Add flag enabling more packets to be retransmittable.

webrtc/modules/rtp_rtcp/source/rtp_format.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 23 matching lines @@
   virtual size_t SetPayloadData(
       const uint8_t* payload_data,
       size_t payload_size,
       const RTPFragmentationHeader* fragmentation) = 0;
 
   // Get the next payload with payload header.
   // Write payload and set marker bit of the |packet|.
   // Returns true on success, false otherwise.
   virtual bool NextPacket(RtpPacketToSend* packet) = 0;
 
-  virtual ProtectionType GetProtectionType() = 0;
-
-  virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
-
   virtual std::string ToString() = 0;
 };
 
 // TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
 // of the parsed payload, rather than just a pointer into the incoming buffer.
 // This way we can move some parsing out from the jitter buffer into here, and
 // the jitter buffer can just store that pointer rather than doing a copy there.
 class RtpDepacketizer {
  public:
   struct ParsedPayload {
     const uint8_t* payload;
     size_t payload_length;
     FrameType frame_type;
     RTPTypeHeader type;
   };
 
   static RtpDepacketizer* Create(RtpVideoCodecTypes type);
 
   virtual ~RtpDepacketizer() {}
 
   // Parses the RTP payload, parsed result will be saved in |parsed_payload|.
   virtual bool Parse(ParsedPayload* parsed_payload,
                      const uint8_t* payload_data,
                      size_t payload_data_length) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_