diff --git a/media/base/mediaengine.h b/media/base/mediaengine.h
index 1ddb36c..01300d4 100644
--- a/media/base/mediaengine.h
+++ b/media/base/mediaengine.h
@@ -111,37 +111,10 @@
   // Initialization
   // Starts the engine.
   virtual bool Init() = 0;
-  // TODO(solenberg): Remove once VoE API refactoring is done.
-  virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
-
-  // MediaChannel creation
-  // Creates a voice media channel. Returns NULL on failure.
-  virtual VoiceMediaChannel* CreateChannel(
-      webrtc::Call* call,
-      const MediaConfig& config,
-      const AudioOptions& options,
-      const webrtc::CryptoOptions& crypto_options) = 0;
-  // Creates a video media channel, paired with the specified voice channel.
-  // Returns NULL on failure.
-  virtual VideoMediaChannel* CreateVideoChannel(
-      webrtc::Call* call,
-      const MediaConfig& config,
-      const VideoOptions& options,
-      const webrtc::CryptoOptions& crypto_options) = 0;
-
-  virtual const std::vector<AudioCodec>& audio_send_codecs() = 0;
-  virtual const std::vector<AudioCodec>& audio_recv_codecs() = 0;
-  virtual RtpCapabilities GetAudioCapabilities() = 0;
-  virtual std::vector<VideoCodec> video_codecs() = 0;
-  virtual RtpCapabilities GetVideoCapabilities() = 0;
-
-  // Starts AEC dump using existing file, a maximum file size in bytes can be
-  // specified. Logging is stopped just before the size limit is exceeded.
-  // If max_size_bytes is set to a value <= 0, no limit will be used.
-  virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
-
-  // Stops recording AEC dump.
-  virtual void StopAecDump() = 0;
+  virtual VoiceEngineInterface& voice() = 0;
+  virtual VideoEngineInterface& video() = 0;
+  virtual const VoiceEngineInterface& voice() const = 0;
+  virtual const VideoEngineInterface& video() const = 0;
 };
 
 // CompositeMediaEngine constructs a MediaEngine from separate
@@ -153,34 +126,10 @@
   ~CompositeMediaEngine() override;
   bool Init() override;
 
-  rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
-  VoiceMediaChannel* CreateChannel(
-      webrtc::Call* call,
-      const MediaConfig& config,
-      const AudioOptions& options,
-      const webrtc::CryptoOptions& crypto_options) override;
-
-  VideoMediaChannel* CreateVideoChannel(
-      webrtc::Call* call,
-      const MediaConfig& config,
-      const VideoOptions& options,
-      const webrtc::CryptoOptions& crypto_options) override;
-
-  const std::vector<AudioCodec>& audio_send_codecs() override;
-  const std::vector<AudioCodec>& audio_recv_codecs() override;
-  RtpCapabilities GetAudioCapabilities() override;
-
-  std::vector<VideoCodec> video_codecs() override;
-  RtpCapabilities GetVideoCapabilities() override;
-
-  bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
-  void StopAecDump() override;
-
- protected:
-  VoiceEngineInterface& voice();
-  VideoEngineInterface& video();
-  const VoiceEngineInterface& voice() const;
-  const VideoEngineInterface& video() const;
+  VoiceEngineInterface& voice() override;
+  VideoEngineInterface& video() override;
+  const VoiceEngineInterface& voice() const override;
+  const VideoEngineInterface& video() const override;
 
  private:
   std::unique_ptr<VoiceEngineInterface> voice_engine_;