Update Chromium to branch 1650 (31.0.1650.63)

Change-Id: I57d8c832eaec1eb2364e0a8e7352a6dd354db99f
Reviewed-by: Jocelyn Turcotte <jocelyn.turcotte@digia.com>

195 files changed, 3918 insertions, 6790 deletions

diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc
index 2b45845756f..9409fd7a210 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.cc
@@ -98,6 +98,7 @@ bool DataChannel::HasNegotiationCompleted() {
 DataChannel::~DataChannel() {
   ClearQueuedReceivedData();
   ClearQueuedSendData();
+  ClearQueuedControlData();
 }
 
 void DataChannel::RegisterObserver(DataChannelObserver* observer) {
@@ -193,7 +194,8 @@ bool DataChannel::SendControl(const talk_base::Buffer* buffer) {
 void DataChannel::SetReceiveSsrc(uint32 receive_ssrc) {
   if (receive_ssrc_set_) {
     ASSERT(session_->data_channel_type() == cricket::DCT_RTP ||
-        receive_ssrc_ == send_ssrc_);
+           !send_ssrc_set_ ||
+           receive_ssrc_ == send_ssrc_);
     return;
   }
   receive_ssrc_ = receive_ssrc;
@@ -209,7 +211,8 @@ void DataChannel::RemotePeerRequestClose() {
 void DataChannel::SetSendSsrc(uint32 send_ssrc) {
   if (send_ssrc_set_) {
     ASSERT(session_->data_channel_type() == cricket::DCT_RTP ||
-        receive_ssrc_ == send_ssrc_);
+           !receive_ssrc_set_ ||
+           receive_ssrc_ == send_ssrc_);
     return;
   }
   send_ssrc_ = send_ssrc;
@@ -250,14 +253,16 @@ void DataChannel::OnChannelReady(bool writable) {
   if (!writable) {
     return;
   }
-  // Update the readyState if the channel is writable for the first time;
-  // otherwise it means the channel was blocked for sending and now unblocked,
-  // so send the queued data now.
+  // Update the readyState and send the queued control message if the channel
+  // is writable for the first time; otherwise it means the channel was blocked
+  // for sending and now unblocked, so send the queued data now.
   if (!was_ever_writable_) {
     was_ever_writable_ = true;
     UpdateState();
+    DeliverQueuedControlData();
+    ASSERT(queued_send_data_.empty());
   } else if (state_ == kOpen) {
-    SendQueuedSendData();
+    DeliverQueuedSendData();
   }
 }
 
@@ -321,15 +326,13 @@ void DataChannel::ConnectToDataSession() {
   data_session_->SignalDataReceived.connect(this, &DataChannel::OnDataReceived);
   cricket::StreamParams params =
     cricket::StreamParams::CreateLegacy(id());
-  data_session_->media_channel()->AddSendStream(params);
-  data_session_->media_channel()->AddRecvStream(params);
+  data_session_->AddRecvStream(params);
+  data_session_->AddSendStream(params);
 }
 
 void DataChannel::DisconnectFromDataSession() {
-  if (data_session_->media_channel() != NULL) {
-    data_session_->media_channel()->RemoveSendStream(id());
-    data_session_->media_channel()->RemoveRecvStream(id());
-  }
+  data_session_->RemoveSendStream(id());
+  data_session_->RemoveRecvStream(id());
   data_session_->SignalReadyToSendData.disconnect(this);
   data_session_->SignalDataReceived.disconnect(this);
   data_session_ = NULL;
@@ -356,7 +359,7 @@ void DataChannel::ClearQueuedReceivedData() {
   }
 }
 
-void DataChannel::SendQueuedSendData() {
+void DataChannel::DeliverQueuedSendData() {
   DeliverQueuedControlData();
   if (!was_ever_writable_) {
     return;
@@ -366,7 +369,7 @@ void DataChannel::SendQueuedSendData() {
     DataBuffer* buffer = queued_send_data_.front();
     cricket::SendDataResult send_result;
     if (!InternalSendWithoutQueueing(*buffer, &send_result)) {
-      LOG(LS_WARNING) << "SendQueuedSendData aborted due to send_result "
+      LOG(LS_WARNING) << "DeliverQueuedSendData aborted due to send_result "
                       << send_result;
       break;
     }
@@ -375,6 +378,14 @@ void DataChannel::SendQueuedSendData() {
   }
 }
 
+void DataChannel::ClearQueuedControlData() {
+  while (!queued_control_data_.empty()) {
+    const talk_base::Buffer *buf = queued_control_data_.front();
+    queued_control_data_.pop();
+    delete buf;
+  }
+}
+
 void DataChannel::DeliverQueuedControlData() {
   if (was_ever_writable_) {
     while (!queued_control_data_.empty()) {
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h
index 28fe3446d5d..3ce3c1b5b4e 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel.h
@@ -67,8 +67,17 @@ class DataChannel : public DataChannelInterface,
   virtual void RegisterObserver(DataChannelObserver* observer);
   virtual void UnregisterObserver();
 
-  virtual std::string label() const  { return label_; }
+  virtual std::string label() const { return label_; }
   virtual bool reliable() const;
+  virtual bool ordered() const { return config_.ordered; }
+  virtual uint16 maxRetransmitTime() const {
+    return config_.maxRetransmitTime;
+  }
+  virtual uint16 maxRetransmits() const {
+    return config_.maxRetransmits;
+  }
+  virtual std::string protocol() const { return config_.protocol; }
+  virtual bool negotiated() const { return config_.negotiated; }
   virtual int id() const { return config_.id; }
   virtual uint64 buffered_amount() const;
   virtual void Close();
@@ -116,9 +125,10 @@ class DataChannel : public DataChannelInterface,
   bool IsConnectedToDataSession() { return data_session_ != NULL; }
   void DeliverQueuedControlData();
   void QueueControl(const talk_base::Buffer* buffer);
+  void ClearQueuedControlData();
   void DeliverQueuedReceivedData();
   void ClearQueuedReceivedData();
-  void SendQueuedSendData();
+  void DeliverQueuedSendData();
   void ClearQueuedSendData();
   bool InternalSendWithoutQueueing(const DataBuffer& buffer,
                                    cricket::SendDataResult* send_result);
@@ -158,6 +168,11 @@ BEGIN_PROXY_MAP(DataChannel)
   PROXY_METHOD0(void, UnregisterObserver)
   PROXY_CONSTMETHOD0(std::string, label)
   PROXY_CONSTMETHOD0(bool, reliable)
+  PROXY_CONSTMETHOD0(bool, ordered)
+  PROXY_CONSTMETHOD0(uint16, maxRetransmitTime)
+  PROXY_CONSTMETHOD0(uint16, maxRetransmits)
+  PROXY_CONSTMETHOD0(std::string, protocol)
+  PROXY_CONSTMETHOD0(bool, negotiated)
   PROXY_CONSTMETHOD0(int, id)
   PROXY_CONSTMETHOD0(DataState, state)
   PROXY_CONSTMETHOD0(uint64, buffered_amount)
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc
index 69060b751d8..2b0a9fe5a65 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannel_unittest.cc
@@ -29,6 +29,7 @@
 #include "talk/app/webrtc/jsep.h"
 #include "talk/app/webrtc/mediastreamsignaling.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
+#include "talk/app/webrtc/test/fakedtlsidentityservice.h"
 #include "talk/app/webrtc/webrtcsession.h"
 #include "talk/base/gunit.h"
 #include "talk/media/base/fakemediaengine.h"
@@ -75,7 +76,7 @@ class SctpDataChannelTest : public testing::Test {
                                         talk_base::Thread::Current())),
         media_stream_signaling_(
             new webrtc::MediaStreamSignaling(talk_base::Thread::Current(),
-                                             NULL)),
+                                             NULL, channel_manager_.get())),
         session_(channel_manager_.get(),
                  talk_base::Thread::Current(),
                  talk_base::Thread::Current(),
@@ -92,20 +93,21 @@ class SctpDataChannelTest : public testing::Test {
     constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true);
     constraints.AddMandatory(MediaConstraintsInterface::kEnableSctpDataChannels,
                              true);
-    ASSERT_TRUE(session_.Initialize(&constraints, NULL));
+    ASSERT_TRUE(session_.Initialize(&constraints,
+                                    new FakeIdentityService()));
+    webrtc_data_channel_ = webrtc::DataChannel::Create(&session_, "test", NULL);
+    ASSERT_TRUE(media_stream_signaling_->AddDataChannel(webrtc_data_channel_));
+
     talk_base::scoped_refptr<CreateSessionDescriptionObserverForTest> observer
         = new CreateSessionDescriptionObserverForTest();
     session_.CreateOffer(observer.get(), NULL);
-    EXPECT_TRUE_WAIT(observer->description() != NULL, 1000);
+    EXPECT_TRUE_WAIT(observer->description() != NULL, 2000);
     ASSERT_TRUE(observer->description() != NULL);
     ASSERT_TRUE(session_.SetLocalDescription(observer->ReleaseDescription(),
                                              NULL));
-
-    webrtc_data_channel_ = webrtc::DataChannel::Create(&session_, "test", NULL);
     // Connect to the media channel.
     webrtc_data_channel_->SetSendSsrc(kFakeSsrc);
     webrtc_data_channel_->SetReceiveSsrc(kFakeSsrc);
-
     session_.data_channel()->SignalReadyToSendData(true);
   }
 
@@ -116,7 +118,6 @@ class SctpDataChannelTest : public testing::Test {
       session_.data_channel()->SignalReadyToSendData(true);
     }
   }
-
   cricket::FakeMediaEngine* media_engine_;
   cricket::FakeDataEngine* data_engine_;
   talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h
index 82d375c1962..7be8a50f5d4 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/datachannelinterface.h
@@ -110,6 +110,16 @@ class DataChannelInterface : public talk_base::RefCountInterface {
   // DataChannel object from other DataChannel objects.
   virtual std::string label() const = 0;
   virtual bool reliable() const = 0;
+
+  // TODO(tommyw): Remove these dummy implementations when all classes have
+  // implemented these APIs. They should all just return the values the
+  // DataChannel was created with.
+  virtual bool ordered() const { return false; }
+  virtual uint16 maxRetransmitTime() const { return 0; }
+  virtual uint16 maxRetransmits() const { return 0; }
+  virtual std::string protocol() const { return std::string(); }
+  virtual bool negotiated() const { return false; }
+
   virtual int id() const = 0;
   virtual DataState state() const = 0;
   // The buffered_amount returns the number of bytes of application data
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/dtmfsender_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/dtmfsender_unittest.cc
index e1c3be9b12b..a4835054ade 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/dtmfsender_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/dtmfsender_unittest.cc
@@ -206,9 +206,9 @@ class DtmfSenderTest : public testing::Test {
     while (it_ref != dtmf_queue_ref.end() && it != dtmf_queue.end()) {
       EXPECT_EQ(it_ref->code, it->code);
       EXPECT_EQ(it_ref->duration, it->duration);
-      // Allow ~20ms error.
-      EXPECT_GE(it_ref->gap, it->gap - 20);
-      EXPECT_LE(it_ref->gap, it->gap + 20);
+      // Allow ~100ms error.
+      EXPECT_GE(it_ref->gap, it->gap - 100);
+      EXPECT_LE(it_ref->gap, it->gap + 100);
       ++it_ref;
       ++it;
     }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/peerconnection_jni.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/peerconnection_jni.cc
deleted file mode 100644
index fa3ad1b451a..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ /dev/null
@@ -1,1551 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// Hints for future visitors:
-// This entire file is an implementation detail of the org.webrtc Java package,
-// the most interesting bits of which are org.webrtc.PeerConnection{,Factory}.
-// The layout of this file is roughly:
-// - various helper C++ functions & classes that wrap Java counterparts and
-//   expose a C++ interface that can be passed to the C++ PeerConnection APIs
-// - implementations of methods declared "static" in the Java package (named
-//   things like Java_org_webrtc_OMG_Can_This_Name_Be_Any_Longer, prescribed by
-//   the JNI spec).
-//
-// Lifecycle notes: objects are owned where they will be called; in other words
-// FooObservers are owned by C++-land, and user-callable objects (e.g.
-// PeerConnection and VideoTrack) are owned by Java-land.
-// When this file allocates C++ RefCountInterfaces it AddRef()s an artificial
-// ref simulating the jlong held in Java-land, and then Release()s the ref in
-// the respective free call.  Sometimes this AddRef is implicit in the
-// construction of a scoped_refptr<> which is then .release()d.
-// Any persistent (non-local) references from C++ to Java must be global or weak
-// (in which case they must be checked before use)!
-//
-// Exception notes: pretty much all JNI calls can throw Java exceptions, so each
-// call through a JNIEnv* pointer needs to be followed by an ExceptionCheck()
-// call.  In this file this is done in CHECK_EXCEPTION, making for much easier
-// debugging in case of failure (the alternative is to wait for control to
-// return to the Java frame that called code in this file, at which point it's
-// impossible to tell which JNI call broke).
-
-#include <jni.h>
-#undef JNIEXPORT
-#define JNIEXPORT __attribute__((visibility("default")))
-
-#include <limits>
-#include <map>
-
-#include "talk/app/webrtc/mediaconstraintsinterface.h"
-#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "talk/app/webrtc/videosourceinterface.h"
-#include "talk/base/logging.h"
-#include "talk/base/ssladapter.h"
-#include "talk/media/base/videocapturer.h"
-#include "talk/media/base/videorenderer.h"
-#include "talk/media/devices/videorendererfactory.h"
-#include "talk/media/webrtc/webrtcvideocapturer.h"
-#include "third_party/icu/source/common/unicode/unistr.h"
-#include "webrtc/system_wrappers/interface/trace.h"
-#include "webrtc/video_engine/include/vie_base.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-
-using icu::UnicodeString;
-using webrtc::AudioSourceInterface;
-using webrtc::AudioTrackInterface;
-using webrtc::AudioTrackVector;
-using webrtc::CreateSessionDescriptionObserver;
-using webrtc::DataBuffer;
-using webrtc::DataChannelInit;
-using webrtc::DataChannelInterface;
-using webrtc::DataChannelObserver;
-using webrtc::IceCandidateInterface;
-using webrtc::MediaConstraintsInterface;
-using webrtc::MediaSourceInterface;
-using webrtc::MediaStreamInterface;
-using webrtc::MediaStreamTrackInterface;
-using webrtc::PeerConnectionFactoryInterface;
-using webrtc::PeerConnectionInterface;
-using webrtc::PeerConnectionObserver;
-using webrtc::SessionDescriptionInterface;
-using webrtc::SetSessionDescriptionObserver;
-using webrtc::StatsObserver;
-using webrtc::StatsReport;
-using webrtc::VideoRendererInterface;
-using webrtc::VideoSourceInterface;
-using webrtc::VideoTrackInterface;
-using webrtc::VideoTrackVector;
-using webrtc::VideoRendererInterface;
-
-// Abort the process if |x| is false, emitting |msg|.
-#define CHECK(x, msg)                                                          \
-  if (x) {} else {                                                             \
-    LOG(LS_ERROR) << __FILE__ << ":" << __LINE__ << ": " << msg;               \
-    abort();                                                                   \
-  }
-// Abort the process if |jni| has a Java exception pending, emitting |msg|.
-#define CHECK_EXCEPTION(jni, msg)                                              \
-  if (0) {} else {                                                             \
-    if (jni->ExceptionCheck()) {                                               \
-      jni->ExceptionDescribe();                                                \
-      jni->ExceptionClear();                                                   \
-      CHECK(0, msg);                                                           \
-    }                                                                          \
-  }
-
-namespace {
-
-static JavaVM* g_jvm = NULL;  // Set in JNI_OnLoad().
-
-static pthread_once_t g_jni_ptr_once = PTHREAD_ONCE_INIT;
-static pthread_key_t g_jni_ptr;  // Key for per-thread JNIEnv* data.
-
-static void ThreadDestructor(void* unused) {
-  jint status = g_jvm->DetachCurrentThread();
-  CHECK(status == JNI_OK, "Failed to detach thread: " << status);
-}
-
-static void CreateJNIPtrKey() {
-  CHECK(!pthread_key_create(&g_jni_ptr, &ThreadDestructor),
-        "pthread_key_create");
-}
-
-// Deal with difference in signatures between Oracle's jni.h and Android's.
-static JNIEnv* AttachCurrentThreadIfNeeded() {
-  CHECK(!pthread_once(&g_jni_ptr_once, &CreateJNIPtrKey),
-        "pthread_once");
-  JNIEnv* jni = reinterpret_cast<JNIEnv*>(pthread_getspecific(g_jni_ptr));
-  if (jni == NULL) {
-#ifdef _JAVASOFT_JNI_H_  // Oracle's jni.h violates the JNI spec!
-    void* env;
-#else
-    JNIEnv* env;
-#endif
-    CHECK(!g_jvm->AttachCurrentThread(&env, NULL), "Failed to attach thread");
-    CHECK(env, "AttachCurrentThread handed back NULL!");
-    jni = reinterpret_cast<JNIEnv*>(env);
-    CHECK(!pthread_setspecific(g_jni_ptr, jni), "pthread_setspecific");
-  }
-  return jni;
-}
-
-// Android's FindClass() is trickier than usual because the app-specific
-// ClassLoader is not consulted when there is no app-specific frame on the
-// stack.  Consequently, we only look up classes once in JNI_OnLoad.
-// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
-class ClassReferenceHolder {
- public:
-  explicit ClassReferenceHolder(JNIEnv* jni) {
-    LoadClass(jni, "java/nio/ByteBuffer");
-    LoadClass(jni, "org/webrtc/AudioTrack");
-    LoadClass(jni, "org/webrtc/DataChannel");
-    LoadClass(jni, "org/webrtc/DataChannel$Buffer");
-    LoadClass(jni, "org/webrtc/DataChannel$Init");
-    LoadClass(jni, "org/webrtc/DataChannel$State");
-    LoadClass(jni, "org/webrtc/IceCandidate");
-    LoadClass(jni, "org/webrtc/MediaSource$State");
-    LoadClass(jni, "org/webrtc/MediaStream");
-    LoadClass(jni, "org/webrtc/MediaStreamTrack$State");
-    LoadClass(jni, "org/webrtc/PeerConnection$SignalingState");
-    LoadClass(jni, "org/webrtc/PeerConnection$IceConnectionState");
-    LoadClass(jni, "org/webrtc/PeerConnection$IceGatheringState");
-    LoadClass(jni, "org/webrtc/SessionDescription");
-    LoadClass(jni, "org/webrtc/SessionDescription$Type");
-    LoadClass(jni, "org/webrtc/StatsReport");
-    LoadClass(jni, "org/webrtc/StatsReport$Value");
-    LoadClass(jni, "org/webrtc/VideoRenderer$I420Frame");
-    LoadClass(jni, "org/webrtc/VideoTrack");
-  }
-
-  ~ClassReferenceHolder() {
-    CHECK(classes_.empty(), "Must call FreeReferences() before dtor!");
-  }
-
-  void FreeReferences(JNIEnv* jni) {
-    for (std::map<std::string, jclass>::const_iterator it = classes_.begin();
-         it != classes_.end(); ++it) {
-      jni->DeleteGlobalRef(it->second);
-    }
-    classes_.clear();
-  }
-
-  jclass GetClass(const std::string& name) {
-    std::map<std::string, jclass>::iterator it = classes_.find(name);
-    CHECK(it != classes_.end(), "Unexpected GetClass() call for: " << name);
-    return it->second;
-  }
-
- private:
-  void LoadClass(JNIEnv* jni, const std::string& name) {
-    jclass localRef = jni->FindClass(name.c_str());
-    CHECK_EXCEPTION(jni, "error during FindClass: " << name);
-    CHECK(localRef, name);
-    jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef));
-    CHECK_EXCEPTION(jni, "error during NewGlobalRef: " << name);
-    CHECK(globalRef, name);
-    bool inserted = classes_.insert(std::make_pair(name, globalRef)).second;
-    CHECK(inserted, "Duplicate class name: " << name);
-  }
-
-  std::map<std::string, jclass> classes_;
-};
-
-// Allocated in JNI_OnLoad(), freed in JNI_OnUnLoad().
-static ClassReferenceHolder* g_class_reference_holder = NULL;
-
-// JNIEnv-helper methods that CHECK success: no Java exception thrown and found
-// object/class/method/field is non-null.
-jmethodID GetMethodID(
-    JNIEnv* jni, jclass c, const std::string& name, const char* signature) {
-  jmethodID m = jni->GetMethodID(c, name.c_str(), signature);
-  CHECK_EXCEPTION(jni,
-                  "error during GetMethodID: " << name << ", " << signature);
-  CHECK(m, name << ", " << signature);
-  return m;
-}
-
-jmethodID GetStaticMethodID(
-    JNIEnv* jni, jclass c, const char* name, const char* signature) {
-  jmethodID m = jni->GetStaticMethodID(c, name, signature);
-  CHECK_EXCEPTION(jni,
-                  "error during GetStaticMethodID: "
-                  << name << ", " << signature);
-  CHECK(m, name << ", " << signature);
-  return m;
-}
-
-jfieldID GetFieldID(
-    JNIEnv* jni, jclass c, const char* name, const char* signature) {
-  jfieldID f = jni->GetFieldID(c, name, signature);
-  CHECK_EXCEPTION(jni, "error during GetFieldID");
-  CHECK(f, name << ", " << signature);
-  return f;
-}
-
-jclass FindClass(JNIEnv* jni, const char* name) {
-  return g_class_reference_holder->GetClass(name);
-}
-
-jclass GetObjectClass(JNIEnv* jni, jobject object) {
-  jclass c = jni->GetObjectClass(object);
-  CHECK_EXCEPTION(jni, "error during GetObjectClass");
-  CHECK(c, "");
-  return c;
-}
-
-jobject GetObjectField(JNIEnv* jni, jobject object, jfieldID id) {
-  jobject o = jni->GetObjectField(object, id);
-  CHECK_EXCEPTION(jni, "error during GetObjectField");
-  CHECK(o, "");
-  return o;
-}
-
-jstring GetStringField(JNIEnv* jni, jobject object, jfieldID id) {
-  return static_cast<jstring>(GetObjectField(jni, object, id));
-}
-
-jlong GetLongField(JNIEnv* jni, jobject object, jfieldID id) {
-  jlong l = jni->GetLongField(object, id);
-  CHECK_EXCEPTION(jni, "error during GetLongField");
-  return l;
-}
-
-jint GetIntField(JNIEnv* jni, jobject object, jfieldID id) {
-  jint i = jni->GetIntField(object, id);
-  CHECK_EXCEPTION(jni, "error during GetIntField");
-  return i;
-}
-
-bool GetBooleanField(JNIEnv* jni, jobject object, jfieldID id) {
-  jboolean b = jni->GetBooleanField(object, id);
-  CHECK_EXCEPTION(jni, "error during GetBooleanField");
-  return b;
-}
-
-jobject NewGlobalRef(JNIEnv* jni, jobject o) {
-  jobject ret = jni->NewGlobalRef(o);
-  CHECK_EXCEPTION(jni, "error during NewGlobalRef");
-  CHECK(ret, "");
-  return ret;
-}
-
-void DeleteGlobalRef(JNIEnv* jni, jobject o) {
-  jni->DeleteGlobalRef(o);
-  CHECK_EXCEPTION(jni, "error during DeleteGlobalRef");
-}
-
-// Given a jweak reference, allocate a (strong) local reference scoped to the
-// lifetime of this object if the weak reference is still valid, or NULL
-// otherwise.
-class WeakRef {
- public:
-  WeakRef(JNIEnv* jni, jweak ref)
-      : jni_(jni), obj_(jni_->NewLocalRef(ref)) {
-    CHECK_EXCEPTION(jni, "error during NewLocalRef");
-  }
-  ~WeakRef() {
-    if (obj_) {
-      jni_->DeleteLocalRef(obj_);
-      CHECK_EXCEPTION(jni_, "error during DeleteLocalRef");
-    }
-  }
-  jobject obj() { return obj_; }
-
- private:
-  JNIEnv* const jni_;
-  jobject const obj_;
-};
-
-// Given a local ref, take ownership of it and delete the ref when this goes out
-// of scope.
-template<class T>  // T is jclass, jobject, jintArray, etc.
-class ScopedLocalRef {
- public:
-  ScopedLocalRef(JNIEnv* jni, T obj)
-      : jni_(jni), obj_(obj) {}
-  ~ScopedLocalRef() {
-    jni_->DeleteLocalRef(obj_);
-  }
-  T operator*() const {
-    return obj_;
-  }
- private:
-  JNIEnv* jni_;
-  T obj_;
-};
-
-// Scoped holder for global Java refs.
-template<class T>  // T is jclass, jobject, jintArray, etc.
-class ScopedGlobalRef {
- public:
-  explicit ScopedGlobalRef(JNIEnv* jni, T obj)
-      : obj_(static_cast<T>(jni->NewGlobalRef(obj))) {}
-  ~ScopedGlobalRef() {
-    DeleteGlobalRef(AttachCurrentThreadIfNeeded(), obj_);
-  }
-  T operator*() const {
-    return obj_;
-  }
- private:
-  T obj_;
-};
-
-// Return the (singleton) Java Enum object corresponding to |index|;
-// |state_class_fragment| is something like "MediaSource$State".
-jobject JavaEnumFromIndex(
-    JNIEnv* jni, const std::string& state_class_fragment, int index) {
-  std::string state_class_name = "org/webrtc/" + state_class_fragment;
-  jclass state_class = FindClass(jni, state_class_name.c_str());
-  jmethodID state_values_id = GetStaticMethodID(
-      jni, state_class, "values", ("()[L" + state_class_name  + ";").c_str());
-  ScopedLocalRef<jobjectArray> state_values(
-      jni,
-      (jobjectArray)jni->CallStaticObjectMethod(state_class, state_values_id));
-  CHECK_EXCEPTION(jni, "error during CallStaticObjectMethod");
-  jobject ret = jni->GetObjectArrayElement(*state_values, index);
-  CHECK_EXCEPTION(jni, "error during GetObjectArrayElement");
-  return ret;
-}
-
-// Given a UTF-8 encoded |native| string return a new (UTF-16) jstring.
-static jstring JavaStringFromStdString(JNIEnv* jni, const std::string& native) {
-  UnicodeString ustr(UnicodeString::fromUTF8(native));
-  jstring jstr = jni->NewString(ustr.getBuffer(), ustr.length());
-  CHECK_EXCEPTION(jni, "error during NewString");
-  return jstr;
-}
-
-// Given a (UTF-16) jstring return a new UTF-8 native string.
-static std::string JavaToStdString(JNIEnv* jni, const jstring& j_string) {
-  const jchar* jchars = jni->GetStringChars(j_string, NULL);
-  CHECK_EXCEPTION(jni, "Error during GetStringChars");
-  UnicodeString ustr(jchars, jni->GetStringLength(j_string));
-  CHECK_EXCEPTION(jni, "Error during GetStringLength");
-  jni->ReleaseStringChars(j_string, jchars);
-  CHECK_EXCEPTION(jni, "Error during ReleaseStringChars");
-  std::string ret;
-  return ustr.toUTF8String(ret);
-}
-
-static DataChannelInit JavaDataChannelInitToNative(
-    JNIEnv* jni, jobject j_init) {
-  DataChannelInit init;
-
-  jclass j_init_class = FindClass(jni, "org/webrtc/DataChannel$Init");
-  jfieldID ordered_id = GetFieldID(jni, j_init_class, "ordered", "Z");
-  jfieldID max_retransmit_time_id =
-      GetFieldID(jni, j_init_class, "maxRetransmitTimeMs", "I");
-  jfieldID max_retransmits_id =
-      GetFieldID(jni, j_init_class, "maxRetransmits", "I");
-  jfieldID protocol_id =
-      GetFieldID(jni, j_init_class, "protocol", "Ljava/lang/String;");
-  jfieldID negotiated_id = GetFieldID(jni, j_init_class, "negotiated", "Z");
-  jfieldID id_id = GetFieldID(jni, j_init_class, "id", "I");
-
-  init.ordered = GetBooleanField(jni, j_init, ordered_id);
-  init.maxRetransmitTime = GetIntField(jni, j_init, max_retransmit_time_id);
-  init.maxRetransmits = GetIntField(jni, j_init, max_retransmits_id);
-  init.protocol = JavaToStdString(
-      jni, GetStringField(jni, j_init, protocol_id));
-  init.negotiated = GetBooleanField(jni, j_init, negotiated_id);
-  init.id = GetIntField(jni, j_init, id_id);
-
-  return init;
-}
-
-class ConstraintsWrapper;
-
-// Adapter between the C++ PeerConnectionObserver interface and the Java
-// PeerConnection.Observer interface.  Wraps an instance of the Java interface
-// and dispatches C++ callbacks to Java.
-class PCOJava : public PeerConnectionObserver {
- public:
-  PCOJava(JNIEnv* jni, jobject j_observer)
-      : j_observer_global_(jni, j_observer),
-        j_observer_class_(jni, GetObjectClass(jni, *j_observer_global_)),
-        j_media_stream_class_(jni, FindClass(jni, "org/webrtc/MediaStream")),
-        j_media_stream_ctor_(GetMethodID(
-            jni, *j_media_stream_class_, "<init>", "(J)V")),
-        j_audio_track_class_(jni, FindClass(jni, "org/webrtc/AudioTrack")),
-        j_audio_track_ctor_(GetMethodID(
-            jni, *j_audio_track_class_, "<init>", "(J)V")),
-        j_video_track_class_(jni, FindClass(jni, "org/webrtc/VideoTrack")),
-        j_video_track_ctor_(GetMethodID(
-            jni, *j_video_track_class_, "<init>", "(J)V")),
-        j_data_channel_class_(jni, FindClass(jni, "org/webrtc/DataChannel")),
-        j_data_channel_ctor_(GetMethodID(
-            jni, *j_data_channel_class_, "<init>", "(J)V")) {
-  }
-
-  virtual ~PCOJava() {}
-
-  virtual void OnIceCandidate(const IceCandidateInterface* candidate) OVERRIDE {
-    std::string sdp;
-    CHECK(candidate->ToString(&sdp), "got so far: " << sdp);
-    jclass candidate_class = FindClass(jni(), "org/webrtc/IceCandidate");
-    jmethodID ctor = GetMethodID(jni(), candidate_class,
-        "<init>", "(Ljava/lang/String;ILjava/lang/String;)V");
-    ScopedLocalRef<jstring> j_mid(
-        jni(), JavaStringFromStdString(jni(), candidate->sdp_mid()));
-    ScopedLocalRef<jstring> j_sdp(jni(), JavaStringFromStdString(jni(), sdp));
-    ScopedLocalRef<jobject> j_candidate(jni(), jni()->NewObject(
-        candidate_class, ctor, *j_mid, candidate->sdp_mline_index(), *j_sdp));
-    CHECK_EXCEPTION(jni(), "error during NewObject");
-    jmethodID m = GetMethodID(jni(), *j_observer_class_,
-                              "onIceCandidate", "(Lorg/webrtc/IceCandidate;)V");
-    jni()->CallVoidMethod(*j_observer_global_, m, *j_candidate);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnError() OVERRIDE {
-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onError", "(V)V");
-    jni()->CallVoidMethod(*j_observer_global_, m);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnSignalingChange(
-      PeerConnectionInterface::SignalingState new_state) OVERRIDE {
-    jmethodID m = GetMethodID(
-        jni(), *j_observer_class_, "onSignalingChange",
-        "(Lorg/webrtc/PeerConnection$SignalingState;)V");
-    ScopedLocalRef<jobject> new_state_enum(jni(), JavaEnumFromIndex(
-        jni(), "PeerConnection$SignalingState", new_state));
-    jni()->CallVoidMethod(*j_observer_global_, m, *new_state_enum);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnIceConnectionChange(
-      PeerConnectionInterface::IceConnectionState new_state) OVERRIDE {
-    jmethodID m = GetMethodID(
-        jni(), *j_observer_class_, "onIceConnectionChange",
-        "(Lorg/webrtc/PeerConnection$IceConnectionState;)V");
-    ScopedLocalRef<jobject> new_state_enum(jni(), JavaEnumFromIndex(
-        jni(), "PeerConnection$IceConnectionState", new_state));
-    jni()->CallVoidMethod(*j_observer_global_, m, *new_state_enum);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnIceGatheringChange(
-      PeerConnectionInterface::IceGatheringState new_state) OVERRIDE {
-    jmethodID m = GetMethodID(
-        jni(), *j_observer_class_, "onIceGatheringChange",
-        "(Lorg/webrtc/PeerConnection$IceGatheringState;)V");
-    ScopedLocalRef<jobject> new_state_enum(jni(), JavaEnumFromIndex(
-        jni(), "PeerConnection$IceGatheringState", new_state));
-    jni()->CallVoidMethod(*j_observer_global_, m, *new_state_enum);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnAddStream(MediaStreamInterface* stream) OVERRIDE {
-    ScopedLocalRef<jobject> j_stream(jni(), jni()->NewObject(
-        *j_media_stream_class_, j_media_stream_ctor_, (jlong)stream));
-    CHECK_EXCEPTION(jni(), "error during NewObject");
-
-    AudioTrackVector audio_tracks = stream->GetAudioTracks();
-    for (size_t i = 0; i < audio_tracks.size(); ++i) {
-      AudioTrackInterface* track = audio_tracks[i];
-      ScopedLocalRef<jstring> id(
-          jni(), JavaStringFromStdString(jni(), track->id()));
-      ScopedLocalRef<jobject> j_track(jni(), jni()->NewObject(
-          *j_audio_track_class_, j_audio_track_ctor_, (jlong)track, *id));
-      CHECK_EXCEPTION(jni(), "error during NewObject");
-      jfieldID audio_tracks_id = GetFieldID(
-          jni(), *j_media_stream_class_, "audioTracks", "Ljava/util/List;");
-      ScopedLocalRef<jobject> audio_tracks(jni(), GetObjectField(
-          jni(), *j_stream, audio_tracks_id));
-      jmethodID add = GetMethodID(jni(),
-          GetObjectClass(jni(), *audio_tracks), "add", "(Ljava/lang/Object;)Z");
-      jboolean added = jni()->CallBooleanMethod(*audio_tracks, add, *j_track);
-      CHECK_EXCEPTION(jni(), "error during CallBooleanMethod");
-      CHECK(added, "");
-    }
-
-    VideoTrackVector video_tracks = stream->GetVideoTracks();
-    for (size_t i = 0; i < video_tracks.size(); ++i) {
-      VideoTrackInterface* track = video_tracks[i];
-      ScopedLocalRef<jstring> id(
-          jni(), JavaStringFromStdString(jni(), track->id()));
-      ScopedLocalRef<jobject> j_track(jni(), jni()->NewObject(
-          *j_video_track_class_, j_video_track_ctor_, (jlong)track, *id));
-      CHECK_EXCEPTION(jni(), "error during NewObject");
-      jfieldID video_tracks_id = GetFieldID(
-          jni(), *j_media_stream_class_, "videoTracks", "Ljava/util/List;");
-      ScopedLocalRef<jobject> video_tracks(jni(), GetObjectField(
-          jni(), *j_stream, video_tracks_id));
-      jmethodID add = GetMethodID(jni(),
-          GetObjectClass(jni(), *video_tracks), "add", "(Ljava/lang/Object;)Z");
-      jboolean added = jni()->CallBooleanMethod(*video_tracks, add, *j_track);
-      CHECK_EXCEPTION(jni(), "error during CallBooleanMethod");
-      CHECK(added, "");
-    }
-    streams_[stream] = jni()->NewWeakGlobalRef(*j_stream);
-    CHECK_EXCEPTION(jni(), "error during NewWeakGlobalRef");
-
-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onAddStream",
-                              "(Lorg/webrtc/MediaStream;)V");
-    jni()->CallVoidMethod(*j_observer_global_, m, *j_stream);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnRemoveStream(MediaStreamInterface* stream) OVERRIDE {
-    NativeToJavaStreamsMap::iterator it = streams_.find(stream);
-    CHECK(it != streams_.end(), "unexpected stream: " << std::hex << stream);
-
-    WeakRef s(jni(), it->second);
-    streams_.erase(it);
-    if (!s.obj())
-      return;
-
-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onRemoveStream",
-                              "(Lorg/webrtc/MediaStream;)V");
-    jni()->CallVoidMethod(*j_observer_global_, m, s.obj());
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnDataChannel(DataChannelInterface* channel) OVERRIDE {
-    ScopedLocalRef<jobject> j_channel(jni(), jni()->NewObject(
-        *j_data_channel_class_, j_data_channel_ctor_, (jlong)channel));
-    CHECK_EXCEPTION(jni(), "error during NewObject");
-    // Channel is now owned by Java object, and will be freed from
-    // DataChannel.dispose().
-    channel->AddRef();
-
-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onDataChannel",
-                              "(Lorg/webrtc/DataChannel;)V");
-    jni()->CallVoidMethod(*j_observer_global_, m, *j_channel);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  void SetConstraints(ConstraintsWrapper* constraints) {
-    CHECK(!constraints_.get(), "constraints already set!");
-    constraints_.reset(constraints);
-  }
-
-  const ConstraintsWrapper* constraints() { return constraints_.get(); }
-
- private:
-  JNIEnv* jni() {
-    return AttachCurrentThreadIfNeeded();
-  }
-
-  const ScopedGlobalRef<jobject> j_observer_global_;
-  const ScopedGlobalRef<jclass> j_observer_class_;
-  const ScopedGlobalRef<jclass> j_media_stream_class_;
-  const jmethodID j_media_stream_ctor_;
-  const ScopedGlobalRef<jclass> j_audio_track_class_;
-  const jmethodID j_audio_track_ctor_;
-  const ScopedGlobalRef<jclass> j_video_track_class_;
-  const jmethodID j_video_track_ctor_;
-  const ScopedGlobalRef<jclass> j_data_channel_class_;
-  const jmethodID j_data_channel_ctor_;
-  typedef std::map<void*, jweak> NativeToJavaStreamsMap;
-  NativeToJavaStreamsMap streams_;  // C++ -> Java streams.
-  talk_base::scoped_ptr<ConstraintsWrapper> constraints_;
-};
-
-// Wrapper for a Java MediaConstraints object.  Copies all needed data so when
-// the constructor returns the Java object is no longer needed.
-class ConstraintsWrapper : public MediaConstraintsInterface {
- public:
-  ConstraintsWrapper(JNIEnv* jni, jobject j_constraints) {
-    PopulateConstraintsFromJavaPairList(
-        jni, j_constraints, "mandatory", &mandatory_);
-    PopulateConstraintsFromJavaPairList(
-        jni, j_constraints, "optional", &optional_);
-  }
-
-  virtual ~ConstraintsWrapper() {}
-
-  // MediaConstraintsInterface.
-  virtual const Constraints& GetMandatory() const OVERRIDE {
-    return mandatory_;
-  }
-
-  virtual const Constraints& GetOptional() const OVERRIDE {
-    return optional_;
-  }
-
- private:
-  // Helper for translating a List<Pair<String, String>> to a Constraints.
-  static void PopulateConstraintsFromJavaPairList(
-      JNIEnv* jni, jobject j_constraints,
-      const char* field_name, Constraints* field) {
-    jfieldID j_id = GetFieldID(jni,
-        GetObjectClass(jni, j_constraints), field_name, "Ljava/util/List;");
-    jobject j_list = GetObjectField(jni, j_constraints, j_id);
-    jmethodID j_iterator_id = GetMethodID(jni,
-        GetObjectClass(jni, j_list), "iterator", "()Ljava/util/Iterator;");
-    jobject j_iterator = jni->CallObjectMethod(j_list, j_iterator_id);
-    CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-    jmethodID j_has_next = GetMethodID(jni,
-        GetObjectClass(jni, j_iterator), "hasNext", "()Z");
-    jmethodID j_next = GetMethodID(jni,
-        GetObjectClass(jni, j_iterator), "next", "()Ljava/lang/Object;");
-    while (jni->CallBooleanMethod(j_iterator, j_has_next)) {
-      CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
-      jobject entry = jni->CallObjectMethod(j_iterator, j_next);
-      CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-      jmethodID get_key = GetMethodID(jni,
-          GetObjectClass(jni, entry), "getKey", "()Ljava/lang/String;");
-      jstring j_key = reinterpret_cast<jstring>(
-          jni->CallObjectMethod(entry, get_key));
-      CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-      jmethodID get_value = GetMethodID(jni,
-          GetObjectClass(jni, entry), "getValue", "()Ljava/lang/String;");
-      jstring j_value = reinterpret_cast<jstring>(
-          jni->CallObjectMethod(entry, get_value));
-      CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-      field->push_back(Constraint(JavaToStdString(jni, j_key),
-                                  JavaToStdString(jni, j_value)));
-    }
-    CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
-  }
-
-  Constraints mandatory_;
-  Constraints optional_;
-};
-
-static jobject JavaSdpFromNativeSdp(
-    JNIEnv* jni, const SessionDescriptionInterface* desc) {
-  std::string sdp;
-  CHECK(desc->ToString(&sdp), "got so far: " << sdp);
-  ScopedLocalRef<jstring> j_description(jni, JavaStringFromStdString(jni, sdp));
-
-  jclass j_type_class = FindClass(
-      jni, "org/webrtc/SessionDescription$Type");
-  jmethodID j_type_from_canonical = GetStaticMethodID(
-      jni, j_type_class, "fromCanonicalForm",
-      "(Ljava/lang/String;)Lorg/webrtc/SessionDescription$Type;");
-  ScopedLocalRef<jstring> j_type_string(
-      jni, JavaStringFromStdString(jni, desc->type()));
-  jobject j_type = jni->CallStaticObjectMethod(
-      j_type_class, j_type_from_canonical, *j_type_string);
-  CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-
-  jclass j_sdp_class = FindClass(jni, "org/webrtc/SessionDescription");
-  jmethodID j_sdp_ctor = GetMethodID(
-      jni, j_sdp_class, "<init>",
-      "(Lorg/webrtc/SessionDescription$Type;Ljava/lang/String;)V");
-  jobject j_sdp = jni->NewObject(
-      j_sdp_class, j_sdp_ctor, j_type, *j_description);
-  CHECK_EXCEPTION(jni, "error during NewObject");
-  return j_sdp;
-}
-
-template <class T>  // T is one of {Create,Set}SessionDescriptionObserver.
-class SdpObserverWrapper : public T {
- public:
-  SdpObserverWrapper(JNIEnv* jni, jobject j_observer,
-                     ConstraintsWrapper* constraints)
-      : constraints_(constraints),
-        j_observer_global_(jni, j_observer),
-        j_observer_class_(jni, GetObjectClass(jni, j_observer)) {
-  }
-
-  virtual ~SdpObserverWrapper() {}
-
-  // Can't mark OVERRIDE because of templating.
-  virtual void OnSuccess() {
-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onSetSuccess", "()V");
-    jni()->CallVoidMethod(*j_observer_global_, m);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  // Can't mark OVERRIDE because of templating.
-  virtual void OnSuccess(SessionDescriptionInterface* desc) {
-    jmethodID m = GetMethodID(
-        jni(), *j_observer_class_, "onCreateSuccess",
-        "(Lorg/webrtc/SessionDescription;)V");
-    ScopedLocalRef<jobject> j_sdp(jni(), JavaSdpFromNativeSdp(jni(), desc));
-    jni()->CallVoidMethod(*j_observer_global_, m, *j_sdp);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
- protected:
-  // Common implementation for failure of Set & Create types, distinguished by
-  // |op| being "Set" or "Create".
-  void OnFailure(const std::string& op, const std::string& error) {
-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "on" + op + "Failure",
-                              "(Ljava/lang/String;)V");
-    ScopedLocalRef<jstring> j_error_string(
-        jni(), JavaStringFromStdString(jni(), error));
-    jni()->CallVoidMethod(*j_observer_global_, m, *j_error_string);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
- private:
-  JNIEnv* jni() {
-    return AttachCurrentThreadIfNeeded();
-  }
-
-  talk_base::scoped_ptr<ConstraintsWrapper> constraints_;
-  const ScopedGlobalRef<jobject> j_observer_global_;
-  const ScopedGlobalRef<jclass> j_observer_class_;
-};
-
-class CreateSdpObserverWrapper
-    : public SdpObserverWrapper<CreateSessionDescriptionObserver> {
- public:
-  CreateSdpObserverWrapper(JNIEnv* jni, jobject j_observer,
-                           ConstraintsWrapper* constraints)
-      : SdpObserverWrapper(jni, j_observer, constraints) {}
-
-  virtual void OnFailure(const std::string& error) OVERRIDE {
-    SdpObserverWrapper::OnFailure(std::string("Create"), error);
-  }
-};
-
-class SetSdpObserverWrapper
-    : public SdpObserverWrapper<SetSessionDescriptionObserver> {
- public:
-  SetSdpObserverWrapper(JNIEnv* jni, jobject j_observer,
-                        ConstraintsWrapper* constraints)
-      : SdpObserverWrapper(jni, j_observer, constraints) {}
-
-  virtual void OnFailure(const std::string& error) OVERRIDE {
-    SdpObserverWrapper::OnFailure(std::string("Set"), error);
-  }
-};
-
-// Adapter for a Java DataChannel$Observer presenting a C++ DataChannelObserver
-// and dispatching the callback from C++ back to Java.
-class DataChannelObserverWrapper : public DataChannelObserver {
- public:
-  DataChannelObserverWrapper(JNIEnv* jni, jobject j_observer)
-      : j_observer_global_(jni, j_observer),
-        j_observer_class_(jni, GetObjectClass(jni, j_observer)),
-        j_on_state_change_mid_(GetMethodID(jni, *j_observer_class_,
-                                           "onStateChange", "()V")),
-        j_on_message_mid_(GetMethodID(jni, *j_observer_class_, "onMessage",
-                                      "(Lorg/webrtc/DataChannel$Buffer;)V")),
-        j_buffer_class_(jni, FindClass(jni, "org/webrtc/DataChannel$Buffer")),
-        j_buffer_ctor_(GetMethodID(jni, *j_buffer_class_,
-                                   "<init>", "(Ljava/nio/ByteBuffer;Z)V")) {
-  }
-
-  virtual ~DataChannelObserverWrapper() {}
-
-  virtual void OnStateChange() OVERRIDE {
-    jni()->CallVoidMethod(*j_observer_global_, j_on_state_change_mid_);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
-  virtual void OnMessage(const DataBuffer& buffer) OVERRIDE {
-    jobject byte_buffer =
-        jni()->NewDirectByteBuffer(const_cast<char*>(buffer.data.data()),
-                                   buffer.data.length());
-    jobject j_buffer = jni()->NewObject(*j_buffer_class_, j_buffer_ctor_,
-                                        byte_buffer, buffer.binary);
-    jni()->CallVoidMethod(*j_observer_global_, j_on_message_mid_, j_buffer);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
- private:
-  JNIEnv* jni() {
-    return AttachCurrentThreadIfNeeded();
-  }
-
-  const ScopedGlobalRef<jobject> j_observer_global_;
-  const ScopedGlobalRef<jclass> j_observer_class_;
-  const ScopedGlobalRef<jclass> j_buffer_class_;
-  const jmethodID j_on_state_change_mid_;
-  const jmethodID j_on_message_mid_;
-  const jmethodID j_buffer_ctor_;
-};
-
-// Adapter for a Java StatsObserver presenting a C++ StatsObserver and
-// dispatching the callback from C++ back to Java.
-class StatsObserverWrapper : public StatsObserver {
- public:
-  StatsObserverWrapper(JNIEnv* jni, jobject j_observer)
-      : j_observer_global_(jni, j_observer),
-        j_observer_class_(jni, GetObjectClass(jni, j_observer)),
-        j_stats_report_class_(jni, FindClass(jni, "org/webrtc/StatsReport")),
-        j_stats_report_ctor_(GetMethodID(
-            jni, *j_stats_report_class_, "<init>",
-            "(Ljava/lang/String;Ljava/lang/String;D"
-            "[Lorg/webrtc/StatsReport$Value;)V")),
-        j_value_class_(jni, FindClass(
-            jni, "org/webrtc/StatsReport$Value")),
-        j_value_ctor_(GetMethodID(
-            jni, *j_value_class_, "<init>",
-            "(Ljava/lang/String;Ljava/lang/String;)V")) {
-  }
-
-  virtual ~StatsObserverWrapper() {}
-
-  virtual void OnComplete(const std::vector<StatsReport>& reports) OVERRIDE {
-    ScopedLocalRef<jobjectArray> j_reports(jni(),
-                                           ReportsToJava(jni(), reports));
-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onComplete",
-                              "([Lorg/webrtc/StatsReport;)V");
-    jni()->CallVoidMethod(*j_observer_global_, m, *j_reports);
-    CHECK_EXCEPTION(jni(), "error during CallVoidMethod");
-  }
-
- private:
-  jobjectArray ReportsToJava(
-      JNIEnv* jni, const std::vector<StatsReport>& reports) {
-    jobjectArray reports_array = jni->NewObjectArray(
-        reports.size(), *j_stats_report_class_, NULL);
-    for (int i = 0; i < reports.size(); ++i) {
-      const StatsReport& report = reports[i];
-      ScopedLocalRef<jstring> j_id(
-          jni, JavaStringFromStdString(jni, report.id));
-      ScopedLocalRef<jstring> j_type(
-          jni, JavaStringFromStdString(jni, report.type));
-      ScopedLocalRef<jobjectArray> j_values(
-          jni, ValuesToJava(jni, report.values));
-      ScopedLocalRef<jobject> j_report(jni, jni->NewObject(
-          *j_stats_report_class_, j_stats_report_ctor_, *j_id, *j_type,
-          report.timestamp, *j_values));
-      jni->SetObjectArrayElement(reports_array, i, *j_report);
-    }
-    return reports_array;
-  }
-
-  jobjectArray ValuesToJava(JNIEnv* jni, const StatsReport::Values& values) {
-    jobjectArray j_values = jni->NewObjectArray(
-        values.size(), *j_value_class_, NULL);
-    for (int i = 0; i < values.size(); ++i) {
-      const StatsReport::Value& value = values[i];
-      ScopedLocalRef<jstring> j_name(
-          jni, JavaStringFromStdString(jni, value.name));
-      ScopedLocalRef<jstring> j_value(
-          jni, JavaStringFromStdString(jni, value.value));
-      ScopedLocalRef<jobject> j_element_value(jni, jni->NewObject(
-          *j_value_class_, j_value_ctor_, *j_name, *j_value));
-      jni->SetObjectArrayElement(j_values, i, *j_element_value);
-    }
-    return j_values;
-  }
-
-  JNIEnv* jni() {
-    return AttachCurrentThreadIfNeeded();
-  }
-
-  const ScopedGlobalRef<jobject> j_observer_global_;
-  const ScopedGlobalRef<jclass> j_observer_class_;
-  const ScopedGlobalRef<jclass> j_stats_report_class_;
-  const jmethodID j_stats_report_ctor_;
-  const ScopedGlobalRef<jclass> j_value_class_;
-  const jmethodID j_value_ctor_;
-};
-
-// Adapter presenting a cricket::VideoRenderer as a
-// webrtc::VideoRendererInterface.
-class VideoRendererWrapper : public VideoRendererInterface {
- public:
-  static VideoRendererWrapper* Create(cricket::VideoRenderer* renderer) {
-    if (renderer)
-      return new VideoRendererWrapper(renderer);
-    return NULL;
-  }
-
-  virtual ~VideoRendererWrapper() {}
-
-  virtual void SetSize(int width, int height) OVERRIDE {
-    const bool kNotReserved = false;  // What does this param mean??
-    renderer_->SetSize(width, height, kNotReserved);
-  }
-
-  virtual void RenderFrame(const cricket::VideoFrame* frame) OVERRIDE {
-    renderer_->RenderFrame(frame);
-  }
-
- private:
-  explicit VideoRendererWrapper(cricket::VideoRenderer* renderer)
-      : renderer_(renderer) {}
-
-  talk_base::scoped_ptr<cricket::VideoRenderer> renderer_;
-};
-
-// Wrapper dispatching webrtc::VideoRendererInterface to a Java VideoRenderer
-// instance.
-class JavaVideoRendererWrapper : public VideoRendererInterface {
- public:
-  JavaVideoRendererWrapper(JNIEnv* jni, jobject j_callbacks)
-      : j_callbacks_(jni, j_callbacks),
-        j_set_size_id_(GetMethodID(
-            jni, GetObjectClass(jni, j_callbacks), "setSize", "(II)V")),
-        j_render_frame_id_(GetMethodID(
-            jni, GetObjectClass(jni, j_callbacks), "renderFrame",
-            "(Lorg/webrtc/VideoRenderer$I420Frame;)V")),
-        j_frame_class_(jni,
-                       FindClass(jni, "org/webrtc/VideoRenderer$I420Frame")),
-        j_frame_ctor_id_(GetMethodID(
-            jni, *j_frame_class_, "<init>", "(II[I[Ljava/nio/ByteBuffer;)V")),
-        j_byte_buffer_class_(jni, FindClass(jni, "java/nio/ByteBuffer")) {
-    CHECK_EXCEPTION(jni, "");
-  }
-
-  virtual ~JavaVideoRendererWrapper() {}
-
-  virtual void SetSize(int width, int height) OVERRIDE {
-    jni()->CallVoidMethod(*j_callbacks_, j_set_size_id_, width, height);
-    CHECK_EXCEPTION(jni(), "");
-  }
-
-  virtual void RenderFrame(const cricket::VideoFrame* frame) OVERRIDE {
-    ScopedLocalRef<jobject> j_frame(jni(), CricketToJavaFrame(frame));
-    jni()->CallVoidMethod(*j_callbacks_, j_render_frame_id_, *j_frame);
-    CHECK_EXCEPTION(jni(), "");
-  }
-
- private:
-  // Return a VideoRenderer.I420Frame referring to the data in |frame|.
-  jobject CricketToJavaFrame(const cricket::VideoFrame* frame) {
-    ScopedLocalRef<jintArray> strides(jni(), jni()->NewIntArray(3));
-    jint* strides_array = jni()->GetIntArrayElements(*strides, NULL);
-    strides_array[0] = frame->GetYPitch();
-    strides_array[1] = frame->GetUPitch();
-    strides_array[2] = frame->GetVPitch();
-    jni()->ReleaseIntArrayElements(*strides, strides_array, 0);
-    ScopedLocalRef<jobjectArray> planes(
-        jni(), jni()->NewObjectArray(3, *j_byte_buffer_class_, NULL));
-    ScopedLocalRef<jobject> y_buffer(jni(), jni()->NewDirectByteBuffer(
-        const_cast<uint8*>(frame->GetYPlane()),
-        frame->GetYPitch() * frame->GetHeight()));
-    ScopedLocalRef<jobject> u_buffer(jni(), jni()->NewDirectByteBuffer(
-        const_cast<uint8*>(frame->GetUPlane()), frame->GetChromaSize()));
-    ScopedLocalRef<jobject> v_buffer(jni(), jni()->NewDirectByteBuffer(
-        const_cast<uint8*>(frame->GetVPlane()), frame->GetChromaSize()));
-    jni()->SetObjectArrayElement(*planes, 0, *y_buffer);
-    jni()->SetObjectArrayElement(*planes, 1, *u_buffer);
-    jni()->SetObjectArrayElement(*planes, 2, *v_buffer);
-    return jni()->NewObject(
-        *j_frame_class_, j_frame_ctor_id_,
-        frame->GetWidth(), frame->GetHeight(), *strides, *planes);
-  }
-
-  JNIEnv* jni() {
-    return AttachCurrentThreadIfNeeded();
-  }
-
-  ScopedGlobalRef<jobject> j_callbacks_;
-  jmethodID j_set_size_id_;
-  jmethodID j_render_frame_id_;
-  ScopedGlobalRef<jclass> j_frame_class_;
-  jmethodID j_frame_ctor_id_;
-  ScopedGlobalRef<jclass> j_byte_buffer_class_;
-};
-
-}  // anonymous namespace
-
-
-// Convenience macro defining JNI-accessible methods in the org.webrtc package.
-// Eliminates unnecessary boilerplate and line-wraps, reducing visual clutter.
-#define JOW(rettype, name) extern "C" rettype JNIEXPORT JNICALL \
-  Java_org_webrtc_##name
-
-extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM *jvm, void *reserved) {
-  CHECK(!g_jvm, "JNI_OnLoad called more than once!");
-  g_jvm = jvm;
-  CHECK(g_jvm, "JNI_OnLoad handed NULL?");
-
-  CHECK(talk_base::InitializeSSL(), "Failed to InitializeSSL()");
-
-  JNIEnv* jni;
-  if (jvm->GetEnv(reinterpret_cast<void**>(&jni), JNI_VERSION_1_6) != JNI_OK)
-    return -1;
-  g_class_reference_holder = new ClassReferenceHolder(jni);
-
-  webrtc::Trace::CreateTrace();
-
-  return JNI_VERSION_1_6;
-}
-
-extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM *jvm, void *reserved) {
-  webrtc::Trace::ReturnTrace();
-  delete g_class_reference_holder;
-  g_class_reference_holder = NULL;
-  CHECK(talk_base::CleanupSSL(), "Failed to CleanupSSL()");
-}
-
-static talk_base::scoped_refptr<DataChannelInterface> ExtractNativeDC(
-    JNIEnv* jni, jobject j_dc) {
-  jfieldID native_dc_id = GetFieldID(jni,
-      GetObjectClass(jni, j_dc), "nativeDataChannel", "J");
-  jlong j_d = GetLongField(jni, j_dc, native_dc_id);
-  return talk_base::scoped_refptr<DataChannelInterface>(
-      reinterpret_cast<DataChannelInterface*>(j_d));
-}
-
-JOW(jlong, DataChannel_registerObserverNative)(
-    JNIEnv* jni, jobject j_dc, jobject j_observer) {
-  talk_base::scoped_ptr<DataChannelObserverWrapper> observer(
-      new DataChannelObserverWrapper(jni, j_observer));
-  ExtractNativeDC(jni, j_dc)->RegisterObserver(observer.get());
-  return reinterpret_cast<jlong>(observer.release());
-}
-
-JOW(void, DataChannel_unregisterObserverNative)(
-    JNIEnv* jni, jobject j_dc, jlong native_observer) {
-  ExtractNativeDC(jni, j_dc)->UnregisterObserver();
-  delete reinterpret_cast<DataChannelObserverWrapper*>(native_observer);
-}
-
-JOW(jstring, DataChannel_label)(JNIEnv* jni, jobject j_dc) {
-  return JavaStringFromStdString(jni, ExtractNativeDC(jni, j_dc)->label());
-}
-
-JOW(jobject, DataChannel_state)(JNIEnv* jni, jobject j_dc) {
-  return JavaEnumFromIndex(
-      jni, "DataChannel$State", ExtractNativeDC(jni, j_dc)->state());
-}
-
-JOW(jlong, DataChannel_bufferedAmount)(JNIEnv* jni, jobject j_dc) {
-  uint64 buffered_amount = ExtractNativeDC(jni, j_dc)->buffered_amount();
-  CHECK(buffered_amount <= std::numeric_limits<int64>::max(),
-        "buffered_amount overflowed jlong!");
-  return static_cast<jlong>(buffered_amount);
-}
-
-JOW(void, DataChannel_close)(JNIEnv* jni, jobject j_dc) {
-  ExtractNativeDC(jni, j_dc)->Close();
-}
-
-JOW(jboolean, DataChannel_sendNative)(JNIEnv* jni, jobject j_dc,
-                                      jbyteArray data, jboolean binary) {
-  jbyte* bytes = jni->GetByteArrayElements(data, NULL);
-  bool ret = ExtractNativeDC(jni, j_dc)->Send(DataBuffer(
-      talk_base::Buffer(bytes, jni->GetArrayLength(data)),
-      binary));
-  jni->ReleaseByteArrayElements(data, bytes, JNI_ABORT);
-  return ret;
-}
-
-JOW(void, DataChannel_dispose)(JNIEnv* jni, jobject j_dc) {
-  ExtractNativeDC(jni, j_dc)->Release();
-}
-
-JOW(void, Logging_nativeEnableTracing)(
-    JNIEnv* jni, jclass, jstring j_path, jint nativeLevels,
-    jint nativeSeverity) {
-  std::string path = JavaToStdString(jni, j_path);
-  if (nativeLevels != webrtc::kTraceNone) {
-    CHECK(!webrtc::Trace::SetTraceFile(path.c_str(), false),
-          "SetTraceFile failed");
-    CHECK(!webrtc::Trace::SetLevelFilter(nativeLevels),
-          "SetLevelFilter failed");
-  }
-  talk_base::LogMessage::LogToDebug(nativeSeverity);
-}
-
-JOW(void, PeerConnection_freePeerConnection)(JNIEnv*, jclass, jlong j_p) {
-  reinterpret_cast<PeerConnectionInterface*>(j_p)->Release();
-}
-
-JOW(void, PeerConnection_freeObserver)(JNIEnv*, jclass, jlong j_p) {
-  PCOJava* p = reinterpret_cast<PCOJava*>(j_p);
-  delete p;
-}
-
-JOW(void, MediaSource_free)(JNIEnv*, jclass, jlong j_p) {
-  reinterpret_cast<MediaSourceInterface*>(j_p)->Release();
-}
-
-JOW(void, VideoCapturer_free)(JNIEnv*, jclass, jlong j_p) {
-  delete reinterpret_cast<cricket::VideoCapturer*>(j_p);
-}
-
-JOW(void, VideoRenderer_free)(JNIEnv*, jclass, jlong j_p) {
-  delete reinterpret_cast<VideoRendererWrapper*>(j_p);
-}
-
-JOW(void, MediaStreamTrack_free)(JNIEnv*, jclass, jlong j_p) {
-  reinterpret_cast<MediaStreamTrackInterface*>(j_p)->Release();
-}
-
-JOW(jboolean, MediaStream_nativeAddAudioTrack)(
-    JNIEnv* jni, jclass, jlong pointer, jlong j_audio_track_pointer) {
-  talk_base::scoped_refptr<MediaStreamInterface> stream(
-      reinterpret_cast<MediaStreamInterface*>(pointer));
-  talk_base::scoped_refptr<AudioTrackInterface> track(
-      reinterpret_cast<AudioTrackInterface*>(j_audio_track_pointer));
-  return stream->AddTrack(track);
-}
-
-JOW(jboolean, MediaStream_nativeAddVideoTrack)(
-    JNIEnv* jni, jclass, jlong pointer, jlong j_video_track_pointer) {
-  talk_base::scoped_refptr<MediaStreamInterface> stream(
-      reinterpret_cast<MediaStreamInterface*>(pointer));
-  talk_base::scoped_refptr<VideoTrackInterface> track(
-      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
-  return stream->AddTrack(track);
-}
-
-JOW(jboolean, MediaStream_nativeRemoveAudioTrack)(
-    JNIEnv* jni, jclass, jlong pointer, jlong j_audio_track_pointer) {
-  talk_base::scoped_refptr<MediaStreamInterface> stream(
-      reinterpret_cast<MediaStreamInterface*>(pointer));
-  talk_base::scoped_refptr<AudioTrackInterface> track(
-      reinterpret_cast<AudioTrackInterface*>(j_audio_track_pointer));
-  return stream->RemoveTrack(track);
-}
-
-JOW(jboolean, MediaStream_nativeRemoveVideoTrack)(
-    JNIEnv* jni, jclass, jlong pointer, jlong j_video_track_pointer) {
-  talk_base::scoped_refptr<MediaStreamInterface> stream(
-      reinterpret_cast<MediaStreamInterface*>(pointer));
-  talk_base::scoped_refptr<VideoTrackInterface> track(
-      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
-  return stream->RemoveTrack(track);
-}
-
-JOW(jstring, MediaStream_nativeLabel)(JNIEnv* jni, jclass, jlong j_p) {
-  return JavaStringFromStdString(
-      jni, reinterpret_cast<MediaStreamInterface*>(j_p)->label());
-}
-
-JOW(void, MediaStream_free)(JNIEnv*, jclass, jlong j_p) {
-  reinterpret_cast<MediaStreamInterface*>(j_p)->Release();
-}
-
-JOW(jlong, PeerConnectionFactory_nativeCreateObserver)(
-    JNIEnv * jni, jclass, jobject j_observer) {
-  return (jlong)new PCOJava(jni, j_observer);
-}
-
-#ifdef ANDROID
-JOW(jboolean, PeerConnectionFactory_initializeAndroidGlobals)(
-    JNIEnv* jni, jclass, jobject context) {
-  CHECK(g_jvm, "JNI_OnLoad failed to run?");
-  bool failure = false;
-  failure |= webrtc::VideoEngine::SetAndroidObjects(g_jvm, context);
-  failure |= webrtc::VoiceEngine::SetAndroidObjects(g_jvm, jni, context);
-  return !failure;
-}
-#endif  // ANDROID
-
-JOW(jlong, PeerConnectionFactory_nativeCreatePeerConnectionFactory)(
-    JNIEnv* jni, jclass) {
-  talk_base::scoped_refptr<PeerConnectionFactoryInterface> factory(
-      webrtc::CreatePeerConnectionFactory());
-  return (jlong)factory.release();
-}
-
-JOW(void, PeerConnectionFactory_freeFactory)(JNIEnv*, jclass, jlong j_p) {
-  reinterpret_cast<PeerConnectionFactoryInterface*>(j_p)->Release();
-}
-
-JOW(jlong, PeerConnectionFactory_nativeCreateLocalMediaStream)(
-    JNIEnv* jni, jclass, jlong native_factory, jstring label) {
-  talk_base::scoped_refptr<PeerConnectionFactoryInterface> factory(
-      reinterpret_cast<PeerConnectionFactoryInterface*>(native_factory));
-  talk_base::scoped_refptr<MediaStreamInterface> stream(
-      factory->CreateLocalMediaStream(JavaToStdString(jni, label)));
-  return (jlong)stream.release();
-}
-
-JOW(jlong, PeerConnectionFactory_nativeCreateVideoSource)(
-    JNIEnv* jni, jclass, jlong native_factory, jlong native_capturer,
-    jobject j_constraints) {
-  talk_base::scoped_ptr<ConstraintsWrapper> constraints(
-      new ConstraintsWrapper(jni, j_constraints));
-  talk_base::scoped_refptr<PeerConnectionFactoryInterface> factory(
-      reinterpret_cast<PeerConnectionFactoryInterface*>(native_factory));
-  talk_base::scoped_refptr<VideoSourceInterface> source(
-      factory->CreateVideoSource(
-          reinterpret_cast<cricket::VideoCapturer*>(native_capturer),
-          constraints.get()));
-  return (jlong)source.release();
-}
-
-JOW(jlong, PeerConnectionFactory_nativeCreateVideoTrack)(
-    JNIEnv* jni, jclass, jlong native_factory, jstring id,
-    jlong native_source) {
-  talk_base::scoped_refptr<PeerConnectionFactoryInterface> factory(
-      reinterpret_cast<PeerConnectionFactoryInterface*>(native_factory));
-  talk_base::scoped_refptr<VideoTrackInterface> track(
-      factory->CreateVideoTrack(
-          JavaToStdString(jni, id),
-          reinterpret_cast<VideoSourceInterface*>(native_source)));
-  return (jlong)track.release();
-}
-
-JOW(jlong, PeerConnectionFactory_nativeCreateAudioTrack)(
-    JNIEnv* jni, jclass, jlong native_factory, jstring id) {
-  talk_base::scoped_refptr<PeerConnectionFactoryInterface> factory(
-      reinterpret_cast<PeerConnectionFactoryInterface*>(native_factory));
-  talk_base::scoped_refptr<AudioTrackInterface> track(
-      factory->CreateAudioTrack(JavaToStdString(jni, id), NULL));
-  return (jlong)track.release();
-}
-
-static void JavaIceServersToJsepIceServers(
-    JNIEnv* jni, jobject j_ice_servers,
-    PeerConnectionInterface::IceServers* ice_servers) {
-  jclass list_class = GetObjectClass(jni, j_ice_servers);
-  jmethodID iterator_id = GetMethodID(
-      jni, list_class, "iterator", "()Ljava/util/Iterator;");
-  jobject iterator = jni->CallObjectMethod(j_ice_servers, iterator_id);
-  CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-  jmethodID iterator_has_next = GetMethodID(
-      jni, GetObjectClass(jni, iterator), "hasNext", "()Z");
-  jmethodID iterator_next = GetMethodID(
-      jni, GetObjectClass(jni, iterator), "next", "()Ljava/lang/Object;");
-  while (jni->CallBooleanMethod(iterator, iterator_has_next)) {
-    CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
-    jobject j_ice_server = jni->CallObjectMethod(iterator, iterator_next);
-    CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-    jclass j_ice_server_class = GetObjectClass(jni, j_ice_server);
-    jfieldID j_ice_server_uri_id =
-        GetFieldID(jni, j_ice_server_class, "uri", "Ljava/lang/String;");
-    jfieldID j_ice_server_username_id =
-        GetFieldID(jni, j_ice_server_class, "username", "Ljava/lang/String;");
-    jfieldID j_ice_server_password_id =
-        GetFieldID(jni, j_ice_server_class, "password", "Ljava/lang/String;");
-    jstring uri = reinterpret_cast<jstring>(
-        GetObjectField(jni, j_ice_server, j_ice_server_uri_id));
-    jstring username = reinterpret_cast<jstring>(
-        GetObjectField(jni, j_ice_server, j_ice_server_username_id));
-    jstring password = reinterpret_cast<jstring>(
-        GetObjectField(jni, j_ice_server, j_ice_server_password_id));
-    PeerConnectionInterface::IceServer server;
-    server.uri = JavaToStdString(jni, uri);
-    server.username = JavaToStdString(jni, username);
-    server.password = JavaToStdString(jni, password);
-    ice_servers->push_back(server);
-  }
-  CHECK_EXCEPTION(jni, "error during CallBooleanMethod");
-}
-
-JOW(jlong, PeerConnectionFactory_nativeCreatePeerConnection)(
-    JNIEnv *jni, jclass, jlong factory, jobject j_ice_servers,
-    jobject j_constraints, jlong observer_p) {
-  talk_base::scoped_refptr<PeerConnectionFactoryInterface> f(
-      reinterpret_cast<PeerConnectionFactoryInterface*>(factory));
-  PeerConnectionInterface::IceServers servers;
-  JavaIceServersToJsepIceServers(jni, j_ice_servers, &servers);
-  PCOJava* observer = reinterpret_cast<PCOJava*>(observer_p);
-  observer->SetConstraints(new ConstraintsWrapper(jni, j_constraints));
-  talk_base::scoped_refptr<PeerConnectionInterface> pc(f->CreatePeerConnection(
-      servers, observer->constraints(), NULL, observer));
-  return (jlong)pc.release();
-}
-
-static talk_base::scoped_refptr<PeerConnectionInterface> ExtractNativePC(
-    JNIEnv* jni, jobject j_pc) {
-  jfieldID native_pc_id = GetFieldID(jni,
-      GetObjectClass(jni, j_pc), "nativePeerConnection", "J");
-  jlong j_p = GetLongField(jni, j_pc, native_pc_id);
-  return talk_base::scoped_refptr<PeerConnectionInterface>(
-      reinterpret_cast<PeerConnectionInterface*>(j_p));
-}
-
-JOW(jobject, PeerConnection_getLocalDescription)(JNIEnv* jni, jobject j_pc) {
-  const SessionDescriptionInterface* sdp =
-      ExtractNativePC(jni, j_pc)->local_description();
-  return sdp ? JavaSdpFromNativeSdp(jni, sdp) : NULL;
-}
-
-JOW(jobject, PeerConnection_getRemoteDescription)(JNIEnv* jni, jobject j_pc) {
-  const SessionDescriptionInterface* sdp =
-      ExtractNativePC(jni, j_pc)->remote_description();
-  return sdp ? JavaSdpFromNativeSdp(jni, sdp) : NULL;
-}
-
-JOW(jobject, PeerConnection_createDataChannel)(
-    JNIEnv* jni, jobject j_pc, jstring j_label, jobject j_init) {
-  DataChannelInit init = JavaDataChannelInitToNative(jni, j_init);
-  talk_base::scoped_refptr<DataChannelInterface> channel(
-      ExtractNativePC(jni, j_pc)->CreateDataChannel(
-          JavaToStdString(jni, j_label), &init));
-  jclass j_data_channel_class = FindClass(jni, "org/webrtc/DataChannel");
-  jmethodID j_data_channel_ctor = GetMethodID(
-      jni, j_data_channel_class, "<init>", "(J)V");
-  jobject j_channel = jni->NewObject(
-      j_data_channel_class, j_data_channel_ctor, channel.get());
-  CHECK_EXCEPTION(jni, "error during NewObject");
-  // Channel is now owned by Java object, and will be freed from there.
-  channel->AddRef();
-  return j_channel;
-}
-
-JOW(void, PeerConnection_createOffer)(
-    JNIEnv* jni, jobject j_pc, jobject j_observer, jobject j_constraints) {
-  ConstraintsWrapper* constraints =
-      new ConstraintsWrapper(jni, j_constraints);
-  talk_base::scoped_refptr<CreateSdpObserverWrapper> observer(
-      new talk_base::RefCountedObject<CreateSdpObserverWrapper>(
-          jni, j_observer, constraints));
-  ExtractNativePC(jni, j_pc)->CreateOffer(observer, constraints);
-}
-
-JOW(void, PeerConnection_createAnswer)(
-    JNIEnv* jni, jobject j_pc, jobject j_observer, jobject j_constraints) {
-  ConstraintsWrapper* constraints =
-      new ConstraintsWrapper(jni, j_constraints);
-  talk_base::scoped_refptr<CreateSdpObserverWrapper> observer(
-      new talk_base::RefCountedObject<CreateSdpObserverWrapper>(
-          jni, j_observer, constraints));
-  ExtractNativePC(jni, j_pc)->CreateAnswer(observer, constraints);
-}
-
-// Helper to create a SessionDescriptionInterface from a SessionDescription.
-static SessionDescriptionInterface* JavaSdpToNativeSdp(
-    JNIEnv* jni, jobject j_sdp) {
-  jfieldID j_type_id = GetFieldID(
-      jni, GetObjectClass(jni, j_sdp), "type",
-      "Lorg/webrtc/SessionDescription$Type;");
-  jobject j_type = GetObjectField(jni, j_sdp, j_type_id);
-  jmethodID j_canonical_form_id = GetMethodID(
-      jni, GetObjectClass(jni, j_type), "canonicalForm",
-      "()Ljava/lang/String;");
-  jstring j_type_string = (jstring)jni->CallObjectMethod(
-      j_type, j_canonical_form_id);
-  CHECK_EXCEPTION(jni, "error during CallObjectMethod");
-  std::string std_type = JavaToStdString(jni, j_type_string);
-
-  jfieldID j_description_id = GetFieldID(
-      jni, GetObjectClass(jni, j_sdp), "description", "Ljava/lang/String;");
-  jstring j_description = (jstring)GetObjectField(jni, j_sdp, j_description_id);
-  std::string std_description = JavaToStdString(jni, j_description);
-
-  return webrtc::CreateSessionDescription(
-      std_type, std_description, NULL);
-}
-
-JOW(void, PeerConnection_setLocalDescription)(
-    JNIEnv* jni, jobject j_pc,
-    jobject j_observer, jobject j_sdp) {
-  talk_base::scoped_refptr<SetSdpObserverWrapper> observer(
-      new talk_base::RefCountedObject<SetSdpObserverWrapper>(
-          jni, j_observer, reinterpret_cast<ConstraintsWrapper*>(NULL)));
-  ExtractNativePC(jni, j_pc)->SetLocalDescription(
-      observer, JavaSdpToNativeSdp(jni, j_sdp));
-}
-
-JOW(void, PeerConnection_setRemoteDescription)(
-    JNIEnv* jni, jobject j_pc,
-    jobject j_observer, jobject j_sdp) {
-  talk_base::scoped_refptr<SetSdpObserverWrapper> observer(
-      new talk_base::RefCountedObject<SetSdpObserverWrapper>(
-          jni, j_observer, reinterpret_cast<ConstraintsWrapper*>(NULL)));
-  ExtractNativePC(jni, j_pc)->SetRemoteDescription(
-      observer, JavaSdpToNativeSdp(jni, j_sdp));
-}
-
-JOW(jboolean, PeerConnection_updateIce)(
-    JNIEnv* jni, jobject j_pc, jobject j_ice_servers, jobject j_constraints) {
-  PeerConnectionInterface::IceServers ice_servers;
-  JavaIceServersToJsepIceServers(jni, j_ice_servers, &ice_servers);
-  talk_base::scoped_ptr<ConstraintsWrapper> constraints(
-      new ConstraintsWrapper(jni, j_constraints));
-  return ExtractNativePC(jni, j_pc)->UpdateIce(ice_servers, constraints.get());
-}
-
-JOW(jboolean, PeerConnection_nativeAddIceCandidate)(
-    JNIEnv* jni, jobject j_pc, jstring j_sdp_mid,
-    jint j_sdp_mline_index, jstring j_candidate_sdp) {
-  std::string sdp_mid = JavaToStdString(jni, j_sdp_mid);
-  std::string sdp = JavaToStdString(jni, j_candidate_sdp);
-  talk_base::scoped_ptr<IceCandidateInterface> candidate(
-      webrtc::CreateIceCandidate(sdp_mid, j_sdp_mline_index, sdp, NULL));
-  return ExtractNativePC(jni, j_pc)->AddIceCandidate(candidate.get());
-}
-
-JOW(jboolean, PeerConnection_nativeAddLocalStream)(
-    JNIEnv* jni, jobject j_pc, jlong native_stream, jobject j_constraints) {
-  talk_base::scoped_ptr<ConstraintsWrapper> constraints(
-      new ConstraintsWrapper(jni, j_constraints));
-  return ExtractNativePC(jni, j_pc)->AddStream(
-      reinterpret_cast<MediaStreamInterface*>(native_stream),
-      constraints.get());
-}
-
-JOW(void, PeerConnection_nativeRemoveLocalStream)(
-    JNIEnv* jni, jobject j_pc, jlong native_stream) {
-  ExtractNativePC(jni, j_pc)->RemoveStream(
-      reinterpret_cast<MediaStreamInterface*>(native_stream));
-}
-
-JOW(bool, PeerConnection_nativeGetStats)(
-    JNIEnv* jni, jobject j_pc, jobject j_observer, jlong native_track) {
-  talk_base::scoped_refptr<StatsObserverWrapper> observer(
-      new talk_base::RefCountedObject<StatsObserverWrapper>(jni, j_observer));
-  return ExtractNativePC(jni, j_pc)->GetStats(
-      observer, reinterpret_cast<MediaStreamTrackInterface*>(native_track));
-}
-
-JOW(jobject, PeerConnection_signalingState)(JNIEnv* jni, jobject j_pc) {
-  PeerConnectionInterface::SignalingState state =
-      ExtractNativePC(jni, j_pc)->signaling_state();
-  return JavaEnumFromIndex(jni, "PeerConnection$SignalingState", state);
-}
-
-JOW(jobject, PeerConnection_iceConnectionState)(JNIEnv* jni, jobject j_pc) {
-  PeerConnectionInterface::IceConnectionState state =
-      ExtractNativePC(jni, j_pc)->ice_connection_state();
-  return JavaEnumFromIndex(jni, "PeerConnection$IceConnectionState", state);
-}
-
-JOW(jobject, PeerGathering_iceGatheringState)(JNIEnv* jni, jobject j_pc) {
-  PeerConnectionInterface::IceGatheringState state =
-      ExtractNativePC(jni, j_pc)->ice_gathering_state();
-  return JavaEnumFromIndex(jni, "PeerGathering$IceGatheringState", state);
-}
-
-JOW(void, PeerConnection_close)(JNIEnv* jni, jobject j_pc) {
-  ExtractNativePC(jni, j_pc)->Close();
-  return;
-}
-
-JOW(jobject, MediaSource_nativeState)(JNIEnv* jni, jclass, jlong j_p) {
-  talk_base::scoped_refptr<MediaSourceInterface> p(
-      reinterpret_cast<MediaSourceInterface*>(j_p));
-  return JavaEnumFromIndex(jni, "MediaSource$State", p->state());
-}
-
-JOW(jlong, VideoCapturer_nativeCreateVideoCapturer)(
-    JNIEnv* jni, jclass, jstring j_device_name) {
-  std::string device_name = JavaToStdString(jni, j_device_name);
-  talk_base::scoped_ptr<cricket::DeviceManagerInterface> device_manager(
-      cricket::DeviceManagerFactory::Create());
-  CHECK(device_manager->Init(), "DeviceManager::Init() failed");
-  cricket::Device device;
-  if (!device_manager->GetVideoCaptureDevice(device_name, &device)) {
-    LOG(LS_ERROR) << "GetVideoCaptureDevice failed";
-    return 0;
-  }
-  talk_base::scoped_ptr<cricket::VideoCapturer> capturer(
-      device_manager->CreateVideoCapturer(device));
-  return (jlong)capturer.release();
-}
-
-JOW(jlong, VideoRenderer_nativeCreateGuiVideoRenderer)(
-    JNIEnv* jni, jclass, int x, int y) {
-  talk_base::scoped_ptr<VideoRendererWrapper> renderer(
-      VideoRendererWrapper::Create(
-          cricket::VideoRendererFactory::CreateGuiVideoRenderer(x, y)));
-  return (jlong)renderer.release();
-}
-
-JOW(jlong, VideoRenderer_nativeWrapVideoRenderer)(
-    JNIEnv* jni, jclass, jobject j_callbacks) {
-  talk_base::scoped_ptr<JavaVideoRendererWrapper> renderer(
-      new JavaVideoRendererWrapper(jni, j_callbacks));
-  return (jlong)renderer.release();
-}
-
-JOW(jstring, MediaStreamTrack_nativeId)(JNIEnv* jni, jclass, jlong j_p) {
-  talk_base::scoped_refptr<MediaStreamTrackInterface> p(
-      reinterpret_cast<MediaStreamTrackInterface*>(j_p));
-  return JavaStringFromStdString(jni, p->id());
-}
-
-JOW(jstring, MediaStreamTrack_nativeKind)(JNIEnv* jni, jclass, jlong j_p) {
-  talk_base::scoped_refptr<MediaStreamTrackInterface> p(
-      reinterpret_cast<MediaStreamTrackInterface*>(j_p));
-  return JavaStringFromStdString(jni, p->kind());
-}
-
-JOW(jboolean, MediaStreamTrack_nativeEnabled)(JNIEnv* jni, jclass, jlong j_p) {
-  talk_base::scoped_refptr<MediaStreamTrackInterface> p(
-      reinterpret_cast<MediaStreamTrackInterface*>(j_p));
-  return p->enabled();
-}
-
-JOW(jobject, MediaStreamTrack_nativeState)(JNIEnv* jni, jclass, jlong j_p) {
-  talk_base::scoped_refptr<MediaStreamTrackInterface> p(
-      reinterpret_cast<MediaStreamTrackInterface*>(j_p));
-  return JavaEnumFromIndex(jni, "MediaStreamTrack$State", p->state());
-}
-
-JOW(jboolean, MediaStreamTrack_nativeSetState)(
-    JNIEnv* jni, jclass, jlong j_p, jint j_new_state) {
-  talk_base::scoped_refptr<MediaStreamTrackInterface> p(
-      reinterpret_cast<MediaStreamTrackInterface*>(j_p));
-  MediaStreamTrackInterface::TrackState new_state =
-      (MediaStreamTrackInterface::TrackState)j_new_state;
-  return p->set_state(new_state);
-}
-
-JOW(jboolean, MediaStreamTrack_nativeSetEnabled)(
-    JNIEnv* jni, jclass, jlong j_p, jboolean enabled) {
-  talk_base::scoped_refptr<MediaStreamTrackInterface> p(
-      reinterpret_cast<MediaStreamTrackInterface*>(j_p));
-  return p->set_enabled(enabled);
-}
-
-JOW(void, VideoTrack_nativeAddRenderer)(
-    JNIEnv* jni, jclass,
-    jlong j_video_track_pointer, jlong j_renderer_pointer) {
-  talk_base::scoped_refptr<VideoTrackInterface> track(
-      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
-  track->AddRenderer(
-      reinterpret_cast<VideoRendererInterface*>(j_renderer_pointer));
-}
-
-JOW(void, VideoTrack_nativeRemoveRenderer)(
-    JNIEnv* jni, jclass,
-    jlong j_video_track_pointer, jlong j_renderer_pointer) {
-  talk_base::scoped_refptr<VideoTrackInterface> track(
-      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
-  track->RemoveRenderer(
-      reinterpret_cast<VideoRendererInterface*>(j_renderer_pointer));
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/AudioSource.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/AudioSource.java
deleted file mode 100644
index 8b7a8f7dc40..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/AudioSource.java
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-/**
- * Java wrapper for a C++ AudioSourceInterface.  Used as the source for one or
- * more {@code AudioTrack} objects.
- */
-public class AudioSource extends MediaSource {
-  public AudioSource(long nativeSource) {
-    super(nativeSource);
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/AudioTrack.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/AudioTrack.java
deleted file mode 100644
index 35d7c41f2bc..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/AudioTrack.java
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-/** Java wrapper for a C++ AudioTrackInterface */
-public class AudioTrack extends MediaStreamTrack {
-  public AudioTrack(long nativeTrack) {
-    super(nativeTrack);
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/DataChannel.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/DataChannel.java
deleted file mode 100644
index d20e42d9834..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/DataChannel.java
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-import java.nio.ByteBuffer;
-
-/** Java wrapper for a C++ DataChannelInterface. */
-public class DataChannel {
-  /** Java wrapper for WebIDL RTCDataChannel. */
-  public static class Init {
-    public boolean ordered = true;
-    // Optional unsigned short in WebIDL, -1 means unspecified.
-    public int maxRetransmitTimeMs = -1;
-    // Optional unsigned short in WebIDL, -1 means unspecified.
-    public int maxRetransmits = -1;
-    public String protocol = "";
-    public boolean negotiated = true;
-    // Optional unsigned short in WebIDL, -1 means unspecified.
-    public int id = -1;
-
-    public Init() {}
-
-    // Called only by native code.
-    private Init(
-        boolean ordered, int maxRetransmitTimeMs, int maxRetransmits,
-        String protocol, boolean negotiated, int id) {
-      this.ordered = ordered;
-      this.maxRetransmitTimeMs = maxRetransmitTimeMs;
-      this.maxRetransmits = maxRetransmits;
-      this.protocol = protocol;
-      this.negotiated = negotiated;
-      this.id = id;
-    }
-  }
-
-  /** Java version of C++ DataBuffer.  The atom of data in a DataChannel. */
-  public static class Buffer {
-    /** The underlying data. */
-    public final ByteBuffer data;
-
-    /**
-     * Indicates whether |data| contains UTF-8 text or "binary data"
-     * (i.e. anything else).
-     */
-    public final boolean binary;
-
-    public Buffer(ByteBuffer data, boolean binary) {
-      this.data = data;
-      this.binary = binary;
-    }
-  }
-
-  /** Java version of C++ DataChannelObserver. */
-  public interface Observer {
-    /** The data channel state has changed. */
-    public void onStateChange();
-    /**
-     * A data buffer was successfully received.  NOTE: |buffer.data| will be
-     * freed once this function returns so callers who want to use the data
-     * asynchronously must make sure to copy it first.
-     */
-    public void onMessage(Buffer buffer);
-  }
-
-  /** Keep in sync with DataChannelInterface::DataState. */
-  public enum State { CONNECTING, OPEN, CLOSING, CLOSED };
-
-  private final long nativeDataChannel;
-  private long nativeObserver;
-
-  public DataChannel(long nativeDataChannel) {
-    this.nativeDataChannel = nativeDataChannel;
-  }
-
-  /** Register |observer|, replacing any previously-registered observer. */
-  public void registerObserver(Observer observer) {
-    if (nativeObserver != 0) {
-      unregisterObserverNative(nativeObserver);
-    }
-    nativeObserver = registerObserverNative(observer);
-  }
-  private native long registerObserverNative(Observer observer);
-
-  /** Unregister the (only) observer. */
-  public void unregisterObserver() {
-    unregisterObserverNative(nativeObserver);
-  }
-  private native void unregisterObserverNative(long nativeObserver);
-
-  public native String label();
-
-  public native State state();
-
-  /**
-   * Return the number of bytes of application data (UTF-8 text and binary data)
-   * that have been queued using SendBuffer but have not yet been transmitted
-   * to the network.
-   */
-  public native long bufferedAmount();
-
-  /** Close the channel. */
-  public native void close();
-
-  /** Send |data| to the remote peer; return success. */
-  public boolean send(Buffer buffer) {
-    // TODO(fischman): this could be cleverer about avoiding copies if the
-    // ByteBuffer is direct and/or is backed by an array.
-    byte[] data = new byte[buffer.data.remaining()];
-    buffer.data.get(data);
-    return sendNative(data, buffer.binary);
-  }
-  private native boolean sendNative(byte[] data, boolean binary);
-
-  /** Dispose of native resources attached to this channel. */
-  public native void dispose();
-};
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/Logging.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/Logging.java
deleted file mode 100644
index f6918b8bf82..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/Logging.java
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-import java.util.EnumSet;
-
-/** Java wrapper for WebRTC & libjingle logging. */
-public class Logging {
-  static {
-    System.loadLibrary("jingle_peerconnection_so");
-  }
-
-  // Keep in sync with webrtc/common_types.h:TraceLevel.
-  public enum TraceLevel {
-    TRACE_NONE(0x0000),
-    TRACE_STATEINFO(0x0001),
-    TRACE_WARNING(0x0002),
-    TRACE_ERROR(0x0004),
-    TRACE_CRITICAL(0x0008),
-    TRACE_APICALL(0x0010),
-    TRACE_DEFAULT(0x00ff),
-    TRACE_MODULECALL(0x0020),
-    TRACE_MEMORY(0x0100),
-    TRACE_TIMER(0x0200),
-    TRACE_STREAM(0x0400),
-    TRACE_DEBUG(0x0800),
-    TRACE_INFO(0x1000),
-    TRACE_TERSEINFO(0x2000),
-    TRACE_ALL(0xffff);
-
-    public final int level;
-    TraceLevel(int level) {
-      this.level = level;
-    }
-  };
-
-  // Keep in sync with talk/base/logging.h:LoggingSeverity.
-  public enum Severity {
-    LS_SENSITIVE, LS_VERBOSE, LS_INFO, LS_WARNING, LS_ERROR,
-  };
-
-
-  // Enable tracing to |path| at |levels| and |severity|.
-  public static void enableTracing(
-      String path, EnumSet<TraceLevel> levels, Severity severity) {
-    int nativeLevel = 0;
-    for (TraceLevel level : levels) {
-      nativeLevel |= level.level;
-    }
-    nativeEnableTracing(path, nativeLevel, severity.ordinal());
-  }
-
-  private static native void nativeEnableTracing(
-      String path, int nativeLevels, int nativeSeverity);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaConstraints.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaConstraints.java
deleted file mode 100644
index ef303019c1b..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaConstraints.java
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-import java.util.LinkedList;
-import java.util.List;
-
-/**
- * Description of media constraints for {@code MediaStream} and
- * {@code PeerConnection}.
- */
-public class MediaConstraints {
-  /** Simple String key/value pair. */
-  public static class KeyValuePair {
-    private final String key;
-    private final String value;
-
-    public KeyValuePair(String key, String value) {
-      this.key = key;
-      this.value = value;
-    }
-
-    public String getKey() {
-      return key;
-    }
-
-    public String getValue() {
-      return value;
-    }
-
-    public String toString() {
-      return key + ": " + value;
-    }
-  }
-
-
-  public final List<KeyValuePair> mandatory;
-  public final List<KeyValuePair> optional;
-
-  public MediaConstraints() {
-    mandatory = new LinkedList<KeyValuePair>();
-    optional = new LinkedList<KeyValuePair>();
-  }
-
-  private static String stringifyKeyValuePairList(List<KeyValuePair> list) {
-    StringBuilder builder = new StringBuilder("[");
-    for (KeyValuePair pair : list) {
-      if (builder.length() > 1) {
-        builder.append(", ");
-      }
-      builder.append(pair.toString());
-    }
-    return builder.append("]").toString();
-  }
-
-  public String toString() {
-    return "mandatory: " + stringifyKeyValuePairList(mandatory) +
-        ", optional: " + stringifyKeyValuePairList(optional);
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaStream.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaStream.java
deleted file mode 100644
index 431c5615827..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaStream.java
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-import java.util.LinkedList;
-import java.util.List;
-
-/** Java wrapper for a C++ MediaStreamInterface. */
-public class MediaStream {
-  public final List<AudioTrack> audioTracks;
-  public final List<VideoTrack> videoTracks;
-  // Package-protected for LocalMediaStream and PeerConnection.
-  final long nativeStream;
-
-  public MediaStream(long nativeStream) {
-    audioTracks = new LinkedList<AudioTrack>();
-    videoTracks = new LinkedList<VideoTrack>();
-    this.nativeStream = nativeStream;
-  }
-
-  public boolean addTrack(AudioTrack track) {
-    if (nativeAddAudioTrack(nativeStream, track.nativeTrack)) {
-      audioTracks.add(track);
-      return true;
-    }
-    return false;
-  }
-
-  public boolean addTrack(VideoTrack track) {
-    if (nativeAddVideoTrack(nativeStream, track.nativeTrack)) {
-      videoTracks.add(track);
-      return true;
-    }
-    return false;
-  }
-
-  public boolean removeTrack(AudioTrack track) {
-    if (nativeRemoveAudioTrack(nativeStream, track.nativeTrack)) {
-      audioTracks.remove(track);
-      return true;
-    }
-    return false;
-  }
-
-  public boolean removeTrack(VideoTrack track) {
-    if (nativeRemoveVideoTrack(nativeStream, track.nativeTrack)) {
-      videoTracks.remove(track);
-      return true;
-    }
-    return false;
-  }
-
-  public void dispose() {
-    for (AudioTrack track : audioTracks) {
-      track.dispose();
-    }
-    audioTracks.clear();
-    for (VideoTrack track : videoTracks) {
-      track.dispose();
-    }
-    videoTracks.clear();
-    free(nativeStream);
-  }
-
-  public String label() {
-    return nativeLabel(nativeStream);
-  }
-
-  public String toString() {
-    return "[" + label() + ":A=" + audioTracks.size() +
-        ":V=" + videoTracks.size() + "]";
-  }
-
-  private static native boolean nativeAddAudioTrack(
-      long nativeStream, long nativeAudioTrack);
-
-  private static native boolean nativeAddVideoTrack(
-      long nativeStream, long nativeVideoTrack);
-
-  private static native boolean nativeRemoveAudioTrack(
-      long nativeStream, long nativeAudioTrack);
-
-  private static native boolean nativeRemoveVideoTrack(
-      long nativeStream, long nativeVideoTrack);
-
-  private static native String nativeLabel(long nativeStream);
-
-  private static native void free(long nativeStream);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
deleted file mode 100644
index 0a0a7f636e8..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
+++ /dev/null
@@ -1,200 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-
-package org.webrtc;
-
-import java.util.LinkedList;
-import java.util.List;
-
-/**
- * Java-land version of the PeerConnection APIs; wraps the C++ API
- * http://www.webrtc.org/reference/native-apis, which in turn is inspired by the
- * JS APIs: http://dev.w3.org/2011/webrtc/editor/webrtc.html and
- * http://www.w3.org/TR/mediacapture-streams/
- */
-public class PeerConnection {
-  static {
-    System.loadLibrary("jingle_peerconnection_so");
-  }
-
-  /** Tracks PeerConnectionInterface::IceGatheringState */
-  public enum IceGatheringState { NEW, GATHERING, COMPLETE };
-
-
-  /** Tracks PeerConnectionInterface::IceConnectionState */
-  public enum IceConnectionState {
-    NEW, CHECKING, CONNECTED, COMPLETED, FAILED, DISCONNECTED, CLOSED
-  };
-
-  /** Tracks PeerConnectionInterface::SignalingState */
-  public enum SignalingState {
-    STABLE, HAVE_LOCAL_OFFER, HAVE_LOCAL_PRANSWER, HAVE_REMOTE_OFFER,
-    HAVE_REMOTE_PRANSWER, CLOSED
-  };
-
-  /** Java version of PeerConnectionObserver. */
-  public static interface Observer {
-    /** Triggered when the SignalingState changes. */
-    public void onSignalingChange(SignalingState newState);
-
-    /** Triggered when the IceConnectionState changes. */
-    public void onIceConnectionChange(IceConnectionState newState);
-
-    /** Triggered when the IceGatheringState changes. */
-    public void onIceGatheringChange(IceGatheringState newState);
-
-    /** Triggered when a new ICE candidate has been found. */
-    public void onIceCandidate(IceCandidate candidate);
-
-    /** Triggered on any error. */
-    public void onError();
-
-    /** Triggered when media is received on a new stream from remote peer. */
-    public void onAddStream(MediaStream stream);
-
-    /** Triggered when a remote peer close a stream. */
-    public void onRemoveStream(MediaStream stream);
-
-    /** Triggered when a remote peer opens a DataChannel. */
-    public void onDataChannel(DataChannel dataChannel);
-  }
-
-  /** Java version of PeerConnectionInterface.IceServer. */
-  public static class IceServer {
-    public final String uri;
-    public final String username;
-    public final String password;
-
-    /** Convenience constructor for STUN servers. */
-    public IceServer(String uri) {
-      this(uri, "", "");
-    }
-
-    public IceServer(String uri, String username, String password) {
-      this.uri = uri;
-      this.username = username;
-      this.password = password;
-    }
-
-    public String toString() {
-      return uri + "[" + username + ":" + password + "]";
-    }
-  }
-
-  private final List<MediaStream> localStreams;
-  private final long nativePeerConnection;
-  private final long nativeObserver;
-
-  PeerConnection(long nativePeerConnection, long nativeObserver) {
-    this.nativePeerConnection = nativePeerConnection;
-    this.nativeObserver = nativeObserver;
-    localStreams = new LinkedList<MediaStream>();
-  }
-
-  // JsepInterface.
-  public native SessionDescription getLocalDescription();
-
-  public native SessionDescription getRemoteDescription();
-
-  public native DataChannel createDataChannel(
-      String label, DataChannel.Init init);
-
-  public native void createOffer(
-      SdpObserver observer, MediaConstraints constraints);
-
-  public native void createAnswer(
-      SdpObserver observer, MediaConstraints constraints);
-
-  public native void setLocalDescription(
-      SdpObserver observer, SessionDescription sdp);
-
-  public native void setRemoteDescription(
-      SdpObserver observer, SessionDescription sdp);
-
-  public native boolean updateIce(
-      List<IceServer> iceServers, MediaConstraints constraints);
-
-  public boolean addIceCandidate(IceCandidate candidate) {
-    return nativeAddIceCandidate(
-        candidate.sdpMid, candidate.sdpMLineIndex, candidate.sdp);
-  }
-
-  public boolean addStream(
-      MediaStream stream, MediaConstraints constraints) {
-    boolean ret = nativeAddLocalStream(stream.nativeStream, constraints);
-    if (!ret) {
-      return false;
-    }
-    localStreams.add(stream);
-    return true;
-  }
-
-  public void removeStream(MediaStream stream) {
-    nativeRemoveLocalStream(stream.nativeStream);
-    localStreams.remove(stream);
-  }
-
-  public boolean getStats(StatsObserver observer, MediaStreamTrack track) {
-    return nativeGetStats(observer, (track == null) ? 0 : track.nativeTrack);
-  }
-
-  // TODO(fischman): add support for DTMF-related methods once that API
-  // stabilizes.
-  public native SignalingState signalingState();
-
-  public native IceConnectionState iceConnectionState();
-
-  public native IceGatheringState iceGatheringState();
-
-  public native void close();
-
-  public void dispose() {
-    close();
-    for (MediaStream stream : localStreams) {
-      stream.dispose();
-    }
-    localStreams.clear();
-    freePeerConnection(nativePeerConnection);
-    freeObserver(nativeObserver);
-  }
-
-  private static native void freePeerConnection(long nativePeerConnection);
-
-  private static native void freeObserver(long nativeObserver);
-
-  private native boolean nativeAddIceCandidate(
-      String sdpMid, int sdpMLineIndex, String iceCandidateSdp);
-
-  private native boolean nativeAddLocalStream(
-      long nativeStream, MediaConstraints constraints);
-
-  private native void nativeRemoveLocalStream(long nativeStream);
-
-  private native boolean nativeGetStats(
-      StatsObserver observer, long nativeTrack);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
deleted file mode 100644
index 03ed03f8e80..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-
-package org.webrtc;
-
-import java.util.List;
-
-/**
- * Java wrapper for a C++ PeerConnectionFactoryInterface.  Main entry point to
- * the PeerConnection API for clients.
- */
-public class PeerConnectionFactory {
-  static {
-    System.loadLibrary("jingle_peerconnection_so");
-  }
-
-  private final long nativeFactory;
-
-  // |context| is an android.content.Context object, but we keep it untyped here
-  // to allow building on non-Android platforms.
-  public static native boolean initializeAndroidGlobals(Object context);
-
-  public PeerConnectionFactory() {
-    nativeFactory = nativeCreatePeerConnectionFactory();
-    if (nativeFactory == 0) {
-      throw new RuntimeException("Failed to initialize PeerConnectionFactory!");
-    }
-  }
-
-
-  public PeerConnection createPeerConnection(
-      List<PeerConnection.IceServer> iceServers,
-      MediaConstraints constraints,
-      PeerConnection.Observer observer) {
-    long nativeObserver = nativeCreateObserver(observer);
-    if (nativeObserver == 0) {
-      return null;
-    }
-    long nativePeerConnection = nativeCreatePeerConnection(
-        nativeFactory, iceServers, constraints, nativeObserver);
-    if (nativePeerConnection == 0) {
-      return null;
-    }
-    return new PeerConnection(nativePeerConnection, nativeObserver);
-  }
-
-  public MediaStream createLocalMediaStream(String label) {
-    return new MediaStream(
-        nativeCreateLocalMediaStream(nativeFactory, label));
-  }
-
-  public VideoSource createVideoSource(
-      VideoCapturer capturer, MediaConstraints constraints) {
-    return new VideoSource(nativeCreateVideoSource(
-        nativeFactory, capturer.nativeVideoCapturer, constraints));
-  }
-
-  public VideoTrack createVideoTrack(String id, VideoSource source) {
-    return new VideoTrack(nativeCreateVideoTrack(
-        nativeFactory, id, source.nativeSource));
-  }
-
-  public AudioTrack createAudioTrack(String id) {
-    return new AudioTrack(nativeCreateAudioTrack(nativeFactory, id));
-  }
-
-  public void dispose() {
-    freeFactory(nativeFactory);
-  }
-
-  private static native long nativeCreatePeerConnectionFactory();
-
-  private static native long nativeCreateObserver(
-      PeerConnection.Observer observer);
-
-  private static native long nativeCreatePeerConnection(
-      long nativeFactory, List<PeerConnection.IceServer> iceServers,
-      MediaConstraints constraints, long nativeObserver);
-
-  private static native long nativeCreateLocalMediaStream(
-      long nativeFactory, String label);
-
-  private static native long nativeCreateVideoSource(
-      long nativeFactory, long nativeVideoCapturer,
-      MediaConstraints constraints);
-
-  private static native long nativeCreateVideoTrack(
-      long nativeFactory, String id, long nativeVideoSource);
-
-  private static native long nativeCreateAudioTrack(
-      long nativeFactory, String id);
-
-  private static native void freeFactory(long nativeFactory);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/SdpObserver.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/SdpObserver.java
deleted file mode 100644
index c9eb14a02bb..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/SdpObserver.java
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-/** Interface for observing SDP-related events. */
-public interface SdpObserver {
-  /** Called on success of Create{Offer,Answer}(). */
-  public void onCreateSuccess(SessionDescription sdp);
-
-  /** Called on success of Set{Local,Remote}Description(). */
-  public void onSetSuccess();
-
-  /** Called on error of Create{Offer,Answer}(). */
-  public void onCreateFailure(String error);
-
-  /** Called on error of Set{Local,Remote}Description(). */
-  public void onSetFailure(String error);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/SessionDescription.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/SessionDescription.java
deleted file mode 100644
index 982db8fc2f3..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/SessionDescription.java
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-
-package org.webrtc;
-
-/**
- * Description of an RFC 4566 Session.
- * SDPs are passed as serialized Strings in Java-land and are materialized
- * to SessionDescriptionInterface as appropriate in the JNI layer.
- */
-public class SessionDescription {
-  /** Java-land enum version of SessionDescriptionInterface's type() string. */
-  public static enum Type {
-    OFFER, PRANSWER, ANSWER;
-
-    public String canonicalForm() {
-      return name().toLowerCase();
-    }
-
-    public static Type fromCanonicalForm(String canonical) {
-      return Type.valueOf(Type.class, canonical.toUpperCase());
-    }
-  }
-
-  public final Type type;
-  public final String description;
-
-  public SessionDescription(Type type, String description) {
-    this.type = type;
-    this.description = description;
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/StatsObserver.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/StatsObserver.java
deleted file mode 100644
index e61d8f74d13..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/StatsObserver.java
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-/** Interface for observing Stats reports (see webrtc::StatsObservers). */
-public interface StatsObserver {
-  /** Called when the reports are ready.*/
-  public void onComplete(StatsReport[] reports);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/StatsReport.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/StatsReport.java
deleted file mode 100644
index 8285ba23250..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/StatsReport.java
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-/** Java version of webrtc::StatsReport. */
-public class StatsReport {
-
-  /** Java version of webrtc::StatsReport::Value. */
-  public static class Value {
-    public final String name;
-    public final String value;
-
-    public Value(String name, String value) {
-      this.name = name;
-      this.value = value;
-    }
-
-    public String toString() {
-      StringBuilder builder = new StringBuilder();
-      builder.append("[").append(name).append(": ").append(value).append("]");
-      return builder.toString();
-    }
-  }
-
-  public final String id;
-  public final String type;
-  // Time since 1970-01-01T00:00:00Z in milliseconds.
-  public final double timestamp;
-  public final Value[] values;
-
-  public StatsReport(String id, String type, double timestamp, Value[] values) {
-    this.id = id;
-    this.type = type;
-    this.timestamp = timestamp;
-    this.values = values;
-  }
-
-  public String toString() {
-    StringBuilder builder = new StringBuilder();
-    builder.append("id: ").append(id).append(", type: ").append(type)
-        .append(", timestamp: ").append(timestamp).append(", values: ");
-    for (int i = 0; i < values.length; ++i) {
-      builder.append(values[i].toString()).append(", ");
-    }
-    return builder.toString();
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java
deleted file mode 100644
index eab5797bfda..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoCapturer.java
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-/** Java version of VideoCapturerInterface. */
-public class VideoCapturer {
-  final long nativeVideoCapturer;
-
-  private VideoCapturer(long nativeVideoCapturer) {
-    this.nativeVideoCapturer = nativeVideoCapturer;
-  }
-
-  public static VideoCapturer create(String deviceName) {
-    long nativeVideoCapturer = nativeCreateVideoCapturer(deviceName);
-    if (nativeVideoCapturer == 0) {
-      return null;
-    }
-    return new VideoCapturer(nativeVideoCapturer);
-  }
-
-  public void dispose() {
-    free(nativeVideoCapturer);
-  }
-
-  private static native long nativeCreateVideoCapturer(String deviceName);
-
-  private static native void free(long nativeVideoCapturer);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java
deleted file mode 100644
index 4cc341a48e1..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-import java.nio.ByteBuffer;
-import java.util.Arrays;
-
-/**
- * Java version of VideoRendererInterface.  In addition to allowing clients to
- * define their own rendering behavior (by passing in a Callbacks object), this
- * class also provides a createGui() method for creating a GUI-rendering window
- * on various platforms.
- */
-public class VideoRenderer {
-
-  /** Java version of cricket::VideoFrame. */
-  public static class I420Frame {
-    public final int width;
-    public final int height;
-    public final int[] yuvStrides;
-    public final ByteBuffer[] yuvPlanes;
-
-    /**
-     * Construct a frame of the given dimensions with the specified planar
-     * data.  If |yuvPlanes| is null, new planes of the appropriate sizes are
-     * allocated.
-     */
-    public I420Frame(
-        int width, int height, int[] yuvStrides, ByteBuffer[] yuvPlanes) {
-      this.width = width;
-      this.height = height;
-      this.yuvStrides = yuvStrides;
-      if (yuvPlanes == null) {
-        yuvPlanes = new ByteBuffer[3];
-        yuvPlanes[0] = ByteBuffer.allocateDirect(yuvStrides[0] * height);
-        yuvPlanes[1] = ByteBuffer.allocateDirect(yuvStrides[1] * height);
-        yuvPlanes[2] = ByteBuffer.allocateDirect(yuvStrides[2] * height);
-      }
-      this.yuvPlanes = yuvPlanes;
-    }
-
-    /**
-     * Copy the planes out of |source| into |this| and return |this|.  Calling
-     * this with mismatched frame dimensions is a programming error and will
-     * likely crash.
-     */
-    public I420Frame copyFrom(I420Frame source) {
-      if (!Arrays.equals(yuvStrides, source.yuvStrides) ||
-          width != source.width || height != source.height) {
-        throw new RuntimeException("Mismatched dimensions!  Source: " +
-            source.toString() + ", destination: " + toString());
-      }
-      copyPlane(source.yuvPlanes[0], yuvPlanes[0]);
-      copyPlane(source.yuvPlanes[1], yuvPlanes[1]);
-      copyPlane(source.yuvPlanes[2], yuvPlanes[2]);
-      return this;
-    }
-
-    @Override
-    public String toString() {
-      return width + "x" + height + ":" + yuvStrides[0] + ":" + yuvStrides[1] +
-          ":" + yuvStrides[2];
-    }
-
-    // Copy the bytes out of |src| and into |dst|, ignoring and overwriting
-    // positon & limit in both buffers.
-    private void copyPlane(ByteBuffer src, ByteBuffer dst) {
-      src.position(0).limit(src.capacity());
-      dst.put(src);
-      dst.position(0).limit(dst.capacity());
-    }
-}
-
-  /** The real meat of VideoRendererInterface. */
-  public static interface Callbacks {
-    public void setSize(int width, int height);
-    public void renderFrame(I420Frame frame);
-  }
-
-  // |this| either wraps a native (GUI) renderer or a client-supplied Callbacks
-  // (Java) implementation; so exactly one of these will be non-0/null.
-  final long nativeVideoRenderer;
-  private final Callbacks callbacks;
-
-  public static VideoRenderer createGui(int x, int y) {
-    long nativeVideoRenderer = nativeCreateGuiVideoRenderer(x, y);
-    if (nativeVideoRenderer == 0) {
-      return null;
-    }
-    return new VideoRenderer(nativeVideoRenderer);
-  }
-
-  public VideoRenderer(Callbacks callbacks) {
-    nativeVideoRenderer = nativeWrapVideoRenderer(callbacks);
-    this.callbacks = callbacks;
-  }
-
-  private VideoRenderer(long nativeVideoRenderer) {
-    this.nativeVideoRenderer = nativeVideoRenderer;
-    callbacks = null;
-  }
-
-  public void dispose() {
-    free(nativeVideoRenderer);
-  }
-
-  private static native long nativeCreateGuiVideoRenderer(int x, int y);
-  private static native long nativeWrapVideoRenderer(Callbacks callbacks);
-
-  private static native void free(long nativeVideoRenderer);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoSource.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoSource.java
deleted file mode 100644
index f29f312c6b7..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoSource.java
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-
-package org.webrtc;
-
-/** Java version of VideoSourceInterface. */
-public class VideoSource extends MediaSource {
-  public VideoSource(long nativeSource) {
-    super(nativeSource);
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java b/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java
deleted file mode 100644
index 90e5c956588..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/VideoTrack.java
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-import java.util.LinkedList;
-
-/** Java version of VideoTrackInterface. */
-public class VideoTrack extends MediaStreamTrack {
-  private final LinkedList<VideoRenderer> renderers;
-
-  public VideoTrack(long nativeTrack) {
-    super(nativeTrack);
-    renderers = new LinkedList<VideoRenderer>();
-  }
-
-  public void addRenderer(VideoRenderer renderer) {
-    renderers.add(renderer);
-    nativeAddRenderer(nativeTrack, renderer.nativeVideoRenderer);
-  }
-
-  public void removeRenderer(VideoRenderer renderer) {
-    if (!renderers.remove(renderer)) {
-      return;
-    }
-    nativeRemoveRenderer(nativeTrack, renderer.nativeVideoRenderer);
-    renderer.dispose();
-  }
-
-  public void dispose() {
-    while (!renderers.isEmpty()) {
-      removeRenderer(renderers.getFirst());
-    }
-  }
-
-  private static native void nativeAddRenderer(
-      long nativeTrack, long nativeRenderer);
-
-  private static native void nativeRemoveRenderer(
-      long nativeTrack, long nativeRenderer);
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java b/chromium/third_party/libjingle/source/talk/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
deleted file mode 100644
index dd4ef487fb0..00000000000
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java
+++ /dev/null
@@ -1,723 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.webrtc;
-
-import junit.framework.TestCase;
-
-import org.junit.Test;
-import org.webrtc.PeerConnection.IceConnectionState;
-import org.webrtc.PeerConnection.IceGatheringState;
-import org.webrtc.PeerConnection.SignalingState;
-
-import java.lang.ref.WeakReference;
-import java.nio.ByteBuffer;
-import java.nio.charset.Charset;
-import java.util.Arrays;
-import java.util.EnumSet;
-import java.util.IdentityHashMap;
-import java.util.LinkedList;
-import java.util.Map;
-import java.util.TreeSet;
-import java.util.concurrent.CountDownLatch;
-import java.util.concurrent.TimeUnit;
-
-/** End-to-end tests for PeerConnection.java. */
-public class PeerConnectionTest extends TestCase {
-  // Set to true to render video.
-  private static final boolean RENDER_TO_GUI = false;
-
-  private static class ObserverExpectations implements PeerConnection.Observer,
-                                            VideoRenderer.Callbacks,
-                                            DataChannel.Observer,
-                                            StatsObserver {
-    private final String name;
-    private int expectedIceCandidates = 0;
-    private int expectedErrors = 0;
-    private int expectedSetSize = 0;
-    private int previouslySeenWidth = 0;
-    private int previouslySeenHeight = 0;
-    private int expectedFramesDelivered = 0;
-    private LinkedList<SignalingState> expectedSignalingChanges =
-        new LinkedList<SignalingState>();
-    private LinkedList<IceConnectionState> expectedIceConnectionChanges =
-        new LinkedList<IceConnectionState>();
-    private LinkedList<IceGatheringState> expectedIceGatheringChanges =
-        new LinkedList<IceGatheringState>();
-    private LinkedList<String> expectedAddStreamLabels =
-        new LinkedList<String>();
-    private LinkedList<String> expectedRemoveStreamLabels =
-        new LinkedList<String>();
-    public LinkedList<IceCandidate> gotIceCandidates =
-        new LinkedList<IceCandidate>();
-    private Map<MediaStream, WeakReference<VideoRenderer>> renderers =
-        new IdentityHashMap<MediaStream, WeakReference<VideoRenderer>>();
-    private DataChannel dataChannel;
-    private LinkedList<DataChannel.Buffer> expectedBuffers =
-        new LinkedList<DataChannel.Buffer>();
-    private LinkedList<DataChannel.State> expectedStateChanges =
-        new LinkedList<DataChannel.State>();
-    private LinkedList<String> expectedRemoteDataChannelLabels =
-        new LinkedList<String>();
-    private int expectedStatsCallbacks = 0;
-    private LinkedList<StatsReport[]> gotStatsReports =
-        new LinkedList<StatsReport[]>();
-
-    public ObserverExpectations(String name) {
-      this.name = name;
-    }
-
-    public synchronized void setDataChannel(DataChannel dataChannel) {
-      assertNull(this.dataChannel);
-      this.dataChannel = dataChannel;
-      this.dataChannel.registerObserver(this);
-      assertNotNull(this.dataChannel);
-    }
-
-    public synchronized void expectIceCandidates(int count) {
-      expectedIceCandidates += count;
-    }
-
-    public synchronized void onIceCandidate(IceCandidate candidate) {
-      --expectedIceCandidates;
-      // We don't assert expectedIceCandidates >= 0 because it's hard to know
-      // how many to expect, in general.  We only use expectIceCandidates to
-      // assert a minimal count.
-      gotIceCandidates.add(candidate);
-    }
-
-    public synchronized void expectError() {
-      ++expectedErrors;
-    }
-
-    public synchronized void onError() {
-      assertTrue(--expectedErrors >= 0);
-    }
-
-    public synchronized void expectSetSize() {
-      if (RENDER_TO_GUI) {
-        // When new frames are delivered to the GUI renderer we don't get
-        // notified of frame size info.
-        return;
-      }
-      ++expectedSetSize;
-    }
-
-    @Override
-    public synchronized void setSize(int width, int height) {
-      assertFalse(RENDER_TO_GUI);
-      assertTrue(--expectedSetSize >= 0);
-      // Because different camera devices (fake & physical) produce different
-      // resolutions, we only sanity-check the set sizes,
-      assertTrue(width > 0);
-      assertTrue(height > 0);
-      if (previouslySeenWidth > 0) {
-        assertEquals(previouslySeenWidth, width);
-        assertEquals(previouslySeenHeight, height);
-      } else {
-        previouslySeenWidth = width;
-        previouslySeenHeight = height;
-      }
-    }
-
-    public synchronized void expectFramesDelivered(int count) {
-      assertFalse(RENDER_TO_GUI);
-      expectedFramesDelivered += count;
-    }
-
-    @Override
-    public synchronized void renderFrame(VideoRenderer.I420Frame frame) {
-      --expectedFramesDelivered;
-    }
-
-    public synchronized void expectSignalingChange(SignalingState newState) {
-      expectedSignalingChanges.add(newState);
-    }
-
-    @Override
-    public synchronized void onSignalingChange(SignalingState newState) {
-      assertEquals(expectedSignalingChanges.removeFirst(), newState);
-    }
-
-    public synchronized void expectIceConnectionChange(
-        IceConnectionState newState) {
-      expectedIceConnectionChanges.add(newState);
-    }
-
-    @Override
-    public synchronized void onIceConnectionChange(
-        IceConnectionState newState) {
-      assertEquals(expectedIceConnectionChanges.removeFirst(), newState);
-    }
-
-    public synchronized void expectIceGatheringChange(
-        IceGatheringState newState) {
-      expectedIceGatheringChanges.add(newState);
-    }
-
-    @Override
-    public synchronized void onIceGatheringChange(IceGatheringState newState) {
-      // It's fine to get a variable number of GATHERING messages before
-      // COMPLETE fires (depending on how long the test runs) so we don't assert
-      // any particular count.
-      if (newState == IceGatheringState.GATHERING) {
-        return;
-      }
-      assertEquals(expectedIceGatheringChanges.removeFirst(), newState);
-    }
-
-    public synchronized void expectAddStream(String label) {
-      expectedAddStreamLabels.add(label);
-    }
-
-    public synchronized void onAddStream(MediaStream stream) {
-      assertEquals(expectedAddStreamLabels.removeFirst(), stream.label());
-      assertEquals(1, stream.videoTracks.size());
-      assertEquals(1, stream.audioTracks.size());
-      assertTrue(stream.videoTracks.get(0).id().endsWith("LMSv0"));
-      assertTrue(stream.audioTracks.get(0).id().endsWith("LMSa0"));
-      assertEquals("video", stream.videoTracks.get(0).kind());
-      assertEquals("audio", stream.audioTracks.get(0).kind());
-      VideoRenderer renderer = createVideoRenderer(this);
-      stream.videoTracks.get(0).addRenderer(renderer);
-      assertNull(renderers.put(
-          stream, new WeakReference<VideoRenderer>(renderer)));
-    }
-
-    public synchronized void expectRemoveStream(String label) {
-      expectedRemoveStreamLabels.add(label);
-    }
-
-    public synchronized void onRemoveStream(MediaStream stream) {
-      assertEquals(expectedRemoveStreamLabels.removeFirst(), stream.label());
-      WeakReference<VideoRenderer> renderer = renderers.remove(stream);
-      assertNotNull(renderer);
-      assertNotNull(renderer.get());
-      assertEquals(1, stream.videoTracks.size());
-      stream.videoTracks.get(0).removeRenderer(renderer.get());
-    }
-
-    public synchronized void expectDataChannel(String label) {
-      expectedRemoteDataChannelLabels.add(label);
-    }
-
-    @Override
-    public synchronized void onDataChannel(DataChannel remoteDataChannel) {
-      assertEquals(expectedRemoteDataChannelLabels.removeFirst(),
-                   remoteDataChannel.label());
-      setDataChannel(remoteDataChannel);
-      assertEquals(DataChannel.State.CONNECTING, dataChannel.state());
-    }
-
-    public synchronized void expectMessage(ByteBuffer expectedBuffer,
-                                           boolean expectedBinary) {
-      expectedBuffers.add(
-          new DataChannel.Buffer(expectedBuffer, expectedBinary));
-    }
-
-    @Override
-    public synchronized void onMessage(DataChannel.Buffer buffer) {
-      DataChannel.Buffer expected = expectedBuffers.removeFirst();
-      assertEquals(expected.binary, buffer.binary);
-      assertTrue(expected.data.equals(buffer.data));
-    }
-
-    @Override
-    public synchronized void onStateChange() {
-      assertEquals(expectedStateChanges.removeFirst(), dataChannel.state());
-    }
-
-    public synchronized void expectStateChange(DataChannel.State state) {
-      expectedStateChanges.add(state);
-    }
-
-    @Override
-    public synchronized void onComplete(StatsReport[] reports) {
-      if (--expectedStatsCallbacks < 0) {
-        throw new RuntimeException("Unexpected stats report: " + reports);
-      }
-      gotStatsReports.add(reports);
-    }
-
-    public synchronized void expectStatsCallback() {
-      ++expectedStatsCallbacks;
-    }
-
-    public synchronized LinkedList<StatsReport[]> takeStatsReports() {
-      LinkedList<StatsReport[]> got = gotStatsReports;
-      gotStatsReports = new LinkedList<StatsReport[]>();
-      return got;
-    }
-
-    // Return a set of expectations that haven't been satisfied yet, possibly
-    // empty if no such expectations exist.
-    public synchronized TreeSet<String> unsatisfiedExpectations() {
-      TreeSet<String> stillWaitingForExpectations = new TreeSet<String>();
-      if (expectedIceCandidates > 0) {  // See comment in onIceCandidate.
-        stillWaitingForExpectations.add("expectedIceCandidates");
-      }
-      if (expectedErrors != 0) {
-        stillWaitingForExpectations.add("expectedErrors: " + expectedErrors);
-      }
-      if (expectedSignalingChanges.size() != 0) {
-        stillWaitingForExpectations.add(
-            "expectedSignalingChanges: " + expectedSignalingChanges.size());
-      }
-      if (expectedIceConnectionChanges.size() != 0) {
-        stillWaitingForExpectations.add("expectedIceConnectionChanges: " +
-                                        expectedIceConnectionChanges.size());
-      }
-      if (expectedIceGatheringChanges.size() != 0) {
-        stillWaitingForExpectations.add("expectedIceGatheringChanges: " +
-                                        expectedIceGatheringChanges.size());
-      }
-      if (expectedAddStreamLabels.size() != 0) {
-        stillWaitingForExpectations.add(
-            "expectedAddStreamLabels: " + expectedAddStreamLabels.size());
-      }
-      if (expectedRemoveStreamLabels.size() != 0) {
-        stillWaitingForExpectations.add(
-            "expectedRemoveStreamLabels: " + expectedRemoveStreamLabels.size());
-      }
-      if (expectedSetSize != 0) {
-        stillWaitingForExpectations.add("expectedSetSize");
-      }
-      if (expectedFramesDelivered > 0) {
-        stillWaitingForExpectations.add(
-            "expectedFramesDelivered: " + expectedFramesDelivered);
-      }
-      if (!expectedBuffers.isEmpty()) {
-        stillWaitingForExpectations.add(
-            "expectedBuffers: " + expectedBuffers.size());
-      }
-      if (!expectedStateChanges.isEmpty()) {
-        stillWaitingForExpectations.add(
-            "expectedStateChanges: " + expectedStateChanges.size());
-      }
-      if (!expectedRemoteDataChannelLabels.isEmpty()) {
-        stillWaitingForExpectations.add("expectedRemoteDataChannelLabels: " +
-                                        expectedRemoteDataChannelLabels.size());
-      }
-      if (expectedStatsCallbacks != 0) {
-        stillWaitingForExpectations.add(
-            "expectedStatsCallbacks: " + expectedStatsCallbacks);
-      }
-      return stillWaitingForExpectations;
-    }
-
-    public void waitForAllExpectationsToBeSatisfied() {
-      // TODO(fischman): problems with this approach:
-      // - come up with something better than a poll loop
-      // - avoid serializing expectations explicitly; the test is not as robust
-      //   as it could be because it must place expectations between wait
-      //   statements very precisely (e.g. frame must not arrive before its
-      //   expectation, and expectation must not be registered so early as to
-      //   stall a wait).  Use callbacks to fire off dependent steps instead of
-      //   explicitly waiting, so there can be just a single wait at the end of
-      //   the test.
-      TreeSet<String> prev = null;
-      TreeSet<String> stillWaitingForExpectations = unsatisfiedExpectations();
-      while (!stillWaitingForExpectations.isEmpty()) {
-        if (!stillWaitingForExpectations.equals(prev)) {
-          System.out.println(
-              name + " still waiting at\n    " +
-              (new Throwable()).getStackTrace()[1] +
-              "\n    for: " +
-              Arrays.toString(stillWaitingForExpectations.toArray()));
-        }
-        try {
-          Thread.sleep(10);
-        } catch (InterruptedException e) {
-          throw new RuntimeException(e);
-        }
-        prev = stillWaitingForExpectations;
-        stillWaitingForExpectations = unsatisfiedExpectations();
-      }
-      if (prev == null) {
-        System.out.println(name + " didn't need to wait at\n    " +
-                           (new Throwable()).getStackTrace()[1]);
-      }
-    }
-  }
-
-  private static class SdpObserverLatch implements SdpObserver {
-    private boolean success = false;
-    private SessionDescription sdp = null;
-    private String error = null;
-    private CountDownLatch latch = new CountDownLatch(1);
-
-    public SdpObserverLatch() {}
-
-    public void onCreateSuccess(SessionDescription sdp) {
-      this.sdp = sdp;
-      onSetSuccess();
-    }
-
-    public void onSetSuccess() {
-      success = true;
-      latch.countDown();
-    }
-
-    public void onCreateFailure(String error) {
-      onSetFailure(error);
-    }
-
-    public void onSetFailure(String error) {
-      this.error = error;
-      latch.countDown();
-    }
-
-    public boolean await() {
-      try {
-        assertTrue(latch.await(1000, TimeUnit.MILLISECONDS));
-        return getSuccess();
-      } catch (Exception e) {
-        throw new RuntimeException(e);
-      }
-    }
-
-    public boolean getSuccess() {
-      return success;
-    }
-
-    public SessionDescription getSdp() {
-      return sdp;
-    }
-
-    public String getError() {
-      return error;
-    }
-  }
-
-  static int videoWindowsMapped = -1;
-
-  private static class TestRenderer implements VideoRenderer.Callbacks {
-    public int width = -1;
-    public int height = -1;
-    public int numFramesDelivered = 0;
-
-    public void setSize(int width, int height) {
-      assertEquals(this.width, -1);
-      assertEquals(this.height, -1);
-      this.width = width;
-      this.height = height;
-    }
-
-    public void renderFrame(VideoRenderer.I420Frame frame) {
-      ++numFramesDelivered;
-    }
-  }
-
-  private static VideoRenderer createVideoRenderer(
-      VideoRenderer.Callbacks videoCallbacks) {
-    if (!RENDER_TO_GUI) {
-      return new VideoRenderer(videoCallbacks);
-    }
-    ++videoWindowsMapped;
-    assertTrue(videoWindowsMapped < 4);
-    int x = videoWindowsMapped % 2 != 0 ? 700 : 0;
-    int y = videoWindowsMapped >= 2 ? 0 : 500;
-    return VideoRenderer.createGui(x, y);
-  }
-
-  // Return a weak reference to test that ownership is correctly held by
-  // PeerConnection, not by test code.
-  private static WeakReference<MediaStream> addTracksToPC(
-      PeerConnectionFactory factory, PeerConnection pc,
-      VideoSource videoSource,
-      String streamLabel, String videoTrackId, String audioTrackId,
-      VideoRenderer.Callbacks videoCallbacks) {
-    MediaStream lMS = factory.createLocalMediaStream(streamLabel);
-    VideoTrack videoTrack =
-        factory.createVideoTrack(videoTrackId, videoSource);
-    assertNotNull(videoTrack);
-    VideoRenderer videoRenderer = createVideoRenderer(videoCallbacks);
-    assertNotNull(videoRenderer);
-    videoTrack.addRenderer(videoRenderer);
-    lMS.addTrack(videoTrack);
-    // Just for fun, let's remove and re-add the track.
-    lMS.removeTrack(videoTrack);
-    lMS.addTrack(videoTrack);
-    lMS.addTrack(factory.createAudioTrack(audioTrackId));
-    pc.addStream(lMS, new MediaConstraints());
-    return new WeakReference<MediaStream>(lMS);
-  }
-
-  private static void assertEquals(
-      SessionDescription lhs, SessionDescription rhs) {
-    assertEquals(lhs.type, rhs.type);
-    assertEquals(lhs.description, rhs.description);
-  }
-
-  @Test
-  public void testCompleteSession() throws Exception {
-    // Uncomment to get ALL WebRTC tracing and SENSITIVE libjingle logging.
-    // Logging.enableTracing(
-    //     "/tmp/AMI-nope.txt",
-    //     EnumSet.of(Logging.TraceLevel.TRACE_ALL),
-    //     Logging.Severity.LS_SENSITIVE);
-
-    CountDownLatch testDone = new CountDownLatch(1);
-
-    PeerConnectionFactory factory = new PeerConnectionFactory();
-    MediaConstraints pcConstraints = new MediaConstraints();
-    pcConstraints.mandatory.add(
-        new MediaConstraints.KeyValuePair("DtlsSrtpKeyAgreement", "true"));
-    pcConstraints.optional.add(
-        new MediaConstraints.KeyValuePair("RtpDataChannels", "true"));
-    // TODO(fischman): replace above with below to test SCTP channels when
-    // supported (https://code.google.com/p/webrtc/issues/detail?id=1408).
-    // pcConstraints.optional.add(new MediaConstraints.KeyValuePair(
-    //     "internalSctpDataChannels", "true"));
-
-    LinkedList<PeerConnection.IceServer> iceServers =
-        new LinkedList<PeerConnection.IceServer>();
-    iceServers.add(new PeerConnection.IceServer(
-        "stun:stun.l.google.com:19302"));
-    iceServers.add(new PeerConnection.IceServer(
-        "turn:fake.example.com", "fakeUsername", "fakePassword"));
-    ObserverExpectations offeringExpectations =
-        new ObserverExpectations("PCTest:offerer");
-    PeerConnection offeringPC = factory.createPeerConnection(
-        iceServers, pcConstraints, offeringExpectations);
-    assertNotNull(offeringPC);
-
-    ObserverExpectations answeringExpectations =
-        new ObserverExpectations("PCTest:answerer");
-    PeerConnection answeringPC = factory.createPeerConnection(
-        iceServers, pcConstraints, answeringExpectations);
-    assertNotNull(answeringPC);
-
-    // We want to use the same camera for offerer & answerer, so create it here
-    // instead of in addTracksToPC.
-    VideoSource videoSource = factory.createVideoSource(
-        VideoCapturer.create(""), new MediaConstraints());
-
-    // TODO(fischman): the track ids here and in the addTracksToPC() call
-    // below hard-code the <mediaStreamLabel>[av]<index> scheme used in the
-    // serialized SDP, because the C++ API doesn't auto-translate.
-    // Drop |label| params from {Audio,Video}Track-related APIs once
-    // https://code.google.com/p/webrtc/issues/detail?id=1253 is fixed.
-    offeringExpectations.expectSetSize();
-    WeakReference<MediaStream> oLMS = addTracksToPC(
-        factory, offeringPC, videoSource, "oLMS", "oLMSv0", "oLMSa0",
-        offeringExpectations);
-
-    offeringExpectations.setDataChannel(offeringPC.createDataChannel(
-        "offeringDC", new DataChannel.Init()));
-
-    SdpObserverLatch sdpLatch = new SdpObserverLatch();
-    offeringPC.createOffer(sdpLatch, new MediaConstraints());
-    assertTrue(sdpLatch.await());
-    SessionDescription offerSdp = sdpLatch.getSdp();
-    assertEquals(offerSdp.type, SessionDescription.Type.OFFER);
-    assertFalse(offerSdp.description.isEmpty());
-
-    sdpLatch = new SdpObserverLatch();
-    answeringExpectations.expectSignalingChange(
-        SignalingState.HAVE_REMOTE_OFFER);
-    answeringExpectations.expectAddStream("oLMS");
-    answeringExpectations.expectDataChannel("offeringDC");
-    answeringPC.setRemoteDescription(sdpLatch, offerSdp);
-    assertEquals(
-        PeerConnection.SignalingState.STABLE, offeringPC.signalingState());
-    assertTrue(sdpLatch.await());
-    assertNull(sdpLatch.getSdp());
-
-    answeringExpectations.expectSetSize();
-    WeakReference<MediaStream> aLMS = addTracksToPC(
-        factory, answeringPC, videoSource, "aLMS", "aLMSv0", "aLMSa0",
-        answeringExpectations);
-
-    sdpLatch = new SdpObserverLatch();
-    answeringPC.createAnswer(sdpLatch, new MediaConstraints());
-    assertTrue(sdpLatch.await());
-    SessionDescription answerSdp = sdpLatch.getSdp();
-    assertEquals(answerSdp.type, SessionDescription.Type.ANSWER);
-    assertFalse(answerSdp.description.isEmpty());
-
-    offeringExpectations.expectIceCandidates(2);
-    answeringExpectations.expectIceCandidates(2);
-
-    offeringExpectations.expectIceGatheringChange(IceGatheringState.COMPLETE);
-    answeringExpectations.expectIceGatheringChange(IceGatheringState.COMPLETE);
-
-    sdpLatch = new SdpObserverLatch();
-    answeringExpectations.expectSignalingChange(SignalingState.STABLE);
-    answeringPC.setLocalDescription(sdpLatch, answerSdp);
-    assertTrue(sdpLatch.await());
-    assertNull(sdpLatch.getSdp());
-
-    sdpLatch = new SdpObserverLatch();
-    offeringExpectations.expectSignalingChange(SignalingState.HAVE_LOCAL_OFFER);
-    offeringPC.setLocalDescription(sdpLatch, offerSdp);
-    assertTrue(sdpLatch.await());
-    assertNull(sdpLatch.getSdp());
-    sdpLatch = new SdpObserverLatch();
-    offeringExpectations.expectSignalingChange(SignalingState.STABLE);
-    offeringExpectations.expectAddStream("aLMS");
-    offeringPC.setRemoteDescription(sdpLatch, answerSdp);
-    assertTrue(sdpLatch.await());
-    assertNull(sdpLatch.getSdp());
-
-    offeringExpectations.waitForAllExpectationsToBeSatisfied();
-    answeringExpectations.waitForAllExpectationsToBeSatisfied();
-
-    assertEquals(offeringPC.getLocalDescription().type, offerSdp.type);
-    assertEquals(offeringPC.getRemoteDescription().type, answerSdp.type);
-    assertEquals(answeringPC.getLocalDescription().type, answerSdp.type);
-    assertEquals(answeringPC.getRemoteDescription().type, offerSdp.type);
-
-    if (!RENDER_TO_GUI) {
-      // Wait for at least some frames to be delivered at each end (number
-      // chosen arbitrarily).
-      offeringExpectations.expectFramesDelivered(10);
-      answeringExpectations.expectFramesDelivered(10);
-      offeringExpectations.expectSetSize();
-      answeringExpectations.expectSetSize();
-    }
-
-    offeringExpectations.expectIceConnectionChange(
-        IceConnectionState.CHECKING);
-    offeringExpectations.expectIceConnectionChange(
-        IceConnectionState.CONNECTED);
-    answeringExpectations.expectIceConnectionChange(
-        IceConnectionState.CHECKING);
-    answeringExpectations.expectIceConnectionChange(
-        IceConnectionState.CONNECTED);
-
-    offeringExpectations.expectStateChange(DataChannel.State.OPEN);
-    answeringExpectations.expectStateChange(DataChannel.State.OPEN);
-
-    for (IceCandidate candidate : offeringExpectations.gotIceCandidates) {
-      answeringPC.addIceCandidate(candidate);
-    }
-    offeringExpectations.gotIceCandidates.clear();
-    for (IceCandidate candidate : answeringExpectations.gotIceCandidates) {
-      offeringPC.addIceCandidate(candidate);
-    }
-    answeringExpectations.gotIceCandidates.clear();
-
-    offeringExpectations.waitForAllExpectationsToBeSatisfied();
-    answeringExpectations.waitForAllExpectationsToBeSatisfied();
-
-    assertEquals(
-        PeerConnection.SignalingState.STABLE, offeringPC.signalingState());
-    assertEquals(
-        PeerConnection.SignalingState.STABLE, answeringPC.signalingState());
-
-    // Test send & receive UTF-8 text.
-    answeringExpectations.expectMessage(
-        ByteBuffer.wrap("hello!".getBytes(Charset.forName("UTF-8"))), false);
-    DataChannel.Buffer buffer = new DataChannel.Buffer(
-        ByteBuffer.wrap("hello!".getBytes(Charset.forName("UTF-8"))), false);
-    assertTrue(offeringExpectations.dataChannel.send(buffer));
-    answeringExpectations.waitForAllExpectationsToBeSatisfied();
-
-    // TODO(fischman): add testing of binary messages when SCTP channels are
-    // supported (https://code.google.com/p/webrtc/issues/detail?id=1408).
-    // // Construct this binary message two different ways to ensure no
-    // // shortcuts are taken.
-    // ByteBuffer expectedBinaryMessage = ByteBuffer.allocateDirect(5);
-    // for (byte i = 1; i < 6; ++i) {
-    //   expectedBinaryMessage.put(i);
-    // }
-    // expectedBinaryMessage.flip();
-    // offeringExpectations.expectMessage(expectedBinaryMessage, true);
-    // assertTrue(answeringExpectations.dataChannel.send(
-    //     new DataChannel.Buffer(
-    //         ByteBuffer.wrap(new byte[] { 1, 2, 3, 4, 5 } ), true)));
-    // offeringExpectations.waitForAllExpectationsToBeSatisfied();
-
-    offeringExpectations.expectStateChange(DataChannel.State.CLOSING);
-    answeringExpectations.expectStateChange(DataChannel.State.CLOSING);
-    answeringExpectations.dataChannel.close();
-    offeringExpectations.dataChannel.close();
-    // TODO(fischman): implement a new offer/answer exchange to finalize the
-    // closing of the channel in order to see the CLOSED state reached.
-    // offeringExpectations.expectStateChange(DataChannel.State.CLOSED);
-    // answeringExpectations.expectStateChange(DataChannel.State.CLOSED);
-
-    if (RENDER_TO_GUI) {
-      try {
-        Thread.sleep(3000);
-      } catch (Throwable t) {
-        throw new RuntimeException(t);
-      }
-    }
-
-    // TODO(fischman) MOAR test ideas:
-    // - Test that PC.removeStream() works; requires a second
-    //   createOffer/createAnswer dance.
-    // - audit each place that uses |constraints| for specifying non-trivial
-    //   constraints (and ensure they're honored).
-    // - test error cases
-    // - ensure reasonable coverage of _jni.cc is achieved.  Coverage is
-    //   extra-important because of all the free-text (class/method names, etc)
-    //   in JNI-style programming; make sure no typos!
-    // - Test that shutdown mid-interaction is crash-free.
-
-    // Free the Java-land objects, collect them, and sleep a bit to make sure we
-    // don't get late-arrival crashes after the Java-land objects have been
-    // freed.
-    shutdownPC(offeringPC, offeringExpectations);
-    offeringPC = null;
-    shutdownPC(answeringPC, answeringExpectations);
-    answeringPC = null;
-    System.gc();
-    Thread.sleep(100);
-  }
-
-  private static void shutdownPC(
-      PeerConnection pc, ObserverExpectations expectations) {
-    expectations.dataChannel.unregisterObserver();
-    expectations.dataChannel.dispose();
-    expectations.expectStatsCallback();
-    assertTrue(pc.getStats(expectations, null));
-    expectations.waitForAllExpectationsToBeSatisfied();
-    expectations.expectIceConnectionChange(IceConnectionState.CLOSED);
-    expectations.expectSignalingChange(SignalingState.CLOSED);
-    pc.close();
-    expectations.waitForAllExpectationsToBeSatisfied();
-    expectations.expectStatsCallback();
-    assertTrue(pc.getStats(expectations, null));
-    expectations.waitForAllExpectationsToBeSatisfied();
-
-    System.out.println("FYI stats: ");
-    int reportIndex = -1;
-    for (StatsReport[] reports : expectations.takeStatsReports()) {
-      System.out.println(" Report #" + (++reportIndex));
-      for (int i = 0; i < reports.length; ++i) {
-        System.out.println("  " + reports[i].toString());
-      }
-    }
-    assertEquals(1, reportIndex);
-    System.out.println("End stats.");
-
-    pc.dispose();
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription_unittest.cc
index 83f67cb3add..e2b59fba205 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/jsepsessiondescription_unittest.cc
@@ -79,16 +79,18 @@ static cricket::SessionDescription* CreateCricketSessionDescription() {
                                  cricket::NS_GINGLE_P2P,
                                  std::vector<std::string>(),
                                  kCandidateUfragVoice, kCandidatePwdVoice,
-                                 cricket::ICEMODE_FULL, NULL,
-                                 cricket::Candidates()))));
+                                 cricket::ICEMODE_FULL,
+                                 cricket::CONNECTIONROLE_NONE,
+                                 NULL, cricket::Candidates()))));
   EXPECT_TRUE(desc->AddTransportInfo(
       cricket::TransportInfo(cricket::CN_VIDEO,
                              cricket::TransportDescription(
                                  cricket::NS_GINGLE_P2P,
                                  std::vector<std::string>(),
                                  kCandidateUfragVideo, kCandidatePwdVideo,
-                                 cricket::ICEMODE_FULL, NULL,
-                                 cricket::Candidates()))));
+                                 cricket::ICEMODE_FULL,
+                                 cricket::CONNECTIONROLE_NONE,
+                                 NULL, cricket::Candidates()))));
   return desc;
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc
index 9706c076796..3663aace528 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/localaudiosource.cc
@@ -51,6 +51,8 @@ const char MediaConstraintsInterface::kNoiseSuppression[] =
     "googNoiseSuppression";
 const char MediaConstraintsInterface::kHighpassFilter[] =
     "googHighpassFilter";
+const char MediaConstraintsInterface::kTypingNoiseDetection[] =
+    "googTypingNoiseDetection";
 const char MediaConstraintsInterface::kInternalAecDump[] = "internalAecDump";
 
