Report total bitrate for all streams in GetStats.

This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d

22 files changed, 22 insertions, 465 deletions

diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index e117b234fd..ca686c08ee 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -42,8 +42,6 @@ class RtpRtcp : public Module {
     *                         will do nothing.
     *  outgoing_transport   - Transport object that will be called when packets
     *                         are ready to be sent out on the network
-    *  rtcp_feedback        - Callback object that will receive the incoming
-    *                         RTCP messages.
     *  intra_frame_callback - Called when the receiver request a intra frame.
     *  bandwidth_callback   - Called when we receive a changed estimate from
     *                         the receiver of out stream.
@@ -60,7 +58,6 @@ class RtpRtcp : public Module {
     RtpRtcp* default_module;
     ReceiveStatistics* receive_statistics;
     Transport* outgoing_transport;
-    RtcpFeedback* rtcp_feedback;
     RtcpIntraFrameObserver* intra_frame_callback;
     RtcpBandwidthObserver* bandwidth_callback;
     RtcpRttStats* rtt_stats;
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 2d6b272af8..dc51467ee1 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -227,26 +227,6 @@ public:
                                    int packet_length) = 0;
 };
 
-class RtcpFeedback
-{
-public:
-    virtual void OnApplicationDataReceived(const int32_t /*id*/,
-                                           const uint8_t /*subType*/,
-                                           const uint32_t /*name*/,
-                                           const uint16_t /*length*/,
-                                           const uint8_t* /*data*/)  {};
-
-    virtual void OnXRVoIPMetricReceived(
-        const int32_t /*id*/,
-        const RTCPVoIPMetric* /*metric*/)  {};
-
-    virtual void OnReceiveReportReceived(const int32_t id,
-                                         const uint32_t senderSSRC)  {};
-
-protected:
-    virtual ~RtcpFeedback() {}
-};
-
 class RtpFeedback
 {
 public:
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 7e187c75a5..584b9ad6b2 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -136,10 +136,9 @@ class MockRtpRtcp : public RtpRtcp {
       int(int bytes));
   MOCK_CONST_METHOD2(GetSendSideDelay,
       bool(int* avg_send_delay_ms, int* max_send_delay_ms));
-  MOCK_METHOD3(RegisterRtcpObservers,
+  MOCK_METHOD2(RegisterRtcpObservers,
       void(RtcpIntraFrameObserver* intraFrameCallback,
-           RtcpBandwidthObserver* bandwidthCallback,
-           RtcpFeedback* callback));
+           RtcpBandwidthObserver* bandwidthCallback));
   MOCK_CONST_METHOD0(RTCP,
       RTCPMethod());
   MOCK_METHOD1(SetRTCPStatus,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index 1d1e29167f..8d044801fc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -38,7 +38,6 @@ RTCPReceiver::RTCPReceiver(const int32_t id, Clock* clock,
     _rtpRtcp(*owner),
       _criticalSectionFeedbacks(
           CriticalSectionWrapper::CreateCriticalSection()),
-    _cbRtcpFeedback(NULL),
     _cbRtcpBandwidthObserver(NULL),
     _cbRtcpIntraFrameObserver(NULL),
     _criticalSectionRTCPReceiver(
@@ -145,12 +144,10 @@ uint32_t RTCPReceiver::RemoteSSRC() const {
 
 void RTCPReceiver::RegisterRtcpObservers(
     RtcpIntraFrameObserver* intra_frame_callback,
-    RtcpBandwidthObserver* bandwidth_callback,
-    RtcpFeedback* feedback_callback) {
+    RtcpBandwidthObserver* bandwidth_callback) {
   CriticalSectionScoped lock(_criticalSectionFeedbacks);
   _cbRtcpIntraFrameObserver = intra_frame_callback;
   _cbRtcpBandwidthObserver = bandwidth_callback;
-  _cbRtcpFeedback = feedback_callback;
 }
 
 void RTCPReceiver::SetSsrcs(uint32_t main_ssrc,
@@ -1442,23 +1439,6 @@ void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
             now);
       }
     }
-    if(_cbRtcpFeedback) {
-      if(!(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)) {
-        _cbRtcpFeedback->OnReceiveReportReceived(_id,
-            rtcpPacketInformation.remoteSSRC);
-      }
-      if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpXrVoipMetric) {
-        _cbRtcpFeedback->OnXRVoIPMetricReceived(_id,
-            rtcpPacketInformation.VoIPMetric);
-      }
-      if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpApp) {
-        _cbRtcpFeedback->OnApplicationDataReceived(_id,
-            rtcpPacketInformation.applicationSubType,
-            rtcpPacketInformation.applicationName,
-            rtcpPacketInformation.applicationLength,
-            rtcpPacketInformation.applicationData);
-      }
-    }
   }
 
