Exposing RtpSenders and RtpReceivers from PeerConnection.

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 e903fdaca308da97ea08952068b854d9dd435339..7e25b66a3cba78495d239520be97189c04ee2667 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -28,6 +28,8 @@
 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
 
+#include "webrtc/base/basictypes.h"
+
 namespace cricket {
 
 class AudioRenderer;
@@ -40,6 +42,14 @@ struct VideoOptions;
 
 namespace webrtc {
 
+// TODO(deadbeef): Change the key from an ssrc to a "sender_id" or
+// "receiver_id" string, which will be the MSID in the short term and MID in
+// the long term.
+
+// TODO(deadbeef): These interfaces are effectively just a way for the
+// 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.
 class AudioProviderInterface {