 namespace {
@@ -90,6 +92,8 @@ bool FromConstraints(const MediaConstraintsInterface::Constraints& constraints,
       options->highpass_filter.Set(value);
     else if (iter->key == MediaConstraintsInterface::kInternalAecDump)
       options->aec_dump.Set(value);
+    else if (iter->key == MediaConstraintsInterface::kTypingNoiseDetection)
+      options->typing_detection.Set(value);
     else
       success = false;
   }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h
index a6b23c63a15..bc3872ce472 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h
@@ -80,12 +80,14 @@ class MediaConstraintsInterface {
   static const char kExperimentalAutoGainControl[];  // googAutoGainControl2
   static const char kNoiseSuppression[];  // googNoiseSuppression
   static const char kHighpassFilter[];  // googHighpassFilter
+  static const char kTypingNoiseDetection[];  // googTypingNoiseDetection
 
   // Google-specific constraint keys for a local video source
   static const char kNoiseReduction[];  // googNoiseReduction
   static const char kLeakyBucket[];  // googLeakyBucket
   // googTemporalLayeredScreencast
   static const char kTemporalLayeredScreencast[];
+  static const char kCpuOveruseDetection[];
 
   // Constraint keys for CreateOffer / CreateAnswer
   // Specified by the W3C PeerConnection spec
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler.cc
index d43c6d5a1d8..b09af7892fb 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler.cc
@@ -28,7 +28,7 @@
 #include "talk/app/webrtc/mediastreamhandler.h"
 
 #include "talk/app/webrtc/localaudiosource.h"
-#include "talk/app/webrtc/localvideosource.h"
+#include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videosourceinterface.h"
 
 namespace webrtc {
@@ -154,6 +154,8 @@ RemoteVideoTrackHandler::RemoteVideoTrackHandler(
       remote_video_track_(track),
       provider_(provider) {
   OnEnabledChanged();
+  provider_->SetVideoPlayout(ssrc, true,
+                             remote_video_track_->GetSource()->FrameInput());
 }
 
 RemoteVideoTrackHandler::~RemoteVideoTrackHandler() {
@@ -169,9 +171,6 @@ void RemoteVideoTrackHandler::OnStateChanged() {
 }
 
 void RemoteVideoTrackHandler::OnEnabledChanged() {
-  provider_->SetVideoPlayout(ssrc(),
-                             remote_video_track_->enabled(),
-                             remote_video_track_->FrameInput());
 }
 
 MediaStreamHandler::MediaStreamHandler(MediaStreamInterface* stream,
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler_unittest.cc
index a874bde0b41..475258e9b98 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamhandler_unittest.cc
@@ -30,9 +30,9 @@
 #include <string>
 
 #include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/localvideosource.h"
 #include "talk/app/webrtc/mediastream.h"
 #include "talk/app/webrtc/streamcollection.h"
+#include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/base/gunit.h"
 #include "talk/media/base/fakevideocapturer.h"
@@ -86,6 +86,7 @@ class FakeVideoSource : public Notifier<VideoSourceInterface> {
   virtual void RemoveSink(cricket::VideoRenderer* output) {}
   virtual SourceState state() const { return state_; }
   virtual const cricket::VideoOptions* options() const { return &options_; }
+  virtual cricket::VideoRenderer* FrameInput() { return NULL; }
 
  protected:
   FakeVideoSource() : state_(kLive) {}
@@ -149,8 +150,8 @@ class MediaStreamHandlerTest : public testing::Test {
   }
 
   void AddRemoteVideoTrack() {
-    EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, true,
-                                                 video_track_->FrameInput()));
+    EXPECT_CALL(video_provider_, SetVideoPlayout(
+        kVideoSsrc, true, video_track_->GetSource()->FrameInput()));
     handlers_.AddRemoteVideoTrack(stream_, stream_->GetVideoTracks()[0],
                                   kVideoSsrc);
   }
@@ -283,11 +284,8 @@ TEST_F(MediaStreamHandlerTest, LocalVideoTrackDisable) {
 TEST_F(MediaStreamHandlerTest, RemoteVideoTrackDisable) {
   AddRemoteVideoTrack();
 
-  EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, false, _));
   video_track_->set_enabled(false);
 
-  EXPECT_CALL(video_provider_, SetVideoPlayout(kVideoSsrc, true,
-                                               video_track_->FrameInput()));
   video_track_->set_enabled(true);
 
   RemoveRemoteVideoTrack();
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h
index 0d3e39d892c..b2c4468fb99 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h
@@ -107,6 +107,9 @@ class MediaStreamTrackInterface : public talk_base::RefCountInterface,
   virtual bool set_enabled(bool enable) = 0;
   // These methods should be called by implementation only.
   virtual bool set_state(TrackState new_state) = 0;
+
+ protected:
+  virtual ~MediaStreamTrackInterface() {}
 };
 
 // Interface for rendering VideoFrames from a VideoTrack
@@ -131,12 +134,6 @@ class VideoTrackInterface : public MediaStreamTrackInterface {
   // Deregister a renderer.
   virtual void RemoveRenderer(VideoRendererInterface* renderer) = 0;
 
-  // Gets a pointer to the frame input of this VideoTrack.
-  // The pointer is valid for the lifetime of this VideoTrack.
-  // VideoFrames rendered to the cricket::VideoRenderer will be rendered on all
-  // registered renderers.
-  virtual cricket::VideoRenderer* FrameInput() = 0;
-
   virtual VideoSourceInterface* GetSource() const = 0;
 
  protected:
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.cc
index c199f1e9c75..a23799f7369 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.cc
@@ -33,8 +33,11 @@
 #include "talk/app/webrtc/mediastreamproxy.h"
 #include "talk/app/webrtc/mediaconstraintsinterface.h"
 #include "talk/app/webrtc/mediastreamtrackproxy.h"
+#include "talk/app/webrtc/remotevideocapturer.h"
+#include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/base/bytebuffer.h"
+#include "talk/media/sctp/sctpdataengine.h"
 
 static const char kDefaultStreamLabel[] = "default";
 static const char kDefaultAudioTrackLabel[] = "defaulta0";
@@ -132,8 +135,10 @@ static bool EvaluateNeedForBundle(const cricket::MediaSessionOptions& options) {
 // Factory class for creating remote MediaStreams and MediaStreamTracks.
 class RemoteMediaStreamFactory {
  public:
-  explicit RemoteMediaStreamFactory(talk_base::Thread* signaling_thread)
-      : signaling_thread_(signaling_thread) {
+  explicit RemoteMediaStreamFactory(talk_base::Thread* signaling_thread,
+                                    cricket::ChannelManager* channel_manager)
+      : signaling_thread_(signaling_thread),
+        channel_manager_(channel_manager) {
   }
 
   talk_base::scoped_refptr<MediaStreamInterface> CreateMediaStream(
@@ -144,21 +149,24 @@ class RemoteMediaStreamFactory {
 
   AudioTrackInterface* AddAudioTrack(webrtc::MediaStreamInterface* stream,
                                      const std::string& track_id) {
-    return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(stream,
-                                                                      track_id);
+    return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(
+        stream, track_id, static_cast<AudioSourceInterface*>(NULL));
   }
 
   VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream,
                                      const std::string& track_id) {
-    return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>(stream,
-                                                                      track_id);
+    return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>(
+        stream, track_id, VideoSource::Create(channel_manager_,
+                                              new RemoteVideoCapturer(),
+                                              NULL).get());
   }
 
  private:
-  template <typename TI, typename T, typename TP>
-  TI* AddTrack(MediaStreamInterface* stream, const std::string& track_id) {
+  template <typename TI, typename T, typename TP, typename S>
+  TI* AddTrack(MediaStreamInterface* stream, const std::string& track_id,
+               S* source) {
     talk_base::scoped_refptr<TI> track(
-        TP::Create(signaling_thread_, T::Create(track_id, NULL)));
+        TP::Create(signaling_thread_, T::Create(track_id, source)));
     track->set_state(webrtc::MediaStreamTrackInterface::kLive);
     if (stream->AddTrack(track)) {
       return track;
@@ -167,17 +175,20 @@ class RemoteMediaStreamFactory {
   }
 
   talk_base::Thread* signaling_thread_;
+  cricket::ChannelManager* channel_manager_;
 };
 
 MediaStreamSignaling::MediaStreamSignaling(
     talk_base::Thread* signaling_thread,
-    MediaStreamSignalingObserver* stream_observer)
+    MediaStreamSignalingObserver* stream_observer,
+    cricket::ChannelManager* channel_manager)
     : signaling_thread_(signaling_thread),
       data_channel_factory_(NULL),
       stream_observer_(stream_observer),
       local_streams_(StreamCollection::Create()),
       remote_streams_(StreamCollection::Create()),
-      remote_stream_factory_(new RemoteMediaStreamFactory(signaling_thread)),
+      remote_stream_factory_(new RemoteMediaStreamFactory(signaling_thread,
+                                                          channel_manager)),
       last_allocated_sctp_id_(0) {
   options_.has_video = false;
   options_.has_audio = false;
@@ -225,6 +236,10 @@ bool MediaStreamSignaling::AllocateSctpId(int* id) {
   return true;
 }
 
+bool MediaStreamSignaling::HasDataChannels() const {
+  return !data_channels_.empty();
+}
+
 bool MediaStreamSignaling::AddDataChannel(DataChannel* data_channel) {
   ASSERT(data_channel != NULL);
   if (data_channels_.find(data_channel->label()) != data_channels_.end()) {
@@ -599,15 +614,19 @@ void MediaStreamSignaling::OnRemoteTrackRemoved(
   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
     talk_base::scoped_refptr<AudioTrackInterface> audio_track =
         stream->FindAudioTrack(track_id);
-    audio_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
-    stream->RemoveTrack(audio_track);
-    stream_observer_->OnRemoveRemoteAudioTrack(stream, audio_track);
+    if (audio_track) {
+      audio_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      stream->RemoveTrack(audio_track);
+      stream_observer_->OnRemoveRemoteAudioTrack(stream, audio_track);
+    }
   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
     talk_base::scoped_refptr<VideoTrackInterface> video_track =
         stream->FindVideoTrack(track_id);
-    video_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
-    stream->RemoveTrack(video_track);
-    stream_observer_->OnRemoveRemoteVideoTrack(stream, video_track);
+    if (video_track) {
+      video_track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      stream->RemoveTrack(video_track);
+      stream_observer_->OnRemoveRemoteVideoTrack(stream, video_track);
+    }
   } else {
     ASSERT(false && "Invalid media type");
   }
@@ -621,11 +640,19 @@ void MediaStreamSignaling::RejectRemoteTracks(cricket::MediaType media_type) {
     MediaStreamInterface* stream = remote_streams_->find(info.stream_label);
     if (media_type == cricket::MEDIA_TYPE_AUDIO) {
       AudioTrackInterface* track = stream->FindAudioTrack(info.track_id);
-      track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      // There's no guarantee the track is still available, e.g. the track may
+      // have been removed from the stream by javascript.
+      if (track) {
+        track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      }
     }
     if (media_type == cricket::MEDIA_TYPE_VIDEO) {
       VideoTrackInterface* track = stream->FindVideoTrack(info.track_id);
-      track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      // There's no guarantee the track is still available, e.g. the track may
+      // have been removed from the stream by javascript.
+      if (track) {
+        track->set_state(webrtc::MediaStreamTrackInterface::kEnded);
+      }
     }
   }
 }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.h b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.h
index f2437d8fcfb..f64bf978ffc 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling.h
@@ -159,7 +159,8 @@ class MediaStreamSignalingObserver {
 class MediaStreamSignaling {
  public:
   MediaStreamSignaling(talk_base::Thread* signaling_thread,
-                       MediaStreamSignalingObserver* stream_observer);
+                       MediaStreamSignalingObserver* stream_observer,
+                       cricket::ChannelManager* channel_manager);
   virtual ~MediaStreamSignaling();
 
   // Notify all referenced objects that MediaStreamSignaling will be teared
@@ -187,6 +188,8 @@ class MediaStreamSignaling {
   // be offered in a SessionDescription.
   void RemoveLocalStream(MediaStreamInterface* local_stream);
 
+  // Checks if any data channel has been added.
+  bool HasDataChannels() const;
   // Adds |data_channel| to the collection of DataChannels that will be
   // be offered in a SessionDescription.
   bool AddDataChannel(DataChannel* data_channel);
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling_unittest.cc
index 8b3dacb52f1..ea1336450da 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamsignaling_unittest.cc
@@ -37,8 +37,11 @@
 #include "talk/base/scoped_ptr.h"
 #include "talk/base/stringutils.h"
 #include "talk/base/thread.h"
+#include "talk/media/base/fakemediaengine.h"
+#include "talk/media/devices/fakedevicemanager.h"
 #include "talk/p2p/base/constants.h"
 #include "talk/p2p/base/sessiondescription.h"
+#include "talk/session/media/channelmanager.h"
 
 static const char kStreams[][8] = {"stream1", "stream2"};
 static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
@@ -285,13 +288,13 @@ class MockSignalingObserver : public webrtc::MediaStreamSignalingObserver {
   }
 
   virtual void OnAddRemoteVideoTrack(MediaStreamInterface* stream,
-                                    VideoTrackInterface* video_track,
-                                    uint32 ssrc) {
+                                     VideoTrackInterface* video_track,
+                                     uint32 ssrc) {
     AddTrack(&remote_video_tracks_, stream, video_track, ssrc);
   }
 
   virtual void OnRemoveRemoteAudioTrack(MediaStreamInterface* stream,
-                                       AudioTrackInterface* audio_track) {
+                                        AudioTrackInterface* audio_track) {
     RemoveTrack(&remote_audio_tracks_, stream, audio_track);
   }
 
@@ -392,8 +395,10 @@ class MockSignalingObserver : public webrtc::MediaStreamSignalingObserver {
 
 class MediaStreamSignalingForTest : public webrtc::MediaStreamSignaling {
  public:
-  explicit MediaStreamSignalingForTest(MockSignalingObserver* observer)
-      : webrtc::MediaStreamSignaling(talk_base::Thread::Current(), observer) {
+  MediaStreamSignalingForTest(MockSignalingObserver* observer,
+                              cricket::ChannelManager* channel_manager)
+      : webrtc::MediaStreamSignaling(talk_base::Thread::Current(), observer,
+                                     channel_manager) {
   };
 
   using webrtc::MediaStreamSignaling::GetOptionsForOffer;
@@ -406,7 +411,12 @@ class MediaStreamSignalingTest: public testing::Test {
  protected:
   virtual void SetUp() {
     observer_.reset(new MockSignalingObserver());
-    signaling_.reset(new MediaStreamSignalingForTest(observer_.get()));
+    channel_manager_.reset(
+        new cricket::ChannelManager(new cricket::FakeMediaEngine(),
+                                    new cricket::FakeDeviceManager(),
+                                    talk_base::Thread::Current()));
+    signaling_.reset(new MediaStreamSignalingForTest(observer_.get(),
+                                                     channel_manager_.get()));
   }
 
   // Create a collection of streams.
@@ -497,6 +507,9 @@ class MediaStreamSignalingTest: public testing::Test {
     ASSERT_TRUE(stream->AddTrack(video_track));
   }
 
+  // ChannelManager is used by VideoSource, so it should be released after all
+  // the video tracks. Put it as the first private variable should ensure that.
+  talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
   talk_base::scoped_refptr<StreamCollection> reference_collection_;
   talk_base::scoped_ptr<MockSignalingObserver> observer_;
   talk_base::scoped_ptr<MediaStreamSignalingForTest> signaling_;
@@ -688,6 +701,9 @@ TEST_F(MediaStreamSignalingTest, UpdateRemoteStreams) {
   observer_->VerifyRemoteAudioTrack(kStreams[0], kAudioTracks[0], 1);
   EXPECT_EQ(1u, observer_->NumberOfRemoteVideoTracks());
   observer_->VerifyRemoteVideoTrack(kStreams[0], kVideoTracks[0], 2);
+  ASSERT_EQ(1u, observer_->remote_streams()->count());
+  MediaStreamInterface* remote_stream =  observer_->remote_streams()->at(0);
+  EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != NULL);
 
   // Create a session description based on another SDP with another
   // MediaStream.
@@ -778,6 +794,33 @@ TEST_F(MediaStreamSignalingTest, RejectMediaContent) {
   EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_audio->state());
 }
 
+// This test that it won't crash if the remote track as been removed outside
+// of MediaStreamSignaling and then MediaStreamSignaling tries to reject
+// this track.
+TEST_F(MediaStreamSignalingTest, RemoveTrackThenRejectMediaContent) {
+  talk_base::scoped_ptr<SessionDescriptionInterface> desc(
+      webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
+                                       kSdpStringWithStream1, NULL));
+  EXPECT_TRUE(desc != NULL);
+  signaling_->OnRemoteDescriptionChanged(desc.get());
+
+  MediaStreamInterface* remote_stream =  observer_->remote_streams()->at(0);
+  remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
+  remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
+
+  cricket::ContentInfo* video_info =
+      desc->description()->GetContentByName("video");
+  video_info->rejected = true;
+  signaling_->OnLocalDescriptionChanged(desc.get());
+
+  cricket::ContentInfo* audio_info =
+      desc->description()->GetContentByName("audio");
+  audio_info->rejected = true;
+  signaling_->OnLocalDescriptionChanged(desc.get());
+
+  // No crash is a pass.
+}
+
 // This tests that a default MediaStream is created if a remote session
 // description doesn't contain any streams and no MSID support.
 // It also tests that the default stream is updated if a video m-line is added
@@ -812,6 +855,28 @@ TEST_F(MediaStreamSignalingTest, SdpWithoutMsidCreatesDefaultStream) {
   observer_->VerifyRemoteVideoTrack("default", "defaultv0", 0);
 }
 
+// This tests that it won't crash when MediaStreamSignaling tries to remove
+//  a remote track that as already been removed from the mediastream.
+TEST_F(MediaStreamSignalingTest, RemoveAlreadyGoneRemoteStream) {
+  talk_base::scoped_ptr<SessionDescriptionInterface> desc_audio_only(
+      webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
+                                       kSdpStringWithoutStreams,
+                                       NULL));
+  ASSERT_TRUE(desc_audio_only != NULL);
+  signaling_->OnRemoteDescriptionChanged(desc_audio_only.get());
+  MediaStreamInterface* remote_stream = observer_->remote_streams()->at(0);
+  remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
+  remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
+
+  talk_base::scoped_ptr<SessionDescriptionInterface> desc(
+      webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
+                                       kSdpStringWithoutStreams, NULL));
+  ASSERT_TRUE(desc != NULL);
+  signaling_->OnRemoteDescriptionChanged(desc.get());
+
+  // No crash is a pass.
+}
+
 // This tests that a default MediaStream is created if the remote session
 // description doesn't contain any streams and don't contain an indication if
 // MSID is supported.
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamtrackproxy.h b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamtrackproxy.h
index 1efc8c616ae..7c622e73332 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamtrackproxy.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/mediastreamtrackproxy.h
@@ -61,7 +61,6 @@ BEGIN_PROXY_MAP(VideoTrack)
 
   PROXY_METHOD1(void, AddRenderer, VideoRendererInterface*)
   PROXY_METHOD1(void, RemoveRenderer, VideoRendererInterface*)
-  PROXY_METHOD0(cricket::VideoRenderer*, FrameInput)
   PROXY_CONSTMETHOD0(VideoSourceInterface*, GetSource)
 
   PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCICEServer.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCICEServer.mm
index cb32673aa90..cc5a84a41fb 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCICEServer.mm
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/RTCICEServer.mm
@@ -33,22 +33,25 @@
 
 @implementation RTCICEServer
 
-- (id)initWithURI:(NSURL *)URI password:(NSString *)password {
-  if (!URI || !password) {
+- (id)initWithURI:(NSURL *)URI
+         username:(NSString *)username
+         password:(NSString *)password {
+  if (!URI || !username || !password) {
     NSAssert(NO, @"nil arguments not allowed");
     self = nil;
     return nil;
   }
   if ((self = [super init])) {
     _URI = URI;
+    _username = [username copy];
     _password = [password copy];
   }
   return self;
 }
 
 - (NSString *)description {
-  return [NSString stringWithFormat:@"Server: [%@]\nPassword: [%@]",
-          [self.URI absoluteString], self.password];
+  return [NSString stringWithFormat:@"RTCICEServer: [%@:%@:%@]",
+                   [self.URI absoluteString], self.username, self.password];
 }
 
 @end
@@ -58,6 +61,7 @@
 - (webrtc::PeerConnectionInterface::IceServer)iceServer {
   webrtc::PeerConnectionInterface::IceServer iceServer;
   iceServer.uri = [[self.URI absoluteString] UTF8String];
+  iceServer.username = [self.username UTF8String];
   iceServer.password = [self.password UTF8String];
   return iceServer;
 }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCICEServer.h b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCICEServer.h
index 717d042f782..63c14efaaf4 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCICEServer.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objc/public/RTCICEServer.h
@@ -30,14 +30,15 @@
 // RTCICEServer allows for the creation of ICEServer structs.
 @interface RTCICEServer : NSObject
 
-// The server URI.
+// The server URI, username, and password.
 @property(nonatomic, strong, readonly) NSURL* URI;
-
-// The server password.
+@property(nonatomic, copy, readonly) NSString* username;
 @property(nonatomic, copy, readonly) NSString* password;
 
 // Initializer for RTCICEServer taking uri and password.
-- (id)initWithURI:(NSString*)URI password:(NSString*)password;
+- (id)initWithURI:(NSString*)URI
+         username:(NSString*)username
+         password:(NSString*)password;
 
 #ifndef DOXYGEN_SHOULD_SKIP_THIS
 // Disallow init and don't add to documentation
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm b/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm
index 826409f561c..0ce8822fd1e 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm
@@ -99,6 +99,7 @@
   NSString *stunURL = @"stun:stun.l.google.com:19302";
   RTCICEServer *stunServer =
       [[RTCICEServer alloc] initWithURI:[NSURL URLWithString:stunURL]
+                               username:@""
                                password:@""];
   NSArray *iceServers = @[stunServer];
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc
index 2be76f439a2..9eaf915c395 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection.cc
@@ -303,7 +303,7 @@ bool PeerConnection::DoInitialize(
   port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
 
   mediastream_signaling_.reset(new MediaStreamSignaling(
-      factory_->signaling_thread(), this));
+      factory_->signaling_thread(), this, factory_->channel_manager()));
 
   session_.reset(new WebRtcSession(factory_->channel_manager(),
                                    factory_->signaling_thread(),
@@ -354,10 +354,10 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream,
 }
 
 void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
+  mediastream_signaling_->RemoveLocalStream(local_stream);
   if (IsClosed()) {
     return;
   }
-  mediastream_signaling_->RemoveLocalStream(local_stream);
   observer_->OnRenegotiationNeeded();
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
index 250e60f4185..522d528020c 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnection_unittest.cc
@@ -990,7 +990,9 @@ class P2PTestConductor : public testing::Test {
 typedef P2PTestConductor<JsepTestClient> JsepPeerConnectionP2PTestClient;
 
 // This test sets up a Jsep call between two parties and test Dtmf.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtmf) {
+// TODO(holmer): Disabled due to sometimes crashing on buildbots.
+// See issue webrtc/2378.
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
   VerifyDtmf();
@@ -1087,7 +1089,8 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestOfferDtlsButNotSdes) {
 
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive video.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) {
+// BUG=https://code.google.com/p/webrtc/issues/detail?id=2288
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestAnswerVideo) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->SetReceiveAudioVideo(false, true);
   LocalP2PTest();
@@ -1095,7 +1098,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) {
 
 // This test sets up a Jsep call between two parties, and the callee only
 // accept to receive audio.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) {
+TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestAnswerAudio) {
   ASSERT_TRUE(CreateTestClients());
   receiving_client()->SetReceiveAudioVideo(true, false);
   LocalP2PTest();
@@ -1369,7 +1372,10 @@ TEST_F(JsepPeerConnectionP2PTestClient, IceRestart) {
 
 // This test sets up a Jsep call between two parties with external
 // VideoDecoderFactory.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestWithVideoDecoderFactory) {
+// TODO(holmer): Disabled due to sometimes crashing on buildbots.
+// See issue webrtc/2378.
+TEST_F(JsepPeerConnectionP2PTestClient,
+       DISABLED_LocalP2PTestWithVideoDecoderFactory) {
   ASSERT_TRUE(CreateTestClients());
   EnableVideoDecoderFactory();
   LocalP2PTest();
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc
index 9c5c126b1a6..7d30fab8ec1 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectionfactory.cc
@@ -29,12 +29,12 @@
 
 #include "talk/app/webrtc/audiotrack.h"
 #include "talk/app/webrtc/localaudiosource.h"
-#include "talk/app/webrtc/localvideosource.h"
 #include "talk/app/webrtc/mediastreamproxy.h"
 #include "talk/app/webrtc/mediastreamtrackproxy.h"
 #include "talk/app/webrtc/peerconnection.h"
 #include "talk/app/webrtc/peerconnectionproxy.h"
 #include "talk/app/webrtc/portallocatorfactory.h"
+#include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videosourceproxy.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/media/devices/dummydevicemanager.h"
@@ -269,9 +269,8 @@ talk_base::scoped_refptr<VideoSourceInterface>
 PeerConnectionFactory::CreateVideoSource_s(
     cricket::VideoCapturer* capturer,
     const MediaConstraintsInterface* constraints) {
-  talk_base::scoped_refptr<LocalVideoSource> source(
-      LocalVideoSource::Create(channel_manager_.get(), capturer,
-                               constraints));
+  talk_base::scoped_refptr<VideoSource> source(
+      VideoSource::Create(channel_manager_.get(), capturer, constraints));
   return VideoSourceProxy::Create(signaling_thread_, source);
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface_unittest.cc
index d743684874d..7aa06ef4248 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -29,18 +29,19 @@
 
 #include "talk/app/webrtc/fakeportallocatorfactory.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
-#include "talk/app/webrtc/localvideosource.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/peerconnectioninterface.h"
 #include "talk/app/webrtc/test/fakeconstraints.h"
 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
 #include "talk/app/webrtc/test/testsdpstrings.h"
+#include "talk/app/webrtc/videosource.h"
 #include "talk/base/gunit.h"
 #include "talk/base/scoped_ptr.h"
 #include "talk/base/sslstreamadapter.h"
 #include "talk/base/stringutils.h"
 #include "talk/base/thread.h"
 #include "talk/media/base/fakevideocapturer.h"
+#include "talk/media/sctp/sctpdataengine.h"
 #include "talk/session/media/mediasession.h"
 
 static const char kStreamLabel1[] = "local_stream_1";
@@ -1193,7 +1194,7 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
   ASSERT_FALSE(local_stream->GetAudioTracks().empty());
   talk_base::scoped_refptr<webrtc::DtmfSenderInterface> dtmf_sender(
       pc_->CreateDtmfSender(local_stream->GetAudioTracks()[0]));
-  EXPECT_FALSE(dtmf_sender->CanInsertDtmf());
+  EXPECT_TRUE(NULL == dtmf_sender);  // local stream has been removed.
 
