Split out dscp option from VideoOptions to new struct MediaChannelOptions.

webrtc/media/webrtc/webrtcvoiceengine.h

Index: webrtc/media/webrtc/webrtcvoiceengine.h
diff --git a/webrtc/media/webrtc/webrtcvoiceengine.h b/webrtc/media/webrtc/webrtcvoiceengine.h
index de12ef3410fae4ad58a0cce36d219077502accc5..f08b76185a28339460c86b7a5adc73cb830b5b58 100644
--- a/webrtc/media/webrtc/webrtcvoiceengine.h
+++ b/webrtc/media/webrtc/webrtcvoiceengine.h
@@ -69,7 +69,8 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
 
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
-                                   const AudioOptions& options);
+                                   const MediaChannelOptions& options,
+                                   const AudioOptions& audio_options);
 
   bool GetOutputVolume(int* level);
   bool SetOutputVolume(int level);
@@ -157,12 +158,15 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
                                       public webrtc::Transport {
  public:
   WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
-                          const AudioOptions& options,
+                          const MediaChannelOptions& options,
+                          const AudioOptions& audio_options,
                           webrtc::Call* call);
   ~WebRtcVoiceMediaChannel() override;
 
   const AudioOptions& options() const { return options_; }
 
+  rtc::DiffServCodePoint MediaTypeDscpValue() const override;
+
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
   bool SetPlayout(bool playout) override;