diff --git a/audio/channel.cc b/audio/channel.cc
index 7dd04d696e..a7325c32a9 100644
--- a/audio/channel.cc
+++ b/audio/channel.cc
@@ -362,10 +362,6 @@ void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
   _rtpRtcpModule->SetRemoteSSRC(ssrc);
 }
 
-void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
-  // TODO(saza): remove.
-}
-
 int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
                                        size_t payloadSize,
                                        const WebRtcRTPHeader* rtpHeader) {
diff --git a/audio/channel.h b/audio/channel.h
index dbb670aa85..ab25aad0ca 100644
--- a/audio/channel.h
+++ b/audio/channel.h
@@ -250,7 +250,6 @@ class Channel
 
   // From RtpFeedback in the RTP/RTCP module
   void OnIncomingSSRCChanged(uint32_t ssrc) override;
-  void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
 
   // From Transport (called by the RTP/RTCP module)
   bool SendRtp(const uint8_t* data,
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index f66ebb3cc4..28ee47b5b5 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -257,8 +257,6 @@ class RtpFeedback {
   virtual ~RtpFeedback() {}
 
   virtual void OnIncomingSSRCChanged(uint32_t ssrc) = 0;
-
-  virtual void OnIncomingCSRCChanged(uint32_t csrc, bool added) = 0;
 };
 
 class RtcpIntraFrameObserver {
@@ -443,7 +441,6 @@ class NullRtpFeedback : public RtpFeedback {
   ~NullRtpFeedback() override {}
 
   void OnIncomingSSRCChanged(uint32_t ssrc) override {}
-  void OnIncomingCSRCChanged(uint32_t csrc, bool added) override {}
 };
 
 // Statistics about packet loss for a single directional connection. All values
diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/modules/rtp_rtcp/source/rtp_receiver_audio.cc
index 26410ae4dc..e3bbdd5e7b 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_audio.cc
@@ -71,11 +71,6 @@ bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type) {
          payload_type == cng_fb_payload_type_;
 }
 
-bool RTPReceiverAudio::ShouldReportCsrcChanges(uint8_t payload_type) const {
-  // Don't do this for DTMF packets, otherwise it's fine.
-  return !TelephoneEventPayloadType(payload_type);
-}
-
 // -   Sample based or frame based codecs based on RFC 3551
 // -
 // -   NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples.
diff --git a/modules/rtp_rtcp/source/rtp_receiver_audio.h b/modules/rtp_rtcp/source/rtp_receiver_audio.h
index 6e6cfc1496..ded9d42426 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -52,8 +52,6 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
 
   RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
 
-  bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
-
   int32_t OnNewPayloadTypeCreated(int payload_type,
                                   const SdpAudioFormat& audio_format) override;
 
diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 1fd45bd3ff..60056c81fd 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -102,11 +102,7 @@ RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
 }
 
-RtpReceiverImpl::~RtpReceiverImpl() {
-  for (int i = 0; i < num_csrcs_; ++i) {
-    cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
-  }
-}
+RtpReceiverImpl::~RtpReceiverImpl() {}
 
 int32_t RtpReceiverImpl::RegisterReceivePayload(
     int payload_type,
@@ -342,84 +338,21 @@ int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
 
 // Implementation note: must not hold critsect when called.
 void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
-  int32_t num_csrcs_diff = 0;
-  uint32_t old_remote_csrc[kRtpCsrcSize];
-  uint8_t old_num_csrcs = 0;
-
+  const uint8_t num_csrcs = rtp_header.header.numCSRCs;
+  if (num_csrcs > kRtpCsrcSize) {
+    // Ignore.
+    return;
+  }
   {
     rtc::CritScope lock(&critical_section_rtp_receiver_);
 
