Add a PacketOptions struct to webrtc::Transport.

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

 /*
 *  Copyright (c) 2013 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 78 matching lines @@
   void DropEveryNthPacket(int n) {
     packet_loss_ = n;
   }
 
   void DropConsecutivePackets(int start, int total) {
     consecutive_drop_start_ = start;
     consecutive_drop_end_ = start + total;
     packet_loss_ = 0;
   }
 
-  bool SendRtp(const uint8_t* data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const PacketOptions& options) override {
     count_++;
     const unsigned char* ptr = static_cast<const unsigned  char*>(data);
     uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
     if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
     uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
     size_t packet_length = len;
     // TODO(pbos): Figure out why this needs to be initialized. Likely this
     // is hiding a bug either in test setup or other code.
     // https://code.google.com/p/webrtc/issues/detail?id=3183
     uint8_t restored_packet[1500] = {0};
@@ skipping 232 matching lines @@
 
 TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
       *(receiver_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }