Introduce struct MediaConfig, with construction-time settings.

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 fb88e49e6f7732765d27b74710f2748fa0e3a3ad..511c372ecf6b87052e719efc08d548d8db49f3b7 100644
--- a/webrtc/media/webrtc/webrtcvoiceengine.h
+++ b/webrtc/media/webrtc/webrtcvoiceengine.h
@@ -52,6 +52,7 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback  {
 
   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
   VoiceMediaChannel* CreateChannel(webrtc::Call* call,
+                                   const MediaConfig& config,
                                    const AudioOptions& options);
 
   bool GetOutputVolume(int* level);
@@ -140,12 +141,15 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
                                       public webrtc::Transport {
  public:
   WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
+                          const MediaConfig& config,
                           const AudioOptions& options,
                           webrtc::Call* call);
   ~WebRtcVoiceMediaChannel() override;
 
   const AudioOptions& options() const { return options_; }
 
+  rtc::DiffServCodePoint PreferredDscp() const override;
+
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
   bool SetPlayout(bool playout) override;