   {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 087722c42f..ecc5ebed73 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -49,8 +49,7 @@ public:
     uint32_t RelaySSRC() const;
 
     void RegisterRtcpObservers(RtcpIntraFrameObserver* intra_frame_callback,
-                               RtcpBandwidthObserver* bandwidth_callback,
-                               RtcpFeedback* feedback_callback);
+                               RtcpBandwidthObserver* bandwidth_callback);
 
     int32_t IncomingRTCPPacket(
         RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
@@ -228,7 +227,6 @@ protected:
   ModuleRtpRtcpImpl&      _rtpRtcp;
 
   CriticalSectionWrapper* _criticalSectionFeedbacks;
-  RtcpFeedback*           _cbRtcpFeedback;
   RtcpBandwidthObserver*  _cbRtcpBandwidthObserver;
   RtcpIntraFrameObserver* _cbRtcpIntraFrameObserver;
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index f8713e2755..6446cb7c49 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -31,7 +31,6 @@ RtpRtcp::Configuration::Configuration()
       default_module(NULL),
       receive_statistics(NullObjectReceiveStatistics()),
       outgoing_transport(NULL),
-      rtcp_feedback(NULL),
       intra_frame_callback(NULL),
       bandwidth_callback(NULL),
       rtt_stats(NULL),
@@ -102,8 +101,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
   }
   // TODO(pwestin) move to constructors of each rtp/rtcp sender/receiver object.
   rtcp_receiver_.RegisterRtcpObservers(configuration.intra_frame_callback,
-                                       configuration.bandwidth_callback,
-                                       configuration.rtcp_feedback);
+                                       configuration.bandwidth_callback);
   rtcp_sender_.RegisterSendTransport(configuration.outgoing_transport);
 
   // Make sure that RTCP objects are aware of our SSRC.
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index 9d0505ef71..8cbdeb5b91 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -24,7 +24,7 @@ using namespace webrtc;
 
 const uint64_t kTestPictureId = 12345678;
 
-class RtcpCallback : public RtcpFeedback, public RtcpIntraFrameObserver {
+class RtcpCallback : public RtcpIntraFrameObserver {
  public:
   void SetModule(RtpRtcp* module) {
     _rtpRtcpModule = module;
@@ -34,27 +34,6 @@ class RtcpCallback : public RtcpFeedback, public RtcpIntraFrameObserver {
   virtual void OnLipSyncUpdate(const int32_t id,
                                const int32_t audioVideoOffset) {
   };
-  virtual void OnXRVoIPMetricReceived(
-      const int32_t id,
-      const RTCPVoIPMetric* metric) {
-  };
-  virtual void OnApplicationDataReceived(const int32_t id,
-                                         const uint8_t subType,
-                                         const uint32_t name,
-                                         const uint16_t length,
-                                         const uint8_t* data) {
-    char print_name[5];
-    print_name[0] = static_cast<char>(name >> 24);
-    print_name[1] = static_cast<char>(name >> 16);
-    print_name[2] = static_cast<char>(name >> 8);
-    print_name[3] = static_cast<char>(name);
-    print_name[4] = 0;
-
-    EXPECT_STRCASEEQ("test", print_name);
-  };
-  virtual void OnReceiveReportReceived(const int32_t id,
-                                       const uint32_t senderSSRC) {
-  };
   virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
   };
   virtual void OnReceivedSLI(uint32_t ssrc,
@@ -112,7 +91,6 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     configuration.clock = &fake_clock;
     configuration.receive_statistics = receive_statistics1_.get();
     configuration.outgoing_transport = transport1;
-    configuration.rtcp_feedback = myRTCPFeedback1;
     configuration.intra_frame_callback = myRTCPFeedback1;
 
     rtp_payload_registry1_.reset(new RTPPayloadRegistry(
@@ -131,7 +109,6 @@ class RtpRtcpRtcpTest : public ::testing::Test {
     configuration.receive_statistics = receive_statistics2_.get();
     configuration.id = test_id + 1;
     configuration.outgoing_transport = transport2;
-    configuration.rtcp_feedback = myRTCPFeedback2;
     configuration.intra_frame_callback = myRTCPFeedback2;
 
     module2 = RtpRtcp::CreateRtpRtcp(configuration);
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index b7ad380b2f..75cc26d860 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1578,6 +1578,8 @@ TEST_F(EndToEndTest, GetStats) {
       expected_cname_ = send_config->rtp.c_name;
     }
 
