diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h
index d3d1541..89356df 100644
--- a/api/frame_transformer_interface.h
+++ b/api/frame_transformer_interface.h
@@ -82,6 +82,11 @@
   // TODO(crbug.com/1456628): Change this to pure virtual after it
   // is implemented everywhere.
   virtual FrameType Type() const { return FrameType::kEmptyFrame; }
+
+  // Audio level in -dBov. Values range from 0 to 127, representing 0 to -127
+  // dBov. 127 represents digital silence. Only present on remote frames if
+  // the audio level header extension was included.
+  virtual absl::optional<uint8_t> AudioLevel() const = 0;
 };
 
 // Objects implement this interface to be notified with the transformed frame.
diff --git a/api/test/mock_transformable_audio_frame.h b/api/test/mock_transformable_audio_frame.h
index 584c77f..f243e38 100644
--- a/api/test/mock_transformable_audio_frame.h
+++ b/api/test/mock_transformable_audio_frame.h
@@ -47,6 +47,7 @@
               Type,
               (),
               (const, override));
+  MOCK_METHOD(absl::optional<uint8_t>, AudioLevel, (), (const, override));
 };
 
 }  // namespace webrtc
diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc
index dbced02..953e27a 100644
--- a/audio/channel_receive_frame_transformer_delegate.cc
+++ b/audio/channel_receive_frame_transformer_delegate.cc
@@ -70,6 +70,13 @@
                : FrameType::kAudioFrameCN;
   }
 
+  absl::optional<uint8_t> AudioLevel() const override {
+    if (header_.extension.audio_level()) {
+      return header_.extension.audio_level()->level();
+    }
+    return absl::nullopt;
+  }
+
  private:
   rtc::Buffer payload_;
   RTPHeader header_;
diff --git a/audio/channel_receive_frame_transformer_delegate_unittest.cc b/audio/channel_receive_frame_transformer_delegate_unittest.cc
index a206a09..8b819f1 100644
--- a/audio/channel_receive_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_receive_frame_transformer_delegate_unittest.cc
@@ -174,5 +174,76 @@
   delegate->Transform(packet, header, /*ssrc=*/1111, /*mimeType=*/"audio/opus");
 }
 
+TEST(ChannelReceiveFrameTransformerDelegateTest,
+     AudioLevelAbsentWithoutExtension) {
+  rtc::AutoThread main_thread;
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          /*receive_frame_callback=*/nullptr, mock_frame_transformer,
+          rtc::Thread::Current());
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  delegate->Init();
+  ASSERT_TRUE(callback);
+
+  const uint8_t data[] = {1, 2, 3, 4};
+  rtc::ArrayView<const uint8_t> packet(data, sizeof(data));
+  RTPHeader header;
+  std::unique_ptr<TransformableFrameInterface> frame;
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault(
+          [&](std::unique_ptr<TransformableFrameInterface> transform_frame) {
+            frame = std::move(transform_frame);
+          });
+  delegate->Transform(packet, header, /*ssrc=*/1111, /*mimeType=*/"audio/opus");
+
+  EXPECT_TRUE(frame);
+  auto* audio_frame =
+      static_cast<TransformableAudioFrameInterface*>(frame.get());
+  EXPECT_FALSE(audio_frame->AudioLevel());
+  EXPECT_EQ(audio_frame->Type(),
+            TransformableAudioFrameInterface::FrameType::kAudioFrameCN);
+}
+
+TEST(ChannelReceiveFrameTransformerDelegateTest,
+     AudioLevelPresentWithExtension) {
+  rtc::AutoThread main_thread;
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          /*receive_frame_callback=*/nullptr, mock_frame_transformer,
+          rtc::Thread::Current());
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  delegate->Init();
+  ASSERT_TRUE(callback);
+
+  const uint8_t data[] = {1, 2, 3, 4};
+  rtc::ArrayView<const uint8_t> packet(data, sizeof(data));
+  RTPHeader header;
+  uint8_t audio_level_dbov = 67;
+  AudioLevel audio_level(/*voice_activity=*/true, audio_level_dbov);
+  header.extension.set_audio_level(audio_level);
+  std::unique_ptr<TransformableFrameInterface> frame;
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault(
+          [&](std::unique_ptr<TransformableFrameInterface> transform_frame) {
+            frame = std::move(transform_frame);
+          });
+  delegate->Transform(packet, header, /*ssrc=*/1111, /*mimeType=*/"audio/opus");
+
+  EXPECT_TRUE(frame);
+  auto* audio_frame =
+      static_cast<TransformableAudioFrameInterface*>(frame.get());
+  EXPECT_EQ(*audio_frame->AudioLevel(), audio_level_dbov);
+  EXPECT_EQ(audio_frame->Type(),
+            TransformableAudioFrameInterface::FrameType::kAudioFrameSpeech);
+}
+
 }  // namespace
 }  // namespace webrtc
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index e8eaa31..1e211ab 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -170,7 +170,8 @@
                        uint32_t rtp_timestamp_without_offset,
                        rtc::ArrayView<const uint8_t> payload,
                        int64_t absolute_capture_timestamp_ms,
-                       rtc::ArrayView<const uint32_t> csrcs)
+                       rtc::ArrayView<const uint32_t> csrcs,
+                       absl::optional<uint8_t> audio_level_dbov)
       RTC_RUN_ON(encoder_queue_checker_);
 
