diff --git a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
index 85a287e126..bee10caa7d 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_common_defs.h
@@ -76,6 +76,9 @@ struct WebRtcACMAudioBuff {
   uint32_t last_in_timestamp;
 };
 
+#define HISTOGRAM_NAME_AUDIO_TARGET_BITRATE_IN_KBPS \
+  "WebRTC.Audio.TargetBitrateInKbps"
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 5536ea4264..c01466f8c0 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -23,6 +23,7 @@
 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 #include "webrtc/typedefs.h"
@@ -293,9 +294,11 @@ int AudioCodingModuleImpl::SendBitrate() const {
 
 void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
   CriticalSectionScoped lock(acm_crit_sect_);
-
   if (codec_manager_.CurrentEncoder()) {
     codec_manager_.CurrentEncoder()->SetTargetBitrate(bitrate_bps);
+    RTC_HISTOGRAM_COUNTS_100(
+        HISTOGRAM_NAME_AUDIO_TARGET_BITRATE_IN_KBPS,
+        codec_manager_.CurrentEncoder()->GetTargetBitrate() / 1000);
   }
 }
 
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index 183d526ffc..568ae1ebe7 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -1564,6 +1564,10 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
       .Times(AtLeast(1))
       .WillRepeatedly(
           Invoke(&encoder, &AudioEncoderMutablePcmU::EncodeInternal));
+  EXPECT_CALL(mock_encoder, GetTargetBitrate())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke(
+          &encoder, &AudioEncoderMutablePcmU::GetTargetBitrate));
   ASSERT_NO_FATAL_FAILURE(
       SetUpTestExternalEncoder(&mock_encoder, codec_inst.pltype));
   Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc b/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
index cad6ee9089..d28dcc6ff5 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_manager.cc
@@ -13,6 +13,8 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
@@ -312,6 +314,9 @@ int CodecManager::RegisterEncoder(const CodecInst& send_codec) {
   // Check if a change in Rate is required.
   if (send_codec.rate != send_codec_inst_.rate) {
     codec_owner_.SpeechEncoder()->SetTargetBitrate(send_codec.rate);
+    RTC_HISTOGRAM_COUNTS_100(
+        HISTOGRAM_NAME_AUDIO_TARGET_BITRATE_IN_KBPS,
+        codec_owner_.SpeechEncoder()->GetTargetBitrate() / 1000);
     send_codec_inst_.rate = send_codec.rate;
   }
 
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc b/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
index 4d214be242..5c1e37b5b0 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_owner.cc
@@ -21,6 +21,8 @@
 #include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/audio_encoder_pcm16b.h"
 #include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
 
 namespace webrtc {
 namespace acm2 {
@@ -180,6 +182,8 @@ void CodecOwner::SetEncoders(const CodecInst& speech_inst,
                       &isac_is_encoder_);
   external_speech_encoder_ = nullptr;
   ChangeCngAndRed(cng_payload_type, vad_mode, red_payload_type);
+  RTC_HISTOGRAM_COUNTS_100(HISTOGRAM_NAME_AUDIO_TARGET_BITRATE_IN_KBPS,
+                           SpeechEncoder()->GetTargetBitrate() / 1000);
 }
 
 void CodecOwner::SetEncoders(AudioEncoderMutable* external_speech_encoder,
@@ -190,6 +194,8 @@ void CodecOwner::SetEncoders(AudioEncoderMutable* external_speech_encoder,
   speech_encoder_.reset();
   isac_is_encoder_ = false;
   ChangeCngAndRed(cng_payload_type, vad_mode, red_payload_type);
+  RTC_HISTOGRAM_COUNTS_100(HISTOGRAM_NAME_AUDIO_TARGET_BITRATE_IN_KBPS,
+                           SpeechEncoder()->GetTargetBitrate() / 1000);
 }
 
 void CodecOwner::ChangeCngAndRed(int cng_payload_type,
diff --git a/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc b/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc
index a1366a9b88..b9d4cdd7d1 100644
--- a/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/codec_owner_unittest.cc
@@ -100,6 +100,8 @@ TEST_F(CodecOwnerTest, VerifyCngFrames) {
 
 TEST_F(CodecOwnerTest, ExternalEncoder) {
   MockAudioEncoderMutable external_encoder;
+  EXPECT_CALL(external_encoder, GetTargetBitrate())
+      .WillRepeatedly(Return(-1));  // Flag adaptive bitrate (doesn't matter).
   codec_owner_.SetEncoders(&external_encoder, -1, VADNormal, -1);
   const int kSampleRateHz = 8000;
   const int kPacketSizeSamples = kSampleRateHz / 100;