+    virtual size_t GetNumStreams() const OVERRIDE { return kNumSsrcs; }
+
     virtual void OnStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) OVERRIDE {
diff --git a/webrtc/video_engine/include/vie_rtp_rtcp.h b/webrtc/video_engine/include/vie_rtp_rtcp.h
index 8771ea8e77..fdf96bbc4c 100644
--- a/webrtc/video_engine/include/vie_rtp_rtcp.h
+++ b/webrtc/video_engine/include/vie_rtp_rtcp.h
@@ -69,25 +69,6 @@ class WEBRTC_DLLEXPORT ViERTPObserver {
   virtual ~ViERTPObserver() {}
 };
 
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterRTCPObserver() and
-// deregistered using DeregisterRTCPObserver().
-
-class WEBRTC_DLLEXPORT ViERTCPObserver {
- public:
-  // This method is called if a application-defined RTCP packet has been
-  // received.
-  virtual void OnApplicationDataReceived(
-      const int video_channel,
-      const unsigned char sub_type,
-      const unsigned int name,
-      const char* data,
-      const unsigned short data_length_in_bytes) = 0;
- protected:
-  virtual ~ViERTCPObserver() {}
-};
-
 class WEBRTC_DLLEXPORT ViERTP_RTCP {
  public:
   enum { KDefaultDeltaTransmitTimeSeconds = 15 };
@@ -468,13 +449,6 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
   // Removes a registered instance of ViERTPObserver.
   virtual int DeregisterRTPObserver(const int video_channel) = 0;
 
-  // Registers an instance of a user implementation of the ViERTCPObserver.
-  virtual int RegisterRTCPObserver(const int video_channel,
-                                   ViERTCPObserver& observer) = 0;
-
-  // Removes a registered instance of ViERTCPObserver.
-  virtual int DeregisterRTCPObserver(const int video_channel) = 0;
-
   // Registers and instance of a user implementation of ViEFrameCountObserver
   virtual int RegisterSendFrameCountObserver(
       int video_channel, FrameCountObserver* observer) = 0;
diff --git a/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc b/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
index e43fc751b0..4508d8e791 100644
--- a/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
+++ b/webrtc/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
@@ -52,9 +52,4 @@ TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
   tests_->ViERenderExtendedTest();
 }
 
-TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
-       DISABLED_RunsRtpRtcpTestWithoutErrors) {
-  tests_->ViERtpRtcpExtendedTest();
-}
-
 }  // namespace
diff --git a/webrtc/video_engine/test/auto_test/interface/vie_autotest.h b/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
index 7a83637860..c584988adf 100644
--- a/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
+++ b/webrtc/video_engine/test/auto_test/interface/vie_autotest.h
@@ -100,7 +100,6 @@ public:
 
     // vie_autotest_rtp_rtcp.cc
     void ViERtpRtcpStandardTest();
-    void ViERtpRtcpExtendedTest();
     void ViERtpRtcpAPITest();
 
 private:
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
index 41fafaef08..aa9dbd54ce 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest.cc
@@ -76,7 +76,6 @@ void ViEAutoTest::ViEExtendedTest()
     ViECodecExtendedTest();
     ViEImageProcessExtendedTest();
     ViERenderExtendedTest();
-    ViERtpRtcpExtendedTest();
 }
 
 void ViEAutoTest::ViEAPITest()
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
index ced235f6b5..ab68634927 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_android.cc
@@ -150,7 +150,8 @@ int ViEAutoTestAndroid::RunAutotest(int testSelection, int subTestSelection,
           break;
 
         case 8: // RTP/RTCP
-          vieAutoTest.ViERtpRtcpExtendedTest();
+          // Note that this test is removed. It hasn't been properly cleaned up
+          // because this hopefully going away soon.
           break;
 
         default:
diff --git a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
index b7d02156c2..923fe41b1b 100644
--- a/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ b/webrtc/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
@@ -40,52 +40,6 @@ public:
     }
 };
 