   void OnReceivedRtt(int64_t rtt_ms);
@@ -280,6 +281,14 @@
                               int64_t absolute_capture_timestamp_ms) {
   RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
   rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
+
+  absl::optional<uint8_t> audio_level_dbov;
+  if (include_audio_level_indication_.load()) {
+    // Take the averaged audio levels from rms_level_ and reset it before
+    // invoking any async transformer.
+    audio_level_dbov = rms_level_.Average();
+  }
+
   if (frame_transformer_delegate_) {
     // Asynchronously transform the payload before sending it. After the payload
     // is transformed, the delegate will call SendRtpAudio to send it.
@@ -290,11 +299,12 @@
     frame_transformer_delegate_->Transform(
         frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
         payloadData, payloadSize, absolute_capture_timestamp_ms,
-        rtp_rtcp_->SSRC(), mime_type.str());
+        rtp_rtcp_->SSRC(), mime_type.str(), audio_level_dbov);
     return 0;
   }
   return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
-                      absolute_capture_timestamp_ms, /*csrcs=*/{});
+                      absolute_capture_timestamp_ms, /*csrcs=*/{},
+                      audio_level_dbov);
 }
 
 int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
@@ -302,7 +312,8 @@
                                   uint32_t rtp_timestamp_without_offset,
                                   rtc::ArrayView<const uint8_t> payload,
                                   int64_t absolute_capture_timestamp_ms,
-                                  rtc::ArrayView<const uint32_t> csrcs) {
+                                  rtc::ArrayView<const uint32_t> csrcs,
+                                  absl::optional<uint8_t> audio_level_dbov) {
   // E2EE Custom Audio Frame Encryption (This is optional).
   // Keep this buffer around for the lifetime of the send call.
   rtc::Buffer encrypted_audio_payload;
@@ -369,8 +380,8 @@
   if (absolute_capture_timestamp_ms > 0) {
     frame.capture_time = Timestamp::Millis(absolute_capture_timestamp_ms);
   }
-  if (include_audio_level_indication_.load()) {
-    frame.audio_level_dbov = rms_level_.Average();
+  if (include_audio_level_indication_.load() && audio_level_dbov) {
+    frame.audio_level_dbov = *audio_level_dbov;
   }
   if (!rtp_sender_audio_->SendAudio(frame)) {
     RTC_DLOG(LS_ERROR)
@@ -866,12 +877,13 @@
              uint32_t rtp_timestamp_with_offset,
              rtc::ArrayView<const uint8_t> payload,
              int64_t absolute_capture_timestamp_ms,
-             rtc::ArrayView<const uint32_t> csrcs) {
+             rtc::ArrayView<const uint32_t> csrcs,
+             absl::optional<uint8_t> audio_level_dbov) {
         RTC_DCHECK_RUN_ON(&encoder_queue_checker_);
         return SendRtpAudio(
             frameType, payloadType,
             rtp_timestamp_with_offset - rtp_rtcp_->StartTimestamp(), payload,
-            absolute_capture_timestamp_ms, csrcs);
+            absolute_capture_timestamp_ms, csrcs, audio_level_dbov);
       };
   frame_transformer_delegate_ =
       rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
diff --git a/audio/channel_send_frame_transformer_delegate.cc b/audio/channel_send_frame_transformer_delegate.cc