   EXPECT_TRUE(pc_->CreateDataChannel("test", NULL) == NULL);
 
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer.cc
new file mode 100644
index 00000000000..072c8d81cc5
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer.cc
@@ -0,0 +1,95 @@
+/*
+ * libjingle
+ * Copyright 2013, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/remotevideocapturer.h"
+
+#include "talk/base/logging.h"
+#include "talk/media/base/videoframe.h"
+
+namespace webrtc {
+
+RemoteVideoCapturer::RemoteVideoCapturer() {}
+
+RemoteVideoCapturer::~RemoteVideoCapturer() {}
+
+cricket::CaptureState RemoteVideoCapturer::Start(
+    const cricket::VideoFormat& capture_format) {
+  if (capture_state() == cricket::CS_RUNNING) {
+    LOG(LS_WARNING)
+        << "RemoteVideoCapturer::Start called when it's already started.";
+    return capture_state();
+  }
+
+  LOG(LS_INFO) << "RemoteVideoCapturer::Start";
+  SetCaptureFormat(&capture_format);
+  return cricket::CS_RUNNING;
+}
+
+void RemoteVideoCapturer::Stop() {
+  if (capture_state() == cricket::CS_STOPPED) {
+    LOG(LS_WARNING)
+        << "RemoteVideoCapturer::Stop called when it's already stopped.";
+    return;
+  }
+
+  LOG(LS_INFO) << "RemoteVideoCapturer::Stop";
+  SetCaptureFormat(NULL);
+  SetCaptureState(cricket::CS_STOPPED);
+}
+
+bool RemoteVideoCapturer::IsRunning() {
+  return capture_state() == cricket::CS_RUNNING;
+}
+
+bool RemoteVideoCapturer::GetPreferredFourccs(std::vector<uint32>* fourccs) {
+  if (!fourccs)
+    return false;
+  fourccs->push_back(cricket::FOURCC_I420);
+  return true;
+}
+
+bool RemoteVideoCapturer::GetBestCaptureFormat(
+    const cricket::VideoFormat& desired, cricket::VideoFormat* best_format) {
+  if (!best_format) {
+    return false;
+  }
+
+  // RemoteVideoCapturer does not support capability enumeration.
+  // Use the desired format as the best format.
+  best_format->width = desired.width;
+  best_format->height = desired.height;
+  best_format->fourcc = cricket::FOURCC_I420;
+  best_format->interval = desired.interval;
+  return true;
+}
+
+bool RemoteVideoCapturer::IsScreencast() const {
+  // TODO(ronghuawu): what about remote screencast stream.
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaStreamTrack.java b/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer.h
index 5cd2f4c11d5..1ea0692006d 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaStreamTrack.java
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer.h
@@ -25,62 +25,41 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-package org.webrtc;
-
-/** Java wrapper for a C++ MediaStreamTrackInterface. */
-public class MediaStreamTrack {
-  /** Tracks MediaStreamTrackInterface.TrackState */
-  public enum State {
-    INITIALIZING, LIVE, ENDED, FAILED
-  }
-
-  final long nativeTrack;
-
-  public MediaStreamTrack(long nativeTrack) {
-    this.nativeTrack = nativeTrack;
-  }
-
-  public String id() {
-    return nativeId(nativeTrack);
-  }
-
-  public String kind() {
-    return nativeKind(nativeTrack);
-  }
-
-  public boolean enabled() {
-    return nativeEnabled(nativeTrack);
-  }
-
-  public boolean setEnabled(boolean enable) {
-    return nativeSetEnabled(nativeTrack, enable);
-  }
-
-  public State state() {
-    return nativeState(nativeTrack);
-  }
-
-  public boolean setState(State newState) {
-    return nativeSetState(nativeTrack, newState.ordinal());
-  }
-
-  public void dispose() {
-    free(nativeTrack);
-  }
-
-  private static native String nativeId(long nativeTrack);
-
-  private static native String nativeKind(long nativeTrack);
-
-  private static native boolean nativeEnabled(long nativeTrack);
-
-  private static native boolean nativeSetEnabled(
-      long nativeTrack, boolean enabled);
-
-  private static native State nativeState(long nativeTrack);
-
-  private static native boolean nativeSetState(
-      long nativeTrack, int newState);
-
-  private static native void free(long nativeTrack);
-}
+#ifndef TALK_APP_WEBRTC_REMOTEVIDEOCAPTURER_H_
+#define TALK_APP_WEBRTC_REMOTEVIDEOCAPTURER_H_
+
+#include <vector>
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/media/base/videocapturer.h"
+#include "talk/media/base/videorenderer.h"
+
+namespace webrtc {
+
+// RemoteVideoCapturer implements a simple cricket::VideoCapturer which
+// gets decoded remote video frames from media channel.
+// It's used as the remote video source's VideoCapturer so that the remote video
+// can be used as a cricket::VideoCapturer and in that way a remote video stream
+// can implement the MediaStreamSourceInterface.
+class RemoteVideoCapturer : public cricket::VideoCapturer {
+ public:
+  RemoteVideoCapturer();
+  virtual ~RemoteVideoCapturer();
+
+  // cricket::VideoCapturer implementation.
+  virtual cricket::CaptureState Start(
+      const cricket::VideoFormat& capture_format) OVERRIDE;
+  virtual void Stop() OVERRIDE;
+  virtual bool IsRunning() OVERRIDE;
+  virtual bool GetPreferredFourccs(std::vector<uint32>* fourccs) OVERRIDE;
+  virtual bool GetBestCaptureFormat(const cricket::VideoFormat& desired,
+                                    cricket::VideoFormat* best_format) OVERRIDE;
+  virtual bool IsScreencast() const OVERRIDE;
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(RemoteVideoCapturer);
+};
+
+}  // namespace webrtc
+
+#endif  // TALK_APP_WEBRTC_REMOTEVIDEOCAPTURER_H_
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer_unittest.cc
new file mode 100644
index 00000000000..68135507cac
--- /dev/null
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/remotevideocapturer_unittest.cc
@@ -0,0 +1,132 @@
+/*
+ * libjingle
+ * Copyright 2013, Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ *  1. Redistributions of source code must retain the above copyright notice,
+ *     this list of conditions and the following disclaimer.
+ *  2. Redistributions in binary form must reproduce the above copyright notice,
+ *     this list of conditions and the following disclaimer in the documentation
+ *     and/or other materials provided with the distribution.
+ *  3. The name of the author may not be used to endorse or promote products
+ *     derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include <string>
+
+#include "talk/app/webrtc/remotevideocapturer.h"
+#include "talk/base/gunit.h"
+#include "talk/media/webrtc/webrtcvideoframe.h"
+
+using cricket::CaptureState;
+using cricket::VideoCapturer;
+using cricket::VideoFormat;
+using cricket::VideoFormatPod;
+using cricket::VideoFrame;
+
+static const int kMaxWaitMs = 1000;
+static const VideoFormatPod kTestFormat =
+    {640, 480, FPS_TO_INTERVAL(30), cricket::FOURCC_ANY};
+
+class RemoteVideoCapturerTest : public testing::Test,
+                                public sigslot::has_slots<> {
+ protected:
+  RemoteVideoCapturerTest()
+      : captured_frame_num_(0),
+        capture_state_(cricket::CS_STOPPED) {}
+
+  virtual void SetUp() {
+    capturer_.SignalStateChange.connect(
+        this, &RemoteVideoCapturerTest::OnStateChange);
+    capturer_.SignalVideoFrame.connect(
+        this, &RemoteVideoCapturerTest::OnVideoFrame);
+  }
+
+  ~RemoteVideoCapturerTest() {
+    capturer_.SignalStateChange.disconnect(this);
+    capturer_.SignalVideoFrame.disconnect(this);
+  }
+
+  int captured_frame_num() const {
+    return captured_frame_num_;
+  }
+
+  CaptureState capture_state() const {
+    return capture_state_;
+  }
+
+  webrtc::RemoteVideoCapturer capturer_;
+
+ private:
+  void OnStateChange(VideoCapturer* capturer,
+                     CaptureState capture_state) {
+    EXPECT_EQ(&capturer_, capturer);
+    capture_state_ = capture_state;
+  }
+
+  void OnVideoFrame(VideoCapturer* capturer, const VideoFrame* frame) {
+    EXPECT_EQ(&capturer_, capturer);
+    ++captured_frame_num_;
+  }
+
+  int captured_frame_num_;
+  CaptureState capture_state_;
+};
+
+TEST_F(RemoteVideoCapturerTest, StartStop) {
+  // Start
+  EXPECT_TRUE(
+      capturer_.StartCapturing(VideoFormat(kTestFormat)));
+  EXPECT_TRUE_WAIT((cricket::CS_RUNNING == capture_state()), kMaxWaitMs);
+  EXPECT_EQ(VideoFormat(kTestFormat),
+            *capturer_.GetCaptureFormat());
+  EXPECT_TRUE(capturer_.IsRunning());
+
+  // Stop
+  capturer_.Stop();
+  EXPECT_TRUE_WAIT((cricket::CS_STOPPED == capture_state()), kMaxWaitMs);
+  EXPECT_TRUE(NULL == capturer_.GetCaptureFormat());
+}
+
+TEST_F(RemoteVideoCapturerTest, GetPreferredFourccs) {
+  EXPECT_FALSE(capturer_.GetPreferredFourccs(NULL));
+
+  std::vector<uint32> fourccs;
+  EXPECT_TRUE(capturer_.GetPreferredFourccs(&fourccs));
+  EXPECT_EQ(1u, fourccs.size());
+  EXPECT_EQ(cricket::FOURCC_I420, fourccs.at(0));
+}
+
+TEST_F(RemoteVideoCapturerTest, GetBestCaptureFormat) {
+  VideoFormat desired = VideoFormat(kTestFormat);
+  EXPECT_FALSE(capturer_.GetBestCaptureFormat(desired, NULL));
+
+  VideoFormat expected_format = VideoFormat(kTestFormat);
+  expected_format.fourcc = cricket::FOURCC_I420;
+  VideoFormat best_format;
+  EXPECT_TRUE(capturer_.GetBestCaptureFormat(desired, &best_format));
+  EXPECT_EQ(expected_format, best_format);
+}
+
+TEST_F(RemoteVideoCapturerTest, InputFrame) {
+  EXPECT_EQ(0, captured_frame_num());
+
+  cricket::WebRtcVideoFrame test_frame;
+  capturer_.SignalVideoFrame(&capturer_, &test_frame);
+  EXPECT_EQ(1, captured_frame_num());
+  capturer_.SignalVideoFrame(&capturer_, &test_frame);
+  EXPECT_EQ(2, captured_frame_num());
+}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc
index b994f2faad7..06c4b44b082 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/statscollector.cc
@@ -94,6 +94,7 @@ const char StatsReport::kStatsValueNameRemoteAddress[] = "googRemoteAddress";
 const char StatsReport::kStatsValueNameRetransmitBitrate[] =
     "googRetransmitBitrate";
 const char StatsReport::kStatsValueNameRtt[] = "googRtt";
+const char StatsReport::kStatsValueNameSsrc[] = "ssrc";
 const char StatsReport::kStatsValueNameTargetEncBitrate[] =
     "googTargetEncBitrate";
 const char StatsReport::kStatsValueNameTransmitBitrate[] =
@@ -101,7 +102,8 @@ const char StatsReport::kStatsValueNameTransmitBitrate[] =
 const char StatsReport::kStatsValueNameTransportId[] = "transportId";
 const char StatsReport::kStatsValueNameTransportType[] = "googTransportType";
 const char StatsReport::kStatsValueNameTrackId[] = "googTrackId";
-const char StatsReport::kStatsValueNameSsrc[] = "ssrc";
+const char StatsReport::kStatsValueNameTypingNoiseState[] =
+    "googTypingNoiseState";
 const char StatsReport::kStatsValueNameWritable[] = "googWritable";
 
 const char StatsReport::kStatsReportTypeSession[] = "googLibjingleSession";
@@ -115,6 +117,7 @@ const char StatsReport::kStatsReportTypeCandidatePair[] = "googCandidatePair";
 
 const char StatsReport::kStatsReportVideoBweId[] = "bweforvideo";
 
+
 // Implementations of functions in statstypes.h
 void StatsReport::AddValue(const std::string& name, const std::string& value) {
   Value temp;
@@ -200,6 +203,8 @@ void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
   report->AddValue(StatsReport::kStatsValueNameEchoReturnLossEnhancement,
                    info.echo_return_loss_enhancement);
   report->AddValue(StatsReport::kStatsValueNameCodecName, info.codec_name);
+  report->AddBoolean(StatsReport::kStatsValueNameTypingNoiseState,
+                     info.typing_noise_detected);
 }
 
 void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
@@ -398,8 +403,8 @@ StatsReport* StatsCollector::PrepareReport(uint32 ssrc,
   std::string track_id;
   if (it == reports_.end()) {
     if (!session()->GetTrackIdBySsrc(ssrc, &track_id)) {
-      LOG(LS_ERROR) << "The SSRC " << ssrc
-                    << " is not associated with a track";
+      LOG(LS_WARNING) << "The SSRC " << ssrc
+                      << " is not associated with a track";
       return NULL;
     }
   } else {
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h b/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h
index 62f8781612a..30a8b84165c 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/statstypes.h
@@ -151,6 +151,7 @@ class StatsReport {
   static const char kStatsValueNameChannelId[];
   static const char kStatsValueNameTrackId[];
   static const char kStatsValueNameSsrc[];
+  static const char kStatsValueNameTypingNoiseState[];
 };
 
 typedef std::vector<StatsReport> StatsReports;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/localvideosource.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/videosource.cc
index 2d43885fccf..f0182f4fb18 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/localvideosource.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videosource.cc
@@ -25,7 +25,7 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#include "talk/app/webrtc/localvideosource.h"
+#include "talk/app/webrtc/videosource.h"
 
 #include <vector>
 
@@ -54,6 +54,9 @@ const char MediaConstraintsInterface::kNoiseReduction[] = "googNoiseReduction";
 const char MediaConstraintsInterface::kLeakyBucket[] = "googLeakyBucket";
 const char MediaConstraintsInterface::kTemporalLayeredScreencast[] =
     "googTemporalLayeredScreencast";
+// TODO(ronghuawu): Remove once cpu overuse detection is stable.
+const char MediaConstraintsInterface::kCpuOveruseDetection[] =
+    "googCpuOveruseDetection";
 
 }  // namespace webrtc
 
@@ -69,7 +72,7 @@ enum {
 
 // Default resolution. If no constraint is specified, this is the resolution we
 // will use.
-static const cricket::VideoFormatPod kDefaultResolution =
+static const cricket::VideoFormatPod kDefaultFormat =
     {640, 480, FPS_TO_INTERVAL(30), cricket::FOURCC_ANY};
 
 // List of formats used if the camera doesn't support capability enumeration.
@@ -202,7 +205,9 @@ bool NewFormatWithConstraints(
   } else if (constraint.key == MediaConstraintsInterface::kNoiseReduction ||
              constraint.key == MediaConstraintsInterface::kLeakyBucket ||
              constraint.key ==
-                 MediaConstraintsInterface::kTemporalLayeredScreencast) {
+                 MediaConstraintsInterface::kTemporalLayeredScreencast ||
+             constraint.key ==
+                 MediaConstraintsInterface::kCpuOveruseDetection) {
     // These are actually options, not constraints, so they can be satisfied
     // regardless of the format.
     return true;
@@ -271,7 +276,7 @@ const cricket::VideoFormat& GetBestCaptureFormat(
     const std::vector<cricket::VideoFormat>& formats) {
   ASSERT(formats.size() > 0);
 
-  int default_area = kDefaultResolution.width * kDefaultResolution.height;
+  int default_area = kDefaultFormat.width * kDefaultFormat.height;
 
   std::vector<cricket::VideoFormat>::const_iterator it = formats.begin();
   std::vector<cricket::VideoFormat>::const_iterator best_it = formats.begin();
@@ -316,42 +321,79 @@ bool ExtractVideoOptions(const MediaConstraintsInterface* all_constraints,
   all_valid &= ExtractOption(all_constraints,
       MediaConstraintsInterface::kTemporalLayeredScreencast,
       &(options->video_temporal_layer_screencast));
+  all_valid &= ExtractOption(all_constraints,
+      MediaConstraintsInterface::kCpuOveruseDetection,
+      &(options->cpu_overuse_detection));
 
   return all_valid;
 }
 
+class FrameInputWrapper : public cricket::VideoRenderer {
+ public:
+  explicit FrameInputWrapper(cricket::VideoCapturer* capturer)
+      : capturer_(capturer) {
+    ASSERT(capturer_ != NULL);
+  }
+
+  virtual ~FrameInputWrapper() {}
+
+  // VideoRenderer implementation.
+  virtual bool SetSize(int width, int height, int reserved) OVERRIDE {
+    return true;
+  }
+
+  virtual bool RenderFrame(const cricket::VideoFrame* frame) OVERRIDE {
+    if (!capturer_->IsRunning()) {
+      return true;
+    }
+
+    // This signal will be made on media engine render thread. The clients
+    // of this signal should have no assumptions on what thread this signal
+    // come from.
+    capturer_->SignalVideoFrame(capturer_, frame);
+    return true;
+  }
+
+ private:
+  cricket::VideoCapturer* capturer_;
+  int width_;
+  int height_;
+
+  DISALLOW_COPY_AND_ASSIGN(FrameInputWrapper);
+};
+
 }  // anonymous namespace
 
 namespace webrtc {
 
-talk_base::scoped_refptr<LocalVideoSource> LocalVideoSource::Create(
+talk_base::scoped_refptr<VideoSource> VideoSource::Create(
     cricket::ChannelManager* channel_manager,
     cricket::VideoCapturer* capturer,
     const webrtc::MediaConstraintsInterface* constraints) {
   ASSERT(channel_manager != NULL);
   ASSERT(capturer != NULL);
-  talk_base::scoped_refptr<LocalVideoSource> source(
-      new talk_base::RefCountedObject<LocalVideoSource>(channel_manager,
-                                                        capturer));
+  talk_base::scoped_refptr<VideoSource> source(
+      new talk_base::RefCountedObject<VideoSource>(channel_manager,
+                                                   capturer));
   source->Initialize(constraints);
   return source;
 }
 
-LocalVideoSource::LocalVideoSource(cricket::ChannelManager* channel_manager,
-                                   cricket::VideoCapturer* capturer)
+VideoSource::VideoSource(cricket::ChannelManager* channel_manager,
+                         cricket::VideoCapturer* capturer)
     : channel_manager_(channel_manager),
       video_capturer_(capturer),
       state_(kInitializing) {
   channel_manager_->SignalVideoCaptureStateChange.connect(
-      this, &LocalVideoSource::OnStateChange);
+      this, &VideoSource::OnStateChange);
 }
 
-LocalVideoSource::~LocalVideoSource() {
+VideoSource::~VideoSource() {
   channel_manager_->StopVideoCapture(video_capturer_.get(), format_);
   channel_manager_->SignalVideoCaptureStateChange.disconnect(this);
 }
 
-void LocalVideoSource::Initialize(
+void VideoSource::Initialize(
     const webrtc::MediaConstraintsInterface* constraints) {
 
   std::vector<cricket::VideoFormat> formats;
@@ -363,7 +405,7 @@ void LocalVideoSource::Initialize(
     // format from the constraints if any.
     // Note that this only affects tab capturing, not desktop capturing,
     // since desktop capturer does not respect the VideoFormat passed in.
-    formats.push_back(cricket::VideoFormat(kDefaultResolution));
+    formats.push_back(cricket::VideoFormat(kDefaultFormat));
   } else {
     // The VideoCapturer implementation doesn't support capability enumeration.
     // We need to guess what the camera support.
@@ -414,25 +456,34 @@ void LocalVideoSource::Initialize(
   // Initialize hasn't succeeded until a successful state change has occurred.
 }
 
-void LocalVideoSource::AddSink(cricket::VideoRenderer* output) {
+cricket::VideoRenderer* VideoSource::FrameInput() {
+  // Defer creation of frame_input_ until it's needed, e.g. the local video
+  // sources will never need it.
+  if (!frame_input_) {
+    frame_input_.reset(new FrameInputWrapper(video_capturer_.get()));
+  }
+  return frame_input_.get();
+}
+
+void VideoSource::AddSink(cricket::VideoRenderer* output) {
   channel_manager_->AddVideoRenderer(video_capturer_.get(), output);
 }
 
-void LocalVideoSource::RemoveSink(cricket::VideoRenderer* output) {
+void VideoSource::RemoveSink(cricket::VideoRenderer* output) {
   channel_manager_->RemoveVideoRenderer(video_capturer_.get(), output);
 }
 
 // OnStateChange listens to the ChannelManager::SignalVideoCaptureStateChange.
 // This signal is triggered for all video capturers. Not only the one we are
 // interested in.
-void LocalVideoSource::OnStateChange(cricket::VideoCapturer* capturer,
+void VideoSource::OnStateChange(cricket::VideoCapturer* capturer,
                                      cricket::CaptureState capture_state) {
   if (capturer == video_capturer_.get()) {
     SetState(GetReadyState(capture_state));
   }
 }
 
-void LocalVideoSource::SetState(SourceState new_state) {
+void VideoSource::SetState(SourceState new_state) {
   if (VERIFY(state_ != new_state)) {
     state_ = new_state;
     FireOnChanged();
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/localvideosource.h b/chromium/third_party/libjingle/source/talk/app/webrtc/videosource.h
index 0a3bac091c7..f58b479c284 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/localvideosource.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videosource.h
@@ -25,18 +25,19 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#ifndef TALK_APP_WEBRTC_LOCALVIDEOSOURCE_H_
-#define TALK_APP_WEBRTC_LOCALVIDEOSOURCE_H_
+#ifndef TALK_APP_WEBRTC_VIDEOSOURCE_H_
+#define TALK_APP_WEBRTC_VIDEOSOURCE_H_
 
 #include "talk/app/webrtc/mediastreaminterface.h"
 #include "talk/app/webrtc/notifier.h"
 #include "talk/app/webrtc/videosourceinterface.h"
+#include "talk/app/webrtc/videotrackrenderers.h"
 #include "talk/base/scoped_ptr.h"
 #include "talk/base/sigslot.h"
 #include "talk/media/base/videocapturer.h"
 #include "talk/media/base/videocommon.h"
 
-// LocalVideoSource implements VideoSourceInterface. It owns a
+// VideoSource implements VideoSourceInterface. It owns a
 // cricket::VideoCapturer and make sure the camera is started at a resolution
 // that honors the constraints.
 // The state is set depending on the result of starting the capturer.
@@ -53,20 +54,21 @@ namespace webrtc {
 
 class MediaConstraintsInterface;
 
-class LocalVideoSource : public Notifier<VideoSourceInterface>,
-                         public sigslot::has_slots<> {
+class VideoSource : public Notifier<VideoSourceInterface>,
+                    public sigslot::has_slots<> {
  public:
-  // Creates an instance of LocalVideoSource.
-  // LocalVideoSource take ownership of |capturer|.
+  // Creates an instance of VideoSource.
+  // VideoSource take ownership of |capturer|.
   // |constraints| can be NULL and in that case the camera is opened using a
   // default resolution.
-  static talk_base::scoped_refptr<LocalVideoSource> Create(
+  static talk_base::scoped_refptr<VideoSource> Create(
       cricket::ChannelManager* channel_manager,
       cricket::VideoCapturer* capturer,
       const webrtc::MediaConstraintsInterface* constraints);
 
   virtual SourceState state() const { return state_; }
   virtual const cricket::VideoOptions* options() const { return &options_; }
+  virtual cricket::VideoRenderer* FrameInput();
 
   virtual cricket::VideoCapturer* GetVideoCapturer() {
     return video_capturer_.get();
@@ -77,18 +79,19 @@ class LocalVideoSource : public Notifier<VideoSourceInterface>,
   virtual void RemoveSink(cricket::VideoRenderer* output);
 
  protected:
-  LocalVideoSource(cricket::ChannelManager* channel_manager,
-                   cricket::VideoCapturer* capturer);
-  ~LocalVideoSource();
+  VideoSource(cricket::ChannelManager* channel_manager,
+              cricket::VideoCapturer* capturer);
+  virtual ~VideoSource();
+  void Initialize(const webrtc::MediaConstraintsInterface* constraints);
 
  private:
-  void Initialize(const webrtc::MediaConstraintsInterface* constraints);
   void OnStateChange(cricket::VideoCapturer* capturer,
                      cricket::CaptureState capture_state);
   void SetState(SourceState new_state);
 
   cricket::ChannelManager* channel_manager_;
   talk_base::scoped_ptr<cricket::VideoCapturer> video_capturer_;
+  talk_base::scoped_ptr<cricket::VideoRenderer> frame_input_;
 
   cricket::VideoFormat format_;
   cricket::VideoOptions options_;
@@ -97,4 +100,4 @@ class LocalVideoSource : public Notifier<VideoSourceInterface>,
 
 }  // namespace webrtc
 
-#endif  // TALK_APP_WEBRTC_LOCALVIDEOSOURCE_H_
+#endif  // TALK_APP_WEBRTC_VIDEOSOURCE_H_
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/localvideosource_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/videosource_unittest.cc
index 24a858886a6..69e9b3f0f74 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/localvideosource_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videosource_unittest.cc
@@ -25,20 +25,21 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-#include "talk/app/webrtc/localvideosource.h"
-
 #include <string>
 #include <vector>
 
 #include "talk/app/webrtc/test/fakeconstraints.h"
+#include "talk/app/webrtc/remotevideocapturer.h"
+#include "talk/app/webrtc/videosource.h"
 #include "talk/base/gunit.h"
 #include "talk/media/base/fakemediaengine.h"
 #include "talk/media/base/fakevideorenderer.h"
 #include "talk/media/devices/fakedevicemanager.h"
+#include "talk/media/webrtc/webrtcvideoframe.h"
 #include "talk/session/media/channelmanager.h"
 
 using webrtc::FakeConstraints;
-using webrtc::LocalVideoSource;
+using webrtc::VideoSource;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaSourceInterface;
 using webrtc::ObserverInterface;
@@ -123,52 +124,54 @@ class StateObserver : public ObserverInterface {
   talk_base::scoped_refptr<VideoSourceInterface> source_;
 };
 
-class LocalVideoSourceTest : public testing::Test {
+class VideoSourceTest : public testing::Test {
  protected:
-  LocalVideoSourceTest()
-      : channel_manager_(new cricket::ChannelManager(
+  VideoSourceTest()
+      : capturer_cleanup_(new TestVideoCapturer()),
+        capturer_(capturer_cleanup_.get()),
+        channel_manager_(new cricket::ChannelManager(
           new cricket::FakeMediaEngine(),
           new cricket::FakeDeviceManager(), talk_base::Thread::Current())) {
   }
 
   void SetUp() {
     ASSERT_TRUE(channel_manager_->Init());
-    capturer_ = new TestVideoCapturer();
   }
 
-  void CreateLocalVideoSource() {
-    CreateLocalVideoSource(NULL);
+  void CreateVideoSource() {
+    CreateVideoSource(NULL);
   }
 
-  void CreateLocalVideoSource(
+  void CreateVideoSource(
       const webrtc::MediaConstraintsInterface* constraints) {
     // VideoSource take ownership of |capturer_|
-    local_source_ = LocalVideoSource::Create(channel_manager_.get(),
-                                             capturer_,
-                                             constraints);
+    source_ = VideoSource::Create(channel_manager_.get(),
+                                  capturer_cleanup_.release(),
+                                  constraints);
 
-    ASSERT_TRUE(local_source_.get() != NULL);
-    EXPECT_EQ(capturer_, local_source_->GetVideoCapturer());
+    ASSERT_TRUE(source_.get() != NULL);
+    EXPECT_EQ(capturer_, source_->GetVideoCapturer());
 
-    state_observer_.reset(new StateObserver(local_source_));
-    local_source_->RegisterObserver(state_observer_.get());
-    local_source_->AddSink(&renderer_);
+    state_observer_.reset(new StateObserver(source_));
+    source_->RegisterObserver(state_observer_.get());
+    source_->AddSink(&renderer_);
   }
 
-  TestVideoCapturer* capturer_;  // Raw pointer. Owned by local_source_.
+  talk_base::scoped_ptr<TestVideoCapturer> capturer_cleanup_;
+  TestVideoCapturer* capturer_;
   cricket::FakeVideoRenderer renderer_;
   talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
   talk_base::scoped_ptr<StateObserver> state_observer_;
-  talk_base::scoped_refptr<LocalVideoSource> local_source_;
+  talk_base::scoped_refptr<VideoSource> source_;
 };
 
 
-// Test that a LocalVideoSource transition to kLive state when the capture
+// Test that a VideoSource transition to kLive state when the capture
 // device have started and kEnded if it is stopped.
 // It also test that an output can receive video frames.
-TEST_F(LocalVideoSourceTest, StartStop) {
+TEST_F(VideoSourceTest, StartStop) {
   // Initialize without constraints.
-  CreateLocalVideoSource();
+  CreateVideoSource();
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
 
@@ -180,10 +183,38 @@ TEST_F(LocalVideoSourceTest, StartStop) {
                  kMaxWaitMs);
 }
 
-// Test that a LocalVideoSource transition to kEnded if the capture device
+// Test start stop with a remote VideoSource - the video source that has a
+// RemoteVideoCapturer and takes video frames from FrameInput.
+TEST_F(VideoSourceTest, StartStopRemote) {
+  source_ = VideoSource::Create(channel_manager_.get(),
+                                new webrtc::RemoteVideoCapturer(),
+                                NULL);
+
+  ASSERT_TRUE(source_.get() != NULL);
+  EXPECT_TRUE(NULL != source_->GetVideoCapturer());
+
+  state_observer_.reset(new StateObserver(source_));
+  source_->RegisterObserver(state_observer_.get());
+  source_->AddSink(&renderer_);
+
+  EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
+                 kMaxWaitMs);
+
+  cricket::VideoRenderer* frameinput = source_->FrameInput();
+  cricket::WebRtcVideoFrame test_frame;
+  frameinput->SetSize(1280, 720, 0);
+  frameinput->RenderFrame(&test_frame);
+  EXPECT_EQ(1, renderer_.num_rendered_frames());
+
+  source_->GetVideoCapturer()->Stop();
+  EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
+                 kMaxWaitMs);
+}
+
+// Test that a VideoSource transition to kEnded if the capture device
 // fails.
-TEST_F(LocalVideoSourceTest, CameraFailed) {
-  CreateLocalVideoSource();
+TEST_F(VideoSourceTest, CameraFailed) {
+  CreateVideoSource();
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
 
@@ -194,13 +225,13 @@ TEST_F(LocalVideoSourceTest, CameraFailed) {
 
 // Test that the capture output is CIF if we set max constraints to CIF.
 // and the capture device support CIF.
-TEST_F(LocalVideoSourceTest, MandatoryConstraintCif5Fps) {
+TEST_F(VideoSourceTest, MandatoryConstraintCif5Fps) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kMaxWidth, 352);
   constraints.AddMandatory(MediaConstraintsInterface::kMaxHeight, 288);
   constraints.AddMandatory(MediaConstraintsInterface::kMaxFrameRate, 5);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -212,7 +243,7 @@ TEST_F(LocalVideoSourceTest, MandatoryConstraintCif5Fps) {
 
 // Test that the capture output is 720P if the camera support it and the
 // optional constraint is set to 720P.
-TEST_F(LocalVideoSourceTest, MandatoryMinVgaOptional720P) {
+TEST_F(VideoSourceTest, MandatoryMinVgaOptional720P) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kMinWidth, 640);
   constraints.AddMandatory(MediaConstraintsInterface::kMinHeight, 480);
@@ -220,7 +251,7 @@ TEST_F(LocalVideoSourceTest, MandatoryMinVgaOptional720P) {
   constraints.AddOptional(MediaConstraintsInterface::kMinAspectRatio,
                           1280.0 / 720);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -233,7 +264,7 @@ TEST_F(LocalVideoSourceTest, MandatoryMinVgaOptional720P) {
 // Test that the capture output have aspect ratio 4:3 if a mandatory constraint
 // require it even if an optional constraint request a higher resolution
 // that don't have this aspect ratio.
-TEST_F(LocalVideoSourceTest, MandatoryAspectRatio4To3) {
+TEST_F(VideoSourceTest, MandatoryAspectRatio4To3) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kMinWidth, 640);
   constraints.AddMandatory(MediaConstraintsInterface::kMinHeight, 480);
@@ -241,7 +272,7 @@ TEST_F(LocalVideoSourceTest, MandatoryAspectRatio4To3) {
                            640.0 / 480);
   constraints.AddOptional(MediaConstraintsInterface::kMinWidth, 1280);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -254,20 +285,20 @@ TEST_F(LocalVideoSourceTest, MandatoryAspectRatio4To3) {
 
 // Test that the source state transition to kEnded if the mandatory aspect ratio
 // is set higher than supported.
-TEST_F(LocalVideoSourceTest, MandatoryAspectRatioTooHigh) {
+TEST_F(VideoSourceTest, MandatoryAspectRatioTooHigh) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kMinAspectRatio, 2);
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
                  kMaxWaitMs);
 }
 
 // Test that the source ignores an optional aspect ratio that is higher than
 // supported.
-TEST_F(LocalVideoSourceTest, OptionalAspectRatioTooHigh) {
+TEST_F(VideoSourceTest, OptionalAspectRatioTooHigh) {
   FakeConstraints constraints;
   constraints.AddOptional(MediaConstraintsInterface::kMinAspectRatio, 2);
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -278,10 +309,10 @@ TEST_F(LocalVideoSourceTest, OptionalAspectRatioTooHigh) {
 
 // Test that the source starts video with the default resolution if the
 // camera doesn't support capability enumeration and there are no constraints.
-TEST_F(LocalVideoSourceTest, NoCameraCapability) {
+TEST_F(VideoSourceTest, NoCameraCapability) {
   capturer_->TestWithoutCameraFormats();
 
-  CreateLocalVideoSource();
+  CreateVideoSource();
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -294,7 +325,7 @@ TEST_F(LocalVideoSourceTest, NoCameraCapability) {
 // Test that the source can start the video and get the requested aspect ratio
 // if the camera doesn't support capability enumeration and the aspect ratio is
 // set.
-TEST_F(LocalVideoSourceTest, NoCameraCapability16To9Ratio) {
+TEST_F(VideoSourceTest, NoCameraCapability16To9Ratio) {
   capturer_->TestWithoutCameraFormats();
 
   FakeConstraints constraints;
@@ -303,7 +334,7 @@ TEST_F(LocalVideoSourceTest, NoCameraCapability16To9Ratio) {
   constraints.AddMandatory(MediaConstraintsInterface::kMinAspectRatio,
                            requested_aspect_ratio);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -313,114 +344,119 @@ TEST_F(LocalVideoSourceTest, NoCameraCapability16To9Ratio) {
 
 // Test that the source state transitions to kEnded if an unknown mandatory
 // constraint is found.
-TEST_F(LocalVideoSourceTest, InvalidMandatoryConstraint) {
+TEST_F(VideoSourceTest, InvalidMandatoryConstraint) {
   FakeConstraints constraints;
   constraints.AddMandatory("weird key", 640);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
                  kMaxWaitMs);
 }
 
 // Test that the source ignores an unknown optional constraint.
-TEST_F(LocalVideoSourceTest, InvalidOptionalConstraint) {
+TEST_F(VideoSourceTest, InvalidOptionalConstraint) {
   FakeConstraints constraints;
   constraints.AddOptional("weird key", 640);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
 }
 
-TEST_F(LocalVideoSourceTest, SetValidOptionValues) {
+TEST_F(VideoSourceTest, SetValidOptionValues) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kNoiseReduction, "false");
   constraints.AddMandatory(
       MediaConstraintsInterface::kTemporalLayeredScreencast, "false");
   constraints.AddOptional(
       MediaConstraintsInterface::kLeakyBucket, "true");
+  constraints.AddOptional(
+      MediaConstraintsInterface::kCpuOveruseDetection, "true");
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
 
   bool value = true;
-  EXPECT_TRUE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
   EXPECT_FALSE(value);
-  EXPECT_TRUE(local_source_->options()->
+  EXPECT_TRUE(source_->options()->
       video_temporal_layer_screencast.Get(&value));
   EXPECT_FALSE(value);
-  EXPECT_TRUE(local_source_->options()->video_leaky_bucket.Get(&value));
+  EXPECT_TRUE(source_->options()->video_leaky_bucket.Get(&value));
   EXPECT_TRUE(value);
+  EXPECT_TRUE(source_->options()->
+      cpu_overuse_detection.GetWithDefaultIfUnset(false));
 }
 
-TEST_F(LocalVideoSourceTest, OptionNotSet) {
+TEST_F(VideoSourceTest, OptionNotSet) {
   FakeConstraints constraints;
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   bool value;
-  EXPECT_FALSE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_FALSE(source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_FALSE(source_->options()->cpu_overuse_detection.Get(&value));
 }
 
-TEST_F(LocalVideoSourceTest, MandatoryOptionOverridesOptional) {
+TEST_F(VideoSourceTest, MandatoryOptionOverridesOptional) {
   FakeConstraints constraints;
   constraints.AddMandatory(
       MediaConstraintsInterface::kNoiseReduction, true);
   constraints.AddOptional(
       MediaConstraintsInterface::kNoiseReduction, false);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
 
   bool value = false;
-  EXPECT_TRUE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
   EXPECT_TRUE(value);
-  EXPECT_FALSE(local_source_->options()->video_leaky_bucket.Get(&value));
+  EXPECT_FALSE(source_->options()->video_leaky_bucket.Get(&value));
 }
 
-TEST_F(LocalVideoSourceTest, InvalidOptionKeyOptional) {
+TEST_F(VideoSourceTest, InvalidOptionKeyOptional) {
   FakeConstraints constraints;
   constraints.AddOptional(
       MediaConstraintsInterface::kNoiseReduction, false);
   constraints.AddOptional("invalidKey", false);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
 
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
       kMaxWaitMs);
   bool value = true;
-  EXPECT_TRUE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
   EXPECT_FALSE(value);
 }
 
-TEST_F(LocalVideoSourceTest, InvalidOptionKeyMandatory) {
+TEST_F(VideoSourceTest, InvalidOptionKeyMandatory) {
   FakeConstraints constraints;
   constraints.AddMandatory(
       MediaConstraintsInterface::kNoiseReduction, false);
   constraints.AddMandatory("invalidKey", false);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
 
   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
       kMaxWaitMs);
   bool value;
-  EXPECT_FALSE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_FALSE(source_->options()->video_noise_reduction.Get(&value));
 }
 
-TEST_F(LocalVideoSourceTest, InvalidOptionValueOptional) {
+TEST_F(VideoSourceTest, InvalidOptionValueOptional) {
   FakeConstraints constraints;
   constraints.AddOptional(
       MediaConstraintsInterface::kNoiseReduction, "true");
   constraints.AddOptional(
       MediaConstraintsInterface::kLeakyBucket, "not boolean");
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
 
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
       kMaxWaitMs);
   bool value = false;
-  EXPECT_TRUE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
   EXPECT_TRUE(value);
-  EXPECT_FALSE(local_source_->options()->video_leaky_bucket.Get(&value));
+  EXPECT_FALSE(source_->options()->video_leaky_bucket.Get(&value));
 }
 
-TEST_F(LocalVideoSourceTest, InvalidOptionValueMandatory) {
+TEST_F(VideoSourceTest, InvalidOptionValueMandatory) {
   FakeConstraints constraints;
   // Optional constraints should be ignored if the mandatory constraints fail.
   constraints.AddOptional(
@@ -429,15 +465,15 @@ TEST_F(LocalVideoSourceTest, InvalidOptionValueMandatory) {
   constraints.AddMandatory(
       MediaConstraintsInterface::kLeakyBucket, "True");
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
 
   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
       kMaxWaitMs);
   bool value;
-  EXPECT_FALSE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_FALSE(source_->options()->video_noise_reduction.Get(&value));
 }
 
-TEST_F(LocalVideoSourceTest, MixedOptionsAndConstraints) {
+TEST_F(VideoSourceTest, MixedOptionsAndConstraints) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kMaxWidth, 352);
   constraints.AddMandatory(MediaConstraintsInterface::kMaxHeight, 288);
@@ -448,7 +484,7 @@ TEST_F(LocalVideoSourceTest, MixedOptionsAndConstraints) {
   constraints.AddOptional(
       MediaConstraintsInterface::kNoiseReduction, true);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -458,18 +494,18 @@ TEST_F(LocalVideoSourceTest, MixedOptionsAndConstraints) {
   EXPECT_EQ(5, format->framerate());
 
   bool value = true;
-  EXPECT_TRUE(local_source_->options()->video_noise_reduction.Get(&value));
+  EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
   EXPECT_FALSE(value);
-  EXPECT_FALSE(local_source_->options()->video_leaky_bucket.Get(&value));
+  EXPECT_FALSE(source_->options()->video_leaky_bucket.Get(&value));
 }
 
 // Tests that the source starts video with the default resolution for
 // screencast if no constraint is set.
-TEST_F(LocalVideoSourceTest, ScreencastResolutionNoConstraint) {
+TEST_F(VideoSourceTest, ScreencastResolutionNoConstraint) {
   capturer_->TestWithoutCameraFormats();
   capturer_->SetScreencast(true);
 
-  CreateLocalVideoSource();
+  CreateVideoSource();
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -481,7 +517,7 @@ TEST_F(LocalVideoSourceTest, ScreencastResolutionNoConstraint) {
 
 // Tests that the source starts video with the max width and height set by
 // constraints for screencast.
-TEST_F(LocalVideoSourceTest, ScreencastResolutionWithConstraint) {
+TEST_F(VideoSourceTest, ScreencastResolutionWithConstraint) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kMaxWidth, 480);
   constraints.AddMandatory(MediaConstraintsInterface::kMaxHeight, 270);
@@ -489,7 +525,7 @@ TEST_F(LocalVideoSourceTest, ScreencastResolutionWithConstraint) {
   capturer_->TestWithoutCameraFormats();
   capturer_->SetScreencast(true);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
@@ -499,21 +535,21 @@ TEST_F(LocalVideoSourceTest, ScreencastResolutionWithConstraint) {
   EXPECT_EQ(30, format->framerate());
 }
 
-TEST_F(LocalVideoSourceTest, MandatorySubOneFpsConstraints) {
+TEST_F(VideoSourceTest, MandatorySubOneFpsConstraints) {
   FakeConstraints constraints;
   constraints.AddMandatory(MediaConstraintsInterface::kMaxFrameRate, 0.5);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
                  kMaxWaitMs);
   ASSERT_TRUE(capturer_->GetCaptureFormat() == NULL);
 }
 
-TEST_F(LocalVideoSourceTest, OptionalSubOneFpsConstraints) {
+TEST_F(VideoSourceTest, OptionalSubOneFpsConstraints) {
   FakeConstraints constraints;
   constraints.AddOptional(MediaConstraintsInterface::kMaxFrameRate, 0.5);
 
-  CreateLocalVideoSource(&constraints);
+  CreateVideoSource(&constraints);
   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
                  kMaxWaitMs);
   const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h b/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h
index ae968f728fe..89c90a1ec53 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h
@@ -47,6 +47,7 @@ class VideoSourceInterface : public MediaSourceInterface {
   virtual void AddSink(cricket::VideoRenderer* output) = 0;
   virtual void RemoveSink(cricket::VideoRenderer* output) = 0;
   virtual const cricket::VideoOptions* options() const = 0;
+  virtual cricket::VideoRenderer* FrameInput() = 0;
 
  protected:
   virtual ~VideoSourceInterface() {}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceproxy.h b/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceproxy.h
index be800777c30..991dd554c63 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceproxy.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videosourceproxy.h
@@ -42,6 +42,7 @@ BEGIN_PROXY_MAP(VideoSource)
   PROXY_METHOD1(void, AddSink, cricket::VideoRenderer*)
   PROXY_METHOD1(void, RemoveSink, cricket::VideoRenderer*)
   PROXY_CONSTMETHOD0(const cricket::VideoOptions*, options)
+  PROXY_METHOD0(cricket::VideoRenderer*, FrameInput)
   PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
   PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
 END_PROXY()
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc
index ec17ec7e718..7ab7815ba2a 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.cc
@@ -39,12 +39,12 @@ VideoTrack::VideoTrack(const std::string& label,
     : MediaStreamTrack<VideoTrackInterface>(label),
       video_source_(video_source) {
   if (video_source_)
-    video_source_->AddSink(FrameInput());
+    video_source_->AddSink(&renderers_);
 }
 
 VideoTrack::~VideoTrack() {
   if (video_source_)
-    video_source_->RemoveSink(FrameInput());
+    video_source_->RemoveSink(&renderers_);
 }
 
 std::string VideoTrack::kind() const {
@@ -59,10 +59,6 @@ void VideoTrack::RemoveRenderer(VideoRendererInterface* renderer) {
   renderers_.RemoveRenderer(renderer);
 }
 
-cricket::VideoRenderer* VideoTrack::FrameInput() {
-  return &renderers_;
-}
-
 bool VideoTrack::set_enabled(bool enable) {
   renderers_.SetEnabled(enable);
   return MediaStreamTrack<VideoTrackInterface>::set_enabled(enable);
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.h b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.h
index aefeb502cbb..acd1b755c58 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack.h
@@ -44,7 +44,6 @@ class VideoTrack : public MediaStreamTrack<VideoTrackInterface> {
 
   virtual void AddRenderer(VideoRendererInterface* renderer);
   virtual void RemoveRenderer(VideoRendererInterface* renderer);
-  virtual cricket::VideoRenderer* FrameInput();
   virtual VideoSourceInterface* GetSource() const {
     return video_source_.get();
   }
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack_unittest.cc
index 671e360f21a..4a30293c7ee 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/videotrack_unittest.cc
@@ -28,38 +28,53 @@
 #include <string>
 
 #include "talk/app/webrtc/test/fakevideotrackrenderer.h"
+#include "talk/app/webrtc/remotevideocapturer.h"
+#include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/base/gunit.h"
 #include "talk/base/scoped_ptr.h"
+#include "talk/media/base/fakemediaengine.h"
+#include "talk/media/devices/fakedevicemanager.h"
 #include "talk/media/webrtc/webrtcvideoframe.h"
+#include "talk/session/media/channelmanager.h"
 
 using webrtc::FakeVideoTrackRenderer;
+using webrtc::VideoSource;
 using webrtc::VideoTrack;
 using webrtc::VideoTrackInterface;
 
 // Test adding renderers to a video track and render to them by providing
-// VideoFrames to the track frame input.
+// frames to the source.
 TEST(VideoTrack, RenderVideo) {
   static const char kVideoTrackId[] = "track_id";
+
+  talk_base::scoped_ptr<cricket::ChannelManager> channel_manager_;
+  channel_manager_.reset(
+    new cricket::ChannelManager(new cricket::FakeMediaEngine(),
+                                new cricket::FakeDeviceManager(),
+                                talk_base::Thread::Current()));
+  ASSERT_TRUE(channel_manager_->Init());
   talk_base::scoped_refptr<VideoTrackInterface> video_track(
-      VideoTrack::Create(kVideoTrackId, NULL));
+      VideoTrack::Create(kVideoTrackId,
+                         VideoSource::Create(channel_manager_.get(),
+                                             new webrtc::RemoteVideoCapturer(),
+                                             NULL)));
   // FakeVideoTrackRenderer register itself to |video_track|
   talk_base::scoped_ptr<FakeVideoTrackRenderer> renderer_1(
       new FakeVideoTrackRenderer(video_track.get()));
 
-  cricket::VideoRenderer* render_input = video_track->FrameInput();
+  cricket::VideoRenderer* render_input = video_track->GetSource()->FrameInput();
   ASSERT_FALSE(render_input == NULL);
 
-  render_input->SetSize(123, 123, 0);
-  EXPECT_EQ(1, renderer_1->num_set_sizes());
-  EXPECT_EQ(123, renderer_1->width());
-  EXPECT_EQ(123, renderer_1->height());
-
   cricket::WebRtcVideoFrame frame;
   frame.InitToBlack(123, 123, 1, 1, 0, 0);
   render_input->RenderFrame(&frame);
   EXPECT_EQ(1, renderer_1->num_rendered_frames());
 
+  EXPECT_EQ(1, renderer_1->num_set_sizes());
+  EXPECT_EQ(123, renderer_1->width());
+  EXPECT_EQ(123, renderer_1->height());
+
   // FakeVideoTrackRenderer register itself to |video_track|
   talk_base::scoped_ptr<FakeVideoTrackRenderer> renderer_2(
       new FakeVideoTrackRenderer(video_track.get()));
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc
index 9910a5101be..60c427d16f1 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp.cc
@@ -42,6 +42,7 @@
 #include "talk/media/base/codec.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/cryptoparams.h"
+#include "talk/media/sctp/sctpdataengine.h"
 #include "talk/p2p/base/candidate.h"
 #include "talk/p2p/base/constants.h"
 #include "talk/p2p/base/port.h"
@@ -141,6 +142,7 @@ static const char kAttributeCandidateUsername[] = "username";
 static const char kAttributeCandidatePassword[] = "password";
 static const char kAttributeCandidateGeneration[] = "generation";
 static const char kAttributeFingerprint[] = "fingerprint";
+static const char kAttributeSetup[] = "setup";
 static const char kAttributeFmtp[] = "fmtp";
 static const char kAttributeRtpmap[] = "rtpmap";
 static const char kAttributeRtcp[] = "rtcp";
@@ -318,6 +320,9 @@ static bool ParseExtmap(const std::string& line,
 static bool ParseFingerprintAttribute(const std::string& line,
                                       talk_base::SSLFingerprint** fingerprint,
                                       SdpParseError* error);
+static bool ParseDtlsSetup(const std::string& line,
+                           cricket::ConnectionRole* role,
+                           SdpParseError* error);
 
 // Helper functions
 
@@ -902,7 +907,8 @@ bool SdpDeserialize(const std::string& message,
                     SdpParseError* error) {
   std::string session_id;
   std::string session_version;
-  TransportDescription session_td(NS_JINGLE_ICE_UDP, Candidates());
+  TransportDescription session_td(NS_JINGLE_ICE_UDP,
+                                  std::string(), std::string());
   RtpHeaderExtensions session_extmaps;
   cricket::SessionDescription* desc = new cricket::SessionDescription();
   std::vector<JsepIceCandidate*> candidates;
@@ -1226,8 +1232,23 @@ void BuildMediaDescription(const ContentInfo* content_info,
       os << kSdpDelimiterColon
          << fp->algorithm << kSdpDelimiterSpace
          << fp->GetRfc4572Fingerprint();
-
       AddLine(os.str(), message);
+
+      // Inserting setup attribute.
+      if (transport_info->description.connection_role !=
+              cricket::CONNECTIONROLE_NONE) {
+        // Making sure we are not using "passive" mode.
+        cricket::ConnectionRole role =
+            transport_info->description.connection_role;
+        ASSERT(role == cricket::CONNECTIONROLE_ACTIVE ||
+               role == cricket::CONNECTIONROLE_ACTPASS);
+        InitAttrLine(kAttributeSetup, &os);
+        std::string dtls_role_str = role == cricket::CONNECTIONROLE_ACTPASS ?
+            cricket::CONNECTIONROLE_ACTPASS_STR :
+            cricket::CONNECTIONROLE_ACTIVE_STR;
+        os << kSdpDelimiterColon << dtls_role_str;
+        AddLine(os.str(), message);
+      }
     }
   }
 
@@ -1296,7 +1317,16 @@ void BuildRtpContentAttributes(
 
   // RFC 4566
   // b=AS:<bandwidth>
-  if (media_desc->bandwidth() >= 1000) {
+  // We should always use the default bandwidth for RTP-based data
+  // channels.  Don't allow SDP to set the bandwidth, because that
+  // would give JS the opportunity to "break the Internet".
+  // TODO(pthatcher): But we need to temporarily allow the SDP to control
+  // this for backwards-compatibility.  Once we don't need that any
+  // more, remove this.
+  bool support_dc_sdp_bandwidth_temporarily = true;
+  if (media_desc->bandwidth() >= 1000 &&
+      (media_type != cricket::MEDIA_TYPE_DATA ||
+       support_dc_sdp_bandwidth_temporarily)) {
     InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
     os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
     AddLine(os.str(), message);
@@ -1796,6 +1826,10 @@ bool ParseSessionDescription(const std::string& message, size_t* pos,
         return false;
       }
       session_td->identity_fingerprint.reset(fingerprint);
+    } else if (HasAttribute(line, kAttributeSetup)) {
+      if (!ParseDtlsSetup(line, &(session_td->connection_role), error)) {
+        return false;
+      }
     } else if (HasAttribute(line, kAttributeMsidSemantics)) {
       std::string semantics;
       if (!GetValue(line, kAttributeMsidSemantics, &semantics, error)) {
@@ -1876,6 +1910,24 @@ static bool ParseFingerprintAttribute(const std::string& line,
   return true;
 }
 
+static bool ParseDtlsSetup(const std::string& line,
+                           cricket::ConnectionRole* role,
+                           SdpParseError* error) {
+  // setup-attr           =  "a=setup:" role
+  // role                 =  "active" / "passive" / "actpass" / "holdconn"
+  std::vector<std::string> fields;
+  talk_base::split(line.substr(kLinePrefixLength), kSdpDelimiterColon, &fields);
+  const size_t expected_fields = 2;
+  if (fields.size() != expected_fields) {
+    return ParseFailedExpectFieldNum(line, expected_fields, error);
+  }
+  std::string role_str = fields[1];
+  if (!cricket::StringToConnectionRole(role_str, role)) {
+    return ParseFailed(line, "Invalid attribute value.", error);
+  }
+  return true;
+}
+
 // RFC 3551
 //  PT   encoding    media type  clock rate   channels
 //                      name                    (Hz)
@@ -2039,6 +2091,7 @@ bool ParseMediaDescription(const std::string& message,
                                    session_td.ice_ufrag,
                                    session_td.ice_pwd,
                                    session_td.ice_mode,
+                                   session_td.connection_role,
                                    session_td.identity_fingerprint.get(),
                                    Candidates());
 
@@ -2062,6 +2115,16 @@ bool ParseMediaDescription(const std::string& message,
                     message, cricket::MEDIA_TYPE_DATA, mline_index, protocol,
                     codec_preference, pos, &content_name,
                     &transport, candidates, error));
+      // We should always use the default bandwidth for RTP-based data
+      // channels.  Don't allow SDP to set the bandwidth, because that
+      // would give JS the opportunity to "break the Internet".
+      // TODO(pthatcher): But we need to temporarily allow the SDP to control
+      // this for backwards-compatibility.  Once we don't need that any
+      // more, remove this.
+      bool support_dc_sdp_bandwidth_temporarily = true;
+      if (content.get() && !support_dc_sdp_bandwidth_temporarily) {
+        content->set_bandwidth(cricket::kAutoBandwidth);
+      }
     } else {
       LOG(LS_WARNING) << "Unsupported media type: " << line;
       continue;
@@ -2378,6 +2441,10 @@ bool ParseContent(const std::string& message,
         return false;
       }
       transport->identity_fingerprint.reset(fingerprint);
+    } else if (HasAttribute(line, kAttributeSetup)) {
+      if (!ParseDtlsSetup(line, &(transport->connection_role), error)) {
+        return false;
+      }
     } else if (is_rtp) {
       //
       // RTP specific attrubtes
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc
index 9e4c66072e5..b1505aae7df 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsdp_unittest.cc
@@ -281,7 +281,7 @@ static const char kSdpSctpDataChannelString[] =
     "a=ice-ufrag:ufrag_data\r\n"
     "a=ice-pwd:pwd_data\r\n"
     "a=mid:data_content_name\r\n"
-    "a=fmtp:5000 protocol=webrtc-datachannel; streams=10\r\n";
+    "a=fmtp:5000 protocol=webrtc-datachannel; streams=65536\r\n";
 
 static const char kSdpSctpDataChannelWithCandidatesString[] =
     "m=application 2345 DTLS/SCTP 5000\r\n"
@@ -296,7 +296,7 @@ static const char kSdpSctpDataChannelWithCandidatesString[] =
     "a=ice-ufrag:ufrag_data\r\n"
     "a=ice-pwd:pwd_data\r\n"
     "a=mid:data_content_name\r\n"
-    "a=fmtp:5000 protocol=webrtc-datachannel; streams=10\r\n";
+    "a=fmtp:5000 protocol=webrtc-datachannel; streams=65536\r\n";
 
 
 // One candidate reference string as per W3c spec.
@@ -485,19 +485,13 @@ class WebRtcSdpTest : public testing::Test {
     EXPECT_TRUE(desc_.AddTransportInfo(
         TransportInfo(kAudioContentName,
                       TransportDescription(NS_JINGLE_ICE_UDP,
-                                           std::vector<std::string>(),
                                            kCandidateUfragVoice,
-                                           kCandidatePwdVoice,
-                                           cricket::ICEMODE_FULL,
-                                           NULL, Candidates()))));
+                                           kCandidatePwdVoice))));
     EXPECT_TRUE(desc_.AddTransportInfo(
         TransportInfo(kVideoContentName,
                       TransportDescription(NS_JINGLE_ICE_UDP,
-                                           std::vector<std::string>(),
                                            kCandidateUfragVideo,
-                                           kCandidatePwdVideo,
-                                           cricket::ICEMODE_FULL,
-                                           NULL, Candidates()))));
+                                           kCandidatePwdVideo))));
 
     // v4 host
     int port = 1234;
@@ -860,9 +854,7 @@ class WebRtcSdpTest : public testing::Test {
     }
     TransportInfo transport_info(
         content_name, TransportDescription(NS_JINGLE_ICE_UDP,
-                                           std::vector<std::string>(),
-                                           ufrag, pwd, cricket::ICEMODE_FULL,
-                                           NULL, Candidates()));
+                                           ufrag, pwd));
     SessionDescription* desc =
         const_cast<SessionDescription*>(jdesc->description());
     desc->RemoveTransportInfoByName(content_name);
@@ -903,16 +895,18 @@ class WebRtcSdpTest : public testing::Test {
                                            std::vector<std::string>(),
                                            kCandidateUfragVoice,
                                            kCandidatePwdVoice,
-                                           cricket::ICEMODE_FULL, &fingerprint,
-                                           Candidates()))));
+                                           cricket::ICEMODE_FULL,
+                                           cricket::CONNECTIONROLE_NONE,
+                                           &fingerprint, Candidates()))));
     EXPECT_TRUE(desc_.AddTransportInfo(
         TransportInfo(kVideoContentName,
                       TransportDescription(NS_JINGLE_ICE_UDP,
                                            std::vector<std::string>(),
                                            kCandidateUfragVideo,
                                            kCandidatePwdVideo,
-                                           cricket::ICEMODE_FULL, &fingerprint,
-                                           Candidates()))));
+                                           cricket::ICEMODE_FULL,
+                                           cricket::CONNECTIONROLE_NONE,
+                                           &fingerprint, Candidates()))));
   }
 
   void AddExtmap() {
@@ -984,11 +978,8 @@ class WebRtcSdpTest : public testing::Test {
     EXPECT_TRUE(desc_.AddTransportInfo(
            TransportInfo(kDataContentName,
                          TransportDescription(NS_JINGLE_ICE_UDP,
-                                              std::vector<std::string>(),
                                               kCandidateUfragData,
-                                              kCandidatePwdData,
-                                              cricket::ICEMODE_FULL,
-                                              NULL, Candidates()))));
+                                              kCandidatePwdData))));
   }
 
   void AddRtpDataChannel() {
@@ -1011,11 +1002,8 @@ class WebRtcSdpTest : public testing::Test {
     EXPECT_TRUE(desc_.AddTransportInfo(
            TransportInfo(kDataContentName,
                          TransportDescription(NS_JINGLE_ICE_UDP,
-                                              std::vector<std::string>(),
                                               kCandidateUfragData,
-                                              kCandidatePwdData,
-                                              cricket::ICEMODE_FULL,
-                                              NULL, Candidates()))));
+                                              kCandidatePwdData))));
   }
 
   bool TestDeserializeDirection(cricket::MediaContentDirection direction) {
@@ -1439,6 +1427,28 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithSctpDataChannel) {
   EXPECT_EQ(message, expected_sdp);
 }
 
+TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
+  AddRtpDataChannel();
+  data_desc_->set_bandwidth(100*1000);
+  JsepSessionDescription jsep_desc(kDummyString);
+
+  ASSERT_TRUE(jsep_desc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+  std::string message = webrtc::SdpSerialize(jsep_desc);
+
+  std::string expected_sdp = kSdpString;
+  expected_sdp.append(kSdpRtpDataChannelString);
+  // We want to test that serializing data content ignores bandwidth
+  // settings (it should always be the default).  Thus, we don't do
+  // the following:
+  // TODO(pthatcher): We need to temporarily allow the SDP to control
+  // this for backwards-compatibility.  Once we don't need that any
+  // more, remove this.
+  InjectAfter("a=mid:data_content_name\r\na=sendrecv\r\n",
+              "b=AS:100\r\n",
+              &expected_sdp);
+  EXPECT_EQ(expected_sdp, message);
+}
+
 TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmap) {
   AddExtmap();
   JsepSessionDescription desc_with_extmap("dummy");
@@ -1751,6 +1761,32 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannels) {
   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
 }
 
+TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannelsAndBandwidth) {
+  AddRtpDataChannel();
+  JsepSessionDescription jdesc(kDummyString);
+  // We want to test that deserializing data content ignores bandwidth
+  // settings (it should always be the default).  Thus, we don't do
+  // the following:
+  // TODO(pthatcher): We need to temporarily allow the SDP to control
+  // this for backwards-compatibility.  Once we don't need that any
+  // more, remove this.
+  DataContentDescription* dcd = static_cast<DataContentDescription*>(
+     GetFirstDataContent(&desc_)->description);
+  dcd->set_bandwidth(100 * 1000);
+  ASSERT_TRUE(jdesc.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
+
+  std::string sdp_with_bandwidth = kSdpString;
+  sdp_with_bandwidth.append(kSdpRtpDataChannelString);
+  InjectAfter("a=mid:data_content_name\r\n",
+              "b=AS:100\r\n",
+              &sdp_with_bandwidth);
+  JsepSessionDescription jdesc_with_bandwidth(kDummyString);
+
+  EXPECT_TRUE(
+      SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
+  EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_with_bandwidth));
+}
+
 TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithSessionLevelExtmap) {
   TestDeserializeExtmap(true, false);
 }
@@ -1966,3 +2002,60 @@ TEST_F(WebRtcSdpTest, RoundTripSdpWithSctpDataChannelsWithCandidates) {
   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
   EXPECT_EQ(sdp_with_data, webrtc::SdpSerialize(jdesc_output));
 }
+
+TEST_F(WebRtcSdpTest, SerializeDtlsSetupAttribute) {
+  AddFingerprint();
+  TransportInfo audio_transport_info =
+      *(desc_.GetTransportInfoByName(kAudioContentName));
+  EXPECT_EQ(cricket::CONNECTIONROLE_NONE,
+            audio_transport_info.description.connection_role);
+  audio_transport_info.description.connection_role =
+        cricket::CONNECTIONROLE_ACTIVE;
+
+  TransportInfo video_transport_info =
+      *(desc_.GetTransportInfoByName(kVideoContentName));
+  EXPECT_EQ(cricket::CONNECTIONROLE_NONE,
+            video_transport_info.description.connection_role);
+  video_transport_info.description.connection_role =
+        cricket::CONNECTIONROLE_ACTIVE;
+
+  desc_.RemoveTransportInfoByName(kAudioContentName);
+  desc_.RemoveTransportInfoByName(kVideoContentName);
+
+  desc_.AddTransportInfo(audio_transport_info);
+  desc_.AddTransportInfo(video_transport_info);
+
+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
+                                jdesc_.session_id(),
+                                jdesc_.session_version()));
+  std::string message = webrtc::SdpSerialize(jdesc_);
+  std::string sdp_with_dtlssetup = kSdpFullString;
+
+  // Fingerprint attribute is necessary to add DTLS setup attribute.
+  InjectAfter(kAttributeIcePwdVoice,
+              kFingerprint, &sdp_with_dtlssetup);
+  InjectAfter(kAttributeIcePwdVideo,
+              kFingerprint, &sdp_with_dtlssetup);
+  // Now adding |setup| attribute.
+  InjectAfter(kFingerprint,
+              "a=setup:active\r\n", &sdp_with_dtlssetup);
+  EXPECT_EQ(sdp_with_dtlssetup, message);
+}
+
+TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttribute) {
+  JsepSessionDescription jdesc_with_dtlssetup(kDummyString);
+  std::string sdp_with_dtlssetup = kSdpFullString;
+  InjectAfter(kSessionTime,
+              "a=setup:actpass\r\n",
+              &sdp_with_dtlssetup);
+  EXPECT_TRUE(SdpDeserialize(sdp_with_dtlssetup, &jdesc_with_dtlssetup));
+  cricket::SessionDescription* desc = jdesc_with_dtlssetup.description();
+  const cricket::TransportInfo* atinfo =
+      desc->GetTransportInfoByName("audio_content_name");
+  EXPECT_EQ(cricket::CONNECTIONROLE_ACTPASS,
+            atinfo->description.connection_role);
+  const cricket::TransportInfo* vtinfo =
+        desc->GetTransportInfoByName("video_content_name");
+  EXPECT_EQ(cricket::CONNECTIONROLE_ACTPASS,
+            vtinfo->description.connection_role);
+}
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc
index b056757df43..f1fb40d51a0 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.cc
@@ -422,6 +422,17 @@ bool WebRtcSession::Initialize(
   // mandatory constraints can be fulfilled. Note that |constraints|
   // can be null.
   bool value;
+
+  // Enable DTLS by default if |dtls_identity_service| is valid.
+  bool dtls_enabled = (dtls_identity_service != NULL);
+  // |constraints| can override the default |dtls_enabled| value.
+  if (FindConstraint(
+        constraints,
+        MediaConstraintsInterface::kEnableDtlsSrtp,
+        &value, NULL)) {
+    dtls_enabled = value;
+  }
+
   // Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
   // It takes precendence over the kEnableSctpDataChannels constraint.
   if (FindConstraint(
@@ -434,11 +445,6 @@ bool WebRtcSession::Initialize(
         constraints,
         MediaConstraintsInterface::kEnableSctpDataChannels,
         &value, NULL) && value;
-    bool dtls_enabled = FindConstraint(
-        constraints,
-        MediaConstraintsInterface::kEnableDtlsSrtp,
-        &value, NULL) && value;
-
     // DTLS has to be enabled to use SCTP.
     if (sctp_enabled && dtls_enabled) {
       LOG(LS_INFO) << "Allowing SCTP data engine.";
@@ -467,7 +473,7 @@ bool WebRtcSession::Initialize(
       this,
       id(),
       data_channel_type_,
-      constraints));
+      dtls_enabled));
 
   webrtc_session_desc_factory_->SignalIdentityReady.connect(
       this, &WebRtcSession::OnIdentityReady);
@@ -505,6 +511,26 @@ cricket::SecureMediaPolicy WebRtcSession::secure_policy() const {
   return webrtc_session_desc_factory_->secure();
 }
 
+bool WebRtcSession::GetSslRole(talk_base::SSLRole* role) {
+  if (local_description() == NULL || remote_description() == NULL) {
+    LOG(LS_INFO) << "Local and Remote descriptions must be applied to get "
+                 << "SSL Role of the session.";
+    return false;
+  }
+
+  // TODO(mallinath) - Return role of each transport, as role may differ from
+  // one another.
+  // In current implementaion we just return the role of first transport in the
+  // transport map.
+  for (cricket::TransportMap::const_iterator iter = transport_proxies().begin();
+       iter != transport_proxies().end(); ++iter) {
+    if (iter->second->impl()) {
+      return iter->second->impl()->GetSslRole(role);
+    }
+  }
+  return false;
+}
+
 void WebRtcSession::CreateOffer(CreateSessionDescriptionObserver* observer,
                                 const MediaConstraintsInterface* constraints) {
   webrtc_session_desc_factory_->CreateOffer(observer, constraints);
@@ -517,42 +543,22 @@ void WebRtcSession::CreateAnswer(CreateSessionDescriptionObserver* observer,
 
 bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
                                         std::string* err_desc) {
-  cricket::SecureMediaPolicy secure_policy =
-      webrtc_session_desc_factory_->secure();
   // Takes the ownership of |desc| regardless of the result.
   talk_base::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
 
-  if (error() != cricket::BaseSession::ERROR_NONE) {
-    return BadLocalSdp(SessionErrorMsg(error()), err_desc);
-  }
-
-  if (!desc || !desc->description()) {
-    return BadLocalSdp(kInvalidSdp, err_desc);
-  }
-
-  if (!VerifyBundleSettings(desc->description())) {
-    return BadLocalSdp(kBundleWithoutRtcpMux, err_desc);
-  }
-
-  Action action = GetAction(desc->type());
-  if (!ExpectSetLocalDescription(action)) {
-    std::string type = desc->type();
-    return BadLocalSdp(BadStateErrMsg(type, state()), err_desc);
-  }
-  if (secure_policy == cricket::SEC_REQUIRED &&
-      !VerifyCrypto(desc->description())) {
-    return BadLocalSdp(kSdpWithoutCrypto, err_desc);
-  }
-  if (action == kAnswer && !VerifyMediaDescriptions(
-          desc->description(), remote_description()->description())) {
-    return BadLocalSdp(kMlineMismatch, err_desc);
+  // Validate SDP.
+  if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {
+    return false;
   }
 
   // Update the initiator flag if this session is the initiator.
+  Action action = GetAction(desc->type());
   if (state() == STATE_INIT && action == kOffer) {
     set_initiator(true);
   }
 
+  cricket::SecureMediaPolicy secure_policy =
+      webrtc_session_desc_factory_->secure();
   // Update the MediaContentDescription crypto settings as per the policy set.
   UpdateSessionDescriptionSecurePolicy(secure_policy, desc->description());
 
@@ -589,40 +595,16 @@ bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
 
 bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,
                                          std::string* err_desc) {
-  cricket::SecureMediaPolicy secure_policy =
-      webrtc_session_desc_factory_->secure();
   // Takes the ownership of |desc| regardless of the result.
   talk_base::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
 
-  if (error() != cricket::BaseSession::ERROR_NONE) {
-    return BadRemoteSdp(SessionErrorMsg(error()), err_desc);
-  }
-
-  if (!desc || !desc->description()) {
-    return BadRemoteSdp(kInvalidSdp, err_desc);
-  }
-
-  if (!VerifyBundleSettings(desc->description())) {
-    return BadRemoteSdp(kBundleWithoutRtcpMux, err_desc);
-  }
-
-  Action action = GetAction(desc->type());
-  if (!ExpectSetRemoteDescription(action)) {
-    std::string type = desc->type();
-    return BadRemoteSdp(BadStateErrMsg(type, state()), err_desc);
-  }
-
-  if (action == kAnswer && !VerifyMediaDescriptions(
-          desc->description(), local_description()->description())) {
-    return BadRemoteSdp(kMlineMismatch, err_desc);
-  }
-
-  if (secure_policy == cricket::SEC_REQUIRED &&
-      !VerifyCrypto(desc->description())) {
-    return BadRemoteSdp(kSdpWithoutCrypto, err_desc);
+  // Validate SDP.
+  if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {
+    return false;
   }
 
   // Transport and Media channels will be created only when offer is set.
+  Action action = GetAction(desc->type());
   if (action == kOffer && !CreateChannels(desc->description())) {
     // TODO(mallinath) - Handle CreateChannel failure, as new local description
     // is applied. Restore back to old description.
@@ -962,7 +944,6 @@ talk_base::scoped_refptr<DataChannel> WebRtcSession::CreateDataChannel(
     if (data_channel_.get()) {
       channel->SetReceiveSsrc(new_config.id);
       channel->SetSendSsrc(new_config.id);
-      channel->ConnectToDataSession();
     }
     if (!config->negotiated) {
       talk_base::Buffer *payload = new talk_base::Buffer;
@@ -1094,36 +1075,6 @@ void WebRtcSession::OnTransportProxyCandidatesReady(
   ProcessNewLocalCandidate(proxy->content_name(), candidates);
 }
 
-bool WebRtcSession::ExpectSetLocalDescription(Action action) {
-  return ((action == kOffer && state() == STATE_INIT) ||
-          // update local offer
-          (action == kOffer && state() == STATE_SENTINITIATE) ||
-          // update the current ongoing session.
-          (action == kOffer && state() == STATE_RECEIVEDACCEPT) ||
-          (action == kOffer && state() == STATE_SENTACCEPT) ||
-          (action == kOffer && state() == STATE_INPROGRESS) ||
-          // accept remote offer
-          (action == kAnswer && state() == STATE_RECEIVEDINITIATE) ||
-          (action == kAnswer && state() == STATE_SENTPRACCEPT) ||
-          (action == kPrAnswer && state() == STATE_RECEIVEDINITIATE) ||
-          (action == kPrAnswer && state() == STATE_SENTPRACCEPT));
-}
-
-bool WebRtcSession::ExpectSetRemoteDescription(Action action) {
-  return ((action == kOffer && state() == STATE_INIT) ||
-          // update remote offer
-          (action == kOffer && state() == STATE_RECEIVEDINITIATE) ||
-          // update the current ongoing session
-          (action == kOffer && state() == STATE_RECEIVEDACCEPT) ||
-          (action == kOffer && state() == STATE_SENTACCEPT) ||
-          (action == kOffer && state() == STATE_INPROGRESS) ||
-          // accept local offer
-          (action == kAnswer && state() == STATE_SENTINITIATE) ||
-          (action == kAnswer && state() == STATE_RECEIVEDPRACCEPT) ||
-          (action == kPrAnswer && state() == STATE_SENTINITIATE) ||
-          (action == kPrAnswer && state() == STATE_RECEIVEDPRACCEPT));
-}
-
 void WebRtcSession::OnCandidatesAllocationDone() {
   ASSERT(signaling_thread()->IsCurrent());
   if (ice_observer_) {
@@ -1378,7 +1329,7 @@ void WebRtcSession::OnDataReceived(
 }
 
 // Returns false if bundle is enabled and rtcp_mux is disabled.
-bool WebRtcSession::VerifyBundleSettings(const SessionDescription* desc) {
+bool WebRtcSession::ValidateBundleSettings(const SessionDescription* desc) {
   bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
   if (!bundle_enabled)
     return true;
@@ -1409,4 +1360,79 @@ bool WebRtcSession::HasRtcpMuxEnabled(
   return description->rtcp_mux();
 }
 
+bool WebRtcSession::ValidateSessionDescription(
+    const SessionDescriptionInterface* sdesc,
+    cricket::ContentSource source, std::string* error_desc) {
+
+  if (error() != cricket::BaseSession::ERROR_NONE) {
+    return BadSdp(source, SessionErrorMsg(error()), error_desc);
+  }
+
+  if (!sdesc || !sdesc->description()) {
+    return BadSdp(source, kInvalidSdp, error_desc);
+  }
+
+  std::string type = sdesc->type();
+  Action action = GetAction(sdesc->type());
+  if (source == cricket::CS_LOCAL) {
+    if (!ExpectSetLocalDescription(action))
+      return BadSdp(source, BadStateErrMsg(type, state()), error_desc);
+  } else {
+    if (!ExpectSetRemoteDescription(action))
+      return BadSdp(source, BadStateErrMsg(type, state()), error_desc);
+  }
+
+  // Verify crypto settings.
+  if (webrtc_session_desc_factory_->secure() == cricket::SEC_REQUIRED &&
+      !VerifyCrypto(sdesc->description())) {
+    return BadSdp(source, kSdpWithoutCrypto, error_desc);
+  }
+
+  if (!ValidateBundleSettings(sdesc->description())) {
+    return BadSdp(source, kBundleWithoutRtcpMux, error_desc);
+  }
+
+  // Verify m-lines in Answer when compared against Offer.
+  if (action == kAnswer) {
+    const cricket::SessionDescription* offer_desc =
+        (source == cricket::CS_LOCAL) ? remote_description()->description() :
+            local_description()->description();
+    if (!VerifyMediaDescriptions(sdesc->description(), offer_desc)) {
+      return BadSdp(source, kMlineMismatch, error_desc);
+    }
+  }
+
+  return true;
+}
+
+bool WebRtcSession::ExpectSetLocalDescription(Action action) {
+  return ((action == kOffer && state() == STATE_INIT) ||
+          // update local offer
+          (action == kOffer && state() == STATE_SENTINITIATE) ||
+          // update the current ongoing session.
+          (action == kOffer && state() == STATE_RECEIVEDACCEPT) ||
+          (action == kOffer && state() == STATE_SENTACCEPT) ||
+          (action == kOffer && state() == STATE_INPROGRESS) ||
+          // accept remote offer
+          (action == kAnswer && state() == STATE_RECEIVEDINITIATE) ||
+          (action == kAnswer && state() == STATE_SENTPRACCEPT) ||
+          (action == kPrAnswer && state() == STATE_RECEIVEDINITIATE) ||
+          (action == kPrAnswer && state() == STATE_SENTPRACCEPT));
+}
+
+bool WebRtcSession::ExpectSetRemoteDescription(Action action) {
+  return ((action == kOffer && state() == STATE_INIT) ||
+          // update remote offer
+          (action == kOffer && state() == STATE_RECEIVEDINITIATE) ||
+          // update the current ongoing session
+          (action == kOffer && state() == STATE_RECEIVEDACCEPT) ||
+          (action == kOffer && state() == STATE_SENTACCEPT) ||
+          (action == kOffer && state() == STATE_INPROGRESS) ||
+          // accept local offer
+          (action == kAnswer && state() == STATE_SENTINITIATE) ||
+          (action == kAnswer && state() == STATE_RECEIVEDPRACCEPT) ||
+          (action == kPrAnswer && state() == STATE_SENTINITIATE) ||
+          (action == kPrAnswer && state() == STATE_RECEIVEDPRACCEPT));
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h
index 202ca66e129..0cb049f4983 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession.h
@@ -130,6 +130,9 @@ class WebRtcSession : public cricket::BaseSession,
   void set_secure_policy(cricket::SecureMediaPolicy secure_policy);
   cricket::SecureMediaPolicy secure_policy() const;
 
+  // Get current ssl role from transport.
+  bool GetSslRole(talk_base::SSLRole* role);
+
   // Generic error message callback from WebRtcSession.
   // TODO - It may be necessary to supply error code as well.
   sigslot::signal0<> SignalError;
@@ -152,9 +155,6 @@ class WebRtcSession : public cricket::BaseSession,
     return remote_desc_.get();
   }
 
-  void set_secure(cricket::SecureMediaPolicy secure_policy);
-  cricket::SecureMediaPolicy secure();
-
   // Get the id used as a media stream track's "id" field from ssrc.
   virtual bool GetTrackIdBySsrc(uint32 ssrc, std::string* id);
 
@@ -223,10 +223,6 @@ class WebRtcSession : public cricket::BaseSession,
       const cricket::Candidates& candidates);
   virtual void OnCandidatesAllocationDone();
 
-  // Check if a call to SetLocalDescription is acceptable with |action|.
-  bool ExpectSetLocalDescription(Action action);
-  // Check if a call to SetRemoteDescription is acceptable with |action|.
-  bool ExpectSetRemoteDescription(Action action);
   // Creates local session description with audio and video contents.
   bool CreateDefaultLocalDescription();
   // Enables media channels to allow sending of media.
@@ -275,8 +271,20 @@ class WebRtcSession : public cricket::BaseSession,
   std::string BadStateErrMsg(const std::string& type, State state);
   void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state);
 
