatsReport* report) {
+  ASSERT(track != NULL);
+  if (report == NULL)
+    return;
+
+  int signal_level = 0;
+  if (track->GetSignalLevel(&signal_level)) {
+    report->ReplaceValue(StatsReport::kStatsValueNameAudioInputLevel,
+                         talk_base::ToString<int>(signal_level));
+  }
+
+  talk_base::scoped_refptr<AudioProcessorInterface> audio_processor(
+      track->GetAudioProcessor());
+  if (audio_processor.get() == NULL)
+    return;
+
+  AudioProcessorInterface::AudioProcessorStats stats;
+  audio_processor->GetStats(&stats);
+  report->ReplaceValue(StatsReport::kStatsValueNameTypingNoiseState,
+                       stats.typing_noise_detected ? "true" : "false");
+  report->ReplaceValue(StatsReport::kStatsValueNameEchoReturnLoss,
+                       talk_base::ToString<int>(stats.echo_return_loss));
+  report->ReplaceValue(
+      StatsReport::kStatsValueNameEchoReturnLossEnhancement,
+      talk_base::ToString<int>(stats.echo_return_loss_enhancement));
+  report->ReplaceValue(StatsReport::kStatsValueNameEchoDelayMedian,
+                       talk_base::ToString<int>(stats.echo_delay_median_ms));
+  report->ReplaceValue(StatsReport::kStatsValueNameEchoCancellationQualityMin,
+                       talk_base::ToString<float>(stats.aec_quality_min));
+  report->ReplaceValue(StatsReport::kStatsValueNameEchoDelayStdDev,
+                       talk_base::ToString<int>(stats.echo_delay_std_ms));
+}
+
 }  // namespace webrtc
diff --git a/talk/app/webrtc/statscollector.h b/talk/app/webrtc/statscollector.h
index 6256d7786..fdb296157 100644
--- a/talk/app/webrtc/statscollector.h
+++ b/talk/app/webrtc/statscollector.h
@@ -31,8 +31,9 @@
 #ifndef TALK_APP_WEBRTC_STATSCOLLECTOR_H_
 #define TALK_APP_WEBRTC_STATSCOLLECTOR_H_
 
-#include <string>
 #include <map>
+#include <string>
+#include <vector>
 
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
@@ -57,6 +58,13 @@ class StatsCollector {
   // to GetStats.
   void AddStream(MediaStreamInterface* stream);
 
+  // Adds a local audio track that is used for getting some voice statistics.
+  void AddLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
+
+  // Removes a local audio tracks that is used for getting some voice
+  // statistics.
+  void RemoveLocalAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc);
+
   // Gather statistics from the session and store them for future use.
   void UpdateStats(PeerConnectionInterface::StatsOutputLevel level);
 
@@ -95,6 +103,13 @@ class StatsCollector {
   WebRtcSession* session() { return session_; }
   webrtc::StatsReport* GetOrCreateReport(const std::string& type,
                                          const std::string& id);
+  webrtc::StatsReport* GetReport(const std::string& type,
+                                 const std::string& id);
+
+  // Helper method to get stats from the local audio tracks.
+  void UpdateStatsFromExistingLocalAudioTracks();
+  void UpdateReportFromAudioTrack(AudioTrackInterface* track,
+                                  StatsReport* report);
 
   // A map from the report id to the report.
   std::map<std::string, StatsReport> reports_;
@@ -103,6 +118,10 @@ class StatsCollector {
   double stats_gathering_started_;
   talk_base::Timing timing_;
   cricket::ProxyTransportMap proxy_to_transport_;
+
+  typedef std::vector<std::pair<AudioTrackInterface*, uint32> >
+      LocalAudioTrackVector;
+  LocalAudioTrackVector local_audio_tracks_;
 };
 
 }  // namespace webrtc
diff --git a/talk/app/webrtc/statscollector_unittest.cc b/talk/app/webrtc/statscollector_unittest.cc
index a7cda1635..83d214c46 100644
--- a/talk/app/webrtc/statscollector_unittest.cc
+++ b/talk/app/webrtc/statscollector_unittest.cc
@@ -29,6 +29,8 @@
 #include "talk/app/webrtc/statscollector.h"
 
 #include "talk/app/webrtc/mediastream.h"
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/app/webrtc/mediastreamtrack.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/base/base64.h"
 #include "talk/base/fakesslidentity.h"
@@ -47,6 +49,8 @@ using testing::Return;
 using testing::ReturnNull;
 using testing::SetArgPointee;
 using webrtc::PeerConnectionInterface;
+using webrtc::StatsReport;
+using webrtc::StatsReports;
 
 namespace cricket {
 
@@ -63,6 +67,7 @@ const char kNoReports[] = "NO REPORTS";
 
 // Constant names for track identification.
 const char kTrackId[] = "somename";
+const char kAudioTrackId[] = "audio_track_id";
 const uint32 kSsrcOfTrack = 1234;
 
 class MockWebRtcSession : public webrtc::WebRtcSession {
@@ -71,6 +76,7 @@ class MockWebRtcSession : public webrtc::WebRtcSession {
     : WebRtcSession(channel_manager, talk_base::Thread::Current(),
                     talk_base::Thread::Current(), NULL, NULL) {
   }
+  MOCK_METHOD0(voice_channel, cricket::VoiceChannel*());
   MOCK_METHOD0(video_channel, cricket::VideoChannel*());
   MOCK_METHOD2(GetTrackIdBySsrc, bool(uint32, std::string*));
   MOCK_METHOD1(GetStats, bool(cricket::SessionStats*));
@@ -86,10 +92,60 @@ class MockVideoMediaChannel : public cricket::FakeVideoMediaChannel {
   MOCK_METHOD2(GetStats, bool(const StatsOptions&, cricket::VideoMediaInfo*));
 };
 
