Reland of Adding the ability to create an RtpSender without a track.

talk/app/webrtc/mediastreamprovider.h

Index: talk/app/webrtc/mediastreamprovider.h
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
index 1c62daf9f16aa33d0874b524970e1b41dbcbff28..a78b55a68c0a73e95cae68b9ab923c0b5a10200b 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -50,8 +50,8 @@ namespace webrtc {
 // RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
 // refactored away eventually, as the classes converge.
 
-// This interface is called by AudioTrackHandler classes in mediastreamhandler.h
-// to change the settings of an audio track connected to certain PeerConnection.
+// This interface is called by AudioRtpSender/Receivers to change the settings
+// of an audio track connected to certain PeerConnection.
 class AudioProviderInterface {
  public:
   // Enable/disable the audio playout of a remote audio track with |ssrc|.
@@ -71,9 +71,8 @@ class AudioProviderInterface {
   virtual ~AudioProviderInterface() {}
 };
 
-// This interface is called by VideoTrackHandler classes in mediastreamhandler.h
-// to change the settings of a video track connected to a certain
-// PeerConnection.
+// This interface is called by VideoRtpSender/Receivers to change the settings
+// of a video track connected to a certain PeerConnection.
 class VideoProviderInterface {
  public:
   virtual bool SetCaptureDevice(uint32_t ssrc,