Add flag enabling more packets to be retransmittable.

webrtc/modules/rtp_rtcp/source/rtp_format_h264.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 @@
 
   size_t SetPayloadData(const uint8_t* payload_data,
                         size_t payload_size,
                         const RTPFragmentationHeader* fragmentation) override;
 
   // Get the next payload with H264 payload header.
   // Write payload and set marker bit of the |packet|.
   // Returns true on success, false otherwise.
   bool NextPacket(RtpPacketToSend* rtp_packet) override;
 
-  ProtectionType GetProtectionType() override;
-
-  StorageType GetStorageType(uint32_t retransmission_settings) override;
-
   std::string ToString() override;
 
  private:
   // Input fragments (NAL units), with an optionally owned temporary buffer,
   // used in case the fragment gets modified.
   struct Fragment {
     Fragment(const uint8_t* buffer, size_t length);
     explicit Fragment(const Fragment& fragment);
     const uint8_t* buffer = nullptr;
     size_t length = 0;
@@ skipping 57 matching lines @@
                     const uint8_t* payload_data);
   bool ProcessStapAOrSingleNalu(RtpDepacketizer::ParsedPayload* parsed_payload,
                                 const uint8_t* payload_data);
 
   size_t offset_;
   size_t length_;
   std::unique_ptr<rtc::Buffer> modified_buffer_;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_