diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 6364202..06c660e 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -19,6 +19,7 @@
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
@@ -40,6 +41,11 @@
 }  // namespace
 
 namespace internal {
+// TODO(elad.alon): Subsequent CL will make these values experiment-dependent.
+constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
+constexpr size_t kPacketLossRateMinNumAckedPackets = 50;
+constexpr size_t kRecoverablePacketLossRateMinNumAckedPairs = 40;
+
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -53,7 +59,10 @@
       config_(config),
       audio_state_(audio_state),
       bitrate_allocator_(bitrate_allocator),
-      send_side_cc_(send_side_cc) {
+      send_side_cc_(send_side_cc),
+      packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
+                           kPacketLossRateMinNumAckedPackets,
+                           kRecoverablePacketLossRateMinNumAckedPairs) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
@@ -72,6 +81,7 @@
                                 config_.rtp.nack.rtp_history_ms / 20);
 
   channel_proxy_->RegisterExternalTransport(config.send_transport);
+  send_side_cc_->RegisterPacketFeedbackObserver(this);
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kAudioLevelUri) {
@@ -91,11 +101,14 @@
   if (!SetupSendCodec()) {
     LOG(LS_ERROR) << "Failed to set up send codec state.";
   }
+
+  pacer_thread_checker_.DetachFromThread();
 }
 
 AudioSendStream::~AudioSendStream() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+  send_side_cc_->DeRegisterPacketFeedbackObserver(this);
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
@@ -103,7 +116,7 @@
 }
 
 void AudioSendStream::Start() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps);
     rtc::Event thread_sync_event(false /* manual_reset */, false);
@@ -123,7 +136,7 @@
 }
 
 void AudioSendStream::Stop() {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   rtc::Event thread_sync_event(false /* manual_reset */, false);
   worker_queue_->PostTask([this, &thread_sync_event] {
     bitrate_allocator_->RemoveObserver(this);
@@ -141,19 +154,19 @@
 bool AudioSendStream::SendTelephoneEvent(int payload_type,
                                          int payload_frequency, int event,
                                          int duration_ms) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type,
                                                           payload_frequency) &&
          channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
 }
 
 void AudioSendStream::SetMuted(bool muted) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   channel_proxy_->SetInputMute(muted);
 }
 
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
 
@@ -217,14 +230,14 @@
 }
 
 void AudioSendStream::SignalNetworkState(NetworkState state) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 }
 
 bool AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
-  // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
+  // RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
@@ -247,13 +260,43 @@
   return 0;
 }
 
+void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
+  RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
+  // Only packets that belong to this stream are of interest.
+  if (ssrc == config_.rtp.ssrc) {
+    rtc::CritScope lock(&packet_loss_tracker_cs_);
+    // TODO(elad.alon): This function call could potentially reset the window,
+    // setting both PLR and RPLR to unknown. Consider (during upcoming
+    // refactoring) passing an indication of such an event.
+    packet_loss_tracker_.OnPacketAdded(seq_num, rtc::TimeMillis());
+  }
+}
+
+void AudioSendStream::OnPacketFeedbackVector(
+    const std::vector<PacketFeedback>& packet_feedback_vector) {
+  // TODO(elad.alon): This fails in UT; fix and uncomment.
+  // RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  rtc::Optional<float> plr;
+  {
+    rtc::CritScope lock(&packet_loss_tracker_cs_);
+    packet_loss_tracker_.OnPacketFeedbackVector(packet_feedback_vector);
+    plr = packet_loss_tracker_.GetPacketLossRate();
+  }
+  // TODO(elad.alon): If PLR goes back to unknown, no indication is given that
+  // the previously sent value is no longer relevant. This will be taken care
+  // of with some refactoring which is now being done.
+  if (plr) {
+    channel_proxy_->OnTwccBasedUplinkPacketLossRate(*plr);
+  }
+}
+
 const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return config_;
 }
 
 void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   send_side_cc_->SetTransportOverhead(transport_overhead_per_packet);
   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
 }
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 436c498..f50f7c4 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -12,12 +12,15 @@
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
+#include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h"
 
 namespace webrtc {
 class SendSideCongestionController;
@@ -33,7 +36,8 @@
 
 namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream,
-                              public webrtc::BitrateAllocatorObserver {
+                              public webrtc::BitrateAllocatorObserver,
+                              public webrtc::PacketFeedbackObserver {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -62,6 +66,11 @@
                             int64_t rtt,
                             int64_t probing_interval_ms) override;
 
