Add a PacketOptions struct to webrtc::Transport.

webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.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 10 matching lines @@
 
 class ExtensionVerifyTransport : public webrtc::Transport {
  public:
   ExtensionVerifyTransport()
       : parser_(webrtc::RtpHeaderParser::Create()),
         received_packets_(0),
         bad_packets_(0),
         audio_level_id_(-1),
         absolute_sender_time_id_(-1) {}
 
-  bool SendRtp(const uint8_t* data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override {
     webrtc::RTPHeader header;
     if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
       bool ok = true;
       if (audio_level_id_ >= 0 &&
           !header.extension.hasAudioLevel) {
         ok = false;
       }
       if (absolute_sender_time_id_ >= 0 &&
           !header.extension.hasAbsoluteSendTime) {
         ok = false;
@@ skipping 102 matching lines @@
   EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true,
                                                               3));
   EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
                                                                 9));
   verifying_transport_.SetAbsoluteSenderTimeId(3);
   // Don't register audio level with header parser - unknown extensions should
   // be ignored when parsing.
   ResumePlaying();
   EXPECT_TRUE(verifying_transport_.Wait());
 }