-  bool VerifyBundleSettings(const cricket::SessionDescription* desc);
+  bool ValidateBundleSettings(const cricket::SessionDescription* desc);
   bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
+  // Below methods are helper methods which verifies SDP.
+  bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc,
+                                  cricket::ContentSource source,
+                                  std::string* error_desc);
+
+  // Check if a call to SetLocalDescription is acceptable with |action|.
+  bool ExpectSetLocalDescription(Action action);
+  // Check if a call to SetRemoteDescription is acceptable with |action|.
+  bool ExpectSetRemoteDescription(Action action);
+  // Verifies a=setup attribute as per RFC 5763.
+  bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc,
+                                  Action action);
 
   talk_base::scoped_ptr<cricket::VoiceChannel> voice_channel_;
   talk_base::scoped_ptr<cricket::VideoChannel> video_channel_;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc
index bcb209a0abe..1531e0efc1a 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsession_unittest.cc
@@ -275,7 +275,8 @@ class WebRtcSessionTest : public testing::Test {
       ss_scope_(fss_.get()),
       stun_server_(talk_base::Thread::Current(), kStunAddr),
       allocator_(&network_manager_, kStunAddr,
-                 SocketAddress(), SocketAddress(), SocketAddress()) {
+                 SocketAddress(), SocketAddress(), SocketAddress()),
+      mediastream_signaling_(channel_manager_.get()) {
     tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID);
     allocator_.set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
                          cricket::PORTALLOCATOR_DISABLE_RELAY |
@@ -313,18 +314,7 @@ class WebRtcSessionTest : public testing::Test {
     Init(NULL);
   }
 
-  void InitWithDtls() {
-    constraints_.reset(new FakeConstraints());
-    constraints_->AddOptional(
-        webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true);
-
-    Init(NULL);
-  }
-
-  void InitWithAsyncDtls(bool identity_request_should_fail) {
-    constraints_.reset(new FakeConstraints());
-    constraints_->AddOptional(
-        webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true);
+  void InitWithDtls(bool identity_request_should_fail = false) {
     FakeIdentityService* identity_service = new FakeIdentityService();
     identity_service->set_should_fail(identity_request_should_fail);
     Init(identity_service);
@@ -348,7 +338,7 @@ class WebRtcSessionTest : public testing::Test {
     session_->CreateOffer(observer, constraints);
     EXPECT_TRUE_WAIT(
         observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
-        1000);
+        2000);
     return observer->ReleaseDescription();
   }
 
@@ -359,7 +349,7 @@ class WebRtcSessionTest : public testing::Test {
     session_->CreateAnswer(observer, constraints);
     EXPECT_TRUE_WAIT(
         observer->state() != WebRtcSessionCreateSDPObserverForTest::kInit,
-        1000);
+        2000);
     return observer->ReleaseDescription();
   }
 
@@ -819,7 +809,7 @@ class WebRtcSessionTest : public testing::Test {
 
   void VerifyMultipleAsyncCreateDescription(
       bool success, CreateSessionDescriptionRequest::Type type) {
-    InitWithAsyncDtls(!success);
+    InitWithDtls(!success);
 
     if (type == CreateSessionDescriptionRequest::kAnswer) {
       cricket::MediaSessionOptions options;
@@ -2490,19 +2480,54 @@ TEST_F(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) {
       webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
   constraints_->AddOptional(
     webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
-  constraints_->AddOptional(
-      webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true);
-  Init(NULL);
+  InitWithDtls(false);
 
   SetLocalDescriptionWithDataChannel();
   EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
 }
 
+TEST_F(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) {
+  MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
+
+  constraints_.reset(new FakeConstraints());
+  constraints_->AddOptional(
+      webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
+  InitWithDtls(false);
+
+  talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
+  EXPECT_TRUE(offer->description()->GetContentByName("data") == NULL);
+  EXPECT_TRUE(offer->description()->GetTransportInfoByName("data") == NULL);
+}
+
+TEST_F(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) {
+  MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
+  SetFactoryDtlsSrtp();
+  constraints_.reset(new FakeConstraints());
+  constraints_->AddOptional(
+      webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
+  InitWithDtls(false);
+
+  // Create remote offer with SCTP.
+  cricket::MediaSessionOptions options;
+  options.data_channel_type = cricket::DCT_SCTP;
+  JsepSessionDescription* offer =
+      CreateRemoteOffer(options, cricket::SEC_ENABLED);
+  SetRemoteDescriptionWithoutError(offer);
+
+  // Verifies the answer contains SCTP.
+  talk_base::scoped_ptr<SessionDescriptionInterface> answer(CreateAnswer(NULL));
+  EXPECT_TRUE(answer != NULL);
+  EXPECT_TRUE(answer->description()->GetContentByName("data") != NULL);
+  EXPECT_TRUE(answer->description()->GetTransportInfoByName("data") != NULL);
+}
+
 TEST_F(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
-  Init(NULL);
+  constraints_->AddOptional(
+      webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
+  InitWithDtls(false);
 
   SetLocalDescriptionWithDataChannel();
   EXPECT_EQ(cricket::DCT_NONE, data_engine_->last_channel_type());
@@ -2514,9 +2539,7 @@ TEST_F(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
   constraints_.reset(new FakeConstraints());
   constraints_->AddOptional(
       webrtc::MediaConstraintsInterface::kEnableSctpDataChannels, true);
-  constraints_->AddOptional(
-      webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true);
-  Init(NULL);
+  InitWithDtls(false);
 
   SetLocalDescriptionWithDataChannel();
   EXPECT_EQ(cricket::DCT_SCTP, data_engine_->last_channel_type());
@@ -2526,7 +2549,7 @@ TEST_F(WebRtcSessionTest, TestSctpDataChannelWithDtls) {
 // identity generation is finished.
 TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
   MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
-  InitWithAsyncDtls(false);
+  InitWithDtls(false);
 
   EXPECT_TRUE(session_->waiting_for_identity());
   talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
@@ -2537,7 +2560,7 @@ TEST_F(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) {
 // identity generation is finished.
 TEST_F(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
   MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
-  InitWithAsyncDtls(false);
+  InitWithDtls(false);
 
   cricket::MediaSessionOptions options;
   scoped_ptr<JsepSessionDescription> offer(
@@ -2553,7 +2576,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) {
 // identity generation is finished.
 TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
   MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
-  InitWithAsyncDtls(false);
+  InitWithDtls(false);
 
   EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
   talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
@@ -2564,7 +2587,7 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) {
 // identity generation fails.
 TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) {
   MAYBE_SKIP_TEST(talk_base::SSLStreamAdapter::HaveDtlsSrtp);
-  InitWithAsyncDtls(true);
+  InitWithDtls(true);
 
   EXPECT_TRUE_WAIT(!session_->waiting_for_identity(), 1000);
   talk_base::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer(NULL));
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
index 2021085aa27..30c49a718ad 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
@@ -113,7 +113,7 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
     WebRtcSession* session,
     const std::string& session_id,
     cricket::DataChannelType dct,
-    const MediaConstraintsInterface* constraints)
+    bool dtls_enabled)
     : signaling_thread_(signaling_thread),
       mediastream_signaling_(mediastream_signaling),
       session_desc_factory_(channel_manager, &transport_desc_factory_),
@@ -132,38 +132,31 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
   // By default SRTP-SDES is enabled in WebRtc.
   set_secure(cricket::SEC_REQUIRED);
 
-  // Enable DTLS-SRTP if the constraint is set.
-  bool dtls_enabled = false;
-  if (!FindConstraint(
-           constraints, MediaConstraintsInterface::kEnableDtlsSrtp,
-           &dtls_enabled, NULL) ||
-      !dtls_enabled) {
-    return;
-  }
-  // DTLS is enabled.
-  if (identity_service_.get()) {
-    identity_request_observer_ =
-      new talk_base::RefCountedObject<WebRtcIdentityRequestObserver>();
-
-    identity_request_observer_->SignalRequestFailed.connect(
-        this, &WebRtcSessionDescriptionFactory::OnIdentityRequestFailed);
-    identity_request_observer_->SignalIdentityReady.connect(
-        this, &WebRtcSessionDescriptionFactory::OnIdentityReady);
-
-    if (identity_service_->RequestIdentity(kWebRTCIdentityName,
-                                           kWebRTCIdentityName,
-                                           identity_request_observer_)) {
-      LOG(LS_VERBOSE) << "DTLS-SRTP enabled; sent DTLS identity request.";
-      identity_request_state_ = IDENTITY_WAITING;
+  if (dtls_enabled) {
+    if (identity_service_.get()) {
+      identity_request_observer_ =
+        new talk_base::RefCountedObject<WebRtcIdentityRequestObserver>();
+
+      identity_request_observer_->SignalRequestFailed.connect(
+          this, &WebRtcSessionDescriptionFactory::OnIdentityRequestFailed);
+      identity_request_observer_->SignalIdentityReady.connect(
+          this, &WebRtcSessionDescriptionFactory::OnIdentityReady);
+
+      if (identity_service_->RequestIdentity(kWebRTCIdentityName,
+                                             kWebRTCIdentityName,
+                                             identity_request_observer_)) {
+        LOG(LS_VERBOSE) << "DTLS-SRTP enabled; sent DTLS identity request.";
+        identity_request_state_ = IDENTITY_WAITING;
+      } else {
+        LOG(LS_ERROR) << "Failed to send DTLS identity request.";
+        identity_request_state_ = IDENTITY_FAILED;
+      }
     } else {
-      LOG(LS_ERROR) << "Failed to send DTLS identity request.";
-      identity_request_state_ = IDENTITY_FAILED;
+      identity_request_state_ = IDENTITY_WAITING;
+      // Do not generate the identity in the constructor since the caller has
+      // not got a chance to connect to SignalIdentityReady.
+      signaling_thread_->Post(this, MSG_GENERATE_IDENTITY, NULL);
     }
-  } else {
-    identity_request_state_ = IDENTITY_WAITING;
-    // Do not generate the identity in the constructor since the caller has
-    // not got a chance to connect to SignalIdentityReady.
-    signaling_thread_->Post(this, MSG_GENERATE_IDENTITY, NULL);
   }
 }
 
@@ -197,7 +190,8 @@ void WebRtcSessionDescriptionFactory::CreateOffer(
     return;
   }
 
-  if (data_channel_type_ == cricket::DCT_SCTP) {
+  if (data_channel_type_ == cricket::DCT_SCTP &&
+      mediastream_signaling_->HasDataChannels()) {
     options.data_channel_type = cricket::DCT_SCTP;
   }
 
@@ -249,6 +243,9 @@ void WebRtcSessionDescriptionFactory::CreateAnswer(
     PostCreateSessionDescriptionFailed(observer, error);
     return;
   }
+  // RTP data channel is handled in MediaSessionOptions::AddStream. SCTP streams
+  // are not signaled in the SDP so does not go through that path and must be
+  // handled here.
   if (data_channel_type_ == cricket::DCT_SCTP) {
     options.data_channel_type = cricket::DCT_SCTP;
   }
@@ -343,6 +340,13 @@ void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
   // an answer should also contain new ice ufrag and password if an offer has
   // been received with new ufrag and password.
   request.options.transport_options.ice_restart = session_->IceRestartPending();
+  // We should pass current ssl role to the transport description factory, if
+  // there is already an existing ongoing session.
+  talk_base::SSLRole ssl_role;
+  if (session_->GetSslRole(&ssl_role)) {
+    request.options.transport_options.prefer_passive_role =
+        (talk_base::SSL_SERVER == ssl_role);
+  }
 
   cricket::SessionDescription* desc(session_desc_factory_.CreateAnswer(
       static_cast<cricket::BaseSession*>(session_)->remote_description(),
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h
index 0c96c69b637..ba34e913749 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h
+++ b/chromium/third_party/libjingle/source/talk/app/webrtc/webrtcsessiondescriptionfactory.h
@@ -102,7 +102,7 @@ class WebRtcSessionDescriptionFactory : public talk_base::MessageHandler,
       WebRtcSession* session,
       const std::string& session_id,
       cricket::DataChannelType dct,
-      const MediaConstraintsInterface* constraints);
+      bool dtls_enabled);
   virtual ~WebRtcSessionDescriptionFactory();
 
   static void CopyCandidatesFromSessionDescription(
diff --git a/chromium/third_party/libjingle/source/talk/base/asyncpacketsocket.h b/chromium/third_party/libjingle/source/talk/base/asyncpacketsocket.h
index a88f770ca9f..3b4748f510f 100644
--- a/chromium/third_party/libjingle/source/talk/base/asyncpacketsocket.h
+++ b/chromium/third_party/libjingle/source/talk/base/asyncpacketsocket.h
@@ -28,6 +28,7 @@
 #ifndef TALK_BASE_ASYNCPACKETSOCKET_H_
 #define TALK_BASE_ASYNCPACKETSOCKET_H_
 
+#include "talk/base/dscp.h"
 #include "talk/base/sigslot.h"
 #include "talk/base/socket.h"
 
@@ -56,8 +57,9 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
   virtual SocketAddress GetRemoteAddress() const = 0;
 
   // Send a packet.
-  virtual int Send(const void *pv, size_t cb) = 0;
-  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr) = 0;
+  virtual int Send(const void *pv, size_t cb, DiffServCodePoint dscp) = 0;
+  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
+                     DiffServCodePoint) = 0;
 
   // Close the socket.
   virtual int Close() = 0;
diff --git a/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.cc b/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.cc
index 095413d3eb8..517e799c422 100644
--- a/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.cc
+++ b/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.cc
@@ -141,10 +141,12 @@ void AsyncTCPSocketBase::SetError(int error) {
   return socket_->SetError(error);
 }
 
+// TODO(mallinath) - Add support of setting DSCP code on AsyncSocket.
 int AsyncTCPSocketBase::SendTo(const void *pv, size_t cb,
-                               const SocketAddress& addr) {
+                               const SocketAddress& addr,
+                               DiffServCodePoint dscp) {
   if (addr == GetRemoteAddress())
-    return Send(pv, cb);
+    return Send(pv, cb, dscp);
 
   ASSERT(false);
   socket_->SetError(ENOTCONN);
@@ -261,7 +263,8 @@ AsyncTCPSocket::AsyncTCPSocket(AsyncSocket* socket, bool listen)
     : AsyncTCPSocketBase(socket, listen, kBufSize) {
 }
 
-int AsyncTCPSocket::Send(const void *pv, size_t cb) {
+// TODO(mallinath) - Add support of setting DSCP code on AsyncSocket.
+int AsyncTCPSocket::Send(const void *pv, size_t cb, DiffServCodePoint dscp) {
   if (cb > kBufSize) {
     SetError(EMSGSIZE);
     return -1;
diff --git a/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.h b/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.h
index b34ce188e9d..a0e7a7e2f41 100644
--- a/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.h
+++ b/chromium/third_party/libjingle/source/talk/base/asynctcpsocket.h
@@ -43,14 +43,15 @@ class AsyncTCPSocketBase : public AsyncPacketSocket {
   virtual ~AsyncTCPSocketBase();
 
   // Pure virtual methods to send and recv data.
-  virtual int Send(const void *pv, size_t cb) = 0;
+  virtual int Send(const void *pv, size_t cb, DiffServCodePoint dscp) = 0;
   virtual void ProcessInput(char* data, size_t* len) = 0;
   // Signals incoming connection.
   virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
 
   virtual SocketAddress GetLocalAddress() const;
   virtual SocketAddress GetRemoteAddress() const;
-  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr);
+  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
+                     DiffServCodePoint dscp);
   virtual int Close();
 
   virtual State GetState() const;
@@ -101,7 +102,7 @@ class AsyncTCPSocket : public AsyncTCPSocketBase {
   AsyncTCPSocket(AsyncSocket* socket, bool listen);
   virtual ~AsyncTCPSocket() {}
 
-  virtual int Send(const void* pv, size_t cb);
+  virtual int Send(const void* pv, size_t cb, DiffServCodePoint dscp);
   virtual void ProcessInput(char* data, size_t* len);
   virtual void HandleIncomingConnection(AsyncSocket* socket);
 
diff --git a/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.cc b/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.cc
index 6388ce7ce6a..97e5dff9836 100644
--- a/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.cc
+++ b/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.cc
@@ -75,12 +75,14 @@ SocketAddress AsyncUDPSocket::GetRemoteAddress() const {
   return socket_->GetRemoteAddress();
 }
 
-int AsyncUDPSocket::Send(const void *pv, size_t cb) {
+// TODO(mallinath) - Add support of setting DSCP code on AsyncSocket.
+int AsyncUDPSocket::Send(const void *pv, size_t cb, DiffServCodePoint dscp) {
   return socket_->Send(pv, cb);
 }
 
-int AsyncUDPSocket::SendTo(
-    const void *pv, size_t cb, const SocketAddress& addr) {
+// TODO(mallinath) - Add support of setting DSCP code on AsyncSocket.
+int AsyncUDPSocket::SendTo(const void *pv, size_t cb,
+                           const SocketAddress& addr, DiffServCodePoint dscp) {
   return socket_->SendTo(pv, cb, addr);
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.h b/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.h
index 1bf2ad22988..17e12a26c31 100644
--- a/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.h
+++ b/chromium/third_party/libjingle/source/talk/base/asyncudpsocket.h
@@ -52,8 +52,9 @@ class AsyncUDPSocket : public AsyncPacketSocket {
 
   virtual SocketAddress GetLocalAddress() const;
   virtual SocketAddress GetRemoteAddress() const;
-  virtual int Send(const void *pv, size_t cb);
-  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr);
+  virtual int Send(const void *pv, size_t cb, DiffServCodePoint dscp);
+  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
+                     DiffServCodePoint dscp);
   virtual int Close();
 
   virtual State GetState() const;
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/IceCandidate.java b/chromium/third_party/libjingle/source/talk/base/dscp.h
index b5d2dc9da80..c71ee3e932c 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/IceCandidate.java
+++ b/chromium/third_party/libjingle/source/talk/base/dscp.h
@@ -25,24 +25,38 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
-package org.webrtc;
+#ifndef TALK_BASE_DSCP_H_
+#define TALK_BASE_DSCP_H_
 
-/**
- * Representation of a single ICE Candidate, mirroring
- * {@code IceCandidateInterface} in the C++ API.
- */
-public class IceCandidate {
-  public final String sdpMid;
-  public final int sdpMLineIndex;
-  public final String sdp;
+namespace talk_base {
+// Differentiated Services Code Point.
+// See http://tools.ietf.org/html/rfc2474 for details.
+enum DiffServCodePoint {
+  DSCP_NO_CHANGE = -1,
+  DSCP_DEFAULT = 0,  // Same as DSCP_CS0
+  DSCP_CS0  = 0,   // The default
+  DSCP_CS1  = 8,   // Bulk/background traffic
+  DSCP_AF11 = 10,
+  DSCP_AF12 = 12,
+  DSCP_AF13 = 14,
+  DSCP_CS2  = 16,
+  DSCP_AF21 = 18,
+  DSCP_AF22 = 20,
+  DSCP_AF23 = 22,
+  DSCP_CS3  = 24,
+  DSCP_AF31 = 26,
+  DSCP_AF32 = 28,
+  DSCP_AF33 = 30,
+  DSCP_CS4  = 32,
+  DSCP_AF41 = 34,  // Video
+  DSCP_AF42 = 36,  // Video
+  DSCP_AF43 = 38,  // Video
+  DSCP_CS5  = 40,  // Video
+  DSCP_EF   = 46,  // Voice
+  DSCP_CS6  = 48,  // Voice
+  DSCP_CS7  = 56,  // Control messages
+};
 
-  public IceCandidate(String sdpMid, int sdpMLineIndex, String sdp) {
-    this.sdpMid = sdpMid;
-    this.sdpMLineIndex = sdpMLineIndex;
-    this.sdp = sdp;
-  }
+}  // namespace talk_base
 
-  public String toString() {
-    return sdpMid + ":" + sdpMLineIndex + ":" + sdp;
-  }
-}
+ #endif  // TALK_BASE_DSCP_H_
diff --git a/chromium/third_party/libjingle/source/talk/base/gunit_prod.h b/chromium/third_party/libjingle/source/talk/base/gunit_prod.h
index 6be5c53f970..bf5e88acb4e 100644
--- a/chromium/third_party/libjingle/source/talk/base/gunit_prod.h
+++ b/chromium/third_party/libjingle/source/talk/base/gunit_prod.h
@@ -28,7 +28,11 @@
 #ifndef TALK_BASE_GUNIT_PROD_H_
 #define TALK_BASE_GUNIT_PROD_H_
 
-#if defined(ANDROID) || defined (GTEST_RELATIVE_PATH)
+#if defined(ANDROID)
+// Android doesn't use gtest at all, so anything that relies on gtest should
+// check this define first.
+#define NO_GTEST
+#elif defined (GTEST_RELATIVE_PATH)
 #include "gtest/gtest_prod.h"
 #else
 #include "testing/base/gunit_prod.h"
diff --git a/chromium/third_party/libjingle/source/talk/base/macsocketserver_unittest.cc b/chromium/third_party/libjingle/source/talk/base/macsocketserver_unittest.cc
index 07cce263c0c..a4a71019dd6 100644
--- a/chromium/third_party/libjingle/source/talk/base/macsocketserver_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/base/macsocketserver_unittest.cc
@@ -132,7 +132,8 @@ TEST_F(MacAsyncSocketTest, TestConnectWithDnsLookupIPv6) {
   SocketTest::TestConnectWithDnsLookupIPv6();
 }
 
-TEST_F(MacAsyncSocketTest, TestConnectFailIPv4) {
+// BUG=https://code.google.com/p/webrtc/issues/detail?id=2272
+TEST_F(MacAsyncSocketTest, DISABLED_TestConnectFailIPv4) {
   SocketTest::TestConnectFailIPv4();
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/base/messagehandler.cc b/chromium/third_party/libjingle/source/talk/base/messagehandler.cc
index 5b3585b1626..16f9a21a165 100644
--- a/chromium/third_party/libjingle/source/talk/base/messagehandler.cc
+++ b/chromium/third_party/libjingle/source/talk/base/messagehandler.cc
@@ -2,26 +2,26 @@
  * libjingle
  * Copyright 2004--2005, Google Inc.
  *
- * Redistribution and use in source and binary forms, with or without 
+ * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
- *  1. Redistributions of source code must retain the above copyright notice, 
+ *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products 
+ *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
@@ -31,7 +31,7 @@
 namespace talk_base {
 
 MessageHandler::~MessageHandler() {
-  MessageQueueManager::Instance()->Clear(this);
+  MessageQueueManager::Clear(this);
 }
 
 } // namespace talk_base
diff --git a/chromium/third_party/libjingle/source/talk/base/messagequeue.cc b/chromium/third_party/libjingle/source/talk/base/messagequeue.cc
index 5c406221622..15b700ffe48 100644
--- a/chromium/third_party/libjingle/source/talk/base/messagequeue.cc
+++ b/chromium/third_party/libjingle/source/talk/base/messagequeue.cc
@@ -42,7 +42,7 @@ const uint32 kMaxMsgLatency = 150;  // 150 ms
 //------------------------------------------------------------------
 // MessageQueueManager
 
-MessageQueueManager* MessageQueueManager::instance_;
+MessageQueueManager* MessageQueueManager::instance_ = NULL;
 
 MessageQueueManager* MessageQueueManager::Instance() {
   // Note: This is not thread safe, but it is first called before threads are
@@ -52,6 +52,10 @@ MessageQueueManager* MessageQueueManager::Instance() {
   return instance_;
 }
 
+bool MessageQueueManager::IsInitialized() {
+  return instance_ != NULL;
+}
+
 MessageQueueManager::MessageQueueManager() {
 }
 
@@ -59,6 +63,9 @@ MessageQueueManager::~MessageQueueManager() {
 }
 
 void MessageQueueManager::Add(MessageQueue *message_queue) {
+  return Instance()->AddInternal(message_queue);
+}
+void MessageQueueManager::AddInternal(MessageQueue *message_queue) {
   // MessageQueueManager methods should be non-reentrant, so we
   // ASSERT that is the case.  If any of these ASSERT, please
   // contact bpm or jbeda.
@@ -68,6 +75,12 @@ void MessageQueueManager::Add(MessageQueue *message_queue) {
 }
 
 void MessageQueueManager::Remove(MessageQueue *message_queue) {
+  // If there isn't a message queue manager instance, then there isn't a queue
+  // to remove.
+  if (!instance_) return;
+  return Instance()->RemoveInternal(message_queue);
+}
+void MessageQueueManager::RemoveInternal(MessageQueue *message_queue) {
   ASSERT(!crit_.CurrentThreadIsOwner());  // See note above.
   // If this is the last MessageQueue, destroy the manager as well so that
   // we don't leak this object at program shutdown. As mentioned above, this is
@@ -91,6 +104,12 @@ void MessageQueueManager::Remove(MessageQueue *message_queue) {
 }
 
 void MessageQueueManager::Clear(MessageHandler *handler) {
+  // If there isn't a message queue manager instance, then there aren't any
+  // queues to remove this handler from.
+  if (!instance_) return;
+  return Instance()->ClearInternal(handler);
+}
+void MessageQueueManager::ClearInternal(MessageHandler *handler) {
   ASSERT(!crit_.CurrentThreadIsOwner());  // See note above.
   CritScope cs(&crit_);
   std::vector<MessageQueue *>::iterator iter;
@@ -122,7 +141,7 @@ MessageQueue::~MessageQueue() {
   // is going away.
   SignalQueueDestroyed();
   if (active_) {
-    MessageQueueManager::Instance()->Remove(this);
+    MessageQueueManager::Remove(this);
     Clear(NULL);
   }
   if (ss_) {
@@ -381,7 +400,7 @@ void MessageQueue::EnsureActive() {
   ASSERT(crit_.CurrentThreadIsOwner());
   if (!active_) {
     active_ = true;
-    MessageQueueManager::Instance()->Add(this);
+    MessageQueueManager::Add(this);
   }
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/base/messagequeue.h b/chromium/third_party/libjingle/source/talk/base/messagequeue.h
index 331f2073636..7b38ba0082d 100644
--- a/chromium/third_party/libjingle/source/talk/base/messagequeue.h
+++ b/chromium/third_party/libjingle/source/talk/base/messagequeue.h
@@ -53,16 +53,26 @@ class MessageQueue;
 
 class MessageQueueManager {
  public:
-  static MessageQueueManager* Instance();
+  static void Add(MessageQueue *message_queue);
+  static void Remove(MessageQueue *message_queue);
+  static void Clear(MessageHandler *handler);
 
-  void Add(MessageQueue *message_queue);
-  void Remove(MessageQueue *message_queue);
-  void Clear(MessageHandler *handler);
+  // For testing purposes, we expose whether or not the MessageQueueManager
+  // instance has been initialized. It has no other use relative to the rest of
+  // the functions of this class, which auto-initialize the underlying
+  // MessageQueueManager instance when necessary.
+  static bool IsInitialized();
 
  private:
+  static MessageQueueManager* Instance();
+
   MessageQueueManager();
   ~MessageQueueManager();
 
+  void AddInternal(MessageQueue *message_queue);
+  void RemoveInternal(MessageQueue *message_queue);
+  void ClearInternal(MessageHandler *handler);
+
   static MessageQueueManager* instance_;
   // This list contains 'active' MessageQueues.
   std::vector<MessageQueue *> message_queues_;
diff --git a/chromium/third_party/libjingle/source/talk/base/messagequeue_unittest.cc b/chromium/third_party/libjingle/source/talk/base/messagequeue_unittest.cc
index 8e555482222..9ce524ec6ac 100644
--- a/chromium/third_party/libjingle/source/talk/base/messagequeue_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/base/messagequeue_unittest.cc
@@ -130,3 +130,28 @@ TEST_F(MessageQueueTest, DiposeHandlerWithPostedMessagePending) {
   EXPECT_TRUE(deleted);
 }
 
+struct UnwrapMainThreadScope {
+  UnwrapMainThreadScope() : rewrap_(Thread::Current() != NULL) {
+    if (rewrap_) ThreadManager::Instance()->UnwrapCurrentThread();
+  }
+  ~UnwrapMainThreadScope() {
+    if (rewrap_) ThreadManager::Instance()->WrapCurrentThread();
+  }
+ private:
+  bool rewrap_;
+};
+
+TEST(MessageQueueManager, Clear) {
+  UnwrapMainThreadScope s;
+  if (MessageQueueManager::IsInitialized()) {
+    LOG(LS_INFO) << "Unable to run MessageQueueManager::Clear test, since the "
+                 << "MessageQueueManager was already initialized by some "
+                 << "other test in this run.";
+    return;
+  }
+  bool deleted = false;
+  DeletedMessageHandler* handler = new DeletedMessageHandler(&deleted);
+  delete handler;
+  EXPECT_TRUE(deleted);
+  EXPECT_FALSE(MessageQueueManager::IsInitialized());
+}
diff --git a/chromium/third_party/libjingle/source/talk/base/natserver.cc b/chromium/third_party/libjingle/source/talk/base/natserver.cc
index 7a3a045099e..483542591e4 100644
--- a/chromium/third_party/libjingle/source/talk/base/natserver.cc
+++ b/chromium/third_party/libjingle/source/talk/base/natserver.cc
@@ -126,7 +126,8 @@ void NATServer::OnInternalPacket(
   iter->second->whitelist->insert(dest_addr);
 
   // Send the packet to its intended destination.
-  iter->second->socket->SendTo(buf + length, size - length, dest_addr);
+  iter->second->socket->SendTo(buf + length, size - length, dest_addr,
+                               DSCP_NO_CHANGE);
 }
 
 void NATServer::OnExternalPacket(
@@ -155,7 +156,7 @@ void NATServer::OnExternalPacket(
   // Copy the data part after the address.
   std::memcpy(real_buf.get() + addrlength, buf, size);
   server_socket_->SendTo(real_buf.get(), size + addrlength,
-                         iter->second->route.source());
+                         iter->second->route.source(), DSCP_NO_CHANGE);
 }
 
 void NATServer::Translate(const SocketAddressPair& route) {
diff --git a/chromium/third_party/libjingle/source/talk/base/openssladapter.cc b/chromium/third_party/libjingle/source/talk/base/openssladapter.cc
index 50391e5c215..af92f0c4534 100644
--- a/chromium/third_party/libjingle/source/talk/base/openssladapter.cc
+++ b/chromium/third_party/libjingle/source/talk/base/openssladapter.cc
@@ -233,7 +233,10 @@ VerificationCallback OpenSSLAdapter::custom_verify_callback_ = NULL;
 bool OpenSSLAdapter::InitializeSSL(VerificationCallback callback) {
   if (!InitializeSSLThread() || !SSL_library_init())
       return false;
+#if !defined(ADDRESS_SANITIZER) || !defined(OSX)
+  // Loading the error strings crashes mac_asan.  Omit this debugging aid there.
   SSL_load_error_strings();
+#endif
   ERR_load_BIO_strings();
   OpenSSL_add_all_algorithms();
   RAND_poll();
diff --git a/chromium/third_party/libjingle/source/talk/base/physicalsocketserver.cc b/chromium/third_party/libjingle/source/talk/base/physicalsocketserver.cc
index 8a1bb5c2656..f14c3bd69c6 100644
--- a/chromium/third_party/libjingle/source/talk/base/physicalsocketserver.cc
+++ b/chromium/third_party/libjingle/source/talk/base/physicalsocketserver.cc
@@ -1265,7 +1265,14 @@ void PhysicalSocketServer::Remove(Dispatcher *pdispatcher) {
   DispatcherList::iterator pos = std::find(dispatchers_.begin(),
                                            dispatchers_.end(),
                                            pdispatcher);
-  ASSERT(pos != dispatchers_.end());
+  // We silently ignore duplicate calls to Add, so we should silently ignore
+  // the (expected) symmetric calls to Remove. Note that this may still hide
+  // a real issue, so we at least log a warning about it.
+  if (pos == dispatchers_.end()) {
+    LOG(LS_WARNING) << "PhysicalSocketServer asked to remove a unknown "
+                    << "dispatcher, potentially from a duplicate call to Add.";
+    return;
+  }
   size_t index = pos - dispatchers_.begin();
   dispatchers_.erase(pos);
   for (IteratorList::iterator it = iterators_.begin(); it != iterators_.end();
diff --git a/chromium/third_party/libjingle/source/talk/base/socket.h b/chromium/third_party/libjingle/source/talk/base/socket.h
index 9932cdada49..e738060f89d 100644
--- a/chromium/third_party/libjingle/source/talk/base/socket.h
+++ b/chromium/third_party/libjingle/source/talk/base/socket.h
@@ -184,7 +184,8 @@ class Socket {
     OPT_RCVBUF,      // receive buffer size
     OPT_SNDBUF,      // send buffer size
     OPT_NODELAY,     // whether Nagle algorithm is enabled
-    OPT_IPV6_V6ONLY  // Whether the socket is IPv6 only.
+    OPT_IPV6_V6ONLY, // Whether the socket is IPv6 only.
+    OPT_DSCP         // DSCP code
   };
   virtual int GetOption(Option opt, int* value) = 0;
   virtual int SetOption(Option opt, int value) = 0;
diff --git a/chromium/third_party/libjingle/source/talk/base/testclient.cc b/chromium/third_party/libjingle/source/talk/base/testclient.cc
index 0e7625f7365..0ef85183117 100644
--- a/chromium/third_party/libjingle/source/talk/base/testclient.cc
+++ b/chromium/third_party/libjingle/source/talk/base/testclient.cc
@@ -25,6 +25,7 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+#include "talk/base/dscp.h"
 #include "talk/base/testclient.h"
 #include "talk/base/thread.h"
 #include "talk/base/timeutils.h"
@@ -58,12 +59,12 @@ bool TestClient::CheckConnState(AsyncPacketSocket::State state) {
 }
 
 int TestClient::Send(const char* buf, size_t size) {
-  return socket_->Send(buf, size);
+  return socket_->Send(buf, size, DSCP_NO_CHANGE);
 }
 
 int TestClient::SendTo(const char* buf, size_t size,
                        const SocketAddress& dest) {
-  return socket_->SendTo(buf, size, dest);
+  return socket_->SendTo(buf, size, dest, DSCP_NO_CHANGE);
 }
 
 TestClient::Packet* TestClient::NextPacket() {
diff --git a/chromium/third_party/libjingle/source/talk/base/testechoserver.h b/chromium/third_party/libjingle/source/talk/base/testechoserver.h
index 9bb5178c0ac..10466fa4359 100644
--- a/chromium/third_party/libjingle/source/talk/base/testechoserver.h
+++ b/chromium/third_party/libjingle/source/talk/base/testechoserver.h
@@ -68,7 +68,7 @@ class TestEchoServer : public sigslot::has_slots<> {
   }
   void OnPacket(AsyncPacketSocket* socket, const char* buf, size_t size,
                 const SocketAddress& remote_addr) {
-    socket->Send(buf, size);
+    socket->Send(buf, size, DSCP_NO_CHANGE);
   }
   void OnClose(AsyncPacketSocket* socket, int err) {
     ClientList::iterator it =
diff --git a/chromium/third_party/libjingle/source/talk/base/timeutils_unittest.cc b/chromium/third_party/libjingle/source/talk/base/timeutils_unittest.cc
index c90f6a49f20..0fc5eb19c8a 100644
--- a/chromium/third_party/libjingle/source/talk/base/timeutils_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/base/timeutils_unittest.cc
@@ -143,7 +143,7 @@ TEST(TimeTest, BoundaryComparison) {
   EXPECT_EQ(-100, TimeDiff(ts_earlier, ts_later));
 }
 
-TEST(TimeTest, CurrentTmTime) {
+TEST(TimeTest, DISABLED_CurrentTmTime) {
   struct tm tm;
   int microseconds;
 
diff --git a/chromium/third_party/libjingle/source/talk/base/virtualsocket_unittest.cc b/chromium/third_party/libjingle/source/talk/base/virtualsocket_unittest.cc
index 244568e5589..f3b13fc0f62 100644
--- a/chromium/third_party/libjingle/source/talk/base/virtualsocket_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/base/virtualsocket_unittest.cc
@@ -69,7 +69,7 @@ struct Sender : public MessageHandler {
 
     count += size;
     memcpy(dummy, &cur_time, sizeof(cur_time));
-    socket->Send(dummy, size);
+    socket->Send(dummy, size, DSCP_NO_CHANGE);
 
     last_send = cur_time;
     thread->PostDelayed(NextDelay(), this, 1);
@@ -315,12 +315,12 @@ class VirtualSocketServerTest : public testing::Test {
         EmptySocketAddressWithFamily(initial_addr.family());
 
     // Create client and server
-    AsyncSocket* client = ss_->CreateAsyncSocket(initial_addr.family(),
-                                                 SOCK_STREAM);
-    sink.Monitor(client);
-    AsyncSocket* server = ss_->CreateAsyncSocket(initial_addr.family(),
-                                                 SOCK_STREAM);
-    sink.Monitor(server);
+    scoped_ptr<AsyncSocket> client(ss_->CreateAsyncSocket(initial_addr.family(),
+                                                          SOCK_STREAM));
+    sink.Monitor(client.get());
+    scoped_ptr<AsyncSocket> server(ss_->CreateAsyncSocket(initial_addr.family(),
+                                                          SOCK_STREAM));
+    sink.Monitor(server.get());
 
     // Initiate connect
     EXPECT_EQ(0, server->Bind(initial_addr));
@@ -330,19 +330,17 @@ class VirtualSocketServerTest : public testing::Test {
     EXPECT_EQ(0, client->Connect(server->GetLocalAddress()));
 
     // Server close before socket enters accept queue
-    EXPECT_FALSE(sink.Check(server, testing::SSE_READ));
+    EXPECT_FALSE(sink.Check(server.get(), testing::SSE_READ));
     server->Close();
 
     ss_->ProcessMessagesUntilIdle();
 
     // Result: connection failed
     EXPECT_EQ(client->GetState(), AsyncSocket::CS_CLOSED);
-    EXPECT_TRUE(sink.Check(client, testing::SSE_ERROR));
+    EXPECT_TRUE(sink.Check(client.get(), testing::SSE_ERROR));
 
-    // New server
-    delete server;
-    server = ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM);
-    sink.Monitor(server);
+    server.reset(ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
+    sink.Monitor(server.get());
 
     // Initiate connect
     EXPECT_EQ(0, server->Bind(initial_addr));
@@ -354,19 +352,18 @@ class VirtualSocketServerTest : public testing::Test {
     ss_->ProcessMessagesUntilIdle();
 
     // Server close while socket is in accept queue
-    EXPECT_TRUE(sink.Check(server, testing::SSE_READ));
+    EXPECT_TRUE(sink.Check(server.get(), testing::SSE_READ));
     server->Close();
 
     ss_->ProcessMessagesUntilIdle();
 
     // Result: connection failed
     EXPECT_EQ(client->GetState(), AsyncSocket::CS_CLOSED);
-    EXPECT_TRUE(sink.Check(client, testing::SSE_ERROR));
+    EXPECT_TRUE(sink.Check(client.get(), testing::SSE_ERROR));
 
     // New server
-    delete server;
-    server = ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM);
-    sink.Monitor(server);
+    server.reset(ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
+    sink.Monitor(server.get());
 
     // Initiate connect
     EXPECT_EQ(0, server->Bind(initial_addr));
@@ -378,10 +375,10 @@ class VirtualSocketServerTest : public testing::Test {
     ss_->ProcessMessagesUntilIdle();
 
     // Server accepts connection
-    EXPECT_TRUE(sink.Check(server, testing::SSE_READ));
-    AsyncSocket* accepted = server->Accept(&accept_addr);
-    ASSERT_TRUE(NULL != accepted);
-    sink.Monitor(accepted);
+    EXPECT_TRUE(sink.Check(server.get(), testing::SSE_READ));
+    scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+    ASSERT_TRUE(NULL != accepted.get());
+    sink.Monitor(accepted.get());
 
     // Client closes before connection complets
     EXPECT_EQ(accepted->GetState(), AsyncSocket::CS_CONNECTED);
@@ -394,8 +391,8 @@ class VirtualSocketServerTest : public testing::Test {
 
     // Result: accepted socket closes
     EXPECT_EQ(accepted->GetState(), AsyncSocket::CS_CLOSED);
-    EXPECT_TRUE(sink.Check(accepted, testing::SSE_CLOSE));
-    EXPECT_FALSE(sink.Check(client, testing::SSE_CLOSE));
+    EXPECT_TRUE(sink.Check(accepted.get(), testing::SSE_CLOSE));
+    EXPECT_FALSE(sink.Check(client.get(), testing::SSE_CLOSE));
   }
 
   void CloseTest(const SocketAddress& initial_addr) {
@@ -409,8 +406,9 @@ class VirtualSocketServerTest : public testing::Test {
     EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family());
 
 
-    AsyncSocket* b = ss_->CreateAsyncSocket(initial_addr.family(), SOCK_STREAM);
-    sink.Monitor(b);
+    scoped_ptr<AsyncSocket> b(ss_->CreateAsyncSocket(initial_addr.family(),
+                                                     SOCK_STREAM));
+    sink.Monitor(b.get());
     b->Bind(initial_addr);
     EXPECT_EQ(b->GetLocalAddress().family(), initial_addr.family());
 
@@ -423,7 +421,7 @@ class VirtualSocketServerTest : public testing::Test {
     EXPECT_EQ(a->GetState(), AsyncSocket::CS_CONNECTED);
     EXPECT_EQ(a->GetRemoteAddress(), b->GetLocalAddress());
 
-    EXPECT_TRUE(sink.Check(b, testing::SSE_OPEN));
+    EXPECT_TRUE(sink.Check(b.get(), testing::SSE_OPEN));
     EXPECT_EQ(b->GetState(), AsyncSocket::CS_CONNECTED);
     EXPECT_EQ(b->GetRemoteAddress(), a->GetLocalAddress());
 
@@ -434,14 +432,15 @@ class VirtualSocketServerTest : public testing::Test {
     ss_->ProcessMessagesUntilIdle();
 
     char buffer[10];
-    EXPECT_FALSE(sink.Check(b, testing::SSE_READ));
+    EXPECT_FALSE(sink.Check(b.get(), testing::SSE_READ));
     EXPECT_EQ(-1, b->Recv(buffer, 10));
 
     EXPECT_TRUE(sink.Check(a, testing::SSE_CLOSE));
     EXPECT_EQ(a->GetState(), AsyncSocket::CS_CLOSED);
     EXPECT_EQ(a->GetRemoteAddress(), kEmptyAddr);
 
-    EXPECT_FALSE(sink.Check(b, testing::SSE_CLOSE));  // No signal for Closer
+    // No signal for Closer
+    EXPECT_FALSE(sink.Check(b.get(), testing::SSE_CLOSE));
     EXPECT_EQ(b->GetState(), AsyncSocket::CS_CLOSED);
     EXPECT_EQ(b->GetRemoteAddress(), kEmptyAddr);
   }
diff --git a/chromium/third_party/libjingle/source/talk/build/common.gypi b/chromium/third_party/libjingle/source/talk/build/common.gypi
index a8fcd57aaa7..50acb39ea95 100644
--- a/chromium/third_party/libjingle/source/talk/build/common.gypi
+++ b/chromium/third_party/libjingle/source/talk/build/common.gypi
@@ -2,26 +2,26 @@
 # libjingle
 # Copyright 2012, Google Inc.
 #
-# Redistribution and use in source and binary forms, with or without 
+# Redistribution and use in source and binary forms, with or without
 # modification, are permitted provided that the following conditions are met:
 #
-#  1. Redistributions of source code must retain the above copyright notice, 
+#  1. Redistributions of source code must retain the above copyright notice,
 #     this list of conditions and the following disclaimer.
 #  2. Redistributions in binary form must reproduce the above copyright notice,
 #     this list of conditions and the following disclaimer in the documentation
 #     and/or other materials provided with the distribution.
-#  3. The name of the author may not be used to endorse or promote products 
+#  3. The name of the author may not be used to endorse or promote products
 #     derived from this software without specific prior written permission.
 #
 # THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
-# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 
+# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
 # MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
-# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 
+# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
 # SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
 # PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
 # OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
-# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 
-# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 
+# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
 # ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 #
 
@@ -36,8 +36,6 @@
     # flood of chromium-style warnings.
     'clang_use_chrome_plugins%': 0,
     'libpeer_target_type%': 'static_library',
-    # Whether or not to build the ObjectiveC PeerConnection API & tests.
-    'libjingle_objc%': 0,
     'conditions': [
       ['OS=="android" or OS=="linux"', {
         # TODO(henrike): make sure waterfall bots have $JAVA_HOME configured
@@ -110,7 +108,7 @@
           'HAVE_OPENSSL_SSL_H=1',
         ],
       }],
-      ['libjingle_objc==1', {
+      ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
         'defines': [
           'CARBON_DEPRECATED=YES',
         ],
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/AppRTCClient.java b/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/AppRTCClient.java
deleted file mode 100644
index fe415643738..00000000000
--- a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/AppRTCClient.java
+++ /dev/null
@@ -1,432 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.appspot.apprtc;
-
-import android.app.Activity;
-import android.os.AsyncTask;
-import android.util.Log;
-
-import org.json.JSONArray;
-import org.json.JSONException;
-import org.json.JSONObject;
-import org.webrtc.MediaConstraints;
-import org.webrtc.PeerConnection;
-
-import java.io.IOException;
-import java.io.InputStream;
-import java.net.HttpURLConnection;
-import java.net.URL;
-import java.net.URLConnection;
-import java.util.LinkedList;
-import java.util.List;
-import java.util.Scanner;
-import java.util.regex.Matcher;
-import java.util.regex.Pattern;
-
-/**
- * Negotiates signaling for chatting with apprtc.appspot.com "rooms".
- * Uses the client<->server specifics of the apprtc AppEngine webapp.
- *
- * To use: create an instance of this object (registering a message handler) and
- * call connectToRoom().  Once that's done call sendMessage() and wait for the
- * registered handler to be called with received messages.
- */
-public class AppRTCClient {
-  private static final String TAG = "AppRTCClient";
-  private GAEChannelClient channelClient;
-  private final Activity activity;
-  private final GAEChannelClient.MessageHandler gaeHandler;
-  private final IceServersObserver iceServersObserver;
-
-  // These members are only read/written under sendQueue's lock.
-  private LinkedList<String> sendQueue = new LinkedList<String>();
-  private AppRTCSignalingParameters appRTCSignalingParameters;
-
-  /**
-   * Callback fired once the room's signaling parameters specify the set of
-   * ICE servers to use.
-   */
-  public static interface IceServersObserver {
-    public void onIceServers(List<PeerConnection.IceServer> iceServers);
-  }
-
-  public AppRTCClient(
-      Activity activity, GAEChannelClient.MessageHandler gaeHandler,
-      IceServersObserver iceServersObserver) {
-    this.activity = activity;
-    this.gaeHandler = gaeHandler;
-    this.iceServersObserver = iceServersObserver;
-  }
-
-  /**
-   * Asynchronously connect to an AppRTC room URL, e.g.
-   * https://apprtc.appspot.com/?r=NNN and register message-handling callbacks
-   * on its GAE Channel.
-   */
-  public void connectToRoom(String url) {
-    while (url.indexOf('?') < 0) {
-      // Keep redirecting until we get a room number.
-      (new RedirectResolver()).execute(url);
-      return;  // RedirectResolver above calls us back with the next URL.
-    }
-    (new RoomParameterGetter()).execute(url);
-  }
-
-  /**
-   * Disconnect from the GAE Channel.
-   */
-  public void disconnect() {
-    if (channelClient != null) {
-      channelClient.close();
-      channelClient = null;
-    }
-  }
-
-  /**
-   * Queue a message for sending to the room's channel and send it if already
-   * connected (other wise queued messages are drained when the channel is
-     eventually established).
-   */
-  public synchronized void sendMessage(String msg) {
-    synchronized (sendQueue) {
-      sendQueue.add(msg);
-    }
-    requestQueueDrainInBackground();
-  }
-
-  public boolean isInitiator() {
-    return appRTCSignalingParameters.initiator;
-  }
-
-  public MediaConstraints pcConstraints() {
-    return appRTCSignalingParameters.pcConstraints;
-  }
-
-  public MediaConstraints videoConstraints() {
-    return appRTCSignalingParameters.videoConstraints;
-  }
-
-  // Struct holding the signaling parameters of an AppRTC room.
-  private class AppRTCSignalingParameters {
-    public final List<PeerConnection.IceServer> iceServers;
-    public final String gaeBaseHref;
-    public final String channelToken;
-    public final String postMessageUrl;
-    public final boolean initiator;
-    public final MediaConstraints pcConstraints;
-    public final MediaConstraints videoConstraints;
-
-    public AppRTCSignalingParameters(
-        List<PeerConnection.IceServer> iceServers,
-        String gaeBaseHref, String channelToken, String postMessageUrl,
-        boolean initiator, MediaConstraints pcConstraints,
-        MediaConstraints videoConstraints) {
-      this.iceServers = iceServers;
-      this.gaeBaseHref = gaeBaseHref;
-      this.channelToken = channelToken;
-      this.postMessageUrl = postMessageUrl;
-      this.initiator = initiator;
-      this.pcConstraints = pcConstraints;
-      this.videoConstraints = videoConstraints;
-    }
-  }
-
-  // Load the given URL and return the value of the Location header of the
-  // resulting 302 response.  If the result is not a 302, throws.
-  private class RedirectResolver extends AsyncTask<String, Void, String> {
-    @Override
-    protected String doInBackground(String... urls) {
-      if (urls.length != 1) {
-        throw new RuntimeException("Must be called with a single URL");
-      }
-      try {
-        return followRedirect(urls[0]);
-      } catch (IOException e) {
-        throw new RuntimeException(e);
-      }
-    }
-
-    @Override
-    protected void onPostExecute(String url) {
-      connectToRoom(url);
-    }
-
-    private String followRedirect(String url) throws IOException {
-      HttpURLConnection connection = (HttpURLConnection)
-          new URL(url).openConnection();
-      connection.setInstanceFollowRedirects(false);
-      int code = connection.getResponseCode();
-      if (code != HttpURLConnection.HTTP_MOVED_TEMP) {
-        throw new IOException("Unexpected response: " + code + " for " + url +
-            ", with contents: " + drainStream(connection.getInputStream()));
-      }
-      int n = 0;
-      String name, value;
-      while ((name = connection.getHeaderFieldKey(n)) != null) {
-        value = connection.getHeaderField(n);
-        if (name.equals("Location")) {
-          return value;
-        }
-        ++n;
-      }
-      throw new IOException("Didn't find Location header!");
-    }
-  }
-
-  // AsyncTask that converts an AppRTC room URL into the set of signaling
-  // parameters to use with that room.
-  private class RoomParameterGetter
-      extends AsyncTask<String, Void, AppRTCSignalingParameters> {
-    @Override
-    protected AppRTCSignalingParameters doInBackground(String... urls) {
-      if (urls.length != 1) {
-        throw new RuntimeException("Must be called with a single URL");
-      }
-      try {
-        return getParametersForRoomUrl(urls[0]);
-      } catch (IOException e) {
-        throw new RuntimeException(e);
-      }
-    }
-
-    @Override
-    protected void onPostExecute(AppRTCSignalingParameters params) {
-      channelClient =
-          new GAEChannelClient(activity, params.channelToken, gaeHandler);
-      synchronized (sendQueue) {
-        appRTCSignalingParameters = params;
-      }
-      requestQueueDrainInBackground();
-      iceServersObserver.onIceServers(appRTCSignalingParameters.iceServers);
-    }
-
-    // Fetches |url| and fishes the signaling parameters out of the HTML via
-    // regular expressions.
-    //
-    // TODO(fischman): replace this hackery with a dedicated JSON-serving URL in
-    // apprtc so that this isn't necessary (here and in other future apps that
-    // want to interop with apprtc).
-    private AppRTCSignalingParameters getParametersForRoomUrl(String url)
-        throws IOException {
-      final Pattern fullRoomPattern = Pattern.compile(
-          ".*\n *Sorry, this room is full\\..*");
-
-      String roomHtml =
-          drainStream((new URL(url)).openConnection().getInputStream());
-
-      Matcher fullRoomMatcher = fullRoomPattern.matcher(roomHtml);
-      if (fullRoomMatcher.find()) {
-        throw new IOException("Room is full!");
-      }
-
-      String gaeBaseHref = url.substring(0, url.indexOf('?'));
-      String token = getVarValue(roomHtml, "channelToken", true);
-      String postMessageUrl = "/message?r=" +
-          getVarValue(roomHtml, "roomKey", true) + "&u=" +
-          getVarValue(roomHtml, "me", true);
-      boolean initiator = getVarValue(roomHtml, "initiator", false).equals("1");
-      LinkedList<PeerConnection.IceServer> iceServers =
-          iceServersFromPCConfigJSON(getVarValue(roomHtml, "pcConfig", false));
-
-      boolean isTurnPresent = false;
-      for (PeerConnection.IceServer server : iceServers) {
-        if (server.uri.startsWith("turn:")) {
-          isTurnPresent = true;
-          break;
-        }
-      }
-      if (!isTurnPresent) {
-        iceServers.add(
-            requestTurnServer(getVarValue(roomHtml, "turnUrl", true)));
-      }
-
-      MediaConstraints pcConstraints = constraintsFromJSON(
-          getVarValue(roomHtml, "pcConstraints", false));
-      Log.d(TAG, "pcConstraints: " + pcConstraints);
-
-      MediaConstraints videoConstraints = constraintsFromJSON(
-          getVideoConstraints(
-              getVarValue(roomHtml, "mediaConstraints", false)));
-      Log.d(TAG, "videoConstraints: " + videoConstraints);
-
-      return new AppRTCSignalingParameters(
-          iceServers, gaeBaseHref, token, postMessageUrl, initiator,
-          pcConstraints, videoConstraints);
-    }
-
-    private String getVideoConstraints(String mediaConstraintsString) {
-      try {
-        JSONObject json = new JSONObject(mediaConstraintsString);
-        JSONObject videoJson = json.optJSONObject("video");
-        if (videoJson == null) {
-          return "";
-        }
-        return videoJson.toString();
-      } catch (JSONException e) {
-        throw new RuntimeException(e);
-      }
-    }
-
-    private MediaConstraints constraintsFromJSON(String jsonString) {
-      try {
-        MediaConstraints constraints = new MediaConstraints();
-        JSONObject json = new JSONObject(jsonString);
-        JSONObject mandatoryJSON = json.optJSONObject("mandatory");
-        if (mandatoryJSON != null) {
-          JSONArray mandatoryKeys = mandatoryJSON.names();
-          if (mandatoryKeys != null) {
-            for (int i = 0; i < mandatoryKeys.length(); ++i) {
-              String key = (String) mandatoryKeys.getString(i);
-              String value = mandatoryJSON.getString(key);
-              constraints.mandatory.add(
-                  new MediaConstraints.KeyValuePair(key, value));
-            }
-          }
-        }
-        JSONArray optionalJSON = json.optJSONArray("optional");
-        if (optionalJSON != null) {
-          for (int i = 0; i < optionalJSON.length(); ++i) {
-            JSONObject keyValueDict = optionalJSON.getJSONObject(i);
-            String key = keyValueDict.names().getString(0);
-            String value = keyValueDict.getString(key);
-            constraints.optional.add(
-                new MediaConstraints.KeyValuePair(key, value));
-          }
-        }
-        return constraints;
-      } catch (JSONException e) {
-        throw new RuntimeException(e);
-      }
-    }
-
-    // Scan |roomHtml| for declaration & assignment of |varName| and return its
-    // value, optionally stripping outside quotes if |stripQuotes| requests it.
-    private String getVarValue(
-        String roomHtml, String varName, boolean stripQuotes)
-        throws IOException {
-      final Pattern pattern = Pattern.compile(
-          ".*\n *var " + varName + " = ([^\n]*);\n.*");
-      Matcher matcher = pattern.matcher(roomHtml);
-      if (!matcher.find()) {
-        throw new IOException("Missing " + varName + " in HTML: " + roomHtml);
-      }
-      String varValue = matcher.group(1);
-      if (matcher.find()) {
-        throw new IOException("Too many " + varName + " in HTML: " + roomHtml);
-      }
-      if (stripQuotes) {
-        varValue = varValue.substring(1, varValue.length() - 1);
-      }
-      return varValue;
-    }
-
-    // Requests & returns a TURN ICE Server based on a request URL.  Must be run
-    // off the main thread!
-    private PeerConnection.IceServer requestTurnServer(String url) {
-      try {
-        URLConnection connection = (new URL(url)).openConnection();
-        connection.addRequestProperty("user-agent", "Mozilla/5.0");
-        connection.addRequestProperty("origin", "https://apprtc.appspot.com");
-        String response = drainStream(connection.getInputStream());
-        JSONObject responseJSON = new JSONObject(response);
-        String uri = responseJSON.getJSONArray("uris").getString(0);
-        String username = responseJSON.getString("username");
-        String password = responseJSON.getString("password");
-        return new PeerConnection.IceServer(uri, username, password);
-      } catch (JSONException e) {
-        throw new RuntimeException(e);
-      } catch (IOException e) {
-        throw new RuntimeException(e);
-      }
-    }
-  }
-
-  // Return the list of ICE servers described by a WebRTCPeerConnection
-  // configuration string.
-  private LinkedList<PeerConnection.IceServer> iceServersFromPCConfigJSON(
-      String pcConfig) {
-    try {
-      JSONObject json = new JSONObject(pcConfig);
-      JSONArray servers = json.getJSONArray("iceServers");
-      LinkedList<PeerConnection.IceServer> ret =
-          new LinkedList<PeerConnection.IceServer>();
-      for (int i = 0; i < servers.length(); ++i) {
-        JSONObject server = servers.getJSONObject(i);
-        String url = server.getString("url");
-        String credential =
-            server.has("credential") ? server.getString("credential") : "";
-        ret.add(new PeerConnection.IceServer(url, "", credential));
-      }
-      return ret;
-    } catch (JSONException e) {
-      throw new RuntimeException(e);
-    }
-  }
-
-  // Request an attempt to drain the send queue, on a background thread.
-  private void requestQueueDrainInBackground() {
-    (new AsyncTask<Void, Void, Void>() {
-      public Void doInBackground(Void... unused) {
-        maybeDrainQueue();
-        return null;
-      }
-    }).execute();
-  }
-
-  // Send all queued messages if connected to the room.
-  private void maybeDrainQueue() {
-    synchronized (sendQueue) {
-      if (appRTCSignalingParameters == null) {
-        return;
-      }
-      try {
-        for (String msg : sendQueue) {
-          URLConnection connection = new URL(
-              appRTCSignalingParameters.gaeBaseHref +
-              appRTCSignalingParameters.postMessageUrl).openConnection();
-          connection.setDoOutput(true);
-          connection.getOutputStream().write(msg.getBytes("UTF-8"));
-          if (!connection.getHeaderField(null).startsWith("HTTP/1.1 200 ")) {
-            throw new IOException(
-                "Non-200 response to POST: " + connection.getHeaderField(null) +
-                " for msg: " + msg);
-          }
-        }
-      } catch (IOException e) {
-        throw new RuntimeException(e);
-      }
-      sendQueue.clear();
-    }
-  }
-
-  // Return the contents of an InputStream as a String.
-  private static String drainStream(InputStream in) {
-    Scanner s = new Scanner(in).useDelimiter("\\A");
-    return s.hasNext() ? s.next() : "";
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java b/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
deleted file mode 100644
index 66187538dfd..00000000000
--- a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
+++ /dev/null
@@ -1,518 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.appspot.apprtc;
-
-import android.app.Activity;
-import android.app.AlertDialog;
-import android.content.DialogInterface;
-import android.content.Intent;
-import android.graphics.Point;
-import android.media.AudioManager;
-import android.os.Bundle;
-import android.os.PowerManager;
-import android.util.Log;
-import android.webkit.JavascriptInterface;
-import android.widget.EditText;
-import android.widget.Toast;
-
-import org.json.JSONException;
-import org.json.JSONObject;
-import org.webrtc.DataChannel;
-import org.webrtc.IceCandidate;
-import org.webrtc.Logging;
-import org.webrtc.MediaConstraints;
-import org.webrtc.MediaStream;
-import org.webrtc.PeerConnection;
-import org.webrtc.PeerConnectionFactory;
-import org.webrtc.SdpObserver;
-import org.webrtc.SessionDescription;
-import org.webrtc.StatsObserver;
-import org.webrtc.StatsReport;
-import org.webrtc.VideoCapturer;
-import org.webrtc.VideoRenderer;
-import org.webrtc.VideoRenderer.I420Frame;
-import org.webrtc.VideoSource;
-import org.webrtc.VideoTrack;
-
-import java.util.EnumSet;
-import java.util.LinkedList;
-import java.util.List;
-
-/**
- * Main Activity of the AppRTCDemo Android app demonstrating interoperability
- * between the Android/Java implementation of PeerConnection and the
- * apprtc.appspot.com demo webapp.
- */
-public class AppRTCDemoActivity extends Activity
-    implements AppRTCClient.IceServersObserver {
-  private static final String TAG = "AppRTCDemoActivity";
-  private PeerConnection pc;
-  private final PCObserver pcObserver = new PCObserver();
-  private final SDPObserver sdpObserver = new SDPObserver();
-  private final GAEChannelClient.MessageHandler gaeHandler = new GAEHandler();
-  private AppRTCClient appRtcClient = new AppRTCClient(this, gaeHandler, this);
-  private VideoStreamsView vsv;
-  private Toast logToast;
-  private LinkedList<IceCandidate> queuedRemoteCandidates =
-      new LinkedList<IceCandidate>();
-  // Synchronize on quit[0] to avoid teardown-related crashes.
-  private final Boolean[] quit = new Boolean[] { false };
-  private MediaConstraints sdpMediaConstraints;
-  private PowerManager.WakeLock wakeLock;
-
-  @Override
-  public void onCreate(Bundle savedInstanceState) {
-    super.onCreate(savedInstanceState);
-
-    // Since the error-handling of this demo consists of throwing
-    // RuntimeExceptions and we assume that'll terminate the app, we install
-    // this default handler so it's applied to background threads as well.
-    Thread.setDefaultUncaughtExceptionHandler(
-        new Thread.UncaughtExceptionHandler() {
-          public void uncaughtException(Thread t, Throwable e) {
-            e.printStackTrace();
-            System.exit(-1);
-          }
-        });
-
-    // Uncomment to get ALL WebRTC tracing and SENSITIVE libjingle logging.
-    // Logging.enableTracing(
-    //     "/sdcard/trace.txt",
-    //     EnumSet.of(Logging.TraceLevel.TRACE_ALL),
-    //     Logging.Severity.LS_SENSITIVE);
-
-    PowerManager powerManager = (PowerManager) getSystemService(POWER_SERVICE);
-    wakeLock = powerManager.newWakeLock(
-        PowerManager.SCREEN_BRIGHT_WAKE_LOCK, "AppRTCDemo");
-    wakeLock.acquire();
-
-    Point displaySize = new Point();
-    getWindowManager().getDefaultDisplay().getSize(displaySize);
-    vsv = new VideoStreamsView(this, displaySize);
-    setContentView(vsv);
-
-    abortUnless(PeerConnectionFactory.initializeAndroidGlobals(this),
-        "Failed to initializeAndroidGlobals");
-
-    AudioManager audioManager =
-        ((AudioManager) getSystemService(AUDIO_SERVICE));
-    audioManager.setMode(audioManager.isWiredHeadsetOn() ?
-        AudioManager.MODE_IN_CALL : AudioManager.MODE_IN_COMMUNICATION);
-    audioManager.setSpeakerphoneOn(!audioManager.isWiredHeadsetOn());
-
-    sdpMediaConstraints = new MediaConstraints();
-    sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair(
-        "OfferToReceiveAudio", "true"));
-    sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair(
-        "OfferToReceiveVideo", "true"));
-
-    final Intent intent = getIntent();
-    if ("android.intent.action.VIEW".equals(intent.getAction())) {
-      connectToRoom(intent.getData().toString());
-      return;
-    }
-    showGetRoomUI();
-  }
-
-  private void showGetRoomUI() {
-    final EditText roomInput = new EditText(this);
-    roomInput.setText("https://apprtc.appspot.com/?r=");
-    roomInput.setSelection(roomInput.getText().length());
-    DialogInterface.OnClickListener listener =
-        new DialogInterface.OnClickListener() {
-          @Override public void onClick(DialogInterface dialog, int which) {
-            abortUnless(which == DialogInterface.BUTTON_POSITIVE, "lolwat?");
-            dialog.dismiss();
-            connectToRoom(roomInput.getText().toString());
-          }
-        };
-    AlertDialog.Builder builder = new AlertDialog.Builder(this);
-    builder
-        .setMessage("Enter room URL").setView(roomInput)
-        .setPositiveButton("Go!", listener).show();
-  }
-
-  private void connectToRoom(String roomUrl) {
-    logAndToast("Connecting to room...");
-    appRtcClient.connectToRoom(roomUrl);
-  }
-
-  @Override
-  public void onPause() {
-    super.onPause();
-    vsv.onPause();
-    // TODO(fischman): IWBN to support pause/resume, but the WebRTC codebase
-    // isn't ready for that yet; e.g.
-    // https://code.google.com/p/webrtc/issues/detail?id=1407
-    // Instead, simply exit instead of pausing (the alternative leads to
-    // system-borking with wedged cameras; e.g. b/8224551)
-    disconnectAndExit();
-  }
-
-  @Override
-  public void onResume() {
-    // The onResume() is a lie!  See TODO(fischman) in onPause() above.
-    super.onResume();
-    vsv.onResume();
-  }
-
-  @Override
-  public void onIceServers(List<PeerConnection.IceServer> iceServers) {
-    PeerConnectionFactory factory = new PeerConnectionFactory();
-
-    pc = factory.createPeerConnection(
-        iceServers, appRtcClient.pcConstraints(), pcObserver);
-
-    {
-      final PeerConnection finalPC = pc;
-      final Runnable repeatedStatsLogger = new Runnable() {
-          public void run() {
-            synchronized (quit[0]) {
-              if (quit[0]) {
-                return;
-              }
-              final Runnable runnableThis = this;
-              boolean success = finalPC.getStats(new StatsObserver() {
-                  public void onComplete(StatsReport[] reports) {
-                    for (StatsReport report : reports) {
-                      Log.d(TAG, "Stats: " + report.toString());
-                    }
-                    vsv.postDelayed(runnableThis, 10000);
-                  }
-                }, null);
-              if (!success) {
-                throw new RuntimeException("getStats() return false!");
-              }
-            }
-          }
-        };
-      vsv.postDelayed(repeatedStatsLogger, 10000);
-    }
-
-    {
-      logAndToast("Creating local video source...");
-      VideoCapturer capturer = getVideoCapturer();
-      VideoSource videoSource = factory.createVideoSource(
-          capturer, appRtcClient.videoConstraints());
-      MediaStream lMS = factory.createLocalMediaStream("ARDAMS");
-      VideoTrack videoTrack = factory.createVideoTrack("ARDAMSv0", videoSource);
-      videoTrack.addRenderer(new VideoRenderer(new VideoCallbacks(
-          vsv, VideoStreamsView.Endpoint.LOCAL)));
-      lMS.addTrack(videoTrack);
-      lMS.addTrack(factory.createAudioTrack("ARDAMSa0"));
-      pc.addStream(lMS, new MediaConstraints());
-    }
-    logAndToast("Waiting for ICE candidates...");
-  }
-
-  // Cycle through likely device names for the camera and return the first
-  // capturer that works, or crash if none do.
-  private VideoCapturer getVideoCapturer() {
-    String[] cameraFacing = { "front", "back" };
-    int[] cameraIndex = { 0, 1 };
-    int[] cameraOrientation = { 0, 90, 180, 270 };
-    for (String facing : cameraFacing) {
-      for (int index : cameraIndex) {
-        for (int orientation : cameraOrientation) {
-          String name = "Camera " + index + ", Facing " + facing +
-              ", Orientation " + orientation;
-          VideoCapturer capturer = VideoCapturer.create(name);
-          if (capturer != null) {
-            logAndToast("Using camera: " + name);
-            return capturer;
-          }
-        }
-      }
-    }
-    throw new RuntimeException("Failed to open capturer");
-  }
-
-  @Override
-  public void onDestroy() {
-    super.onDestroy();
-  }
-
-  // Poor-man's assert(): die with |msg| unless |condition| is true.
-  private static void abortUnless(boolean condition, String msg) {
-    if (!condition) {
-      throw new RuntimeException(msg);
-    }
-  }
-
-  // Log |msg| and Toast about it.
-  private void logAndToast(String msg) {
-    Log.d(TAG, msg);
-    if (logToast != null) {
-      logToast.cancel();
-    }
-    logToast = Toast.makeText(this, msg, Toast.LENGTH_SHORT);
-    logToast.show();
-  }
-
-  // Send |json| to the underlying AppEngine Channel.
-  private void sendMessage(JSONObject json) {
-    appRtcClient.sendMessage(json.toString());
-  }
-
-  // Put a |key|->|value| mapping in |json|.
-  private static void jsonPut(JSONObject json, String key, Object value) {
-    try {
-      json.put(key, value);
-    } catch (JSONException e) {
-      throw new RuntimeException(e);
-    }
-  }
-
-  // Implementation detail: observe ICE & stream changes and react accordingly.
-  private class PCObserver implements PeerConnection.Observer {
-    @Override public void onIceCandidate(final IceCandidate candidate){
-      runOnUiThread(new Runnable() {
-          public void run() {
-            JSONObject json = new JSONObject();
-            jsonPut(json, "type", "candidate");
-            jsonPut(json, "label", candidate.sdpMLineIndex);
-            jsonPut(json, "id", candidate.sdpMid);
-            jsonPut(json, "candidate", candidate.sdp);
-            sendMessage(json);
-          }
-        });
-    }
-
-    @Override public void onError(){
-      runOnUiThread(new Runnable() {
-          public void run() {
-            throw new RuntimeException("PeerConnection error!");
-          }
-        });
-    }
-
-    @Override public void onSignalingChange(
-        PeerConnection.SignalingState newState) {
-    }
-
-    @Override public void onIceConnectionChange(
-        PeerConnection.IceConnectionState newState) {
-    }
-
-    @Override public void onIceGatheringChange(
-        PeerConnection.IceGatheringState newState) {
-    }
-
-    @Override public void onAddStream(final MediaStream stream){
-      runOnUiThread(new Runnable() {
-          public void run() {
-            abortUnless(stream.audioTracks.size() == 1 &&
-                stream.videoTracks.size() == 1,
-                "Weird-looking stream: " + stream);
-            stream.videoTracks.get(0).addRenderer(new VideoRenderer(
-                new VideoCallbacks(vsv, VideoStreamsView.Endpoint.REMOTE)));
-          }
-        });
-    }
-
-    @Override public void onRemoveStream(final MediaStream stream){
-      runOnUiThread(new Runnable() {
-          public void run() {
-            stream.videoTracks.get(0).dispose();
-          }
-        });
-    }
-
-    @Override public void onDataChannel(final DataChannel dc) {
-      runOnUiThread(new Runnable() {
-          public void run() {
-            throw new RuntimeException(
-                "AppRTC doesn't use data channels, but got: " + dc.label() +
-                " anyway!");
-          }
-        });
-    }
-  }
-
-  // Implementation detail: handle offer creation/signaling and answer setting,
-  // as well as adding remote ICE candidates once the answer SDP is set.
-  private class SDPObserver implements SdpObserver {
-    @Override public void onCreateSuccess(final SessionDescription sdp) {
-      runOnUiThread(new Runnable() {
-          public void run() {
-            logAndToast("Sending " + sdp.type);
-            JSONObject json = new JSONObject();
-            jsonPut(json, "type", sdp.type.canonicalForm());
-            jsonPut(json, "sdp", sdp.description);
-            sendMessage(json);
-            pc.setLocalDescription(sdpObserver, sdp);
-          }
-        });
-    }
-
-    @Override public void onSetSuccess() {
-      runOnUiThread(new Runnable() {
-          public void run() {
-            if (appRtcClient.isInitiator()) {
-              if (pc.getRemoteDescription() != null) {
-                // We've set our local offer and received & set the remote
-                // answer, so drain candidates.
-                drainRemoteCandidates();
-              }
-            } else {
-              if (pc.getLocalDescription() == null) {
-                // We just set the remote offer, time to create our answer.
-                logAndToast("Creating answer");
-                pc.createAnswer(SDPObserver.this, sdpMediaConstraints);
-              } else {
-                // Sent our answer and set it as local description; drain
-                // candidates.
-                drainRemoteCandidates();
-              }
-            }
-          }
-        });
-    }
-
-    @Override public void onCreateFailure(final String error) {
-      runOnUiThread(new Runnable() {
-          public void run() {
-            throw new RuntimeException("createSDP error: " + error);
-          }
-        });
-    }
-
-    @Override public void onSetFailure(final String error) {
-      runOnUiThread(new Runnable() {
-          public void run() {
-            throw new RuntimeException("setSDP error: " + error);
-          }
-        });
-    }
-
-    private void drainRemoteCandidates() {
-      for (IceCandidate candidate : queuedRemoteCandidates) {
-        pc.addIceCandidate(candidate);
-      }
-      queuedRemoteCandidates = null;
-    }
-  }
-
-  // Implementation detail: handler for receiving GAE messages and dispatching
-  // them appropriately.
-  private class GAEHandler implements GAEChannelClient.MessageHandler {
-    @JavascriptInterface public void onOpen() {
-      if (!appRtcClient.isInitiator()) {
-        return;
-      }
-      logAndToast("Creating offer...");
-      pc.createOffer(sdpObserver, sdpMediaConstraints);
-    }
-
-    @JavascriptInterface public void onMessage(String data) {
-      try {
-        JSONObject json = new JSONObject(data);
-        String type = (String) json.get("type");
-        if (type.equals("candidate")) {
-          IceCandidate candidate = new IceCandidate(
-              (String) json.get("id"),
-              json.getInt("label"),
-              (String) json.get("candidate"));
-          if (queuedRemoteCandidates != null) {
-            queuedRemoteCandidates.add(candidate);
-          } else {
-            pc.addIceCandidate(candidate);
-          }
-        } else if (type.equals("answer") || type.equals("offer")) {
-          SessionDescription sdp = new SessionDescription(
-              SessionDescription.Type.fromCanonicalForm(type),
-              (String) json.get("sdp"));
-          pc.setRemoteDescription(sdpObserver, sdp);
-        } else if (type.equals("bye")) {
-          logAndToast("Remote end hung up; dropping PeerConnection");
-          disconnectAndExit();
-        } else {
-          throw new RuntimeException("Unexpected message: " + data);
-        }
-      } catch (JSONException e) {
-        throw new RuntimeException(e);
-      }
-    }
-
-    @JavascriptInterface public void onClose() {
-      disconnectAndExit();
-    }
-
-    @JavascriptInterface public void onError(int code, String description) {
-      disconnectAndExit();
-    }
-  }
-
-  // Disconnect from remote resources, dispose of local resources, and exit.
-  private void disconnectAndExit() {
-    synchronized (quit[0]) {
-      if (quit[0]) {
-        return;
-      }
-      quit[0] = true;
-      wakeLock.release();
-      if (pc != null) {
-        pc.dispose();
-        pc = null;
-      }
-      if (appRtcClient != null) {
-        appRtcClient.sendMessage("{\"type\": \"bye\"}");
-        appRtcClient.disconnect();
-        appRtcClient = null;
-      }
-      finish();
-    }
-  }
-
-  // Implementation detail: bridge the VideoRenderer.Callbacks interface to the
-  // VideoStreamsView implementation.
-  private class VideoCallbacks implements VideoRenderer.Callbacks {
-    private final VideoStreamsView view;
-    private final VideoStreamsView.Endpoint stream;
-
-    public VideoCallbacks(
-        VideoStreamsView view, VideoStreamsView.Endpoint stream) {
-      this.view = view;
-      this.stream = stream;
-    }
-
-    @Override
-    public void setSize(final int width, final int height) {
-      view.queueEvent(new Runnable() {
-          public void run() {
-            view.setSize(stream, width, height);
-          }
-        });
-    }
-
-    @Override
-    public void renderFrame(I420Frame frame) {
-      view.queueFrame(stream, frame);
-    }
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/FramePool.java b/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/FramePool.java
deleted file mode 100644
index 6f112865053..00000000000
--- a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/FramePool.java
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.appspot.apprtc;
-
-import org.webrtc.VideoRenderer.I420Frame;
-
-import java.util.HashMap;
-import java.util.LinkedList;
-
-/**
- * This class acts as an allocation pool meant to minimize GC churn caused by
- * frame allocation & disposal.  The public API comprises of just two methods:
- * copyFrame(), which allocates as necessary and copies, and
- * returnFrame(), which returns frame ownership to the pool for use by a later
- * call to copyFrame().
- *
- * This class is thread-safe; calls to copyFrame() and returnFrame() are allowed
- * to happen on any thread.
- */
-class FramePool {
-  // Maps each summary code (see summarizeFrameDimensions()) to a list of frames
-  // of that description.
-  private final HashMap<Long, LinkedList<I420Frame>> availableFrames =
-      new HashMap<Long, LinkedList<I420Frame>>();
-  // Every dimension (e.g. width, height, stride) of a frame must be less than
-  // this value.
-  private static final long MAX_DIMENSION = 4096;
-
-  public I420Frame takeFrame(I420Frame source) {
-    long desc = summarizeFrameDimensions(source);
-    I420Frame dst = null;
-    synchronized (availableFrames) {
-      LinkedList<I420Frame> frames = availableFrames.get(desc);
-      if (frames == null) {
-        frames = new LinkedList<I420Frame>();
-        availableFrames.put(desc, frames);
-      }
-      if (!frames.isEmpty()) {
-        dst = frames.pop();
-      } else {
-        dst = new I420Frame(
-            source.width, source.height, source.yuvStrides, null);
-      }
-    }
-    return dst;
-  }
-
-  public void returnFrame(I420Frame frame) {
-    long desc = summarizeFrameDimensions(frame);
-    synchronized (availableFrames) {
-      LinkedList<I420Frame> frames = availableFrames.get(desc);
-      if (frames == null) {
-        throw new IllegalArgumentException("Unexpected frame dimensions");
-      }
-      frames.add(frame);
-    }
-  }
-
-  /** Validate that |frame| can be managed by the pool. */
-  public static boolean validateDimensions(I420Frame frame) {
-    return frame.width < MAX_DIMENSION && frame.height < MAX_DIMENSION &&
-        frame.yuvStrides[0] < MAX_DIMENSION &&
-        frame.yuvStrides[1] < MAX_DIMENSION &&
-        frame.yuvStrides[2] < MAX_DIMENSION;
-  }
-
-  // Return a code summarizing the dimensions of |frame|.  Two frames that
-  // return the same summary are guaranteed to be able to store each others'
-  // contents.  Used like Object.hashCode(), but we need all the bits of a long
-  // to do a good job, and hashCode() returns int, so we do this.
-  private static long summarizeFrameDimensions(I420Frame frame) {
-    long ret = frame.width;
-    ret = ret * MAX_DIMENSION + frame.height;
-    ret = ret * MAX_DIMENSION + frame.yuvStrides[0];
-    ret = ret * MAX_DIMENSION + frame.yuvStrides[1];
-    ret = ret * MAX_DIMENSION + frame.yuvStrides[2];
-    return ret;
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/GAEChannelClient.java b/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/GAEChannelClient.java
deleted file mode 100644
index 29a91134808..00000000000
--- a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/GAEChannelClient.java
+++ /dev/null
@@ -1,173 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.appspot.apprtc;
-
-import android.annotation.SuppressLint;
-import android.app.Activity;
-import android.util.Log;
-import android.webkit.ConsoleMessage;
-import android.webkit.JavascriptInterface;
-import android.webkit.WebChromeClient;
-import android.webkit.WebView;
-import android.webkit.WebViewClient;
-
-/**
- * Java-land version of Google AppEngine's JavaScript Channel API:
- * https://developers.google.com/appengine/docs/python/channel/javascript
- *
- * Requires a hosted HTML page that opens the desired channel and dispatches JS
- * on{Open,Message,Close,Error}() events to a global object named
- * "androidMessageHandler".
- */
-public class GAEChannelClient {
-  private static final String TAG = "GAEChannelClient";
-  private WebView webView;
-  private final ProxyingMessageHandler proxyingMessageHandler;
-
-  /**
-   * Callback interface for messages delivered on the Google AppEngine channel.
-   *
-   * Methods are guaranteed to be invoked on the UI thread of |activity| passed
-   * to GAEChannelClient's constructor.
-   */
-  public interface MessageHandler {
-    public void onOpen();
-    public void onMessage(String data);
-    public void onClose();
-    public void onError(int code, String description);
-  }
-
-  /** Asynchronously open an AppEngine channel. */
-  @SuppressLint("SetJavaScriptEnabled")
-  public GAEChannelClient(
-      Activity activity, String token, MessageHandler handler) {
-    webView = new WebView(activity);
-    webView.getSettings().setJavaScriptEnabled(true);
-    webView.setWebChromeClient(new WebChromeClient() {  // Purely for debugging.
-        public boolean onConsoleMessage (ConsoleMessage msg) {
-          Log.d(TAG, "console: " + msg.message() + " at " +
-              msg.sourceId() + ":" + msg.lineNumber());
-          return false;
-        }
-      });
-    webView.setWebViewClient(new WebViewClient() {  // Purely for debugging.
-        public void onReceivedError(
-            WebView view, int errorCode, String description,
-            String failingUrl) {
-          Log.e(TAG, "JS error: " + errorCode + " in " + failingUrl +
-              ", desc: " + description);
-        }
-      });
-    proxyingMessageHandler =
-        new ProxyingMessageHandler(activity, handler, token);
-    webView.addJavascriptInterface(
-        proxyingMessageHandler, "androidMessageHandler");
-    webView.loadUrl("file:///android_asset/channel.html");
-  }
-
-  /** Close the connection to the AppEngine channel. */
-  public void close() {
-    if (webView == null) {
-      return;
-    }
-    proxyingMessageHandler.disconnect();
-    webView.removeJavascriptInterface("androidMessageHandler");
-    webView.loadUrl("about:blank");
-    webView = null;
-  }
-
-  // Helper class for proxying callbacks from the Java<->JS interaction
-  // (private, background) thread to the Activity's UI thread.
-  private static class ProxyingMessageHandler {
-    private final Activity activity;
-    private final MessageHandler handler;
-    private final boolean[] disconnected = { false };
-    private final String token;
-
-    public
-     ProxyingMessageHandler(Activity activity, MessageHandler handler,
-                            String token) {
-      this.activity = activity;
-      this.handler = handler;
-      this.token = token;
-    }
-
-    public void disconnect() {
-      disconnected[0] = true;
-    }
-
-    private boolean disconnected() {
-      return disconnected[0];
-    }
-
-    @JavascriptInterface public String getToken() {
-      return token;
-    }
-
-    @JavascriptInterface public void onOpen() {
-      activity.runOnUiThread(new Runnable() {
-          public void run() {
-            if (!disconnected()) {
-              handler.onOpen();
-            }
-          }
-        });
-    }
-
-    @JavascriptInterface public void onMessage(final String data) {
-      activity.runOnUiThread(new Runnable() {
-          public void run() {
-            if (!disconnected()) {
-              handler.onMessage(data);
-            }
-          }
-        });
-    }
-
-    @JavascriptInterface public void onClose() {
-      activity.runOnUiThread(new Runnable() {
-          public void run() {
-            if (!disconnected()) {
-              handler.onClose();
-            }
-          }
-        });
-    }
-
-    @JavascriptInterface public void onError(
-        final int code, final String description) {
-      activity.runOnUiThread(new Runnable() {
-          public void run() {
-            if (!disconnected()) {
-              handler.onError(code, description);
-            }
-          }
-        });
-    }
-  }
-}
diff --git a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/VideoStreamsView.java b/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/VideoStreamsView.java
deleted file mode 100644
index 30c9bf8f207..00000000000
--- a/chromium/third_party/libjingle/source/talk/examples/android/src/org/appspot/apprtc/VideoStreamsView.java
+++ /dev/null
@@ -1,323 +0,0 @@
-/*
- * libjingle
- * Copyright 2013, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-package org.appspot.apprtc;
-
-import android.content.Context;
-import android.graphics.Point;
-import android.graphics.Rect;
-import android.opengl.GLES20;
-import android.opengl.GLSurfaceView;
-import android.util.Log;
-
-import org.webrtc.VideoRenderer.I420Frame;
-
-import java.nio.ByteBuffer;
-import java.nio.ByteOrder;
-import java.nio.FloatBuffer;
-import java.util.EnumMap;
-
-import javax.microedition.khronos.egl.EGLConfig;
-import javax.microedition.khronos.opengles.GL10;
-
-/**
- * A GLSurfaceView{,.Renderer} that efficiently renders YUV frames from local &
- * remote VideoTracks using the GPU for CSC.  Clients will want to call the
- * constructor, setSize() and updateFrame() as appropriate, but none of the
- * other public methods of this class are of interest to clients (only to system
- * classes).
- */
-public class VideoStreamsView
-    extends GLSurfaceView
-    implements GLSurfaceView.Renderer {
-
-  /** Identify which of the two video streams is being addressed. */
-  public static enum Endpoint { LOCAL, REMOTE };
-
-  private final static String TAG = "VideoStreamsView";
-  private EnumMap<Endpoint, Rect> rects =
-      new EnumMap<Endpoint, Rect>(Endpoint.class);
-  private Point screenDimensions;
-  // [0] are local Y,U,V, [1] are remote Y,U,V.
-  private int[][] yuvTextures = { { -1, -1, -1}, {-1, -1, -1 }};
-  private int posLocation = -1;
-  private long lastFPSLogTime = System.nanoTime();
-  private long numFramesSinceLastLog = 0;
-  private FramePool framePool = new FramePool();
-  // Accessed on multiple threads!  Must be synchronized.
-  private EnumMap<Endpoint, I420Frame> framesToRender =
-      new EnumMap<Endpoint, I420Frame>(Endpoint.class);
-
-  public VideoStreamsView(Context c, Point screenDimensions) {
-    super(c);
-    this.screenDimensions = screenDimensions;
-    setEGLContextClientVersion(2);
-    setRenderer(this);
-    setRenderMode(RENDERMODE_WHEN_DIRTY);
-  }
-
-  /** Queue |frame| to be uploaded. */
-  public void queueFrame(final Endpoint stream, I420Frame frame) {
-    // Paying for the copy of the YUV data here allows CSC and painting time
-    // to get spent on the render thread instead of the UI thread.
-    abortUnless(framePool.validateDimensions(frame), "Frame too large!");
-    final I420Frame frameCopy = framePool.takeFrame(frame).copyFrom(frame);
-    boolean needToScheduleRender;
-    synchronized (framesToRender) {
-      // A new render needs to be scheduled (via updateFrames()) iff there isn't
-      // already a render scheduled, which is true iff framesToRender is empty.
-      needToScheduleRender = framesToRender.isEmpty();
-      I420Frame frameToDrop = framesToRender.put(stream, frameCopy);
-      if (frameToDrop != null) {
-        framePool.returnFrame(frameToDrop);
-      }
-    }
-    if (needToScheduleRender) {
-      queueEvent(new Runnable() {
-          public void run() {
-            updateFrames();
-          }
-        });
-    }
-  }
-
-  // Upload the planes from |framesToRender| to the textures owned by this View.
-  private void updateFrames() {
-    I420Frame localFrame = null;
-    I420Frame remoteFrame = null;
-    synchronized (framesToRender) {
-      localFrame = framesToRender.remove(Endpoint.LOCAL);
-      remoteFrame = framesToRender.remove(Endpoint.REMOTE);
-    }
-    if (localFrame != null) {
-      texImage2D(localFrame, yuvTextures[0]);
-      framePool.returnFrame(localFrame);
-    }
-    if (remoteFrame != null) {
-      texImage2D(remoteFrame, yuvTextures[1]);
-      framePool.returnFrame(remoteFrame);
-    }
-    abortUnless(localFrame != null || remoteFrame != null,
-                "Nothing to render!");
-    requestRender();
-  }
-
-  /** Inform this View of the dimensions of frames coming from |stream|. */
-  public void setSize(Endpoint stream, int width, int height) {
-    // Generate 3 texture ids for Y/U/V and place them into |textures|,
-    // allocating enough storage for |width|x|height| pixels.
-    int[] textures = yuvTextures[stream == Endpoint.LOCAL ? 0 : 1];
-    GLES20.glGenTextures(3, textures, 0);
-    for (int i = 0; i < 3; ++i) {
-      int w = i == 0 ? width : width / 2;
-      int h = i == 0 ? height : height / 2;
-      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
-      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textures[i]);
-      GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE, w, h, 0,
-          GLES20.GL_LUMINANCE, GLES20.GL_UNSIGNED_BYTE, null);
-      GLES20.glTexParameterf(GLES20.GL_TEXTURE_2D,
-          GLES20.GL_TEXTURE_MIN_FILTER, GLES20.GL_LINEAR);
-      GLES20.glTexParameterf(GLES20.GL_TEXTURE_2D,
-          GLES20.GL_TEXTURE_MAG_FILTER, GLES20.GL_LINEAR);
-      GLES20.glTexParameterf(GLES20.GL_TEXTURE_2D,
-          GLES20.GL_TEXTURE_WRAP_S, GLES20.GL_CLAMP_TO_EDGE);
-      GLES20.glTexParameterf(GLES20.GL_TEXTURE_2D,
-          GLES20.GL_TEXTURE_WRAP_T, GLES20.GL_CLAMP_TO_EDGE);
-    }
-    checkNoGLES2Error();
-  }
-
-  @Override
-  protected void onMeasure(int unusedX, int unusedY) {
-    // Go big or go home!
-    setMeasuredDimension(screenDimensions.x, screenDimensions.y);
-  }
-
-  @Override
-  public void onSurfaceChanged(GL10 unused, int width, int height) {
-    GLES20.glViewport(0, 0, width, height);
-    checkNoGLES2Error();
-  }
-
-  @Override
-  public void onDrawFrame(GL10 unused) {
-    GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
-    drawRectangle(yuvTextures[1], remoteVertices);
-    drawRectangle(yuvTextures[0], localVertices);
-    ++numFramesSinceLastLog;
-    long now = System.nanoTime();
-    if (lastFPSLogTime == -1 || now - lastFPSLogTime > 1e9) {
-      double fps = numFramesSinceLastLog / ((now - lastFPSLogTime) / 1e9);
-      Log.d(TAG, "Rendered FPS: " + fps);
-      lastFPSLogTime = now;
-      numFramesSinceLastLog = 1;
-    }
-    checkNoGLES2Error();
-  }
-
-  @Override
-  public void onSurfaceCreated(GL10 unused, EGLConfig config) {
-    int program = GLES20.glCreateProgram();
-    addShaderTo(GLES20.GL_VERTEX_SHADER, VERTEX_SHADER_STRING, program);
-    addShaderTo(GLES20.GL_FRAGMENT_SHADER, FRAGMENT_SHADER_STRING, program);
-
-    GLES20.glLinkProgram(program);
-    int[] result = new int[] { GLES20.GL_FALSE };
-    result[0] = GLES20.GL_FALSE;
-    GLES20.glGetProgramiv(program, GLES20.GL_LINK_STATUS, result, 0);
-    abortUnless(result[0] == GLES20.GL_TRUE,
-        GLES20.glGetProgramInfoLog(program));
-    GLES20.glUseProgram(program);
-
-    GLES20.glUniform1i(GLES20.glGetUniformLocation(program, "y_tex"), 0);
-    GLES20.glUniform1i(GLES20.glGetUniformLocation(program, "u_tex"), 1);
-    GLES20.glUniform1i(GLES20.glGetUniformLocation(program, "v_tex"), 2);
-
-    // Actually set in drawRectangle(), but queried only once here.
-    posLocation = GLES20.glGetAttribLocation(program, "in_pos");
-
-    int tcLocation = GLES20.glGetAttribLocation(program, "in_tc");
-    GLES20.glEnableVertexAttribArray(tcLocation);
-    GLES20.glVertexAttribPointer(
-        tcLocation, 2, GLES20.GL_FLOAT, false, 0, textureCoords);
-
-    GLES20.glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
-    checkNoGLES2Error();
-  }
-
-  // Wrap a float[] in a direct FloatBuffer using native byte order.
-  private static FloatBuffer directNativeFloatBuffer(float[] array) {
-    FloatBuffer buffer = ByteBuffer.allocateDirect(array.length * 4).order(
-        ByteOrder.nativeOrder()).asFloatBuffer();
-    buffer.put(array);
-    buffer.flip();
-    return buffer;
-  }
-
-  // Upload the YUV planes from |frame| to |textures|.
-  private void texImage2D(I420Frame frame, int[] textures) {
-    for (int i = 0; i < 3; ++i) {
-      ByteBuffer plane = frame.yuvPlanes[i];
-      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
-      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textures[i]);
-      int w = i == 0 ? frame.width : frame.width / 2;
-      int h = i == 0 ? frame.height : frame.height / 2;
-      abortUnless(w == frame.yuvStrides[i], frame.yuvStrides[i] + "!=" + w);
-      GLES20.glTexImage2D(
-          GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE, w, h, 0,
-          GLES20.GL_LUMINANCE, GLES20.GL_UNSIGNED_BYTE, plane);
-    }
-    checkNoGLES2Error();
-  }
-
-  // Draw |textures| using |vertices| (X,Y coordinates).
-  private void drawRectangle(int[] textures, FloatBuffer vertices) {
-    for (int i = 0; i < 3; ++i) {
-      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
-      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textures[i]);
-    }
-
-    GLES20.glVertexAttribPointer(
-        posLocation, 2, GLES20.GL_FLOAT, false, 0, vertices);
-    GLES20.glEnableVertexAttribArray(posLocation);
-
-    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
-    checkNoGLES2Error();
-  }
-
-  // Compile & attach a |type| shader specified by |source| to |program|.
-  private static void addShaderTo(
-      int type, String source, int program) {
-    int[] result = new int[] { GLES20.GL_FALSE };
-    int shader = GLES20.glCreateShader(type);
-    GLES20.glShaderSource(shader, source);
-    GLES20.glCompileShader(shader);
-    GLES20.glGetShaderiv(shader, GLES20.GL_COMPILE_STATUS, result, 0);
-    abortUnless(result[0] == GLES20.GL_TRUE,
-        GLES20.glGetShaderInfoLog(shader) + ", source: " + source);
-    GLES20.glAttachShader(program, shader);
-    GLES20.glDeleteShader(shader);
-    checkNoGLES2Error();
-  }
-
-  // Poor-man's assert(): die with |msg| unless |condition| is true.
-  private static void abortUnless(boolean condition, String msg) {
-    if (!condition) {
-      throw new RuntimeException(msg);
-    }
-  }
-
-  // Assert that no OpenGL ES 2.0 error has been raised.
-  private static void checkNoGLES2Error() {
-    int error = GLES20.glGetError();
-    abortUnless(error == GLES20.GL_NO_ERROR, "GLES20 error: " + error);
-  }
-
-  // Remote image should span the full screen.
-  private static final FloatBuffer remoteVertices = directNativeFloatBuffer(
-      new float[] { -1, 1, -1, -1, 1, 1, 1, -1 });
-
-  // Local image should be thumbnailish.
-  private static final FloatBuffer localVertices = directNativeFloatBuffer(
-      new float[] { 0.6f, 0.9f, 0.6f, 0.6f, 0.9f, 0.9f, 0.9f, 0.6f });
-
-  // Texture Coordinates mapping the entire texture.
-  private static final FloatBuffer textureCoords = directNativeFloatBuffer(
-      new float[] { 0, 0, 0, 1, 1, 0, 1, 1 });
-
-  // Pass-through vertex shader.
-  private static final String VERTEX_SHADER_STRING =
-      "varying vec2 interp_tc;\n" +
-      "\n" +
-      "attribute vec4 in_pos;\n" +
-      "attribute vec2 in_tc;\n" +
-      "\n" +
-      "void main() {\n" +
-      "  gl_Position = in_pos;\n" +
-      "  interp_tc = in_tc;\n" +
-      "}\n";
-
-  // YUV to RGB pixel shader. Loads a pixel from each plane and pass through the
-  // matrix.
-  private static final String FRAGMENT_SHADER_STRING =
-      "precision mediump float;\n" +
-      "varying vec2 interp_tc;\n" +
-      "\n" +
-      "uniform sampler2D y_tex;\n" +
-      "uniform sampler2D u_tex;\n" +
-      "uniform sampler2D v_tex;\n" +
-      "\n" +
-      "void main() {\n" +
-      "  float y = texture2D(y_tex, interp_tc).r;\n" +
-      "  float u = texture2D(u_tex, interp_tc).r - .5;\n" +
-      "  float v = texture2D(v_tex, interp_tc).r - .5;\n" +
-      // CSC according to http://www.fourcc.org/fccyvrgb.php
-      "  gl_FragColor = vec4(y + 1.403 * v, " +
-      "                      y - 0.344 * u - 0.714 * v, " +
-      "                      y + 1.77 * u, 1);\n" +
-      "}\n";
-}
diff --git a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppClient.m b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppClient.m
index 93f0d36a316..99f5166669b 100644
--- a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppClient.m
+++ b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppClient.m
@@ -34,7 +34,7 @@
 