-bool GetValue(const webrtc::StatsReport* report,
+class MockVoiceMediaChannel : public cricket::FakeVoiceMediaChannel {
+ public:
+  MockVoiceMediaChannel() : cricket::FakeVoiceMediaChannel(NULL) {
+  }
+  MOCK_METHOD1(GetStats, bool(cricket::VoiceMediaInfo*));
+};
+
+class FakeAudioProcessor : public webrtc::AudioProcessorInterface {
+ public:
+  FakeAudioProcessor() {}
+  ~FakeAudioProcessor() {}
+
+ private:
+  virtual void GetStats(
+      AudioProcessorInterface::AudioProcessorStats* stats) OVERRIDE {
+    stats->typing_noise_detected = true;
+    stats->echo_return_loss = 2;
+    stats->echo_return_loss_enhancement = 3;
+    stats->echo_delay_median_ms = 4;
+    stats->aec_quality_min = 5.1f;
+    stats->echo_delay_std_ms = 6;
+  }
+};
+
+class FakeLocalAudioTrack
+    : public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface> {
+ public:
+  explicit FakeLocalAudioTrack(const std::string& id)
+      : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(id),
+        processor_(new talk_base::RefCountedObject<FakeAudioProcessor>()) {}
+  std::string kind() const OVERRIDE {
+    return "audio";
+  }
+  virtual webrtc::AudioSourceInterface* GetSource() const OVERRIDE {
+    return NULL;
+  }
+  virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE {}
+  virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE {}
+  virtual bool GetSignalLevel(int* level) OVERRIDE {
+    *level = 1;
+    return true;
+  }
+  virtual webrtc::AudioProcessorInterface* GetAudioProcessor() OVERRIDE {
+    return processor_.get();
+  }
+
+ private:
+  talk_base::scoped_refptr<FakeAudioProcessor> processor_;
+};
+
+bool GetValue(const StatsReport* report,
               const std::string& name,
               std::string* value) {
-  webrtc::StatsReport::Values::const_iterator it = report->values.begin();
+  StatsReport::Values::const_iterator it = report->values.begin();
   for (; it != report->values.end(); ++it) {
     if (it->name == name) {
       *value = it->value;
@@ -100,7 +156,7 @@ bool GetValue(const webrtc::StatsReport* report,
 }
 
 std::string ExtractStatsValue(const std::string& type,
-                              const webrtc::StatsReports& reports,
+                              const StatsReports& reports,
                               const std::string name) {
   if (reports.empty()) {
     return kNoReports;
@@ -119,8 +175,8 @@ std::string ExtractStatsValue(const std::string& type,
 
 // Finds the |n|-th report of type |type| in |reports|.
 // |n| starts from 1 for finding the first report.
-const webrtc::StatsReport* FindNthReportByType(
-    const webrtc::StatsReports& reports, const std::string& type, int n) {
+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) {
       n--;
@@ -131,8 +187,8 @@ const webrtc::StatsReport* FindNthReportByType(
   return NULL;
 }
 
-const webrtc::StatsReport* FindReportById(const webrtc::StatsReports& reports,
-                                          const std::string& id) {
+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];
@@ -141,16 +197,16 @@ const webrtc::StatsReport* FindReportById(const webrtc::StatsReports& reports,
   return NULL;
 }
 
-std::string ExtractSsrcStatsValue(webrtc::StatsReports reports,
+std::string ExtractSsrcStatsValue(StatsReports reports,
                                   const std::string& name) {
   return ExtractStatsValue(
-      webrtc::StatsReport::kStatsReportTypeSsrc, reports, name);
+      StatsReport::kStatsReportTypeSsrc, reports, name);
 }
 
-std::string ExtractBweStatsValue(webrtc::StatsReports reports,
+std::string ExtractBweStatsValue(StatsReports reports,
                                   const std::string& name) {
   return ExtractStatsValue(
-      webrtc::StatsReport::kStatsReportTypeBwe, reports, name);
+      StatsReport::kStatsReportTypeBwe, reports, name);
 }
 
 std::string DerToPem(const std::string& der) {
@@ -167,18 +223,18 @@ std::vector<std::string> DersToPems(
   return pems;
 }
 
-void CheckCertChainReports(const webrtc::StatsReports& reports,
+void CheckCertChainReports(const StatsReports& reports,
                            const std::vector<std::string>& ders,
                            const std::string& start_id) {
   std::string certificate_id = start_id;
   size_t i = 0;
   while (true) {
-    const webrtc::StatsReport* report = FindReportById(reports, certificate_id);
+    const StatsReport* report = FindReportById(reports, certificate_id);
     ASSERT_TRUE(report != NULL);
 
     std::string der_base64;
     EXPECT_TRUE(GetValue(
-        report, webrtc::StatsReport::kStatsValueNameDer, &der_base64));
+        report, StatsReport::kStatsValueNameDer, &der_base64));
     std::string der = talk_base::Base64::Decode(der_base64,
                                                 talk_base::Base64::DO_STRICT);
     EXPECT_EQ(ders[i], der);
@@ -186,7 +242,7 @@ void CheckCertChainReports(const webrtc::StatsReports& reports,
     std::string fingerprint_algorithm;
     EXPECT_TRUE(GetValue(
         report,
-        webrtc::StatsReport::kStatsValueNameFingerprintAlgorithm,
+        StatsReport::kStatsValueNameFingerprintAlgorithm,
         &fingerprint_algorithm));
     // The digest algorithm for a FakeSSLCertificate is always SHA-1.
     std::string sha_1_str = talk_base::DIGEST_SHA_1;
@@ -195,17 +251,68 @@ void CheckCertChainReports(const webrtc::StatsReports& reports,
     std::string dummy_fingerprint;  // Value is not checked.
     EXPECT_TRUE(GetValue(
         report,
-        webrtc::StatsReport::kStatsValueNameFingerprint,
+        StatsReport::kStatsValueNameFingerprint,
         &dummy_fingerprint));
 
