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.cc

@@ -313,6 +313,18 @@ bool OldTimestamp(uint32_t newTimestamp,
  * Misc utility routines
  */
 
+const WebRtc_UWord8* GetPayloadData(const WebRtcRTPHeader* rtp_header,
+                                    const WebRtc_UWord8* packet) {
+  return packet + rtp_header->header.headerLength;
+}
+
+WebRtc_UWord16 GetPayloadDataLength(const WebRtcRTPHeader* rtp_header,
+                                    const WebRtc_UWord16 packet_length) {
+  WebRtc_UWord16 length = packet_length - rtp_header->header.paddingLength -
+      rtp_header->header.headerLength;
+  return static_cast<WebRtc_UWord16>(length);
+}
+
 #if defined(_WIN32)
 bool StringCompare(const char* str1, const char* str2,
                    const WebRtc_UWord32 length) {