Add a PacketOptions struct to webrtc::Transport.

webrtc/modules/rtp_rtcp/source/rtp_sender.h

 /*
  *  Copyright (c) 2012 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 332 matching lines @@
   // return a larger value that their argument.
   size_t TrySendRedundantPayloads(size_t bytes);
 
   void BuildPaddingPacket(uint8_t* packet,
                           size_t header_length,
                           size_t padding_length);
 
   void BuildRtxPacket(uint8_t* buffer, size_t* length,
                       uint8_t* buffer_rtx);
 
-  bool SendPacketToNetwork(const uint8_t *packet, size_t size);
+  bool SendPacketToNetwork(const uint8_t* packet,
+                           size_t size,
+                           const PacketOptions& options);
 
   void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms);
 
   // Find the byte position of the RTP extension as indicated by |type| in
   // |rtp_packet|. Return false if such extension doesn't exist.
   bool FindHeaderExtensionPosition(RTPExtensionType type,
                                    const uint8_t* rtp_packet,
                                    size_t rtp_packet_length,
                                    const RTPHeader& rtp_header,
                                    size_t* position) const;
@@ skipping 97 matching lines @@
   // SetTargetBitrateKbps or GetTargetBitrateKbps. Also remember
   // that by the time the function returns there is no guarantee
   // that the target bitrate is still valid.
   rtc::scoped_ptr<CriticalSectionWrapper> target_bitrate_critsect_;
   uint32_t target_bitrate_ GUARDED_BY(target_bitrate_critsect_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_