     ++i;
     if (!GetValue(
-        report, webrtc::StatsReport::kStatsValueNameIssuerId, &certificate_id))
+        report, StatsReport::kStatsValueNameIssuerId, &certificate_id))
       break;
   }
   EXPECT_EQ(ders.size(), i);
 }
 
+void VerifyVoiceSenderInfoReport(const StatsReport* report,
+                                 const cricket::VoiceSenderInfo& sinfo) {
+  std::string value_in_report;
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameCodecName, &value_in_report));
+  EXPECT_EQ(sinfo.codec_name, value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameBytesSent, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int64>(sinfo.bytes_sent), value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNamePacketsSent, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.packets_sent), value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameRtt, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.rtt_ms), value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameRtt, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.rtt_ms), value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameJitterReceived, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.jitter_ms), value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameEchoCancellationQualityMin,
+      &value_in_report));
+  EXPECT_EQ(talk_base::ToString<float>(sinfo.aec_quality_min), value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameEchoDelayMedian, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.echo_delay_median_ms),
+            value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameEchoDelayStdDev, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.echo_delay_std_ms),
+            value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameEchoReturnLoss, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.echo_return_loss),
+            value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameEchoReturnLossEnhancement,
+      &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.echo_return_loss_enhancement),
+            value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameAudioInputLevel, &value_in_report));
+  EXPECT_EQ(talk_base::ToString<int>(sinfo.audio_level), value_in_report);
+  EXPECT_TRUE(GetValue(
+      report, StatsReport::kStatsValueNameTypingNoiseState, &value_in_report));
+  std::string typing_detected = sinfo.typing_noise_detected ? "true" : "false";
+  EXPECT_EQ(typing_detected, value_in_report);
+}
+
 class StatsCollectorTest : public testing::Test {
  protected:
   StatsCollectorTest()
@@ -245,12 +352,25 @@ class StatsCollectorTest : public testing::Test {
                             Return(true)));
   }
 
+  // Adds a local audio track with a given SSRC into the stats.
+  void AddLocalAudioTrackStats() {
+    if (stream_ == NULL)
+      stream_ = webrtc::MediaStream::Create("streamlabel");
+
+    audio_track_ =
+        new talk_base::RefCountedObject<FakeLocalAudioTrack>(kAudioTrackId);
+    stream_->AddTrack(audio_track_);
+    EXPECT_CALL(session_, GetTrackIdBySsrc(kSsrcOfTrack, _))
+        .WillRepeatedly(DoAll(SetArgPointee<1>(kAudioTrackId),
+                              Return(true)));
+  }
+
   void TestCertificateReports(const talk_base::FakeSSLCertificate& local_cert,
                               const std::vector<std::string>& local_ders,
                               const talk_base::FakeSSLCertificate& remote_cert,
                               const std::vector<std::string>& remote_ders) {
     webrtc::StatsCollector stats;  // Implementation under test.
-    webrtc::StatsReports reports;  // returned values.
+    StatsReports reports;  // returned values.
     stats.set_session(&session_);
 
     // Fake stats to process.
@@ -289,22 +409,22 @@ class StatsCollectorTest : public testing::Test {
     EXPECT_CALL(session_, GetStats(_))
       .WillOnce(DoAll(SetArgPointee<0>(session_stats),
                       Return(true)));
-    EXPECT_CALL(session_, video_channel())
-      .WillRepeatedly(ReturnNull());
+    EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+    EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
 
     stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
 
     stats.GetStats(NULL, &reports);
 
-    const webrtc::StatsReport* channel_report = FindNthReportByType(
-        reports, webrtc::StatsReport::kStatsReportTypeComponent, 1);
+    const StatsReport* channel_report = FindNthReportByType(
+        reports, StatsReport::kStatsReportTypeComponent, 1);
     EXPECT_TRUE(channel_report != NULL);
 
     // Check local certificate chain.
     std::string local_certificate_id = ExtractStatsValue(
-        webrtc::StatsReport::kStatsReportTypeComponent,
+        StatsReport::kStatsReportTypeComponent,
         reports,
-        webrtc::StatsReport::kStatsValueNameLocalCertificateId);
+        StatsReport::kStatsValueNameLocalCertificateId);
     if (local_ders.size() > 0) {
       EXPECT_NE(kNotFound, local_certificate_id);
       CheckCertChainReports(reports, local_ders, local_certificate_id);
@@ -314,9 +434,9 @@ class StatsCollectorTest : public testing::Test {
 
     // Check remote certificate chain.
     std::string remote_certificate_id = ExtractStatsValue(
-        webrtc::StatsReport::kStatsReportTypeComponent,
+        StatsReport::kStatsReportTypeComponent,
         reports,
-        webrtc::StatsReport::kStatsValueNameRemoteCertificateId);
+        StatsReport::kStatsValueNameRemoteCertificateId);
     if (remote_ders.size() > 0) {
       EXPECT_NE(kNotFound, remote_certificate_id);
       CheckCertChainReports(reports, remote_ders, remote_certificate_id);
@@ -331,6 +451,7 @@ class StatsCollectorTest : public testing::Test {
   cricket::SessionStats session_stats_;
   talk_base::scoped_refptr<webrtc::MediaStream> stream_;
   talk_base::scoped_refptr<webrtc::VideoTrack> track_;
+  talk_base::scoped_refptr<FakeLocalAudioTrack> audio_track_;
   std::string track_id_;
 };
 
