Add a PacketOptions struct to webrtc::Transport.

webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc

 /*
  *  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 12 matching lines @@
   rtp_rtcp_module_ = rtp_rtcp_module;
   rtp_payload_registry_ = payload_registry;
   rtp_receiver_ = receiver;
   receive_statistics_ = receive_statistics;
 }
 
 void LoopBackTransport::DropEveryNthPacket(int n) {
   packet_loss_ = n;
 }
 
-bool LoopBackTransport::SendRtp(const uint8_t* data, size_t len) {
+bool LoopBackTransport::SendRtp(const uint8_t* data,
+                                size_t len,
+                                const PacketOptions& options) {
   count_++;
   if (packet_loss_ > 0) {
     if ((count_ % packet_loss_) == 0) {
       return true;
     }
   }
   RTPHeader header;
   rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
   if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
     return false;
@@ skipping 131 matching lines @@
   RTPHeader rtx_header;
   rtx_header.ssrc = kRtxSsrc;
   rtx_header.payloadType = kRtxPayloadType;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = 0;
   EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
   rtx_header.ssrc = kRtxSsrc;
   rtx_header.payloadType = 0;
   EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
 }