diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
index ed44dad5e5..2b85452ac3 100644
--- a/webrtc/call/rampup_tests.cc
+++ b/webrtc/call/rampup_tests.cc
@@ -207,8 +207,9 @@ void RampUpTester::ModifyVideoConfigs(
       recv_config.rtp.ulpfec.ulpfec_payload_type =
           send_config->rtp.ulpfec.ulpfec_payload_type;
       if (rtx_) {
-        recv_config.rtp.ulpfec.red_rtx_payload_type =
-            send_config->rtp.ulpfec.red_rtx_payload_type;
+        recv_config.rtp.rtx_associated_payload_types
+            [send_config->rtp.ulpfec.red_rtx_payload_type] =
+            send_config->rtp.ulpfec.red_payload_type;
       }
     }
 
diff --git a/webrtc/call/rtx_receive_stream.cc b/webrtc/call/rtx_receive_stream.cc
index 16463525c7..6a5432fea6 100644
--- a/webrtc/call/rtx_receive_stream.cc
+++ b/webrtc/call/rtx_receive_stream.cc
@@ -11,17 +11,21 @@
 #include <utility>
 
 #include "webrtc/call/rtx_receive_stream.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/rtc_base/logging.h"
 
 namespace webrtc {
 
-RtxReceiveStream::RtxReceiveStream(RtpPacketSinkInterface* media_sink,
-                                   std::map<int, int> associated_payload_types,
-                                   uint32_t media_ssrc)
+RtxReceiveStream::RtxReceiveStream(
+    RtpPacketSinkInterface* media_sink,
+    std::map<int, int> associated_payload_types,
+    uint32_t media_ssrc,
+    ReceiveStatistics* rtp_receive_statistics /* = nullptr */)
     : media_sink_(media_sink),
       associated_payload_types_(std::move(associated_payload_types)),
-      media_ssrc_(media_ssrc) {
+      media_ssrc_(media_ssrc),
+      rtp_receive_statistics_(rtp_receive_statistics) {
   if (associated_payload_types_.empty()) {
     LOG(LS_WARNING)
         << "RtxReceiveStream created with empty payload type mapping.";
@@ -31,6 +35,12 @@ RtxReceiveStream::RtxReceiveStream(RtpPacketSinkInterface* media_sink,
 RtxReceiveStream::~RtxReceiveStream() = default;
 
 void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
+  if (rtp_receive_statistics_) {
+    RTPHeader header;
+    rtx_packet.GetHeader(&header);
+    rtp_receive_statistics_->IncomingPacket(header, rtx_packet.size(),
+                                            false /* retransmitted */);
+  }
   rtc::ArrayView<const uint8_t> payload = rtx_packet.payload();
 
   if (payload.size() < kRtxHeaderSize) {
diff --git a/webrtc/call/rtx_receive_stream.h b/webrtc/call/rtx_receive_stream.h
index 418775c778..c288a27eda 100644
--- a/webrtc/call/rtx_receive_stream.h
+++ b/webrtc/call/rtx_receive_stream.h
@@ -17,13 +17,20 @@
 
 namespace webrtc {
 
+class ReceiveStatistics;
+
 // This class is responsible for RTX decapsulation. The resulting media packets
 // are passed on to a sink representing the associated media stream.
 class RtxReceiveStream : public RtpPacketSinkInterface {
  public:
   RtxReceiveStream(RtpPacketSinkInterface* media_sink,
                    std::map<int, int> associated_payload_types,
-                   uint32_t media_ssrc);
+                   uint32_t media_ssrc,
+                   // TODO(nisse): Delete this argument, and
+                   // corresponding member variable, by moving the
+                   // responsibility for rtcp feedback to
+                   // RtpStreamReceiverController.
+                   ReceiveStatistics* rtp_receive_statistics = nullptr);
   ~RtxReceiveStream() override;
   // RtpPacketSinkInterface.
   void OnRtpPacket(const RtpPacketReceived& packet) override;
@@ -35,6 +42,7 @@ class RtxReceiveStream : public RtpPacketSinkInterface {
   // TODO(nisse): Ultimately, the media receive stream shouldn't care about the
   // ssrc, and we should delete this.
   const uint32_t media_ssrc_;
+  ReceiveStatistics* const rtp_receive_statistics_;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/call/video_receive_stream.h b/webrtc/call/video_receive_stream.h
index e51184e2dd..f199ca1f85 100644
--- a/webrtc/call/video_receive_stream.h
+++ b/webrtc/call/video_receive_stream.h
@@ -167,6 +167,11 @@ class VideoReceiveStream {
       NackConfig nack;
 
       // See UlpfecConfig for description.
+      // TODO(nisse): UlpfecConfig includes the field red_rtx_payload_type,
+      // which duplicates info in the rtx_associated_payload_types mapping. So
+      // delete the use of UlpfecConfig here, and replace by the values which
+      // make sense in this context, likely those are ulpfec_payload_type_ and
+      // red_payload_type_.
       UlpfecConfig ulpfec;
 
       // SSRC for retransmissions.
diff --git a/webrtc/media/engine/webrtcvideoengine.cc b/webrtc/media/engine/webrtcvideoengine.cc
index 9c0031ab48..a613dc6de6 100644
--- a/webrtc/media/engine/webrtcvideoengine.cc
+++ b/webrtc/media/engine/webrtcvideoengine.cc
@@ -2187,6 +2187,11 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureCodecs(
 
   config_.rtp.nack.rtp_history_ms =
       HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
+  if (config_.rtp.ulpfec.red_rtx_payload_type != -1) {
+    config_.rtp
+        .rtx_associated_payload_types[config_.rtp.ulpfec.red_rtx_payload_type] =
+        config_.rtp.ulpfec.red_payload_type;
+  }
 }
 
 void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureFlexfecCodec(
diff --git a/webrtc/media/engine/webrtcvideoengine_unittest.cc b/webrtc/media/engine/webrtcvideoengine_unittest.cc
index c6adfd3976..1306082abf 100644
--- a/webrtc/media/engine/webrtcvideoengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvideoengine_unittest.cc
@@ -83,6 +83,32 @@ bool HasRtxCodec(const std::vector<cricket::VideoCodec>& codecs,
   return false;
 }
 
+// TODO(nisse): Duplicated in call.cc.
+const int* FindKeyByValue(const std::map<int, int>& m, int v) {
+  for (const auto& kv : m) {
+    if (kv.second == v)
+      return &kv.first;
+  }
+  return nullptr;
+}
+
+bool HasRtxReceiveAssociation(
+    const webrtc::VideoReceiveStream::Config& config,
+    int payload_type) {
+  return FindKeyByValue(config.rtp.rtx_associated_payload_types,
+                        payload_type) != nullptr;
+}
+
+// Check that there's an Rtx payload type for each decoder.
+bool VerifyRtxReceiveAssociations(
+    const webrtc::VideoReceiveStream::Config& config) {
+  for (const auto& decoder : config.decoders) {
+    if (!HasRtxReceiveAssociation(config, decoder.payload_type))
+      return false;
+  }
+  return true;
+}
+
 rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer(
     int width,
     int height) {
@@ -112,15 +138,6 @@ cricket::MediaConfig GetMediaConfig() {
   return media_config;
 }
 
-// TODO(nisse): Duplicated in call.cc.
-const int* FindKeyByValue(const std::map<int, int>& m, int v) {
-  for (const auto& kv : m) {
-    if (kv.second == v)
-      return &kv.first;
-  }
-  return nullptr;
-}
-
 }  // namespace
 
 namespace cricket {
@@ -1316,9 +1333,12 @@ TEST_F(WebRtcVideoChannelTest, RecvStreamWithSimAndRtx) {
       cricket::CreateSimWithRtxStreamParams("cname", ssrcs, rtx_ssrcs));
   EXPECT_FALSE(
       recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
-  EXPECT_EQ(recv_stream->GetConfig().decoders.size(),
-            recv_stream->GetConfig().rtp.rtx_associated_payload_types.size())
+  EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
       << "RTX should be mapped for all decoders/payload types.";
+  EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
+                                          GetEngineCodec("red").id))
+      << "RTX should be mapped for the RED payload type";
+
   EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
 }
 
@@ -1329,6 +1349,12 @@ TEST_F(WebRtcVideoChannelTest, RecvStreamWithRtx) {
   params.AddFidSsrc(kSsrcs1[0], kRtxSsrcs1[0]);
   FakeVideoReceiveStream* recv_stream = AddRecvStream(params);
   EXPECT_EQ(kRtxSsrcs1[0], recv_stream->GetConfig().rtp.rtx_ssrc);
+
+  EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
+      << "RTX should be mapped for all decoders/payload types.";
+  EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
+                                          GetEngineCodec("red").id))
+      << "RTX should be mapped for the RED payload type";
 }
 
 TEST_F(WebRtcVideoChannelTest, RecvStreamNoRtx) {
@@ -3796,9 +3822,11 @@ TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) {
   recv_stream = fake_call_->GetVideoReceiveStreams()[0];
   EXPECT_FALSE(
       recv_stream->GetConfig().rtp.rtx_associated_payload_types.empty());
-  EXPECT_EQ(recv_stream->GetConfig().decoders.size(),
-            recv_stream->GetConfig().rtp.rtx_associated_payload_types.size())
+  EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
       << "RTX should be mapped for all decoders/payload types.";
+  EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
+                                          GetEngineCodec("red").id))
+      << "RTX should be mapped also for the RED payload type";
   EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
 }
 
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 7756ff416d..16f6bd0050 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -61,6 +61,9 @@ rtc_static_library("video") {
     "../call:call_interfaces",
     "../call:rtp_interfaces",
     "../call:video_stream_api",
+
+    # For RtxReceiveStream.
+    "../call:rtp_receiver",
     "../common_video",
     "../logging:rtc_event_log_api",
     "../media:rtc_media_base",
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index ffd52fd531..e92d8fe2d4 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1184,7 +1184,9 @@ void EndToEndTest::DecodesRetransmittedFrame(bool enable_rtx, bool enable_red) {
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
         (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
         (*receive_configs)[0]
-            .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
+            .rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType)
+                                                  ? kRtxRedPayloadType
+                                                  : kSendRtxPayloadType] =
             payload_type_;
       }
       // Configure encoding and decoding with VP8, since generic packetization
diff --git a/webrtc/video/rtp_video_stream_receiver.cc b/webrtc/video/rtp_video_stream_receiver.cc
index a47055abf0..1dbd86988d 100644
--- a/webrtc/video/rtp_video_stream_receiver.cc
+++ b/webrtc/video/rtp_video_stream_receiver.cc
@@ -86,6 +86,7 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
     RtcpRttStats* rtt_stats,
     PacketRouter* packet_router,
     const VideoReceiveStream::Config* config,
+    ReceiveStatistics* rtp_receive_statistics,
     ReceiveStatisticsProxy* receive_stats_proxy,
     ProcessThread* process_thread,
     NackSender* nack_sender,
@@ -102,12 +103,11 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
                                                      this,
                                                      this,
                                                      &rtp_payload_registry_)),
-      rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
+      rtp_receive_statistics_(rtp_receive_statistics),
       ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, this)),
       receiving_(false),
-      restored_packet_in_use_(false),
       last_packet_log_ms_(-1),
-      rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(),
+      rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_,
                                     transport,
                                     rtt_stats,
                                     receive_stats_proxy,
@@ -146,12 +146,8 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
   rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
 
   if (config_.rtp.rtx_ssrc) {
+    // Needed for rtp_payload_registry_.RtxEnabled().
     rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc);
-
-    for (const auto& kv : config_.rtp.rtx_associated_payload_types) {
-      RTC_DCHECK_NE(kv.first, 0);
-      rtp_payload_registry_.SetRtxPayloadType(kv.first, kv.second);
-    }
   }
 
   if (IsUlpfecEnabled()) {
@@ -168,11 +164,6 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
     strncpy(red_codec.plName, "red", sizeof(red_codec.plName));
     red_codec.plType = config_.rtp.ulpfec.red_payload_type;
     RTC_CHECK(AddReceiveCodec(red_codec));
-    if (config_.rtp.ulpfec.red_rtx_payload_type != -1) {
-      rtp_payload_registry_.SetRtxPayloadType(
-          config_.rtp.ulpfec.red_rtx_payload_type,
-          config_.rtp.ulpfec.red_payload_type);
-    }
   }
 
