Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional.

The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch.

BUG=
TEST=vie & voe_auto_test full runs

Review URL: https://webrtc-codereview.appspot.com/1014006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

@@ -269,5 +269,63 @@ class RtpRtcpClock {
   virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) = 0;
 };
 
+// Null object version of RtpFeedback.
+class NullRtpFeedback : public RtpFeedback {
+ public:
+  virtual ~NullRtpFeedback() {}
+
+  virtual WebRtc_Word32 OnInitializeDecoder(
+      const WebRtc_Word32 id,
+      const WebRtc_Word8 payloadType,
+      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+      const int frequency,
+      const WebRtc_UWord8 channels,
+      const WebRtc_UWord32 rate) {
+   return 0;
+ }
+
+ virtual void OnPacketTimeout(const WebRtc_Word32 id) {}
+
+ virtual void OnReceivedPacket(const WebRtc_Word32 id,
+                               const RtpRtcpPacketType packetType) {}
+
+ virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
+                                    const RTPAliveType alive) {}
+
+ virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
+                                    const WebRtc_UWord32 SSRC) {}
+
+ virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
+                                    const WebRtc_UWord32 CSRC,
+                                    const bool added) {}
+};
+
+// Null object version of RtpData.
+class NullRtpData : public RtpData {
+ public:
+  virtual ~NullRtpData() {}
+  virtual WebRtc_Word32 OnReceivedPayloadData(
+      const WebRtc_UWord8* payloadData,
+      const WebRtc_UWord16 payloadSize,
+      const WebRtcRTPHeader* rtpHeader) {
+   return 0;
+ }
+};
+
+// Null object version of RtpAudioFeedback.
+class NullRtpAudioFeedback : public RtpAudioFeedback {
+ public:
+  virtual ~NullRtpAudioFeedback() {}
+
+  virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
+                                        const WebRtc_UWord8 event,
+                                        const bool endOfEvent) {}
+
+  virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
+                                    const WebRtc_UWord8 event,
+                                    const WebRtc_UWord16 lengthMs,
+                                    const WebRtc_UWord8 volume) {}
+};
+
 } // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_