 @interface APPRTCAppClient ()
 
-@property(nonatomic, assign) dispatch_queue_t backgroundQueue;
+@property(nonatomic, strong) dispatch_queue_t backgroundQueue;
 @property(nonatomic, copy) NSString *baseURL;
 @property(nonatomic, strong) GAEChannelClient *gaeChannel;
 @property(nonatomic, copy) NSString *postMessageUrl;
@@ -52,7 +52,6 @@
 - (id)init {
   if (self = [super init]) {
     _backgroundQueue = dispatch_queue_create("RTCBackgroundQueue", NULL);
-    dispatch_retain(_backgroundQueue);
     _sendQueue = [NSMutableArray array];
     // Uncomment to see Request/Response logging.
     // _verboseLogging = YES;
@@ -60,10 +59,6 @@
   return self;
 }
 
-- (void)dealloc {
-  dispatch_release(_backgroundQueue);
-}
-
 #pragma mark - Public methods
 
 - (void)connectToRoom:(NSURL *)url {
@@ -199,14 +194,17 @@
                                                              error:&error];
       NSAssert(!error, @"Unable to parse.  %@", error.localizedDescription);
       NSString *username = json[@"username"];
-      NSString *turnServer = json[@"turn"];
       NSString *password = json[@"password"];
-      NSString *fullUrl =
-          [NSString stringWithFormat:@"turn:%@@%@", username, turnServer];
-      RTCICEServer *ICEServer =
-          [[RTCICEServer alloc] initWithURI:[NSURL URLWithString:fullUrl]
+      NSArray* uris = json[@"uris"];
+      for (int i = 0; i < [uris count]; ++i) {
+        NSString *turnServer = [uris objectAtIndex:i];
+        RTCICEServer *ICEServer =
+          [[RTCICEServer alloc] initWithURI:[NSURL URLWithString:turnServer]
+                                   username:username
                                    password:password];
-      [ICEServers addObject:ICEServer];
+        NSLog(@"Added ICE Server: %@", ICEServer);
+        [ICEServers addObject:ICEServer];
+      }
     } else {
       NSLog(@"Unable to get TURN server.  Error: %@", error.description);
     }
@@ -246,9 +244,10 @@
     [NSRegularExpression regularExpressionWithPattern:@"room is full"
                                               options:0
                                                 error:nil];
-  if ([fullRegex numberOfMatchesInString:self.roomHtml
-                                 options:0
-                                   range:NSMakeRange(0, [self.roomHtml length])]) {
+  if ([fullRegex
+          numberOfMatchesInString:self.roomHtml
+                          options:0
+                            range:NSMakeRange(0, [self.roomHtml length])]) {
     [self showMessage:@"Room full"];
     return;
   }
@@ -257,7 +256,8 @@
   NSString *fullUrl = [[[connection originalRequest] URL] absoluteString];
   NSRange queryRange = [fullUrl rangeOfString:@"?"];
   self.baseURL = [fullUrl substringToIndex:queryRange.location];
-  [self maybeLogMessage:[NSString stringWithFormat:@"Base URL: %@", self.baseURL]];
+  [self maybeLogMessage:
+      [NSString stringWithFormat:@"Base URL: %@", self.baseURL]];
 
   self.token = [self findVar:@"channelToken" strippingQuotes:YES];
   if (!self.token)
@@ -291,11 +291,15 @@
   NSDictionary *json =
       [NSJSONSerialization JSONObjectWithData:pcData options:0 error:&error];
   NSAssert(!error, @"Unable to parse.  %@", error.localizedDescription);
-  NSArray *servers = [json objectForKey:@"ICEServers"];
+  NSArray *servers = [json objectForKey:@"iceServers"];
   NSMutableArray *ICEServers = [NSMutableArray array];
   for (NSDictionary *server in servers) {
     NSString *url = [server objectForKey:@"url"];
+    NSString *username = json[@"username"];
     NSString *credential = [server objectForKey:@"credential"];
+    if (!username) {
+      username = @"";
+    }
     if (!credential) {
       credential = @"";
     }
@@ -305,7 +309,9 @@
                 credential]];
     RTCICEServer *ICEServer =
         [[RTCICEServer alloc] initWithURI:[NSURL URLWithString:url]
+                                 username:username
                                  password:credential];
+    NSLog(@"Added ICE Server: %@", ICEServer);
     [ICEServers addObject:ICEServer];
   }
   [self updateICEServers:ICEServers withTurnServer:turnServerUrl];
diff --git a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.h b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.h
index ad1c5127600..22754e3ad26 100644
--- a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.h
+++ b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.h
@@ -35,7 +35,8 @@
 @protocol APPRTCSendMessage<NSObject>
 
 - (void)sendData:(NSData *)data;
-
+// Logging helper.
+- (void)displayLogMessage:(NSString *)message;
 @end
 
 @class APPRTCViewController;
diff --git a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m
index 710f4ad5eca..34aa7520cbf 100644
--- a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m
+++ b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCAppDelegate.m
@@ -62,7 +62,7 @@
 
 - (void)peerConnection:(RTCPeerConnection *)peerConnection
     signalingStateChanged:(RTCSignalingState)stateChanged {
-  NSLog(@"PCO onSignalingStateChange.");
+  NSLog(@"PCO onSignalingStateChange: %d", stateChanged);
 }
 
 - (void)peerConnection:(RTCPeerConnection *)peerConnection
@@ -119,6 +119,13 @@
 - (void)peerConnection:(RTCPeerConnection *)peerConnection
     iceConnectionChanged:(RTCICEConnectionState)newState {
   NSLog(@"PCO onIceConnectionChange. %d", newState);
+  if (newState == RTCICEConnectionConnected)
+    [self displayLogMessage:@"ICE Connection Connected."];
+  NSAssert(newState != RTCICEConnectionFailed, @"ICE Connection failed!");
+}
+
+- (void)displayLogMessage:(NSString *)message {
+  [_delegate displayLogMessage:message];
 }
 
 @end
@@ -258,8 +265,8 @@
   } else if (([value compare:@"offer"] == NSOrderedSame) ||
              ([value compare:@"answer"] == NSOrderedSame)) {
     NSString *sdpString = [objects objectForKey:@"sdp"];
-    RTCSessionDescription *sdp =
-        [[RTCSessionDescription alloc] initWithType:value sdp:sdpString];
+    RTCSessionDescription *sdp = [[RTCSessionDescription alloc]
+        initWithType:value sdp:[APPRTCAppDelegate preferISAC:sdpString]];
     [self.peerConnection setRemoteDescriptionWithDelegate:self
                                        sessionDescription:sdp];
     [self displayLogMessage:@"PC - setRemoteDescription."];
@@ -283,8 +290,71 @@
 
 #pragma mark - RTCSessionDescriptonDelegate methods
 
+// Match |pattern| to |string| and return the first group of the first
+// match, or nil if no match was found.
++ (NSString *)firstMatch:(NSRegularExpression *)pattern
+              withString:(NSString *)string {
+  NSTextCheckingResult* result =
+    [pattern firstMatchInString:string
+                        options:0
+                          range:NSMakeRange(0, [string length])];
+  if (!result)
+    return nil;
+  return [string substringWithRange:[result rangeAtIndex:1]];
+}
+
+// Mangle |origSDP| to prefer the ISAC/16k audio codec.
++ (NSString *)preferISAC:(NSString *)origSDP {
+  int mLineIndex = -1;
+  NSString* isac16kRtpMap = nil;
+  NSArray* lines = [origSDP componentsSeparatedByString:@"\n"];
+  NSRegularExpression* isac16kRegex = [NSRegularExpression
+      regularExpressionWithPattern:@"^a=rtpmap:(\\d+) ISAC/16000[\r]?$"
+                           options:0
+                             error:nil];
+  for (int i = 0;
+       (i < [lines count]) && (mLineIndex == -1 || isac16kRtpMap == nil);
+       ++i) {
+    NSString* line = [lines objectAtIndex:i];
+    if ([line hasPrefix:@"m=audio "]) {
+      mLineIndex = i;
+      continue;
+    }
+    isac16kRtpMap = [self firstMatch:isac16kRegex withString:line];
+  }
+  if (mLineIndex == -1) {
+    NSLog(@"No m=audio line, so can't prefer iSAC");
+    return origSDP;
+  }
+  if (isac16kRtpMap == nil) {
+    NSLog(@"No ISAC/16000 line, so can't prefer iSAC");
+    return origSDP;
+  }
+  NSArray* origMLineParts =
+      [[lines objectAtIndex:mLineIndex] componentsSeparatedByString:@" "];
+  NSMutableArray* newMLine =
+      [NSMutableArray arrayWithCapacity:[origMLineParts count]];
+  int origPartIndex = 0;
+  // Format is: m=<media> <port> <proto> <fmt> ...
+  [newMLine addObject:[origMLineParts objectAtIndex:origPartIndex++]];
+  [newMLine addObject:[origMLineParts objectAtIndex:origPartIndex++]];
+  [newMLine addObject:[origMLineParts objectAtIndex:origPartIndex++]];
+  [newMLine addObject:isac16kRtpMap];
+  for (; origPartIndex < [origMLineParts count]; ++origPartIndex) {
+    if ([isac16kRtpMap compare:[origMLineParts objectAtIndex:origPartIndex]]
+        != NSOrderedSame) {
+      [newMLine addObject:[origMLineParts objectAtIndex:origPartIndex]];
+    }
+  }
+  NSMutableArray* newLines = [NSMutableArray arrayWithCapacity:[lines count]];
+  [newLines addObjectsFromArray:lines];
+  [newLines replaceObjectAtIndex:mLineIndex
+                      withObject:[newMLine componentsJoinedByString:@" "]];
+  return [newLines componentsJoinedByString:@"\n"];
+}
+
 - (void)peerConnection:(RTCPeerConnection *)peerConnection
-    didCreateSessionDescription:(RTCSessionDescription *)sdp
+    didCreateSessionDescription:(RTCSessionDescription *)origSdp
                           error:(NSError *)error {
   if (error) {
     [self displayLogMessage:@"SDP onFailure."];
@@ -293,6 +363,10 @@
   }
 
   [self displayLogMessage:@"SDP onSuccess(SDP) - set local description."];
+  RTCSessionDescription* sdp =
+      [[RTCSessionDescription alloc]
+          initWithType:origSdp.type
+                   sdp:[APPRTCAppDelegate preferISAC:origSdp.description]];
   [self.peerConnection setLocalDescriptionWithDelegate:self
                                     sessionDescription:sdp];
   [self displayLogMessage:@"PC setLocalDescription."];
diff --git a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCViewController.m b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCViewController.m
index 928686b6853..ab84c0932b4 100644
--- a/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCViewController.m
+++ b/chromium/third_party/libjingle/source/talk/examples/ios/AppRTCDemo/APPRTCViewController.m
@@ -36,6 +36,7 @@
 - (void)viewDidLoad {
   [super viewDidLoad];
   self.textField.delegate = self;
+  [self.textField becomeFirstResponder];
 }
 
 - (void)displayText:(NSString *)text {
diff --git a/chromium/third_party/libjingle/source/talk/libjingle.gyp b/chromium/third_party/libjingle/source/talk/libjingle.gyp
index 15d92d10d98..5250eec1b8f 100755
--- a/chromium/third_party/libjingle/source/talk/libjingle.gyp
+++ b/chromium/third_party/libjingle/source/talk/libjingle.gyp
@@ -42,7 +42,6 @@
        }],
      ],
     }],
-
     ['OS=="linux" or OS=="android"', {
       'targets': [
         {
@@ -107,6 +106,7 @@
                 # and include it here.
                 'android_java_files': [
                   '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRTCAudioDevice.java',
+                  '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/AudioManagerAndroid.java',
                   '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/CaptureCapabilityAndroid.java',
                   '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java',
                   '<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java',
@@ -150,7 +150,7 @@
         },
       ],
     }],
-    ['libjingle_objc == 1', {
+    ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
       'targets': [
         {
           'target_name': 'libjingle_peerconnection_objc',
@@ -229,13 +229,14 @@
           'link_settings': {
             'libraries': [
               '$(SDKROOT)/System/Library/Frameworks/Foundation.framework',
+              '-lstdc++',
             ],
           },
           'xcode_settings': {
             'CLANG_ENABLE_OBJC_ARC': 'YES',
           },
-        }
-      ]
+        },  # target libjingle_peerconnection_objc
+      ],
     }],
   ],
 
@@ -591,17 +592,6 @@
           ],
         }],
         ['OS=="mac"', {
-          'conditions': [
-            ['libjingle_objc != 1', {
-              'link_settings' :{
-                'xcode_settings': {
-                  'OTHER_LDFLAGS': [
-                    '-framework Carbon',
-                  ],
-                },
-              },
-            }],
-          ],
           'sources': [
             'base/macasyncsocket.cc',
             'base/macasyncsocket.h',
@@ -619,18 +609,36 @@
           ],
           'link_settings': {
             'libraries': [
-             '$(SDKROOT)/usr/lib/libcrypto.dylib',
-             '$(SDKROOT)/usr/lib/libssl.dylib',
+              '$(SDKROOT)/usr/lib/libcrypto.dylib',
+              '$(SDKROOT)/usr/lib/libssl.dylib',
             ],
-            'xcode_settings': {
-              'OTHER_LDFLAGS': [
-                '-framework Cocoa',
-                '-framework IOKit',
-                '-framework Security',
-                '-framework SystemConfiguration',
-              ],
+          },
+          'all_dependent_settings': {
+            'link_settings': {
+              'xcode_settings': {
+                'OTHER_LDFLAGS': [
+                  '-framework Cocoa',
+                  '-framework Foundation',
+                  '-framework IOKit',
+                  '-framework Security',
+                  '-framework SystemConfiguration',
+                ],
+              },
             },
           },
+          'conditions': [
+            ['target_arch=="ia32"', {
+              'all_dependent_settings': {
+                'link_settings': {
+                  'xcode_settings': {
+                    'OTHER_LDFLAGS': [
+                      '-framework Carbon',
+                    ],
+                  },
+                },
+              },
+            }],
+          ],
         }],
         ['OS=="ios"', {
           'sources': [
@@ -639,13 +647,16 @@
           'dependencies': [
             '../net/third_party/nss/ssl.gyp:libssl',
           ],
-          'xcode_settings': {
-            'OTHER_LDFLAGS': [
-              '-framework IOKit',
-              '-framework Security',
-              '-framework SystemConfiguration',
-              '-framework UIKit',
-            ],
+          'all_dependent_settings': {
+            'xcode_settings': {
+              'OTHER_LDFLAGS': [
+                '-framework Foundation',
+                '-framework IOKit',
+                '-framework Security',
+                '-framework SystemConfiguration',
+                '-framework UIKit',
+              ],
+            },
           },
         }],
         ['OS=="win"', {
@@ -903,12 +914,18 @@
             'media/devices/macdevicemanagermm.mm',
           ],
           'conditions': [
-            # TODO(hughv):  Investigate if this is needed.
-            [ 'libjingle_objc != 1', {
+            ['target_arch=="ia32"', {
               'sources': [
                 'media/devices/carbonvideorenderer.cc',
                 'media/devices/carbonvideorenderer.h',
               ],
+              'link_settings': {
+                'xcode_settings': {
+                  'OTHER_LDFLAGS': [
+                    '-framework Carbon',
+                  ],
+                },
+              },
             }],
           ],
           'xcode_settings': {
@@ -1032,6 +1049,7 @@
         'p2p/base/transportchannelimpl.h',
         'p2p/base/transportchannelproxy.cc',
         'p2p/base/transportchannelproxy.h',
+        'p2p/base/transportdescription.cc',
         'p2p/base/transportdescription.h',
         'p2p/base/transportdescriptionfactory.cc',
         'p2p/base/transportdescriptionfactory.h',
@@ -1119,8 +1137,6 @@
         'app/webrtc/jsepsessiondescription.h',
         'app/webrtc/localaudiosource.cc',
         'app/webrtc/localaudiosource.h',
-        'app/webrtc/localvideosource.cc',
-        'app/webrtc/localvideosource.h',
         'app/webrtc/mediaconstraintsinterface.cc',
         'app/webrtc/mediaconstraintsinterface.h',
         'app/webrtc/mediastream.cc',
@@ -1144,10 +1160,14 @@
         'app/webrtc/portallocatorfactory.cc',
         'app/webrtc/portallocatorfactory.h',
         'app/webrtc/proxy.h',
+        'app/webrtc/remotevideocapturer.cc',
+        'app/webrtc/remotevideocapturer.h',
         'app/webrtc/statscollector.cc',
         'app/webrtc/statscollector.h',
         'app/webrtc/statstypes.h',
         'app/webrtc/streamcollection.h',
+        'app/webrtc/videosource.cc',
+        'app/webrtc/videosource.h',
         'app/webrtc/videosourceinterface.h',
         'app/webrtc/videosourceproxy.h',
         'app/webrtc/videotrack.cc',
diff --git a/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp b/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp
index 78bfe654f13..d4aa4a42712 100755
--- a/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp
+++ b/chromium/third_party/libjingle/source/talk/libjingle_examples.gyp
@@ -218,7 +218,7 @@
       ], # targets
     }],  # OS=="linux" or OS=="win"
 
-    ['libjingle_objc==1 and OS=="ios"', {
+    ['OS=="ios"', {
       'targets': [
         {
           'target_name': 'AppRTCDemo',
@@ -276,17 +276,23 @@
                 # we could pick more intelligently among the keys, but as a
                 # first cut just tell the developer to specify a key identity
                 # explicitly.
-                'ensure_single_key': '<!(python -c "assert len(\'\'\'<(key_id)\'\'\') > 0 and \'\\n\' not in \'\'\'<(key_id)\'\'\', \'key_id gyp variable needs to be set explicitly because there are multiple codesigning keys, or none!\'")',
+                'ensure_single_key': '<!(python -c "assert \'\\n\' not in \'\'\'<(key_id)\'\'\', \'key_id gyp variable needs to be set explicitly because there are multiple codesigning keys!\'")',
               },
-              'action': [
-                '/usr/bin/codesign', '-v', '--force', '--sign', '<(key_id)',
-                '${BUILT_PRODUCTS_DIR}/AppRTCDemo.app',
+              'conditions': [
+                ['key_id==""', {
+                  'action': [ 'echo', 'Skipping signing' ],
+                }, {
+                  'action': [
+                    '/usr/bin/codesign', '-v', '--force', '--sign', '<(key_id)',
+                    '${BUILT_PRODUCTS_DIR}/AppRTCDemo.app',
+                  ],
+                }],
               ],
             },
           ],
         },  # target AppRTCDemo
       ],  # targets
-    }],  # libjingle_objc==1
+    }],  # OS=="ios"
 
     ['OS=="android"', {
       'targets': [
diff --git a/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp b/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp
index 9ea81afd06f..546767d1fb5 100755
--- a/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp
+++ b/chromium/third_party/libjingle/source/talk/libjingle_tests.gyp
@@ -379,7 +379,6 @@
         'app/webrtc/dtmfsender_unittest.cc',
         'app/webrtc/jsepsessiondescription_unittest.cc',
         'app/webrtc/localaudiosource_unittest.cc',
-        'app/webrtc/localvideosource_unittest.cc',
         # 'app/webrtc/mediastream_unittest.cc',
         # 'app/webrtc/mediastreamhandler_unittest.cc',
         'app/webrtc/mediastreamsignaling_unittest.cc',
@@ -387,6 +386,7 @@
         'app/webrtc/peerconnectionfactory_unittest.cc',
         'app/webrtc/peerconnectioninterface_unittest.cc',
         # 'app/webrtc/peerconnectionproxy_unittest.cc',
+        'app/webrtc/remotevideocapturer_unittest.cc',
         'app/webrtc/test/fakeaudiocapturemodule.cc',
         'app/webrtc/test/fakeaudiocapturemodule.h',
         'app/webrtc/test/fakeaudiocapturemodule_unittest.cc',
@@ -397,6 +397,7 @@
         'app/webrtc/test/fakevideotrackrenderer.h',
         'app/webrtc/test/mockpeerconnectionobservers.h',
         'app/webrtc/test/testsdpstrings.h',
+        'app/webrtc/videosource_unittest.cc',
         'app/webrtc/videotrack_unittest.cc',
         'app/webrtc/webrtcsdp_unittest.cc',
         'app/webrtc/webrtcsession_unittest.cc',
@@ -464,13 +465,16 @@
         },
       ],
     }],
-    ['libjingle_objc == 1', {
+    ['OS=="ios" or (OS=="mac" and target_arch!="ia32" and mac_sdk>="10.7")', {
+      # The >=10.7 above is required to make ARC link cleanly (e.g. as
+      # opposed to _compile_ cleanly, which the library under test
+      # does just fine on 10.6 too).
       'targets': [
         {
+        'target_name': 'libjingle_peerconnection_objc_test',
           'variables': {
             'infoplist_file': './app/webrtc/objctests/Info.plist',
           },
-          'target_name': 'libjingle_peerconnection_objc_test',
           'type': 'executable',
           'mac_bundle': 1,
           'mac_bundle_resources': [
@@ -510,7 +514,7 @@
               ],
             }],
           ],
-        },
+        },  # target libjingle_peerconnection_objc_test
       ],
     }],
   ],
diff --git a/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h b/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h
index ded5698ba42..7ef0c9b1ab3 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/fakemediaengine.h
@@ -864,8 +864,6 @@ class FakeVideoEngine : public FakeBaseEngine {
     renderer_ = r;
     return true;
   }
-  bool SetVideoCapturer(VideoCapturer* /*capturer*/) { return true; }
-  VideoCapturer* GetVideoCapturer() const { return NULL; }
   bool SetCapture(bool capture) {
     capture_ = capture;
     return true;
diff --git a/chromium/third_party/libjingle/source/talk/media/base/fakenetworkinterface.h b/chromium/third_party/libjingle/source/talk/media/base/fakenetworkinterface.h
index 2fdd1d4a39a..d0f277e8f42 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/fakenetworkinterface.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/fakenetworkinterface.h
@@ -130,7 +130,8 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
   int recvbuf_size() const { return recvbuf_size_; }
 
  protected:
-  virtual bool SendPacket(talk_base::Buffer* packet) {
+  virtual bool SendPacket(talk_base::Buffer* packet,
+                          talk_base::DiffServCodePoint dscp) {
     talk_base::CritScope cs(&crit_);
 
     uint32 cur_ssrc = 0;
@@ -164,7 +165,8 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
     return true;
   }
 
-  virtual bool SendRtcp(talk_base::Buffer* packet) {
+  virtual bool SendRtcp(talk_base::Buffer* packet,
+                        talk_base::DiffServCodePoint dscp) {
     talk_base::CritScope cs(&crit_);
     rtcp_packets_.push_back(*packet);
     if (!conf_) {
diff --git a/chromium/third_party/libjingle/source/talk/media/base/fakevideorenderer.h b/chromium/third_party/libjingle/source/talk/media/base/fakevideorenderer.h
index 4000d5e2045..362e592951e 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/fakevideorenderer.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/fakevideorenderer.h
@@ -28,6 +28,7 @@
 #ifndef TALK_MEDIA_BASE_FAKEVIDEORENDERER_H_
 #define TALK_MEDIA_BASE_FAKEVIDEORENDERER_H_
 
+#include "talk/base/logging.h"
 #include "talk/base/sigslot.h"
 #include "talk/media/base/videoframe.h"
 #include "talk/media/base/videorenderer.h"
@@ -62,6 +63,13 @@ class FakeVideoRenderer : public VideoRenderer {
     if (!frame ||
         frame->GetWidth() != static_cast<size_t>(width_) ||
         frame->GetHeight() != static_cast<size_t>(height_)) {
+      if (!frame) {
+        LOG(LS_WARNING) << "RenderFrame expected non-null frame.";
+      } else {
+        LOG(LS_WARNING) << "RenderFrame expected frame of size " << width_
+                        << "x" << height_ << " but received frame of size "
+                        << frame->GetWidth() << "x" << frame->GetHeight();
+      }
       ++errors_;
       return false;
     }
diff --git a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine_unittest.cc b/chromium/third_party/libjingle/source/talk/media/base/filemediaengine_unittest.cc
index 703fc118d54..e4d72bbe610 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/filemediaengine_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/filemediaengine_unittest.cc
@@ -58,7 +58,8 @@ class FileNetworkInterface : public MediaChannel::NetworkInterface {
   }
 
   // Implement pure virtual methods of NetworkInterface.
-  virtual bool SendPacket(talk_base::Buffer* packet) {
+  virtual bool SendPacket(talk_base::Buffer* packet,
+                          talk_base::DiffServCodePoint dscp) {
     if (!packet) return false;
 
     if (media_channel_) {
@@ -74,11 +75,13 @@ class FileNetworkInterface : public MediaChannel::NetworkInterface {
     return true;
   }
 
-  virtual bool SendRtcp(talk_base::Buffer* packet) { return false; }
+  virtual bool SendRtcp(talk_base::Buffer* packet,
+                        talk_base::DiffServCodePoint dscp) { return false; }
   virtual int SetOption(MediaChannel::NetworkInterface::SocketType type,
       talk_base::Socket::Option opt, int option) {
     return 0;
   }
+  virtual void SetDefaultDSCPCode(talk_base::DiffServCodePoint dscp) {}
 
   size_t num_sent_packets() const { return num_sent_packets_; }
 
diff --git a/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h b/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h
index a9e37781b39..7431bc1001f 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/mediachannel.h
@@ -33,6 +33,7 @@
 
 #include "talk/base/basictypes.h"
 #include "talk/base/buffer.h"
+#include "talk/base/dscp.h"
 #include "talk/base/logging.h"
 #include "talk/base/sigslot.h"
 #include "talk/base/socket.h"
@@ -236,6 +237,7 @@ struct VideoOptions {
     adapt_input_to_cpu_usage.SetFrom(change.adapt_input_to_cpu_usage);
     adapt_cpu_with_smoothing.SetFrom(change.adapt_cpu_with_smoothing);
     adapt_view_switch.SetFrom(change.adapt_view_switch);
+    video_adapt_third.SetFrom(change.video_adapt_third);
     video_noise_reduction.SetFrom(change.video_noise_reduction);
     video_three_layers.SetFrom(change.video_three_layers);
     video_enable_camera_list.SetFrom(change.video_enable_camera_list);
@@ -245,6 +247,7 @@ struct VideoOptions {
     video_temporal_layer_screencast.SetFrom(
         change.video_temporal_layer_screencast);
     video_leaky_bucket.SetFrom(change.video_leaky_bucket);
+    cpu_overuse_detection.SetFrom(change.cpu_overuse_detection);
     conference_mode.SetFrom(change.conference_mode);
     process_adaptation_threshhold.SetFrom(change.process_adaptation_threshhold);
     system_low_adaptation_threshhold.SetFrom(
@@ -259,6 +262,7 @@ struct VideoOptions {
         adapt_input_to_cpu_usage == o.adapt_input_to_cpu_usage &&
         adapt_cpu_with_smoothing == o.adapt_cpu_with_smoothing &&
         adapt_view_switch == o.adapt_view_switch &&
+        video_adapt_third == o.video_adapt_third &&
         video_noise_reduction == o.video_noise_reduction &&
         video_three_layers == o.video_three_layers &&
         video_enable_camera_list == o.video_enable_camera_list &&
@@ -267,6 +271,7 @@ struct VideoOptions {
         video_watermark == o.video_watermark &&
         video_temporal_layer_screencast == o.video_temporal_layer_screencast &&
         video_leaky_bucket == o.video_leaky_bucket &&
+        cpu_overuse_detection == o.cpu_overuse_detection &&
         conference_mode == o.conference_mode &&
         process_adaptation_threshhold == o.process_adaptation_threshhold &&
         system_low_adaptation_threshhold ==
@@ -283,16 +288,17 @@ struct VideoOptions {
     ost << ToStringIfSet("cpu adaption", adapt_input_to_cpu_usage);
     ost << ToStringIfSet("cpu adaptation smoothing", adapt_cpu_with_smoothing);
     ost << ToStringIfSet("adapt view switch", adapt_view_switch);
+    ost << ToStringIfSet("video adapt third", video_adapt_third);
     ost << ToStringIfSet("noise reduction", video_noise_reduction);
     ost << ToStringIfSet("3 layers", video_three_layers);
     ost << ToStringIfSet("camera list", video_enable_camera_list);
-    ost << ToStringIfSet("1 layer screencast",
-                        video_one_layer_screencast);
+    ost << ToStringIfSet("1 layer screencast", video_one_layer_screencast);
     ost << ToStringIfSet("high bitrate", video_high_bitrate);
     ost << ToStringIfSet("watermark", video_watermark);
     ost << ToStringIfSet("video temporal layer screencast",
                          video_temporal_layer_screencast);
     ost << ToStringIfSet("leaky bucket", video_leaky_bucket);
+    ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection);
     ost << ToStringIfSet("conference mode", conference_mode);
     ost << ToStringIfSet("process", process_adaptation_threshhold);
     ost << ToStringIfSet("low", system_low_adaptation_threshhold);
@@ -310,6 +316,8 @@ struct VideoOptions {
   Settable<bool> adapt_cpu_with_smoothing;
   // Enable Adapt View Switch?
   Settable<bool> adapt_view_switch;
+  // Enable video adapt third?
+  Settable<bool> video_adapt_third;
   // Enable denoising?
   Settable<bool> video_noise_reduction;
   // Experimental: Enable multi layer?
@@ -326,6 +334,10 @@ struct VideoOptions {
   Settable<bool> video_temporal_layer_screencast;
   // Enable WebRTC leaky bucket when sending media packets.
   Settable<bool> video_leaky_bucket;
+  // Enable WebRTC Cpu Overuse Detection, which is a new version of the CPU
+  // adaptation algorithm. So this option will override the
+  // |adapt_input_to_cpu_usage|.
+  Settable<bool> cpu_overuse_detection;
   // Use conference mode?
   Settable<bool> conference_mode;
   // Threshhold for process cpu adaptation.  (Process limit)
@@ -403,8 +415,12 @@ class MediaChannel : public sigslot::has_slots<> {
   class NetworkInterface {
    public:
     enum SocketType { ST_RTP, ST_RTCP };
-    virtual bool SendPacket(talk_base::Buffer* packet) = 0;
-    virtual bool SendRtcp(talk_base::Buffer* packet) = 0;
+    virtual bool SendPacket(
+        talk_base::Buffer* packet,
+        talk_base::DiffServCodePoint dscp = talk_base::DSCP_NO_CHANGE) = 0;
+    virtual bool SendRtcp(
+        talk_base::Buffer* packet,
+        talk_base::DiffServCodePoint dscp = talk_base::DSCP_NO_CHANGE) = 0;
     virtual int SetOption(SocketType type, talk_base::Socket::Option opt,
                           int option) = 0;
     virtual ~NetworkInterface() {}
@@ -508,7 +524,8 @@ struct VoiceSenderInfo {
         echo_delay_median_ms(0),
         echo_delay_std_ms(0),
         echo_return_loss(0),
-        echo_return_loss_enhancement(0) {
+        echo_return_loss_enhancement(0),
+        typing_noise_detected(false) {
   }
 
   uint32 ssrc;
@@ -526,6 +543,7 @@ struct VoiceSenderInfo {
   int echo_delay_std_ms;
   int echo_return_loss;
   int echo_return_loss_enhancement;
+  bool typing_noise_detected;
 };
 
 struct VoiceReceiverInfo {
@@ -849,10 +867,12 @@ class VideoMediaChannel : public MediaChannel {
 };
 
 enum DataMessageType {
-  // TODO(pthatcher):  Make this enum match the SCTP PPIDs that WebRTC uses?
-  DMT_CONTROL = 0,
-  DMT_BINARY = 1,
-  DMT_TEXT = 2,
+  // Chrome-Internal use only.  See SctpDataMediaChannel for the actual PPID
+  // values.
+  DMT_NONE = 0,
+  DMT_CONTROL = 1,
+  DMT_BINARY = 2,
+  DMT_TEXT = 3,
 };
 
 // Info about data received in DataMediaChannel.  For use in
diff --git a/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h b/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h
index 7a1244dfae7..8ebc13b1086 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/mediaengine.h
@@ -259,10 +259,10 @@ class CompositeMediaEngine : public MediaEngineInterface {
   }
 
   virtual void SetVoiceLogging(int min_sev, const char* filter) {
-    return voice_.SetLogging(min_sev, filter);
+    voice_.SetLogging(min_sev, filter);
   }
   virtual void SetVideoLogging(int min_sev, const char* filter) {
-    return video_.SetLogging(min_sev, filter);
+    video_.SetLogging(min_sev, filter);
   }
 
   virtual bool RegisterVoiceProcessor(uint32 ssrc,
diff --git a/chromium/third_party/libjingle/source/talk/media/base/testutils.h b/chromium/third_party/libjingle/source/talk/media/base/testutils.h
index 7bc7dc3ae02..4d037b7ba38 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/testutils.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/testutils.h
@@ -28,8 +28,16 @@
 #ifndef TALK_MEDIA_BASE_TESTUTILS_H_
 #define TALK_MEDIA_BASE_TESTUTILS_H_
 
+#ifdef LINUX
+#include <X11/Xlib.h>
+// X defines a few macros that stomp on types that gunit.h uses.
+#undef None
+#undef Bool
+#endif
+
 #include <string>
 #include <vector>
+
 #if !defined(DISABLE_YUV)
 #include "libyuv/compare.h"
 #endif
@@ -237,6 +245,37 @@ bool ContainsMatchingCodec(const std::vector<C>& codecs, const C& codec) {
   return false;
 }
 
+#define MAYBE_SKIP_SCREENCAST_TEST() \
+  if (!cricket::IsScreencastingAvailable()) { \
+    LOG(LS_WARNING) << "Skipping test, since it doesn't have the requisite " \
+                    << "X environment for screen capture."; \
+    return; \
+  } \
+
+#ifdef LINUX
+struct XDisplay {
+  XDisplay() : display_(XOpenDisplay(NULL)) { }
+  ~XDisplay() { if (display_) XCloseDisplay(display_); }
+  bool IsValid() const { return display_ != NULL; }
+  operator Display*() { return display_; }
+ private:
+  Display* display_;
+};
+#endif
+
+// Returns true if screencasting is available. When false, anything that uses
+// screencasting features may fail.
+inline bool IsScreencastingAvailable() {
+#ifdef LINUX
+  XDisplay display;
+  if (!display.IsValid()) {
+    LOG(LS_WARNING) << "No X Display available.";
+    return false;
+  }
+#endif
+  return true;
+}
+
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_BASE_TESTUTILS_H_
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videoadapter.cc b/chromium/third_party/libjingle/source/talk/media/base/videoadapter.cc
index cef4248d38a..20114fbd383 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videoadapter.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/videoadapter.cc
@@ -45,86 +45,100 @@ static const float kCpuLoadWeightCoefficient = 0.4f;
 // The seed value for the cpu load moving average.
 static const float kCpuLoadInitialAverage = 0.5f;
 
-// TODO(fbarchard): Consider making scale factor table settable, to allow
-// application to select quality vs performance tradeoff.
-// TODO(fbarchard): Add framerate scaling to tables for 1/2 framerate.
-// List of scale factors that adapter will scale by.
-#if defined(IOS) || defined(ANDROID)
-// Mobile needs 1/4 scale for VGA (640 x 360) to QQVGA (160 x 90)
-// or 1/4 scale for HVGA (480 x 270) to QQHVGA (120 x 67)
-static const int kMinNumPixels = 120 * 67;
-static float kScaleFactors[] = {
-  1.f / 1.f,  // Full size.
-  3.f / 4.f,  // 3/4 scale.
-  1.f / 2.f,  // 1/2 scale.
-  3.f / 8.f,  // 3/8 scale.
-  1.f / 4.f,  // 1/4 scale.
-};
-#else
 // Desktop needs 1/8 scale for HD (1280 x 720) to QQVGA (160 x 90)
-static const int kMinNumPixels = 160 * 100;
-static float kScaleFactors[] = {
-  1.f / 1.f,  // Full size.
-  3.f / 4.f,  // 3/4 scale.
-  1.f / 2.f,  // 1/2 scale.
-  3.f / 8.f,  // 3/8 scale.
-  1.f / 4.f,  // 1/4 scale.
+static const float kScaleFactors[] = {
+  1.f / 1.f,   // Full size.
+  3.f / 4.f,   // 3/4 scale.
+  1.f / 2.f,   // 1/2 scale.
+  3.f / 8.f,   // 3/8 scale.
+  1.f / 4.f,   // 1/4 scale.
   3.f / 16.f,  // 3/16 scale.
-  1.f / 8.f  // 1/8 scale.
+  1.f / 8.f,   // 1/8 scale.
+  0.f  // End of table.
 };
-#endif
 
-static const int kNumScaleFactors = ARRAY_SIZE(kScaleFactors);
+// TODO(fbarchard): Use this table (optionally) for CPU and GD as well.
+static const float kViewScaleFactors[] = {
+  1.f / 1.f,   // Full size.
+  3.f / 4.f,   // 3/4 scale.
+  2.f / 3.f,   // 2/3 scale.  // Allow 1080p to 720p.
+  1.f / 2.f,   // 1/2 scale.
+  3.f / 8.f,   // 3/8 scale.
+  1.f / 3.f,   // 1/3 scale.  // Allow 1080p to 360p.
+  1.f / 4.f,   // 1/4 scale.
+  3.f / 16.f,  // 3/16 scale.
+  1.f / 8.f,   // 1/8 scale.
+  0.f  // End of table.
+};
 
+const float* VideoAdapter::GetViewScaleFactors() const {
+  return scale_third_ ? kViewScaleFactors : kScaleFactors;
+}
+
+// For resolutions that would scale down a little instead of up a little,
+// bias toward scaling up a little.  This will tend to choose 3/4 scale instead
+// of 2/3 scale, when the 2/3 is not an exact match.
+static const float kUpBias = -0.9f;
 // Find the scale factor that, when applied to width and height, is closest
 // to num_pixels.
-float VideoAdapter::FindClosestScale(int width, int height,
-                                     int target_num_pixels) {
+float VideoAdapter::FindScale(const float* scale_factors,
+                              const float upbias,
+                              int width, int height,
+                              int target_num_pixels) {
+  const float kMinNumPixels = 160 * 90;
   if (!target_num_pixels) {
     return 0.f;
   }
-  int best_distance = INT_MAX;
-  int best_index = kNumScaleFactors - 1;  // Default to max scale.
-  for (int i = 0; i < kNumScaleFactors; ++i) {
-    int test_num_pixels = static_cast<int>(width * kScaleFactors[i] *
-                                           height * kScaleFactors[i]);
-    int diff = test_num_pixels - target_num_pixels;
+  float best_distance = static_cast<float>(INT_MAX);
+  float best_scale = 1.f;  // Default to unscaled if nothing matches.
+  float pixels = static_cast<float>(width * height);
+  for (int i = 0; ; ++i) {
+    float scale = scale_factors[i];
+    float test_num_pixels = pixels * scale * scale;
+    // Do not consider scale factors that produce too small images.
+    // Scale factor of 0 at end of table will also exit here.
+    if (test_num_pixels < kMinNumPixels) {
+      break;
+    }
+    float diff = target_num_pixels - test_num_pixels;
+    // If resolution is higher than desired, bias the difference based on
+    // preference for slightly larger for nearest, or avoid completely if
+    // looking for lower resolutions only.
     if (diff < 0) {
-      diff = -diff;
+      diff = diff * kUpBias;
     }
     if (diff < best_distance) {
       best_distance = diff;
-      best_index = i;
+      best_scale = scale;
       if (best_distance == 0) {  // Found exact match.
         break;
       }
     }
   }
-  return kScaleFactors[best_index];
+  return best_scale;
+}
+
+// Find the closest scale factor.
+float VideoAdapter::FindClosestScale(int width, int height,
+                                         int target_num_pixels) {
+  return FindScale(kScaleFactors, kUpBias,
+                   width, height, target_num_pixels);
+}
+
+// Find the closest view scale factor.
+float VideoAdapter::FindClosestViewScale(int width, int height,
+                                         int target_num_pixels) {
+  return FindScale(GetViewScaleFactors(), kUpBias,
+                   width, height, target_num_pixels);
 }
 
 // Finds the scale factor that, when applied to width and height, produces
 // fewer than num_pixels.
+static const float kUpAvoidBias = -1000000000.f;
 float VideoAdapter::FindLowerScale(int width, int height,
                                    int target_num_pixels) {
-  if (!target_num_pixels) {
-    return 0.f;
-  }
-  int best_distance = INT_MAX;
-  int best_index = kNumScaleFactors - 1;  // Default to max scale.
-  for (int i = 0; i < kNumScaleFactors; ++i) {
-    int test_num_pixels = static_cast<int>(width * kScaleFactors[i] *
-                                           height * kScaleFactors[i]);
-    int diff = target_num_pixels - test_num_pixels;
-    if (diff >= 0 && diff < best_distance) {
-      best_distance = diff;
-      best_index = i;
-      if (best_distance == 0) {  // Found exact match.
-        break;
-      }
-    }
-  }
-  return kScaleFactors[best_index];
+  return FindScale(GetViewScaleFactors(), kUpAvoidBias,
+                   width, height, target_num_pixels);
 }
 
 // There are several frame sizes used by Adapter.  This explains them
@@ -147,6 +161,12 @@ float VideoAdapter::FindLowerScale(int width, int height,
 // Implementation of VideoAdapter
 VideoAdapter::VideoAdapter()
     : output_num_pixels_(INT_MAX),
+      scale_third_(false),
+      frames_(0),
+      adapted_frames_(0),
+      adaption_changes_(0),
+      previous_width(0),
+      previous_height(0),
       black_output_(false),
       is_black_(false),
       interval_next_frame_(0) {
@@ -155,12 +175,6 @@ VideoAdapter::VideoAdapter()
 VideoAdapter::~VideoAdapter() {
 }
 
-void VideoAdapter::SetInputFormat(const VideoFrame& in_frame) {
-  talk_base::CritScope cs(&critical_section_);
-  input_format_.width = static_cast<int>(in_frame.GetWidth());
-  input_format_.height = static_cast<int>(in_frame.GetHeight());
-}
-
 void VideoAdapter::SetInputFormat(const VideoFormat& format) {
   talk_base::CritScope cs(&critical_section_);
   input_format_ = format;
@@ -168,6 +182,29 @@ void VideoAdapter::SetInputFormat(const VideoFormat& format) {
       output_format_.interval, input_format_.interval);
 }
 
+void CoordinatedVideoAdapter::SetInputFormat(const VideoFormat& format) {
+  int previous_width = input_format().width;
+  int previous_height = input_format().height;
+  bool is_resolution_change = previous_width > 0 && format.width > 0 &&
+                              (previous_width != format.width ||
+                               previous_height != format.height);
+  VideoAdapter::SetInputFormat(format);
+  if (is_resolution_change) {
+    int width, height;
+    // Trigger the adaptation logic again, to potentially reset the adaptation
+    // state for things like view requests that may not longer be capping
+    // output (or may now cap output).
+    AdaptToMinimumFormat(&width, &height);
+    LOG(LS_INFO) << "VAdapt Input Resolution Change: "
+                 << "Previous input resolution: "
+                 << previous_width << "x" << previous_height
+                 << " New input resolution: "
+                 << format.width << "x" << format.height
+                 << " New output resolution: "
+                 << width << "x" << height;
+  }
+}
+
 void VideoAdapter::SetOutputFormat(const VideoFormat& format) {
   talk_base::CritScope cs(&critical_section_);
   output_format_ = format;
@@ -208,9 +245,13 @@ bool VideoAdapter::AdaptFrame(const VideoFrame* in_frame,
   if (!in_frame || !out_frame) {
     return false;
   }
+  ++frames_;
 
   // Update input to actual frame dimensions.
-  SetInputFormat(*in_frame);
+  VideoFormat format(static_cast<int>(in_frame->GetWidth()),
+                     static_cast<int>(in_frame->GetHeight()),
+                     input_format_.interval, input_format_.fourcc);
+  SetInputFormat(format);
 
   // Drop the input frame if necessary.
   bool should_drop = false;
@@ -236,8 +277,9 @@ bool VideoAdapter::AdaptFrame(const VideoFrame* in_frame,
     return true;
   }
 
+  float scale = 1.f;
   if (output_num_pixels_) {
-    float scale = VideoAdapter::FindClosestScale(
+    scale = VideoAdapter::FindClosestViewScale(
         static_cast<int>(in_frame->GetWidth()),
         static_cast<int>(in_frame->GetHeight()),
         output_num_pixels_);
@@ -251,9 +293,45 @@ bool VideoAdapter::AdaptFrame(const VideoFrame* in_frame,
   }
 
   *out_frame = output_frame_.get();
+
+  // Show VAdapt log every 300 frames. (10 seconds)
+  // TODO(fbarchard): Consider GetLogSeverity() to change interval to less
+  // for LS_VERBOSE and more for LS_INFO.
+  bool show = frames_ % 300 == 0;
+  if (in_frame->GetWidth() != (*out_frame)->GetWidth() ||
+      in_frame->GetHeight() != (*out_frame)->GetHeight()) {
+    ++adapted_frames_;
+  }
+  // TODO(fbarchard): LOG the previous output resolution and track input
+  // resolution changes as well.  Consider dropping the statistics into their
+  // own class which could be queried publically.
+  bool changed = false;
+  if (previous_width && (previous_width != (*out_frame)->GetWidth() ||
+      previous_height != (*out_frame)->GetHeight())) {
+    show = true;
+    ++adaption_changes_;
+    changed = true;
+  }
+  if (show) {
+    // TODO(fbarchard): Reduce to LS_VERBOSE when adapter info is not needed
+    // in default calls.
+    LOG(LS_INFO) << "VAdapt Frame: " << adapted_frames_
+                 << " / " << frames_
+                 << " Changes: " << adaption_changes_
+                 << " Input: " << in_frame->GetWidth()
+                 << "x" << in_frame->GetHeight()
+                 << " Scale: " << scale
+                 << " Output: " << (*out_frame)->GetWidth()
+                 << "x" << (*out_frame)->GetHeight()
+                 << " Changed: " << (changed ? "true" : "false");
+  }
+  previous_width = (*out_frame)->GetWidth();
+  previous_height = (*out_frame)->GetHeight();
+
   return true;
 }
 
+// Scale or Blacken the frame.  Returns true if successful.
 bool VideoAdapter::StretchToOutputFrame(const VideoFrame* in_frame) {
   int output_width = output_format_.width;
   int output_height = output_format_.height;
@@ -409,37 +487,12 @@ void CoordinatedVideoAdapter::OnEncoderResolutionRequest(
                << " To: " << new_width << "x" << new_height;
 }
 
-// A CPU request for new resolution
-void CoordinatedVideoAdapter::OnCpuLoadUpdated(
-    int current_cpus, int max_cpus, float process_load, float system_load) {
+// A Bandwidth GD request for new resolution
+void CoordinatedVideoAdapter::OnCpuResolutionRequest(AdaptRequest request) {
   talk_base::CritScope cs(&request_critical_section_);
   if (!cpu_adaptation_) {
     return;
   }
-  // Update the moving average of system load. Even if we aren't smoothing,
-  // we'll still calculate this information, in case smoothing is later enabled.
-  system_load_average_ = kCpuLoadWeightCoefficient * system_load +
-      (1.0f - kCpuLoadWeightCoefficient) * system_load_average_;
-  if (cpu_smoothing_) {
-    system_load = system_load_average_;
-  }
-  // If we haven't started taking samples yet, wait until we have at least
-  // the correct number of samples per the wait time.
-  if (cpu_adapt_wait_time_ == 0) {
-    cpu_adapt_wait_time_ = talk_base::TimeAfter(kCpuLoadMinSampleTime);
-  }
-  AdaptRequest request = FindCpuRequest(current_cpus, max_cpus,
-                                        process_load, system_load);
-  // Make sure we're not adapting too quickly.
-  if (request != KEEP) {
-    if (talk_base::TimeIsLater(talk_base::Time(),
-                               cpu_adapt_wait_time_)) {
-      LOG(LS_VERBOSE) << "VAdapt CPU load high/low but do not adapt until "
-                      << talk_base::TimeUntil(cpu_adapt_wait_time_) << " ms";
-      request = KEEP;
-    }
-  }
-
   // Update how many times we have downgraded due to the cpu load.
   switch (request) {
     case DOWNGRADE:
@@ -482,13 +535,46 @@ void CoordinatedVideoAdapter::OnCpuLoadUpdated(
   LOG(LS_INFO) << "VAdapt CPU Request: "
                << (DOWNGRADE == request ? "down" :
                    (UPGRADE == request ? "up" : "keep"))
-               << " Process: " << process_load
-               << " System: " << system_load
                << " Steps: " << cpu_downgrade_count_
                << " Changed: " << (changed ? "true" : "false")
                << " To: " << new_width << "x" << new_height;
 }
 
+// A CPU request for new resolution
+// TODO(fbarchard): Move outside adapter.
+void CoordinatedVideoAdapter::OnCpuLoadUpdated(
+    int current_cpus, int max_cpus, float process_load, float system_load) {
+  talk_base::CritScope cs(&request_critical_section_);
+  if (!cpu_adaptation_) {
+    return;
+  }
+  // Update the moving average of system load. Even if we aren't smoothing,
+  // we'll still calculate this information, in case smoothing is later enabled.
+  system_load_average_ = kCpuLoadWeightCoefficient * system_load +
+      (1.0f - kCpuLoadWeightCoefficient) * system_load_average_;
+  if (cpu_smoothing_) {
+    system_load = system_load_average_;
+  }
+  // If we haven't started taking samples yet, wait until we have at least
+  // the correct number of samples per the wait time.
+  if (cpu_adapt_wait_time_ == 0) {
+    cpu_adapt_wait_time_ = talk_base::TimeAfter(kCpuLoadMinSampleTime);
+  }
+  AdaptRequest request = FindCpuRequest(current_cpus, max_cpus,
+                                        process_load, system_load);
+  // Make sure we're not adapting too quickly.
+  if (request != KEEP) {
+    if (talk_base::TimeIsLater(talk_base::Time(),
+                               cpu_adapt_wait_time_)) {
+      LOG(LS_VERBOSE) << "VAdapt CPU load high/low but do not adapt until "
+                      << talk_base::TimeUntil(cpu_adapt_wait_time_) << " ms";
+      request = KEEP;
+    }
+  }
+
+  OnCpuResolutionRequest(request);
+}
+
 // Called by cpu adapter on up requests.
 bool CoordinatedVideoAdapter::IsMinimumFormat(int pixels) {
   // Find closest scale factor that matches input resolution to min_num_pixels
@@ -522,51 +608,46 @@ bool CoordinatedVideoAdapter::AdaptToMinimumFormat(int* new_width,
     input = new_output;
   }
   int old_num_pixels = GetOutputNumPixels();
-  // Find resolution that respects ViewRequest or less pixels.
-  int view_desired_num_pixels = view_desired_num_pixels_;
-  int min_num_pixels = view_desired_num_pixels_;
-  if (!input.IsSize0x0()) {
-    float scale = FindLowerScale(input.width, input.height, min_num_pixels);
-    min_num_pixels = view_desired_num_pixels =
-        static_cast<int>(input.width * input.height * scale * scale + .5f);
-  }
-  // Reduce resolution further, if necessary, based on encoder bandwidth (GD).
+  int min_num_pixels = INT_MAX;
+  adapt_reason_ = 0;
+
+  // Reduce resolution based on encoder bandwidth (GD).
   if (encoder_desired_num_pixels_ &&
       (encoder_desired_num_pixels_ < min_num_pixels)) {
+    adapt_reason_ |= ADAPTREASON_BANDWIDTH;
     min_num_pixels = encoder_desired_num_pixels_;
   }
-  // Reduce resolution further, if necessary, based on CPU.
+  // Reduce resolution based on CPU.
   if (cpu_adaptation_ && cpu_desired_num_pixels_ &&
-      (cpu_desired_num_pixels_ < min_num_pixels)) {
+      (cpu_desired_num_pixels_ <= min_num_pixels)) {
+    if (cpu_desired_num_pixels_ < min_num_pixels) {
+      adapt_reason_ = ADAPTREASON_CPU;
+    } else {
+      adapt_reason_ |= ADAPTREASON_CPU;
+    }
     min_num_pixels = cpu_desired_num_pixels_;
   }
-
-  // Determine which factors are keeping adapter resolution low.
-  // Caveat: Does not consider framerate.
-  adapt_reason_ = static_cast<AdaptReason>(0);
-  if (view_desired_num_pixels == min_num_pixels) {
-    adapt_reason_ |= ADAPTREASON_VIEW;
-  }
-  if (encoder_desired_num_pixels_ == min_num_pixels) {
-    adapt_reason_ |= ADAPTREASON_BANDWIDTH;
-  }
-  if (cpu_desired_num_pixels_ == min_num_pixels) {
-    adapt_reason_ |= ADAPTREASON_CPU;
+  // Round resolution for GD or CPU to allow 1/2 to map to 9/16.
+  if (!input.IsSize0x0() && min_num_pixels != INT_MAX) {
+    float scale = FindClosestScale(input.width, input.height, min_num_pixels);
+    min_num_pixels = static_cast<int>(input.width * scale + .5f) *
+        static_cast<int>(input.height * scale + .5f);
   }
-
-  // Prevent going below QQVGA.
-  if (min_num_pixels > 0 && min_num_pixels < kMinNumPixels) {
-    min_num_pixels = kMinNumPixels;
+  // Reduce resolution based on View Request.
+  if (view_desired_num_pixels_ <= min_num_pixels) {
+    if (view_desired_num_pixels_ < min_num_pixels) {
+      adapt_reason_ = ADAPTREASON_VIEW;
+    } else {
+      adapt_reason_ |= ADAPTREASON_VIEW;
+    }
+    min_num_pixels = view_desired_num_pixels_;
   }
-  SetOutputNumPixels(min_num_pixels);
-
-  // Find closest scale factor that matches input resolution to min_num_pixels
-  // and set that for output resolution.  This is not needed for VideoAdapter,
-  // but provides feedback to unittests and users on expected resolution.
-  // Actual resolution is based on input frame.
+  // Snap to a scale factor.
   float scale = 1.0f;
   if (!input.IsSize0x0()) {
-    scale = FindClosestScale(input.width, input.height, min_num_pixels);
+    scale = FindLowerScale(input.width, input.height, min_num_pixels);
+    min_num_pixels = static_cast<int>(input.width * scale + .5f) *
+        static_cast<int>(input.height * scale + .5f);
   }
   if (scale == 1.0f) {
     adapt_reason_ = 0;
@@ -574,6 +655,8 @@ bool CoordinatedVideoAdapter::AdaptToMinimumFormat(int* new_width,
   *new_width = new_output.width = static_cast<int>(input.width * scale + .5f);
   *new_height = new_output.height = static_cast<int>(input.height * scale +
                                                      .5f);
+  SetOutputNumPixels(min_num_pixels);
+
   new_output.interval = view_desired_interval_;
   SetOutputFormat(new_output);
   int new_num_pixels = GetOutputNumPixels();
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videoadapter.h b/chromium/third_party/libjingle/source/talk/media/base/videoadapter.h
index c41ac6e5cdc..2bd31d5df5c 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videoadapter.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/videoadapter.h
@@ -45,8 +45,7 @@ class VideoAdapter {
   VideoAdapter();
   virtual ~VideoAdapter();
 
-  void SetInputFormat(const VideoFrame& in_frame);
-  void SetInputFormat(const VideoFormat& format);
+  virtual void SetInputFormat(const VideoFormat& format);
   void SetOutputFormat(const VideoFormat& format);
   // Constrain output resolution to this many pixels overall
   void SetOutputNumPixels(int num_pixels);
@@ -65,16 +64,34 @@ class VideoAdapter {
   // the output frame.
   bool AdaptFrame(const VideoFrame* in_frame, const VideoFrame** out_frame);
 
+  void set_scale_third(bool enable) {
+    LOG(LS_INFO) << "Video Adapter third scaling is now "
+                 << (enable ? "enabled" : "disabled");
+    scale_third_ = enable;
+  }
+  bool scale_third() const { return scale_third_; }
+
  protected:
   float FindClosestScale(int width, int height, int target_num_pixels);
+  float FindClosestViewScale(int width, int height, int target_num_pixels);
   float FindLowerScale(int width, int height, int target_num_pixels);
 
  private:
+  const float* GetViewScaleFactors() const;
+  float FindScale(const float* scale_factors,
+                  const float upbias, int width, int height,
+                  int target_num_pixels);
   bool StretchToOutputFrame(const VideoFrame* in_frame);
 
   VideoFormat input_format_;
   VideoFormat output_format_;
   int output_num_pixels_;
+  bool scale_third_;  // True if adapter allows scaling to 1/3 and 2/3.
+  int frames_;  // Number of input frames.
+  int adapted_frames_;  // Number of frames scaled.
+  int adaption_changes_;  // Number of changes in scale factor.
+  size_t previous_width;  // Previous adapter output width.
+  size_t previous_height;  // Previous adapter output height.
   bool black_output_;  // Flag to tell if we need to black output_frame_.
   bool is_black_;  // Flag to tell if output_frame_ is currently black.
   int64 interval_next_frame_;
@@ -102,6 +119,8 @@ class CoordinatedVideoAdapter
   CoordinatedVideoAdapter();
   virtual ~CoordinatedVideoAdapter() {}
 
+  virtual void SetInputFormat(const VideoFormat& format);
+
   // Enable or disable video adaptation due to the change of the CPU load.
   void set_cpu_adaptation(bool enable) { cpu_adaptation_ = enable; }
   bool cpu_adaptation() const { return cpu_adaptation_; }
@@ -176,6 +195,8 @@ class CoordinatedVideoAdapter
   void OnOutputFormatRequest(const VideoFormat& format);
   // Handle the resolution request from the encoder due to bandwidth changes.
   void OnEncoderResolutionRequest(int width, int height, AdaptRequest request);
+  // Handle the resolution request for CPU overuse.
+  void OnCpuResolutionRequest(AdaptRequest request);
   // Handle the CPU load provided by a CPU monitor.
   void OnCpuLoadUpdated(int current_cpus, int max_cpus,
                         float process_load, float system_load);
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videocapturer.cc b/chromium/third_party/libjingle/source/talk/media/base/videocapturer.cc
index 3bc23731db0..acab19d6616 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videocapturer.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/videocapturer.cc
@@ -103,10 +103,12 @@ VideoCapturer::VideoCapturer(talk_base::Thread* thread) : thread_(thread) {
 void VideoCapturer::Construct() {
   ClearAspectRatio();
   enable_camera_list_ = false;
+  square_pixel_aspect_ratio_ = false;
   capture_state_ = CS_STOPPED;
   SignalFrameCaptured.connect(this, &VideoCapturer::OnFrameCaptured);
   scaled_width_ = 0;
   scaled_height_ = 0;
+  screencast_max_pixels_ = 0;
   muted_ = false;
   black_frame_count_down_ = kNumBlackFramesOnMute;
 }
@@ -323,24 +325,30 @@ void VideoCapturer::OnFrameCaptured(VideoCapturer*,
 #if !defined(DISABLE_YUV)
   if (IsScreencast()) {
     int scaled_width, scaled_height;
-    int desired_screencast_fps = capture_format_.get() ?
-        VideoFormat::IntervalToFps(capture_format_->interval) :
-        kDefaultScreencastFps;
-    ComputeScale(captured_frame->width, captured_frame->height,
-                 desired_screencast_fps, &scaled_width, &scaled_height);
-
-    if (scaled_width != scaled_width_ || scaled_height != scaled_height_) {
-      LOG(LS_VERBOSE) << "Scaling Screencast from "
-                      << captured_frame->width << "x"
-                      << captured_frame->height << " to "
-                      << scaled_width << "x" << scaled_height;
-      scaled_width_ = scaled_width;
-      scaled_height_ = scaled_height;
+    if (screencast_max_pixels_ > 0) {
+      ComputeScaleMaxPixels(captured_frame->width, captured_frame->height,
+          screencast_max_pixels_, &scaled_width, &scaled_height);
+    } else {
+      int desired_screencast_fps = capture_format_.get() ?
+          VideoFormat::IntervalToFps(capture_format_->interval) :
+          kDefaultScreencastFps;
+      ComputeScale(captured_frame->width, captured_frame->height,
+                   desired_screencast_fps, &scaled_width, &scaled_height);
     }
+
     if (FOURCC_ARGB == captured_frame->fourcc &&
-        (scaled_width != captured_frame->height ||
-         scaled_height != captured_frame->height)) {
-      CapturedFrame* scaled_frame = const_cast<CapturedFrame*>(captured_frame);
+        (scaled_width != captured_frame->width ||
+        scaled_height != captured_frame->height)) {
+      if (scaled_width != scaled_width_ || scaled_height != scaled_height_) {
+        LOG(LS_INFO) << "Scaling Screencast from "
+                     << captured_frame->width << "x"
+                     << captured_frame->height << " to "
+                     << scaled_width << "x" << scaled_height;
+        scaled_width_ = scaled_width;
+        scaled_height_ = scaled_height;
+      }
+      CapturedFrame* modified_frame =
+          const_cast<CapturedFrame*>(captured_frame);
       // Compute new width such that width * height is less than maximum but
       // maintains original captured frame aspect ratio.
       // Round down width to multiple of 4 so odd width won't round up beyond
@@ -349,17 +357,88 @@ void VideoCapturer::OnFrameCaptured(VideoCapturer*,
       libyuv::ARGBScale(reinterpret_cast<const uint8*>(captured_frame->data),
                         captured_frame->width * 4, captured_frame->width,
                         captured_frame->height,
-                        reinterpret_cast<uint8*>(scaled_frame->data),
+                        reinterpret_cast<uint8*>(modified_frame->data),
                         scaled_width * 4, scaled_width, scaled_height,
                         libyuv::kFilterBilinear);
-      scaled_frame->width = scaled_width;
-      scaled_frame->height = scaled_height;
-      scaled_frame->data_size = scaled_width * 4 * scaled_height;
+      modified_frame->width = scaled_width;
+      modified_frame->height = scaled_height;
+      modified_frame->data_size = scaled_width * 4 * scaled_height;
     }
   }
+
+  const int kYuy2Bpp = 2;
+  const int kArgbBpp = 4;
+  // TODO(fbarchard): Make a helper function to adjust pixels to square.
+  // TODO(fbarchard): Hook up experiment to scaling.
+  // TODO(fbarchard): Avoid scale and convert if muted.
+  // Temporary buffer is scoped here so it will persist until i420_frame.Init()
+  // makes a copy of the frame, converting to I420.
+  talk_base::scoped_array<uint8> temp_buffer;
+  // YUY2 can be scaled vertically using an ARGB scaler.  Aspect ratio is only
+  // a problem on OSX.  OSX always converts webcams to YUY2 or UYVY.
+  bool can_scale =
+      FOURCC_YUY2 == CanonicalFourCC(captured_frame->fourcc) ||
+      FOURCC_UYVY == CanonicalFourCC(captured_frame->fourcc);
+
+  // If pixels are not square, optionally use vertical scaling to make them
+  // square.  Square pixels simplify the rest of the pipeline, including
+  // effects and rendering.
+  if (can_scale && square_pixel_aspect_ratio_ &&
+      captured_frame->pixel_width != captured_frame->pixel_height) {
+    int scaled_width, scaled_height;
+    // modified_frame points to the captured_frame but with const casted away
+    // so it can be modified.
+    CapturedFrame* modified_frame = const_cast<CapturedFrame*>(captured_frame);
+    // Compute the frame size that makes pixels square pixel aspect ratio.
+    ComputeScaleToSquarePixels(captured_frame->width, captured_frame->height,
+                               captured_frame->pixel_width,
+                               captured_frame->pixel_height,
+                               &scaled_width, &scaled_height);
+
+    if (scaled_width != scaled_width_ || scaled_height != scaled_height_) {
+      LOG(LS_INFO) << "Scaling WebCam from "
+                   << captured_frame->width << "x"
+                   << captured_frame->height << " to "
+                   << scaled_width << "x" << scaled_height
+                   << " for PAR "
+                   << captured_frame->pixel_width << "x"
+                   << captured_frame->pixel_height;
+      scaled_width_ = scaled_width;
+      scaled_height_ = scaled_height;
+    }
+    const int modified_frame_size = scaled_width * scaled_height * kYuy2Bpp;
+    uint8* temp_buffer_data;
+    // Pixels are wide and short; Increasing height. Requires temporary buffer.
+    if (scaled_height > captured_frame->height) {
+      temp_buffer.reset(new uint8[modified_frame_size]);
+      temp_buffer_data = temp_buffer.get();
+    } else {
+      // Pixels are narrow and tall; Decreasing height. Scale will be done
+      // in place.
+      temp_buffer_data = reinterpret_cast<uint8*>(captured_frame->data);
+    }
+
+    // Use ARGBScaler to vertically scale the YUY2 image, adjusting for 16 bpp.
+    libyuv::ARGBScale(reinterpret_cast<const uint8*>(captured_frame->data),
+                      captured_frame->width * kYuy2Bpp,  // Stride for YUY2.
+                      captured_frame->width * kYuy2Bpp / kArgbBpp,  // Width.
+                      abs(captured_frame->height),  // Height.
+                      temp_buffer_data,
+                      scaled_width * kYuy2Bpp,  // Stride for YUY2.
+                      scaled_width * kYuy2Bpp / kArgbBpp,  // Width.
+                      abs(scaled_height),  // New height.
+                      libyuv::kFilterBilinear);
+    modified_frame->width = scaled_width;
+    modified_frame->height = scaled_height;
+    modified_frame->pixel_width = 1;
+    modified_frame->pixel_height = 1;
+    modified_frame->data_size = modified_frame_size;
+    modified_frame->data = temp_buffer_data;
+  }
 #endif  // !DISABLE_YUV
-        // Size to crop captured frame to.  This adjusts the captured frames
-        // aspect ratio to match the final view aspect ratio, considering pixel
+
+  // Size to crop captured frame to.  This adjusts the captured frames
+  // aspect ratio to match the final view aspect ratio, considering pixel
   // aspect ratio and rotation.  The final size may be scaled down by video
   // adapter to better match ratio_w_ x ratio_h_.
   // Note that abs() of frame height is passed in, because source may be
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videocapturer.h b/chromium/third_party/libjingle/source/talk/media/base/videocapturer.h
index 3997976f893..933fc825000 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videocapturer.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/videocapturer.h
@@ -236,11 +236,18 @@ class VideoCapturer
   bool enable_camera_list() {
     return enable_camera_list_;
   }
+
+  // Enable scaling to ensure square pixels.
+  void set_square_pixel_aspect_ratio(bool square_pixel_aspect_ratio) {
+    square_pixel_aspect_ratio_ = square_pixel_aspect_ratio;
+  }
+  bool square_pixel_aspect_ratio() {
+    return square_pixel_aspect_ratio_;
+  }
+
   // Signal all capture state changes that are not a direct result of calling
   // Start().
   sigslot::signal2<VideoCapturer*, CaptureState> SignalStateChange;
-  // TODO(hellner): rename |SignalFrameCaptured| to something like
-  //                |SignalRawFrame| or |SignalNativeFrame|.
   // Frame callbacks are multithreaded to allow disconnect and connect to be
   // called concurrently. It also ensures that it is safe to call disconnect
   // at any time which is needed since the signal may be called from an
@@ -254,6 +261,17 @@ class VideoCapturer
 
   const VideoProcessors& video_processors() const { return video_processors_; }
 
+  // If 'screencast_max_pixels' is set greater than zero, screencasts will be
+  // scaled to be no larger than this value.
+  // If set to zero, the max pixels will be limited to
+  // Retina MacBookPro 15" resolution of 2880 x 1800.
+  // For high fps, maximum pixels limit is set based on common 24" monitor
+  // resolution of 2048 x 1280.
+  int screencast_max_pixels() const { return screencast_max_pixels_; }
+  void set_screencast_max_pixels(int p) {
+    screencast_max_pixels_ = talk_base::_max(0, p);
+  }
+
  protected:
   // Callback attached to SignalFrameCaptured where SignalVideoFrames is called.
   void OnFrameCaptured(VideoCapturer* video_capturer,
@@ -311,8 +329,10 @@ class VideoCapturer
   int ratio_w_;  // View resolution. e.g. 1280 x 720.
   int ratio_h_;
   bool enable_camera_list_;
+  bool square_pixel_aspect_ratio_;  // Enable scaling to square pixels.
   int scaled_width_;  // Current output size from ComputeScale.
   int scaled_height_;
+  int screencast_max_pixels_;  // Downscale screencasts further if requested.
   bool muted_;
   int black_frame_count_down_;
 
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videocapturer_unittest.cc b/chromium/third_party/libjingle/source/talk/media/base/videocapturer_unittest.cc
index a6ce3ba9b31..82a95fb637d 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videocapturer_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/videocapturer_unittest.cc
@@ -194,6 +194,40 @@ TEST_F(VideoCapturerTest, CameraOffOnMute) {
   EXPECT_EQ(33, video_frames_received());
 }
 
+TEST_F(VideoCapturerTest, ScreencastScaledMaxPixels) {
+  capturer_.SetScreencast(true);
+
+  int kWidth = 1280;
+  int kHeight = 720;
+
+  // Screencasts usually have large weird dimensions and are ARGB.
+  std::vector<cricket::VideoFormat> formats;
+  formats.push_back(cricket::VideoFormat(kWidth, kHeight,
+      cricket::VideoFormat::FpsToInterval(5), cricket::FOURCC_ARGB));
+  formats.push_back(cricket::VideoFormat(2 * kWidth, 2 * kHeight,
+      cricket::VideoFormat::FpsToInterval(5), cricket::FOURCC_ARGB));
+  capturer_.ResetSupportedFormats(formats);
+
+
+  EXPECT_EQ(0, capturer_.screencast_max_pixels());
+  EXPECT_EQ(cricket::CS_RUNNING, capturer_.Start(cricket::VideoFormat(
+      2 * kWidth,
+      2 * kHeight,
+      cricket::VideoFormat::FpsToInterval(30),
+      cricket::FOURCC_ARGB)));
+  EXPECT_TRUE(capturer_.IsRunning());
+  EXPECT_EQ(0, renderer_.num_rendered_frames());
+  renderer_.SetSize(2 * kWidth, 2 * kHeight, 0);
+  EXPECT_TRUE(capturer_.CaptureFrame());
+  EXPECT_EQ(1, renderer_.num_rendered_frames());
+
+  capturer_.set_screencast_max_pixels(kWidth * kHeight);
+  renderer_.SetSize(kWidth, kHeight, 0);
+  EXPECT_TRUE(capturer_.CaptureFrame());
+  EXPECT_EQ(2, renderer_.num_rendered_frames());
+}
+
+
 TEST_F(VideoCapturerTest, TestFourccMatch) {
   cricket::VideoFormat desired(640, 480,
                                cricket::VideoFormat::FpsToInterval(30),
@@ -681,3 +715,29 @@ TEST_F(VideoCapturerTest, Whitelist) {
   capturer_.ConstrainSupportedFormats(vga_format);
   EXPECT_TRUE(HdFormatInList(*capturer_.GetSupportedFormats()));
 }
+
+TEST_F(VideoCapturerTest, BlacklistAllFormats) {
+  cricket::VideoFormat vga_format(640, 480,
+                                  cricket::VideoFormat::FpsToInterval(30),
+                                  cricket::FOURCC_I420);
+  std::vector<cricket::VideoFormat> supported_formats;
+  // Mock a device that only supports HD formats.
+  supported_formats.push_back(cricket::VideoFormat(1280, 720,
+      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
+  supported_formats.push_back(cricket::VideoFormat(1920, 1080,
+      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
+  capturer_.ResetSupportedFormats(supported_formats);
+  EXPECT_EQ(2u, capturer_.GetSupportedFormats()->size());
+  // Now, enable the list, which would exclude both formats. However, since
+  // only HD formats are available, we refuse to filter at all, so we don't
+  // break this camera.
+  capturer_.set_enable_camera_list(true);
+  capturer_.ConstrainSupportedFormats(vga_format);
+  EXPECT_EQ(2u, capturer_.GetSupportedFormats()->size());
+  // To make sure it's not just the camera list being broken, add in VGA and
+  // try again. This time, only the VGA format should be there.
+  supported_formats.push_back(vga_format);
+  capturer_.ResetSupportedFormats(supported_formats);
+  ASSERT_EQ(1u, capturer_.GetSupportedFormats()->size());
+  EXPECT_EQ(vga_format.height, capturer_.GetSupportedFormats()->at(0).height);
+}
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videocommon.cc b/chromium/third_party/libjingle/source/talk/media/base/videocommon.cc
index 12f3bb36886..5dd45d73075 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videocommon.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/videocommon.cc
@@ -98,23 +98,17 @@ static float FindLowerScale(int width, int height, int target_num_pixels) {
   return kScaleFactors[best_index];
 }
 
-// Compute a size to scale frames to that is below maximum compression
-// and rendering size with the same aspect ratio.
-void ComputeScale(int frame_width, int frame_height, int fps,
-                  int* scaled_width, int* scaled_height) {
+// Computes a scale less to fit in max_pixels while maintaining aspect ratio.
+void ComputeScaleMaxPixels(int frame_width, int frame_height, int max_pixels,
+    int* scaled_width, int* scaled_height) {
   ASSERT(scaled_width != NULL);
   ASSERT(scaled_height != NULL);
+  ASSERT(max_pixels > 0);
   // For VP8 the values for max width and height can be found here
   // webrtc/src/video_engine/vie_defines.h (kViEMaxCodecWidth and
   // kViEMaxCodecHeight)
   const int kMaxWidth = 4096;
   const int kMaxHeight = 3072;
-  // Maximum pixels limit is set to Retina MacBookPro 15" resolution of
-  // 2880 x 1800 as of 4/18/2013.
-  // For high fps, maximum pixels limit is set based on common 24" monitor
-  // resolution of 2048 x 1280 as of 6/13/2013. The Retina resolution is
-  // therefore reduced to 1440 x 900.
-  int kMaxPixels = (fps > 5) ? 2048 * 1280 : 2880 * 1800;
   int new_frame_width = frame_width;
   int new_frame_height = frame_height;
 
@@ -129,12 +123,12 @@ void ComputeScale(int frame_width, int frame_height, int fps,
     new_frame_height = kMaxHeight;
   }
   // Limit number of pixels.
-  if (new_frame_width * new_frame_height > kMaxPixels) {
+  if (new_frame_width * new_frame_height > max_pixels) {
     // Compute new width such that width * height is less than maximum but
     // maintains original captured frame aspect ratio.
     new_frame_width = static_cast<int>(sqrtf(static_cast<float>(
-        kMaxPixels) * new_frame_width / new_frame_height));
-    new_frame_height = kMaxPixels / new_frame_width;
+        max_pixels) * new_frame_width / new_frame_height));
+    new_frame_height = max_pixels / new_frame_width;
   }
   // Snap to a scale factor that is less than or equal to target pixels.
   float scale = FindLowerScale(frame_width, frame_height,
@@ -143,6 +137,20 @@ void ComputeScale(int frame_width, int frame_height, int fps,
   *scaled_height = static_cast<int>(frame_height * scale + .5f);
 }
 
+// Compute a size to scale frames to that is below maximum compression
+// and rendering size with the same aspect ratio.
+void ComputeScale(int frame_width, int frame_height, int fps,
+                  int* scaled_width, int* scaled_height) {
+  // Maximum pixels limit is set to Retina MacBookPro 15" resolution of
+  // 2880 x 1800 as of 4/18/2013.
+  // For high fps, maximum pixels limit is set based on common 24" monitor
+  // resolution of 2048 x 1280 as of 6/13/2013. The Retina resolution is
+  // therefore reduced to 1440 x 900.
+  int max_pixels = (fps > 5) ? 2048 * 1280 : 2880 * 1800;
+  ComputeScaleMaxPixels(
+      frame_width, frame_height, max_pixels, scaled_width, scaled_height);
+}
+
 // Compute size to crop video frame to.
 // If cropped_format_* is 0, return the frame_* size as is.
 void ComputeCrop(int cropped_format_width,
@@ -209,6 +217,14 @@ void ComputeCrop(int cropped_format_width,
   }
 }
 
+// Compute the frame size that makes pixels square pixel aspect ratio.
+void ComputeScaleToSquarePixels(int in_width, int in_height,
+                                int pixel_width, int pixel_height,
+                                int* scaled_width, int* scaled_height) {
+  *scaled_width = in_width;  // Keep width the same.
+  *scaled_height = in_height * pixel_height / pixel_width;
+}
+
 // The C++ standard requires a namespace-scope definition of static const
 // integral types even when they are initialized in the declaration (see
 // [class.static.data]/4), but MSVC with /Ze is non-conforming and treats that
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videocommon.h b/chromium/third_party/libjingle/source/talk/media/base/videocommon.h
index 098651f3df8..cf24f6fbb39 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videocommon.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/videocommon.h
@@ -25,7 +25,7 @@
 //
 // Common definition for video, including fourcc and VideoFormat.
 