-class ViERtcpObserver: public webrtc::ViERTCPObserver
-{
-public:
-    int _channel;
-    unsigned char _subType;
-    unsigned int _name;
-    char* _data;
-    unsigned short _dataLength;
-
-    ViERtcpObserver() :
-        _channel(-1),
-        _subType(0),
-        _name(0),
-        _data(NULL),
-        _dataLength(0)
-    {
-    }
-    ~ViERtcpObserver()
-    {
-        if (_data)
-        {
-            delete[] _data;
-        }
-    }
-    virtual void OnApplicationDataReceived(
-        const int videoChannel, const unsigned char subType,
-        const unsigned int name, const char* data,
-        const unsigned short dataLengthInBytes)
-    {
-        _channel = videoChannel;
-        _subType = subType;
-        _name = name;
-        if (dataLengthInBytes > _dataLength)
-        {
-            delete[] _data;
-            _data = NULL;
-        }
-        if (_data == NULL)
-        {
-            _data = new char[dataLengthInBytes];
-        }
-        memcpy(_data, data, dataLengthInBytes);
-        _dataLength = dataLengthInBytes;
-    }
-};
-
 void ViEAutoTest::ViERtpRtcpStandardTest()
 {
     // ***************************************************************
@@ -630,70 +584,6 @@ void ViEAutoTest::ViERtpRtcpStandardTest()
     //***************************************************************
 }
 
-void ViEAutoTest::ViERtpRtcpExtendedTest()
-{
-    //***************************************************************
-    //  Begin create/initialize WebRTC Video Engine for testing
-    //***************************************************************
-    // Create VIE
-    TbInterfaces ViE("ViERtpRtcpExtendedTest");
-    // Create a video channel
-    TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-    // Create a capture device
-    TbCaptureDevice tbCapture(ViE);
-    tbCapture.ConnectTo(tbChannel.videoChannel);
-
-    //tbChannel.StartReceive(rtpPort);
-    //tbChannel.StartSend(rtpPort);
-    TbExternalTransport myTransport(*(ViE.network), tbChannel.videoChannel,
-                                    NULL);
-
-    EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.network->RegisterSendTransport(
-        tbChannel.videoChannel, myTransport));
-    EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
-    //***************************************************************
-    //  Engine ready. Begin testing class
-    //***************************************************************
-
-    //
-    // Application specific RTCP
-    //
-    //
-
-    ViERtcpObserver rtcpObserver;
-    EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
-        tbChannel.videoChannel, rtcpObserver));
-
-    unsigned char subType = 3;
-    unsigned int name = static_cast<unsigned int> (0x41424344); // 'ABCD';
-    const char* data = "ViEAutoTest Data of length 32 -\0";
-    const unsigned short numBytes = 32;
-
-    EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
-        tbChannel.videoChannel, subType, name, data, numBytes));
-
-    ViETest::Log("Sending RTCP application data...\n");
-    AutoTestSleep(kAutoTestSleepTimeMs);
-
-    EXPECT_EQ(subType, rtcpObserver._subType);
-    EXPECT_STRCASEEQ(data, rtcpObserver._data);
-    EXPECT_EQ(name, rtcpObserver._name);
-    EXPECT_EQ(numBytes, rtcpObserver._dataLength);
-
-    ViETest::Log("\t RTCP application data received\n");
-
-    //***************************************************************
-    //  Testing finished. Tear down Video Engine
-    //***************************************************************
-    EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-    EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
-    EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
-}
-
 void ViEAutoTest::ViERtpRtcpAPITest()
 {
     //***************************************************************
@@ -853,16 +743,6 @@ void ViEAutoTest::ViERtpRtcpAPITest()
             tbChannel.videoChannel));
         EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTPObserver(
             tbChannel.videoChannel));
-
-        ViERtcpObserver rtcpObserver;
-        EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
-            tbChannel.videoChannel, rtcpObserver));
-        EXPECT_NE(0, ViE.rtp_rtcp->RegisterRTCPObserver(
-            tbChannel.videoChannel, rtcpObserver));
-        EXPECT_EQ(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
-            tbChannel.videoChannel));
-        EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
-            tbChannel.videoChannel));
     }
     //
     // PLI
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index ded21407c9..270da2a252 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -130,7 +130,6 @@ ViEChannel::ViEChannel(int32_t channel_id,
       codec_observer_(NULL),
       do_key_frame_callbackRequest_(false),
       rtp_observer_(NULL),
-      rtcp_observer_(NULL),
       intra_frame_observer_(intra_frame_observer),
       rtt_stats_(rtt_stats),
       paced_sender_(paced_sender),
@@ -150,22 +149,9 @@ ViEChannel::ViEChannel(int32_t channel_id,
       max_nack_reordering_threshold_(kMaxPacketAgeToNack),
       pre_render_callback_(NULL),
       start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()) {
-  RtpRtcp::Configuration configuration;
-  configuration.id = ViEModuleId(engine_id, channel_id);
-  configuration.audio = false;
-  configuration.default_module = default_rtp_rtcp;
-  configuration.outgoing_transport = &vie_sender_;
-  configuration.rtcp_feedback = this;
-  configuration.intra_frame_callback = intra_frame_observer;
-  configuration.bandwidth_callback = bandwidth_observer;
-  configuration.rtt_stats = rtt_stats;
+  RtpRtcp::Configuration configuration = CreateRtpRtcpConfiguration();
   configuration.remote_bitrate_estimator = remote_bitrate_estimator;
-  configuration.paced_sender = paced_sender;
   configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
-  configuration.send_bitrate_observer = &send_bitrate_observer_;
-  configuration.send_frame_count_observer = &send_frame_count_observer_;
-  configuration.send_side_delay_observer = &send_side_delay_observer_;
-
   rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
   vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
   vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
@@ -1013,20 +999,6 @@ int32_t ViEChannel::RegisterRtpObserver(ViERTPObserver* observer) {
   return 0;
 }
 