@@ -340,7 +461,7 @@ TEST_F(StatsCollectorTest, BytesCounterHandles64Bits) {
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel;
   cricket::VideoChannel video_channel(talk_base::Thread::Current(),
       media_engine_, media_channel, &session_, "", false, NULL);
-  webrtc::StatsReports reports;  // returned values.
+  StatsReports reports;  // returned values.
   cricket::VideoSenderInfo video_sender_info;
   cricket::VideoMediaInfo stats_read;
   // The number of bytes must be larger than 0xFFFFFFFF for this test.
@@ -356,8 +477,8 @@ TEST_F(StatsCollectorTest, BytesCounterHandles64Bits) {
   video_sender_info.bytes_sent = kBytesSent;
   stats_read.senders.push_back(video_sender_info);
 
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
   EXPECT_CALL(*media_channel, GetStats(_, _))
     .WillOnce(DoAll(SetArgPointee<1>(stats_read),
                     Return(true)));
@@ -373,7 +494,7 @@ TEST_F(StatsCollectorTest, BandwidthEstimationInfoIsReported) {
   MockVideoMediaChannel* media_channel = new MockVideoMediaChannel;
   cricket::VideoChannel video_channel(talk_base::Thread::Current(),
       media_engine_, media_channel, &session_, "", false, NULL);
-  webrtc::StatsReports reports;  // returned values.
+  StatsReports reports;  // returned values.
   cricket::VideoSenderInfo video_sender_info;
   cricket::VideoMediaInfo stats_read;
   // Set up an SSRC just to test that we get both kinds of stats back: SSRC and
@@ -395,8 +516,8 @@ TEST_F(StatsCollectorTest, BandwidthEstimationInfoIsReported) {
   bwe.target_enc_bitrate = kTargetEncBitrate;
   stats_read.bw_estimations.push_back(bwe);
 
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
   EXPECT_CALL(*media_channel, GetStats(_, _))
     .WillOnce(DoAll(SetArgPointee<1>(stats_read),
                     Return(true)));
@@ -413,14 +534,14 @@ TEST_F(StatsCollectorTest, BandwidthEstimationInfoIsReported) {
 // exists in the returned stats.
 TEST_F(StatsCollectorTest, SessionObjectExists) {
   webrtc::StatsCollector stats;  // Implementation under test.
-  webrtc::StatsReports reports;  // returned values.
+  StatsReports reports;  // returned values.
   stats.set_session(&session_);
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
-  const webrtc::StatsReport* session_report = FindNthReportByType(
-      reports, webrtc::StatsReport::kStatsReportTypeSession, 1);
+  const StatsReport* session_report = FindNthReportByType(
+      reports, StatsReport::kStatsReportTypeSession, 1);
   EXPECT_FALSE(session_report == NULL);
 }
 
@@ -428,18 +549,18 @@ TEST_F(StatsCollectorTest, SessionObjectExists) {
 // in the returned stats.
 TEST_F(StatsCollectorTest, OnlyOneSessionObjectExists) {
   webrtc::StatsCollector stats;  // Implementation under test.
-  webrtc::StatsReports reports;  // returned values.
+  StatsReports reports;  // returned values.
   stats.set_session(&session_);
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
-  const webrtc::StatsReport* session_report = FindNthReportByType(
-      reports, webrtc::StatsReport::kStatsReportTypeSession, 1);
+  const StatsReport* session_report = FindNthReportByType(
+      reports, StatsReport::kStatsReportTypeSession, 1);
   EXPECT_FALSE(session_report == NULL);
   session_report = FindNthReportByType(
-      reports, webrtc::StatsReport::kStatsReportTypeSession, 2);
+      reports, StatsReport::kStatsReportTypeSession, 2);
   EXPECT_EQ(NULL, session_report);
 }
 
@@ -455,18 +576,18 @@ TEST_F(StatsCollectorTest, TrackObjectExistsWithoutUpdateStats) {
 
   stats.set_session(&session_);
 
-  webrtc::StatsReports reports;
+  StatsReports reports;
 
   // Verfies the existence of the track report.
   stats.GetStats(NULL, &reports);
   EXPECT_EQ((size_t)1, reports.size());
-  EXPECT_EQ(std::string(webrtc::StatsReport::kStatsReportTypeTrack),
+  EXPECT_EQ(std::string(StatsReport::kStatsReportTypeTrack),
             reports[0].type);
 
   std::string trackValue =
-      ExtractStatsValue(webrtc::StatsReport::kStatsReportTypeTrack,
+      ExtractStatsValue(StatsReport::kStatsReportTypeTrack,
                         reports,
-                        webrtc::StatsReport::kStatsValueNameTrackId);
+                        StatsReport::kStatsValueNameTrackId);
   EXPECT_EQ(kTrackId, trackValue);
 }
 
@@ -482,7 +603,7 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
 
   stats.set_session(&session_);
 
-  webrtc::StatsReports reports;
+  StatsReports reports;
 
   // Constructs an ssrc stats update.
   cricket::VideoSenderInfo video_sender_info;
@@ -494,8 +615,8 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
   video_sender_info.bytes_sent = kBytesSent;
   stats_read.senders.push_back(video_sender_info);
 