+  // From PacketFeedbackObserver.
+  void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
+  void OnPacketFeedbackVector(
+      const std::vector<PacketFeedback>& packet_feedback_vector) override;
+
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
@@ -70,7 +79,8 @@
 
   bool SetupSendCodec();
 
-  rtc::ThreadChecker thread_checker_;
+  rtc::ThreadChecker worker_thread_checker_;
+  rtc::ThreadChecker pacer_thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
@@ -80,6 +90,10 @@
   SendSideCongestionController* const send_side_cc_;
   std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
 
+  rtc::CriticalSection packet_loss_tracker_cs_;
+  TransportFeedbackPacketLossTracker packet_loss_tracker_
+      GUARDED_BY(&packet_loss_tracker_cs_);
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
diff --git a/webrtc/modules/congestion_controller/congestion_controller.cc b/webrtc/modules/congestion_controller/congestion_controller.cc
index 9b836be..62772b7 100644
--- a/webrtc/modules/congestion_controller/congestion_controller.cc
+++ b/webrtc/modules/congestion_controller/congestion_controller.cc
@@ -106,10 +106,11 @@
   receive_side_cc_.Process();
 }
 
-void CongestionController::AddPacket(uint16_t sequence_number,
+void CongestionController::AddPacket(uint32_t ssrc,
+                                     uint16_t sequence_number,
                                      size_t length,
                                      const PacedPacketInfo& pacing_info) {
-  send_side_cc_.AddPacket(sequence_number, length, pacing_info);
+  send_side_cc_.AddPacket(ssrc, sequence_number, length, pacing_info);
 }
 
 void CongestionController::OnTransportFeedback(
diff --git a/webrtc/modules/congestion_controller/congestion_controller_unittest.cc b/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
index 8073ae1..5621936 100644
--- a/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
+++ b/webrtc/modules/congestion_controller/congestion_controller_unittest.cc
@@ -79,7 +79,8 @@
   }
 
   void OnSentPacket(const PacketFeedback& packet_feedback) {
-    controller_->AddPacket(packet_feedback.sequence_number,
+    constexpr uint32_t ssrc = 0;
+    controller_->AddPacket(ssrc, packet_feedback.sequence_number,
                            packet_feedback.payload_size,
                            packet_feedback.pacing_info);
     controller_->OnSentPacket(rtc::SentPacket(packet_feedback.sequence_number,
diff --git a/webrtc/modules/congestion_controller/include/congestion_controller.h b/webrtc/modules/congestion_controller/include/congestion_controller.h
index e962f9f..be2d68d 100644
--- a/webrtc/modules/congestion_controller/include/congestion_controller.h
+++ b/webrtc/modules/congestion_controller/include/congestion_controller.h
@@ -119,7 +119,8 @@
   void Process() override;
 
   // Implements TransportFeedbackObserver.
-  void AddPacket(uint16_t sequence_number,
+  void AddPacket(uint32_t ssrc,
+                 uint16_t sequence_number,
                  size_t length,
                  const PacedPacketInfo& pacing_info) override;
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
diff --git a/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h b/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
index 9aa4727..8130feb 100644
--- a/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
+++ b/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
@@ -67,6 +67,9 @@
                                std::unique_ptr<PacedSender> pacer);
   virtual ~SendSideCongestionController();
 
+  void RegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
+  void DeRegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
+
   virtual void SetBweBitrates(int min_bitrate_bps,
                               int start_bitrate_bps,
                               int max_bitrate_bps);
@@ -111,7 +114,8 @@
   void Process() override;
 
   // Implements TransportFeedbackObserver.
-  void AddPacket(uint16_t sequence_number,
+  void AddPacket(uint32_t ssrc,
+                 uint16_t sequence_number,
                  size_t length,
                  const PacedPacketInfo& pacing_info) override;
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
diff --git a/webrtc/modules/congestion_controller/send_side_congestion_controller.cc b/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
index f3cbed9..782c060 100644
--- a/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
+++ b/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
@@ -83,6 +83,16 @@
 