index 6d3c011..8bf1963 100644
--- a/audio/channel_send_frame_transformer_delegate.cc
+++ b/audio/channel_send_frame_transformer_delegate.cc
@@ -59,7 +59,8 @@
       uint32_t ssrc,
       std::vector<uint32_t> csrcs,
       const std::string& codec_mime_type,
-      absl::optional<uint16_t> sequence_number)
+      absl::optional<uint16_t> sequence_number,
+      absl::optional<uint8_t> audio_level_dbov)
       : frame_type_(frame_type),
         payload_type_(payload_type),
         rtp_timestamp_with_offset_(rtp_timestamp_with_offset),
@@ -68,7 +69,8 @@
         ssrc_(ssrc),
         csrcs_(std::move(csrcs)),
         codec_mime_type_(codec_mime_type),
-        sequence_number_(sequence_number) {}
+        sequence_number_(sequence_number),
+        audio_level_dbov_(audio_level_dbov) {}
   ~TransformableOutgoingAudioFrame() override = default;
   rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
   void SetData(rtc::ArrayView<const uint8_t> data) override {
@@ -101,6 +103,10 @@
     return absolute_capture_timestamp_ms_;
   }
 
+  absl::optional<uint8_t> AudioLevel() const override {
+    return audio_level_dbov_;
+  }
+
  private:
   AudioFrameType frame_type_;
   uint8_t payload_type_;
@@ -111,6 +117,7 @@
   std::vector<uint32_t> csrcs_;
   std::string codec_mime_type_;
   absl::optional<uint16_t> sequence_number_;
+  absl::optional<uint8_t> audio_level_dbov_;
 };
 }  // namespace
 
@@ -143,14 +150,15 @@
     size_t payload_size,
     int64_t absolute_capture_timestamp_ms,
     uint32_t ssrc,
-    const std::string& codec_mimetype) {
+    const std::string& codec_mimetype,
+    absl::optional<uint8_t> audio_level_dbov) {
   {
     MutexLock lock(&send_lock_);
     if (short_circuit_) {
       send_frame_callback_(
           frame_type, payload_type, rtp_timestamp,
           rtc::ArrayView<const uint8_t>(payload_data, payload_size),
-          absolute_capture_timestamp_ms, /*csrcs=*/{});
+          absolute_capture_timestamp_ms, /*csrcs=*/{}, audio_level_dbov);
       return;
     }
   }
@@ -159,7 +167,7 @@
           frame_type, payload_type, rtp_timestamp, payload_data, payload_size,
           absolute_capture_timestamp_ms, ssrc,
           /*csrcs=*/std::vector<uint32_t>(), codec_mimetype,
-          /*sequence_number=*/absl::nullopt));
+          /*sequence_number=*/absl::nullopt, audio_level_dbov));
 }
 
 void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
@@ -194,7 +202,8 @@
       transformed_frame->AbsoluteCaptureTimestamp()
           ? *transformed_frame->AbsoluteCaptureTimestamp()
           : 0,
-      transformed_frame->GetContributingSources());
+      transformed_frame->GetContributingSources(),
+      transformed_frame->AudioLevel());
 }
 
 std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
@@ -207,7 +216,8 @@
       original->GetPayloadType(), original->GetTimestamp(),
       original->GetData().data(), original->GetData().size(),
       original->AbsoluteCaptureTimestamp(), original->GetSsrc(),
-      std::move(csrcs), original->GetMimeType(), original->SequenceNumber());
+      std::move(csrcs), original->GetMimeType(), original->SequenceNumber(),
+      original->AudioLevel());
 }
 