-#ifndef TALK_MEDIA_BASE_VIDEOCOMMON_H_
+#ifndef TALK_MEDIA_BASE_VIDEOCOMMON_H_  // NOLINT
 #define TALK_MEDIA_BASE_VIDEOCOMMON_H_
 
 #include <string>
@@ -147,6 +147,15 @@ inline std::string GetFourccName(uint32 fourcc) {
   return name;
 }
 
+// Computes a scale less to fit in max_pixels while maintaining aspect ratio.
+void ComputeScaleMaxPixels(int frame_width, int frame_height, int max_pixels,
+                           int* scaled_width, int* scaled_height);
+
+// For low fps, max pixels limit is set to Retina MacBookPro 15" resolution of
+// 2880 x 1800 as of 4/18/2013.
+// For high fps, maximum pixels limit is set based on common 24" monitor
+// resolution of 2048 x 1280 as of 6/13/2013. The Retina resolution is
+// therefore reduced to 1440 x 900.
 void ComputeScale(int frame_width, int frame_height, int fps,
                   int* scaled_width, int* scaled_height);
 
@@ -158,6 +167,11 @@ void ComputeCrop(int cropped_format_width, int cropped_format_height,
                  int rotation,
                  int* cropped_width, int* cropped_height);
 
+// Compute the frame size that makes pixels square pixel aspect ratio.
+void ComputeScaleToSquarePixels(int in_width, int in_height,
+                                int pixel_width, int pixel_height,
+                                int* scaled_width, int* scaled_height);
+
 //////////////////////////////////////////////////////////////////////////////
 // Definition of VideoFormat.
 //////////////////////////////////////////////////////////////////////////////
@@ -239,4 +253,4 @@ struct VideoFormat : VideoFormatPod {
 
 }  // namespace cricket
 
-#endif  // TALK_MEDIA_BASE_VIDEOCOMMON_H_
+#endif  // TALK_MEDIA_BASE_VIDEOCOMMON_H_  // NOLINT
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videocommon_unittest.cc b/chromium/third_party/libjingle/source/talk/media/base/videocommon_unittest.cc
index e9cd26a1729..91228437351 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videocommon_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/base/videocommon_unittest.cc
@@ -287,4 +287,26 @@ TEST(VideoCommonTest, TestComputeCrop) {
   EXPECT_EQ(768, cropped_height);
 }
 
+TEST(VideoCommonTest, TestComputeScaleToSquarePixels) {
+  int scaled_width, scaled_height;
+
+  // Pixel aspect ratio is 4:3.  Logical aspect ratio is 16:9.  Expect scale
+  // to square pixels with physical aspect ratio of 16:9.
+  ComputeScaleToSquarePixels(640, 480,
+                             4, 3,  // 4 x 3 pixel aspect ratio
+                             &scaled_width, &scaled_height);
+  EXPECT_EQ(640, scaled_width);
+  EXPECT_EQ(360, scaled_height);
+
+  // Pixel aspect ratio is 3:8.  Physical aspect ratio is 4:3.  Expect scale
+  // to square pixels with logical aspect ratio of 1:2.
+  // Note that 640x1280 will be scaled down by video adapter to view request
+  // of 640*360 and will end up using 320x640.
+  ComputeScaleToSquarePixels(640, 480,
+                             3, 8,  // 4 x 3 pixel aspect ratio
+                             &scaled_width, &scaled_height);
+  EXPECT_EQ(640, scaled_width);
+  EXPECT_EQ(1280, scaled_height);
+}
+
 }  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h b/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h
index e9f7612dfa2..d8b9bcb2955 100644
--- a/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h
+++ b/chromium/third_party/libjingle/source/talk/media/base/videoengine_unittest.h
@@ -23,7 +23,7 @@
 // OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
 // ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 
-#ifndef TALK_MEDIA_BASE_VIDEOENGINE_UNITTEST_H_
+#ifndef TALK_MEDIA_BASE_VIDEOENGINE_UNITTEST_H_  // NOLINT
 #define TALK_MEDIA_BASE_VIDEOENGINE_UNITTEST_H_
 
 #include <string>
@@ -1373,8 +1373,10 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_EQ(0, renderer.num_rendered_frames());
 
     EXPECT_TRUE(SendFrame());
-    EXPECT_FRAME_ON_RENDERER_WAIT(renderer, 1, codec.width, codec.height,
-                                  kTimeout);
+    EXPECT_TRUE_WAIT(renderer.num_rendered_frames() >= 1 &&
+                     codec.width == renderer.width() &&
+                     codec.height == renderer.height(), kTimeout);
+    EXPECT_EQ(0, renderer.errors());
 
     // Registering an external capturer is currently the same as screen casting
     // (update the test when this changes).
@@ -1392,8 +1394,8 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_TRUE(capturer->CaptureCustomFrame(kWidth, kHeight,
                                              cricket::FOURCC_ARGB));
     EXPECT_TRUE(capturer->CaptureFrame());
-    EXPECT_EQ_WAIT(2, renderer.num_rendered_frames(), kTimeout);
-    EXPECT_TRUE_WAIT(kScaledWidth == renderer.width() &&
+    EXPECT_TRUE_WAIT(renderer.num_rendered_frames() >= 2 &&
+                     kScaledWidth == renderer.width() &&
                      kScaledHeight == renderer.height(), kTimeout);
     EXPECT_TRUE(channel_->SetCapturer(kSsrc, NULL));
   }
@@ -1665,4 +1667,4 @@ class VideoMediaChannelTest : public testing::Test,
   cricket::FakeVideoRenderer renderer2_;
 };
 
-#endif  // TALK_MEDIA_BASE_VIDEOENGINE_UNITTEST_H_
+#endif  // TALK_MEDIA_BASE_VIDEOENGINE_UNITTEST_H_  NOLINT
diff --git a/chromium/third_party/libjingle/source/talk/media/devices/devicemanager.cc b/chromium/third_party/libjingle/source/talk/media/devices/devicemanager.cc
index 2ce5eb08b82..6f4aa33ffca 100644
--- a/chromium/third_party/libjingle/source/talk/media/devices/devicemanager.cc
+++ b/chromium/third_party/libjingle/source/talk/media/devices/devicemanager.cc
@@ -149,9 +149,9 @@ bool DeviceManager::GetAudioOutputDevice(const std::string& name, Device* out) {
 
 bool DeviceManager::GetVideoCaptureDevices(std::vector<Device>* devices) {
   devices->clear();
-#if defined(IOS)
-  // On iOS, we treat the camera(s) as a single device. Even if there are
-  // multiple cameras, that's abstracted away at a higher level.
+#if defined(ANDROID) || defined(IOS)
+  // On Android and iOS, we treat the camera(s) as a single device. Even if
+  // there are multiple cameras, that's abstracted away at a higher level.
   Device dev("camera", "1");    // name and ID
   devices->push_back(dev);
   return true;
diff --git a/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.cc b/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.cc
index 2f9b117b913..3d450c6863b 100644
--- a/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.cc
+++ b/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.cc
@@ -64,28 +64,58 @@ struct SctpInboundPacket {
   int flags;
 };
 
-// Helper for logging SCTP data. Given a buffer, returns a readable string.
+// Helper for logging SCTP messages.
 static void debug_sctp_printf(const char *format, ...) {
   char s[255];
   va_list ap;
   va_start(ap, format);
   vsnprintf(s, sizeof(s), format, ap);
-  LOG(LS_INFO) << s;
-  // vprintf(format, ap);
+  LOG(LS_INFO) << "SCTP: " << s;
   va_end(ap);
 }
 
-// Helper for make a string dump of some SCTP data. Used for LOG
-// debugging messages.
-static std::string SctpDataToDebugString(void* buffer, size_t length,
-                                         int dump_type) {
-  char *dump_buf = usrsctp_dumppacket(buffer, length, dump_type);
-  if (!dump_buf) {
-      return "";
-  }
-  std::string s = std::string(dump_buf);
-  usrsctp_freedumpbuffer(dump_buf);
-  return s;
+// Get the PPID to use for the terminating fragment of this type.
+static SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid(
+    cricket::DataMessageType type) {
+  switch (type) {
+  default:
+  case cricket::DMT_NONE:
+    return SctpDataMediaChannel::PPID_NONE;
+  case cricket::DMT_CONTROL:
+    return SctpDataMediaChannel::PPID_CONTROL;
+  case cricket::DMT_BINARY:
+    return SctpDataMediaChannel::PPID_BINARY_LAST;
+  case cricket::DMT_TEXT:
+    return SctpDataMediaChannel::PPID_TEXT_LAST;
+  };
+}
+
+static bool GetDataMediaType(
+    SctpDataMediaChannel::PayloadProtocolIdentifier ppid,
+    cricket::DataMessageType *dest) {
+  ASSERT(dest != NULL);
+  switch (ppid) {
+    case SctpDataMediaChannel::PPID_BINARY_PARTIAL:
+    case SctpDataMediaChannel::PPID_BINARY_LAST:
+      *dest = cricket::DMT_BINARY;
+      return true;
+
+    case SctpDataMediaChannel::PPID_TEXT_PARTIAL:
+    case SctpDataMediaChannel::PPID_TEXT_LAST:
+      *dest = cricket::DMT_TEXT;
+      return true;
+
+    case SctpDataMediaChannel::PPID_CONTROL:
+      *dest = cricket::DMT_CONTROL;
+      return true;
+
+    case SctpDataMediaChannel::PPID_NONE:
+      *dest = cricket::DMT_NONE;
+      return true;
+
+    default:
+      return false;
+  }
 }
 
 // This is the callback usrsctp uses when there's data to send on the network
@@ -96,9 +126,7 @@ static int OnSctpOutboundPacket(void* addr, void* data, size_t length,
   LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
                   << "addr: " << addr << "; length: " << length
                   << "; tos: " << std::hex << static_cast<int>(tos)
-                  << "; set_df: " << std::hex << static_cast<int>(set_df)
-                  << "; data:" << SctpDataToDebugString(data, length,
-                                                        SCTP_DUMP_OUTBOUND);
+                  << "; set_df: " << std::hex << static_cast<int>(set_df);
   // Note: We have to copy the data; the caller will delete it.
   talk_base::Buffer* buffer = new talk_base::Buffer(data, length);
   channel->worker_thread()->Post(channel, MSG_SCTPOUTBOUNDPACKET,
@@ -114,37 +142,29 @@ static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr,
                                void* data, size_t length,
                                struct sctp_rcvinfo rcv, int flags,
                                void* ulp_info) {
-  LOG(LS_VERBOSE) << "global OnSctpInboundPacket... Msg of length "
-                  << length << " received via " << addr.sconn.sconn_addr << ":"
-                  << talk_base::NetworkToHost16(addr.sconn.sconn_port)
-                  << " on stream " << rcv.rcv_sid
-                  << " with SSN " << rcv.rcv_ssn
-                  << " and TSN " << rcv.rcv_tsn << ", PPID "
-                  << talk_base::NetworkToHost32(rcv.rcv_ppid)
-                  << ", context " << rcv.rcv_context
-                  << ", data: " << data
-                  << ", ulp_info:" << ulp_info
-                  << ", flags:" << std::hex << flags;
   SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(ulp_info);
-  // The second log call is useful when the defines flags are incorrect. In
-  // this case, ulp_info ends up being bad and the second log message will
-  // cause a crash.
-  LOG(LS_VERBOSE) << "global OnSctpInboundPacket. channel="
-                  << channel->debug_name() << "...";
   // Post data to the channel's receiver thread (copying it).
   // TODO(ldixon): Unclear if copy is needed as this method is responsible for
   // memory cleanup. But this does simplify code.
-  const uint32 native_ppid = talk_base::HostToNetwork32(rcv.rcv_ppid);
-  SctpInboundPacket* packet = new SctpInboundPacket();
-  packet->buffer.SetData(data, length);
-  packet->params.ssrc = rcv.rcv_sid;
-  packet->params.seq_num = rcv.rcv_ssn;
-  packet->params.timestamp = rcv.rcv_tsn;
-  packet->params.type =
-      static_cast<cricket::DataMessageType>(native_ppid);
-  packet->flags = flags;
-  channel->worker_thread()->Post(channel, MSG_SCTPINBOUNDPACKET,
-                                 talk_base::WrapMessageData(packet));
+  const SctpDataMediaChannel::PayloadProtocolIdentifier ppid =
+      static_cast<SctpDataMediaChannel::PayloadProtocolIdentifier>(
+          talk_base::HostToNetwork32(rcv.rcv_ppid));
+  cricket::DataMessageType type = cricket::DMT_NONE;
+  if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) {
+    // It's neither a notification nor a recognized data packet.  Drop it.
+    LOG(LS_ERROR) << "Received an unknown PPID " << ppid
+                  << " on an SCTP packet.  Dropping.";
+  } else {
+    SctpInboundPacket* packet = new SctpInboundPacket;
+    packet->buffer.SetData(data, length);
+    packet->params.ssrc = rcv.rcv_sid;
+    packet->params.seq_num = rcv.rcv_ssn;
+    packet->params.timestamp = rcv.rcv_tsn;
+    packet->params.type = type;
+    packet->flags = flags;
+    channel->worker_thread()->Post(channel, MSG_SCTPINBOUNDPACKET,
+                                   talk_base::WrapMessageData(packet));
+  }
   free(data);
   return 1;
 }
@@ -181,6 +201,14 @@ SctpDataEngine::SctpDataEngine() {
     // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
     // See: http://svnweb.freebsd.org/base?view=revision&revision=229805
     // usrsctp_sysctl_set_sctp_blackhole(2);
+
+    // Set the number of default outgoing streams.  This is the number we'll
+    // send in the SCTP INIT message.  The 'appropriate default' in the
+    // second paragraph of
+    // http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2
+    // is cricket::kMaxSctpSid.
+    usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(
+        cricket::kMaxSctpSid);
   }
   usrsctp_engines_count++;
 
@@ -270,6 +298,13 @@ bool SctpDataMediaChannel::OpenSctpSocket() {
     return false;
   }
 
+  uint32_t nodelay = 1;
+  if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay,
+                         sizeof(nodelay))) {
+    LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_NODELAY.";
+    return false;
+  }
+
   // Subscribe to SCTP event notifications.
   int event_types[] = {SCTP_ASSOC_CHANGE,
                        SCTP_PEER_ADDR_CHANGE,
@@ -430,7 +465,8 @@ bool SctpDataMediaChannel::SendData(
     const talk_base::Buffer& payload,
     SendDataResult* result) {
   if (result) {
-    // If we return true, we'll set this to SDR_SUCCESS.
+    // Preset |result| to assume an error.  If SendData succeeds, we'll
+    // overwrite |*result| once more at the end.
     *result = SDR_ERROR;
   }
 
@@ -450,41 +486,36 @@ bool SctpDataMediaChannel::SendData(
     return false;
   }
 
-  // TODO(ldixon): Experiment with sctp_sendv_spa instead of sctp_sndinfo. e.g.
-  // struct sctp_sendv_spa spa = {0};
-  // spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID;
-  // spa.sendv_sndinfo.snd_sid = params.ssrc;
-  // spa.sendv_sndinfo.snd_context = 0;
-  // spa.sendv_sndinfo.snd_assoc_id = 0;
-  // TODO(pthatcher): Support different types of protocols (e.g. SSL) and
-  // messages (e.g. Binary) via SendDataParams.
-  // spa.sendv_sndinfo.snd_ppid = htonl(PPID_NONE);
-  // TODO(pthatcher): Support different reliability semantics.
-  // For reliable: Remove SCTP_UNORDERED.
-  // For partially-reliable: Add rtx or ttl.
-  // spa.sendv_sndinfo.snd_flags = SCTP_UNORDERED;
-  // TODO(phatcher): Try some of these things.
-  // spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
-  // spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX;
-  // spa.sendv_prinfo.pr_value = htons(max_retransmit_count);
-  // spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL;
-  // spa.sendv_prinfo.pr_value = htons(max_retransmit_time);
   //
   // Send data using SCTP.
-  sctp_sndinfo sndinfo = {0};
-  sndinfo.snd_sid = params.ssrc;
-  sndinfo.snd_flags = 0;
-  // TODO(pthatcher): Once data types are added to SendParams, this can be set
-  // from SendParams.
-  sndinfo.snd_ppid = talk_base::HostToNetwork32(params.type);
-  sndinfo.snd_context = 0;
-  sndinfo.snd_assoc_id = 0;
-  ssize_t res = usrsctp_sendv(sock_, payload.data(),
-                              static_cast<size_t>(payload.length()),
-                              NULL, 0, &sndinfo,
-                              static_cast<socklen_t>(sizeof(sndinfo)),
-                              SCTP_SENDV_SNDINFO, 0);
-  if (res < 0) {
+  ssize_t send_res = 0;  // result from usrsctp_sendv.
+  struct sctp_sendv_spa spa = {0};
+  spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID;
+  spa.sendv_sndinfo.snd_sid = params.ssrc;
+  spa.sendv_sndinfo.snd_ppid = talk_base::HostToNetwork32(
+      GetPpid(params.type));
+
+  // Ordered implies reliable.
+  if (!params.ordered) {
+    spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED;
+    if (params.max_rtx_count >= 0 || params.max_rtx_ms == 0) {
+      spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
+      spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX;
+      spa.sendv_prinfo.pr_value = params.max_rtx_count;
+    } else {
+      spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
+      spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL;
+      spa.sendv_prinfo.pr_value = params.max_rtx_ms;
+    }
+  }
+
+  // We don't fragment.
+  send_res = usrsctp_sendv(sock_, payload.data(),
+                           static_cast<size_t>(payload.length()),
+                           NULL, 0, &spa,
+                           static_cast<socklen_t>(sizeof(spa)),
+                           SCTP_SENDV_SPA, 0);
+  if (send_res < 0) {
     if (errno == EWOULDBLOCK) {
       *result = SDR_BLOCK;
       LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned";
@@ -496,7 +527,7 @@ bool SctpDataMediaChannel::SendData(
     return false;
   }
   if (result) {
-    // If we return true, we'll set this to SDR_SUCCESS.
+    // Only way out now is success.
     *result = SDR_SUCCESS;
   }
   return true;
@@ -504,17 +535,13 @@ bool SctpDataMediaChannel::SendData(
 
 // Called by network interface when a packet has been received.
 void SctpDataMediaChannel::OnPacketReceived(talk_base::Buffer* packet) {
-  LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...): "
-                  << " length=" << packet->length() << "; data="
-                  << SctpDataToDebugString(packet->data(), packet->length(),
-                                           SCTP_DUMP_INBOUND);
+  LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...): " << " length="
+                  << packet->length() << ", sending: " << sending_;
   // Only give receiving packets to usrsctp after if connected. This enables two
   // peers to each make a connect call, but for them not to receive an INIT
   // packet before they have called connect; least the last receiver of the INIT
   // packet will have called connect, and a connection will be established.
   if (sending_) {
-    LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...):"
-                    << " Passed packet to sctp.";
     // Pass received packet to SCTP stack. Once processed by usrsctp, the data
     // will be will be given to the global OnSctpInboundData, and then,
     // marshalled by a Post and handled with OnMessage.
@@ -522,8 +549,6 @@ void SctpDataMediaChannel::OnPacketReceived(talk_base::Buffer* packet) {
   } else {
     // TODO(ldixon): Consider caching the packet for very slightly better
     // reliability.
-    LOG(LS_INFO) << debug_name_ << "->OnPacketReceived(...):"
-                 << " Threw packet (probably an INIT) away.";
   }
 }
 
@@ -532,10 +557,8 @@ void SctpDataMediaChannel::OnInboundPacketFromSctpToChannel(
   LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
                   << "Received SCTP data:"
                   << " ssrc=" << packet->params.ssrc
-                  << " data='" << std::string(packet->buffer.data(),
-                                              packet->buffer.length())
                   << " notification: " << (packet->flags & MSG_NOTIFICATION)
-                  << "' length=" << packet->buffer.length();
+                  << " length=" << packet->buffer.length();
   // Sending a packet with data == NULL (no data) is SCTPs "close the
   // connection" message. This sets sock_ = NULL;
   if (!packet->buffer.length() || !packet->buffer.data()) {
diff --git a/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.h b/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.h
index d62eff1aeaa..429016e0e6a 100644
--- a/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine.h
@@ -54,6 +54,10 @@ struct sctp_assoc_change;
 struct socket;
 
 namespace cricket {
+// The highest stream ID (Sid) that SCTP allows, and the number of streams we
+// tell SCTP we're going to use.
+const uint32 kMaxSctpSid = USHRT_MAX;
+
 // A DataEngine that interacts with usrsctp.
 //
 // From channel calls, data flows like this:
@@ -108,12 +112,14 @@ class SctpDataMediaChannel : public DataMediaChannel,
   // on top of SCTP.
   enum PayloadProtocolIdentifier {
     PPID_NONE = 0,  // No protocol is specified.
-    // Specified by Mozilla. Not clear that this is actually part of the
-    // standard. Use with caution!
-    // http://mxr.mozilla.org/mozilla-central/source/netwerk/sctp/datachannel/DataChannelProtocol.h#22
+    // Matches the PPIDs in mozilla source and
+    // https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol Sec. 9
+    // They're not yet assigned by IANA.
     PPID_CONTROL = 50,
-    PPID_TEXT = 51,
-    PPID_BINARY = 52,
+    PPID_BINARY_PARTIAL = 52,
+    PPID_BINARY_LAST = 53,
+    PPID_TEXT_PARTIAL = 54,
+    PPID_TEXT_LAST = 51
   };
 
   // Given a thread which will be used to post messages (received data) to this
@@ -208,11 +214,7 @@ class SctpDataMediaChannel : public DataMediaChannel,
   // related to the ports at the IP level.
   int local_port_;
   int remote_port_;
-  // TODO(ldixon): investigate why removing 'struct' makes the compiler
-  // complain.
-  //
-  // The socket created by usrsctp_socket(...).
-  struct socket* sock_;
+  struct socket* sock_;  // The socket created by usrsctp_socket(...).
 
   // sending_ is true iff there is a connected socket.
   bool sending_;
diff --git a/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine_unittest.cc b/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine_unittest.cc
index 2b8787f3aac..363e7dfeb64 100644
--- a/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/sctp/sctpdataengine_unittest.cc
@@ -60,7 +60,8 @@ class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
 
  protected:
   // Called to send raw packet down the wire (e.g. SCTP an packet).
-  virtual bool SendPacket(talk_base::Buffer* packet) {
+  virtual bool SendPacket(talk_base::Buffer* packet,
+                          talk_base::DiffServCodePoint dscp) {
     LOG(LS_VERBOSE) << "SctpFakeNetworkInterface::SendPacket";
 
     // TODO(ldixon): Can/should we use Buffer.TransferTo here?
@@ -89,7 +90,8 @@ class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
   // Unsupported functions required to exist by NetworkInterface.
   // TODO(ldixon): Refactor parent NetworkInterface class so these are not
   // required. They are RTC specific and should be in an appropriate subclass.
-  virtual bool SendRtcp(talk_base::Buffer* packet) {
+  virtual bool SendRtcp(talk_base::Buffer* packet,
+                        talk_base::DiffServCodePoint dscp) {
     LOG(LS_WARNING) << "Unsupported: SctpFakeNetworkInterface::SendRtcp.";
     return false;
   }
@@ -98,6 +100,9 @@ class SctpFakeNetworkInterface : public cricket::MediaChannel::NetworkInterface,
     LOG(LS_WARNING) << "Unsupported: SctpFakeNetworkInterface::SetOption.";
     return 0;
   }
+  virtual void SetDefaultDSCPCode(talk_base::DiffServCodePoint dscp) {
+    LOG(LS_WARNING) << "Unsupported: SctpFakeNetworkInterface::SetOption.";
+  }
 
  private:
   // Not owned by this class.
diff --git a/chromium/third_party/libjingle/source/talk/media/testdata/1.frame_plus_1.byte b/chromium/third_party/libjingle/source/talk/media/testdata/1.frame_plus_1.byte
deleted file mode 100644
index b619edeed23..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/testdata/1.frame_plus_1.byte
+++ /dev/null
diff --git a/chromium/third_party/libjingle/source/talk/media/testdata/captured-320x240-2s-48.frames b/chromium/third_party/libjingle/source/talk/media/testdata/captured-320x240-2s-48.frames
deleted file mode 100644
index 292a1700f15..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/testdata/captured-320x240-2s-48.frames
+++ /dev/null
diff --git a/chromium/third_party/libjingle/source/talk/media/testdata/h264-svc-99-640x360.rtpdump b/chromium/third_party/libjingle/source/talk/media/testdata/h264-svc-99-640x360.rtpdump
deleted file mode 100644
index ffa521df066..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/testdata/h264-svc-99-640x360.rtpdump
+++ /dev/null
diff --git a/chromium/third_party/libjingle/source/talk/media/testdata/video.rtpdump b/chromium/third_party/libjingle/source/talk/media/testdata/video.rtpdump
deleted file mode 100644
index 7be863e5013..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/testdata/video.rtpdump
+++ /dev/null
diff --git a/chromium/third_party/libjingle/source/talk/media/testdata/voice.rtpdump b/chromium/third_party/libjingle/source/talk/media/testdata/voice.rtpdump
deleted file mode 100644
index 8f0ec15c996..00000000000
--- a/chromium/third_party/libjingle/source/talk/media/testdata/voice.rtpdump
+++ /dev/null
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h
index df539048105..31de172a211 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -787,10 +787,6 @@ class FakeWebRtcVideoEngine
   // Not using WEBRTC_STUB due to bool return value
   virtual bool IsIPv6Enabled(int channel) { return true; }
   WEBRTC_STUB(SetMTU, (int, unsigned int));
-  WEBRTC_STUB(SetPacketTimeoutNotification, (const int, bool, int));
-  WEBRTC_STUB(RegisterObserver, (const int, webrtc::ViENetworkObserver&));
-  WEBRTC_STUB(SetPeriodicDeadOrAliveStatus, (const int, const bool,
-    const unsigned int));
 
   // webrtc::ViERender
   WEBRTC_STUB(RegisterVideoRenderModule, (webrtc::VideoRender&));
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h
index 4ed38d1d3f6..c3cd786009d 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -266,9 +266,6 @@ class FakeWebRtcVoiceEngine
   virtual webrtc::AudioProcessing* audio_processing() OVERRIDE {
     return NULL;
   }
-#ifndef USE_WEBRTC_DEV_BRANCH
-  WEBRTC_STUB(MaxNumOfChannels, ());
-#endif
   WEBRTC_FUNC(CreateChannel, ()) {
     if (fail_create_channel_) {
       return -1;
@@ -619,18 +616,6 @@ class FakeWebRtcVoiceEngine
   }
   WEBRTC_STUB(ReceivedRTCPPacket, (int channel, const void* data,
                                    unsigned int length));
-  // Not using WEBRTC_STUB due to bool return value
-  WEBRTC_STUB(SetPacketTimeoutNotification, (int channel, bool enable,
-                                             int timeoutSeconds));
-  WEBRTC_STUB(GetPacketTimeoutNotification, (int channel, bool& enable,
-                                             int& timeoutSeconds));
-  WEBRTC_STUB(RegisterDeadOrAliveObserver, (int channel,
-      webrtc::VoEConnectionObserver& observer));
-  WEBRTC_STUB(DeRegisterDeadOrAliveObserver, (int channel));
-  WEBRTC_STUB(GetPeriodicDeadOrAliveStatus, (int channel, bool& enabled,
-                                             int& sampleTimeSeconds));
-  WEBRTC_STUB(SetPeriodicDeadOrAliveStatus, (int channel, bool enable,
-                                             int sampleTimeSeconds));
 
   // webrtc::VoERTP_RTCP
   WEBRTC_STUB(RegisterRTPObserver, (int channel,
@@ -751,7 +736,7 @@ class FakeWebRtcVoiceEngine
   // webrtc::VoEVideoSync
   WEBRTC_STUB(GetPlayoutBufferSize, (int& bufferMs));
   WEBRTC_STUB(GetPlayoutTimestamp, (int channel, unsigned int& timestamp));
-  WEBRTC_STUB(GetRtpRtcp, (int, webrtc::RtpRtcp*&));
+  WEBRTC_STUB(GetRtpRtcp, (int, webrtc::RtpRtcp**, webrtc::RtpReceiver**));
   WEBRTC_STUB(SetInitTimestamp, (int channel, unsigned int timestamp));
   WEBRTC_STUB(SetInitSequenceNumber, (int channel, short sequenceNumber));
   WEBRTC_STUB(SetMinimumPlayoutDelay, (int channel, int delayMs));
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtctexturevideoframe.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtctexturevideoframe.h
index 05b50f74c28..691c8142558 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtctexturevideoframe.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtctexturevideoframe.h
@@ -31,14 +31,7 @@
 #include "talk/base/refcount.h"
 #include "talk/base/scoped_ref_ptr.h"
 #include "talk/media/base/videoframe.h"
-#ifdef USE_WEBRTC_DEV_BRANCH
 #include "webrtc/common_video/interface/native_handle.h"
-#else
-#include "webrtc/common_video/interface/i420_video_frame.h"
-// Define NativeHandle to an existing type so we don't need to add lots of
-// USE_WEBRTC_DEV_BRANCH.
-#define NativeHandle I420VideoFrame
-#endif
 
 namespace cricket {
 
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc
index 873b249b70f..fd7e5bfa3f6 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine.cc
@@ -221,9 +221,7 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
 
   virtual int DeliverFrame(unsigned char* buffer, int buffer_size,
                            uint32_t time_stamp, int64_t render_time
-#ifdef USE_WEBRTC_DEV_BRANCH
                            , void* handle
-#endif
                           ) {
     talk_base::CritScope cs(&crit_);
     frame_rate_tracker_.Update(1);
@@ -238,17 +236,13 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
         talk_base::kNumNanosecsPerMillisec;
     // Send the rtp timestamp to renderer as the VideoFrame timestamp.
     // and the render timestamp as the VideoFrame elapsed_time.
-#ifdef USE_WEBRTC_DEV_BRANCH
     if (handle == NULL) {
-#endif
       return DeliverBufferFrame(buffer, buffer_size, render_time_stamp_in_ns,
                                 rtp_time_stamp_in_ns);
-#ifdef USE_WEBRTC_DEV_BRANCH
     } else {
       return DeliverTextureFrame(handle, render_time_stamp_in_ns,
                                  rtp_time_stamp_in_ns);
     }
-#endif
   }
 
   virtual bool IsTextureSupported() { return true; }
@@ -465,6 +459,46 @@ class WebRtcVideoChannelRecvInfo  {
   DecoderMap registered_decoders_;
 };
 
+class WebRtcOveruseObserver : public webrtc::CpuOveruseObserver {
+ public:
+  explicit WebRtcOveruseObserver(CoordinatedVideoAdapter* video_adapter)
+      : video_adapter_(video_adapter),
+        enabled_(false) {
+  }
+
+  // TODO(mflodman): Consider sending resolution as part of event, to let
+  // adapter know what resolution the request is based on. Helps eliminate stale
+  // data, race conditions.
+  virtual void OveruseDetected() OVERRIDE {
+    talk_base::CritScope cs(&crit_);
+    if (!enabled_) {
+      return;
+    }
+
+    video_adapter_->OnCpuResolutionRequest(CoordinatedVideoAdapter::DOWNGRADE);
+  }
+
+  virtual void NormalUsage() OVERRIDE {
+    talk_base::CritScope cs(&crit_);
+    if (!enabled_) {
+      return;
+    }
+
+    video_adapter_->OnCpuResolutionRequest(CoordinatedVideoAdapter::UPGRADE);
+  }
+
+  void Enable(bool enable) {
+    talk_base::CritScope cs(&crit_);
+    enabled_ = enable;
+  }
+
+ private:
+  CoordinatedVideoAdapter* video_adapter_;
+  bool enabled_;
+  talk_base::CriticalSection crit_;
+};
+
+
 class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
  public:
   typedef std::map<int, webrtc::VideoEncoder*> EncoderMap;  // key: payload type
@@ -481,6 +515,7 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
         capturer_updated_(false),
         interval_(0),
         video_adapter_(new CoordinatedVideoAdapter) {
+    overuse_observer_.reset(new WebRtcOveruseObserver(video_adapter_.get()));
     SignalCpuAdaptationUnable.repeat(video_adapter_->SignalCpuAdaptationUnable);
     if (cpu_monitor) {
       cpu_monitor->SignalUpdate.connect(
@@ -534,6 +569,9 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
   int CurrentAdaptReason() const {
     return video_adapter_->adapt_reason();
   }
+  webrtc::CpuOveruseObserver* overuse_observer() {
+    return overuse_observer_.get();
+  }
 
   StreamParams* stream_params() { return stream_params_.get(); }
   void set_stream_params(const StreamParams& sp) {
@@ -572,7 +610,7 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
   }
 
   void ApplyCpuOptions(const VideoOptions& options) {
-    bool cpu_adapt, cpu_smoothing;
+    bool cpu_adapt, cpu_smoothing, adapt_third;
     float low, med, high;
     if (options.adapt_input_to_cpu_usage.Get(&cpu_adapt)) {
       video_adapter_->set_cpu_adaptation(cpu_adapt);
@@ -589,7 +627,16 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
     if (options.system_high_adaptation_threshhold.Get(&high)) {
       video_adapter_->set_high_system_threshold(high);
     }
+    if (options.video_adapt_third.Get(&adapt_third)) {
+      video_adapter_->set_scale_third(adapt_third);
+    }
   }
+
+  void SetCpuOveruseDetection(bool enable) {
+    overuse_observer_->Enable(enable);
+    video_adapter_->set_cpu_adaptation(enable);
+  }
+
   void ProcessFrame(const VideoFrame& original_frame, bool mute,
                     VideoFrame** processed_frame) {
     if (!mute) {
@@ -642,6 +689,7 @@ class WebRtcVideoChannelSendInfo : public sigslot::has_slots<> {
   int64 interval_;
 
   talk_base::scoped_ptr<CoordinatedVideoAdapter> video_adapter_;
+  talk_base::scoped_ptr<WebRtcOveruseObserver> overuse_observer_;
 };
 
 const WebRtcVideoEngine::VideoCodecPref
@@ -1175,7 +1223,7 @@ static void AddDefaultFeedbackParams(VideoCodec* codec) {
 }
 
 // Rebuilds the codec list to be only those that are less intensive
-// than the specified codec.
+// than the specified codec. Prefers internal codec over external.
 bool WebRtcVideoEngine::RebuildCodecList(const VideoCodec& in_codec) {
   if (!FindCodec(in_codec))
     return false;
@@ -1183,32 +1231,12 @@ bool WebRtcVideoEngine::RebuildCodecList(const VideoCodec& in_codec) {
   video_codecs_.clear();
 
   bool found = false;
-  std::set<std::string> external_codec_names;
-  if (encoder_factory_) {
-    const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
-        encoder_factory_->codecs();
-    for (size_t i = 0; i < codecs.size(); ++i) {
-      if (!found)
-        found = (in_codec.name == codecs[i].name);
-      VideoCodec codec(
-          GetExternalVideoPayloadType(static_cast<int>(i)),
-          codecs[i].name,
-          codecs[i].max_width,
-          codecs[i].max_height,
-          codecs[i].max_fps,
-          static_cast<int>(codecs.size() + ARRAY_SIZE(kVideoCodecPrefs) - i));
-      AddDefaultFeedbackParams(&codec);
-      video_codecs_.push_back(codec);
-      external_codec_names.insert(codecs[i].name);
-    }
-  }
+  std::set<std::string> internal_codec_names;
   for (size_t i = 0; i < ARRAY_SIZE(kVideoCodecPrefs); ++i) {
     const VideoCodecPref& pref(kVideoCodecPrefs[i]);
     if (!found)
       found = (in_codec.name == pref.name);
-    bool is_external_codec = external_codec_names.find(pref.name) !=
-        external_codec_names.end();
-    if (found && !is_external_codec) {
+    if (found) {
       VideoCodec codec(pref.payload_type, pref.name,
                        in_codec.width, in_codec.height, in_codec.framerate,
                        static_cast<int>(ARRAY_SIZE(kVideoCodecPrefs) - i));
@@ -1216,6 +1244,28 @@ bool WebRtcVideoEngine::RebuildCodecList(const VideoCodec& in_codec) {
         AddDefaultFeedbackParams(&codec);
       }
       video_codecs_.push_back(codec);
+      internal_codec_names.insert(codec.name);
+    }
+  }
+  if (encoder_factory_) {
+    const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
+        encoder_factory_->codecs();
+    for (size_t i = 0; i < codecs.size(); ++i) {
+      bool is_internal_codec = internal_codec_names.find(codecs[i].name) !=
+          internal_codec_names.end();
+      if (!is_internal_codec) {
+        if (!found)
+          found = (in_codec.name == codecs[i].name);
+        VideoCodec codec(
+            GetExternalVideoPayloadType(static_cast<int>(i)),
+            codecs[i].name,
+            codecs[i].max_width,
+            codecs[i].max_height,
+            codecs[i].max_fps,
+            static_cast<int>(codecs.size() + ARRAY_SIZE(kVideoCodecPrefs) - i));
+        AddDefaultFeedbackParams(&codec);
+        video_codecs_.push_back(codec);
+      }
     }
   }
   ASSERT(found);
@@ -2193,7 +2243,10 @@ bool WebRtcVideoMediaChannel::GetStats(VideoMediaInfo* info) {
 
     unsigned int ssrc;
     // Get receiver statistics and build VideoReceiverInfo, if we have data.
-    if (engine_->vie()->rtp()->GetRemoteSSRC(channel->channel_id(), ssrc) != 0)
+    // Skip the default channel (ssrc == 0).
+    if (engine_->vie()->rtp()->GetRemoteSSRC(
+            channel->channel_id(), ssrc) != 0 ||
+        ssrc == 0)
       continue;
 
     unsigned int bytes_sent, packets_sent, bytes_recv, packets_recv;
@@ -2481,6 +2534,9 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
   bool buffer_latency_changed = options.buffered_mode_latency.IsSet() &&
       (options_.buffered_mode_latency != options.buffered_mode_latency);
 
+  bool cpu_overuse_detection_changed = options.cpu_overuse_detection.IsSet() &&
+      (options_.cpu_overuse_detection != options.cpu_overuse_detection);
+
   bool conference_mode_turned_off = false;
   if (options_.conference_mode.IsSet() && options.conference_mode.IsSet() &&
       options_.conference_mode.GetWithDefaultIfUnset(false) &&
@@ -2558,6 +2614,15 @@ bool WebRtcVideoMediaChannel::SetOptions(const VideoOptions &options) {
       }
     }
   }
+  if (cpu_overuse_detection_changed) {
+    bool cpu_overuse_detection =
+        options_.cpu_overuse_detection.GetWithDefaultIfUnset(false);
+    for (SendChannelMap::iterator iter = send_channels_.begin();
+         iter != send_channels_.end(); ++iter) {
+      WebRtcVideoChannelSendInfo* send_channel = iter->second;
+      send_channel->SetCpuOveruseDetection(cpu_overuse_detection);
+    }
+  }
   return true;
 }
 
@@ -2702,8 +2767,8 @@ bool WebRtcVideoMediaChannel::SendFrame(
   frame_i420.y_pitch = frame_out->GetYPitch();
   frame_i420.u_pitch = frame_out->GetUPitch();
   frame_i420.v_pitch = frame_out->GetVPitch();
-  frame_i420.width = static_cast<unsigned short>(frame_out->GetWidth());
-  frame_i420.height = static_cast<unsigned short>(frame_out->GetHeight());
+  frame_i420.width = static_cast<uint16>(frame_out->GetWidth());
+  frame_i420.height = static_cast<uint16>(frame_out->GetHeight());
 
   int64 timestamp_ntp_ms = 0;
   // TODO(justinlin): Reenable after Windows issues with clock drift are fixed.
@@ -2966,10 +3031,19 @@ bool WebRtcVideoMediaChannel::ConfigureSending(int channel_id,
       new WebRtcVideoChannelSendInfo(channel_id, vie_capture,
                                      external_capture,
                                      engine()->cpu_monitor()));
+  if (engine()->vie()->base()->RegisterCpuOveruseObserver(
+      channel_id, send_channel->overuse_observer())) {
+    LOG_RTCERR1(RegisterCpuOveruseObserver, channel_id);
+    return false;
+  }
   send_channel->ApplyCpuOptions(options_);
   send_channel->SignalCpuAdaptationUnable.connect(this,
       &WebRtcVideoMediaChannel::OnCpuAdaptationUnable);
 
+  if (options_.cpu_overuse_detection.GetWithDefaultIfUnset(false)) {
+    send_channel->SetCpuOveruseDetection(true);
+  }
+
   // Register encoder observer for outgoing framerate and bitrate.
   if (engine()->vie()->codec()->RegisterEncoderObserver(
       channel_id, *send_channel->encoder_observer()) != 0) {
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine_unittest.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine_unittest.cc
index 840fcdd0868..04662caf7c1 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvideoengine_unittest.cc
@@ -86,8 +86,7 @@ class FakeViEWrapper : public cricket::ViEWrapper {
 
 // Test fixture to test WebRtcVideoEngine with a fake webrtc::VideoEngine.
 // Useful for testing failure paths.
-class WebRtcVideoEngineTestFake :
-  public testing::Test,
+class WebRtcVideoEngineTestFake : public testing::Test,
   public sigslot::has_slots<> {
  public:
   WebRtcVideoEngineTestFake()
@@ -1759,18 +1758,57 @@ TEST_F(WebRtcVideoEngineTestFake, FeedbackParamsForNonVP8) {
   EXPECT_TRUE(SetupEngine());
 
   std::vector<cricket::VideoCodec> codecs(engine_.codecs());
-  EXPECT_EQ("GENERIC", codecs[0].name);
-  EXPECT_TRUE(codecs[0].HasFeedbackParam(
+  // The external codec will appear at last.
+  size_t pos = codecs.size() - 1;
+  EXPECT_EQ("GENERIC", codecs[pos].name);
+  EXPECT_TRUE(codecs[pos].HasFeedbackParam(
       cricket::FeedbackParam(cricket::kRtcpFbParamNack,
                              cricket::kParamValueEmpty)));
-  EXPECT_TRUE(codecs[0].HasFeedbackParam(
+  EXPECT_TRUE(codecs[pos].HasFeedbackParam(
       cricket::FeedbackParam(cricket::kRtcpFbParamRemb,
                              cricket::kParamValueEmpty)));
-  EXPECT_TRUE(codecs[0].HasFeedbackParam(
+  EXPECT_TRUE(codecs[pos].HasFeedbackParam(
       cricket::FeedbackParam(cricket::kRtcpFbParamCcm,
                              cricket::kRtcpFbCcmParamFir)));
 }
 
+// Test external codec with be added to the end of the supported codec list.
+TEST_F(WebRtcVideoEngineTestFake, ExternalCodecAddedToTheEnd) {
+  EXPECT_TRUE(SetupEngine());
+
+  std::vector<cricket::VideoCodec> codecs(engine_.codecs());
+  EXPECT_EQ("VP8", codecs[0].name);
+
+  encoder_factory_.AddSupportedVideoCodecType(webrtc::kVideoCodecGeneric,
+                                              "GENERIC");
+  engine_.SetExternalEncoderFactory(&encoder_factory_);
+  encoder_factory_.NotifyCodecsAvailable();
+
+  codecs = engine_.codecs();
+  // The external codec will appear at last.
+  EXPECT_EQ("GENERIC", codecs[codecs.size() - 1].name);
+}
+
+// Test that external codec with be ignored if it has the same name as one of
+// the internal codecs.
+TEST_F(WebRtcVideoEngineTestFake, ExternalCodecIgnored) {
+  EXPECT_TRUE(SetupEngine());
+
+  std::vector<cricket::VideoCodec> internal_codecs(engine_.codecs());
+  EXPECT_EQ("VP8", internal_codecs[0].name);
+
+  encoder_factory_.AddSupportedVideoCodecType(webrtc::kVideoCodecVP8, "VP8");
+  engine_.SetExternalEncoderFactory(&encoder_factory_);
+  encoder_factory_.NotifyCodecsAvailable();
+
+  std::vector<cricket::VideoCodec> codecs = engine_.codecs();
+  EXPECT_EQ("VP8", codecs[0].name);
+  EXPECT_EQ(internal_codecs[0].height, codecs[0].height);
+  EXPECT_EQ(internal_codecs[0].width, codecs[0].width);
+  // Verify the last codec is not the external codec.
+  EXPECT_NE("VP8", codecs[codecs.size() - 1].name);
+}
+
 TEST_F(WebRtcVideoEngineTestFake, UpdateEncoderCodecsAfterSetFactory) {
   engine_.SetExternalEncoderFactory(&encoder_factory_);
   EXPECT_TRUE(SetupEngine());
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc
index 433138562bc..86860389314 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.cc
@@ -125,16 +125,19 @@ static const int kOpusStereoBitrate = 64000;
 static const int kOpusMinBitrate = 6000;
 static const int kOpusMaxBitrate = 510000;
 
-#if defined(CHROMEOS)
-// Ensure we open the file in a writeable path on ChromeOS. This workaround
-// can be removed when it's possible to specify a filename for audio option
-// based AEC dumps.
+// Ensure we open the file in a writeable path on ChromeOS and Android. This
+// workaround can be removed when it's possible to specify a filename for audio
+// option based AEC dumps.
 //
 // TODO(grunell): Use a string in the options instead of hardcoding it here
 // and let the embedder choose the filename (crbug.com/264223).
 //
-// NOTE(ajm): Don't use this hardcoded /tmp path on non-ChromeOS platforms.
+// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
+// below.
+#if defined(CHROMEOS)
 static const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
+#elif defined(ANDROID)
+static const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
 #else
 static const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
 #endif
@@ -1482,6 +1485,7 @@ WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine *engine)
       desired_playout_(false),
       nack_enabled_(false),
       playout_(false),
+      typing_noise_detected_(false),
       desired_send_(SEND_NOTHING),
       send_(SEND_NOTHING),
       default_receive_ssrc_(0) {
@@ -1619,18 +1623,11 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
 }
 
 bool WebRtcVoiceMediaChannel::SetSendCodecs(
-    const std::vector<AudioCodec>& codecs) {
-  // TODO(xians): Break down this function into SetSendCodecs(channel, codecs)
-  // to support per-channel codecs.
-
-  // Disable DTMF, VAD, and FEC unless we know the other side wants them.
-  dtmf_allowed_ = false;
-  for (ChannelMap::iterator iter = send_channels_.begin();
-       iter != send_channels_.end(); ++iter) {
-    engine()->voe()->codec()->SetVADStatus(iter->second.channel, false);
-    engine()->voe()->rtp()->SetNACKStatus(iter->second.channel, false, 0);
-    engine()->voe()->rtp()->SetFECStatus(iter->second.channel, false);
-  }
+    int channel, const std::vector<AudioCodec>& codecs) {
+  // Disable VAD, and FEC unless we know the other side wants them.
+  engine()->voe()->codec()->SetVADStatus(channel, false);
+  engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
+  engine()->voe()->rtp()->SetFECStatus(channel, false);
 
   // Scan through the list to figure out the codec to use for sending, along
   // with the proper configuration for VAD and DTMF.
@@ -1687,16 +1684,11 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
     // about it.
     if (_stricmp(it->name.c_str(), "telephone-event") == 0 ||
         _stricmp(it->name.c_str(), "audio/telephone-event") == 0) {
-      for (ChannelMap::iterator iter = send_channels_.begin();
-           iter != send_channels_.end(); ++iter) {
-        if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
-                iter->second.channel, it->id) == -1) {
-          LOG_RTCERR2(SetSendTelephoneEventPayloadType,
-                      iter->second.channel, it->id);
-          return false;
-        }
+      if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
+              channel, it->id) == -1) {
+        LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, it->id);
+        return false;
       }
-      dtmf_allowed_ = true;
     }
 
     // Turn voice activity detection/comfort noise on if supported.
@@ -1719,35 +1711,30 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
                           << " not supported.";
           continue;
       }
-      // Loop through the existing send channels and set the CN payloadtype
-      // and the VAD status.
-      for (ChannelMap::iterator iter = send_channels_.begin();
-           iter != send_channels_.end(); ++iter) {
-        int channel = iter->second.channel;
-        // The CN payload type for 8000 Hz clockrate is fixed at 13.
-        if (cn_freq != webrtc::kFreq8000Hz) {
-          if (engine()->voe()->codec()->SetSendCNPayloadType(
-                  channel, it->id, cn_freq) == -1) {
-            LOG_RTCERR3(SetSendCNPayloadType, channel, it->id, cn_freq);
-            // TODO(ajm): This failure condition will be removed from VoE.
-            // Restore the return here when we update to a new enough webrtc.
-            //
-            // Not returning false because the SetSendCNPayloadType will fail if
-            // the channel is already sending.
-            // This can happen if the remote description is applied twice, for
-            // example in the case of ROAP on top of JSEP, where both side will
-            // send the offer.
-          }
+      // Set the CN payloadtype and the VAD status.
+      // The CN payload type for 8000 Hz clockrate is fixed at 13.
+      if (cn_freq != webrtc::kFreq8000Hz) {
+        if (engine()->voe()->codec()->SetSendCNPayloadType(
+                channel, it->id, cn_freq) == -1) {
+          LOG_RTCERR3(SetSendCNPayloadType, channel, it->id, cn_freq);
+          // TODO(ajm): This failure condition will be removed from VoE.
+          // Restore the return here when we update to a new enough webrtc.
+          //
+          // Not returning false because the SetSendCNPayloadType will fail if
+          // the channel is already sending.
+          // This can happen if the remote description is applied twice, for
+          // example in the case of ROAP on top of JSEP, where both side will
+          // send the offer.
         }
+      }
 
-        // Only turn on VAD if we have a CN payload type that matches the
-        // clockrate for the codec we are going to use.
-        if (it->clockrate == send_codec.plfreq) {
-          LOG(LS_INFO) << "Enabling VAD";
-          if (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
-            LOG_RTCERR2(SetVADStatus, channel, true);
-            return false;
-          }
+      // Only turn on VAD if we have a CN payload type that matches the
+      // clockrate for the codec we are going to use.
+      if (it->clockrate == send_codec.plfreq) {
+        LOG(LS_INFO) << "Enabling VAD";
+        if (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
+          LOG_RTCERR2(SetVADStatus, channel, true);
+          return false;
         }
       }
     }
@@ -1767,28 +1754,22 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
         // Enable redundant encoding of the specified codec. Treat any
         // failure as a fatal internal error.
         LOG(LS_INFO) << "Enabling FEC";
-        for (ChannelMap::iterator iter = send_channels_.begin();
-             iter != send_channels_.end(); ++iter) {
-          if (engine()->voe()->rtp()->SetFECStatus(iter->second.channel,
-                                                   true, it->id) == -1) {
-            LOG_RTCERR3(SetFECStatus, iter->second.channel, true, it->id);
-            return false;
-          }
+        if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) {
+          LOG_RTCERR3(SetFECStatus, channel, true, it->id);
+          return false;
         }
       } else {
         send_codec = voe_codec;
         nack_enabled_ = IsNackEnabled(*it);
-        SetNack(send_channels_, nack_enabled_);
+        SetNack(channel, nack_enabled_);
       }
       first = false;
       // Set the codec immediately, since SetVADStatus() depends on whether
       // the current codec is mono or stereo.
-      if (!SetSendCodec(send_codec))
+      if (!SetSendCodec(channel, send_codec))
         return false;
     }
   }
-  SetNack(receive_channels_, nack_enabled_);
-
 
   // If we're being asked to set an empty list of codecs, due to a buggy client,
   // choose the most common format: PCMU
@@ -1796,10 +1777,39 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
     LOG(LS_WARNING) << "Received empty list of codecs; using PCMU/8000";
     AudioCodec codec(0, "PCMU", 8000, 0, 1, 0);
     engine()->FindWebRtcCodec(codec, &send_codec);
-    if (!SetSendCodec(send_codec))
+    if (!SetSendCodec(channel, send_codec))
       return false;
   }
 
+  // Always update the |send_codec_| to the currently set send codec.
+  send_codec_.reset(new webrtc::CodecInst(send_codec));
+
+  return true;
+}
+
+bool WebRtcVoiceMediaChannel::SetSendCodecs(
+    const std::vector<AudioCodec>& codecs) {
+  dtmf_allowed_ = false;
+  for (std::vector<AudioCodec>::const_iterator it = codecs.begin();
+       it != codecs.end(); ++it) {
+    // Find the DTMF telephone event "codec".
+    if (_stricmp(it->name.c_str(), "telephone-event") == 0 ||
+        _stricmp(it->name.c_str(), "audio/telephone-event") == 0) {
+      dtmf_allowed_ = true;
+    }
+  }
+
+  // Cache the codecs in order to configure the channel created later.
+  send_codecs_ = codecs;
+  for (ChannelMap::iterator iter = send_channels_.begin();
+       iter != send_channels_.end(); ++iter) {
+    if (!SetSendCodecs(iter->second.channel, codecs)) {
+      return false;
+    }
+  }
+
+  SetNack(receive_channels_, nack_enabled_);
+
   return true;
 }
 
@@ -1807,17 +1817,16 @@ void WebRtcVoiceMediaChannel::SetNack(const ChannelMap& channels,
                                       bool nack_enabled) {
   for (ChannelMap::const_iterator it = channels.begin();
        it != channels.end(); ++it) {
-    SetNack(it->first, it->second.channel, nack_enabled_);
+    SetNack(it->second.channel, nack_enabled);
   }
 }
 
-void WebRtcVoiceMediaChannel::SetNack(uint32 ssrc, int channel,
-                                      bool nack_enabled) {
+void WebRtcVoiceMediaChannel::SetNack(int channel, bool nack_enabled) {
   if (nack_enabled) {
-    LOG(LS_INFO) << "Enabling NACK for stream " << ssrc;
+    LOG(LS_INFO) << "Enabling NACK for channel " << channel;
     engine()->voe()->rtp()->SetNACKStatus(channel, true, kNackMaxPackets);
   } else {
-    LOG(LS_INFO) << "Disabling NACK for stream " << ssrc;
+    LOG(LS_INFO) << "Disabling NACK for channel " << channel;
     engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
   }
 }
@@ -1832,10 +1841,6 @@ bool WebRtcVoiceMediaChannel::SetSendCodec(
       return false;
   }
 
-  // All SetSendCodec calls were successful. Update the global state
-  // accordingly.
-  send_codec_.reset(new webrtc::CodecInst(send_codec));
-
   return true;
 }
 
@@ -2085,8 +2090,8 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
      return false;
   }
 
-  // Set the current codec to be used for the new channel.
-  if (send_codec_ && !SetSendCodec(channel, *send_codec_))
+  // Set the current codecs to be used for the new channel.
+  if (!send_codecs_.empty() && !SetSendCodecs(channel, send_codecs_))
     return false;
 
   return ChangeSend(channel, desired_send_);
@@ -2210,7 +2215,7 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
       SetPlayout(voe_channel(), false);
     }
   }
-  SetNack(ssrc, channel, nack_enabled_);
+  SetNack(channel, nack_enabled_);
 
   receive_channels_.insert(
       std::make_pair(ssrc, WebRtcVoiceChannelInfo(channel, NULL)));
@@ -2534,7 +2539,24 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32 ssrc, int event,
 
   // Send the event.
   if (flags & cricket::DF_SEND) {
-    int channel = (ssrc == 0) ? voe_channel() : GetSendChannelNum(ssrc);
+    int channel = -1;
+    if (ssrc == 0) {
+      bool default_channel_is_inuse = false;
+      for (ChannelMap::const_iterator iter = send_channels_.begin();
+           iter != send_channels_.end(); ++iter) {
+        if (IsDefaultChannel(iter->second.channel)) {
+          default_channel_is_inuse = true;
+          break;
+        }
+      }
+      if (default_channel_is_inuse) {
+        channel = voe_channel();
+      } else if (!send_channels_.empty()) {
+        channel = send_channels_.begin()->second.channel;
+      }
+    } else {
+      channel = GetSendChannelNum(ssrc);
+    }
     if (channel == -1) {
       LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
                       << ssrc << " is not in use.";
@@ -2784,6 +2806,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement;
     sinfo.echo_delay_median_ms = echo_delay_median_ms;
     sinfo.echo_delay_std_ms = echo_delay_std_ms;
+    sinfo.typing_noise_detected = typing_noise_detected_;
 
     info->senders.push_back(sinfo);
   }
@@ -2892,6 +2915,13 @@ bool WebRtcVoiceMediaChannel::FindSsrc(int channel_num, uint32* ssrc) {
 }
 
 void WebRtcVoiceMediaChannel::OnError(uint32 ssrc, int error) {
+#ifdef USE_WEBRTC_DEV_BRANCH
+  if (error == VE_TYPING_NOISE_WARNING) {
+    typing_noise_detected_ = true;
+  } else if (error == VE_TYPING_NOISE_OFF_WARNING) {
+    typing_noise_detected_ = false;
+  }
+#endif
   SignalMediaError(ssrc, WebRtcErrorToChannelError(error));
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h
index 76f5be7af39..62d3bc10a25 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine.h
@@ -388,7 +388,7 @@ class WebRtcVoiceMediaChannel
   };
   typedef std::map<uint32, WebRtcVoiceChannelInfo> ChannelMap;
 
-  void SetNack(uint32 ssrc, int channel, bool nack_enabled);
+  void SetNack(int channel, bool nack_enabled);
   void SetNack(const ChannelMap& channels, bool nack_enabled);
   bool SetSendCodec(const webrtc::CodecInst& send_codec);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
@@ -403,16 +403,19 @@ class WebRtcVoiceMediaChannel
   bool IsDefaultChannel(int channel_id) const {
     return channel_id == voe_channel();
   }
+  bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
 
   talk_base::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
   std::set<int> ringback_channels_;  // channels playing ringback
   std::vector<AudioCodec> recv_codecs_;
+  std::vector<AudioCodec> send_codecs_;
   talk_base::scoped_ptr<webrtc::CodecInst> send_codec_;
   AudioOptions options_;
   bool dtmf_allowed_;
   bool desired_playout_;
   bool nack_enabled_;
   bool playout_;
+  bool typing_noise_detected_;
   SendFlags desired_send_;
   SendFlags send_;
 
diff --git a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 31596cd726d..3710e7ebc93 100644
--- a/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -125,7 +125,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     EXPECT_TRUE(SetupEngine());
     // Remove stream added in Setup, which is corresponding to default channel.
     int default_channel_num = voe_.GetLastChannel();
-    uint32 default_send_ssrc;
+    uint32 default_send_ssrc = 0u;
     EXPECT_EQ(0, voe_.GetLocalSSRC(default_channel_num, default_send_ssrc));
     EXPECT_EQ(kSsrc1, default_send_ssrc);
     EXPECT_TRUE(channel_->RemoveSendStream(default_send_ssrc));
@@ -143,7 +143,18 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     engine_.Terminate();
   }
 
-  void TestInsertDtmf(uint32 ssrc, int channel_id) {
+  void TestInsertDtmf(uint32 ssrc, bool caller) {
+    EXPECT_TRUE(engine_.Init(talk_base::Thread::Current()));
+    channel_ = engine_.CreateChannel();
+    EXPECT_TRUE(channel_ != NULL);
+    if (caller) {
+      // if this is a caller, local description will be applied and add the
+      // send stream.
+      EXPECT_TRUE(channel_->AddSendStream(
+          cricket::StreamParams::CreateLegacy(kSsrc1)));
+    }
+    int channel_id = voe_.GetLastChannel();
+
     // Test we can only InsertDtmf when the other side supports telephone-event.
     std::vector<cricket::AudioCodec> codecs;
     codecs.push_back(kPcmuCodec);
@@ -154,6 +165,14 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     codecs.push_back(kTelephoneEventCodec);
     EXPECT_TRUE(channel_->SetSendCodecs(codecs));
     EXPECT_TRUE(channel_->CanInsertDtmf());
+
+    if (!caller) {
+      // There's no active send channel yet.
+      EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123, cricket::DF_SEND));
+      EXPECT_TRUE(channel_->AddSendStream(
+          cricket::StreamParams::CreateLegacy(kSsrc1)));
+    }
+
     // Check we fail if the ssrc is invalid.
     EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111, cricket::DF_SEND));
 
@@ -923,8 +942,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusMaxAverageBitrate) {
   EXPECT_EQ(200000, gcodec.rate);
 }
 
-// Test that we can enable NACK with opus.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNack) {
+// Test that we can enable NACK with opus as caller.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCaller) {
   EXPECT_TRUE(SetupEngine());
   int channel_num = voe_.GetLastChannel();
   std::vector<cricket::AudioCodec> codecs;
@@ -936,6 +955,26 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNack) {
   EXPECT_TRUE(voe_.GetNACK(channel_num));
 }
 
+// Test that we can enable NACK with opus as callee.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
+  EXPECT_TRUE(engine_.Init(talk_base::Thread::Current()));
+  channel_ = engine_.CreateChannel();
+  EXPECT_TRUE(channel_ != NULL);
+
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kOpusCodec);
+  codecs[0].AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamNack,
+                                                    cricket::kParamValueEmpty));
+  EXPECT_FALSE(voe_.GetNACK(channel_num));
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_FALSE(voe_.GetNACK(channel_num));
+
+  EXPECT_TRUE(channel_->AddSendStream(
+      cricket::StreamParams::CreateLegacy(kSsrc1)));
+  EXPECT_TRUE(voe_.GetNACK(channel_num));
+}
+
 // Test that we can enable NACK on receive streams.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
   EXPECT_TRUE(SetupEngine());
@@ -1136,8 +1175,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) {
   EXPECT_EQ(106, voe_.GetSendTelephoneEventPayloadType(channel_num));
 }
 
-// Test that we set VAD and DTMF types correctly.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMF) {
+// Test that we set VAD and DTMF types correctly as caller.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
   EXPECT_TRUE(SetupEngine());
   int channel_num = voe_.GetLastChannel();
   std::vector<cricket::AudioCodec> codecs;
@@ -1163,6 +1202,39 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMF) {
   EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
 }
 
+// Test that we set VAD and DTMF types correctly as callee.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
+  EXPECT_TRUE(engine_.Init(talk_base::Thread::Current()));
+  channel_ = engine_.CreateChannel();
+  EXPECT_TRUE(channel_ != NULL);
+
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kIsacCodec);
+  codecs.push_back(kPcmuCodec);
+  // TODO(juberti): cn 32000
+  codecs.push_back(kCn16000Codec);
+  codecs.push_back(kCn8000Codec);
+  codecs.push_back(kTelephoneEventCodec);
+  codecs.push_back(kRedCodec);
+  codecs[0].id = 96;
+  codecs[2].id = 97;  // wideband CN
+  codecs[4].id = 98;  // DTMF
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_TRUE(channel_->AddSendStream(
+      cricket::StreamParams::CreateLegacy(kSsrc1)));
+
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_EQ(96, gcodec.pltype);
+  EXPECT_STREQ("ISAC", gcodec.plname);
+  EXPECT_TRUE(voe_.GetVAD(channel_num));
+  EXPECT_FALSE(voe_.GetFEC(channel_num));
+  EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
+  EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
+  EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+}
+
 // Test that we only apply VAD if we have a CN codec that matches the
 // send codec clockrate.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
@@ -1227,8 +1299,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
   EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
 }
 
-// Test that we set up FEC correctly.
-TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsRED) {
+// Test that we set up FEC correctly as caller.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCaller) {
   EXPECT_TRUE(SetupEngine());
   int channel_num = voe_.GetLastChannel();
   std::vector<cricket::AudioCodec> codecs;
@@ -1247,6 +1319,31 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsRED) {
   EXPECT_EQ(127, voe_.GetSendFECPayloadType(channel_num));
 }
 
+// Test that we set up FEC correctly as callee.
+TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDAsCallee) {
+  EXPECT_TRUE(engine_.Init(talk_base::Thread::Current()));
+  channel_ = engine_.CreateChannel();
+  EXPECT_TRUE(channel_ != NULL);
+
+  int channel_num = voe_.GetLastChannel();
+  std::vector<cricket::AudioCodec> codecs;
+  codecs.push_back(kRedCodec);
+  codecs.push_back(kIsacCodec);
+  codecs.push_back(kPcmuCodec);
+  codecs[0].id = 127;
+  codecs[0].params[""] = "96/96";
+  codecs[1].id = 96;
+  EXPECT_TRUE(channel_->SetSendCodecs(codecs));
+  EXPECT_TRUE(channel_->AddSendStream(
+      cricket::StreamParams::CreateLegacy(kSsrc1)));
+  webrtc::CodecInst gcodec;
+  EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
+  EXPECT_EQ(96, gcodec.pltype);
+  EXPECT_STREQ("ISAC", gcodec.plname);
+  EXPECT_TRUE(voe_.GetFEC(channel_num));
+  EXPECT_EQ(127, voe_.GetSendFECPayloadType(channel_num));
+}
+
 // Test that we set up FEC correctly if params are omitted.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsREDNoParams) {
   EXPECT_TRUE(SetupEngine());
@@ -1947,18 +2044,24 @@ TEST_F(WebRtcVoiceEngineTestFake, StreamCleanup) {
   EXPECT_EQ(0, voe_.GetNumChannels());
 }
 
-// Test the InsertDtmf on default send stream.
-TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStream) {
-  EXPECT_TRUE(SetupEngine());
-  int channel_num = voe_.GetLastChannel();
-  TestInsertDtmf(0, channel_num);
+// Test the InsertDtmf on default send stream as caller.
+TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) {
+  TestInsertDtmf(0, true);
 }
 
-// Test the InsertDtmf on specified send stream.
-TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStream) {
-  EXPECT_TRUE(SetupEngine());
-  int channel_num = voe_.GetLastChannel();
-  TestInsertDtmf(kSsrc1, channel_num);
+// Test the InsertDtmf on default send stream as callee
+TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) {
+  TestInsertDtmf(0, false);
+}
+
+// Test the InsertDtmf on specified send stream as caller.
+TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) {
+  TestInsertDtmf(kSsrc1, true);
+}
+
+// Test the InsertDtmf on specified send stream as callee.
+TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
+  TestInsertDtmf(kSsrc1, false);
 }
 
 // Test that we can play a ringback tone properly in a single-stream call.
@@ -2722,8 +2825,6 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
   EXPECT_TRUE(engine.FindCodec(
       cricket::AudioCodec(96, "red", 8000, 0, 1, 0)));
   EXPECT_TRUE(engine.FindCodec(
-      cricket::AudioCodec(96, "CN", 48000, 0, 1, 0)));
-  EXPECT_TRUE(engine.FindCodec(
       cricket::AudioCodec(96, "CN", 32000, 0, 1, 0)));
   EXPECT_TRUE(engine.FindCodec(
       cricket::AudioCodec(96, "CN", 16000, 0, 1, 0)));
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.cc b/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.cc
index 2f616410fc7..ec00c048288 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.cc
@@ -65,7 +65,9 @@ AsyncStunTCPSocket::AsyncStunTCPSocket(
     : talk_base::AsyncTCPSocketBase(socket, listen, kBufSize) {
 }
 
-int AsyncStunTCPSocket::Send(const void *pv, size_t cb) {
+// TODO(mallinath) - Add support of setting DSCP code on AsyncSocket.
+int AsyncStunTCPSocket::Send(const void *pv, size_t cb,
+                             talk_base::DiffServCodePoint dscp) {
   if (cb > kBufSize || cb < kPacketLenSize + kPacketLenOffset) {
     SetError(EMSGSIZE);
     return -1;
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.h b/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.h
index 2380c4c0dac..ff748d1f287 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket.h
@@ -47,7 +47,8 @@ class AsyncStunTCPSocket : public talk_base::AsyncTCPSocketBase {
   AsyncStunTCPSocket(talk_base::AsyncSocket* socket, bool listen);
   virtual ~AsyncStunTCPSocket() {}
 
-  virtual int Send(const void* pv, size_t cb);
+  virtual int Send(const void* pv, size_t cb,
+                   talk_base::DiffServCodePoint dscp);
   virtual void ProcessInput(char* data, size_t* len);
   virtual void HandleIncomingConnection(talk_base::AsyncSocket* socket);
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket_unittest.cc
index a6757126418..7cb380b0aa4 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/asyncstuntcpsocket_unittest.cc
@@ -121,7 +121,8 @@ class AsyncStunTCPSocketTest : public testing::Test,
   }
 
   bool Send(const void* data, size_t len) {
-    size_t ret = send_socket_->Send(reinterpret_cast<const char*>(data), len);
+    size_t ret = send_socket_->Send(
+        reinterpret_cast<const char*>(data), len, talk_base::DSCP_NO_CHANGE);
     vss_->ProcessMessagesUntilIdle();
     return (ret == len);
   }
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/constants.cc b/chromium/third_party/libjingle/source/talk/p2p/base/constants.cc
index 12336d41016..27d43096b9d 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/constants.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/constants.cc
@@ -262,4 +262,10 @@ const char NS_VOICEMAIL[] = "http://www.google.com/session/voicemail";
 const buzz::StaticQName QN_VOICEMAIL_REGARDING = { NS_VOICEMAIL, "regarding" };
 #endif
 
+// From RFC 4145, SDP setup attribute values.
+const char CONNECTIONROLE_ACTIVE_STR[] = "active";
+const char CONNECTIONROLE_PASSIVE_STR[] = "passive";
+const char CONNECTIONROLE_ACTPASS_STR[] = "actpass";
+const char CONNECTIONROLE_HOLDCONN_STR[] = "holdconn";
+
 }  // namespace cricket
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/constants.h b/chromium/third_party/libjingle/source/talk/p2p/base/constants.h
index f7e5671b267..99e006a0185 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/constants.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/constants.h
@@ -261,6 +261,12 @@ extern const char NS_VOICEMAIL[];
 extern const buzz::StaticQName QN_VOICEMAIL_REGARDING;
 #endif
 
+// RFC 4145, SDP setup attribute values.
+extern const char CONNECTIONROLE_ACTIVE_STR[];
+extern const char CONNECTIONROLE_PASSIVE_STR[];
+extern const char CONNECTIONROLE_ACTPASS_STR[];
+extern const char CONNECTIONROLE_HOLDCONN_STR[];
+
 }  // namespace cricket
 
 #endif  // TALK_P2P_BASE_CONSTANTS_H_
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransport.h b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransport.h
index a6e3b82c1f0..93da1033e8b 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransport.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransport.h
@@ -55,7 +55,6 @@ class DtlsTransport : public Base {
   ~DtlsTransport() {
     Base::DestroyAllChannels();
   }
-
   virtual void SetIdentity_w(talk_base::SSLIdentity* identity) {
     identity_ = identity;
   }
@@ -93,6 +92,12 @@ class DtlsTransport : public Base {
   }
 
   virtual bool NegotiateTransportDescription_w(ContentAction local_role) {
+    if (!Base::local_description() || !Base::remote_description()) {
+      LOG(LS_INFO) << "Local and Remote description must be set before "
+                   << "transport descriptions are negotiated";
+      return false;
+    }
+
     talk_base::SSLFingerprint* local_fp =
         Base::local_description()->identity_fingerprint.get();
     talk_base::SSLFingerprint* remote_fp =
@@ -100,6 +105,74 @@ class DtlsTransport : public Base {
 
     if (remote_fp && local_fp) {
       remote_fingerprint_.reset(new talk_base::SSLFingerprint(*remote_fp));
+
+      // From RFC 4145, section-4.1, The following are the values that the
+      // 'setup' attribute can take in an offer/answer exchange:
+      //       Offer      Answer
+      //      ________________
+      //      active     passive / holdconn
+      //      passive    active / holdconn
+      //      actpass    active / passive / holdconn
+      //      holdconn   holdconn
+      //
+      // Set the role that is most conformant with RFC 5763, Section 5, bullet 1
+      // The endpoint MUST use the setup attribute defined in [RFC4145].
+      // The endpoint that is the offerer MUST use the setup attribute
+      // value of setup:actpass and be prepared to receive a client_hello
+      // before it receives the answer.  The answerer MUST use either a
+      // setup attribute value of setup:active or setup:passive.  Note that
+      // if the answerer uses setup:passive, then the DTLS handshake will
+      // not begin until the answerer is received, which adds additional
+      // latency. setup:active allows the answer and the DTLS handshake to
+      // occur in parallel.  Thus, setup:active is RECOMMENDED.  Whichever
+      // party is active MUST initiate a DTLS handshake by sending a
+      // ClientHello over each flow (host/port quartet).
+      // IOW - actpass and passive modes should be treated as server and
+      // active as client.
+      ConnectionRole local_connection_role =
+          Base::local_description()->connection_role;
+      ConnectionRole remote_connection_role =
+          Base::remote_description()->connection_role;
+
+      bool is_remote_server = false;
+      if (local_role == CA_OFFER) {
+        if (local_connection_role != CONNECTIONROLE_ACTPASS) {
+          LOG(LS_ERROR) << "Offerer must use actpass value for setup attribute";
+          return false;
+        }
+
+        if (remote_connection_role == CONNECTIONROLE_ACTIVE ||
+            remote_connection_role == CONNECTIONROLE_PASSIVE ||
+            remote_connection_role == CONNECTIONROLE_NONE) {
+          is_remote_server = (remote_connection_role == CONNECTIONROLE_PASSIVE);
+        } else {
+          LOG(LS_ERROR) << "Answerer must use either active or passive value "
+                        << "for setup attribute";
+          return false;
+        }
+        // If remote is NONE or ACTIVE it will act as client.
+      } else {
+        if (remote_connection_role != CONNECTIONROLE_ACTPASS &&
+            remote_connection_role != CONNECTIONROLE_NONE) {
+          LOG(LS_ERROR) << "Offerer must use actpass value for setup attribute";
+          return false;
+        }
+
+        if (local_connection_role == CONNECTIONROLE_ACTIVE ||
+            local_connection_role == CONNECTIONROLE_PASSIVE) {
+          is_remote_server = (local_connection_role == CONNECTIONROLE_ACTIVE);
+        } else {
+          LOG(LS_ERROR) << "Answerer must use either active or passive value "
+                        << "for setup attribute";
+          return false;
+        }
+
+        // If local is passive, local will act as server.
+      }
+
+      secure_role_ = is_remote_server ? talk_base::SSL_CLIENT :
+                                        talk_base::SSL_SERVER;
+
     } else if (local_fp && (local_role == CA_ANSWER)) {
       LOG(LS_ERROR)
           << "Local fingerprint supplied when caller didn't offer DTLS";
@@ -128,18 +201,34 @@ class DtlsTransport : public Base {
     Base::DestroyTransportChannel(base_channel);
   }
 
+  virtual bool GetSslRole_w(talk_base::SSLRole* ssl_role) const {
+    ASSERT(ssl_role != NULL);
+    *ssl_role = secure_role_;
+    return true;
+  }
+
  private:
-  virtual void ApplyNegotiatedTransportDescription_w(
+  virtual bool ApplyNegotiatedTransportDescription_w(
       TransportChannelImpl* channel) {
-    channel->SetRemoteFingerprint(
+    // Set ssl role. Role must be set before fingerprint is applied, which
+    // initiates DTLS setup.
+    if (!channel->SetSslRole(secure_role_)) {
+      LOG(LS_INFO) << "Failed to set ssl role for the channel.";
+      return false;
+    }
+    // Apply remote fingerprint.
+    if (!channel->SetRemoteFingerprint(
         remote_fingerprint_->algorithm,
         reinterpret_cast<const uint8 *>(remote_fingerprint_->
                                     digest.data()),
-        remote_fingerprint_->digest.length());
-    Base::ApplyNegotiatedTransportDescription_w(channel);
+        remote_fingerprint_->digest.length())) {
+      return false;
+    }
+    return Base::ApplyNegotiatedTransportDescription_w(channel);
   }
 
   talk_base::SSLIdentity* identity_;
+  talk_base::SSLRole secure_role_;
   talk_base::scoped_ptr<talk_base::SSLFingerprint> remote_fingerprint_;
 };
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.cc b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.cc
index 6cf400c97ec..dead3a550be 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.cc
@@ -29,6 +29,7 @@
 #include "talk/p2p/base/dtlstransportchannel.h"
 