-int32_t ViEChannel::RegisterRtcpObserver(ViERTCPObserver* observer) {
-  CriticalSectionScoped cs(callback_cs_.get());
-  if (observer) {
-    if (rtcp_observer_) {
-      LOG_F(LS_ERROR) << "Observer already registered.";
-      return -1;
-    }
-    rtcp_observer_ = observer;
-  } else {
-    rtcp_observer_ = NULL;
-  }
-  return 0;
-}
-
 int32_t ViEChannel::SendApplicationDefinedRTCPPacket(
     const uint8_t sub_type,
     uint32_t name,
@@ -1640,19 +1612,25 @@ RtpRtcp* ViEChannel::GetRtpRtcpModule(size_t index) const {
   return *it;
 }
 
-RtpRtcp* ViEChannel::CreateRtpRtcpModule() {
+RtpRtcp::Configuration ViEChannel::CreateRtpRtcpConfiguration() {
   RtpRtcp::Configuration configuration;
   configuration.id = ViEModuleId(engine_id_, channel_id_);
-  configuration.audio = false;  // Video.
+  configuration.audio = false;
   configuration.default_module = default_rtp_rtcp_;
   configuration.outgoing_transport = &vie_sender_;
   configuration.intra_frame_callback = intra_frame_observer_;
   configuration.bandwidth_callback = bandwidth_observer_.get();
   configuration.rtt_stats = rtt_stats_;
   configuration.paced_sender = paced_sender_;
+  configuration.send_bitrate_observer = &send_bitrate_observer_;
+  configuration.send_frame_count_observer = &send_frame_count_observer_;
   configuration.send_side_delay_observer = &send_side_delay_observer_;
 
-  return RtpRtcp::CreateRtpRtcp(configuration);
+  return configuration;
+}
+
+RtpRtcp* ViEChannel::CreateRtpRtcpModule() {
+  return RtpRtcp::CreateRtpRtcp(CreateRtpRtcpConfiguration());
 }
 
 int32_t ViEChannel::StartDecodeThread() {
@@ -1732,24 +1710,6 @@ void ViEChannel::RegisterPreDecodeImageCallback(
   vcm_->RegisterPreDecodeImageCallback(pre_decode_callback);
 }
 
-void ViEChannel::OnApplicationDataReceived(const int32_t id,
-                                           const uint8_t sub_type,
-                                           const uint32_t name,
-                                           const uint16_t length,
-                                           const uint8_t* data) {
-  if (channel_id_ != ChannelId(id)) {
-    return;
-  }
-  CriticalSectionScoped cs(callback_cs_.get());
-  {
-    if (rtcp_observer_) {
-      rtcp_observer_->OnApplicationDataReceived(
-          channel_id_, sub_type, name, reinterpret_cast<const char*>(data),
-          length);
-    }
-  }
-}
-
 int32_t ViEChannel::OnInitializeDecoder(
     const int32_t id,
     const int8_t payload_type,
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index 3b8d96a283..97b158784b 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -14,6 +14,7 @@
 #include <list>
 
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
@@ -39,11 +40,9 @@ class I420FrameCallback;
 class PacedSender;
 class ProcessThread;
 class RtcpRttStats;
-class RtpRtcp;
 class ThreadWrapper;
 class ViEDecoderObserver;
 class ViEEffectFilter;
-class ViERTCPObserver;
 class ViERTPObserver;
 class VideoCodingModule;
 class VideoDecoder;
@@ -56,7 +55,6 @@ class ViEChannel
       public VCMReceiveStatisticsCallback,
       public VCMDecoderTimingCallback,
       public VCMPacketRequestCallback,
-      public RtcpFeedback,
       public RtpFeedback,
       public ViEFrameProviderBase {
  public:
@@ -162,7 +160,6 @@ class ViEChannel
   // Gets the CName of the incoming stream.
   int32_t GetRemoteRTCPCName(char rtcp_cname[]);
   int32_t RegisterRtpObserver(ViERTPObserver* observer);
-  int32_t RegisterRtcpObserver(ViERTCPObserver* observer);
   int32_t SendApplicationDefinedRTCPPacket(
       const uint8_t sub_type,
       uint32_t name,
@@ -226,13 +223,6 @@ class ViEChannel
                        RTPDirections direction);
   int32_t StopRTPDump(RTPDirections direction);
 
-  // Implements RtcpFeedback.
-  // TODO(pwestin) Depricate this functionality.
-  virtual void OnApplicationDataReceived(const int32_t id,
-                                         const uint8_t sub_type,
-                                         const uint32_t name,
-                                         const uint16_t length,
-                                         const uint8_t* data);
   // Implements RtpFeedback.
   virtual int32_t OnInitializeDecoder(
       const int32_t id,
@@ -370,6 +360,7 @@ class ViEChannel
       EXCLUSIVE_LOCKS_REQUIRED(rtp_rtcp_cs_);
   RtpRtcp* GetRtpRtcpModule(size_t simulcast_idx) const
       EXCLUSIVE_LOCKS_REQUIRED(rtp_rtcp_cs_);
+  RtpRtcp::Configuration CreateRtpRtcpConfiguration();
   RtpRtcp* CreateRtpRtcpModule();
   // Assumed to be protected.
   int32_t StartDecodeThread();
@@ -475,7 +466,6 @@ class ViEChannel
   ViEDecoderObserver* codec_observer_;
   bool do_key_frame_callbackRequest_;
   ViERTPObserver* rtp_observer_;
-  ViERTCPObserver* rtcp_observer_;
   RtcpIntraFrameObserver* intra_frame_observer_;
   RtcpRttStats* rtt_stats_;
   PacedSender* paced_sender_;
diff --git a/webrtc/video_engine/vie_rtp_rtcp_impl.cc b/webrtc/video_engine/vie_rtp_rtcp_impl.cc
index 36e60040de..853d778168 100644
--- a/webrtc/video_engine/vie_rtp_rtcp_impl.cc
+++ b/webrtc/video_engine/vie_rtp_rtcp_impl.cc
@@ -906,37 +906,6 @@ int ViERTP_RTCPImpl::DeregisterRTPObserver(const int video_channel) {
   return 0;
 }
 
-int ViERTP_RTCPImpl::RegisterRTCPObserver(const int video_channel,
-                                          ViERTCPObserver& observer) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterRtcpObserver(&observer) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpObserverAlreadyRegistered);
-    return -1;
-  }
-  return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterRTCPObserver(const int video_channel) {
-  LOG_F(LS_INFO) << "channel " << video_channel;
-  ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
-  ViEChannel* vie_channel = cs.Channel(video_channel);
-  if (!vie_channel) {
-    shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
-    return -1;
-  }
-  if (vie_channel->RegisterRtcpObserver(NULL) != 0) {
-    shared_data_->SetLastError(kViERtpRtcpObserverNotRegistered);
-    return -1;
-  }
-  return 0;
-}
-
 int ViERTP_RTCPImpl::RegisterSendChannelRtcpStatisticsCallback(
   int video_channel, RtcpStatisticsCallback* callback) {
   LOG_F(LS_INFO) << "channel " << video_channel;
diff --git a/webrtc/video_engine/vie_rtp_rtcp_impl.h b/webrtc/video_engine/vie_rtp_rtcp_impl.h
index 5566069b8d..99b97583ef 100644
--- a/webrtc/video_engine/vie_rtp_rtcp_impl.h
+++ b/webrtc/video_engine/vie_rtp_rtcp_impl.h
@@ -133,9 +133,6 @@ class ViERTP_RTCPImpl
   virtual int RegisterRTPObserver(const int video_channel,
                                   ViERTPObserver& observer);
   virtual int DeregisterRTPObserver(const int video_channel);
-  virtual int RegisterRTCPObserver(const int video_channel,
-                                   ViERTCPObserver& observer);
-  virtual int DeregisterRTCPObserver(const int video_channel);
 