 }  // namespace webrtc
diff --git a/audio/channel_send_frame_transformer_delegate.h b/audio/channel_send_frame_transformer_delegate.h
index 30e63ff..5573052 100644
--- a/audio/channel_send_frame_transformer_delegate.h
+++ b/audio/channel_send_frame_transformer_delegate.h
@@ -36,7 +36,8 @@
                             uint32_t rtp_timestamp_with_offset,
                             rtc::ArrayView<const uint8_t> payload,
                             int64_t absolute_capture_timestamp_ms,
-                            rtc::ArrayView<const uint32_t> csrcs)>;
+                            rtc::ArrayView<const uint32_t> csrcs,
+                            absl::optional<uint8_t> audio_level_dbov)>;
   ChannelSendFrameTransformerDelegate(
       SendFrameCallback send_frame_callback,
       rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
@@ -60,7 +61,8 @@
                  size_t payload_size,
                  int64_t absolute_capture_timestamp_ms,
                  uint32_t ssrc,
-                 const std::string& codec_mime_type);
+                 const std::string& codec_mime_type,
+                 absl::optional<uint8_t> audio_level_dbov);
 
   // Implements TransformedFrameCallback. Can be called on any thread.
   void OnTransformedFrame(
diff --git a/audio/channel_send_frame_transformer_delegate_unittest.cc b/audio/channel_send_frame_transformer_delegate_unittest.cc
index 5c025bb..e8b7aef 100644
--- a/audio/channel_send_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_send_frame_transformer_delegate_unittest.cc
@@ -28,6 +28,7 @@
 using ::testing::ElementsAre;
 using ::testing::ElementsAreArray;
 using ::testing::NiceMock;
+using ::testing::Optional;
 using ::testing::Return;
 using ::testing::SaveArg;
 
@@ -45,21 +46,24 @@
                uint32_t rtp_timestamp,
                rtc::ArrayView<const uint8_t> payload,
                int64_t absolute_capture_timestamp_ms,
-               rtc::ArrayView<const uint32_t> csrcs));
+               rtc::ArrayView<const uint32_t> csrcs,
+               absl::optional<uint8_t> audio_level_dbov));
 
   ChannelSendFrameTransformerDelegate::SendFrameCallback callback() {
     return [this](AudioFrameType frameType, uint8_t payloadType,
                   uint32_t rtp_timestamp, rtc::ArrayView<const uint8_t> payload,
                   int64_t absolute_capture_timestamp_ms,
-                  rtc::ArrayView<const uint32_t> csrcs) {
+                  rtc::ArrayView<const uint32_t> csrcs,
+                  absl::optional<uint8_t> audio_level_dbov) {
       return SendFrame(frameType, payloadType, rtp_timestamp, payload,
-                       absolute_capture_timestamp_ms, csrcs);
+                       absolute_capture_timestamp_ms, csrcs, audio_level_dbov);
     };
   }
 };
 
 std::unique_ptr<TransformableAudioFrameInterface> CreateMockReceiverFrame(
-    const std::vector<uint32_t>& csrcs) {
+    const std::vector<uint32_t>& csrcs,
+    absl::optional<uint8_t> audio_level_dbov) {
   std::unique_ptr<MockTransformableAudioFrame> mock_frame =
       std::make_unique<NiceMock<MockTransformableAudioFrame>>();
   rtc::ArrayView<const uint8_t> payload(mock_data);
@@ -69,6 +73,7 @@
       .WillByDefault(Return(TransformableFrameInterface::Direction::kReceiver));
   ON_CALL(*mock_frame, GetContributingSources).WillByDefault(Return(csrcs));
   ON_CALL(*mock_frame, SequenceNumber).WillByDefault(Return(987654321));
+  ON_CALL(*mock_frame, AudioLevel).WillByDefault(Return(audio_level_dbov));
   return mock_frame;
 }
 