   if (config_.rtp.rtcp_xr.receiver_reference_time_report)
@@ -495,31 +486,7 @@ void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
     }
     ulpfec_receiver_->ProcessReceivedFec();
   } else if (rtp_payload_registry_.IsRtx(header)) {
-    if (header.headerLength + header.paddingLength == packet_length) {
-      // This is an empty packet and should be silently dropped before trying to
-      // parse the RTX header.
-      return;
-    }
-    // Remove the RTX header and parse the original RTP header.
-    if (packet_length < header.headerLength)
-      return;
-    if (packet_length > sizeof(restored_packet_))
-      return;
-    if (restored_packet_in_use_) {
-      LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
-      return;
-    }
-    if (!rtp_payload_registry_.RestoreOriginalPacket(
-            restored_packet_, packet, &packet_length, config_.rtp.remote_ssrc,
-            header)) {
-      LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
-                      << header.ssrc << " payload type: "
-                      << static_cast<int>(header.payloadType);
-      return;
-    }
-    restored_packet_in_use_ = true;
-    OnRecoveredPacket(restored_packet_, packet_length);
-    restored_packet_in_use_ = false;
+    LOG(LS_WARNING) << "Unexpected RTX packet on media ssrc";
   }
 }
 
diff --git a/webrtc/video/rtp_video_stream_receiver.h b/webrtc/video/rtp_video_stream_receiver.h
index 3c8b2aa167..994ef55e89 100644
--- a/webrtc/video/rtp_video_stream_receiver.h
+++ b/webrtc/video/rtp_video_stream_receiver.h
@@ -72,6 +72,7 @@ class RtpVideoStreamReceiver : public RtpData,
       RtcpRttStats* rtt_stats,
       PacketRouter* packet_router,
       const VideoReceiveStream::Config* config,
+      ReceiveStatistics* rtp_receive_statistics,
       ReceiveStatisticsProxy* receive_stats_proxy,
       ProcessThread* process_thread,
       NackSender* nack_sender,
@@ -180,13 +181,11 @@ class RtpVideoStreamReceiver : public RtpData,
 
   const std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   const std::unique_ptr<RtpReceiver> rtp_receiver_;
-  const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+  ReceiveStatistics* const rtp_receive_statistics_;
   std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
 
   rtc::SequencedTaskChecker worker_task_checker_;
   bool receiving_ RTC_GUARDED_BY(worker_task_checker_);
-  uint8_t restored_packet_[IP_PACKET_SIZE] RTC_GUARDED_BY(worker_task_checker_);
-  bool restored_packet_in_use_ RTC_GUARDED_BY(worker_task_checker_);
   int64_t last_packet_log_ms_ RTC_GUARDED_BY(worker_task_checker_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
diff --git a/webrtc/video/rtp_video_stream_receiver_unittest.cc b/webrtc/video/rtp_video_stream_receiver_unittest.cc
index d309397dd4..4f2dca7338 100644
--- a/webrtc/video/rtp_video_stream_receiver_unittest.cc
+++ b/webrtc/video/rtp_video_stream_receiver_unittest.cc
@@ -126,11 +126,14 @@ class RtpVideoStreamReceiverTest : public testing::Test {
         process_thread_(ProcessThread::Create("TestThread")) {}
 
   void SetUp() {
-    rtp_video_stream_receiver_.reset(new RtpVideoStreamReceiver(
+    rtp_receive_statistics_ =
+        rtc::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
+    rtp_video_stream_receiver_ = rtc::MakeUnique<RtpVideoStreamReceiver>(
         &mock_transport_, nullptr, &packet_router_, &config_,
-        nullptr, process_thread_.get(), &mock_nack_sender_,
+        rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
+        &mock_nack_sender_,
         &mock_key_frame_request_sender_, &mock_on_complete_frame_callback_,
-        &timing_));
+        &timing_);
   }
 
   WebRtcRTPHeader GetDefaultPacket() {
@@ -196,6 +199,7 @@ class RtpVideoStreamReceiverTest : public testing::Test {
   PacketRouter packet_router_;
   VCMTiming timing_;
   std::unique_ptr<ProcessThread> process_thread_;
+  std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_;
 };
 