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
   EXPECT_CALL(*media_channel, GetStats(_, _))
     .WillOnce(DoAll(SetArgPointee<1>(stats_read),
                     Return(true)));
@@ -505,8 +626,8 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
   // |reports| should contain at least one session report, one track report,
   // and one ssrc report.
   EXPECT_LE((size_t)3, reports.size());
-  const webrtc::StatsReport* track_report = FindNthReportByType(
-      reports, webrtc::StatsReport::kStatsReportTypeTrack, 1);
+  const StatsReport* track_report = FindNthReportByType(
+      reports, StatsReport::kStatsReportTypeTrack, 1);
   EXPECT_FALSE(track_report == NULL);
 
   stats.GetStats(track_, &reports);
@@ -514,15 +635,15 @@ TEST_F(StatsCollectorTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
   // and one ssrc report.
   EXPECT_LE((size_t)3, reports.size());
   track_report = FindNthReportByType(
-      reports, webrtc::StatsReport::kStatsReportTypeTrack, 1);
+      reports, StatsReport::kStatsReportTypeTrack, 1);
   EXPECT_FALSE(track_report == NULL);
 
   std::string ssrc_id = ExtractSsrcStatsValue(
-      reports, webrtc::StatsReport::kStatsValueNameSsrc);
+      reports, StatsReport::kStatsValueNameSsrc);
   EXPECT_EQ(talk_base::ToString<uint32>(kSsrcOfTrack), ssrc_id);
 
   std::string track_id = ExtractSsrcStatsValue(
-      reports, webrtc::StatsReport::kStatsValueNameTrackId);
+      reports, StatsReport::kStatsValueNameTrackId);
   EXPECT_EQ(kTrackId, track_id);
 }
 
@@ -540,7 +661,7 @@ TEST_F(StatsCollectorTest, TransportObjectLinkedFromSsrcObject) {
 
   stats.set_session(&session_);
 
-  webrtc::StatsReports reports;
+  StatsReports reports;
 
   // Constructs an ssrc stats update.
   cricket::VideoSenderInfo video_sender_info;
@@ -552,26 +673,26 @@ TEST_F(StatsCollectorTest, TransportObjectLinkedFromSsrcObject) {
   video_sender_info.bytes_sent = kBytesSent;
   stats_read.senders.push_back(video_sender_info);
 
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
   EXPECT_CALL(*media_channel, GetStats(_, _))
     .WillRepeatedly(DoAll(SetArgPointee<1>(stats_read),
                           Return(true)));
 
   InitSessionStats(kVcName);
   EXPECT_CALL(session_, GetStats(_))
-    .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
-                          Return(true)));
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
+                            Return(true)));
 
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
   std::string transport_id = ExtractStatsValue(
-      webrtc::StatsReport::kStatsReportTypeSsrc,
+      StatsReport::kStatsReportTypeSsrc,
       reports,
-      webrtc::StatsReport::kStatsValueNameTransportId);
+      StatsReport::kStatsValueNameTransportId);
   ASSERT_NE(kNotFound, transport_id);
-  const webrtc::StatsReport* transport_report = FindReportById(reports,
-                                                               transport_id);
+  const StatsReport* transport_report = FindReportById(reports,
+                                                       transport_id);
   ASSERT_FALSE(transport_report == NULL);
 }
 
@@ -589,14 +710,14 @@ TEST_F(StatsCollectorTest, RemoteSsrcInfoIsAbsent) {
 
   stats.set_session(&session_);
 
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
 
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
-  webrtc::StatsReports reports;
+  StatsReports reports;
   stats.GetStats(NULL, &reports);
-  const webrtc::StatsReport* remote_report = FindNthReportByType(reports,
-      webrtc::StatsReport::kStatsReportTypeRemoteSsrc, 1);
+  const StatsReport* remote_report = FindNthReportByType(reports,
+      StatsReport::kStatsReportTypeRemoteSsrc, 1);
   EXPECT_TRUE(remote_report == NULL);
 }
 
@@ -614,13 +735,13 @@ TEST_F(StatsCollectorTest, RemoteSsrcInfoIsPresent) {
 
   stats.set_session(&session_);
 
-  webrtc::StatsReports reports;
+  StatsReports reports;
 
   // Instruct the session to return stats containing the transport channel.
   InitSessionStats(kVcName);
   EXPECT_CALL(session_, GetStats(_))
-    .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
-                          Return(true)));
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
+                            Return(true)));
 
   // Constructs an ssrc stats update.
   cricket::VideoMediaInfo stats_read;
@@ -633,16 +754,16 @@ TEST_F(StatsCollectorTest, RemoteSsrcInfoIsPresent) {
   video_sender_info.remote_stats.push_back(remote_ssrc_stats);
   stats_read.senders.push_back(video_sender_info);
 
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
   EXPECT_CALL(*media_channel, GetStats(_, _))
     .WillRepeatedly(DoAll(SetArgPointee<1>(stats_read),
                           Return(true)));
 
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
-  const webrtc::StatsReport* remote_report = FindNthReportByType(reports,
-      webrtc::StatsReport::kStatsReportTypeRemoteSsrc, 1);
+  const StatsReport* remote_report = FindNthReportByType(reports,
+      StatsReport::kStatsReportTypeRemoteSsrc, 1);
   EXPECT_FALSE(remote_report == NULL);
   EXPECT_NE(0, remote_report->timestamp);
 }
@@ -689,7 +810,7 @@ TEST_F(StatsCollectorTest, ChainlessCertificateReportsCreated) {
 // transport is present.
 TEST_F(StatsCollectorTest, NoTransport) {
   webrtc::StatsCollector stats;  // Implementation under test.
-  webrtc::StatsReports reports;  // returned values.
+  StatsReports reports;  // returned values.
   stats.set_session(&session_);
 
   // Fake stats to process.
@@ -711,24 +832,24 @@ TEST_F(StatsCollectorTest, NoTransport) {
     .WillOnce(DoAll(SetArgPointee<0>(session_stats),
                     Return(true)));
 
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
 
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
 
   // Check that the local certificate is absent.
   std::string local_certificate_id = ExtractStatsValue(
-      webrtc::StatsReport::kStatsReportTypeComponent,
+      StatsReport::kStatsReportTypeComponent,
       reports,
-      webrtc::StatsReport::kStatsValueNameLocalCertificateId);
+      StatsReport::kStatsValueNameLocalCertificateId);
   ASSERT_EQ(kNotFound, local_certificate_id);
 
