diff --git a/audio/audio_state.h b/audio/audio_state.h
index 023c7b1efd..f4bddbfa85 100644
--- a/audio/audio_state.h
+++ b/audio/audio_state.h
@@ -35,6 +35,9 @@ class AudioState final : public webrtc::AudioState {
     RTC_DCHECK(config_.audio_processing);
     return config_.audio_processing.get();
   }
+  AudioTransport* audio_transport() override {
+    return &audio_transport_proxy_;
+  }
 
   void SetPlayout(bool enabled) override;
   void SetRecording(bool enabled) override;
diff --git a/call/audio_state.h b/call/audio_state.h
index ad411d1faa..e4a281aec2 100644
--- a/call/audio_state.h
+++ b/call/audio_state.h
@@ -17,6 +17,7 @@
 namespace webrtc {
 
 class AudioProcessing;
+class AudioTransport;
 class VoiceEngine;
 
 // WORK IN PROGRESS
@@ -43,6 +44,7 @@ class AudioState : public rtc::RefCountInterface {
   };
 
   virtual AudioProcessing* audio_processing() = 0;
+  virtual AudioTransport* audio_transport() = 0;
 
   // Enable/disable playout of the audio channels. Enabled by default.
   // This will stop playout of the underlying audio device but start a task