@@ -88,9 +93,9 @@
               std::unique_ptr<TransformableFrameInterface> transform_frame) {
             frame = std::move(transform_frame);
           });
-  delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, mock_data,
-                      sizeof(mock_data), 0,
-                      /*ssrc=*/0, /*mimeType=*/"audio/opus");
+  delegate->Transform(
+      AudioFrameType::kEmptyFrame, 0, 0, mock_data, sizeof(mock_data), 0,
+      /*ssrc=*/0, /*mimeType=*/"audio/opus", /*audio_level_dbov=*/123);
   return absl::WrapUnique(
       static_cast<webrtc::TransformableAudioFrameInterface*>(frame.release()));
 }
@@ -147,7 +152,8 @@
             callback->OnTransformedFrame(std::move(frame));
           });
   delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, data, sizeof(data), 0,
-                      /*ssrc=*/0, /*mimeType=*/"audio/opus");
+                      /*ssrc=*/0, /*mimeType=*/"audio/opus",
+                      /*audio_level_dbov=*/31);
   channel_queue.WaitForPreviouslyPostedTasks();
 }
 
@@ -169,16 +175,20 @@
   ASSERT_TRUE(callback);
 
   const std::vector<uint32_t> csrcs = {123, 234, 345, 456};
+  const uint8_t audio_level_dbov = 17;
   EXPECT_CALL(mock_channel, SendFrame).Times(0);
-  EXPECT_CALL(mock_channel, SendFrame(_, 0, 0, ElementsAreArray(mock_data), _,
-                                      ElementsAreArray(csrcs)));
+  EXPECT_CALL(mock_channel,
+              SendFrame(_, 0, 0, ElementsAreArray(mock_data), _,
+                        ElementsAreArray(csrcs), Optional(audio_level_dbov)));
   ON_CALL(*mock_frame_transformer, Transform)
       .WillByDefault([&](std::unique_ptr<TransformableFrameInterface> frame) {
-        callback->OnTransformedFrame(CreateMockReceiverFrame(csrcs));
+        callback->OnTransformedFrame(CreateMockReceiverFrame(
+            csrcs, absl::optional<uint8_t>(audio_level_dbov)));
       });
   delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, mock_data,
                       sizeof(mock_data), 0,
-                      /*ssrc=*/0, /*mimeType=*/"audio/opus");
+                      /*ssrc=*/0, /*mimeType=*/"audio/opus",
+                      /*audio_level_dbov=*/absl::nullopt);
   channel_queue.WaitForPreviouslyPostedTasks();
 }
 
@@ -218,7 +228,8 @@
   EXPECT_CALL(mock_channel, SendFrame);
   const uint8_t data[] = {1, 2, 3, 4};
   delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, data, sizeof(data), 0,
-                      /*ssrc=*/0, /*mimeType=*/"audio/opus");
+                      /*ssrc=*/0, /*mimeType=*/"audio/opus",
+                      /*audio_level_dbov=*/absl::nullopt);
 }
 
 TEST(ChannelSendFrameTransformerDelegateTest,
@@ -234,11 +245,13 @@
   EXPECT_EQ(cloned_frame->GetMimeType(), frame->GetMimeType());
   EXPECT_THAT(cloned_frame->GetContributingSources(),
               ElementsAreArray(frame->GetContributingSources()));
+  EXPECT_EQ(cloned_frame->AudioLevel(), frame->AudioLevel());
 }
 