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 11e060566b..6294184f2a 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -1555,6 +1555,9 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
             video_send_config_.rtp.ulpfec.ulpfec_payload_type;
         it->rtp.ulpfec.red_rtx_payload_type =
             video_send_config_.rtp.ulpfec.red_rtx_payload_type;
+        it->rtp.rtx_associated_payload_types[video_send_config_.rtp.ulpfec
+                                                 .red_rtx_payload_type] =
+            video_send_config_.rtp.ulpfec.red_payload_type;
       }
     } else {
       video_receive_configs_[params_.ss.selected_stream]
@@ -1566,6 +1569,10 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
       video_receive_configs_[params_.ss.selected_stream]
           .rtp.ulpfec.red_rtx_payload_type =
           video_send_config_.rtp.ulpfec.red_rtx_payload_type;
+      video_receive_configs_[params_.ss.selected_stream]
+          .rtp.rtx_associated_payload_types[video_send_config_.rtp.ulpfec
+                                                .red_rtx_payload_type] =
+          video_send_config_.rtp.ulpfec.red_payload_type;
     }
   }
 }
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 74c90134b9..a89e3ce366 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -18,6 +18,7 @@
 
 #include "webrtc/api/optional.h"
 #include "webrtc/call/rtp_stream_receiver_controller_interface.h"
+#include "webrtc/call/rtx_receive_stream.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/h264/profile_level_id.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@@ -32,6 +33,7 @@
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/location.h"
 #include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
@@ -88,6 +90,7 @@ VideoReceiveStream::VideoReceiveStream(
                      "DecodingThread",
                      rtc::kHighestPriority),
       call_stats_(call_stats),
+      rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
       timing_(new VCMTiming(clock_)),
       video_receiver_(clock_, nullptr, this, timing_.get(), this, this),
       stats_proxy_(&config_, clock_),
@@ -95,6 +98,7 @@ VideoReceiveStream::VideoReceiveStream(
                                  call_stats_->rtcp_rtt_stats(),
                                  packet_router,
                                  &config_,
+                                 rtp_receive_statistics_.get(),
                                  &stats_proxy_,
                                  process_thread_,
                                  this,  // NackSender
@@ -120,7 +124,7 @@ VideoReceiveStream::VideoReceiveStream(
     decoder_payload_types.insert(decoder.payload_type);
   }
 
-  video_receiver_.SetRenderDelay(config.render_delay_ms);
+  video_receiver_.SetRenderDelay(config_.render_delay_ms);
 
   jitter_estimator_.reset(new VCMJitterEstimator(clock_));
   frame_buffer_.reset(new video_coding::FrameBuffer(
@@ -131,9 +135,12 @@ VideoReceiveStream::VideoReceiveStream(
   // Register with RtpStreamReceiverController.
   media_receiver_ = receiver_controller->CreateReceiver(
       config_.rtp.remote_ssrc, &rtp_video_stream_receiver_);
-  if (config.rtp.rtx_ssrc) {
+  if (config_.rtp.rtx_ssrc) {
+    rtx_receive_stream_ = rtc::MakeUnique<RtxReceiveStream>(
+        &rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types,
+        config_.rtp.remote_ssrc, rtp_receive_statistics_.get());
     rtx_receiver_ = receiver_controller->CreateReceiver(
-        config_.rtp.rtx_ssrc, &rtp_video_stream_receiver_);
+        config_.rtp.rtx_ssrc, rtx_receive_stream_.get());
   }
 }
 
diff --git a/webrtc/video/video_receive_stream.h b/webrtc/video/video_receive_stream.h
index 32ce812326..8b680aeb1e 100644
--- a/webrtc/video/video_receive_stream.h
+++ b/webrtc/video/video_receive_stream.h
@@ -38,6 +38,7 @@ class ProcessThread;
 class RTPFragmentationHeader;
 class RtpStreamReceiverInterface;
 class RtpStreamReceiverControllerInterface;
+class RtxReceiveStream;
 class VCMTiming;
 class VCMJitterEstimator;
 
@@ -125,6 +126,10 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
 
   CallStats* const call_stats_;
 
+  // Shared by media and rtx stream receivers, since the latter has no RtpRtcp
+  // module of its own.
+  const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+
   std::unique_ptr<VCMTiming> timing_;  // Jitter buffer experiment.
   vcm::VideoReceiver video_receiver_;
   std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
@@ -141,6 +146,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
   std::unique_ptr<video_coding::FrameBuffer> frame_buffer_;
 
   std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
+  std::unique_ptr<RtxReceiveStream> rtx_receive_stream_;
   std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
 
   // Whenever we are in an undecodable state (stream has just started or due to