   // Check that the remote certificate is absent.
   std::string remote_certificate_id = ExtractStatsValue(
-      webrtc::StatsReport::kStatsReportTypeComponent,
+      StatsReport::kStatsReportTypeComponent,
       reports,
-      webrtc::StatsReport::kStatsValueNameRemoteCertificateId);
+      StatsReport::kStatsValueNameRemoteCertificateId);
   ASSERT_EQ(kNotFound, remote_certificate_id);
 }
 
@@ -736,7 +857,7 @@ TEST_F(StatsCollectorTest, NoTransport) {
 // does not have any certificates.
 TEST_F(StatsCollectorTest, NoCertificates) {
   webrtc::StatsCollector stats;  // Implementation under test.
-  webrtc::StatsReports reports;  // returned values.
+  StatsReports reports;  // returned values.
   stats.set_session(&session_);
 
   // Fake stats to process.
@@ -764,24 +885,24 @@ TEST_F(StatsCollectorTest, NoCertificates) {
   EXPECT_CALL(session_, GetStats(_))
     .WillOnce(DoAll(SetArgPointee<0>(session_stats),
                     Return(true)));
-  EXPECT_CALL(session_, video_channel())
-    .WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
 
   stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
   stats.GetStats(NULL, &reports);
 
   // Check that the local certificate is absent.
   std::string local_certificate_id = ExtractStatsValue(
-      webrtc::StatsReport::kStatsReportTypeComponent,
+      StatsReport::kStatsReportTypeComponent,
       reports,
-      webrtc::StatsReport::kStatsValueNameLocalCertificateId);
+      StatsReport::kStatsValueNameLocalCertificateId);
   ASSERT_EQ(kNotFound, local_certificate_id);
 
   // Check that the remote certificate is absent.
   std::string remote_certificate_id = ExtractStatsValue(
-      webrtc::StatsReport::kStatsReportTypeComponent,
+      StatsReport::kStatsReportTypeComponent,
       reports,
-      webrtc::StatsReport::kStatsValueNameRemoteCertificateId);
+      StatsReport::kStatsValueNameRemoteCertificateId);
   ASSERT_EQ(kNotFound, remote_certificate_id);
 }
 
@@ -810,7 +931,7 @@ TEST_F(StatsCollectorTest, StatsOutputLevelVerbose) {
       media_engine_, media_channel, &session_, "", false, NULL);
   stats.set_session(&session_);
 
-  webrtc::StatsReports reports;  // returned values.
+  StatsReports reports;  // returned values.
   cricket::VideoMediaInfo stats_read;
   cricket::BandwidthEstimationInfo bwe;
   bwe.total_received_propagation_delta_ms = 10;
@@ -822,6 +943,7 @@ TEST_F(StatsCollectorTest, StatsOutputLevelVerbose) {
 
   EXPECT_CALL(session_, video_channel())
     .WillRepeatedly(Return(&video_channel));
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(ReturnNull());
 
   StatsOptions options;
   options.include_received_propagation_stats = true;
@@ -844,4 +966,175 @@ TEST_F(StatsCollectorTest, StatsOutputLevelVerbose) {
   EXPECT_EQ("[1000, 2000]", result);
 }
 
