[Cast] Clean-up RtpCastHeader and RtpParser, removing the last WebRTC dependency.

media/cast/rtp_receiver/rtp_receiver.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 // Interface to the rtp receiver.
 
 #ifndef MEDIA_CAST_RTP_RECEIVER_RTP_RECEIVER_H_
 #define MEDIA_CAST_RTP_RECEIVER_RTP_RECEIVER_H_
 
 #include "base/memory/scoped_ptr.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/rtcp/rtcp.h"
 #include "media/cast/rtp_receiver/receiver_stats.h"
 #include "media/cast/rtp_receiver/rtp_parser/rtp_parser.h"
 #include "media/cast/rtp_receiver/rtp_receiver_defines.h"
 
 namespace media {
 namespace cast {
 
 class RtpReceiver : public RtpParser {

[hubbe, 2014/04/24 00:31:13]

No need to inherit the parser anymore.
Please change to composition.
(Maybe add a todo to do the same sort of thing one step up, where
audio/video_receiver inherits RtpReceiver)

[miu, 2014/04/24 21:03:55]

Done.

  public:
   RtpReceiver(base::TickClock* clock,
               const AudioReceiverConfig* audio_config,
               const VideoReceiverConfig* video_config);
   virtual ~RtpReceiver();
 
   static uint32 GetSsrcOfSender(const uint8* rtcp_buffer, size_t length);
 
   bool ReceivedPacket(const uint8* packet, size_t length);
 
   RtpReceiverStatistics* GetStatistics() {
     return &stats_;
   }
 
+ protected:
+  // Subclasses implement this to consume and process deserialized packets.
+  virtual void OnReceivedPayloadData(const uint8* payload_data,
+                                     size_t payload_size,
+                                     const RtpCastHeader& rtp_header) = 0;
+
  private:
   ReceiverStats stats_;
 
   DISALLOW_COPY_AND_ASSIGN(RtpReceiver);
 };
 
 }  // namespace cast
 }  // namespace media
 
 #endif  // MEDIA_CAST_RTP_RECEIVER_RTP_RECEIVER_H_