 SendSideCongestionController::~SendSideCongestionController() {}
 
+void SendSideCongestionController::RegisterPacketFeedbackObserver(
+    PacketFeedbackObserver* observer) {
+  transport_feedback_adapter_.RegisterPacketFeedbackObserver(observer);
+}
+
+void SendSideCongestionController::DeRegisterPacketFeedbackObserver(
+    PacketFeedbackObserver* observer) {
+  transport_feedback_adapter_.DeRegisterPacketFeedbackObserver(observer);
+}
+
 void SendSideCongestionController::SetBweBitrates(int min_bitrate_bps,
                                                   int start_bitrate_bps,
                                                   int max_bitrate_bps) {
@@ -203,10 +213,12 @@
 }
 
 void SendSideCongestionController::AddPacket(
+    uint32_t ssrc,
     uint16_t sequence_number,
     size_t length,
     const PacedPacketInfo& pacing_info) {
-  transport_feedback_adapter_.AddPacket(sequence_number, length, pacing_info);
+  transport_feedback_adapter_.AddPacket(ssrc, sequence_number, length,
+                                        pacing_info);
 }
 
 void SendSideCongestionController::OnTransportFeedback(
diff --git a/webrtc/modules/congestion_controller/transport_feedback_adapter.cc b/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
index ddffa72..9fb1af5 100644
--- a/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
+++ b/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
@@ -36,19 +36,49 @@
       local_net_id_(0),
       remote_net_id_(0) {}
 
-TransportFeedbackAdapter::~TransportFeedbackAdapter() {}
+TransportFeedbackAdapter::~TransportFeedbackAdapter() {
+  RTC_DCHECK(observers_.empty());
+}
 
-void TransportFeedbackAdapter::AddPacket(uint16_t sequence_number,
+void TransportFeedbackAdapter::RegisterPacketFeedbackObserver(
+    PacketFeedbackObserver* observer) {
+  rtc::CritScope cs(&observers_lock_);
+  RTC_DCHECK(observer);
+  RTC_DCHECK(std::find(observers_.begin(), observers_.end(), observer) ==
+             observers_.end());
+  observers_.push_back(observer);
+}
+
+void TransportFeedbackAdapter::DeRegisterPacketFeedbackObserver(
+    PacketFeedbackObserver* observer) {
+  rtc::CritScope cs(&observers_lock_);
+  RTC_DCHECK(observer);
+  const auto it = std::find(observers_.begin(), observers_.end(), observer);
+  RTC_DCHECK(it != observers_.end());
+  observers_.erase(it);
+}
+
+void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
+                                         uint16_t sequence_number,
                                          size_t length,
                                          const PacedPacketInfo& pacing_info) {
-  rtc::CritScope cs(&lock_);
-  if (send_side_bwe_with_overhead_) {
-    length += transport_overhead_bytes_per_packet_;
+  {
+    rtc::CritScope cs(&lock_);
+    if (send_side_bwe_with_overhead_) {
+      length += transport_overhead_bytes_per_packet_;
+    }
+    const int64_t creation_time_ms = clock_->TimeInMilliseconds();
+    send_time_history_.AddAndRemoveOld(
+        PacketFeedback(creation_time_ms, sequence_number, length, local_net_id_,
+                       remote_net_id_, pacing_info));
   }
-  const int64_t creation_time_ms = clock_->TimeInMilliseconds();
-  send_time_history_.AddAndRemoveOld(
-      PacketFeedback(creation_time_ms, sequence_number, length, local_net_id_,
-                     remote_net_id_, pacing_info));
+
+  {
+    rtc::CritScope cs(&observers_lock_);
+    for (auto observer : observers_) {
+      observer->OnPacketAdded(ssrc, sequence_number);
+    }
+  }
 }
 
 void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
@@ -154,6 +184,12 @@
 void TransportFeedbackAdapter::OnTransportFeedback(
     const rtcp::TransportFeedback& feedback) {
   last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
+  {
+    rtc::CritScope cs(&observers_lock_);
+    for (auto observer : observers_) {
+      observer->OnPacketFeedbackVector(last_packet_feedback_vector_);
+    }
+  }
 }
 
