Add flag enabling more packets to be retransmittable.

webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

 /*
  *  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 371 matching lines @@
   uint8_t* buffer =
       rtp_packet->AllocatePayload(kFuAHeaderSize + fragment.length);
   buffer[0] = fu_indicator;
   buffer[1] = fu_header;
   memcpy(buffer + kFuAHeaderSize, fragment.buffer, fragment.length);
   if (packet->last_fragment)
     input_fragments_.pop_front();
   packets_.pop();
 }
 
-ProtectionType RtpPacketizerH264::GetProtectionType() {
-  return kProtectedPacket;
-}
-
-StorageType RtpPacketizerH264::GetStorageType(
-    uint32_t retransmission_settings) {
-  return kAllowRetransmission;
-}
-
 std::string RtpPacketizerH264::ToString() {
   return "RtpPacketizerH264";
 }
 
 RtpDepacketizerH264::RtpDepacketizerH264() : offset_(0), length_(0) {}
 RtpDepacketizerH264::~RtpDepacketizerH264() {}
 
 bool RtpDepacketizerH264::Parse(ParsedPayload* parsed_payload,
                                 const uint8_t* payload_data,
                                 size_t payload_data_length) {
@@ skipping 257 matching lines @@
   h264->packetization_type = kH264FuA;
   h264->nalu_type = original_nal_type;
   if (first_fragment) {
     h264->nalus[h264->nalus_length] = nalu;
     h264->nalus_length = 1;
   }
   return true;
 }
 
 }  // namespace webrtc