   virtual int RegisterSendChannelRtcpStatisticsCallback(
       int channel, RtcpStatisticsCallback* callback);
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 58e0055de6..d651d8f76d 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -323,36 +323,6 @@ void Channel::ResetStatistics(uint32_t ssrc) {
   statistics_proxy_->ResetStatistics();
 }
 
-void
-Channel::OnApplicationDataReceived(int32_t id,
-                                   uint8_t subType,
-                                   uint32_t name,
-                                   uint16_t length,
-                                   const uint8_t* data)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::OnApplicationDataReceived(id=%d, subType=%u,"
-                 " name=%u, length=%u)",
-                 id, subType, name, length);
-
-    int32_t channel = VoEChannelId(id);
-    assert(channel == _channelId);
-
-    if (_rtcpObserver)
-    {
-        CriticalSectionScoped cs(&_callbackCritSect);
-
-        if (_rtcpObserverPtr)
-        {
-            _rtcpObserverPtr->OnApplicationDataReceived(channel,
-                                                        subType,
-                                                        name,
-                                                        data,
-                                                        length);
-        }
-    }
-}
-
 int32_t
 Channel::OnInitializeDecoder(
     int32_t id,
@@ -790,10 +760,8 @@ Channel::Channel(int32_t channelId,
     _rxVadObserverPtr(NULL),
     _oldVadDecision(-1),
     _sendFrameType(0),
-    _rtcpObserverPtr(NULL),
     _externalMixing(false),
     _mixFileWithMicrophone(false),
-    _rtcpObserver(false),
     _mute(false),
     _panLeft(1.0f),
     _panRight(1.0f),
@@ -832,7 +800,6 @@ Channel::Channel(int32_t channelId,
     configuration.id = VoEModuleId(instanceId, channelId);
     configuration.audio = true;
     configuration.outgoing_transport = this;
-    configuration.rtcp_feedback = this;
     configuration.audio_messages = this;
     configuration.receive_statistics = rtp_receive_statistics_.get();
     configuration.bandwidth_callback = rtcp_bandwidth_observer_.get();
@@ -2872,48 +2839,6 @@ Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
 #endif // #ifdef WEBRTC_VOICE_ENGINE_NR
 
 int
-Channel::RegisterRTCPObserver(VoERTCPObserver& observer)
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
-                 "Channel::RegisterRTCPObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
-
-    if (_rtcpObserverPtr)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceError,
-            "RegisterRTCPObserver() observer already enabled");
-        return -1;
-    }
-
-    _rtcpObserverPtr = &observer;
-    _rtcpObserver = true;
-
-    return 0;
-}
-
-int
-Channel::DeRegisterRTCPObserver()
-{
-    WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-                 "Channel::DeRegisterRTCPObserver()");
-    CriticalSectionScoped cs(&_callbackCritSect);
-
-    if (!_rtcpObserverPtr)
-    {
-        _engineStatisticsPtr->SetLastError(
-            VE_INVALID_OPERATION, kTraceWarning,
-            "DeRegisterRTCPObserver() observer already disabled");
-        return 0;
-    }
-
-    _rtcpObserver = false;
-    _rtcpObserverPtr = NULL;
-
-    return 0;
-}
-
-int
 Channel::SetLocalSSRC(unsigned int ssrc)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 1c2fc4ac8d..94a34328c0 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -59,7 +59,6 @@ class RtpRtcp;
 class TelephoneEventHandler;
 class ViENetwork;
 class VoEMediaProcess;
-class VoERTCPObserver;
 class VoERTPObserver;
 class VoiceEngineObserver;
 
@@ -156,7 +155,6 @@ private:
 class Channel:
     public RtpData,
     public RtpFeedback,
-    public RtcpFeedback,
     public FileCallback, // receiving notification from file player & recorder
     public Transport,
     public RtpAudioFeedback,
@@ -314,8 +312,6 @@ public:
 #endif
 