-    if (!rtp_media_receiver_->ShouldReportCsrcChanges(
-        rtp_header.header.payloadType)) {
-      return;
-    }
-    old_num_csrcs  = num_csrcs_;
-    if (old_num_csrcs > 0) {
-      // Make a copy of old.
-      memcpy(old_remote_csrc, current_remote_csrc_,
-             num_csrcs_ * sizeof(uint32_t));
-    }
-    const uint8_t num_csrcs = rtp_header.header.numCSRCs;
-    if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) {
-      // Copy new.
-      memcpy(current_remote_csrc_,
-             rtp_header.header.arrOfCSRCs,
-             num_csrcs * sizeof(uint32_t));
-    }
-    if (num_csrcs > 0 || old_num_csrcs > 0) {
-      num_csrcs_diff = num_csrcs - old_num_csrcs;
-      num_csrcs_ = num_csrcs;  // Update stored CSRCs.
-    } else {
-      // No change.
-      return;
-    }
+    // Copy new.
+    memcpy(current_remote_csrc_,
+           rtp_header.header.arrOfCSRCs,
+           num_csrcs * sizeof(uint32_t));
+
+    num_csrcs_ = num_csrcs;
   }  // End critsect.
-
-  bool have_called_callback = false;
-  // Search for new CSRC in old array.
-  for (uint8_t i = 0; i < rtp_header.header.numCSRCs; ++i) {
-    const uint32_t csrc = rtp_header.header.arrOfCSRCs[i];
-
-    bool found_match = false;
-    for (uint8_t j = 0; j < old_num_csrcs; ++j) {
-      if (csrc == old_remote_csrc[j]) {  // old list
-        found_match = true;
-        break;
-      }
-    }
-    if (!found_match && csrc) {
-      // Didn't find it, report it as new.
-      have_called_callback = true;
-      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, true);
-    }
-  }
-  // Search for old CSRC in new array.
-  for (uint8_t i = 0; i < old_num_csrcs; ++i) {
-    const uint32_t csrc = old_remote_csrc[i];
-
-    bool found_match = false;
-    for (uint8_t j = 0; j < rtp_header.header.numCSRCs; ++j) {
-      if (csrc == rtp_header.header.arrOfCSRCs[j]) {
-        found_match = true;
-        break;
-      }
-    }
-    if (!found_match && csrc) {
-      // Did not find it, report as removed.
-      have_called_callback = true;
-      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, false);
-    }
-  }
-  if (!have_called_callback) {
-    // If the CSRC list contain non-unique entries we will end up here.
-    // Using CSRC 0 to signal this event, not interop safe, other
-    // implementations might have CSRC 0 as a valid value.
-    if (num_csrcs_diff > 0) {
-      cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
-    } else if (num_csrcs_diff < 0) {
-      cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
-    }
-  }
 }
 
 void RtpReceiverImpl::UpdateSources(
diff --git a/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index d5c51a278c..52cb5935fc 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -49,10 +49,6 @@ class RTPReceiverStrategy {
   virtual RTPAliveType ProcessDeadOrAlive(
       uint16_t last_payload_length) const = 0;
 
-  // Returns true if we should report CSRC changes for this payload type.
-  // TODO(phoglund): should move out of here along with other payload stuff.
-  virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
-
   // Notifies the strategy that we have created a new non-RED audio payload type
   // in the payload registry.
   virtual int32_t OnNewPayloadTypeCreated(
diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.cc b/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 6efe6b52c3..454d3ef827 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -38,11 +38,6 @@ RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
 RTPReceiverVideo::~RTPReceiverVideo() {
 }
 
-bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const {
-  // Always do this for video packets.
-  return true;
-}
-
 int32_t RTPReceiverVideo::OnNewPayloadTypeCreated(
     int payload_type,
     const SdpAudioFormat& audio_format) {
diff --git a/modules/rtp_rtcp/source/rtp_receiver_video.h b/modules/rtp_rtcp/source/rtp_receiver_video.h
index 3e95456de2..d9404d1ef7 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -35,8 +35,6 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
 
   RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
 
-  bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
-
   int32_t OnNewPayloadTypeCreated(int payload_type,
                                   const SdpAudioFormat& audio_format) override;
 
diff --git a/video/rtp_video_stream_receiver.h b/video/rtp_video_stream_receiver.h
index 83d52d681a..b101e03de9 100644
--- a/video/rtp_video_stream_receiver.h
+++ b/video/rtp_video_stream_receiver.h
@@ -109,7 +109,6 @@ class RtpVideoStreamReceiver : public RtpData,
 
   // Implements RtpFeedback.
   void OnIncomingSSRCChanged(uint32_t ssrc) override {}
-  void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override {}
 
   // Implements VCMFrameTypeCallback.
   int32_t RequestKeyFrame() override;