 #include "talk/base/buffer.h"
+#include "talk/base/dscp.h"
 #include "talk/base/messagequeue.h"
 #include "talk/base/stream.h"
 #include "talk/base/sslstreamadapter.h"
@@ -69,7 +70,8 @@ talk_base::StreamResult StreamInterfaceChannel::Write(const void* data,
                                                       int* error) {
   // Always succeeds, since this is an unreliable transport anyway.
   // TODO: Should this block if channel_'s temporarily unwritable?
-  channel_->SendPacket(static_cast<const char*>(data), data_len);
+  channel_->SendPacket(
+      static_cast<const char*>(data), data_len, talk_base::DSCP_NO_CHANGE);
   if (written) {
     *written = data_len;
   }
@@ -102,7 +104,7 @@ DtlsTransportChannelWrapper::DtlsTransportChannelWrapper(
       downward_(NULL),
       dtls_state_(STATE_NONE),
       local_identity_(NULL),
-      dtls_role_(talk_base::SSL_CLIENT) {
+      ssl_role_(talk_base::SSL_CLIENT) {
   channel_->SignalReadableState.connect(this,
       &DtlsTransportChannelWrapper::OnReadableState);
   channel_->SignalWritableState.connect(this,
@@ -171,18 +173,22 @@ bool DtlsTransportChannelWrapper::SetLocalIdentity(
   return true;
 }
 
-void DtlsTransportChannelWrapper::SetIceRole(IceRole role) {
-  // TODO(ekr@rtfm.com): Forbid this if Connect() has been called.
-  ASSERT(dtls_state_ < STATE_ACCEPTED);
+bool DtlsTransportChannelWrapper::SetSslRole(talk_base::SSLRole role) {
+  if (dtls_state_ == STATE_OPEN) {
+    if (ssl_role_ != role) {
+      LOG(LS_ERROR) << "SSL Role can't be reversed after the session is setup.";
+      return false;
+    }
+    return true;
+  }
 
-  // Set the role that is most conformant with RFC 5763, Section 5, bullet 1:
-  //     The endpoint that is the offerer MUST [...] be prepared to receive
-  //     a client_hello before it receives the answer.
-  // (IOW, the offerer is the server, and the answerer is the client).
-  dtls_role_ = (role == ICEROLE_CONTROLLING) ?
-      talk_base::SSL_SERVER : talk_base::SSL_CLIENT;
+  ssl_role_ = role;
+  return true;
+}
 
-  channel_->SetIceRole(role);
+bool DtlsTransportChannelWrapper::GetSslRole(talk_base::SSLRole* role) const {
+  *role = ssl_role_;
+  return true;
 }
 
 bool DtlsTransportChannelWrapper::SetRemoteFingerprint(
@@ -201,12 +207,12 @@ bool DtlsTransportChannelWrapper::SetRemoteFingerprint(
   // hasn't been called.
   if (dtls_state_ > STATE_OFFERED ||
       (dtls_state_ == STATE_NONE && !digest_alg.empty())) {
-    LOG_J(LS_ERROR, this) << "Can't set DTLS remote settings in this state";
+    LOG_J(LS_ERROR, this) << "Can't set DTLS remote settings in this state.";
     return false;
   }
 
   if (digest_alg.empty()) {
-    LOG_J(LS_INFO, this) << "Other side didn't support DTLS";
+    LOG_J(LS_INFO, this) << "Other side didn't support DTLS.";
     dtls_state_ = STATE_NONE;
     return true;
   }
@@ -230,7 +236,7 @@ bool DtlsTransportChannelWrapper::SetupDtls() {
 
   dtls_.reset(talk_base::SSLStreamAdapter::Create(downward));
   if (!dtls_) {
-    LOG_J(LS_ERROR, this) << "Failed to create DTLS adapter";
+    LOG_J(LS_ERROR, this) << "Failed to create DTLS adapter.";
     delete downward;
     return false;
   }
@@ -239,27 +245,27 @@ bool DtlsTransportChannelWrapper::SetupDtls() {
 
   dtls_->SetIdentity(local_identity_->GetReference());
   dtls_->SetMode(talk_base::SSL_MODE_DTLS);
-  dtls_->SetServerRole(dtls_role_);
+  dtls_->SetServerRole(ssl_role_);
   dtls_->SignalEvent.connect(this, &DtlsTransportChannelWrapper::OnDtlsEvent);
   if (!dtls_->SetPeerCertificateDigest(
           remote_fingerprint_algorithm_,
           reinterpret_cast<unsigned char *>(remote_fingerprint_value_.data()),
           remote_fingerprint_value_.length())) {
-    LOG_J(LS_ERROR, this) << "Couldn't set DTLS certificate digest";
+    LOG_J(LS_ERROR, this) << "Couldn't set DTLS certificate digest.";
     return false;
   }
 
   // Set up DTLS-SRTP, if it's been enabled.
   if (!srtp_ciphers_.empty()) {
     if (!dtls_->SetDtlsSrtpCiphers(srtp_ciphers_)) {
-      LOG_J(LS_ERROR, this) << "Couldn't set DTLS-SRTP ciphers";
+      LOG_J(LS_ERROR, this) << "Couldn't set DTLS-SRTP ciphers.";
       return false;
     }
   } else {
-    LOG_J(LS_INFO, this) << "Not using DTLS";
+    LOG_J(LS_INFO, this) << "Not using DTLS.";
   }
 
-  LOG_J(LS_INFO, this) << "DTLS setup complete";
+  LOG_J(LS_INFO, this) << "DTLS setup complete.";
   return true;
 }
 
@@ -293,6 +299,7 @@ bool DtlsTransportChannelWrapper::GetSrtpCipher(std::string* cipher) {
 
 // Called from upper layers to send a media packet.
 int DtlsTransportChannelWrapper::SendPacket(const char* data, size_t size,
+                                            talk_base::DiffServCodePoint dscp,
                                             int flags) {
   int result = -1;
 
@@ -317,7 +324,7 @@ int DtlsTransportChannelWrapper::SendPacket(const char* data, size_t size,
           break;
         }
 
-        result = channel_->SendPacket(data, size);
+        result = channel_->SendPacket(data, size, dscp);
       } else {
         result = (dtls_->WriteAll(data, size, NULL, NULL) ==
           talk_base::SR_SUCCESS) ? static_cast<int>(size) : -1;
@@ -325,7 +332,7 @@ int DtlsTransportChannelWrapper::SendPacket(const char* data, size_t size,
       break;
       // Not doing DTLS.
     case STATE_NONE:
-      result = channel_->SendPacket(data, size);
+      result = channel_->SendPacket(data, size, dscp);
       break;
 
     case STATE_CLOSED:  // Can't send anything when we're closed.
@@ -349,7 +356,7 @@ void DtlsTransportChannelWrapper::OnReadableState(TransportChannel* channel) {
   ASSERT(talk_base::Thread::Current() == worker_thread_);
   ASSERT(channel == channel_);
   LOG_J(LS_VERBOSE, this)
-      << "DTLSTransportChannelWrapper: channel readable state changed";
+      << "DTLSTransportChannelWrapper: channel readable state changed.";
 
   if (dtls_state_ == STATE_NONE || dtls_state_ == STATE_OPEN) {
     set_readable(channel_->readable());
@@ -361,7 +368,7 @@ void DtlsTransportChannelWrapper::OnWritableState(TransportChannel* channel) {
   ASSERT(talk_base::Thread::Current() == worker_thread_);
   ASSERT(channel == channel_);
   LOG_J(LS_VERBOSE, this)
-      << "DTLSTransportChannelWrapper: channel writable state changed";
+      << "DTLSTransportChannelWrapper: channel writable state changed.";
 
   switch (dtls_state_) {
     case STATE_NONE:
@@ -416,13 +423,13 @@ void DtlsTransportChannelWrapper::OnReadPacket(TransportChannel* channel,
       // decide to take this as evidence that the other
       // side is ready to do DTLS and start the handshake
       // on our end
-      LOG_J(LS_WARNING, this) << "Received packet before we know if we are doing "
-                              << "DTLS or not; dropping";
+      LOG_J(LS_WARNING, this) << "Received packet before we know if we are "
+                              << "doing DTLS or not; dropping.";
       break;
 
     case STATE_ACCEPTED:
       // Drop packets received before DTLS has actually started
-      LOG_J(LS_INFO, this) << "Dropping packet received before DTLS started";
+      LOG_J(LS_INFO, this) << "Dropping packet received before DTLS started.";
       break;
 
     case STATE_STARTED:
@@ -431,19 +438,20 @@ void DtlsTransportChannelWrapper::OnReadPacket(TransportChannel* channel,
       // Is this potentially a DTLS packet?
       if (IsDtlsPacket(data, size)) {
         if (!HandleDtlsPacket(data, size)) {
-          LOG_J(LS_ERROR, this) << "Failed to handle DTLS packet";
+          LOG_J(LS_ERROR, this) << "Failed to handle DTLS packet.";
           return;
         }
       } else {
         // Not a DTLS packet; our handshake should be complete by now.
         if (dtls_state_ != STATE_OPEN) {
-          LOG_J(LS_ERROR, this) << "Received non-DTLS packet before DTLS complete";
+          LOG_J(LS_ERROR, this) << "Received non-DTLS packet before DTLS "
+                                << "complete.";
           return;
         }
 
         // And it had better be a SRTP packet.
         if (!IsRtpPacket(data, size)) {
-          LOG_J(LS_ERROR, this) << "Received unexpected non-DTLS packet";
+          LOG_J(LS_ERROR, this) << "Received unexpected non-DTLS packet.";
           return;
         }
 
@@ -472,7 +480,7 @@ void DtlsTransportChannelWrapper::OnDtlsEvent(talk_base::StreamInterface* dtls,
   ASSERT(dtls == dtls_.get());
   if (sig & talk_base::SE_OPEN) {
     // This is the first time.
-    LOG_J(LS_INFO, this) << "DTLS handshake complete";
+    LOG_J(LS_INFO, this) << "DTLS handshake complete.";
     if (dtls_->GetState() == talk_base::SS_OPEN) {
       // The check for OPEN shouldn't be necessary but let's make
       // sure we don't accidentally frob the state if it's closed.
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.h b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.h
index 8321024df22..aec8c7ac428 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel.h
@@ -121,8 +121,9 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
                               TransportChannelImpl* channel);
   virtual ~DtlsTransportChannelWrapper();
 
-  virtual void SetIceRole(IceRole ice_role);
-  // Returns current transport role of the channel.
+  virtual void SetIceRole(IceRole role) {
+    channel_->SetIceRole(role);
+  }
   virtual IceRole GetIceRole() const {
     return channel_->GetIceRole();
   }
@@ -134,7 +135,9 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
   virtual bool IsDtlsActive() const { return dtls_state_ != STATE_NONE; }
 
   // Called to send a packet (via DTLS, if turned on).
-  virtual int SendPacket(const char* data, size_t size, int flags);
+  virtual int SendPacket(const char* data, size_t size,
+                         talk_base::DiffServCodePoint dscp,
+                         int flags);
 
   // TransportChannel calls that we forward to the wrapped transport.
   virtual int SetOption(talk_base::Socket::Option opt, int value) {
@@ -158,6 +161,9 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
   // Find out which DTLS-SRTP cipher was negotiated
   virtual bool GetSrtpCipher(std::string* cipher);
 
+  virtual bool GetSslRole(talk_base::SSLRole* role) const;
+  virtual bool SetSslRole(talk_base::SSLRole role);
+
   // Once DTLS has established (i.e., this channel is writable), this method
   // extracts the keys negotiated during the DTLS handshake, for use in external
   // encryption. DTLS-SRTP uses this to extract the needed SRTP keys.
@@ -234,7 +240,7 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
   std::vector<std::string> srtp_ciphers_;  // SRTP ciphers to use with DTLS.
   State dtls_state_;
   talk_base::SSLIdentity* local_identity_;
-  talk_base::SSLRole dtls_role_;
+  talk_base::SSLRole ssl_role_;
   talk_base::Buffer remote_fingerprint_value_;
   std::string remote_fingerprint_algorithm_;
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel_unittest.cc
index c46839f5ece..267d60be167 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/dtlstransportchannel_unittest.cc
@@ -29,6 +29,7 @@
 #include <set>
 
 #include "talk/base/common.h"
+#include "talk/base/dscp.h"
 #include "talk/base/gunit.h"
 #include "talk/base/helpers.h"
 #include "talk/base/scoped_ptr.h"
@@ -56,6 +57,10 @@ static bool IsRtpLeadByte(uint8 b) {
   return ((b & 0xC0) == 0x80);
 }
 
+using cricket::ConnectionRole;
+
+enum Flags { NF_REOFFER = 0x1, NF_EXPECT_FAILURE = 0x2 };
+
 class DtlsTestClient : public sigslot::has_slots<> {
  public:
   DtlsTestClient(const std::string& name,
@@ -77,6 +82,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
   void CreateIdentity() {
     identity_.reset(talk_base::SSLIdentity::Generate(name_));
   }
+  talk_base::SSLIdentity* identity() { return identity_.get(); }
   void SetupSrtp() {
     ASSERT(identity_.get() != NULL);
     use_dtls_srtp_ = true;
@@ -108,6 +114,9 @@ class DtlsTestClient : public sigslot::has_slots<> {
           this, &DtlsTestClient::OnFakeTransportChannelReadPacket);
     }
   }
+
+  cricket::Transport* transport() { return transport_.get(); }
+
   cricket::FakeTransportChannel* GetFakeChannel(int component) {
     cricket::TransportChannelImpl* ch = transport_->GetChannel(component);
     cricket::DtlsTransportChannelWrapper* wrapper =
@@ -118,13 +127,20 @@ class DtlsTestClient : public sigslot::has_slots<> {
 
   // Offer DTLS if we have an identity; pass in a remote fingerprint only if
   // both sides support DTLS.
-  void Negotiate(DtlsTestClient* peer) {
-    Negotiate(identity_.get(), (identity_) ? peer->identity_.get() : NULL);
+  void Negotiate(DtlsTestClient* peer, cricket::ContentAction action,
+                 ConnectionRole local_role, ConnectionRole remote_role,
+                 int flags) {
+    Negotiate(identity_.get(), (identity_) ? peer->identity_.get() : NULL,
+              action, local_role, remote_role, flags);
   }
 
   // Allow any DTLS configuration to be specified (including invalid ones).
   void Negotiate(talk_base::SSLIdentity* local_identity,
-                 talk_base::SSLIdentity* remote_identity) {
+                 talk_base::SSLIdentity* remote_identity,
+                 cricket::ContentAction action,
+                 ConnectionRole local_role,
+                 ConnectionRole remote_role,
+                 int flags) {
     talk_base::scoped_ptr<talk_base::SSLFingerprint> local_fingerprint;
     talk_base::scoped_ptr<talk_base::SSLFingerprint> remote_fingerprint;
     if (local_identity) {
@@ -137,7 +153,9 @@ class DtlsTestClient : public sigslot::has_slots<> {
           talk_base::DIGEST_SHA_1, remote_identity));
       ASSERT_TRUE(remote_fingerprint.get() != NULL);
     }
-    if (use_dtls_srtp_) {
+
+    if (use_dtls_srtp_ && !(flags & NF_REOFFER)) {
+      // SRTP ciphers will be set only in the beginning.
       for (std::vector<cricket::DtlsTransportChannelWrapper*>::iterator it =
            channels_.begin(); it != channels_.end(); ++it) {
         std::vector<std::string> ciphers;
@@ -150,17 +168,32 @@ class DtlsTestClient : public sigslot::has_slots<> {
         cricket::NS_GINGLE_P2P : cricket::NS_JINGLE_ICE_UDP;
     cricket::TransportDescription local_desc(
         transport_type, std::vector<std::string>(), kIceUfrag1, kIcePwd1,
-        cricket::ICEMODE_FULL, local_fingerprint.get(),
+        cricket::ICEMODE_FULL, local_role,
+         // If remote if the offerer and has no DTLS support, answer will be
+        // without any fingerprint.
+        (action == cricket::CA_ANSWER && !remote_identity) ?
+            NULL : local_fingerprint.get(),
         cricket::Candidates());
-    ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
-                                                         cricket::CA_OFFER));
+
     cricket::TransportDescription remote_desc(
         transport_type, std::vector<std::string>(), kIceUfrag1, kIcePwd1,
-        cricket::ICEMODE_FULL, remote_fingerprint.get(),
+        cricket::ICEMODE_FULL, remote_role, remote_fingerprint.get(),
         cricket::Candidates());
-    ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
-                                                          cricket::CA_ANSWER));
 
+    bool expect_success = (flags & NF_EXPECT_FAILURE) ? false : true;
+    // If |expect_success| is false, expect SRTD or SLTD to fail when
+    // content action is CA_ANSWER.
+    if (action == cricket::CA_OFFER) {
+      ASSERT_TRUE(transport_->SetLocalTransportDescription(
+          local_desc, cricket::CA_OFFER));
+      ASSERT_EQ(expect_success, transport_->SetRemoteTransportDescription(
+          remote_desc, cricket::CA_ANSWER));
+    } else {
+      ASSERT_TRUE(transport_->SetRemoteTransportDescription(
+          remote_desc, cricket::CA_OFFER));
+      ASSERT_EQ(expect_success, transport_->SetLocalTransportDescription(
+          local_desc, cricket::CA_ANSWER));
+    }
     negotiated_dtls_ = (local_identity && remote_identity);
   }
 
@@ -212,7 +245,8 @@ class DtlsTestClient : public sigslot::has_slots<> {
 
       // Only set the bypass flag if we've activated DTLS.
       int flags = (identity_.get() && srtp) ? cricket::PF_SRTP_BYPASS : 0;
-      int rv = channels_[channel]->SendPacket(packet.get(), size, flags);
+      int rv = channels_[channel]->SendPacket(
+          packet.get(), size, talk_base::DSCP_NO_CHANGE, flags);
       ASSERT_GT(rv, 0);
       ASSERT_EQ(size, static_cast<size_t>(rv));
       ++sent;
@@ -373,8 +407,8 @@ class DtlsTransportChannelTest : public testing::Test {
       use_dtls_srtp_ = true;
   }
 
-  bool Connect() {
-    Negotiate();
+  bool Connect(ConnectionRole client1_role, ConnectionRole client2_role) {
+    Negotiate(client1_role, client2_role);
 
     bool rv = client1_.Connect(&client2_);
     EXPECT_TRUE(rv);
@@ -387,8 +421,20 @@ class DtlsTransportChannelTest : public testing::Test {
 
     // Check that we used the right roles.
     if (use_dtls_) {
-      client1_.CheckRole(talk_base::SSL_SERVER);
-      client2_.CheckRole(talk_base::SSL_CLIENT);
+      talk_base::SSLRole client1_ssl_role =
+          (client1_role == cricket::CONNECTIONROLE_ACTIVE ||
+           (client2_role == cricket::CONNECTIONROLE_PASSIVE &&
+            client1_role == cricket::CONNECTIONROLE_ACTPASS)) ?
+              talk_base::SSL_CLIENT : talk_base::SSL_SERVER;
+
+      talk_base::SSLRole client2_ssl_role =
+          (client2_role == cricket::CONNECTIONROLE_ACTIVE ||
+           (client1_role == cricket::CONNECTIONROLE_PASSIVE &&
+            client2_role == cricket::CONNECTIONROLE_ACTPASS)) ?
+              talk_base::SSL_CLIENT : talk_base::SSL_SERVER;
+
+      client1_.CheckRole(client1_ssl_role);
+      client2_.CheckRole(client2_ssl_role);
     }
 
     // Check that we negotiated the right ciphers.
@@ -402,11 +448,55 @@ class DtlsTransportChannelTest : public testing::Test {
 
     return true;
   }
+
+  bool Connect() {
+    // By default, Client1 will be Server and Client2 will be Client.
+    return Connect(cricket::CONNECTIONROLE_ACTPASS,
+                   cricket::CONNECTIONROLE_ACTIVE);
+  }
+
   void Negotiate() {
+    Negotiate(cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTIVE);
+  }
+
+  void Negotiate(ConnectionRole client1_role, ConnectionRole client2_role) {
+    client1_.SetupChannels(channel_ct_, cricket::ICEROLE_CONTROLLING);
+    client2_.SetupChannels(channel_ct_, cricket::ICEROLE_CONTROLLED);
+    // Expect success from SLTD and SRTD.
+    client1_.Negotiate(&client2_, cricket::CA_OFFER,
+                       client1_role, client2_role, 0);
+    client2_.Negotiate(&client1_, cricket::CA_ANSWER,
+                       client2_role, client1_role, 0);
+  }
+
+  // Negotiate with legacy client |client2|. Legacy client doesn't use setup
+  // attributes, except NONE.
+  void NegotiateWithLegacy() {
     client1_.SetupChannels(channel_ct_, cricket::ICEROLE_CONTROLLING);
     client2_.SetupChannels(channel_ct_, cricket::ICEROLE_CONTROLLED);
-    client2_.Negotiate(&client1_);
-    client1_.Negotiate(&client2_);
+    // Expect success from SLTD and SRTD.
+    client1_.Negotiate(&client2_, cricket::CA_OFFER,
+                       cricket::CONNECTIONROLE_ACTPASS,
+                       cricket::CONNECTIONROLE_NONE, 0);
+    client2_.Negotiate(&client1_, cricket::CA_ANSWER,
+                       cricket::CONNECTIONROLE_ACTIVE,
+                       cricket::CONNECTIONROLE_NONE, 0);
+  }
+
+  void Renegotiate(DtlsTestClient* reoffer_initiator,
+                   ConnectionRole client1_role, ConnectionRole client2_role,
+                   int flags) {
+    if (reoffer_initiator == &client1_) {
+      client1_.Negotiate(&client2_, cricket::CA_OFFER,
+                         client1_role, client2_role, flags);
+      client2_.Negotiate(&client1_, cricket::CA_ANSWER,
+                         client2_role, client1_role, flags);
+    } else {
+      client2_.Negotiate(&client1_, cricket::CA_OFFER,
+                         client2_role, client1_role, flags);
+      client1_.Negotiate(&client2_, cricket::CA_ANSWER,
+                         client1_role, client2_role, flags);
+    }
   }
 
   void TestTransfer(size_t channel, size_t size, size_t count, bool srtp) {
@@ -568,3 +658,96 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) {
   TestTransfer(0, 1000, 100, false);
   TestTransfer(0, 1000, 100, true);
 }
+
+// Testing when the remote is passive.
+TEST_F(DtlsTransportChannelTest, TestTransferDtlsAnswererIsPassive) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  SetChannelCount(2);
+  PrepareDtls(true, true);
+  PrepareDtlsSrtp(true, true);
+  ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
+                      cricket::CONNECTIONROLE_PASSIVE));
+  TestTransfer(0, 1000, 100, true);
+  TestTransfer(1, 1000, 100, true);
+}
+
+// Testing with the legacy DTLS client which doesn't use setup attribute.
+// In this case legacy is the answerer.
+TEST_F(DtlsTransportChannelTest, TestDtlsSetupWithLegacyAsAnswerer) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  PrepareDtls(true, true);
+  NegotiateWithLegacy();
+  talk_base::SSLRole channel1_role;
+  talk_base::SSLRole channel2_role;
+  EXPECT_TRUE(client1_.transport()->GetSslRole(&channel1_role));
+  EXPECT_TRUE(client2_.transport()->GetSslRole(&channel2_role));
+  EXPECT_EQ(talk_base::SSL_SERVER, channel1_role);
+  EXPECT_EQ(talk_base::SSL_CLIENT, channel2_role);
+}
+
+// Testing re offer/answer after the session is estbalished. Roles will be
+// kept same as of the previous negotiation.
+TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromOfferer) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  SetChannelCount(2);
+  PrepareDtls(true, true);
+  PrepareDtlsSrtp(true, true);
+  // Initial role for client1 is ACTPASS and client2 is ACTIVE.
+  ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
+                      cricket::CONNECTIONROLE_ACTIVE));
+  TestTransfer(0, 1000, 100, true);
+  TestTransfer(1, 1000, 100, true);
+  // Using input roles for the re-offer.
+  Renegotiate(&client1_, cricket::CONNECTIONROLE_ACTPASS,
+              cricket::CONNECTIONROLE_ACTIVE, NF_REOFFER);
+  TestTransfer(0, 1000, 100, true);
+  TestTransfer(1, 1000, 100, true);
+}
+
+TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromAnswerer) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  SetChannelCount(2);
+  PrepareDtls(true, true);
+  PrepareDtlsSrtp(true, true);
+  // Initial role for client1 is ACTPASS and client2 is ACTIVE.
+  ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
+                      cricket::CONNECTIONROLE_ACTIVE));
+  TestTransfer(0, 1000, 100, true);
+  TestTransfer(1, 1000, 100, true);
+  // Using input roles for the re-offer.
+  Renegotiate(&client2_, cricket::CONNECTIONROLE_PASSIVE,
+              cricket::CONNECTIONROLE_ACTPASS, NF_REOFFER);
+  TestTransfer(0, 1000, 100, true);
+  TestTransfer(1, 1000, 100, true);
+}
+
+// Test that any change in role after the intial setup will result in failure.
+TEST_F(DtlsTransportChannelTest, TestDtlsRoleReversal) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  SetChannelCount(2);
+  PrepareDtls(true, true);
+  PrepareDtlsSrtp(true, true);
+  ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
+                      cricket::CONNECTIONROLE_PASSIVE));
+
+  // Renegotiate from client2 with actpass and client1 as active.
+  Renegotiate(&client2_, cricket::CONNECTIONROLE_ACTPASS,
+              cricket::CONNECTIONROLE_ACTIVE,
+              NF_REOFFER | NF_EXPECT_FAILURE);
+}
+
+// Test that using different setup attributes which results in similar ssl
+// role as the initial negotiation will result in success.
+TEST_F(DtlsTransportChannelTest, TestDtlsReOfferWithDifferentSetupAttr) {
+  MAYBE_SKIP_TEST(HaveDtlsSrtp);
+  SetChannelCount(2);
+  PrepareDtls(true, true);
+  PrepareDtlsSrtp(true, true);
+  ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS,
+                      cricket::CONNECTIONROLE_PASSIVE));
+  // Renegotiate from client2 with actpass and client1 as active.
+  Renegotiate(&client2_, cricket::CONNECTIONROLE_ACTIVE,
+              cricket::CONNECTIONROLE_ACTPASS, NF_REOFFER);
+  TestTransfer(0, 1000, 100, true);
+  TestTransfer(1, 1000, 100, true);
+}
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/fakesession.h b/chromium/third_party/libjingle/source/talk/p2p/base/fakesession.h
index 6b96c60e630..d162950a3bb 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/fakesession.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/fakesession.h
@@ -71,7 +71,8 @@ class FakeTransportChannel : public TransportChannelImpl,
         tiebreaker_(0),
         ice_proto_(ICEPROTO_HYBRID),
         remote_ice_mode_(ICEMODE_FULL),
-        dtls_fingerprint_("", NULL, 0) {
+        dtls_fingerprint_("", NULL, 0),
+        ssl_role_(talk_base::SSL_CLIENT) {
   }
   ~FakeTransportChannel() {
     Reset();
@@ -117,6 +118,14 @@ class FakeTransportChannel : public TransportChannelImpl,
     dtls_fingerprint_ = talk_base::SSLFingerprint(alg, digest, digest_len);
     return true;
   }
+  virtual bool SetSslRole(talk_base::SSLRole role) {
+    ssl_role_ = role;
+    return true;
+  }
+  virtual bool GetSslRole(talk_base::SSLRole* role) const {
+    *role = ssl_role_;
+    return true;
+  }
 
   virtual void Connect() {
     if (state_ == STATE_INIT) {
@@ -160,7 +169,8 @@ class FakeTransportChannel : public TransportChannelImpl,
     }
   }
 
-  virtual int SendPacket(const char* data, size_t len, int flags) {
+  virtual int SendPacket(const char* data, size_t len,
+                         talk_base::DiffServCodePoint dscp, int flags) {
     if (state_ != STATE_CONNECTED) {
       return -1;
     }
@@ -199,7 +209,6 @@ class FakeTransportChannel : public TransportChannelImpl,
 
   bool SetLocalIdentity(talk_base::SSLIdentity* identity) {
     identity_ = identity;
-
     return true;
   }
 
@@ -275,6 +284,7 @@ class FakeTransportChannel : public TransportChannelImpl,
   std::string remote_ice_pwd_;
   IceMode remote_ice_mode_;
   talk_base::SSLFingerprint dtls_fingerprint_;
+  talk_base::SSLRole ssl_role_;
 };
 
 // Fake transport class, which can be passed to anything that needs a Transport.
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.cc b/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.cc
index 7a72d1008ed..d45a66c40b6 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.cc
@@ -518,7 +518,20 @@ void P2PTransportChannel::OnUnknownAddress(
     // request came from.
 
     // There shouldn't be an existing connection with this remote address.
-    ASSERT(port->GetConnection(new_remote_candidate.address()) == NULL);
+    // When ports are muxed, this channel might get multiple unknown addres
+    // signals. In that case if the connection is already exists, we should
+    // simply ignore the signal othewise send server error.
+    if (port->GetConnection(new_remote_candidate.address()) && port_muxed) {
+      LOG(LS_INFO) << "Connection already exist for PeerReflexive candidate: "
+                   << new_remote_candidate.ToString();
+      return;
+    } else if (port->GetConnection(new_remote_candidate.address())) {
+      ASSERT(false);
+      port->SendBindingErrorResponse(stun_msg, address,
+                                     STUN_ERROR_SERVER_ERROR,
+                                     STUN_ERROR_REASON_SERVER_ERROR);
+      return;
+    }
 
     Connection* connection = port->CreateConnection(
         new_remote_candidate, cricket::PortInterface::ORIGIN_THIS_PORT);
@@ -773,7 +786,9 @@ int P2PTransportChannel::SetOption(talk_base::Socket::Option opt, int value) {
 }
 
 // Send data to the other side, using our best connection.
-int P2PTransportChannel::SendPacket(const char *data, size_t len, int flags) {
+int P2PTransportChannel::SendPacket(const char *data, size_t len,
+                                    talk_base::DiffServCodePoint dscp,
+                                    int flags) {
   ASSERT(worker_thread_ == talk_base::Thread::Current());
   if (flags != 0) {
     error_ = EINVAL;
@@ -783,7 +798,7 @@ int P2PTransportChannel::SendPacket(const char *data, size_t len, int flags) {
     error_ = EWOULDBLOCK;
     return -1;
   }
-  int sent = best_connection_->Send(data, len);
+  int sent = best_connection_->Send(data, len, dscp);
   if (sent <= 0) {
     ASSERT(sent < 0);
     error_ = best_connection_->GetError();
@@ -823,6 +838,14 @@ bool P2PTransportChannel::GetStats(ConnectionInfos *infos) {
   return true;
 }
 
+talk_base::DiffServCodePoint P2PTransportChannel::DefaultDscpValue() const {
+  OptionMap::const_iterator it = options_.find(talk_base::Socket::OPT_DSCP);
+  if (it == options_.end()) {
+    return talk_base::DSCP_NO_CHANGE;
+  }
+  return static_cast<talk_base::DiffServCodePoint> (it->second);
+}
+
 // Begin allocate (or immediately re-allocate, if MSG_ALLOCATE pending)
 void P2PTransportChannel::Allocate() {
   // Time for a new allocator, lets make sure we have a signalling channel
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.h b/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.h
index 74a4483801f..2fc718641fb 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel.h
@@ -90,7 +90,8 @@ class P2PTransportChannel : public TransportChannelImpl,
   virtual void OnCandidate(const Candidate& candidate);
 
   // From TransportChannel:
-  virtual int SendPacket(const char *data, size_t len, int flags);
+  virtual int SendPacket(const char *data, size_t len,
+                         talk_base::DiffServCodePoint dscp, int flags);
   virtual int SetOption(talk_base::Socket::Option opt, int value);
   virtual int GetError() { return error_; }
   virtual bool GetStats(std::vector<ConnectionInfo>* stats);
@@ -104,6 +105,54 @@ class P2PTransportChannel : public TransportChannelImpl,
 
   IceMode remote_ice_mode() const { return remote_ice_mode_; }
 
+  // DTLS methods.
+  virtual bool IsDtlsActive() const { return false; }
+
+  // Default implementation.
+  virtual bool GetSslRole(talk_base::SSLRole* role) const {
+    return false;
+  }
+
+  virtual bool SetSslRole(talk_base::SSLRole role) {
+    return false;
+  }
+
+  // Set up the ciphers to use for DTLS-SRTP.
+  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
+    return false;
+  }
+
+  // Find out which DTLS-SRTP cipher was negotiated
+  virtual bool GetSrtpCipher(std::string* cipher) {
+    return false;
+  }
+
+  // Allows key material to be extracted for external encryption.
+  virtual bool ExportKeyingMaterial(
+      const std::string& label,
+      const uint8* context,
+      size_t context_len,
+      bool use_context,
+      uint8* result,
+      size_t result_len) {
+    return false;
+  }
+
+  virtual bool SetLocalIdentity(talk_base::SSLIdentity* identity) {
+    return false;
+  }
+
+  // Set DTLS Remote fingerprint. Must be after local identity set.
+  virtual bool SetRemoteFingerprint(
+    const std::string& digest_alg,
+    const uint8* digest,
+    size_t digest_len) {
+    return false;
+  }
+
+  // Helper method used only in unittest.
+  talk_base::DiffServCodePoint DefaultDscpValue() const;
+
  private:
   talk_base::Thread* thread() { return worker_thread_; }
   PortAllocatorSession* allocator_session() {
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel_unittest.cc
index 32504debba0..e3cddc0e22e 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/p2ptransportchannel_unittest.cc
@@ -25,6 +25,7 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+#include "talk/base/dscp.h"
 #include "talk/base/fakenetwork.h"
 #include "talk/base/firewallsocketserver.h"
 #include "talk/base/gunit.h"
@@ -620,7 +621,7 @@ class P2PTransportChannelTestBase : public testing::Test,
   }
   int SendData(cricket::TransportChannel* channel,
                const char* data, size_t len) {
-    return channel->SendPacket(data, len, 0);
+    return channel->SendPacket(data, len, talk_base::DSCP_NO_CHANGE, 0);
   }
   bool CheckDataOnChannel(cricket::TransportChannel* channel,
                           const char* data, int len) {
@@ -1022,13 +1023,15 @@ const P2PTransportChannelTest::Result*
 #define FLAKY_P2P_TEST(x, y) \
   P2P_TEST_DECLARATION(x, y, DISABLED_)
 
+// TODO(holmer): Disabled due to randomly failing on webrtc buildbots.
+// Issue: webrtc/2383
 #define P2P_TEST_SET(x) \
   P2P_TEST(x, OPEN) \
-  P2P_TEST(x, NAT_FULL_CONE) \
-  P2P_TEST(x, NAT_ADDR_RESTRICTED) \
+  FLAKY_P2P_TEST(x, NAT_FULL_CONE) \
+  FLAKY_P2P_TEST(x, NAT_ADDR_RESTRICTED) \
   P2P_TEST(x, NAT_PORT_RESTRICTED) \
   P2P_TEST(x, NAT_SYMMETRIC) \
-  P2P_TEST(x, NAT_DOUBLE_CONE) \
+  FLAKY_P2P_TEST(x, NAT_DOUBLE_CONE) \
   P2P_TEST(x, NAT_SYMMETRIC_THEN_CONE) \
   P2P_TEST(x, BLOCK_UDP) \
   P2P_TEST(x, BLOCK_UDP_AND_INCOMING_TCP) \
@@ -1144,7 +1147,7 @@ TEST_F(P2PTransportChannelTest, GetStats) {
 }
 
 // Test that we properly handle getting a STUN error due to slow signaling.
-TEST_F(P2PTransportChannelTest, SlowSignaling) {
+TEST_F(P2PTransportChannelTest, DISABLED_SlowSignaling) {
   ConfigureEndpoints(OPEN, NAT_SYMMETRIC,
                      kDefaultPortAllocatorFlags,
                      kDefaultPortAllocatorFlags,
@@ -1266,6 +1269,150 @@ TEST_F(P2PTransportChannelTest, TestTcpConnectionsFromActiveToPassive) {
   DestroyChannels();
 }
 
+TEST_F(P2PTransportChannelTest, TestBundleAllocatorToBundleAllocator) {
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+  SetAllocatorFlags(0, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
+  SetAllocatorFlags(1, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
+
+  CreateChannels(2);
+
+  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
+                   ep1_ch1()->writable() &&
+                   ep2_ch1()->readable() &&
+                   ep2_ch1()->writable(),
+                   1000);
+  EXPECT_TRUE(ep1_ch1()->best_connection() &&
+              ep2_ch1()->best_connection());
+
+  EXPECT_FALSE(ep1_ch2()->readable());
+  EXPECT_FALSE(ep1_ch2()->writable());
+  EXPECT_FALSE(ep2_ch2()->readable());
+  EXPECT_FALSE(ep2_ch2()->writable());
+
+  TestSendRecv(1);  // Only 1 channel is writable per Endpoint.
+  DestroyChannels();
+}
+
+TEST_F(P2PTransportChannelTest, TestBundleAllocatorToNonBundleAllocator) {
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+  // Enable BUNDLE flag at one side.
+  SetAllocatorFlags(0, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
+
+  CreateChannels(2);
+
+  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
+                   ep1_ch1()->writable() &&
+                   ep2_ch1()->readable() &&
+                   ep2_ch1()->writable(),
+                   1000);
+  EXPECT_TRUE_WAIT(ep1_ch2()->readable() &&
+                   ep1_ch2()->writable() &&
+                   ep2_ch2()->readable() &&
+                   ep2_ch2()->writable(),
+                   1000);
+
+  EXPECT_TRUE(ep1_ch1()->best_connection() &&
+              ep2_ch1()->best_connection());
+  EXPECT_TRUE(ep1_ch2()->best_connection() &&
+              ep2_ch2()->best_connection());
+
+  TestSendRecv(2);
+  DestroyChannels();
+}
+
+TEST_F(P2PTransportChannelTest, TestIceRoleConflictWithoutBundle) {
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+  TestSignalRoleConflict();
+}
+
+TEST_F(P2PTransportChannelTest, TestIceRoleConflictWithBundle) {
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+  SetAllocatorFlags(0, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
+  SetAllocatorFlags(1, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
+  TestSignalRoleConflict();
+}
+
+// Tests that the ice configs (protocol, tiebreaker and role) can be passed
+// down to ports.
+TEST_F(P2PTransportChannelTest, TestIceConfigWillPassDownToPort) {
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+
+  SetIceRole(0, cricket::ICEROLE_CONTROLLING);
+  SetIceProtocol(0, cricket::ICEPROTO_GOOGLE);
+  SetIceTiebreaker(0, kTiebreaker1);
+  SetIceRole(1, cricket::ICEROLE_CONTROLLING);
+  SetIceProtocol(1, cricket::ICEPROTO_RFC5245);
+  SetIceTiebreaker(1, kTiebreaker2);
+
+  CreateChannels(1);
+
+  EXPECT_EQ_WAIT(2u, ep1_ch1()->ports().size(), 1000);
+
+  const std::vector<cricket::PortInterface *> ports_before = ep1_ch1()->ports();
+  for (size_t i = 0; i < ports_before.size(); ++i) {
+    EXPECT_EQ(cricket::ICEROLE_CONTROLLING, ports_before[i]->GetIceRole());
+    EXPECT_EQ(cricket::ICEPROTO_GOOGLE, ports_before[i]->IceProtocol());
+    EXPECT_EQ(kTiebreaker1, ports_before[i]->IceTiebreaker());
+  }
+
+  ep1_ch1()->SetIceRole(cricket::ICEROLE_CONTROLLED);
+  ep1_ch1()->SetIceProtocolType(cricket::ICEPROTO_RFC5245);
+  ep1_ch1()->SetIceTiebreaker(kTiebreaker2);
+
+  const std::vector<cricket::PortInterface *> ports_after = ep1_ch1()->ports();
+  for (size_t i = 0; i < ports_after.size(); ++i) {
+    EXPECT_EQ(cricket::ICEROLE_CONTROLLED, ports_before[i]->GetIceRole());
+    EXPECT_EQ(cricket::ICEPROTO_RFC5245, ports_before[i]->IceProtocol());
+    // SetIceTiebreaker after Connect() has been called will fail. So expect the
+    // original value.
+    EXPECT_EQ(kTiebreaker1, ports_before[i]->IceTiebreaker());
+  }
+
+  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
+                   ep1_ch1()->writable() &&
+                   ep2_ch1()->readable() &&
+                   ep2_ch1()->writable(),
+                   1000);
+
+  EXPECT_TRUE(ep1_ch1()->best_connection() &&
+              ep2_ch1()->best_connection());
+
+  TestSendRecv(1);
+}
+
+// Verify that we can set DSCP value and retrieve properly from P2PTC.
+TEST_F(P2PTransportChannelTest, TestDefaultDscpValue) {
+  AddAddress(0, kPublicAddrs[0]);
+  AddAddress(1, kPublicAddrs[1]);
+
+  CreateChannels(1);
+  EXPECT_EQ(talk_base::DSCP_NO_CHANGE,
+            GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
+  EXPECT_EQ(talk_base::DSCP_NO_CHANGE,
+            GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
+  GetEndpoint(0)->cd1_.ch_->SetOption(
+      talk_base::Socket::OPT_DSCP, talk_base::DSCP_CS6);
+  GetEndpoint(1)->cd1_.ch_->SetOption(
+      talk_base::Socket::OPT_DSCP, talk_base::DSCP_CS6);
+  EXPECT_EQ(talk_base::DSCP_CS6,
+            GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
+  EXPECT_EQ(talk_base::DSCP_CS6,
+            GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
+  GetEndpoint(0)->cd1_.ch_->SetOption(
+      talk_base::Socket::OPT_DSCP, talk_base::DSCP_AF41);
+  GetEndpoint(1)->cd1_.ch_->SetOption(
+      talk_base::Socket::OPT_DSCP, talk_base::DSCP_AF41);
+  EXPECT_EQ(talk_base::DSCP_AF41,
+            GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
+  EXPECT_EQ(talk_base::DSCP_AF41,
+            GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
+}
+
 // Test what happens when we have 2 users behind the same NAT. This can lead
 // to interesting behavior because the STUN server will only give out the
 // address of the outermost NAT.
@@ -1306,7 +1453,7 @@ class P2PTransportChannelMultihomedTest : public P2PTransportChannelTestBase {
 };
 
 // Test that we can establish connectivity when both peers are multihomed.
-TEST_F(P2PTransportChannelMultihomedTest, TestBasic) {
+TEST_F(P2PTransportChannelMultihomedTest, DISABLED_TestBasic) {
   AddAddress(0, kPublicAddrs[0]);
   AddAddress(0, kAlternateAddrs[0]);
   AddAddress(1, kPublicAddrs[1]);
@@ -1387,119 +1534,3 @@ TEST_F(P2PTransportChannelMultihomedTest, TestDrain) {
 
   DestroyChannels();
 }
-
-TEST_F(P2PTransportChannelTest, TestBundleAllocatorToBundleAllocator) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-  SetAllocatorFlags(0, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
-  SetAllocatorFlags(1, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
-
-  CreateChannels(2);
-
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
-                   ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() &&
-                   ep2_ch1()->writable(),
-                   1000);
-  EXPECT_TRUE(ep1_ch1()->best_connection() &&
-              ep2_ch1()->best_connection());
-
-  EXPECT_FALSE(ep1_ch2()->readable());
-  EXPECT_FALSE(ep1_ch2()->writable());
-  EXPECT_FALSE(ep2_ch2()->readable());
-  EXPECT_FALSE(ep2_ch2()->writable());
-
-  TestSendRecv(1);  // Only 1 channel is writable per Endpoint.
-  DestroyChannels();
-}
-
-TEST_F(P2PTransportChannelTest, TestBundleAllocatorToNonBundleAllocator) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-  // Enable BUNDLE flag at one side.
-  SetAllocatorFlags(0, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
-
-  CreateChannels(2);
-
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
-                   ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() &&
-                   ep2_ch1()->writable(),
-                   1000);
-  EXPECT_TRUE_WAIT(ep1_ch2()->readable() &&
-                   ep1_ch2()->writable() &&
-                   ep2_ch2()->readable() &&
-                   ep2_ch2()->writable(),
-                   1000);
-
-  EXPECT_TRUE(ep1_ch1()->best_connection() &&
-              ep2_ch1()->best_connection());
-  EXPECT_TRUE(ep1_ch2()->best_connection() &&
-              ep2_ch2()->best_connection());
-
-  TestSendRecv(2);
-  DestroyChannels();
-}
-
-TEST_F(P2PTransportChannelTest, TestIceRoleConflictWithoutBundle) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-  TestSignalRoleConflict();
-}
-
-TEST_F(P2PTransportChannelTest, TestIceRoleConflictWithBundle) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-  SetAllocatorFlags(0, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
-  SetAllocatorFlags(1, cricket::PORTALLOCATOR_ENABLE_BUNDLE);
-  TestSignalRoleConflict();
-}
-
-// Tests that the ice configs (protocol, tiebreaker and role) can be passed
-// down to ports.
-TEST_F(P2PTransportChannelTest, TestIceConfigWillPassDownToPort) {
-  AddAddress(0, kPublicAddrs[0]);
-  AddAddress(1, kPublicAddrs[1]);
-
-  SetIceRole(0, cricket::ICEROLE_CONTROLLING);
-  SetIceProtocol(0, cricket::ICEPROTO_GOOGLE);
-  SetIceTiebreaker(0, kTiebreaker1);
-  SetIceRole(1, cricket::ICEROLE_CONTROLLING);
-  SetIceProtocol(1, cricket::ICEPROTO_RFC5245);
-  SetIceTiebreaker(1, kTiebreaker2);
-
-  CreateChannels(1);
-
-  EXPECT_EQ_WAIT(2u, ep1_ch1()->ports().size(), 1000);
-
-  const std::vector<cricket::PortInterface *> ports_before = ep1_ch1()->ports();
-  for (size_t i = 0; i < ports_before.size(); ++i) {
-    EXPECT_EQ(cricket::ICEROLE_CONTROLLING, ports_before[i]->GetIceRole());
-    EXPECT_EQ(cricket::ICEPROTO_GOOGLE, ports_before[i]->IceProtocol());
-    EXPECT_EQ(kTiebreaker1, ports_before[i]->IceTiebreaker());
-  }
-
-  ep1_ch1()->SetIceRole(cricket::ICEROLE_CONTROLLED);
-  ep1_ch1()->SetIceProtocolType(cricket::ICEPROTO_RFC5245);
-  ep1_ch1()->SetIceTiebreaker(kTiebreaker2);
-
-  const std::vector<cricket::PortInterface *> ports_after = ep1_ch1()->ports();
-  for (size_t i = 0; i < ports_after.size(); ++i) {
-    EXPECT_EQ(cricket::ICEROLE_CONTROLLED, ports_before[i]->GetIceRole());
-    EXPECT_EQ(cricket::ICEPROTO_RFC5245, ports_before[i]->IceProtocol());
-    // SetIceTiebreaker after Connect() has been called will fail. So expect the
-    // original value.
-    EXPECT_EQ(kTiebreaker1, ports_before[i]->IceTiebreaker());
-  }
-
-  EXPECT_TRUE_WAIT(ep1_ch1()->readable() &&
-                   ep1_ch1()->writable() &&
-                   ep2_ch1()->readable() &&
-                   ep2_ch1()->writable(),
-                   1000);
-
-  EXPECT_TRUE(ep1_ch1()->best_connection() &&
-              ep2_ch1()->best_connection());
-
-  TestSendRecv(1);
-}
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/port.cc b/chromium/third_party/libjingle/source/talk/p2p/base/port.cc
index b2bfea1859c..6e688dace56 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/port.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/port.cc
@@ -112,7 +112,7 @@ const int RTT_RATIO = 3;  // 3 : 1
 // The delay before we begin checking if this port is useless.
 const int kPortTimeoutDelay = 30 * 1000;  // 30 seconds
 
-const uint32 MSG_CHECKTIMEOUT = 1;
+// Used by the Connection.
 const uint32 MSG_DELETE = 1;
 }
 
@@ -181,7 +181,8 @@ Port::Port(talk_base::Thread* thread, talk_base::Network* network,
       ice_protocol_(ICEPROTO_GOOGLE),
       ice_role_(ICEROLE_UNKNOWN),
       tiebreaker_(0),
-      shared_socket_(true) {
+      shared_socket_(true),
+      default_dscp_(talk_base::DSCP_NO_CHANGE) {
   Construct();
 }
 
@@ -207,7 +208,8 @@ Port::Port(talk_base::Thread* thread, const std::string& type,
       ice_protocol_(ICEPROTO_GOOGLE),
       ice_role_(ICEROLE_UNKNOWN),
       tiebreaker_(0),
-      shared_socket_(false) {
+      shared_socket_(false),
+      default_dscp_(talk_base::DSCP_NO_CHANGE) {
   ASSERT(factory_ != NULL);
   Construct();
 }
@@ -606,7 +608,7 @@ void Port::SendBindingResponse(StunMessage* request,
   // Send the response message.
   talk_base::ByteBuffer buf;
   response.Write(&buf);
-  if (SendTo(buf.Data(), buf.Length(), addr, false) < 0) {
+  if (SendTo(buf.Data(), buf.Length(), addr, DefaultDscpValue(), false) < 0) {
     LOG_J(LS_ERROR, this) << "Failed to send STUN ping response to "
                           << addr.ToSensitiveString();
   }
@@ -660,7 +662,7 @@ void Port::SendBindingErrorResponse(StunMessage* request,
   // Send the response message.
   talk_base::ByteBuffer buf;
   response.Write(&buf);
-  SendTo(buf.Data(), buf.Length(), addr, false);
+  SendTo(buf.Data(), buf.Length(), addr, DefaultDscpValue(), false);
   LOG_J(LS_INFO, this) << "Sending STUN binding error: reason=" << reason
                        << " to " << addr.ToSensitiveString();
 }
@@ -916,7 +918,8 @@ void Connection::set_use_candidate_attr(bool enable) {
 
 void Connection::OnSendStunPacket(const void* data, size_t size,
                                   StunRequest* req) {
-  if (port_->SendTo(data, size, remote_candidate_.address(), false) < 0) {
+  if (port_->SendTo(data, size, remote_candidate_.address(),
+                    port_->DefaultDscpValue(), false) < 0) {
     LOG_J(LS_WARNING, this) << "Failed to send STUN ping " << req->id();
   }
 }
@@ -1069,7 +1072,12 @@ void Connection::UpdateState(uint32 now) {
   // test we can do is a simple window.
   // If other side has not sent ping after connection has become readable, use
   // |last_data_received_| as the indication.
-  if ((read_state_ == STATE_READABLE) &&
+  // If remote endpoint is doing RFC 5245, it's not required to send ping
+  // after connection is established. If this connection is serving a data
+  // channel, it may not be in a position to send media continuously. Do not
+  // mark connection timeout if it's in RFC5245 mode.
+  // Below check will be performed with end point if it's doing google-ice.
+  if (port_->IsGoogleIce() && (read_state_ == STATE_READABLE) &&
       (last_ping_received_ + CONNECTION_READ_TIMEOUT <= now) &&
       (last_data_received_ + CONNECTION_READ_TIMEOUT <= now)) {
     LOG_J(LS_INFO, this) << "Unreadable after "
@@ -1384,12 +1392,13 @@ ProxyConnection::ProxyConnection(Port* port, size_t index,
   : Connection(port, index, candidate), error_(0) {
 }
 
-int ProxyConnection::Send(const void* data, size_t size) {
+int ProxyConnection::Send(const void* data, size_t size,
+                          talk_base::DiffServCodePoint dscp) {
   if (write_state_ == STATE_WRITE_INIT || write_state_ == STATE_WRITE_TIMEOUT) {
     error_ = EWOULDBLOCK;
     return SOCKET_ERROR;
   }
-  int sent = port_->SendTo(data, size, remote_candidate_.address(), true);
+  int sent = port_->SendTo(data, size, remote_candidate_.address(), dscp, true);
   if (sent <= 0) {
     ASSERT(sent < 0);
     error_ = port_->GetError();
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/port.h b/chromium/third_party/libjingle/source/talk/p2p/base/port.h
index f533f627b6f..7b89e5546e8 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/port.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/port.h
@@ -304,7 +304,17 @@ class Port : public PortInterface, public talk_base::MessageHandler,
   // Returns if Google ICE protocol is used.
   bool IsGoogleIce() const;
 
+  // Returns default DSCP value.
+  talk_base::DiffServCodePoint DefaultDscpValue() const {
+    return default_dscp_;
+  }
+
  protected:
+  enum {
+    MSG_CHECKTIMEOUT = 0,
+    MSG_FIRST_AVAILABLE
+  };
+
   void set_type(const std::string& type) { type_ = type; }
   // Fills in the local address of the port.
   void AddAddress(const talk_base::SocketAddress& address,
@@ -334,6 +344,11 @@ class Port : public PortInterface, public talk_base::MessageHandler,
   // Checks if the address in addr is compatible with the port's ip.
   bool IsCompatibleAddress(const talk_base::SocketAddress& addr);
 
+  // Default DSCP value for this port. Set by TransportChannel.
+  void SetDefaultDscpValue(talk_base::DiffServCodePoint dscp) {
+    default_dscp_ = dscp;
+  }
+
  private:
   void Construct();
   // Called when one of our connections deletes itself.
@@ -372,7 +387,9 @@ class Port : public PortInterface, public talk_base::MessageHandler,
   IceRole ice_role_;
   uint64 tiebreaker_;
   bool shared_socket_;
-
+  // DSCP value for ICE/STUN messages. Set by the P2PTransportChannel after
+  // port becomes ready.
+  talk_base::DiffServCodePoint default_dscp_;
   // Information to use when going through a proxy.
   std::string user_agent_;
   talk_base::ProxyInfo proxy_;
@@ -447,7 +464,8 @@ class Connection : public talk_base::MessageHandler,
   // The connection can send and receive packets asynchronously.  This matches
   // the interface of AsyncPacketSocket, which may use UDP or TCP under the
   // covers.
-  virtual int Send(const void* data, size_t size) = 0;
+  virtual int Send(const void* data, size_t size,
+                   talk_base::DiffServCodePoint dscp) = 0;
 
   // Error if Send() returns < 0
   virtual int GetError() = 0;
@@ -576,7 +594,8 @@ class ProxyConnection : public Connection {
  public:
   ProxyConnection(Port* port, size_t index, const Candidate& candidate);
 
-  virtual int Send(const void* data, size_t size);
+  virtual int Send(const void* data, size_t size,
+                   talk_base::DiffServCodePoint dscp);
   virtual int GetError() { return error_; }
 
  private:
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/port_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/port_unittest.cc
index cecefdaeebf..d3e02ac9f58 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/port_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/port_unittest.cc
@@ -172,7 +172,7 @@ class TestPort : public Port {
   }
   virtual int SendTo(
       const void* data, size_t size, const talk_base::SocketAddress& addr,
-      bool payload) {
+      talk_base::DiffServCodePoint dscp, bool payload) {
     if (!payload) {
       IceMessage* msg = new IceMessage;
       ByteBuffer* buf = new ByteBuffer(static_cast<const char*>(data), size);
@@ -787,10 +787,12 @@ class FakeAsyncPacketSocket : public AsyncPacketSocket {
   }
 
   // Send a packet.
-  virtual int Send(const void *pv, size_t cb) {
+  virtual int Send(const void *pv, size_t cb,
+                   talk_base::DiffServCodePoint dscp) {
     return static_cast<int>(cb);
   }
-  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr) {
+  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp) {
     return static_cast<int>(cb);
   }
   virtual int Close() {
@@ -1224,6 +1226,26 @@ TEST_F(PortTest, TestSkipCrossFamilyUdp) {
   TestCrossFamilyPorts(SOCK_DGRAM);
 }
 
+// This test verifies DSCP value set through SetOption interface can be
+// get through DefaultDscpValue.
+TEST_F(PortTest, TestDefaultDscpValue) {
+  talk_base::scoped_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
+  udpport->SetOption(talk_base::Socket::OPT_DSCP, talk_base::DSCP_CS6);
+  EXPECT_EQ(talk_base::DSCP_CS6, udpport->DefaultDscpValue());
+  talk_base::scoped_ptr<TCPPort> tcpport(CreateTcpPort(kLocalAddr1));
+  tcpport->SetOption(talk_base::Socket::OPT_DSCP, talk_base::DSCP_AF31);
+  EXPECT_EQ(talk_base::DSCP_AF31, tcpport->DefaultDscpValue());
+  talk_base::scoped_ptr<StunPort> stunport(
+      CreateStunPort(kLocalAddr1, nat_socket_factory1()));
+  stunport->SetOption(talk_base::Socket::OPT_DSCP, talk_base::DSCP_AF41);
+  EXPECT_EQ(talk_base::DSCP_AF41, stunport->DefaultDscpValue());
+  talk_base::scoped_ptr<TurnPort> turnport(CreateTurnPort(
+      kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
+  turnport->SetOption(talk_base::Socket::OPT_DSCP, talk_base::DSCP_CS7);
+  EXPECT_EQ(talk_base::DSCP_CS7, turnport->DefaultDscpValue());
+  // TODO(mallinath) - Test DSCP through GetOption.
+}
+
 // Test sending STUN messages in GICE format.
 TEST_F(PortTest, TestSendStunMessageAsGice) {
   talk_base::scoped_ptr<TestPort> lport(
@@ -2127,14 +2149,16 @@ TEST_F(PortTest, TestWritableState) {
   // Data should be unsendable until the connection is accepted.
   char data[] = "abcd";
   int data_size = ARRAY_SIZE(data);
-  EXPECT_EQ(SOCKET_ERROR, ch1.conn()->Send(data, data_size));
+  EXPECT_EQ(SOCKET_ERROR,
+            ch1.conn()->Send(data, data_size, talk_base::DSCP_NO_CHANGE));
 
   // Accept the connection to return the binding response, transition to
   // writable, and allow data to be sent.
   ch2.AcceptConnection();
   EXPECT_EQ_WAIT(Connection::STATE_WRITABLE, ch1.conn()->write_state(),
                  kTimeout);
-  EXPECT_EQ(data_size, ch1.conn()->Send(data, data_size));
+  EXPECT_EQ(data_size,
+            ch1.conn()->Send(data, data_size, talk_base::DSCP_NO_CHANGE));
 
   // Ask the connection to update state as if enough time has passed to lose
   // full writability and 5 pings went unresponded to. We'll accomplish the
@@ -2147,7 +2171,8 @@ TEST_F(PortTest, TestWritableState) {
   EXPECT_EQ(Connection::STATE_WRITE_UNRELIABLE, ch1.conn()->write_state());
 
   // Data should be able to be sent in this state.
-  EXPECT_EQ(data_size, ch1.conn()->Send(data, data_size));
+  EXPECT_EQ(data_size,
+            ch1.conn()->Send(data, data_size, talk_base::DSCP_NO_CHANGE));
 
   // And now allow the other side to process the pings and send binding
   // responses.
@@ -2164,7 +2189,8 @@ TEST_F(PortTest, TestWritableState) {
   EXPECT_EQ(Connection::STATE_WRITE_TIMEOUT, ch1.conn()->write_state());
 
   // Now that the connection has completely timed out, data send should fail.
-  EXPECT_EQ(SOCKET_ERROR, ch1.conn()->Send(data, data_size));
+  EXPECT_EQ(SOCKET_ERROR,
+            ch1.conn()->Send(data, data_size, talk_base::DSCP_NO_CHANGE));
 
   ch1.Stop();
   ch2.Stop();
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/portallocatorsessionproxy_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/portallocatorsessionproxy_unittest.cc
index 2d3ec3dbe61..689fb968d5d 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/portallocatorsessionproxy_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/portallocatorsessionproxy_unittest.cc
@@ -59,7 +59,9 @@ class TestSessionChannel : public sigslot::has_slots<> {
     proxy_session_->SignalPortReady.connect(
         this, &TestSessionChannel::OnPortReady);
   }
-  virtual ~TestSessionChannel() {}
+  virtual ~TestSessionChannel() {
+    delete proxy_session_;
+  }
   void OnCandidatesReady(PortAllocatorSession* session,
                          const std::vector<Candidate>& candidates) {
     EXPECT_EQ(proxy_session_, session);
@@ -102,10 +104,11 @@ class PortAllocatorSessionProxyTest : public testing::Test {
   PortAllocatorSessionProxyTest()
       : socket_factory_(talk_base::Thread::Current()),
         allocator_(talk_base::Thread::Current(), NULL),
-        session_(talk_base::Thread::Current(), &socket_factory_,
-                 "test content", 1,
-                 kIceUfrag0, kIcePwd0),
-        session_muxer_(new PortAllocatorSessionMuxer(&session_)) {
+        session_(new cricket::FakePortAllocatorSession(
+                     talk_base::Thread::Current(), &socket_factory_,
+                     "test content", 1,
+                     kIceUfrag0, kIcePwd0)),
+        session_muxer_(new PortAllocatorSessionMuxer(session_)) {
   }
   virtual ~PortAllocatorSessionProxyTest() {}
   void RegisterSessionProxy(PortAllocatorSessionProxy* proxy) {
@@ -124,7 +127,7 @@ class PortAllocatorSessionProxyTest : public testing::Test {
  protected:
   talk_base::BasicPacketSocketFactory socket_factory_;
   cricket::FakePortAllocator allocator_;
-  cricket::FakePortAllocatorSession session_;
+  cricket::FakePortAllocatorSession* session_;
   // Muxer object will be delete itself after all registered session proxies
   // are deleted.
   PortAllocatorSessionMuxer* session_muxer_;
@@ -143,7 +146,7 @@ TEST_F(PortAllocatorSessionProxyTest, TestLateBinding) {
   EXPECT_EQ_WAIT(1, channel1->candidates_count(), 1000);
   EXPECT_EQ(1, channel1->ports_count());
   EXPECT_TRUE(channel1->allocation_complete());
-  EXPECT_EQ(1, session_.port_config_count());
+  EXPECT_EQ(1, session_->port_config_count());
   // Creating another PortAllocatorSessionProxy and it also should receive
   // already happened events.
   PortAllocatorSessionProxy* proxy =
@@ -154,7 +157,7 @@ TEST_F(PortAllocatorSessionProxyTest, TestLateBinding) {
   EXPECT_EQ_WAIT(1, channel2->candidates_count(), 1000);
   EXPECT_EQ(1, channel2->ports_count());
   EXPECT_TRUE_WAIT(channel2->allocation_complete(), 1000);
-  EXPECT_EQ(1, session_.port_config_count());
+  EXPECT_EQ(1, session_->port_config_count());
   delete channel1;
   delete channel2;
 }
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/portinterface.h b/chromium/third_party/libjingle/source/talk/p2p/base/portinterface.h
index b956f9abe50..6ea63466c9d 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/portinterface.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/portinterface.h
@@ -30,6 +30,7 @@
 
 #include <string>
 
+#include "talk/base/dscp.h"
 #include "talk/base/socketaddress.h"
 #include "talk/p2p/base/transport.h"
 
@@ -90,16 +91,16 @@ class PortInterface {
 
   // Functions on the underlying socket(s).
   virtual int SetOption(talk_base::Socket::Option opt, int value) = 0;
-  virtual int GetError() = 0;
-
   virtual int GetOption(talk_base::Socket::Option opt, int* value) = 0;
+  virtual int GetError() = 0;
 
   virtual const std::vector<Candidate>& Candidates() const = 0;
 
   // Sends the given packet to the given address, provided that the address is
   // that of a connection or an address that has sent to us already.
   virtual int SendTo(const void* data, size_t size,
-                     const talk_base::SocketAddress& addr, bool payload) = 0;
+                     const talk_base::SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp, bool payload) = 0;
 
   // Indicates that we received a successful STUN binding request from an
   // address that doesn't correspond to any current connection.  To turn this
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.cc b/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.cc
index 2c6119aa5d3..eae39f1612d 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.cc
@@ -97,9 +97,10 @@ Connection* PortProxy::CreateConnection(const Candidate& remote_candidate,
 int PortProxy::SendTo(const void* data,
                       size_t size,
                       const talk_base::SocketAddress& addr,
+                      talk_base::DiffServCodePoint dscp,
                       bool payload) {
   ASSERT(impl_ != NULL);
-  return impl_->SendTo(data, size, addr, payload);
+  return impl_->SendTo(data, size, addr, dscp, payload);
 }
 
 int PortProxy::SetOption(talk_base::Socket::Option opt,
@@ -114,7 +115,6 @@ int PortProxy::GetOption(talk_base::Socket::Option opt,
   return impl_->GetOption(opt, value);
 }
 
-
 int PortProxy::GetError() {
   ASSERT(impl_ != NULL);
   return impl_->GetError();
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.h b/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.h
index 25808ea361a..da326646dcc 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/portproxy.h
@@ -68,7 +68,9 @@ class PortProxy : public PortInterface, public sigslot::has_slots<> {
       const talk_base::SocketAddress& remote_addr);
 
   virtual int SendTo(const void* data, size_t size,
-                     const talk_base::SocketAddress& addr, bool payload);
+                     const talk_base::SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp,
+                     bool payload);
   virtual int SetOption(talk_base::Socket::Option opt, int value);
   virtual int GetOption(talk_base::Socket::Option opt, int* value);
   virtual int GetError();
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/pseudotcp.cc b/chromium/third_party/libjingle/source/talk/p2p/base/pseudotcp.cc
index 6a2c1d26b7e..b647fbf3dec 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/pseudotcp.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/pseudotcp.cc
@@ -36,6 +36,7 @@
 #include "talk/base/byteorder.h"
 #include "talk/base/common.h"
 #include "talk/base/logging.h"
+#include "talk/base/scoped_ptr.h"
 #include "talk/base/socket.h"
 #include "talk/base/stringutils.h"
 #include "talk/base/timeutils.h"
@@ -538,25 +539,24 @@ IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32 seq, uint8 flags,
 
   uint32 now = Now();
 
-  uint8 buffer[MAX_PACKET];
-  long_to_bytes(m_conv, buffer);
-  long_to_bytes(seq, buffer + 4);
-  long_to_bytes(m_rcv_nxt, buffer + 8);
+  talk_base::scoped_array<uint8> buffer(new uint8[MAX_PACKET]);
+  long_to_bytes(m_conv, buffer.get());
+  long_to_bytes(seq, buffer.get() + 4);
+  long_to_bytes(m_rcv_nxt, buffer.get() + 8);
   buffer[12] = 0;
   buffer[13] = flags;
-  short_to_bytes(static_cast<uint16>(m_rcv_wnd >> m_rwnd_scale), buffer + 14);
+  short_to_bytes(
+      static_cast<uint16>(m_rcv_wnd >> m_rwnd_scale), buffer.get() + 14);
 
   // Timestamp computations
-  long_to_bytes(now, buffer + 16);
-  long_to_bytes(m_ts_recent, buffer + 20);
+  long_to_bytes(now, buffer.get() + 16);
+  long_to_bytes(m_ts_recent, buffer.get() + 20);
   m_ts_lastack = m_rcv_nxt;
 
   if (len) {
     size_t bytes_read = 0;
-    talk_base::StreamResult result = m_sbuf.ReadOffset(buffer + HEADER_SIZE,
-                                                       len,
-                                                       offset,
-                                                       &bytes_read);
+    talk_base::StreamResult result = m_sbuf.ReadOffset(
+        buffer.get() + HEADER_SIZE, len, offset, &bytes_read);
     UNUSED(result);
     ASSERT(result == talk_base::SR_SUCCESS);
     ASSERT(static_cast<uint32>(bytes_read) == len);
@@ -573,7 +573,8 @@ IPseudoTcpNotify::WriteResult PseudoTcp::packet(uint32 seq, uint8 flags,
                << "><LEN=" << len << ">";
 #endif // _DEBUGMSG
 
-  IPseudoTcpNotify::WriteResult wres = m_notify->TcpWritePacket(this, reinterpret_cast<char *>(buffer), len + HEADER_SIZE);
+  IPseudoTcpNotify::WriteResult wres = m_notify->TcpWritePacket(
+      this, reinterpret_cast<char *>(buffer.get()), len + HEADER_SIZE);
   // Note: When len is 0, this is an ACK packet.  We don't read the return value for those,
   // and thus we won't retry.  So go ahead and treat the packet as a success (basically simulate
   // as if it were dropped), which will prevent our timers from being messed up.
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.cc b/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.cc
index 54adab13ca9..ec225029b8b 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.cc
@@ -74,14 +74,16 @@ RawTransportChannel::~RawTransportChannel() {
   delete allocator_session_;
 }
 
-int RawTransportChannel::SendPacket(const char *data, size_t size, int flags) {
+int RawTransportChannel::SendPacket(const char *data, size_t size,
+                                    talk_base::DiffServCodePoint dscp,
+                                    int flags) {
   if (port_ == NULL)
     return -1;
   if (remote_address_.IsNil())
     return -1;
   if (flags != 0)
     return -1;
-  return port_->SendTo(data, size, remote_address_, true);
+  return port_->SendTo(data, size, remote_address_, dscp, true);
 }
 
 int RawTransportChannel::SetOption(talk_base::Socket::Option opt, int value) {
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.h b/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.h
index 0f606b72797..2aac2b5edf6 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/rawtransportchannel.h
@@ -64,7 +64,8 @@ class RawTransportChannel : public TransportChannelImpl,
   virtual ~RawTransportChannel();
 
   // Implementation of normal channel packet sending.
-  virtual int SendPacket(const char *data, size_t len, int flags);
+  virtual int SendPacket(const char *data, size_t len,
+                         talk_base::DiffServCodePoint dscp, int flags);
   virtual int SetOption(talk_base::Socket::Option opt, int value);
   virtual int GetError();
 
@@ -101,6 +102,55 @@ class RawTransportChannel : public TransportChannelImpl,
   virtual void SetIcePwd(const std::string& ice_pwd) {}
   virtual void SetRemoteIceMode(IceMode mode) {}
 
+  virtual bool GetStats(ConnectionInfos* infos) {
+    return false;
+  }
+
+  // DTLS methods.
+  virtual bool IsDtlsActive() const { return false; }
+
+  // Default implementation.
+  virtual bool GetSslRole(talk_base::SSLRole* role) const {
+    return false;
+  }
+
+  virtual bool SetSslRole(talk_base::SSLRole role) {
+    return false;
+  }
+
+  // Set up the ciphers to use for DTLS-SRTP.
+  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
+    return false;
+  }
+
+  // Find out which DTLS-SRTP cipher was negotiated
+  virtual bool GetSrtpCipher(std::string* cipher) {
+    return false;
+  }
+
+  // Allows key material to be extracted for external encryption.
+  virtual bool ExportKeyingMaterial(
+      const std::string& label,
+      const uint8* context,
+      size_t context_len,
+      bool use_context,
+      uint8* result,
+      size_t result_len) {
+    return false;
+  }
+
+  virtual bool SetLocalIdentity(talk_base::SSLIdentity* identity) {
+    return false;
+  }
+
+  // Set DTLS Remote fingerprint. Must be after local identity set.
+  virtual bool SetRemoteFingerprint(
+    const std::string& digest_alg,
+    const uint8* digest,
+    size_t digest_len) {
+    return false;
+  }
+
  private:
   RawTransport* raw_transport_;
   talk_base::Thread *worker_thread_;
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/relayport.cc b/chromium/third_party/libjingle/source/talk/p2p/base/relayport.cc
index 0cd40e5aec5..ff8c07c5514 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/relayport.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/relayport.cc
@@ -67,7 +67,7 @@ class RelayConnection : public sigslot::has_slots<> {
   bool CheckResponse(StunMessage* msg);
 
   // Sends data to the relay server.
-  int Send(const void* pv, size_t cb);
+  int Send(const void* pv, size_t cb, talk_base::DiffServCodePoint dscp);
 
   // Sends a STUN allocate request message to the relay server.
   void SendAllocateRequest(RelayEntry* entry, int delay);
@@ -123,7 +123,8 @@ class RelayEntry : public talk_base::MessageHandler,
   // Sends a packet to the given destination address using the socket of this
   // entry.  This will wrap the packet in STUN if necessary.
   int SendTo(const void* data, size_t size,
-    const talk_base::SocketAddress& addr);
+             const talk_base::SocketAddress& addr,
+             talk_base::DiffServCodePoint dscp);
 
   // Schedules a keep-alive allocate request.
   void ScheduleKeepAlive();
@@ -163,7 +164,8 @@ class RelayEntry : public talk_base::MessageHandler,
 
   // Sends the given data on the socket to the server with no wrapping.  This
   // returns the number of bytes written or -1 if an error occurred.
-  int SendPacket(const void* data, size_t size);
+  int SendPacket(const void* data, size_t size,
+                 talk_base::DiffServCodePoint dscp);
 };
 
 // Handles an allocate request for a particular RelayEntry.
@@ -300,7 +302,9 @@ Connection* RelayPort::CreateConnection(const Candidate& address,
 }
 
 int RelayPort::SendTo(const void* data, size_t size,
-                      const talk_base::SocketAddress& addr, bool payload) {
+                      const talk_base::SocketAddress& addr,
+                      talk_base::DiffServCodePoint dscp,
+                      bool payload) {
   // Try to find an entry for this specific address.  Note that the first entry
   // created was not given an address initially, so it can be set to the first
   // address that comes along.
@@ -341,7 +345,7 @@ int RelayPort::SendTo(const void* data, size_t size,
   }
 
   // Send the actual contents to the server using the usual mechanism.
-  int sent = entry->SendTo(data, size, addr);
+  int sent = entry->SendTo(data, size, addr, dscp);
   if (sent <= 0) {
     ASSERT(sent < 0);
     error_ = entry->GetError();
@@ -354,6 +358,14 @@ int RelayPort::SendTo(const void* data, size_t size,
 
 int RelayPort::SetOption(talk_base::Socket::Option opt, int value) {
   int result = 0;
+  // DSCP option is not passed to the socket.
+  // TODO(mallinath) - After we have the support on socket,
+  // remove this specialization.
+  if (opt == talk_base::Socket::OPT_DSCP) {
+    SetDefaultDscpValue(static_cast<talk_base::DiffServCodePoint>(value));
+    return result;
+  }
+
   for (size_t i = 0; i < entries_.size(); ++i) {
     if (entries_[i]->SetSocketOption(opt, value) < 0) {
       result = -1;
@@ -418,7 +430,9 @@ bool RelayConnection::CheckResponse(StunMessage* msg) {
 
 void RelayConnection::OnSendPacket(const void* data, size_t size,
                                    StunRequest* req) {
-  int sent = socket_->SendTo(data, size, GetAddress());
+  // TODO(mallinath) Find a way to get DSCP value from Port.
+  int sent = socket_->SendTo(
+      data, size, GetAddress(), talk_base::DSCP_NO_CHANGE);
   if (sent <= 0) {
     LOG(LS_VERBOSE) << "OnSendPacket: failed sending to " << GetAddress() <<
         std::strerror(socket_->GetError());
@@ -426,8 +440,9 @@ void RelayConnection::OnSendPacket(const void* data, size_t size,
   }
 }
 
-int RelayConnection::Send(const void* pv, size_t cb) {
-  return socket_->SendTo(pv, cb, GetAddress());
+int RelayConnection::Send(const void* pv, size_t cb,
+                          talk_base::DiffServCodePoint dscp) {
+  return socket_->SendTo(pv, cb, GetAddress(), dscp);
 }
 
 void RelayConnection::SendAllocateRequest(RelayEntry* entry, int delay) {
@@ -546,11 +561,12 @@ void RelayEntry::OnConnect(const talk_base::SocketAddress& mapped_addr,
 }
 
 int RelayEntry::SendTo(const void* data, size_t size,
-                       const talk_base::SocketAddress& addr) {
+                       const talk_base::SocketAddress& addr,
+                       talk_base::DiffServCodePoint dscp) {
   // If this connection is locked to the address given, then we can send the
   // packet with no wrapper.
   if (locked_ && (ext_addr_ == addr))
-    return SendPacket(data, size);
+    return SendPacket(data, size, dscp);
 
   // Otherwise, we must wrap the given data in a STUN SEND request so that we
   // can communicate the destination address to the server.
@@ -598,7 +614,7 @@ int RelayEntry::SendTo(const void* data, size_t size,
   talk_base::ByteBuffer buf;
   request.Write(&buf);
 
-  return SendPacket(buf.Data(), buf.Length());
+  return SendPacket(buf.Data(), buf.Length(), dscp);
 }
 
 void RelayEntry::ScheduleKeepAlive() {
@@ -744,12 +760,13 @@ void RelayEntry::OnReadyToSend(talk_base::AsyncPacketSocket* socket) {
   }
 }
 
-int RelayEntry::SendPacket(const void* data, size_t size) {
+int RelayEntry::SendPacket(const void* data, size_t size,
+                           talk_base::DiffServCodePoint dscp) {
   int sent = 0;
   if (current_connection_) {
     // We are connected, no need to send packets anywere else than to
     // the current connection.
-    sent = current_connection_->Send(data, size);
+    sent = current_connection_->Send(data, size, dscp);
   }
   return sent;
 }
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/relayport.h b/chromium/third_party/libjingle/source/talk/p2p/base/relayport.h
index a2bfb74425b..c15e7e01069 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/relayport.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/relayport.h
@@ -92,7 +92,9 @@ class RelayPort : public Port {
   void SetReady();
 
   virtual int SendTo(const void* data, size_t size,
-                     const talk_base::SocketAddress& addr, bool payload);
+                     const talk_base::SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp,
+                     bool payload);
 
   // Dispatches the given packet to the port or connection as appropriate.
   void OnReadPacket(const char* data, size_t size,
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/relayserver.cc b/chromium/third_party/libjingle/source/talk/p2p/base/relayserver.cc
index 95aa08c39c2..c2cf472d3ba 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/relayserver.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/relayserver.cc
@@ -51,7 +51,7 @@ static const uint32 kMessageAcceptConnection = 1;
 // Calls SendTo on the given socket and logs any bad results.
 void Send(talk_base::AsyncPacketSocket* socket, const char* bytes, size_t size,
           const talk_base::SocketAddress& addr) {
-  int result = socket->SendTo(bytes, size, addr);
+  int result = socket->SendTo(bytes, size, addr, talk_base::DSCP_NO_CHANGE);
   if (result < static_cast<int>(size)) {
     LOG(LS_ERROR) << "SendTo wrote only " << result << " of " << size
                   << " bytes";
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/session.cc b/chromium/third_party/libjingle/source/talk/p2p/base/session.cc
index 74eda019c34..3128393c349 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/session.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/session.cc
@@ -999,9 +999,10 @@ TransportInfos Session::GetEmptyTransportInfos(
   TransportInfos tinfos;
   for (ContentInfos::const_iterator content = contents.begin();
        content != contents.end(); ++content) {
-    tinfos.push_back(
-        TransportInfo(content->name,
-            TransportDescription(transport_type(), Candidates())));
+    tinfos.push_back(TransportInfo(content->name,
+                                   TransportDescription(transport_type(),
+                                                        std::string(),
+                                                        std::string())));
   }
   return tinfos;
 }
@@ -1558,7 +1559,9 @@ bool Session::SendTransportInfoMessage(const TransportProxy* transproxy,
                                        const Candidates& candidates,
                                        SessionError* error) {
   return SendTransportInfoMessage(TransportInfo(transproxy->content_name(),
-      TransportDescription(transproxy->type(), candidates)), error);
+      TransportDescription(transproxy->type(), std::vector<std::string>(),
+                           std::string(), std::string(), ICEMODE_FULL,
+                           CONNECTIONROLE_NONE, NULL, candidates)), error);
 }
 
 bool Session::WriteSessionAction(SignalingProtocol protocol,
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/session_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/session_unittest.cc
index 1d072ae0a99..b64e7374221 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/session_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/session_unittest.cc
@@ -32,6 +32,7 @@
 
 #include "talk/base/base64.h"
 #include "talk/base/common.h"
+#include "talk/base/dscp.h"
 #include "talk/base/gunit.h"
 #include "talk/base/helpers.h"
 #include "talk/base/logging.h"
@@ -713,7 +714,7 @@ cricket::SessionDescription* NewTestSessionDescription(
                     new TestContentDescription(gingle_content_type,
                                                content_type_a));
   cricket::TransportDescription desc(cricket::NS_GINGLE_P2P,
-                                     cricket::Candidates());
+                                     std::string(), std::string());
   offer->AddTransportInfo(cricket::TransportInfo(content_name_a, desc));
 
   if (content_name_a != content_name_b) {
@@ -735,7 +736,7 @@ cricket::SessionDescription* NewTestSessionDescription(
   offer->AddTransportInfo(cricket::TransportInfo
                           (content_name, cricket::TransportDescription(
                           cricket::NS_GINGLE_P2P,
-                          cricket::Candidates())));
+                          std::string(), std::string())));
   return offer;
 }
 
@@ -830,7 +831,7 @@ struct ChannelHandler : sigslot::has_slots<> {
     std::string data_with_id(name);
     data_with_id += data;
     int result = channel->SendPacket(data_with_id.c_str(), data_with_id.size(),
-                                     0);
+                                     talk_base::DSCP_NO_CHANGE, 0);
     EXPECT_EQ(static_cast<int>(data_with_id.size()), result);
   }
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/sessionmanager.cc b/chromium/third_party/libjingle/source/talk/p2p/base/sessionmanager.cc
index 7aa52b35968..15b745239d4 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/sessionmanager.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/sessionmanager.cc
@@ -210,6 +210,12 @@ void SessionManager::OnIncomingResponse(const buzz::XmlElement* orig_stanza,
   }
 
   Session* session = FindSession(msg.sid, msg.to);
+  if (!session) {
+    // Also try the QN_FROM in the response stanza, in case we sent the request
+    // to a bare JID but got the response from a full JID.
+    std::string ack_from = response_stanza->Attr(buzz::QN_FROM);
+    session = FindSession(msg.sid, ack_from);
+  }
   if (session) {
     session->OnIncomingResponse(orig_stanza, response_stanza, msg);
   }
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/sessionmessages.cc b/chromium/third_party/libjingle/source/talk/p2p/base/sessionmessages.cc
index 031c3d6f6aa..7a03d765064 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/sessionmessages.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/sessionmessages.cc
@@ -359,7 +359,7 @@ bool ParseGingleTransportInfos(const buzz::XmlElement* action_elem,
   // If we don't have media, no need to separate the candidates.
   if (!has_audio && !has_video) {
     TransportInfo tinfo(CN_OTHER,
-        TransportDescription(NS_GINGLE_P2P, Candidates()));
+        TransportDescription(NS_GINGLE_P2P, std::string(), std::string()));
     if (!ParseGingleCandidates(action_elem, trans_parsers, translators,
                                CN_OTHER, &tinfo.description.candidates,
                                error)) {
@@ -371,10 +371,12 @@ bool ParseGingleTransportInfos(const buzz::XmlElement* action_elem,
   }
 