 std::vector<PacketFeedback>
diff --git a/webrtc/modules/congestion_controller/transport_feedback_adapter.h b/webrtc/modules/congestion_controller/transport_feedback_adapter.h
index 616bebe..2e70e44 100644
--- a/webrtc/modules/congestion_controller/transport_feedback_adapter.h
+++ b/webrtc/modules/congestion_controller/transport_feedback_adapter.h
@@ -21,6 +21,8 @@
 
 namespace webrtc {
 
+class PacketFeedbackObserver;
+
 namespace rtcp {
 class TransportFeedback;
 }  // namespace rtcp
@@ -30,7 +32,11 @@
   explicit TransportFeedbackAdapter(const Clock* clock);
   virtual ~TransportFeedbackAdapter();
 
-  void AddPacket(uint16_t sequence_number,
+  void RegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
+  void DeRegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
+
+  void AddPacket(uint32_t ssrc,
+                 uint16_t sequence_number,
                  size_t length,
                  const PacedPacketInfo& pacing_info);
   void OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
@@ -57,8 +63,11 @@
   int64_t current_offset_ms_;
   int64_t last_timestamp_us_;
   std::vector<PacketFeedback> last_packet_feedback_vector_;
-  uint16_t local_net_id_;
-  uint16_t remote_net_id_;
+  uint16_t local_net_id_ GUARDED_BY(&lock_);
+  uint16_t remote_net_id_ GUARDED_BY(&lock_);
+
+  rtc::CriticalSection observers_lock_;
+  std::vector<PacketFeedbackObserver*> observers_ GUARDED_BY(&observers_lock_);
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc b/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
index 22e8b59..97b7f5f 100644
--- a/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
+++ b/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
@@ -38,6 +38,13 @@
 
 namespace test {
 
+class MockPacketFeedbackObserver : public webrtc::PacketFeedbackObserver {
+ public:
+  MOCK_METHOD2(OnPacketAdded, void(uint32_t ssrc, uint16_t seq_num));
+  MOCK_METHOD1(OnPacketFeedbackVector,
+               void(const std::vector<PacketFeedback>& packet_feedback_vector));
+};
+
 class TransportFeedbackAdapterTest : public ::testing::Test {
  public:
   TransportFeedbackAdapterTest() : clock_(0) {}
@@ -58,17 +65,75 @@
                                     int64_t now_ms) {}
 
   void OnSentPacket(const PacketFeedback& packet_feedback) {
-    adapter_->AddPacket(packet_feedback.sequence_number,
+    adapter_->AddPacket(kSsrc, packet_feedback.sequence_number,
                         packet_feedback.payload_size,
                         packet_feedback.pacing_info);
     adapter_->OnSentPacket(packet_feedback.sequence_number,
                            packet_feedback.send_time_ms);
   }
 
+  static constexpr uint32_t kSsrc = 8492;
+
   SimulatedClock clock_;
   std::unique_ptr<TransportFeedbackAdapter> adapter_;
 };
 
+TEST_F(TransportFeedbackAdapterTest, ObserverSanity) {
+  MockPacketFeedbackObserver mock;
+  adapter_->RegisterPacketFeedbackObserver(&mock);
+
+  const std::vector<PacketFeedback> packets = {
+      PacketFeedback(100, 200, 0, 1000, kPacingInfo0),
+      PacketFeedback(110, 210, 1, 2000, kPacingInfo0),
+      PacketFeedback(120, 220, 2, 3000, kPacingInfo0)
+  };
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(packets[0].sequence_number,
+                   packets[0].arrival_time_ms * 1000);
+
+  for (const PacketFeedback& packet : packets) {
+    EXPECT_CALL(mock, OnPacketAdded(kSsrc, packet.sequence_number)).Times(1);
+    OnSentPacket(packet);
+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
+                                           packet.arrival_time_ms * 1000));
+  }
+
+  EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(1);
+  adapter_->OnTransportFeedback(feedback);
+
+  adapter_->DeRegisterPacketFeedbackObserver(&mock);
+
+  // After deregistration, the observer no longers gets indications.
+  EXPECT_CALL(mock, OnPacketAdded(_, _)).Times(0);
+  const PacketFeedback new_packet(130, 230, 3, 4000, kPacingInfo0);
+  OnSentPacket(new_packet);
+
+  rtcp::TransportFeedback second_feedback;
+  second_feedback.SetBase(new_packet.sequence_number,
+                          new_packet.arrival_time_ms * 1000);
+  EXPECT_TRUE(feedback.AddReceivedPacket(new_packet.sequence_number,
+                                         new_packet.arrival_time_ms * 1000));
+  EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(0);
+  adapter_->OnTransportFeedback(second_feedback);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(TransportFeedbackAdapterTest, ObserverDoubleRegistrationDeathTest) {
+  MockPacketFeedbackObserver mock;
+  adapter_->RegisterPacketFeedbackObserver(&mock);
+  EXPECT_DEATH(adapter_->RegisterPacketFeedbackObserver(&mock), "");
+  adapter_->DeRegisterPacketFeedbackObserver(&mock);
+}
+
+TEST_F(TransportFeedbackAdapterTest, ObserverMissingDeRegistrationDeathTest) {
+  MockPacketFeedbackObserver mock;
+  adapter_->RegisterPacketFeedbackObserver(&mock);
+  EXPECT_DEATH(adapter_.reset(), "");
+  adapter_->DeRegisterPacketFeedbackObserver(&mock);
+}
+#endif
+
 TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
   std::vector<PacketFeedback> packets;
   packets.push_back(PacketFeedback(100, 200, 0, 1500, kPacingInfo0));
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 7df8a13..ddfec4d 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -342,7 +342,8 @@
   virtual ~TransportFeedbackObserver() {}
 
   // Note: Transport-wide sequence number as sequence number.
-  virtual void AddPacket(uint16_t sequence_number,
+  virtual void AddPacket(uint32_t ssrc,
+                         uint16_t sequence_number,
                          size_t length,
                          const PacedPacketInfo& pacing_info) = 0;
 
@@ -351,6 +352,15 @@
   virtual std::vector<PacketFeedback> GetTransportFeedbackVector() const = 0;
 };
 
+class PacketFeedbackObserver {
+ public:
+  virtual ~PacketFeedbackObserver() = default;
+
+  virtual void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) = 0;
+  virtual void OnPacketFeedbackVector(
+      const std::vector<PacketFeedback>& packet_feedback_vector) = 0;
+};
+
 class RtcpRttStats {
  public:
   virtual void OnRttUpdate(int64_t rtt) = 0;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
index 4b0255f..d419caf 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
@@ -73,8 +73,9 @@
 
 class MockTransportFeedbackObserver : public TransportFeedbackObserver {
  public:
-  MOCK_METHOD2(AddPacket, void(uint16_t, size_t));
-  MOCK_METHOD3(AddPacket, void(uint16_t, size_t, const PacedPacketInfo&));
+  MOCK_METHOD3(AddPacket, void(uint32_t, uint16_t, size_t));
+  MOCK_METHOD4(AddPacket,
+               void(uint32_t, uint16_t, size_t, const PacedPacketInfo&));
   MOCK_METHOD1(OnTransportFeedback, void(const rtcp::TransportFeedback&));
   MOCK_CONST_METHOD0(GetTransportFeedbackVector, std::vector<PacketFeedback>());
 };
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 2fd8b3e..52b0891 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -1256,7 +1256,7 @@
   }
 
   if (transport_feedback_observer_) {
-    transport_feedback_observer_->AddPacket(packet_id, packet_size,
+    transport_feedback_observer_->AddPacket(SSRC(), packet_id, packet_size,
                                             pacing_info);
   }
 }
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index fed52ee..6539f73 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -126,7 +126,8 @@
 
 class MockTransportFeedbackObserver : public TransportFeedbackObserver {
  public:
-  MOCK_METHOD3(AddPacket, void(uint16_t, size_t, const PacedPacketInfo&));
+  MOCK_METHOD4(AddPacket,
+               void(uint32_t, uint16_t, size_t, const PacedPacketInfo&));
   MOCK_METHOD1(OnTransportFeedback, void(const rtcp::TransportFeedback&));
   MOCK_CONST_METHOD0(GetTransportFeedbackVector, std::vector<PacketFeedback>());
 };
@@ -355,7 +356,7 @@
       .Times(1);
   EXPECT_CALL(
       feedback_observer_,
-      AddPacket(kTransportSequenceNumber,
+      AddPacket(rtp_sender_->SSRC(), kTransportSequenceNumber,
                 sizeof(kPayloadData) + kGenericHeaderLength, PacedPacketInfo()))
       .Times(1);
 
