Replaced the _audio parameter with a strategy.

The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.

In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.

BUG=
TEST=vie/voe_auto_test, trybots

Review URL: https://webrtc-codereview.appspot.com/1001006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/source/rtp_utility.h

@@ -59,6 +59,8 @@ namespace ModuleRTPUtility
         PayloadUnion typeSpecific;
     };
 
+    typedef std::map<WebRtc_Word8, Payload*> PayloadTypeMap;
+
     // Return a clock that reads the time as reported by the operating
     // system. The returned instances are guaranteed to read the same
     // times; in particular, they return relative times relative to
@@ -85,6 +87,14 @@ namespace ModuleRTPUtility
 
     WebRtc_UWord32 pow2(WebRtc_UWord8 exp);
 
+    // Returns a pointer to the payload data given a packet.
+    const WebRtc_UWord8* GetPayloadData(const WebRtcRTPHeader* rtp_header,
+                                        const WebRtc_UWord8* packet);
+
+    // Returns payload length given a packet.
+    WebRtc_UWord16 GetPayloadDataLength(const WebRtcRTPHeader* rtp_header,
+                                        const WebRtc_UWord16 packet_length);
+
     // Returns true if |newTimestamp| is older than |existingTimestamp|.
     // |wrapped| will be set to true if there has been a wraparound between the
     // two timestamps.
@@ -170,7 +180,7 @@ namespace ModuleRTPUtility
         int                  frameWidth;
         int                  frameHeight;
 
-        const WebRtc_UWord8*   data; 
+        const WebRtc_UWord8*   data;
         WebRtc_UWord16         dataLength;
     };