+// This test verifies that a local stats object can get statistics via
+// AudioTrackInterface::GetStats() method.
+TEST_F(StatsCollectorTest, GetStatsFromLocalAudioTrack) {
+  webrtc::StatsCollector stats;  // Implementation under test.
+  MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
+  // The content_name known by the voice channel.
+  const std::string kVcName("vcname");
+  cricket::VoiceChannel voice_channel(talk_base::Thread::Current(),
+      media_engine_, media_channel, &session_, kVcName, false);
+  AddLocalAudioTrackStats();
+  stats.AddStream(stream_);
+  stats.AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
+
+  stats.set_session(&session_);
+
+  // Instruct the session to return stats containing the transport channel.
+  InitSessionStats(kVcName);
+  EXPECT_CALL(session_, GetStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
+                            Return(true)));
+
+  cricket::VoiceSenderInfo voice_sender_info;
+  // Contents won't be modified by the AudioTrackInterface::GetStats().
+  voice_sender_info.add_ssrc(kSsrcOfTrack);
+  voice_sender_info.codec_name = "fake_codec";
+  voice_sender_info.bytes_sent = 100;
+  voice_sender_info.packets_sent = 101;
+  voice_sender_info.rtt_ms = 102;
+  voice_sender_info.fraction_lost = 103;
+  voice_sender_info.jitter_ms = 104;
+  voice_sender_info.packets_lost = 105;
+  voice_sender_info.ext_seqnum = 106;
+
+  // Contents will be modified by the AudioTrackInterface::GetStats().
+  voice_sender_info.audio_level = 107;
+  voice_sender_info.echo_return_loss = 108;;
+  voice_sender_info.echo_return_loss_enhancement = 109;
+  voice_sender_info.echo_delay_median_ms = 110;
+  voice_sender_info.echo_delay_std_ms = 111;
+  voice_sender_info.aec_quality_min = 112.0f;
+  voice_sender_info.typing_noise_detected = false;
+
+  // Constructs an ssrc stats update.
+  cricket::VoiceMediaInfo stats_read;
+  stats_read.senders.push_back(voice_sender_info);
+
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(Return(&voice_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(*media_channel, GetStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
+                            Return(true)));
+
+  StatsReports reports;  // returned values.
+  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
+  stats.GetStats(NULL, &reports);
+
+  // Verfy the existence of the track report.
+  const StatsReport* report = FindNthReportByType(
+      reports, StatsReport::kStatsReportTypeSsrc, 1);
+  EXPECT_FALSE(report == NULL);
+  std::string track_id = ExtractSsrcStatsValue(
+      reports, StatsReport::kStatsValueNameTrackId);
+  EXPECT_EQ(kAudioTrackId, track_id);
+  std::string ssrc_id = ExtractSsrcStatsValue(
+      reports, StatsReport::kStatsValueNameSsrc);
+  EXPECT_EQ(talk_base::ToString<uint32>(kSsrcOfTrack), ssrc_id);
+
+  // Verifies the values in the track report.
+  audio_track_->GetSignalLevel(&voice_sender_info.audio_level);
+  webrtc::AudioProcessorInterface::AudioProcessorStats audio_processor_stats;
+  audio_track_->GetAudioProcessor()->GetStats(&audio_processor_stats);
+  voice_sender_info.typing_noise_detected =
+      audio_processor_stats.typing_noise_detected;
+  voice_sender_info.echo_return_loss = audio_processor_stats.echo_return_loss;
+  voice_sender_info.echo_return_loss_enhancement =
+      audio_processor_stats.echo_return_loss_enhancement;
+  voice_sender_info.echo_delay_median_ms =
+      audio_processor_stats.echo_delay_median_ms;
+  voice_sender_info.aec_quality_min = audio_processor_stats.aec_quality_min;
+  voice_sender_info.echo_delay_std_ms = audio_processor_stats.echo_delay_std_ms;
+  VerifyVoiceSenderInfoReport(report, voice_sender_info);
+
+  // Verify we get the same result by passing a track to GetStats().
+  StatsReports track_reports;  // returned values.
+  stats.GetStats(audio_track_.get(), &track_reports);
+  const StatsReport* track_report = FindNthReportByType(
+      track_reports, StatsReport::kStatsReportTypeSsrc, 1);
+  EXPECT_FALSE(track_report == NULL);
+  track_id = ExtractSsrcStatsValue(track_reports,
+                                   StatsReport::kStatsValueNameTrackId);
+  EXPECT_EQ(kAudioTrackId, track_id);
+  ssrc_id = ExtractSsrcStatsValue(track_reports,
+                                  StatsReport::kStatsValueNameSsrc);
+  EXPECT_EQ(talk_base::ToString<uint32>(kSsrcOfTrack), ssrc_id);
+  VerifyVoiceSenderInfoReport(track_report, voice_sender_info);
+}
+
+// This test verifies that a local stats object won't update its statistics
+// after a RemoveLocalAudioTrack() call.
+TEST_F(StatsCollectorTest, GetStatsAfterRemoveAudioStream) {
+  webrtc::StatsCollector stats;  // Implementation under test.
+  MockVoiceMediaChannel* media_channel = new MockVoiceMediaChannel();
+  // The content_name known by the voice channel.
+  const std::string kVcName("vcname");
+  cricket::VoiceChannel voice_channel(talk_base::Thread::Current(),
+      media_engine_, media_channel, &session_, kVcName, false);
+  AddLocalAudioTrackStats();
+  stats.AddStream(stream_);
+  stats.AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
+
+  stats.set_session(&session_);
+
+  // Instruct the session to return stats containing the transport channel.
+  InitSessionStats(kVcName);
+  EXPECT_CALL(session_, GetStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats_),
+                            Return(true)));
+
+  stats.RemoveLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
+  cricket::VoiceSenderInfo voice_sender_info;
+  // Contents won't be modified by the AudioTrackInterface::GetStats().
+  voice_sender_info.add_ssrc(kSsrcOfTrack);
+  voice_sender_info.codec_name = "fake_codec";
+  voice_sender_info.bytes_sent = 100;
+  voice_sender_info.packets_sent = 101;
+  voice_sender_info.rtt_ms = 102;
+  voice_sender_info.fraction_lost = 103;
+  voice_sender_info.jitter_ms = 104;
+  voice_sender_info.packets_lost = 105;
+  voice_sender_info.ext_seqnum = 106;
+
+  // Contents will be modified by the AudioTrackInterface::GetStats().
+  voice_sender_info.audio_level = 107;
+  voice_sender_info.echo_return_loss = 108;;
+  voice_sender_info.echo_return_loss_enhancement = 109;
+  voice_sender_info.echo_delay_median_ms = 110;
+  voice_sender_info.echo_delay_std_ms = 111;
+  voice_sender_info.aec_quality_min = 112;
+  voice_sender_info.typing_noise_detected = false;
+
+  // Constructs an ssrc stats update.
+  cricket::VoiceMediaInfo stats_read;
+  stats_read.senders.push_back(voice_sender_info);
+
+  EXPECT_CALL(session_, voice_channel()).WillRepeatedly(Return(&voice_channel));
+  EXPECT_CALL(session_, video_channel()).WillRepeatedly(ReturnNull());
+  EXPECT_CALL(*media_channel, GetStats(_))
+      .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
+                            Return(true)));
+
+  StatsReports reports;  // returned values.
+  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
+  stats.GetStats(NULL, &reports);
+
+  // The report will exist since we don't remove them in RemoveStream().
+  const StatsReport* report = FindNthReportByType(
+      reports, StatsReport::kStatsReportTypeSsrc, 1);
+  EXPECT_FALSE(report == NULL);
+  std::string track_id = ExtractSsrcStatsValue(
+      reports, StatsReport::kStatsValueNameTrackId);
+  EXPECT_EQ(kAudioTrackId, track_id);
+  std::string ssrc_id = ExtractSsrcStatsValue(
+      reports, StatsReport::kStatsValueNameSsrc);
+  EXPECT_EQ(talk_base::ToString<uint32>(kSsrcOfTrack), ssrc_id);
+
+  // Verifies the values in the track report, no value will be changed by the
+  // AudioTrackInterface::GetSignalValue() and
+  // AudioProcessorInterface::AudioProcessorStats::GetStats();
+  VerifyVoiceSenderInfoReport(report, voice_sender_info);
+}
+
 }  // namespace
diff --git a/talk/app/webrtc/statstypes.h b/talk/app/webrtc/statstypes.h
index 2e7fb8127..c15dfaaed 100644
--- a/talk/app/webrtc/statstypes.h
+++ b/talk/app/webrtc/statstypes.h
@@ -57,6 +57,8 @@ class StatsReport {
   void AddValue(const std::string& name, const std::vector<T>& value);
   void AddBoolean(const std::string& name, bool value);
 
