Add flag enabling more packets to be retransmittable.

webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.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 90 matching lines @@
   payload_size_ -= next_packet_payload_len;
   --num_packets_left_;
   // Packets left to produce and data left to split should end at the same time.
   RTC_DCHECK_EQ(num_packets_left_ == 0, payload_size_ == 0);
 
   packet->SetMarker(payload_size_ == 0);
 
   return true;
 }
 
-ProtectionType RtpPacketizerGeneric::GetProtectionType() {
-  return kProtectedPacket;
-}
-
-StorageType RtpPacketizerGeneric::GetStorageType(
-    uint32_t retransmission_settings) {
-  return kAllowRetransmission;
-}
-
 std::string RtpPacketizerGeneric::ToString() {
   return "RtpPacketizerGeneric";
 }
 
 bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
                                    const uint8_t* payload_data,
                                    size_t payload_data_length) {
   assert(parsed_payload != NULL);
   if (payload_data_length == 0) {
     LOG(LS_ERROR) << "Empty payload.";
@@ skipping 11 matching lines @@
       (generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
   parsed_payload->type.Video.codec = kRtpVideoGeneric;
   parsed_payload->type.Video.width = 0;
   parsed_payload->type.Video.height = 0;
 
   parsed_payload->payload = payload_data;
   parsed_payload->payload_length = payload_data_length;
   return true;
 }
 }  // namespace webrtc