diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 7742538ca7..3795d79f5e 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -529,7 +529,7 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
   RTCStatsMember<uint64_t> bytes_sent;
   RTCStatsMember<uint64_t> header_bytes_sent;
   RTCStatsMember<uint64_t> retransmitted_bytes_sent;
-  // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
+  // TODO(https://crbug.com/webrtc/13394): Also collect this metric for video.
   RTCStatsMember<double> target_bitrate;
   RTCStatsMember<uint32_t> frames_encoded;
   RTCStatsMember<uint32_t> key_frames_encoded;
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7d70592f76..08bb4e6bbf 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -427,7 +427,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   webrtc::AudioSendStream::Stats stats;
   stats.local_ssrc = config_.rtp.ssrc;
-  stats.target_bitrate_bps = channel_send_->GetBitrate();
+  stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
 
   webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
   stats.payload_bytes_sent = call_stats.payload_bytes_sent;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 3c5b1423b2..9228611e31 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -300,7 +300,7 @@ struct ConfigHelper {
         .WillRepeatedly(Return(report_blocks));
     EXPECT_CALL(*channel_send_, GetANAStatistics())
         .WillRepeatedly(Return(ANAStats()));
-    EXPECT_CALL(*channel_send_, GetBitrate()).WillRepeatedly(Return(0));
+    EXPECT_CALL(*channel_send_, GetTargetBitrate()).WillRepeatedly(Return(0));
 
     audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
     audio_processing_stats_.echo_return_loss_enhancement =
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 6a65655d70..2ad031e543 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -96,7 +96,7 @@ class ChannelSend : public ChannelSendInterface,
 
   // Codecs
   void OnBitrateAllocation(BitrateAllocationUpdate update) override;
-  int GetBitrate() const override;
+  int GetTargetBitrate() const override;
 
   // Network
   void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
@@ -238,9 +238,6 @@ class ChannelSend : public ChannelSendInterface,
   rtc::scoped_refptr<ChannelSendFrameTransformerDelegate>
       frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_);
 
-  mutable Mutex bitrate_mutex_;
-  int configured_bitrate_bps_ RTC_GUARDED_BY(bitrate_mutex_) = 0;
-
   // Defined last to ensure that there are no running tasks when the other
   // members are destroyed.
   rtc::TaskQueue encoder_queue_;
@@ -616,18 +613,14 @@ void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
   // rules.
   // RTC_DCHECK(worker_thread_checker_.IsCurrent() ||
   //            module_process_thread_checker_.IsCurrent());
-  MutexLock lock(&bitrate_mutex_);
-
   CallEncoder([&](AudioEncoder* encoder) {
     encoder->OnReceivedUplinkAllocation(update);
   });
   retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
-  configured_bitrate_bps_ = update.target_bitrate.bps();
 }
 
-int ChannelSend::GetBitrate() const {
-  MutexLock lock(&bitrate_mutex_);
-  return configured_bitrate_bps_;
+int ChannelSend::GetTargetBitrate() const {
+  return audio_coding_->GetTargetBitrate();
 }
 
 void ChannelSend::OnUplinkPacketLossRate(float packet_loss_rate) {
diff --git a/audio/channel_send.h b/audio/channel_send.h
index 663b947036..e100725460 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -91,7 +91,7 @@ class ChannelSendInterface {
                                                 int payload_frequency) = 0;
   virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
   virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
-  virtual int GetBitrate() const = 0;
+  virtual int GetTargetBitrate() const = 0;
   virtual void SetInputMute(bool muted) = 0;
 
   virtual void ProcessAndEncodeAudio(
diff --git a/audio/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
index d445b51312..a02bee38ad 100644
--- a/audio/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -166,7 +166,7 @@ class MockChannelSend : public voe::ChannelSendInterface {
               (std::unique_ptr<AudioFrame>),
               (override));
   MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override));
-  MOCK_METHOD(int, GetBitrate, (), (const, override));
+  MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
   MOCK_METHOD(int64_t, GetRTT, (), (const, override));
   MOCK_METHOD(void, StartSend, (), (override));
   MOCK_METHOD(void, StopSend, (), (override));
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 2607a7a4aa..519ca5dd6d 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -375,6 +375,8 @@ struct MediaSenderInfo {
   uint64_t retransmitted_packets_sent = 0;
   // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-nackcount
   uint32_t nacks_rcvd = 0;
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-targetbitrate
+  double target_bitrate = 0.0;
   int packets_lost = 0;
   float fraction_lost = 0.0f;
   int64_t rtt_ms = 0;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 5d4455c2de..6934a0ca54 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2369,6 +2369,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info,
     sinfo.packets_lost = stats.packets_lost;
     sinfo.fraction_lost = stats.fraction_lost;
     sinfo.nacks_rcvd = stats.nacks_rcvd;
+    sinfo.target_bitrate = stats.target_bitrate_bps;
     sinfo.codec_name = stats.codec_name;
     sinfo.codec_payload_type = stats.codec_payload_type;
     sinfo.jitter_ms = stats.jitter_ms;
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 8ba1b9f3de..b742a824a9 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -92,6 +92,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
 