   // If we have media, separate the candidates.
-  TransportInfo audio_tinfo(CN_AUDIO,
-                            TransportDescription(NS_GINGLE_P2P, Candidates()));
-  TransportInfo video_tinfo(CN_VIDEO,
-                            TransportDescription(NS_GINGLE_P2P, Candidates()));
+  TransportInfo audio_tinfo(
+      CN_AUDIO,
+      TransportDescription(NS_GINGLE_P2P, std::string(), std::string()));
+  TransportInfo video_tinfo(
+      CN_VIDEO,
+      TransportDescription(NS_GINGLE_P2P, std::string(), std::string()));
   for (const buzz::XmlElement* candidate_elem = action_elem->FirstElement();
        candidate_elem != NULL;
        candidate_elem = candidate_elem->NextElement()) {
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/stunport.cc b/chromium/third_party/libjingle/source/talk/p2p/base/stunport.cc
index e182a51cbae..5e0e5002970 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/stunport.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/stunport.cc
@@ -216,8 +216,10 @@ Connection* UDPPort::CreateConnection(const Candidate& address,
 }
 
 int UDPPort::SendTo(const void* data, size_t size,
-                     const talk_base::SocketAddress& addr, bool payload) {
-  int sent = socket_->SendTo(data, size, addr);
+                    const talk_base::SocketAddress& addr,
+                    talk_base::DiffServCodePoint dscp,
+                    bool payload) {
+  int sent = socket_->SendTo(data, size, addr, dscp);
   if (sent < 0) {
     error_ = socket_->GetError();
     LOG_J(LS_ERROR, this) << "UDP send of " << size
@@ -227,6 +229,12 @@ int UDPPort::SendTo(const void* data, size_t size,
 }
 
 int UDPPort::SetOption(talk_base::Socket::Option opt, int value) {
+  // TODO(mallinath) - After we have the support on socket,
+  // remove this specialization.
+  if (opt == talk_base::Socket::OPT_DSCP) {
+    SetDefaultDscpValue(static_cast<talk_base::DiffServCodePoint>(value));
+    return 0;
+  }
   return socket_->SetOption(opt, value);
 }
 
@@ -254,8 +262,7 @@ void UDPPort::OnReadPacket(talk_base::AsyncPacketSocket* socket,
   // Even if the response doesn't match one of our outstanding requests, we
   // will eat it because it might be a response to a retransmitted packet, and
   // we already cleared the request when we got the first response.
-  ASSERT(!server_addr_.IsUnresolved());
-  if (remote_addr == server_addr_) {
+  if (!server_addr_.IsUnresolved() && remote_addr == server_addr_) {
     requests_.CheckResponse(data, size);
     return;
   }
@@ -346,7 +353,7 @@ void UDPPort::SetResult(bool success) {
 // TODO: merge this with SendTo above.
 void UDPPort::OnSendPacket(const void* data, size_t size, StunRequest* req) {
   StunBindingRequest* sreq = static_cast<StunBindingRequest*>(req);
-  if (socket_->SendTo(data, size, sreq->server_addr()) < 0)
+  if (socket_->SendTo(data, size, sreq->server_addr(), DefaultDscpValue()) < 0)
     PLOG(LERROR, socket_->GetError()) << "sendto";
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/stunport.h b/chromium/third_party/libjingle/source/talk/p2p/base/stunport.h
index 3f982d57be2..7cfed4b7cb8 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/stunport.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/stunport.h
@@ -121,7 +121,9 @@ class UDPPort : public Port {
   bool Init();
 
   virtual int SendTo(const void* data, size_t size,
-                     const talk_base::SocketAddress& addr, bool payload);
+                     const talk_base::SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp,
+                     bool payload);
 
   void OnLocalAddressReady(talk_base::AsyncPacketSocket* socket,
                            const talk_base::SocketAddress& address);
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/stunport_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/stunport_unittest.cc
index ba36c480ec9..3c1c6836f52 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/stunport_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/stunport_unittest.cc
@@ -71,10 +71,36 @@ class StunPortTest : public testing::Test,
         &StunPortTest::OnPortError);
   }
 
+  void CreateSharedStunPort(const talk_base::SocketAddress& server_addr) {
+    socket_.reset(socket_factory_.CreateUdpSocket(
+        talk_base::SocketAddress(kLocalAddr.ipaddr(), 0), 0, 0));
+    ASSERT_TRUE(socket_ != NULL);
+    socket_->SignalReadPacket.connect(this, &StunPortTest::OnReadPacket);
+    stun_port_.reset(cricket::UDPPort::Create(
+        talk_base::Thread::Current(), &network_, socket_.get(),
+        talk_base::CreateRandomString(16), talk_base::CreateRandomString(22)));
+    ASSERT_TRUE(stun_port_ != NULL);
+    stun_port_->set_server_addr(server_addr);
+    stun_port_->SignalPortComplete.connect(this,
+        &StunPortTest::OnPortComplete);
+    stun_port_->SignalPortError.connect(this,
+        &StunPortTest::OnPortError);
+  }
+
   void PrepareAddress() {
     stun_port_->PrepareAddress();
   }
 
+  void OnReadPacket(talk_base::AsyncPacketSocket* socket, const char* data,
+                    size_t size, const talk_base::SocketAddress& remote_addr) {
+    stun_port_->HandleIncomingPacket(socket, data, size, remote_addr);
+  }
+
+  void SendData(const char* data, size_t len) {
+    stun_port_->HandleIncomingPacket(
+        socket_.get(), data, len, talk_base::SocketAddress("22.22.22.22", 0));
+  }
+
  protected:
   static void SetUpTestCase() {
     // Ensure the RNG is inited.
@@ -96,8 +122,9 @@ class StunPortTest : public testing::Test,
  private:
   talk_base::Network network_;
   talk_base::BasicPacketSocketFactory socket_factory_;
-  talk_base::scoped_ptr<cricket::StunPort> stun_port_;
+  talk_base::scoped_ptr<cricket::UDPPort> stun_port_;
   talk_base::scoped_ptr<cricket::TestStunServer> stun_server_;
+  talk_base::scoped_ptr<talk_base::AsyncPacketSocket> socket_;
   bool done_;
   bool error_;
   int stun_keepalive_delay_;
@@ -164,3 +191,28 @@ TEST_F(StunPortTest, TestKeepAliveResponse) {
   ASSERT_EQ(1U, port()->Candidates().size());
 }
 
+// Test that a local candidate can be generated using a shared socket.
+TEST_F(StunPortTest, TestSharedSocketPrepareAddress) {
+  CreateSharedStunPort(kStunAddr);
+  PrepareAddress();
+  EXPECT_TRUE_WAIT(done(), kTimeoutMs);
+  ASSERT_EQ(1U, port()->Candidates().size());
+  EXPECT_TRUE(kLocalAddr.EqualIPs(port()->Candidates()[0].address()));
+}
+
+// Test that we still a get a local candidate with invalid stun server hostname.
+// Also verifing that UDPPort can receive packets when stun address can't be
+// resolved.
+TEST_F(StunPortTest, TestSharedSocketPrepareAddressInvalidHostname) {
+  CreateSharedStunPort(kBadHostnameAddr);
+  PrepareAddress();
+  EXPECT_TRUE_WAIT(done(), kTimeoutMs);
+  ASSERT_EQ(1U, port()->Candidates().size());
+  EXPECT_TRUE(kLocalAddr.EqualIPs(port()->Candidates()[0].address()));
+
+  // Send data to port after it's ready. This is to make sure, UDP port can
+  // handle data with unresolved stun server address.
+  std::string data = "some random data, sending to cricket::Port.";
+  SendData(data.c_str(), data.length());
+  // No crash is success.
+}
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/stunserver.cc b/chromium/third_party/libjingle/source/talk/p2p/base/stunserver.cc
index 05292e8e181..80719b4aa5e 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/stunserver.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/stunserver.cc
@@ -102,7 +102,8 @@ void StunServer::SendResponse(
     const StunMessage& msg, const talk_base::SocketAddress& addr) {
   talk_base::ByteBuffer buf;
   msg.Write(&buf);
-  if (socket_->SendTo(buf.Data(), buf.Length(), addr) < 0)
+  if (socket_->SendTo(
+      buf.Data(), buf.Length(), addr, talk_base::DSCP_NO_CHANGE) < 0)
     LOG_ERR(LS_ERROR) << "sendto";
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.cc b/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.cc
index 037abeccc88..11334c6225e 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.cc
@@ -67,6 +67,10 @@ bool TCPPort::Init() {
 
 TCPPort::~TCPPort() {
   delete socket_;
+  std::list<Incoming>::iterator it;
+  for (it = incoming_.begin(); it != incoming_.end(); ++it)
+    delete it->socket;
+  incoming_.clear();
 }
 
 Connection* TCPPort::CreateConnection(const Candidate& address,
@@ -130,7 +134,9 @@ void TCPPort::PrepareAddress() {
 }
 
 int TCPPort::SendTo(const void* data, size_t size,
-                    const talk_base::SocketAddress& addr, bool payload) {
+                    const talk_base::SocketAddress& addr,
+                    talk_base::DiffServCodePoint dscp,
+                    bool payload) {
   talk_base::AsyncPacketSocket * socket = NULL;
   if (TCPConnection * conn = static_cast<TCPConnection*>(GetConnection(addr))) {
     socket = conn->socket();
@@ -143,7 +149,7 @@ int TCPPort::SendTo(const void* data, size_t size,
     return -1;  // TODO: Set error_
   }
 
-  int sent = socket->Send(data, size);
+  int sent = socket->Send(data, size, dscp);
   if (sent < 0) {
     error_ = socket->GetError();
     LOG_J(LS_ERROR, this) << "TCP send of " << size
@@ -161,6 +167,14 @@ int TCPPort::GetOption(talk_base::Socket::Option opt, int* value) {
 }
 
 int TCPPort::SetOption(talk_base::Socket::Option opt, int value) {
+  // If we are setting DSCP value, pass value to base Port and return.
+  // TODO(mallinath) - After we have the support on socket,
+  // remove this specialization.
+  if (opt == talk_base::Socket::OPT_DSCP) {
+    SetDefaultDscpValue(static_cast<talk_base::DiffServCodePoint>(value));
+    return 0;
+  }
+
   if (socket_) {
     return socket_->SetOption(opt, value);
   } else {
@@ -257,7 +271,8 @@ TCPConnection::~TCPConnection() {
   delete socket_;
 }
 
-int TCPConnection::Send(const void* data, size_t size) {
+int TCPConnection::Send(const void* data, size_t size,
+                        talk_base::DiffServCodePoint dscp) {
   if (!socket_) {
     error_ = ENOTCONN;
     return SOCKET_ERROR;
@@ -268,7 +283,7 @@ int TCPConnection::Send(const void* data, size_t size) {
     error_ = EWOULDBLOCK;
     return SOCKET_ERROR;
   }
-  int sent = socket_->Send(data, size);
+  int sent = socket_->Send(data, size, dscp);
   if (sent < 0) {
     error_ = socket_->GetError();
   } else {
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.h b/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.h
index 8136176664c..599d3c66bef 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/tcpport.h
@@ -82,7 +82,9 @@ class TCPPort : public Port {
 
   // Handles sending using the local TCP socket.
   virtual int SendTo(const void* data, size_t size,
-                     const talk_base::SocketAddress& addr, bool payload);
+                     const talk_base::SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp,
+                     bool payload);
 
   // Accepts incoming TCP connection.
   void OnNewConnection(talk_base::AsyncPacketSocket* socket,
@@ -124,7 +126,8 @@ class TCPConnection : public Connection {
                 talk_base::AsyncPacketSocket* socket = 0);
   virtual ~TCPConnection();
 
-  virtual int Send(const void* data, size_t size);
+  virtual int Send(const void* data, size_t size,
+                   talk_base::DiffServCodePoint dscp);
   virtual int GetError();
 
   talk_base::AsyncPacketSocket* socket() { return socket_; }
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transport.cc b/chromium/third_party/libjingle/source/talk/p2p/base/transport.cc
index 6ccd90b02dd..3e4ad704068 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transport.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transport.cc
@@ -27,6 +27,7 @@
 
 #include "talk/p2p/base/transport.h"
 
+#include "talk/base/bind.h"
 #include "talk/base/common.h"
 #include "talk/base/logging.h"
 #include "talk/p2p/base/candidate.h"
@@ -39,27 +40,19 @@
 
 namespace cricket {
 
+using talk_base::Bind;
+
 enum {
-  MSG_CREATECHANNEL = 1,
-  MSG_DESTROYCHANNEL = 2,
-  MSG_DESTROYALLCHANNELS = 3,
-  MSG_CONNECTCHANNELS = 4,
-  MSG_RESETCHANNELS = 5,
-  MSG_ONSIGNALINGREADY = 6,
-  MSG_ONREMOTECANDIDATE = 7,
-  MSG_READSTATE = 8,
-  MSG_WRITESTATE = 9,
-  MSG_REQUESTSIGNALING = 10,
-  MSG_CANDIDATEREADY = 11,
-  MSG_ROUTECHANGE = 12,
-  MSG_CONNECTING = 13,
-  MSG_CANDIDATEALLOCATIONCOMPLETE = 14,
-  MSG_ROLECONFLICT = 15,
-  MSG_SETICEROLE = 16,
-  MSG_SETLOCALDESCRIPTION = 17,
-  MSG_SETREMOTEDESCRIPTION = 18,
-  MSG_GETSTATS = 19,
-  MSG_SETIDENTITY = 20,
+  MSG_ONSIGNALINGREADY = 1,
+  MSG_ONREMOTECANDIDATE,
+  MSG_READSTATE,
+  MSG_WRITESTATE,
+  MSG_REQUESTSIGNALING,
+  MSG_CANDIDATEREADY,
+  MSG_ROUTECHANGE,
+  MSG_CONNECTING,
+  MSG_CANDIDATEALLOCATIONCOMPLETE,
+  MSG_ROLECONFLICT,
 };
 
 struct ChannelParams : public talk_base::MessageData {
@@ -80,36 +73,6 @@ struct ChannelParams : public talk_base::MessageData {
   Candidate* candidate;
 };
 
-struct TransportDescriptionParams : public talk_base::MessageData {
-  TransportDescriptionParams(const TransportDescription& desc,
-                             ContentAction action)
-      : desc(desc), action(action), result(false) {}
-  const TransportDescription& desc;
-  ContentAction action;
-  bool result;
-};
-
-struct IceRoleParam : public talk_base::MessageData {
-  explicit IceRoleParam(IceRole role) : role(role) {}
-
-  IceRole role;
-};
-
-struct StatsParam : public talk_base::MessageData {
-  explicit StatsParam(TransportStats* stats)
-      : stats(stats), result(false) {}
-
-  TransportStats* stats;
-  bool result;
-};
-
-struct IdentityParam : public talk_base::MessageData {
-  explicit IdentityParam(talk_base::SSLIdentity* identity)
-      : identity(identity) {}
-
-  talk_base::SSLIdentity* identity;
-};
-
 Transport::Transport(talk_base::Thread* signaling_thread,
                      talk_base::Thread* worker_thread,
                      const std::string& content_name,
@@ -137,33 +100,28 @@ Transport::~Transport() {
 }
 
 void Transport::SetIceRole(IceRole role) {
-  IceRoleParam param(role);
-  worker_thread()->Send(this, MSG_SETICEROLE, &param);
+  worker_thread_->Invoke<void>(Bind(&Transport::SetIceRole_w, this, role));
 }
 
 void Transport::SetIdentity(talk_base::SSLIdentity* identity) {
-  IdentityParam params(identity);
-  worker_thread()->Send(this, MSG_SETIDENTITY, &params);
+  worker_thread_->Invoke<void>(Bind(&Transport::SetIdentity_w, this, identity));
 }
 
 bool Transport::SetLocalTransportDescription(
     const TransportDescription& description, ContentAction action) {
-  TransportDescriptionParams params(description, action);
-  worker_thread()->Send(this, MSG_SETLOCALDESCRIPTION, &params);
-  return params.result;
+  return worker_thread_->Invoke<bool>(Bind(
+      &Transport::SetLocalTransportDescription_w, this, description, action));
 }
 
 bool Transport::SetRemoteTransportDescription(
     const TransportDescription& description, ContentAction action) {
-  TransportDescriptionParams params(description, action);
-  worker_thread()->Send(this, MSG_SETREMOTEDESCRIPTION, &params);
-  return params.result;
+  return worker_thread_->Invoke<bool>(Bind(
+      &Transport::SetRemoteTransportDescription_w, this, description, action));
 }
 
 TransportChannelImpl* Transport::CreateChannel(int component) {
-  ChannelParams params(component);
-  worker_thread()->Send(this, MSG_CREATECHANNEL, &params);
-  return params.channel;
+  return worker_thread_->Invoke<TransportChannelImpl*>(Bind(
+      &Transport::CreateChannel_w, this, component));
 }
 
 TransportChannelImpl* Transport::CreateChannel_w(int component) {
@@ -235,8 +193,8 @@ bool Transport::HasChannels() {
 }
 
 void Transport::DestroyChannel(int component) {
-  ChannelParams params(component);
-  worker_thread()->Send(this, MSG_DESTROYCHANNEL, &params);
+  worker_thread_->Invoke<void>(Bind(
+      &Transport::DestroyChannel_w, this, component));
 }
 
 void Transport::DestroyChannel_w(int component) {
@@ -270,7 +228,7 @@ void Transport::DestroyChannel_w(int component) {
 
 void Transport::ConnectChannels() {
   ASSERT(signaling_thread()->IsCurrent());
-  worker_thread()->Send(this, MSG_CONNECTCHANNELS, NULL);
+  worker_thread_->Invoke<void>(Bind(&Transport::ConnectChannels_w, this));
 }
 
 void Transport::ConnectChannels_w() {
@@ -293,7 +251,8 @@ void Transport::ConnectChannels_w() {
     TransportDescription desc(NS_GINGLE_P2P, std::vector<std::string>(),
                               talk_base::CreateRandomString(ICE_UFRAG_LENGTH),
                               talk_base::CreateRandomString(ICE_PWD_LENGTH),
-                              ICEMODE_FULL, NULL, Candidates());
+                              ICEMODE_FULL, CONNECTIONROLE_NONE, NULL,
+                              Candidates());
     SetLocalTransportDescription_w(desc, CA_OFFER);
   }
 
@@ -310,7 +269,8 @@ void Transport::OnConnecting_s() {
 
 void Transport::DestroyAllChannels() {
   ASSERT(signaling_thread()->IsCurrent());
-  worker_thread()->Send(this, MSG_DESTROYALLCHANNELS, NULL);
+  worker_thread_->Invoke<void>(
+      Bind(&Transport::DestroyAllChannels_w, this));
   worker_thread()->Clear(this);
   signaling_thread()->Clear(this);
   destroyed_ = true;
@@ -338,7 +298,7 @@ void Transport::DestroyAllChannels_w() {
 
 void Transport::ResetChannels() {
   ASSERT(signaling_thread()->IsCurrent());
-  worker_thread()->Send(this, MSG_RESETCHANNELS, NULL);
+  worker_thread_->Invoke<void>(Bind(&Transport::ResetChannels_w, this));
 }
 
 void Transport::ResetChannels_w() {
@@ -402,9 +362,8 @@ bool Transport::VerifyCandidate(const Candidate& cand, std::string* error) {
 
 bool Transport::GetStats(TransportStats* stats) {
   ASSERT(signaling_thread()->IsCurrent());
-  StatsParam params(stats);
-  worker_thread()->Send(this, MSG_GETSTATS, &params);
-  return params.result;
+  return worker_thread_->Invoke<bool>(Bind(
+      &Transport::GetStats_w, this, stats));
 }
 
 bool Transport::GetStats_w(TransportStats* stats) {
@@ -424,6 +383,11 @@ bool Transport::GetStats_w(TransportStats* stats) {
   return true;
 }
 
+bool Transport::GetSslRole(talk_base::SSLRole* ssl_role) const {
+  return worker_thread_->Invoke<bool>(Bind(
+      &Transport::GetSslRole_w, this, ssl_role));
+}
+
 void Transport::OnRemoteCandidates(const std::vector<Candidate>& candidates) {
   for (std::vector<Candidate>::const_iterator iter = candidates.begin();
        iter != candidates.end();
@@ -668,19 +632,20 @@ bool Transport::ApplyRemoteTransportDescription_w(TransportChannelImpl* ch) {
   return true;
 }
 
-void Transport::ApplyNegotiatedTransportDescription_w(
+bool Transport::ApplyNegotiatedTransportDescription_w(
     TransportChannelImpl* channel) {
   channel->SetIceProtocolType(protocol_);
   channel->SetRemoteIceMode(remote_ice_mode_);
+  return true;
 }
 
-bool Transport::NegotiateTransportDescription_w(ContentAction local_role_) {
+bool Transport::NegotiateTransportDescription_w(ContentAction local_role) {
   // TODO(ekr@rtfm.com): This is ICE-specific stuff. Refactor into
   // P2PTransport.
   const TransportDescription* offer;
   const TransportDescription* answer;
 
-  if (local_role_ == CA_OFFER) {
+  if (local_role == CA_OFFER) {
     offer = local_description_.get();
     answer = remote_description_.get();
   } else {
@@ -724,32 +689,14 @@ bool Transport::NegotiateTransportDescription_w(ContentAction local_role_) {
   for (ChannelMap::iterator iter = channels_.begin();
        iter != channels_.end();
        ++iter) {
-    ApplyNegotiatedTransportDescription_w(iter->second.get());
+    if (!ApplyNegotiatedTransportDescription_w(iter->second.get()))
+      return false;
   }
   return true;
 }
 
 void Transport::OnMessage(talk_base::Message* msg) {
   switch (msg->message_id) {
-    case MSG_CREATECHANNEL: {
-        ChannelParams* params = static_cast<ChannelParams*>(msg->pdata);
-        params->channel = CreateChannel_w(params->component);
-      }
-      break;
-    case MSG_DESTROYCHANNEL: {
-        ChannelParams* params = static_cast<ChannelParams*>(msg->pdata);
-        DestroyChannel_w(params->component);
-      }
-      break;
-    case MSG_CONNECTCHANNELS:
-      ConnectChannels_w();
-      break;
-    case MSG_RESETCHANNELS:
-      ResetChannels_w();
-      break;
-    case MSG_DESTROYALLCHANNELS:
-      DestroyAllChannels_w();
-      break;
     case MSG_ONSIGNALINGREADY:
       CallChannels_w(&TransportChannelImpl::OnSignalingReady);
       break;
@@ -789,36 +736,6 @@ void Transport::OnMessage(talk_base::Message* msg) {
     case MSG_ROLECONFLICT:
       SignalRoleConflict();
       break;
-    case MSG_SETICEROLE: {
-        IceRoleParam* param =
-            static_cast<IceRoleParam*>(msg->pdata);
-        SetIceRole_w(param->role);
-      }
-      break;
-    case MSG_SETLOCALDESCRIPTION: {
-        TransportDescriptionParams* params =
-            static_cast<TransportDescriptionParams*>(msg->pdata);
-        params->result = SetLocalTransportDescription_w(params->desc,
-                                                        params->action);
-      }
-      break;
-    case MSG_SETREMOTEDESCRIPTION: {
-        TransportDescriptionParams* params =
-            static_cast<TransportDescriptionParams*>(msg->pdata);
-        params->result = SetRemoteTransportDescription_w(params->desc,
-                                                         params->action);
-      }
-      break;
-    case MSG_GETSTATS: {
-        StatsParam* params = static_cast<StatsParam*>(msg->pdata);
-        params->result = GetStats_w(params->stats);
-      }
-      break;
-    case MSG_SETIDENTITY: {
-        IdentityParam* params = static_cast<IdentityParam*>(msg->pdata);
-        SetIdentity_w(params->identity);
-      }
-      break;
   }
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transport.h b/chromium/third_party/libjingle/source/talk/p2p/base/transport.h
index 63c37343abe..381215f5d1c 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transport.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transport.h
@@ -52,6 +52,7 @@
 #include "talk/base/criticalsection.h"
 #include "talk/base/messagequeue.h"
 #include "talk/base/sigslot.h"
+#include "talk/base/sslstreamadapter.h"
 #include "talk/p2p/base/candidate.h"
 #include "talk/p2p/base/constants.h"
 #include "talk/p2p/base/sessiondescription.h"
@@ -323,6 +324,8 @@ class Transport : public talk_base::MessageHandler,
   // Forwards the signal from TransportChannel to BaseSession.
   sigslot::signal0<> SignalRoleConflict;
 
+  virtual bool GetSslRole(talk_base::SSLRole* ssl_role) const;
+
  protected:
   // These are called by Create/DestroyChannel above in order to create or
   // destroy the appropriate type of channel.
@@ -366,9 +369,13 @@ class Transport : public talk_base::MessageHandler,
   // Pushes down the transport parameters obtained via negotiation.
   // Derived classes can set their specific parameters here, but must call the
   // base as well.
-  virtual void ApplyNegotiatedTransportDescription_w(
+  virtual bool ApplyNegotiatedTransportDescription_w(
       TransportChannelImpl* channel);
 
+  virtual bool GetSslRole_w(talk_base::SSLRole* ssl_role) const {
+    return false;
+  }
+
  private:
   struct ChannelMapEntry {
     ChannelMapEntry() : impl_(NULL), candidates_allocated_(false), ref_(0) {}
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transport_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/transport_unittest.cc
index f9bebdfa535..e3b7badfff7 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transport_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transport_unittest.cc
@@ -145,8 +145,7 @@ TEST_F(TransportTest, TestChannelIceParameters) {
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLING);
   transport_->SetIceTiebreaker(99U);
   cricket::TransportDescription local_desc(
-      cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
-      kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL, NULL, cricket::Candidates());
+      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
                                                        cricket::CA_OFFER));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, transport_->ice_role());
@@ -157,8 +156,7 @@ TEST_F(TransportTest, TestChannelIceParameters) {
   EXPECT_EQ(kIcePwd1, channel_->ice_pwd());
 
   cricket::TransportDescription remote_desc(
-      cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
-      kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL, NULL, cricket::Candidates());
+      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
                                                         cricket::CA_ANSWER));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel_->GetIceRole());
@@ -177,12 +175,12 @@ TEST_F(TransportTest, TestSetRemoteIceLiteInOffer) {
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLED);
   cricket::TransportDescription remote_desc(
       cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
-      kIceUfrag1, kIcePwd1, cricket::ICEMODE_LITE, NULL, cricket::Candidates());
+      kIceUfrag1, kIcePwd1, cricket::ICEMODE_LITE,
+      cricket::CONNECTIONROLE_ACTPASS, NULL, cricket::Candidates());
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
                                                         cricket::CA_OFFER));
   cricket::TransportDescription local_desc(
-      cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
-      kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL, NULL, cricket::Candidates());
+      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
                                                        cricket::CA_ANSWER));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, transport_->ice_role());
@@ -195,8 +193,7 @@ TEST_F(TransportTest, TestSetRemoteIceLiteInOffer) {
 TEST_F(TransportTest, TestSetRemoteIceLiteInAnswer) {
   transport_->SetIceRole(cricket::ICEROLE_CONTROLLING);
   cricket::TransportDescription local_desc(
-      cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
-      kIceUfrag1, kIcePwd1, cricket::ICEMODE_FULL, NULL, cricket::Candidates());
+      cricket::NS_JINGLE_ICE_UDP, kIceUfrag1, kIcePwd1);
   ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
                                                        cricket::CA_OFFER));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, transport_->ice_role());
@@ -206,7 +203,8 @@ TEST_F(TransportTest, TestSetRemoteIceLiteInAnswer) {
   EXPECT_EQ(cricket::ICEMODE_FULL, channel_->remote_ice_mode());
   cricket::TransportDescription remote_desc(
       cricket::NS_JINGLE_ICE_UDP, std::vector<std::string>(),
-      kIceUfrag1, kIcePwd1, cricket::ICEMODE_LITE, NULL, cricket::Candidates());
+      kIceUfrag1, kIcePwd1, cricket::ICEMODE_LITE,
+      cricket::CONNECTIONROLE_NONE, NULL, cricket::Candidates());
   ASSERT_TRUE(transport_->SetRemoteTransportDescription(remote_desc,
                                                         cricket::CA_ANSWER));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING, channel_->GetIceRole());
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannel.h b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannel.h
index 2b09f54b83a..85fff7a9f92 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannel.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannel.h
@@ -32,6 +32,7 @@
 #include <vector>
 
 #include "talk/base/basictypes.h"
+#include "talk/base/dscp.h"
 #include "talk/base/sigslot.h"
 #include "talk/base/socket.h"
 #include "talk/base/sslidentity.h"
@@ -80,7 +81,9 @@ class TransportChannel : public sigslot::has_slots<> {
 
   // Attempts to send the given packet.  The return value is < 0 on failure.
   // TODO: Remove the default argument once channel code is updated.
-  virtual int SendPacket(const char* data, size_t len, int flags = 0) = 0;
+  virtual int SendPacket(const char* data, size_t len,
+                         talk_base::DiffServCodePoint dscp,
+                         int flags = 0) = 0;
 
   // Sets a socket option on this channel.  Note that not all options are
   // supported by all transport types.
@@ -89,30 +92,20 @@ class TransportChannel : public sigslot::has_slots<> {
   // Returns the most recent error that occurred on this channel.
   virtual int GetError() = 0;
 
-  // TODO(mallinath) - Move this to TransportChannelImpl, after channel.cc
-  // no longer needs it.
-  // Returns current transportchannel ICE role.
-  virtual IceRole GetIceRole() const = 0;
-
   // Returns the current stats for this connection.
-  virtual bool GetStats(ConnectionInfos* infos) {
-    return false;
-  }
+  virtual bool GetStats(ConnectionInfos* infos) = 0;
 
   // Is DTLS active?
-  virtual bool IsDtlsActive() const {
-    return false;
-  }
+  virtual bool IsDtlsActive() const = 0;
+
+  // Default implementation.
+  virtual bool GetSslRole(talk_base::SSLRole* role) const = 0;
 
   // Set up the ciphers to use for DTLS-SRTP.
-  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) {
-    return false;
-  }
+  virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers) = 0;
 
   // Find out which DTLS-SRTP cipher was negotiated
-  virtual bool GetSrtpCipher(std::string* cipher) {
-    return false;
-  }
+  virtual bool GetSrtpCipher(std::string* cipher) = 0;
 
   // Allows key material to be extracted for external encryption.
   virtual bool ExportKeyingMaterial(const std::string& label,
@@ -120,9 +113,7 @@ class TransportChannel : public sigslot::has_slots<> {
       size_t context_len,
       bool use_context,
       uint8* result,
-      size_t result_len) {
-    return false;
-  }
+      size_t result_len) = 0;
 
   // Signalled each time a packet is received on this channel.
   sigslot::signal4<TransportChannel*, const char*,
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelimpl.h b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelimpl.h
index 32e2471f120..cde2441307b 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelimpl.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelimpl.h
@@ -50,6 +50,7 @@ class TransportChannelImpl : public TransportChannel {
   virtual Transport* GetTransport() = 0;
 
   // For ICE channels.
+  virtual IceRole GetIceRole() const = 0;
   virtual void SetIceRole(IceRole role) = 0;
   virtual void SetIceTiebreaker(uint64 tiebreaker) = 0;
   // To toggle G-ICE/ICE.
@@ -93,16 +94,14 @@ class TransportChannelImpl : public TransportChannel {
 
   // DTLS methods
   // Set DTLS local identity.
-  virtual bool SetLocalIdentity(talk_base::SSLIdentity* identity) {
-    return false;
-  }
+  virtual bool SetLocalIdentity(talk_base::SSLIdentity* identity) = 0;
 
   // Set DTLS Remote fingerprint. Must be after local identity set.
   virtual bool SetRemoteFingerprint(const std::string& digest_alg,
     const uint8* digest,
-    size_t digest_len) {
-    return false;
-  }
+    size_t digest_len) = 0;
+
+  virtual bool SetSslRole(talk_base::SSLRole role) = 0;
 
   // TransportChannel is forwarding this signal from PortAllocatorSession.
   sigslot::signal1<TransportChannelImpl*> SignalCandidatesAllocationDone;
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.cc b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.cc
index b25f75751d6..04b32ce6499 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.cc
@@ -93,13 +93,15 @@ void TransportChannelProxy::SetImplementation(TransportChannelImpl* impl) {
   worker_thread_->Post(this, MSG_UPDATESTATE);
 }
 
-int TransportChannelProxy::SendPacket(const char* data, size_t len, int flags) {
+int TransportChannelProxy::SendPacket(const char* data, size_t len,
+                                      talk_base::DiffServCodePoint dscp,
+                                      int flags) {
   ASSERT(talk_base::Thread::Current() == worker_thread_);
   // Fail if we don't have an impl yet.
   if (!impl_) {
     return -1;
   }
-  return impl_->SendPacket(data, len, flags);
+  return impl_->SendPacket(data, len, dscp, flags);
 }
 
 int TransportChannelProxy::SetOption(talk_base::Socket::Option opt, int value) {
@@ -135,6 +137,22 @@ bool TransportChannelProxy::IsDtlsActive() const {
   return impl_->IsDtlsActive();
 }
 
+bool TransportChannelProxy::GetSslRole(talk_base::SSLRole* role) const {
+  ASSERT(talk_base::Thread::Current() == worker_thread_);
+  if (!impl_) {
+    return false;
+  }
+  return impl_->GetSslRole(role);
+}
+
+bool TransportChannelProxy::SetSslRole(talk_base::SSLRole role) {
+  ASSERT(talk_base::Thread::Current() == worker_thread_);
+  if (!impl_) {
+    return false;
+  }
+  return impl_->SetSslRole(role);
+}
+
 bool TransportChannelProxy::SetSrtpCiphers(const std::vector<std::string>&
                                            ciphers) {
   ASSERT(talk_base::Thread::Current() == worker_thread_);
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.h b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.h
index 828c0ae05b4..29f46634198 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportchannelproxy.h
@@ -63,12 +63,16 @@ class TransportChannelProxy : public TransportChannel,
 
   // Implementation of the TransportChannel interface.  These simply forward to
   // the implementation.
-  virtual int SendPacket(const char* data, size_t len, int flags);
+  virtual int SendPacket(const char* data, size_t len,
+                         talk_base::DiffServCodePoint dscp,
+                         int flags);
   virtual int SetOption(talk_base::Socket::Option opt, int value);
   virtual int GetError();
   virtual IceRole GetIceRole() const;
   virtual bool GetStats(ConnectionInfos* infos);
   virtual bool IsDtlsActive() const;
+  virtual bool GetSslRole(talk_base::SSLRole* role) const;
+  virtual bool SetSslRole(talk_base::SSLRole role);
   virtual bool SetSrtpCiphers(const std::vector<std::string>& ciphers);
   virtual bool GetSrtpCipher(std::string* cipher);
   virtual bool ExportKeyingMaterial(const std::string& label,
diff --git a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaSource.java b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescription.cc
index 29490490006..56873427284 100644
--- a/chromium/third_party/libjingle/source/talk/app/webrtc/java/src/org/webrtc/MediaSource.java
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescription.cc
@@ -1,6 +1,6 @@
 /*
  * libjingle
- * Copyright 2013, Google Inc.
+ * Copyright 2013 Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
@@ -25,31 +25,28 @@
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+#include "talk/p2p/base/transportdescription.h"
 
-package org.webrtc;
+#include "talk/p2p/base/constants.h"
 
-/** Java wrapper for a C++ MediaSourceInterface. */
-public class MediaSource {
-  /** Tracks MediaSourceInterface.SourceState */
-  public enum State {
-    INITIALIZING, LIVE, ENDED, MUTED
-  }
-
-  final long nativeSource;  // Package-protected for PeerConnectionFactory.
+namespace cricket {
 
-  public MediaSource(long nativeSource) {
-    this.nativeSource = nativeSource;
-  }
+bool StringToConnectionRole(const std::string& role_str, ConnectionRole* role) {
+  const char* const roles[] = {
+      CONNECTIONROLE_ACTIVE_STR,
+      CONNECTIONROLE_PASSIVE_STR,
+      CONNECTIONROLE_ACTPASS_STR,
+      CONNECTIONROLE_HOLDCONN_STR
+  };
 
-  public State state() {
-    return nativeState(nativeSource);
-  }
-
-  void dispose() {
-    free(nativeSource);
+  for (size_t i = 0; i < ARRAY_SIZE(roles); ++i) {
+    if (stricmp(roles[i], role_str.c_str()) == 0) {
+      *role = static_cast<ConnectionRole>(CONNECTIONROLE_ACTIVE + i);
+      return true;
+    }
   }
+  return false;
+}
 
-  private static native State nativeState(long pointer);
+}  // namespace cricket
 
-  private static native void free(long nativeSource);
-}
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transportdescription.h b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescription.h
index 92da8d66c29..64fbb89a808 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transportdescription.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescription.h
@@ -75,6 +75,26 @@ enum IceMode {
   ICEMODE_LITE   // As defined in http://tools.ietf.org/html/rfc5245#section-4.2
 };
 
+// RFC 4145 - http://tools.ietf.org/html/rfc4145#section-4
+// 'active':  The endpoint will initiate an outgoing connection.
+// 'passive': The endpoint will accept an incoming connection.
+// 'actpass': The endpoint is willing to accept an incoming
+//            connection or to initiate an outgoing connection.
+enum ConnectionRole {
+  CONNECTIONROLE_NONE = 0,
+  CONNECTIONROLE_ACTIVE,
+  CONNECTIONROLE_PASSIVE,
+  CONNECTIONROLE_ACTPASS,
+  CONNECTIONROLE_HOLDCONN,
+};
+
+extern const char CONNECTIONROLE_ACTIVE_STR[];
+extern const char CONNECTIONROLE_PASSIVE_STR[];
+extern const char CONNECTIONROLE_ACTPASS_STR[];
+extern const char CONNECTIONROLE_HOLDCONN_STR[];
+
+bool StringToConnectionRole(const std::string& role_str, ConnectionRole* role);
+
 typedef std::vector<Candidate> Candidates;
 
 struct TransportDescription {
@@ -85,6 +105,7 @@ struct TransportDescription {
                        const std::string& ice_ufrag,
                        const std::string& ice_pwd,
                        IceMode ice_mode,
+                       ConnectionRole role,
                        const talk_base::SSLFingerprint* identity_fingerprint,
                        const Candidates& candidates)
       : transport_type(transport_type),
@@ -92,19 +113,24 @@ struct TransportDescription {
         ice_ufrag(ice_ufrag),
         ice_pwd(ice_pwd),
         ice_mode(ice_mode),
+        connection_role(role),
         identity_fingerprint(CopyFingerprint(identity_fingerprint)),
         candidates(candidates) {}
   TransportDescription(const std::string& transport_type,
-                       const Candidates& candidates)
+                       const std::string& ice_ufrag,
+                       const std::string& ice_pwd)
       : transport_type(transport_type),
+        ice_ufrag(ice_ufrag),
+        ice_pwd(ice_pwd),
         ice_mode(ICEMODE_FULL),
-        candidates(candidates) {}
+        connection_role(CONNECTIONROLE_NONE) {}
   TransportDescription(const TransportDescription& from)
       : transport_type(from.transport_type),
         transport_options(from.transport_options),
         ice_ufrag(from.ice_ufrag),
         ice_pwd(from.ice_pwd),
         ice_mode(from.ice_mode),
+        connection_role(from.connection_role),
         identity_fingerprint(CopyFingerprint(from.identity_fingerprint.get())),
         candidates(from.candidates) {}
 
@@ -118,6 +144,7 @@ struct TransportDescription {
     ice_ufrag = from.ice_ufrag;
     ice_pwd = from.ice_pwd;
     ice_mode = from.ice_mode;
+    connection_role = from.connection_role;
 
     identity_fingerprint.reset(CopyFingerprint(
         from.identity_fingerprint.get()));
@@ -147,6 +174,7 @@ struct TransportDescription {
   std::string ice_ufrag;
   std::string ice_pwd;
   IceMode ice_mode;
+  ConnectionRole connection_role;
 
   talk_base::scoped_ptr<talk_base::SSLFingerprint> identity_fingerprint;
   Candidates candidates;
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.cc b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.cc
index 8fbfff144fa..0c129437ccb 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.cc
@@ -73,10 +73,12 @@ TransportDescription* TransportDescriptionFactory::CreateOffer(
   // If we are trying to establish a secure transport, add a fingerprint.
   if (secure_ == SEC_ENABLED || secure_ == SEC_REQUIRED) {
     // Fail if we can't create the fingerprint.
-    if (!CreateIdentityDigest(desc.get())) {
+    // If we are the initiator set role to "actpass".
+    if (!SetSecurityInfo(desc.get(), CONNECTIONROLE_ACTPASS)) {
       return NULL;
     }
   }
+
   return desc.release();
 }
 
@@ -101,7 +103,7 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
     desc->transport_type = NS_GINGLE_P2P;
     // Offer is hybrid, we support GICE: use GICE.
   } else if ((!offer || offer->transport_type == NS_GINGLE_P2P) &&
-           (protocol_ == ICEPROTO_HYBRID || protocol_ == ICEPROTO_GOOGLE)) {
+             (protocol_ == ICEPROTO_HYBRID || protocol_ == ICEPROTO_GOOGLE)) {
     // Offer is GICE, we support hybrid or GICE: use GICE.
     desc->transport_type = NS_GINGLE_P2P;
   } else {
@@ -126,7 +128,11 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
     // The offer supports DTLS, so answer with DTLS, as long as we support it.
     if (secure_ == SEC_ENABLED || secure_ == SEC_REQUIRED) {
       // Fail if we can't create the fingerprint.
-      if (!CreateIdentityDigest(desc.get())) {
+      // Setting DTLS role to active.
+      ConnectionRole role = (options.prefer_passive_role) ?
+          CONNECTIONROLE_PASSIVE : CONNECTIONROLE_ACTIVE;
+
+      if (!SetSecurityInfo(desc.get(), role)) {
         return NULL;
       }
     }
@@ -140,8 +146,8 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
   return desc.release();
 }
 
-bool TransportDescriptionFactory::CreateIdentityDigest(
-    TransportDescription* desc) const {
+bool TransportDescriptionFactory::SetSecurityInfo(
+    TransportDescription* desc, ConnectionRole role) const {
   if (!identity_) {
     LOG(LS_ERROR) << "Cannot create identity digest with no identity";
     return false;
@@ -154,6 +160,8 @@ bool TransportDescriptionFactory::CreateIdentityDigest(
     return false;
   }
 
+  // Assign security role.
+  desc->connection_role = role;
   return true;
 }
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.h b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.h
index 32836f3e814..ddf27998188 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/transportdescriptionfactory.h
@@ -37,8 +37,9 @@ class SSLIdentity;
 namespace cricket {
 
 struct TransportOptions {
-  TransportOptions() : ice_restart(false) {}
+  TransportOptions() : ice_restart(false), prefer_passive_role(false) {}
   bool ice_restart;
+  bool prefer_passive_role;
 };
 
 // Creates transport descriptions according to the supplied configuration.
@@ -71,7 +72,8 @@ class TransportDescriptionFactory {
       const TransportDescription* current_description) const;
 
  private:
-  bool CreateIdentityDigest(TransportDescription* description) const;
+  bool SetSecurityInfo(TransportDescription* description,
+                       ConnectionRole role) const;
 
   TransportProtocol protocol_;
   SecurePolicy secure_;
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/turnport.cc b/chromium/third_party/libjingle/source/talk/p2p/base/turnport.cc
index a302b713524..35e51fc2da8 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/turnport.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/turnport.cc
@@ -52,10 +52,6 @@ static const int TURN_PERMISSION_TIMEOUT = 5 * 60 * 1000;  // 5 minutes
 
 static const size_t TURN_CHANNEL_HEADER_SIZE = 4U;
 
-enum {
-  MSG_PORT_ERROR = 1
-};
-
 inline bool IsTurnChannelData(uint16 msg_type) {
   return ((msg_type & 0xC000) == 0x4000);  // MSB are 0b01
 }
@@ -156,7 +152,8 @@ class TurnEntry : public sigslot::has_slots<> {
   void SendChannelBindRequest(int delay);
   // Sends a packet to the given destination address.
   // This will wrap the packet in STUN if necessary.
-  int Send(const void* data, size_t size, bool payload);
+  int Send(const void* data, size_t size, bool payload,
+           talk_base::DiffServCodePoint dscp);
 
   void OnCreatePermissionSuccess();
   void OnCreatePermissionError(StunMessage* response, int code);
@@ -296,6 +293,14 @@ Connection* TurnPort::CreateConnection(const Candidate& address,
 }
 
 int TurnPort::SetOption(talk_base::Socket::Option opt, int value) {
+  // DSCP option is not passed to the socket.
+  // TODO(mallinath) - After we have the support on socket,
+  // remove this specialization.
+  if (opt == talk_base::Socket::OPT_DSCP) {
+    SetDefaultDscpValue(static_cast<talk_base::DiffServCodePoint>(value));
+    return 0;
+  }
+
   if (!socket_) {
     // If socket is not created yet, these options will be applied during socket
     // creation.
@@ -318,6 +323,7 @@ int TurnPort::GetError() {
 
 int TurnPort::SendTo(const void* data, size_t size,
                      const talk_base::SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp,
                      bool payload) {
   // Try to find an entry for this specific address; we should have one.
   TurnEntry* entry = FindEntry(addr);
@@ -332,7 +338,7 @@ int TurnPort::SendTo(const void* data, size_t size,
   }
 
   // Send the actual contents to the server using the usual mechanism.
-  int sent = entry->Send(data, size, payload);
+  int sent = entry->Send(data, size, payload, dscp);
   if (sent <= 0) {
     return SOCKET_ERROR;
   }
@@ -406,7 +412,7 @@ void TurnPort::OnResolveResult(talk_base::SignalThread* signal_thread) {
 
 void TurnPort::OnSendStunPacket(const void* data, size_t size,
                                 StunRequest* request) {
-  if (Send(data, size) < 0) {
+  if (Send(data, size, DefaultDscpValue()) < 0) {
     LOG_J(LS_ERROR, this) << "Failed to send TURN message, err="
                           << socket_->GetError();
   }
@@ -431,15 +437,16 @@ void TurnPort::OnAllocateError() {
   // We will send SignalPortError asynchronously as this can be sent during
   // port initialization. This way it will not be blocking other port
   // creation.
-  thread()->Post(this, MSG_PORT_ERROR);
+  thread()->Post(this, MSG_ERROR);
 }
 
 void TurnPort::OnMessage(talk_base::Message* message) {
-  if (message->message_id == MSG_PORT_ERROR) {
+  if (message->message_id == MSG_ERROR) {
     SignalPortError(this);
-  } else {
-    Port::OnMessage(message);
+    return;
   }
+
+  Port::OnMessage(message);
 }
 
 void TurnPort::OnAllocateRequestTimeout() {
@@ -557,8 +564,9 @@ void TurnPort::AddRequestAuthInfo(StunMessage* msg) {
   VERIFY(msg->AddMessageIntegrity(hash()));
 }
 
-int TurnPort::Send(const void* data, size_t len) {
-  return socket_->SendTo(data, len, server_address_.address);
+int TurnPort::Send(const void* data, size_t len,
+                   talk_base::DiffServCodePoint dscp) {
+  return socket_->SendTo(data, len, server_address_.address, dscp);
 }
 
 void TurnPort::UpdateHash() {
@@ -890,7 +898,8 @@ void TurnEntry::SendChannelBindRequest(int delay) {
       port_, this, channel_id_, ext_addr_), delay);
 }
 
-int TurnEntry::Send(const void* data, size_t size, bool payload) {
+int TurnEntry::Send(const void* data, size_t size, bool payload,
+                    talk_base::DiffServCodePoint dscp) {
   talk_base::ByteBuffer buf;
   if (state_ != STATE_BOUND) {
     // If we haven't bound the channel yet, we have to use a Send Indication.
@@ -915,7 +924,7 @@ int TurnEntry::Send(const void* data, size_t size, bool payload) {
     buf.WriteUInt16(static_cast<uint16>(size));
     buf.WriteBytes(reinterpret_cast<const char*>(data), size);
   }
-  return port_->Send(buf.Data(), buf.Length());
+  return port_->Send(buf.Data(), buf.Length(), dscp);
 }
 
 void TurnEntry::OnCreatePermissionSuccess() {
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/turnport.h b/chromium/third_party/libjingle/source/talk/p2p/base/turnport.h
index fa23d53475b..4462b0c8c99 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/turnport.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/turnport.h
@@ -74,6 +74,7 @@ class TurnPort : public Port {
       const Candidate& c, PortInterface::CandidateOrigin origin);
   virtual int SendTo(const void* data, size_t size,
                      const talk_base::SocketAddress& addr,
+                     talk_base::DiffServCodePoint dscp,
                      bool payload);
   virtual int SetOption(talk_base::Socket::Option opt, int value);
   virtual int GetOption(talk_base::Socket::Option opt, int* value);
@@ -106,6 +107,8 @@ class TurnPort : public Port {
            const RelayCredentials& credentials);
 
  private:
+  enum { MSG_ERROR = MSG_FIRST_AVAILABLE };
+
   typedef std::list<TurnEntry*> EntryList;
   typedef std::map<talk_base::Socket::Option, int> SocketOptionsMap;
 
@@ -138,7 +141,7 @@ class TurnPort : public Port {
 
   bool ScheduleRefresh(int lifetime);
   void SendRequest(StunRequest* request, int delay);
-  int Send(const void* data, size_t size);
+  int Send(const void* data, size_t size, talk_base::DiffServCodePoint dscp);
   void UpdateHash();
   bool UpdateNonce(StunMessage* response);
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/turnport_unittest.cc b/chromium/third_party/libjingle/source/talk/p2p/base/turnport_unittest.cc
index 6304ce67898..726175c5c2f 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/turnport_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/turnport_unittest.cc
@@ -27,6 +27,7 @@
 
 #include "talk/base/asynctcpsocket.h"
 #include "talk/base/buffer.h"
+#include "talk/base/dscp.h"
 #include "talk/base/firewallsocketserver.h"
 #include "talk/base/logging.h"
 #include "talk/base/gunit.h"
@@ -217,8 +218,8 @@ class TurnPortTest : public testing::Test,
       for (size_t j = 0; j < i + 1; ++j) {
         buf[j] = 0xFF - j;
       }
-      conn1->Send(buf, i + 1);
-      conn2->Send(buf, i + 1);
+      conn1->Send(buf, i + 1, talk_base::DSCP_NO_CHANGE);
+      conn2->Send(buf, i + 1, talk_base::DSCP_NO_CHANGE);
       main_->ProcessMessages(0);
     }
 
diff --git a/chromium/third_party/libjingle/source/talk/p2p/base/turnserver.cc b/chromium/third_party/libjingle/source/talk/p2p/base/turnserver.cc
index 8260f3dbe71..17ecf3507c6 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/base/turnserver.cc
+++ b/chromium/third_party/libjingle/source/talk/p2p/base/turnserver.cc
@@ -564,7 +564,8 @@ void TurnServer::SendStun(Connection* conn, StunMessage* msg) {
 
 void TurnServer::Send(Connection* conn,
                       const talk_base::ByteBuffer& buf) {
-  conn->socket()->SendTo(buf.Data(), buf.Length(), conn->src());
+  conn->socket()->SendTo(buf.Data(), buf.Length(), conn->src(),
+                         talk_base::DSCP_NO_CHANGE);
 }
 
 void TurnServer::OnAllocationDestroyed(Allocation* allocation) {
@@ -936,7 +937,7 @@ void TurnServer::Allocation::SendErrorResponse(const TurnMessage* req, int code,
 
 void TurnServer::Allocation::SendExternal(const void* data, size_t size,
                                   const talk_base::SocketAddress& peer) {
-  external_socket_->SendTo(data, size, peer);
+  external_socket_->SendTo(data, size, peer, talk_base::DSCP_NO_CHANGE);
 }
 
 void TurnServer::Allocation::OnMessage(talk_base::Message* msg) {
diff --git a/chromium/third_party/libjingle/source/talk/p2p/client/httpportallocator.h b/chromium/third_party/libjingle/source/talk/p2p/client/httpportallocator.h
index cb4c8f82bba..a0ef3b722de 100644
--- a/chromium/third_party/libjingle/source/talk/p2p/client/httpportallocator.h
+++ b/chromium/third_party/libjingle/source/talk/p2p/client/httpportallocator.h
@@ -32,7 +32,6 @@
 #include <string>
 #include <vector>
 
-#include "talk/base/gunit_prod.h"
 #include "talk/p2p/client/basicportallocator.h"
 
 class HttpPortAllocatorTest_TestSessionRequestUrl_Test;
@@ -129,6 +128,9 @@ class HttpPortAllocatorSessionBase : public BasicPortAllocatorSession {
   virtual void SendSessionRequest(const std::string& host, int port) = 0;
   virtual void ReceiveSessionResponse(const std::string& response);
 
+  // Made public for testing. Should be protected.
+  std::string GetSessionRequestUrl();
+
  protected:
   virtual void GetPortConfigurations();
   void TryCreateRelaySession();
@@ -137,11 +139,7 @@ class HttpPortAllocatorSessionBase : public BasicPortAllocatorSession {
         BasicPortAllocatorSession::allocator());
   }
 
-  std::string GetSessionRequestUrl();
-
  private:
-  FRIEND_TEST(::HttpPortAllocatorTest, TestSessionRequestUrl);
-
   std::vector<std::string> relay_hosts_;
   std::vector<talk_base::SocketAddress> stun_hosts_;
   std::string relay_token_;
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channel.cc b/chromium/third_party/libjingle/source/talk/session/media/channel.cc
index 1bce2acdccb..f6259e9a34b 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channel.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channel.cc
@@ -30,6 +30,7 @@
 #include "talk/base/buffer.h"
 #include "talk/base/byteorder.h"
 #include "talk/base/common.h"
+#include "talk/base/dscp.h"
 #include "talk/base/logging.h"
 #include "talk/media/base/rtputils.h"
 #include "talk/p2p/base/transportchannel.h"
@@ -55,6 +56,8 @@ enum {
   MSG_SETRENDERER,
   MSG_ADDRECVSTREAM,
   MSG_REMOVERECVSTREAM,
+  MSG_ADDSENDSTREAM,
+  MSG_REMOVESENDSTREAM,
   MSG_SETRINGBACKTONE,
   MSG_PLAYRINGBACKTONE,
   MSG_SETMAXSENDBANDWIDTH,
@@ -74,7 +77,7 @@ enum {
   MSG_DATARECEIVED,
   MSG_SETCAPTURER,
   MSG_ISSCREENCASTING,
-  MSG_SCREENCASTFPS,
+  MSG_GETSCREENCASTDETAILS,
   MSG_SETSCREENCASTFACTORY,
   MSG_FIRSTPACKETRECEIVED,
   MSG_SESSION_ERROR,
@@ -187,6 +190,7 @@ struct VideoStatsMessageData : public talk_base::MessageData {
 
 struct PacketMessageData : public talk_base::MessageData {
   talk_base::Buffer packet;
+  talk_base::DiffServCodePoint dscp;
 };
 
 struct AudioRenderMessageData: public talk_base::MessageData {
@@ -334,12 +338,14 @@ struct IsScreencastingMessageData : public talk_base::MessageData {
   bool result;
 };
 
-struct ScreencastFpsMessageData : public talk_base::MessageData {
-  explicit ScreencastFpsMessageData(uint32 s)
-      : ssrc(s), result(0) {
+struct VideoChannel::ScreencastDetailsMessageData :
+    public talk_base::MessageData {
+  explicit ScreencastDetailsMessageData(uint32 s)
+      : ssrc(s), fps(0), screencast_max_pixels(0) {
   }
   uint32 ssrc;
-  int result;
+  int fps;
+  int screencast_max_pixels;
 };
 
 struct SetScreenCaptureFactoryMessageData : public talk_base::MessageData {
@@ -480,6 +486,18 @@ bool BaseChannel::RemoveRecvStream(uint32 ssrc) {
   return data.result;
 }
 
+bool BaseChannel::AddSendStream(const StreamParams& sp) {
+  StreamMessageData data(sp);
+  Send(MSG_ADDSENDSTREAM, &data);
+  return data.result;
+}
+
+bool BaseChannel::RemoveSendStream(uint32 ssrc) {
+  SsrcMessageData data(ssrc);
+  Send(MSG_REMOVESENDSTREAM, &data);
+  return data.result;
+}
+
 bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
                                   ContentAction action) {
   SetContentData data(content, action);
@@ -550,12 +568,14 @@ bool BaseChannel::IsReadyToSend() const {
          was_ever_writable();
 }
 
-bool BaseChannel::SendPacket(talk_base::Buffer* packet) {
-  return SendPacket(false, packet);
+bool BaseChannel::SendPacket(talk_base::Buffer* packet,
+                             talk_base::DiffServCodePoint dscp) {
+  return SendPacket(false, packet, dscp);
 }
 
-bool BaseChannel::SendRtcp(talk_base::Buffer* packet) {
-  return SendPacket(true, packet);
+bool BaseChannel::SendRtcp(talk_base::Buffer* packet,
+                           talk_base::DiffServCodePoint dscp) {
+  return SendPacket(true, packet, dscp);
 }
 
 int BaseChannel::SetOption(SocketType type, talk_base::Socket::Option opt,
@@ -619,7 +639,8 @@ bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
           rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
 }
 
-bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet) {
+bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet,
+                             talk_base::DiffServCodePoint dscp) {
   // Unless we're sending optimistically, we only allow packets through when we
   // are completely writable.
   if (!optimistic_data_send_ && !writable_) {
@@ -638,6 +659,7 @@ bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet) {
     int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
     PacketMessageData* data = new PacketMessageData;
     packet->TransferTo(&data->packet);
+    data->dscp = dscp;
     worker_thread_->Post(this, message_id, data);
     return true;
   }
@@ -715,7 +737,7 @@ bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet) {
   }
 
   // Bon voyage.
-  int ret = channel->SendPacket(packet->data(), packet->length(),
+  int ret = channel->SendPacket(packet->data(), packet->length(), dscp,
       (secure() && secure_dtls()) ? PF_SRTP_BYPASS : 0);
   if (ret != static_cast<int>(packet->length())) {
     if (channel->GetError() == EWOULDBLOCK) {
@@ -1003,8 +1025,13 @@ bool BaseChannel::SetupDtlsSrtp(bool rtcp_channel) {
     &dtls_buffer[offset], SRTP_MASTER_KEY_SALT_LEN);
 
   std::vector<unsigned char> *send_key, *recv_key;
+  talk_base::SSLRole role;
+  if (!channel->GetSslRole(&role)) {
+    LOG(LS_WARNING) << "GetSslRole failed";
+    return false;
+  }
 
-  if (channel->GetIceRole() == ICEROLE_CONTROLLING) {
+  if (role == talk_base::SSL_SERVER) {
     send_key = &server_write_key;
     recv_key = &client_write_key;
   } else {
@@ -1144,6 +1171,16 @@ bool BaseChannel::RemoveRecvStream_w(uint32 ssrc) {
   return media_channel()->RemoveRecvStream(ssrc);
 }
 
+bool BaseChannel::AddSendStream_w(const StreamParams& sp) {
+  ASSERT(worker_thread() == talk_base::Thread::Current());
+  return media_channel()->AddSendStream(sp);
+}
+
+bool BaseChannel::RemoveSendStream_w(uint32 ssrc) {
+  ASSERT(worker_thread() == talk_base::Thread::Current());
+  return media_channel()->RemoveSendStream(ssrc);
+}
+
 bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
                                        ContentAction action) {
   if (!VERIFY(action == CA_OFFER || action == CA_ANSWER ||
@@ -1354,6 +1391,16 @@ void BaseChannel::OnMessage(talk_base::Message *pmsg) {
       data->result = RemoveRecvStream_w(data->ssrc);
       break;
     }
+    case MSG_ADDSENDSTREAM: {
+      StreamMessageData* data = static_cast<StreamMessageData*>(pmsg->pdata);
+      data->result = AddSendStream_w(data->sp);
+      break;
+    }
+    case MSG_REMOVESENDSTREAM: {
+      SsrcMessageData* data = static_cast<SsrcMessageData*>(pmsg->pdata);
+      data->result = RemoveSendStream_w(data->ssrc);
+      break;
+    }
     case MSG_SETMAXSENDBANDWIDTH: {
       SetBandwidthData* data = static_cast<SetBandwidthData*>(pmsg->pdata);
       data->result = SetMaxSendBandwidth_w(data->value);
@@ -1363,7 +1410,7 @@ void BaseChannel::OnMessage(talk_base::Message *pmsg) {
     case MSG_RTPPACKET:
     case MSG_RTCPPACKET: {
       PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
-      SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet);
+      SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet, data->dscp);
       delete data;  // because it is Posted
       break;
     }
@@ -1959,10 +2006,16 @@ bool VideoChannel::IsScreencasting() {
   return data.result;
 }
 
-int VideoChannel::ScreencastFps(uint32 ssrc) {
-  ScreencastFpsMessageData data(ssrc);
-  Send(MSG_SCREENCASTFPS, &data);
-  return data.result;
+int VideoChannel::GetScreencastFps(uint32 ssrc) {
+  ScreencastDetailsMessageData data(ssrc);
+  Send(MSG_GETSCREENCASTDETAILS, &data);
+  return data.fps;
+}
+
+int VideoChannel::GetScreencastMaxPixels(uint32 ssrc) {
+  ScreencastDetailsMessageData data(ssrc);
+  Send(MSG_GETSCREENCASTDETAILS, &data);
+  return data.screencast_max_pixels;
 }
 
 bool VideoChannel::SendIntraFrame() {
@@ -2179,14 +2232,16 @@ bool VideoChannel::IsScreencasting_w() const {
   return !screencast_capturers_.empty();
 }
 
-int VideoChannel::ScreencastFps_w(uint32 ssrc) const {
-  ScreencastMap::const_iterator iter = screencast_capturers_.find(ssrc);
+void VideoChannel::ScreencastDetails_w(
+    ScreencastDetailsMessageData* data) const {
+  ScreencastMap::const_iterator iter = screencast_capturers_.find(data->ssrc);
   if (iter == screencast_capturers_.end()) {
-    return 0;
+    return;
   }
   VideoCapturer* capturer = iter->second;
   const VideoFormat* video_format = capturer->GetCaptureFormat();
-  return VideoFormat::IntervalToFps(video_format->interval);
+  data->fps = VideoFormat::IntervalToFps(video_format->interval);
+  data->screencast_max_pixels = capturer->screencast_max_pixels();
 }
 
 void VideoChannel::SetScreenCaptureFactory_w(
@@ -2257,10 +2312,10 @@ void VideoChannel::OnMessage(talk_base::Message *pmsg) {
       data->result = IsScreencasting_w();
       break;
     }
-    case MSG_SCREENCASTFPS: {
-      ScreencastFpsMessageData* data =
-          static_cast<ScreencastFpsMessageData*>(pmsg->pdata);
-      data->result = ScreencastFps_w(data->ssrc);
+    case MSG_GETSCREENCASTDETAILS: {
+      ScreencastDetailsMessageData* data =
+          static_cast<ScreencastDetailsMessageData*>(pmsg->pdata);
+      ScreencastDetails_w(data);
       break;
     }
     case MSG_SENDINTRAFRAME: {
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channel.h b/chromium/third_party/libjingle/source/talk/session/media/channel.h
index eccadd32d60..0d66be9a90d 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channel.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/channel.h
@@ -119,6 +119,8 @@ class BaseChannel
   // Multiplexing
   bool AddRecvStream(const StreamParams& sp);
   bool RemoveRecvStream(uint32 ssrc);
+  bool AddSendStream(const StreamParams& sp);
+  bool RemoveSendStream(uint32 ssrc);
 
   // Monitoring
   void StartConnectionMonitor(int cms);
@@ -249,8 +251,10 @@ class BaseChannel
   void FlushRtcpMessages();
 
   // NetworkInterface implementation, called by MediaEngine
-  virtual bool SendPacket(talk_base::Buffer* packet);
-  virtual bool SendRtcp(talk_base::Buffer* packet);
+  virtual bool SendPacket(talk_base::Buffer* packet,
+                          talk_base::DiffServCodePoint dscp);
+  virtual bool SendRtcp(talk_base::Buffer* packet,
+                        talk_base::DiffServCodePoint dscp);
   virtual int SetOption(SocketType type, talk_base::Socket::Option o, int val);
 
   // From TransportChannel
@@ -261,7 +265,8 @@ class BaseChannel
 
   bool PacketIsRtcp(const TransportChannel* channel, const char* data,
                     size_t len);
-  bool SendPacket(bool rtcp, talk_base::Buffer* packet);
+  bool SendPacket(bool rtcp, talk_base::Buffer* packet,
+                  talk_base::DiffServCodePoint dscp);
   virtual bool WantsPacket(bool rtcp, talk_base::Buffer* packet);
   void HandlePacket(bool rtcp, talk_base::Buffer* packet);
 
@@ -277,6 +282,8 @@ class BaseChannel
   void ChannelNotWritable_w();
   bool AddRecvStream_w(const StreamParams& sp);
   bool RemoveRecvStream_w(uint32 ssrc);
+  bool AddSendStream_w(const StreamParams& sp);
+  bool RemoveSendStream_w(uint32 ssrc);
   virtual bool ShouldSetupDtlsSrtp() const;
   // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
   // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
@@ -488,13 +495,13 @@ class VideoChannel : public BaseChannel {
   // TODO(pthatcher): Refactor to use a "capture id" instead of an
   // ssrc here as the "key".
   VideoCapturer* AddScreencast(uint32 ssrc, const ScreencastId& id);
-  VideoCapturer* GetScreencastCapturer(uint32 ssrc);
   bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
   bool RemoveScreencast(uint32 ssrc);
   // True if we've added a screencast.  Doesn't matter if the capturer
   // has been started or not.
   bool IsScreencasting();
-  int ScreencastFps(uint32 ssrc);
+  int GetScreencastFps(uint32 ssrc);
+  int GetScreencastMaxPixels(uint32 ssrc);
   // Get statistics about the current media session.
   bool GetStats(VideoMediaInfo* stats);
 
@@ -525,6 +532,7 @@ class VideoChannel : public BaseChannel {
 
  private:
   typedef std::map<uint32, VideoCapturer*> ScreencastMap;
+  struct ScreencastDetailsMessageData;
 
   // overrides from BaseChannel
   virtual void ChangeState();
@@ -544,12 +552,11 @@ class VideoChannel : public BaseChannel {
   void SetRenderer_w(uint32 ssrc, VideoRenderer* renderer);
 
   VideoCapturer* AddScreencast_w(uint32 ssrc, const ScreencastId& id);
-  VideoCapturer* GetScreencastCapturer_w(uint32 ssrc);
   bool SetCapturer_w(uint32 ssrc, VideoCapturer* capturer);
   bool RemoveScreencast_w(uint32 ssrc);
   void OnScreencastWindowEvent_s(uint32 ssrc, talk_base::WindowEvent we);
   bool IsScreencasting_w() const;
-  int ScreencastFps_w(uint32 ssrc) const;
+  void ScreencastDetails_w(ScreencastDetailsMessageData* d) const;
   void SetScreenCaptureFactory_w(
       ScreenCapturerFactory* screencapture_factory);
   bool GetStats_w(VideoMediaInfo* stats);
@@ -590,11 +597,6 @@ class DataChannel : public BaseChannel {
   ~DataChannel();
   bool Init();
 
-  // downcasts a MediaChannel
-  virtual DataMediaChannel* media_channel() const {
-    return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
-  }
-
   virtual bool SendData(const SendDataParams& params,
                         const talk_base::Buffer& payload,
                         SendDataResult* result);
@@ -616,6 +618,12 @@ class DataChannel : public BaseChannel {
   // both local and remote descriptions are set, and the channel is unblocked.
   sigslot::signal1<bool> SignalReadyToSendData;
 
+ protected:
+  // downcasts a MediaChannel.
+  virtual DataMediaChannel* media_channel() const {
+    return static_cast<DataMediaChannel*>(BaseChannel::media_channel());
+  }
+
  private:
   struct SendDataMessageData : public talk_base::MessageData {
     SendDataMessageData(const SendDataParams& params,
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc
index ff03b49805f..fda89b3c6ed 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channel_unittest.cc
@@ -1794,7 +1794,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
     transport_channel->SignalReadPacket(
         transport_channel, reinterpret_cast<const char*>(kBadPacket),
         sizeof(kBadPacket), 0);
-    EXPECT_EQ_WAIT(T::MediaChannel::ERROR_PLAY_SRTP_AUTH_FAILED, error_, 500);
+    EXPECT_EQ_WAIT(T::MediaChannel::ERROR_PLAY_SRTP_ERROR, error_, 500);
   }
 
   void TestOnReadyToSend() {
@@ -2189,7 +2189,7 @@ TEST_F(VoiceChannelTest, TestVoiceSpecificMuteStream) {
 }
 
 // Test that keyboard automute works correctly and signals upwards.
-TEST_F(VoiceChannelTest, TestKeyboardMute) {
+TEST_F(VoiceChannelTest, DISABLED_TestKeyboardMute) {
   CreateChannels(0, 0);
   EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
   EXPECT_EQ(cricket::VoiceMediaChannel::ERROR_NONE, error_);
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc
index c16b066be11..36c71832da3 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.cc
@@ -670,7 +670,16 @@ VideoCapturer* ChannelManager::CreateVideoCapturer() {
     }
     return NULL;
   }
-  return device_manager_->CreateVideoCapturer(device);
+  VideoCapturer* capturer = device_manager_->CreateVideoCapturer(device);
+  if (capturer && default_video_encoder_config_.max_codec.id != 0) {
+    // For now, use the aspect ratio of the default_video_encoder_config_,
+    // which may be different than the native aspect ratio of the start
+    // format the camera may use.
+    capturer->UpdateAspectRatio(
+        default_video_encoder_config_.max_codec.width,
+        default_video_encoder_config_.max_codec.height);
+  }
+  return capturer;
 }
 
 bool ChannelManager::SetCaptureDevice_w(const Device* cam_device) {
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h
index b1967bfcd50..04af5e19632 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/channelmanager.h
@@ -163,9 +163,6 @@ class ChannelManager : public talk_base::MessageHandler,
   bool monitoring() const { return monitoring_; }
   // Sets the local renderer where to renderer the local camera.
   bool SetLocalRenderer(VideoRenderer* renderer);
-  // Sets the externally provided video capturer. The ssrc is the ssrc of the
-  // (video) stream for which the video capturer should be set.
-  bool SetVideoCapturer(VideoCapturer* capturer);
   bool capturing() const { return capturing_; }
 
   // Configures the logging output of the mediaengine(s).
diff --git a/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc
index 32321ebcd01..6f7c7687154 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/channelmanager_unittest.cc
@@ -149,7 +149,8 @@ TEST_F(ChannelManagerTest, CreateDestroyChannels) {
 }
 
 // Test that we can create and destroy a voice and video channel with a worker.
-TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
+// BUG=https://code.google.com/p/webrtc/issues/detail?id=2355
+TEST_F(ChannelManagerTest, DISABLED_CreateDestroyChannelsOnThread) {
   worker_.Start();
   EXPECT_TRUE(cm_->set_worker_thread(&worker_));
   EXPECT_TRUE(cm_->Init());
@@ -203,6 +204,45 @@ TEST_F(ChannelManagerTest, SetDefaultVideoEncoderConfig) {
   EXPECT_EQ(config, fme_->default_video_encoder_config());
 }
 
+struct GetCapturerFrameSize : public sigslot::has_slots<> {
+  void OnVideoFrame(VideoCapturer* capturer, const VideoFrame* frame) {
+    width = frame->GetWidth();
+    height = frame->GetHeight();
+  }
+  GetCapturerFrameSize(VideoCapturer* capturer) : width(0), height(0) {
+    capturer->SignalVideoFrame.connect(this,
+                                       &GetCapturerFrameSize::OnVideoFrame);
+    static_cast<FakeVideoCapturer*>(capturer)->CaptureFrame();
+  }
+  size_t width;
+  size_t height;
+};
+
+TEST_F(ChannelManagerTest, DefaultCapturerAspectRatio) {
+  VideoCodec codec(100, "VP8", 640, 360, 30, 0);
+  VideoFormat format(640, 360, 33, FOURCC_ANY);
+  VideoEncoderConfig config(codec, 1, 2);
+  EXPECT_TRUE(cm_->Init());
+  // A capturer created before the default encoder config is set will have no
+  // set aspect ratio, so it'll be 4:3 (based on the fake video capture impl).
+  VideoCapturer* capturer = cm_->CreateVideoCapturer();
+  ASSERT_TRUE(capturer != NULL);
+  EXPECT_EQ(CS_RUNNING, capturer->Start(format));
+  GetCapturerFrameSize size(capturer);
+  EXPECT_EQ(640u, size.width);
+  EXPECT_EQ(480u, size.height);
+  delete capturer;
+  // Try again, but with the encoder config set to 16:9.
+  EXPECT_TRUE(cm_->SetDefaultVideoEncoderConfig(config));
+  capturer = cm_->CreateVideoCapturer();
+  ASSERT_TRUE(capturer != NULL);
+  EXPECT_EQ(CS_RUNNING, capturer->Start(format));
+  GetCapturerFrameSize cropped_size(capturer);
+  EXPECT_EQ(640u, cropped_size.width);
+  EXPECT_EQ(360u, cropped_size.height);
+  delete capturer;
+}
+
 // Test that SetDefaultVideoCodec passes through the right values.
 TEST_F(ChannelManagerTest, SetDefaultVideoCodecBeforeInit) {
   cricket::VideoCodec codec(96, "G264", 1280, 720, 60, 0);
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc b/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc
index 1215008b05d..85612308211 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/mediasession.cc
@@ -38,6 +38,7 @@
 #include "talk/base/stringutils.h"
 #include "talk/media/base/constants.h"
 #include "talk/media/base/cryptoparams.h"
+#include "talk/media/sctp/sctpdataengine.h"
 #include "talk/p2p/base/constants.h"
 #include "talk/session/media/channelmanager.h"
 #include "talk/session/media/srtpfilter.h"
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediasession.h b/chromium/third_party/libjingle/source/talk/session/media/mediasession.h
index 327480466a7..5dfc765e7d2 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/mediasession.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/mediasession.h
@@ -83,10 +83,6 @@ extern const char kMediaProtocolDtlsSctp[];
 // Options to control how session descriptions are generated.
 const int kAutoBandwidth = -1;
 const int kBufferedModeDisabled = 0;
-// TODO(pthatcher): This is imposed by usrsctp lib.  I have no idea
-// why it is 9.  Figure out why, and make it bigger, hopefully up to
-// 2^16-1.
-const uint32 kMaxSctpSid = 9;
 
 struct MediaSessionOptions {
   MediaSessionOptions() :
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc b/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc
index 6e04915704a..f2e576ca92c 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/mediasession_unittest.cc
@@ -219,25 +219,19 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
       current_desc.reset(new SessionDescription());
       EXPECT_TRUE(current_desc->AddTransportInfo(
           TransportInfo("audio",
-                        TransportDescription("", std::vector<std::string>(),
+                        TransportDescription("",
                                              current_audio_ufrag,
-                                             current_audio_pwd,
-                                             cricket::ICEMODE_FULL,
-                                             NULL, Candidates()))));
+                                             current_audio_pwd))));
       EXPECT_TRUE(current_desc->AddTransportInfo(
           TransportInfo("video",
-                        TransportDescription("", std::vector<std::string>(),
+                        TransportDescription("",
                                              current_video_ufrag,
-                                             current_video_pwd,
-                                             cricket::ICEMODE_FULL,
-                                             NULL, Candidates()))));
+                                             current_video_pwd))));
       EXPECT_TRUE(current_desc->AddTransportInfo(
           TransportInfo("data",
-                        TransportDescription("", std::vector<std::string>(),
+                        TransportDescription("",
                                              current_data_ufrag,
-                                             current_data_pwd,
-                                             cricket::ICEMODE_FULL,
-                                             NULL, Candidates()))));
+                                             current_data_pwd))));
     }
     if (offer) {
       desc.reset(f1_.CreateOffer(options, current_desc.get()));
diff --git a/chromium/third_party/libjingle/source/talk/session/media/mediasessionclient.cc b/chromium/third_party/libjingle/source/talk/session/media/mediasessionclient.cc
index b54891e8ccb..246592c617f 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/mediasessionclient.cc
+++ b/chromium/third_party/libjingle/source/talk/session/media/mediasessionclient.cc
@@ -35,6 +35,7 @@
 #include "talk/base/stringutils.h"
 #include "talk/media/base/cryptoparams.h"
 #include "talk/media/base/capturemanager.h"
+#include "talk/media/sctp/sctpdataengine.h"
 #include "talk/p2p/base/constants.h"
 #include "talk/p2p/base/parsing.h"
 #include "talk/session/media/mediamessages.h"
diff --git a/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h b/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h
index 9b48dcd957c..b6a269952a4 100644
--- a/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h
+++ b/chromium/third_party/libjingle/source/talk/session/media/srtpfilter.h
@@ -271,7 +271,10 @@ class SrtpStat {
           error(in_error) {
     }
     bool operator <(const FailureKey& key) const {
-      return ssrc < key.ssrc || mode < key.mode || error < key.error;
+      return
+          (ssrc < key.ssrc) ||
+          (ssrc == key.ssrc && mode < key.mode) ||
+          (ssrc == key.ssrc && mode == key.mode && error < key.error);
     }
     uint32 ssrc;
     SrtpFilter::Mode mode;
diff --git a/chromium/third_party/libjingle/source/talk/session/tunnel/pseudotcpchannel.cc b/chromium/third_party/libjingle/source/talk/session/tunnel/pseudotcpchannel.cc
index 8b9a19f0b32..92e9e0ea5d0 100644
--- a/chromium/third_party/libjingle/source/talk/session/tunnel/pseudotcpchannel.cc
+++ b/chromium/third_party/libjingle/source/talk/session/tunnel/pseudotcpchannel.cc
@@ -502,7 +502,7 @@ IPseudoTcpNotify::WriteResult PseudoTcpChannel::TcpWritePacket(
   ASSERT(cs_.CurrentThreadIsOwner());
   ASSERT(tcp == tcp_);
   ASSERT(NULL != channel_);
-  int sent = channel_->SendPacket(buffer, len);
+  int sent = channel_->SendPacket(buffer, len, talk_base::DSCP_NO_CHANGE);
   if (sent > 0) {
     //LOG_F(LS_VERBOSE) << "(" << sent << ") Sent";
     return IPseudoTcpNotify::WR_SUCCESS;
diff --git a/chromium/third_party/libjingle/source/talk/xmpp/constants.cc b/chromium/third_party/libjingle/source/talk/xmpp/constants.cc
index 196a1ec2c45..c56796bc9e7 100644
--- a/chromium/third_party/libjingle/source/talk/xmpp/constants.cc
+++ b/chromium/third_party/libjingle/source/talk/xmpp/constants.cc
@@ -363,6 +363,7 @@ const StaticQName QN_ATTR_STATUS = { STR_EMPTY, "status" };
 
 // Presence connection status
 const char STR_PSTN_CONFERENCE_STATUS_CONNECTING[] = "connecting";
+const char STR_PSTN_CONFERENCE_STATUS_JOINING[] = "joining";
 const char STR_PSTN_CONFERENCE_STATUS_CONNECTED[] = "connected";
 const char STR_PSTN_CONFERENCE_STATUS_HANGUP[] = "hangup";
 
diff --git a/chromium/third_party/libjingle/source/talk/xmpp/constants.h b/chromium/third_party/libjingle/source/talk/xmpp/constants.h
index cd6d2b7c2c4..c53abb5b86f 100644
--- a/chromium/third_party/libjingle/source/talk/xmpp/constants.h
+++ b/chromium/third_party/libjingle/source/talk/xmpp/constants.h
@@ -322,6 +322,7 @@ extern const StaticQName QN_ATTR_STATUS;
 
 // Presence connection status
 extern const char STR_PSTN_CONFERENCE_STATUS_CONNECTING[];
+extern const char STR_PSTN_CONFERENCE_STATUS_JOINING[];
 extern const char STR_PSTN_CONFERENCE_STATUS_CONNECTED[];
 extern const char STR_PSTN_CONFERENCE_STATUS_HANGUP[];
 
diff --git a/chromium/third_party/libjingle/source/talk/xmpp/hangoutpubsubclient.cc b/chromium/third_party/libjingle/source/talk/xmpp/hangoutpubsubclient.cc
index edbf4dddbeb..b6669a10865 100644
--- a/chromium/third_party/libjingle/source/talk/xmpp/hangoutpubsubclient.cc
+++ b/chromium/third_party/libjingle/source/talk/xmpp/hangoutpubsubclient.cc
@@ -238,7 +238,10 @@ class PubSubStateClient : public sigslot::has_slots<> {
     }
 
     PubSubStateChange<C> change;
-    change.publisher_nick = info.publisher_nick;
+    if (!retracted) {
+      // Retracts do not have publisher information.
+      change.publisher_nick = info.publisher_nick;
+    }
     change.published_nick = info.published_nick;
     change.old_state = old_state;
     change.new_state = new_state;
diff --git a/chromium/third_party/libjingle/source/talk/xmpp/rostermodule.h b/chromium/third_party/libjingle/source/talk/xmpp/rostermodule.h
index eafd5954446..7e14dc1314b 100644
--- a/chromium/third_party/libjingle/source/talk/xmpp/rostermodule.h
+++ b/chromium/third_party/libjingle/source/talk/xmpp/rostermodule.h
@@ -81,9 +81,14 @@ enum XmppPresenceAvailable {
 
 enum XmppPresenceConnectionStatus {
   XMPP_CONNECTION_STATUS_UNKNOWN    = 0,
+  // Status set by the server while the user is being rung.
   XMPP_CONNECTION_STATUS_CONNECTING = 1,
-  XMPP_CONNECTION_STATUS_CONNECTED  = 2,
-  XMPP_CONNECTION_STATUS_HANGUP     = 3,
+  // Status set by the client when the user has accepted the ring but before
+  // the client has joined the call.
+  XMPP_CONNECTION_STATUS_JOINING    = 2,
+  // Status set by the client as part of joining the call.
+  XMPP_CONNECTION_STATUS_CONNECTED  = 3,
+  XMPP_CONNECTION_STATUS_HANGUP     = 4,
 };
 
 //! Presence Information
diff --git a/chromium/third_party/libjingle/source/talk/xmpp/rostermoduleimpl.cc b/chromium/third_party/libjingle/source/talk/xmpp/rostermoduleimpl.cc
index 24228803286..31b3abdf0fb 100644
--- a/chromium/third_party/libjingle/source/talk/xmpp/rostermoduleimpl.cc
+++ b/chromium/third_party/libjingle/source/talk/xmpp/rostermoduleimpl.cc
@@ -300,6 +300,8 @@ XmppPresenceImpl::connection_status() const {
       return XMPP_CONNECTION_STATUS_CONNECTING;
     else if (status == STR_PSTN_CONFERENCE_STATUS_CONNECTED)
       return XMPP_CONNECTION_STATUS_CONNECTED;
+    else if (status == STR_PSTN_CONFERENCE_STATUS_JOINING)
+            return XMPP_CONNECTION_STATUS_JOINING;
     else if (status == STR_PSTN_CONFERENCE_STATUS_HANGUP)
         return XMPP_CONNECTION_STATUS_HANGUP;
   }
@@ -349,8 +351,11 @@ XmppPresenceImpl::set_raw_xml(const XmlElement * xml) {
       xml->Name() != QN_PRESENCE)
     return XMPP_RETURN_BADARGUMENT;
 
-  raw_xml_.reset(new XmlElement(*xml));
+  const std::string& type = xml->Attr(QN_TYPE);
+  if (type != STR_EMPTY && type != "unavailable")
+    return XMPP_RETURN_BADARGUMENT;
 
+  raw_xml_.reset(new XmlElement(*xml));
   return XMPP_RETURN_OK;
 }