diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index 49247ce424..f50933701b 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -610,13 +610,6 @@ class RtpRtcp : public Module {
     *
     ***************************************************************************/
 
-    /*
-    *   Set the estimated camera delay in MS
-    *
-    *   return -1 on failure else 0
-    */
-     virtual int32_t SetCameraDelay(int32_t delayMS) = 0;
-
     /*
     *   Set the target send bitrate
     */
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index 57de5fed6e..6c8dff41f6 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -231,8 +231,6 @@ class MockRtpRtcp : public RtpRtcp {
       int32_t(bool& enable, uint8_t& ID));
   MOCK_METHOD1(SetAudioLevel,
       int32_t(const uint8_t level_dBov));
-  MOCK_METHOD1(SetCameraDelay,
-      int32_t(const int32_t delayMS));
   MOCK_METHOD1(SetTargetSendBitrate,
       void(const std::vector<uint32_t>& stream_bitrates));
   MOCK_METHOD3(SetGenericFECStatus,
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
index 7bf38a6f9e..fbfe832991 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
@@ -108,9 +108,6 @@ RTCPSender::RTCPSender(int32_t id,
       internal_report_blocks_(),
       external_report_blocks_(),
       _csrcCNAMEs(),
-
-      _cameraDelayMS(0),
-
       _lastSendReport(),
       _lastRTCPTime(),
 
@@ -302,18 +299,6 @@ void RTCPSender::SetRemoteSSRC(uint32_t ssrc)
     _remoteSSRC = ssrc;
 }
 
-int32_t RTCPSender::SetCameraDelay(int32_t delayMS) {
-    CriticalSectionScoped lock(_criticalSectionRTCPSender);
-    if(delayMS > 1000 || delayMS < -1000)
-    {
-        LOG(LS_WARNING) << "Delay can't be larger than 1 second: "
-                        << delayMS << " ms";
-        return -1;
-    }
-    _cameraDelayMS = delayMS;
-    return 0;
-}
-
 int32_t RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
   if (!cName)
     return -1;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 874adbfa77..fcc79b64b4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -93,8 +93,6 @@ public:
 
     void SetRemoteSSRC(uint32_t ssrc);
 
-    int32_t SetCameraDelay(int32_t delayMS);
-
     int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]);
 
     int32_t AddMixedCNAME(uint32_t SSRC, const char cName[RTCP_CNAME_SIZE]);
@@ -303,8 +301,6 @@ private:
     std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs
         GUARDED_BY(_criticalSectionRTCPSender);
 
-    int32_t _cameraDelayMS GUARDED_BY(_criticalSectionRTCPSender);
-
     // Sent
     uint32_t _lastSendReport[RTCP_NUMBER_OF_SR] GUARDED_BY(
         _criticalSectionRTCPSender);  // allow packet loss and RTT above 1 sec
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 89ac9a0204..98c9bf9713 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -959,22 +959,6 @@ int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(
       GetFeedbackState(), kRtcpSli, 0, 0, false, picture_id);
 }
 
-int32_t ModuleRtpRtcpImpl::SetCameraDelay(const int32_t delay_ms) {
-  if (IsDefaultModule()) {
-    CriticalSectionScoped lock(critical_section_module_ptrs_.get());
-    std::vector<ModuleRtpRtcpImpl*>::iterator it = child_modules_.begin();
-    while (it != child_modules_.end()) {
-      RtpRtcp* module = *it;
-      if (module) {
-        module->SetCameraDelay(delay_ms);
-      }
-      it++;
-    }
-    return 0;
-  }
-  return rtcp_sender_.SetCameraDelay(delay_ms);
-}
-
 int32_t ModuleRtpRtcpImpl::SetGenericFECStatus(
     const bool enable,
     const uint8_t payload_type_red,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 7f61b7df1d..0c028bb575 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -292,8 +292,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
   // Send a request for a keyframe.
   virtual int32_t RequestKeyFrame() OVERRIDE;
 
-  virtual int32_t SetCameraDelay(int32_t delay_ms) OVERRIDE;
-
   virtual void SetTargetSendBitrate(
       const std::vector<uint32_t>& stream_bitrates) OVERRIDE;
 
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 7e010b1f06..90abacc164 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -618,7 +618,6 @@ void ViEEncoder::DeliverFrame(int id,
 }
 
 void ViEEncoder::DelayChanged(int id, int frame_delay) {
-  default_rtp_rtcp_->SetCameraDelay(frame_delay);
 }
 
 int ViEEncoder::GetPreferedFrameSettings(int* width,