[Cast] Remove AudioDecoder's dependency on WebRTC, and refactor/clean-up AudioReceiver.

media/cast/rtp_receiver/rtp_receiver.h

Index: media/cast/rtp_receiver/rtp_receiver.h
diff --git a/media/cast/rtp_receiver/rtp_receiver.h b/media/cast/rtp_receiver/rtp_receiver.h
index 5cfa8c69a9094f85aeb6516cf2cf2cba15c2c67f..db0e1731a1d25c905a67aeb342be0f67ca5bdf07 100644
--- a/media/cast/rtp_receiver/rtp_receiver.h
+++ b/media/cast/rtp_receiver/rtp_receiver.h
@@ -19,7 +19,7 @@ class RtpData {
  public:
   virtual void OnReceivedPayloadData(const uint8* payload_data,
                                      size_t payload_size,
-                                     const RtpCastHeader* rtp_header) = 0;
+                                     const RtpCastHeader& rtp_header) = 0;
 
  protected:
   virtual ~RtpData() {}
@@ -28,22 +28,22 @@ class RtpData {
 class ReceiverStats;
 class RtpParser;
 
-class RtpReceiver {
+class RtpReceiver : public RtpReceiverStatistics {
  public:
   RtpReceiver(base::TickClock* clock,
               const AudioReceiverConfig* audio_config,
               const VideoReceiverConfig* video_config,
               RtpData* incoming_payload_callback);
-  ~RtpReceiver();
+  virtual ~RtpReceiver();
 
   static uint32 GetSsrcOfSender(const uint8* rtcp_buffer, size_t length);
 
   bool ReceivedPacket(const uint8* packet, size_t length);
 
-  void GetStatistics(uint8* fraction_lost,
-                     uint32* cumulative_lost,  // 24 bits valid.
-                     uint32* extended_high_sequence_number,
-                     uint32* jitter);
+  virtual void GetStatistics(uint8* fraction_lost,
+                             uint32* cumulative_lost,  // 24 bits valid.
+                             uint32* extended_high_sequence_number,
+                             uint32* jitter) OVERRIDE;
 
  private:
   scoped_ptr<ReceiverStats> stats_;