Move some send stream configuration into webrtc::AudioSendStream.

webrtc/config.h

Index: webrtc/config.h
diff --git a/webrtc/config.h b/webrtc/config.h
index 4b863c8d23746e84a188a01039e8906f08172aca..114303e6169a55510adfa1d1884a260aef99f092 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -49,10 +49,13 @@ struct FecConfig {
   int red_rtx_payload_type;
 };
 
-// RTP header extension to use for the video stream, see RFC 5285.
+// RTP header extension, see RFC 5285.
 struct RtpExtension {
   RtpExtension(const std::string& name, int id) : name(name), id(id) {}
   std::string ToString() const;
+  bool operator==(const RtpExtension& rhs) const {
+    return name == rhs.name && id == rhs.id;
+  }
   static bool IsSupportedForAudio(const std::string& name);
   static bool IsSupportedForVideo(const std::string& name);