     // VoERTP_RTCP
-    int RegisterRTCPObserver(VoERTCPObserver& observer);
-    int DeRegisterRTCPObserver();
     int SetLocalSSRC(unsigned int ssrc);
     int GetLocalSSRC(unsigned int& ssrc);
     int GetRemoteSSRC(unsigned int& ssrc);
@@ -386,13 +382,6 @@ public:
                                        uint32_t CSRC, bool added) OVERRIDE;
     virtual void ResetStatistics(uint32_t ssrc) OVERRIDE;
 
-    // From RtcpFeedback in the RTP/RTCP module
-    virtual void OnApplicationDataReceived(int32_t id,
-                                           uint8_t subType,
-                                           uint32_t name,
-                                           uint16_t length,
-                                           const uint8_t* data) OVERRIDE;
-
     // From RtpAudioFeedback in the RTP/RTCP module
     virtual void OnPlayTelephoneEvent(int32_t id,
                                       uint8_t event,
@@ -565,11 +554,9 @@ private:
     VoERxVadCallback* _rxVadObserverPtr;
     int32_t _oldVadDecision;
     int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
-    VoERTCPObserver* _rtcpObserverPtr;
     // VoEBase
     bool _externalMixing;
     bool _mixFileWithMicrophone;
-    bool _rtcpObserver;
     // VoEVolumeControl
     bool _mute;
     float _panLeft;
diff --git a/webrtc/voice_engine/include/voe_rtp_rtcp.h b/webrtc/voice_engine/include/voe_rtp_rtcp.h
index 6223bd6e0b..c5ab861509 100644
--- a/webrtc/voice_engine/include/voe_rtp_rtcp.h
+++ b/webrtc/voice_engine/include/voe_rtp_rtcp.h
@@ -61,19 +61,6 @@ protected:
     virtual ~VoERTPObserver() {}
 };
 
-// VoERTCPObserver
-class WEBRTC_DLLEXPORT VoERTCPObserver
-{
-public:
-    virtual void OnApplicationDataReceived(
-        int channel, unsigned char subType,
-        unsigned int name, const unsigned char* data,
-        unsigned short dataLengthInBytes) = 0;
-
-protected:
-    virtual ~VoERTCPObserver() {}
-};
-
 // CallStatistics
 struct CallStatistics
 {
@@ -268,9 +255,6 @@ public:
     virtual int RegisterRTPObserver(int channel,
             VoERTPObserver& observer) { return -1; };
     virtual int DeRegisterRTPObserver(int channel) { return -1; };
-    virtual int RegisterRTCPObserver(
-            int channel, VoERTCPObserver& observer) { return -1; };
-    virtual int DeRegisterRTCPObserver(int channel) { return -1; };
     virtual int GetRemoteCSRCs(int channel,
             unsigned int arrCSRC[15]) { return -1; };
     virtual int InsertExtraRTPPacket(
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index 9a732c0e2b..5ba84216e5 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -55,23 +55,6 @@ void TestRtpObserver::OnIncomingSSRCChanged(int channel,
   }
 }
 
-class RtcpAppHandler : public webrtc::VoERTCPObserver {
- public:
-  RtcpAppHandler() : length_in_bytes_(0), sub_type_(0), name_(0) {}
-  void OnApplicationDataReceived(int channel,
-                                 unsigned char sub_type,
-                                 unsigned int name,
-                                 const unsigned char* data,
-                                 unsigned short length_in_bytes);
-  void Reset();
-  ~RtcpAppHandler() {}
-  unsigned short length_in_bytes_;
-  unsigned char data_[256];
-  unsigned char sub_type_;
-  unsigned int name_;
-};
-
-
 static const char* const RTCP_CNAME = "Whatever";
 
 class RtpRtcpTest : public AfterStreamingFixture {
@@ -104,23 +87,6 @@ class RtpRtcpTest : public AfterStreamingFixture {
   LoopBackTransport* transport_;
 };
 
-void RtcpAppHandler::OnApplicationDataReceived(
-    const int /*channel*/, unsigned char sub_type,
-    unsigned int name, const unsigned char* data,
-    unsigned short length_in_bytes) {
-  length_in_bytes_ = length_in_bytes;
-  memcpy(data_, &data[0], length_in_bytes);
-  sub_type_ = sub_type;
-  name_ = name;
-}
-
-void RtcpAppHandler::Reset() {
-  length_in_bytes_ = 0;
-  memset(data_, 0, sizeof(data_));
-  sub_type_ = 0;
-  name_ = 0;
-}
-
 TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
   if (!FLAGS_include_timing_dependent_tests) {
     TEST_LOG("Skipping test - running in slow execution environment...\n");