 TEST(ChannelSendFrameTransformerDelegateTest, CloningReceiverFrameWithCsrcs) {
   std::unique_ptr<TransformableAudioFrameInterface> frame =
-      CreateMockReceiverFrame(/*csrcs=*/{123, 234, 345});
+      CreateMockReceiverFrame(/*csrcs=*/{123, 234, 345},
+                              absl::optional<uint8_t>(72));
   std::unique_ptr<TransformableAudioFrameInterface> cloned_frame =
       CloneSenderAudioFrame(frame.get());
 
@@ -254,6 +267,7 @@
   EXPECT_THAT(cloned_frame->GetContributingSources(),
               ElementsAreArray(frame->GetContributingSources()));
   EXPECT_EQ(cloned_frame->SequenceNumber(), frame->SequenceNumber());
+  EXPECT_EQ(cloned_frame->AudioLevel(), frame->AudioLevel());
 }
 
 }  // namespace
diff --git a/audio/channel_send_unittest.cc b/audio/channel_send_unittest.cc
index 77d8479..523408e 100644
--- a/audio/channel_send_unittest.cc
+++ b/audio/channel_send_unittest.cc
@@ -18,6 +18,7 @@
 #include "api/environment/environment_factory.h"
 #include "api/scoped_refptr.h"
 #include "api/test/mock_frame_transformer.h"
+#include "api/test/mock_transformable_audio_frame.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "call/rtp_transport_controller_send.h"
@@ -76,22 +77,29 @@
     ON_CALL(transport_, SendRtp).WillByDefault(Return(true));
   }
 
-  std::unique_ptr<AudioFrame> CreateAudioFrame() {
+  std::unique_ptr<AudioFrame> CreateAudioFrame(uint8_t data_init_value = 0) {
     auto frame = std::make_unique<AudioFrame>();
     frame->sample_rate_hz_ = kSampleRateHz;
     frame->samples_per_channel_ = kSampleRateHz / 100;
     frame->num_channels_ = 1;
     frame->set_absolute_capture_timestamp_ms(
         time_controller_.GetClock()->TimeInMilliseconds());
+    int16_t* dest = frame->mutable_data();
+    for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+         i++, dest++) {
+      *dest = data_init_value;
+    }
     return frame;
   }
 
-  void ProcessNextFrame() {
-    channel_->ProcessAndEncodeAudio(CreateAudioFrame());
+  void ProcessNextFrame(std::unique_ptr<AudioFrame> audio_frame) {
+    channel_->ProcessAndEncodeAudio(std::move(audio_frame));
     // Advance time to process the task queue.
     time_controller_.AdvanceTime(TimeDelta::Millis(10));
   }
 