@@ -406,7 +407,7 @@
       .Times(1);
   EXPECT_CALL(
       feedback_observer_,
-      AddPacket(kTransportSequenceNumber,
+      AddPacket(rtp_sender_->SSRC(), kTransportSequenceNumber,
                 sizeof(kPayloadData) + kGenericHeaderLength, PacedPacketInfo()))
       .Times(1);
 
@@ -1487,7 +1488,7 @@
   EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
       .WillOnce(testing::Return(kTransportSequenceNumber));
   EXPECT_CALL(feedback_observer_,
-              AddPacket(kTransportSequenceNumber,
+              AddPacket(rtp_sender_->SSRC(), kTransportSequenceNumber,
                         sizeof(kPayloadData) + kGenericHeaderLength +
                             kRtpOverheadBytesPerPacket,
                         PacedPacketInfo()))
diff --git a/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc b/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
index bcb075d..cf19573 100644
--- a/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
+++ b/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
@@ -245,7 +245,7 @@
     if (!may_continue) {
       return false;
     }
-    tracker->OnNewTransportFeedbackVector(feedback_vector);
+    tracker->OnPacketFeedbackVector(feedback_vector);
     tracker->Validate();
   }
 
diff --git a/webrtc/test/mock_voe_channel_proxy.h b/webrtc/test/mock_voe_channel_proxy.h
index 9aeb2d1..24adcc2 100644
--- a/webrtc/test/mock_voe_channel_proxy.h
+++ b/webrtc/test/mock_voe_channel_proxy.h
@@ -87,6 +87,7 @@
   MOCK_METHOD1(SetSendCodec, bool(const CodecInst& codec_inst));
   MOCK_METHOD2(SetSendCNPayloadType,
                bool(int type, PayloadFrequencies frequency));
+  MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index fa772d7..c986ff7 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -1070,7 +1070,8 @@
       const LoggedRtpPacket& rtp = *rtp_iterator->second;
       if (rtp.header.extension.hasTransportSequenceNumber) {
         RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
-        cc.AddPacket(rtp.header.extension.transportSequenceNumber,
+        cc.AddPacket(rtp.header.ssrc,
+                     rtp.header.extension.transportSequenceNumber,
                      rtp.total_length, PacedPacketInfo());
         rtc::SentPacket sent_packet(
             rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
@@ -1169,7 +1170,8 @@
       const LoggedRtpPacket& rtp = *rtp_iterator->second;
       if (rtp.header.extension.hasTransportSequenceNumber) {
         RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
-        feedback_adapter.AddPacket(rtp.header.extension.transportSequenceNumber,
+        feedback_adapter.AddPacket(rtp.header.ssrc,
+                                   rtp.header.extension.transportSequenceNumber,
                                    rtp.total_length, PacedPacketInfo());
         feedback_adapter.OnSentPacket(
             rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 0f0831c..7e49e32 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -246,13 +246,14 @@
   }
 
   // Implements TransportFeedbackObserver.
-  void AddPacket(uint16_t sequence_number,
+  void AddPacket(uint32_t ssrc,
+                 uint16_t sequence_number,
                  size_t length,
                  const PacedPacketInfo& pacing_info) override {
     RTC_DCHECK(pacer_thread_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
     if (feedback_observer_)
-      feedback_observer_->AddPacket(sequence_number, length, pacing_info);
+      feedback_observer_->AddPacket(ssrc, sequence_number, length, pacing_info);
   }
 
