Split out dscp option from VideoOptions to new struct MediaChannelOptions.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 51 matching lines @@
 
   WebRtcVoiceEngine();
   // Dependency injection for testing.
   explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
   ~WebRtcVoiceEngine();
   bool Init(rtc::Thread* worker_thread);
   void Terminate();
 
   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);
   int GetInputLevel();
 
   const std::vector<AudioCodec>& codecs();
   RtpCapabilities GetCapabilities() const;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
@@ skipping 67 matching lines @@
 
   RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
 };
 
 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
 // WebRtc Voice Engine.
 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;
   bool PausePlayout();
   bool ResumePlayout();
   bool SetSend(SendFlags send) override;
   bool PauseSend();
   bool ResumeSend();
   bool SetAudioSend(uint32_t ssrc,
                     bool enable,
@@ skipping 109 matching lines @@
 
   class WebRtcAudioReceiveStream;
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_