+  void ProcessNextFrame() { ProcessNextFrame(CreateAudioFrame()); }
+
   GlobalSimulatedTimeController time_controller_;
   webrtc::test::ScopedKeyValueConfig field_trials_;
   Environment env_;
@@ -189,6 +197,117 @@
   EXPECT_TRUE_WAIT(sent_timestamp, 1000);
   EXPECT_EQ(*sent_timestamp, transformable_frame_timestamp);
 }
+
+// Ensure that AudioLevel calculations are performed correctly per-packet even
+// if there's an async Encoded Frame Transform happening.
+TEST_F(ChannelSendTest, AudioLevelsAttachedToCorrectTransformedFrame) {
+  channel_->SetSendAudioLevelIndicationStatus(true, /*id=*/1);
+  RtpPacketReceived::ExtensionManager extension_manager;
+  extension_manager.RegisterByType(1, kRtpExtensionAudioLevel);
+
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<MockFrameTransformer>();
+  channel_->SetEncoderToPacketizerFrameTransformer(mock_frame_transformer);
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
+
+  std::vector<uint8_t> sent_audio_levels;
+  auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
+                      const PacketOptions& options) {
+    RtpPacketReceived packet(&extension_manager);
+    packet.Parse(data);
+    RTPHeader header;
+    packet.GetHeader(&header);
+    sent_audio_levels.push_back(header.extension.audio_level()->level());
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));
+
+  channel_->StartSend();
+  std::vector<std::unique_ptr<TransformableFrameInterface>> frames;
+  EXPECT_CALL(*mock_frame_transformer, Transform)
+      .Times(2)
+      .WillRepeatedly([&](std::unique_ptr<TransformableFrameInterface> frame) {
+        frames.push_back(std::move(frame));
+      });
+
+  // Insert two frames of 7s which should trigger a new packet.
+  ProcessNextFrame(CreateAudioFrame(/*data_init_value=*/7));
+  ProcessNextFrame(CreateAudioFrame(/*data_init_value=*/7));
+
+  // Insert two more frames of 3s, meaning a second packet is
+  // prepared and sent to the transform before the first packet has
+  // been sent.
+  ProcessNextFrame(CreateAudioFrame(/*data_init_value=*/3));
+  ProcessNextFrame(CreateAudioFrame(/*data_init_value=*/3));
+
+  // Wait for both packets to be encoded and sent to the transform.
+  EXPECT_EQ_WAIT(frames.size(), 2ul, 1000);
+  // Complete the transforms on both frames at the same time
+  callback->OnTransformedFrame(std::move(frames[0]));
+  callback->OnTransformedFrame(std::move(frames[1]));
+
+  // Allow things posted back to the encoder queue to run.
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+
+  // Ensure the audio levels on both sent packets is present and
+  // matches their contents.
+  EXPECT_EQ_WAIT(sent_audio_levels.size(), 2ul, 1000);
+  // rms dbov of the packet with raw audio of 7s is 73.
+  EXPECT_EQ(sent_audio_levels[0], 73);
+  // rms dbov of the second packet with raw audio of 3s is 81.
+  EXPECT_EQ(sent_audio_levels[1], 81);
+}
+
+// Ensure that AudioLevels are attached to frames injected into the
+// Encoded Frame transform.
+TEST_F(ChannelSendTest, AudioLevelsAttachedToInsertedTransformedFrame) {
+  channel_->SetSendAudioLevelIndicationStatus(true, /*id=*/1);
+  RtpPacketReceived::ExtensionManager extension_manager;
+  extension_manager.RegisterByType(1, kRtpExtensionAudioLevel);
+
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<MockFrameTransformer>();
+  channel_->SetEncoderToPacketizerFrameTransformer(mock_frame_transformer);
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
+
+  std::optional<uint8_t> sent_audio_level;
+  auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
+                      const PacketOptions& options) {
+    RtpPacketReceived packet(&extension_manager);
+    packet.Parse(data);
+    RTPHeader header;
+    packet.GetHeader(&header);
+    sent_audio_level = header.extension.audio_level()->level();
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));
+
+  channel_->StartSend();
+
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  // Inject a frame encoded elsewhere.
+  auto mock_frame = std::make_unique<NiceMock<MockTransformableAudioFrame>>();
+  uint8_t audio_level = 67;
+  ON_CALL(*mock_frame, AudioLevel()).WillByDefault(Return(audio_level));
+  uint8_t payload[10];
+  ON_CALL(*mock_frame, GetData())
+      .WillByDefault(Return(rtc::ArrayView<uint8_t>(&payload[0], 10)));
+  EXPECT_TRUE_WAIT(callback, 1000);
+  callback->OnTransformedFrame(std::move(mock_frame));
+
+  // Allow things posted back to the encoder queue to run.
+  time_controller_.AdvanceTime(TimeDelta::Millis(10));
+
+  // Ensure the audio levels is set on the sent packet.
+  EXPECT_TRUE_WAIT(sent_audio_level, 1000);
+  EXPECT_EQ(*sent_audio_level, audio_level);
+}
 }  // namespace
 }  // namespace voe
 }  // namespace webrtc