Add a PacketOptions struct to webrtc::Transport.

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

 /*
  *  Copyright (c) 2015 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 80 matching lines @@
    *
    * Input:
    *   id : stream id;
    *   stats : pointer to a CallStatistics to store the result.
    *
    * Returns false if the specified stream does not exist, true if succeeds.
    */
   bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
 
   // Inherit from class webrtc::Transport.
-  bool SendRtp(const uint8_t *data, size_t len) override;
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override;
   bool SendRtcp(const uint8_t *data, size_t len) override;
 
  private:
   struct Packet {
     enum Type { Rtp, Rtcp, } type_;
 
     Packet() : len_(0) {}
     Packet(Type type, const void* data, size_t len, uint32 time_ms)
         : type_(type), len_(len), send_time_ms_(time_ms) {
       EXPECT_LE(len_, kMaxPacketSizeByte);
@@ skipping 41 matching lines @@
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
 
   LoudestFilter loudest_filter_;
 
   const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
 };
 }  // namespace voetest
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_