   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
@@ -395,7 +396,7 @@
           (fraction_lost_aggregate + total_number_of_packets / 2) /
           total_number_of_packets;
     }
-    owner_->OnIncomingFractionLoss(weighted_fraction_lost);
+    owner_->OnUplinkPacketLossRate(weighted_fraction_lost / 255.0f);
   }
 
  private:
@@ -902,7 +903,9 @@
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
       retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
                                                    kMaxRetransmissionWindowMs)),
-      decoder_factory_(config.acm_config.decoder_factory) {
+      decoder_factory_(config.acm_config.decoder_factory),
+      // TODO(elad.alon): Subsequent CL experiments with PLR source.
+      use_twcc_plr_for_ana_(false) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config(config.acm_config);
@@ -1301,10 +1304,23 @@
   retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
 }
 
-void Channel::OnIncomingFractionLoss(int fraction_lost) {
+void Channel::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
+  if (!use_twcc_plr_for_ana_)
+    return;
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
-    if (*encoder)
-      (*encoder)->OnReceivedUplinkPacketLossFraction(fraction_lost / 255.0f);
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
+    }
+  });
+}
+
+void Channel::OnUplinkPacketLossRate(float packet_loss_rate) {
+  if (use_twcc_plr_for_ana_)
+    return;
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
+    }
   });
 }
 
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 68d022d..d24eb5f 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -378,10 +378,15 @@
   // From OverheadObserver in the RTP/RTCP module
   void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
 
- protected:
-  void OnIncomingFractionLoss(int fraction_lost);
+  // The existence of this function alongside OnUplinkPacketLossRate is
+  // a compromise. We want the encoder to be agnostic of the PLR source, but
+  // we also don't want it to receive conflicting information from TWCC and
+  // from RTCP-XR.
+  void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
 
  private:
+  void OnUplinkPacketLossRate(float packet_loss_rate);
+
   bool InputMute() const;
   bool OnRtpPacketWithHeader(const uint8_t* received_packet,
                              size_t length,
@@ -508,6 +513,8 @@
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 
   rtc::ThreadChecker construction_thread_;
+
+  const bool use_twcc_plr_for_ana_;
 };
 
 }  // namespace voe
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 7f6e96c..5388b8d 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -367,6 +367,11 @@
   return channel()->SetSendCNPayloadType(type, frequency) == 0;
 }
 
+void ChannelProxy::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  channel()->OnTwccBasedUplinkPacketLossRate(packet_loss_rate);
+}
+
 Channel* ChannelProxy::channel() const {
   RTC_DCHECK(channel_owner_.channel());
   return channel_owner_.channel();
diff --git a/webrtc/voice_engine/channel_proxy.h b/webrtc/voice_engine/channel_proxy.h
index 28d0364..fd25378 100644
--- a/webrtc/voice_engine/channel_proxy.h
+++ b/webrtc/voice_engine/channel_proxy.h
@@ -116,6 +116,7 @@
   virtual bool SetOpusMaxPlaybackRate(int frequency_hz);
   virtual bool SetSendCodec(const CodecInst& codec_inst);
   virtual bool SetSendCNPayloadType(int type, PayloadFrequencies frequency);
+  virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
 
  private:
   Channel* channel() const;
diff --git a/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc b/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc
index 6cdeec2..07af861 100644
--- a/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc
+++ b/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc
@@ -98,7 +98,7 @@
   }
 }
 
-void TransportFeedbackPacketLossTracker::OnNewTransportFeedbackVector(
+void TransportFeedbackPacketLossTracker::OnPacketFeedbackVector(
     const std::vector<PacketFeedback>& packet_feedback_vector) {
   for (const PacketFeedback& packet : packet_feedback_vector) {
     const auto& it = packet_status_window_.find(packet.sequence_number);
diff --git a/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h b/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
index 231fb70..ea82f74 100644
--- a/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
+++ b/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
@@ -38,7 +38,7 @@
 
   void OnPacketAdded(uint16_t seq_num, int64_t send_time_ms);
 
-  void OnNewTransportFeedbackVector(
+  void OnPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedbacks_vector);
 
   // Returns the packet loss rate, if the window has enough packet statuses to
diff --git a/webrtc/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc b/webrtc/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc
index 1e463dd..9f7d578 100644
--- a/webrtc/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc
+++ b/webrtc/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc
@@ -86,7 +86,7 @@
       ++seq_num;
     }
 
-    tracker->OnNewTransportFeedbackVector(packet_feedback_vector);
+    tracker->OnPacketFeedbackVector(packet_feedback_vector);
     tracker->Validate();
   }