   ANAStats GetANAStats() const override;
 
+  int GetTargetBitrate() const override;
+
  private:
   struct InputData {
     InputData() : buffer(kInitialInputDataBufferSize) {}
@@ -603,6 +605,14 @@ ANAStats AudioCodingModuleImpl::GetANAStats() const {
   return ANAStats();
 }
 
+int AudioCodingModuleImpl::GetTargetBitrate() const {
+  MutexLock lock(&acm_mutex_);
+  if (!encoder_stack_) {
+    return -1;
+  }
+  return encoder_stack_->GetTargetBitrate();
+}
+
 }  // namespace
 
 AudioCodingModule::Config::Config(
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index 7551814023..9b1dce8e74 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -237,6 +237,8 @@ class AudioCodingModule {
       NetworkStatistics* network_statistics) = 0;
 
   virtual ANAStats GetANAStats() const = 0;
+
+  virtual int GetTargetBitrate() const = 0;
 };
 
 }  // namespace webrtc
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 7637c07c09..23e24dc849 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -536,6 +536,9 @@ void SetOutboundRTPStreamStatsFromVoiceSenderInfo(
                                                outbound_audio);
   outbound_audio->media_type = "audio";
   outbound_audio->kind = "audio";
+  if (voice_sender_info.target_bitrate > 0) {
+    outbound_audio->target_bitrate = voice_sender_info.target_bitrate;
+  }
   if (voice_sender_info.codec_payload_type) {
     outbound_audio->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
         mid, /*inbound=*/false, *voice_sender_info.codec_payload_type);
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index b8c321394b..8c3c42f8ad 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -2180,6 +2180,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   voice_media_info.senders[0].header_and_padding_bytes_sent = 12;
   voice_media_info.senders[0].retransmitted_bytes_sent = 30;
   voice_media_info.senders[0].nacks_rcvd = 31;
+  voice_media_info.senders[0].target_bitrate = 32000;
   voice_media_info.senders[0].codec_payload_type = 42;
 
   RtpCodecParameters codec_parameters;
@@ -2214,6 +2215,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
   expected_audio.header_bytes_sent = 12;
   expected_audio.retransmitted_bytes_sent = 30;
   expected_audio.nack_count = 31;
+  expected_audio.target_bitrate = 32000;
 
   ASSERT_TRUE(report->Get(expected_audio.id()));
   EXPECT_EQ(
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 81ead00efe..32c567ee45 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -964,7 +964,6 @@ class RTCStatsReportVerifier {
         outbound_stream.header_bytes_sent);
     verifier.TestMemberIsNonNegative<uint64_t>(
         outbound_stream.retransmitted_bytes_sent);
-    verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
     if (outbound_stream.media_type.is_defined() &&
         *outbound_stream.media_type == "video") {
       verifier.TestMemberIsDefined(outbound_stream.frames_encoded);
@@ -1000,6 +999,7 @@ class RTCStatsReportVerifier {
       verifier.TestMemberIsNonNegative<uint32_t>(outbound_stream.frames_sent);
       verifier.TestMemberIsNonNegative<uint32_t>(
           outbound_stream.huge_frames_sent);
+      verifier.TestMemberIsUndefined(outbound_stream.target_bitrate);
       verifier.MarkMemberTested(outbound_stream.rid, true);
     } else {
       verifier.TestMemberIsUndefined(outbound_stream.frames_encoded);
@@ -1023,6 +1023,7 @@ class RTCStatsReportVerifier {
       verifier.TestMemberIsUndefined(outbound_stream.frame_width);
       verifier.TestMemberIsUndefined(outbound_stream.frames_sent);
       verifier.TestMemberIsUndefined(outbound_stream.huge_frames_sent);
+      verifier.TestMemberIsNonNegative<double>(outbound_stream.target_bitrate);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }