diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index c2b0d1a878..f60f9208ae 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -89,7 +89,7 @@ AudioSendStream::AudioSendStream(
     rtc::TaskQueue* worker_queue,
     ProcessThread* module_process_thread,
     RtpTransportControllerSendInterface* transport,
-    BitrateAllocator* bitrate_allocator,
+    BitrateAllocatorInterface* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats,
     const absl::optional<RtpState>& suspended_rtp_state,
@@ -115,7 +115,7 @@ AudioSendStream::AudioSendStream(
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     rtc::TaskQueue* worker_queue,
     RtpTransportControllerSendInterface* transport,
-    BitrateAllocator* bitrate_allocator,
+    BitrateAllocatorInterface* bitrate_allocator,
     RtcEventLog* event_log,
     RtcpRttStats* rtcp_rtt_stats,
     const absl::optional<RtpState>& suspended_rtp_state,
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index 1ea676b65d..46c68b3c75 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -46,7 +46,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                   rtc::TaskQueue* worker_queue,
                   ProcessThread* module_process_thread,
                   RtpTransportControllerSendInterface* transport,
-                  BitrateAllocator* bitrate_allocator,
+                  BitrateAllocatorInterface* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats,
                   const absl::optional<RtpState>& suspended_rtp_state,
@@ -56,7 +56,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
                   RtpTransportControllerSendInterface* transport,
-                  BitrateAllocator* bitrate_allocator,
+                  BitrateAllocatorInterface* bitrate_allocator,
                   RtcEventLog* event_log,
                   RtcpRttStats* rtcp_rtt_stats,
                   const absl::optional<RtpState>& suspended_rtp_state,
@@ -140,7 +140,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   size_t encoder_num_channels_ = 0;
   bool sending_ = false;
 
-  BitrateAllocator* const bitrate_allocator_;
+  BitrateAllocatorInterface* const bitrate_allocator_;
   RtpTransportControllerSendInterface* const transport_;
 
   rtc::CriticalSection packet_loss_tracker_cs_;
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index 054d27e421..799931235c 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -62,7 +62,7 @@ VideoSendStream::VideoSendStream(
     rtc::TaskQueue* worker_queue,
     CallStats* call_stats,
     RtpTransportControllerSendInterface* transport,
-    BitrateAllocator* bitrate_allocator,
+    BitrateAllocatorInterface* bitrate_allocator,
     SendDelayStats* send_delay_stats,
     RtcEventLog* event_log,
     VideoSendStream::Config config,
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index 1ff64dc777..22aa4b1c6d 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -59,7 +59,7 @@ class VideoSendStream : public webrtc::VideoSendStream {
       rtc::TaskQueue* worker_queue,
       CallStats* call_stats,
       RtpTransportControllerSendInterface* transport,
-      BitrateAllocator* bitrate_allocator,
+      BitrateAllocatorInterface* bitrate_allocator,
       SendDelayStats* send_delay_stats,
       RtcEventLog* event_log,
       VideoSendStream::Config config,