+  void ReplaceValue(const std::string& name, const std::string& value);
+
   double timestamp;  // Time since 1970-01-01T00:00:00Z in milliseconds.
   typedef std::vector<Value> Values;
   Values values;
diff --git a/talk/app/webrtc/test/fakemediastreamsignaling.h b/talk/app/webrtc/test/fakemediastreamsignaling.h
index f0c19cf6d..c7b30aaa5 100644
--- a/talk/app/webrtc/test/fakemediastreamsignaling.h
+++ b/talk/app/webrtc/test/fakemediastreamsignaling.h
@@ -122,7 +122,8 @@ class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
 
   virtual void OnRemoveLocalAudioTrack(
       webrtc::MediaStreamInterface* stream,
-      webrtc::AudioTrackInterface* audio_track) {
+      webrtc::AudioTrackInterface* audio_track,
+      uint32 ssrc) {
   }
   virtual void OnRemoveLocalVideoTrack(
       webrtc::MediaStreamInterface* stream,
diff --git a/talk/examples/call/call_main.cc b/talk/examples/call/call_main.cc
index 2ee796b1c..25c33ff0e 100644
--- a/talk/examples/call/call_main.cc
+++ b/talk/examples/call/call_main.cc
@@ -45,9 +45,6 @@
 #include "talk/examples/call/console.h"
 #include "talk/examples/call/mediaenginefactory.h"
 #include "talk/p2p/base/constants.h"
-#ifdef ANDROID
-#include "talk/media/other/androidmediaengine.h"
-#endif
 #include "talk/session/media/mediasessionclient.h"
 #include "talk/session/media/srtpfilter.h"
 #include "talk/xmpp/xmppauth.h"
@@ -185,7 +182,7 @@ static const int DEFAULT_PORT = 5222;
 static std::vector<cricket::AudioCodec> codecs;
 static const cricket::AudioCodec ISAC(103, "ISAC", 40000, 16000, 1, 0);
 
-cricket::MediaEngine *AndroidMediaEngineFactory() {
+cricket::MediaEngineInterface *CreateAndroidMediaEngine() {
     cricket::FakeMediaEngine *engine = new cricket::FakeMediaEngine();
 
     codecs.push_back(ISAC);
@@ -438,7 +435,7 @@ int main(int argc, char **argv) {
   }
 
 #ifdef ANDROID
-  InitAndroidMediaEngineFactory(AndroidMediaEngineFactory);
+  MediaEngineFactory::SetCreateFunction(&CreateAndroidMediaEngine);
 #endif
 
 #if WIN32
diff --git a/talk/media/base/mediaengine.cc b/talk/media/base/mediaengine.cc
index 021cf81f1..289f2290a 100644
--- a/talk/media/base/mediaengine.cc
+++ b/talk/media/base/mediaengine.cc
@@ -42,6 +42,13 @@ const int MediaEngineInterface::kDefaultAudioDelayOffset = 0;
 #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
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif  // HAVE_CONFIG
 
 namespace cricket {
 #if defined(HAVE_WEBRTC_VOICE)
@@ -51,22 +58,59 @@ namespace cricket {
 #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;
+};
+
 MediaEngineInterface* MediaEngineFactory::Create() {
+  if (create_function_) {
+    return create_function_();
+  } else {
 #if defined(HAVE_LINPHONE)
-  return new LinphoneMediaEngine("", "");
+    return new LinphoneMediaEngine("", "");
 #elif defined(AUDIO_ENG_NAME) && defined(VIDEO_ENG_NAME)
-  return new CompositeMediaEngine<AUDIO_ENG_NAME, VIDEO_ENG_NAME>();
+    return new CompositeMediaEngine<AUDIO_ENG_NAME, VIDEO_ENG_NAME>();
 #else
-  return new NullMediaEngine();
+    return new NullMediaEngine();
 #endif
+  }
 }
 
 };  // namespace cricket
diff --git a/talk/media/base/mediaengine.h b/talk/media/base/mediaengine.h
index 93586bb5b..bccae9951 100644
--- a/talk/media/base/mediaengine.h
+++ b/talk/media/base/mediaengine.h
@@ -157,7 +157,18 @@ class MediaEngineInterface {
 #if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
 class MediaEngineFactory {
  public:
+  typedef cricket::MediaEngineInterface* (*MediaEngineCreateFunction)();
+  // Creates a media engine, using either the compiled system default or the
+  // creation function specified in SetCreateFunction, if specified.
   static MediaEngineInterface* Create();
+  // Sets the function used when calling Create. If unset, the compiled system
+  // default will be used. Returns the old create function, or NULL if one
+  // wasn't set. Likewise, NULL can be used as the |function| parameter to
+  // reset to the default behavior.
+  static MediaEngineCreateFunction SetCreateFunction(
+      MediaEngineCreateFunction function);
+ private:
+  static MediaEngineCreateFunction create_function_;
 };
 #endif
 
diff --git a/talk/media/other/androidmediaengine.cc b/talk/media/other/androidmediaengine.cc
new file mode 100644
index 000000000..e69de29bb
diff --git a/talk/media/other/androidmediaengine.h b/talk/media/other/androidmediaengine.h
new file mode 100644
index 000000000..e69de29bb