Add a PacketOptions struct to webrtc::Transport.

webrtc/video/video_quality_test.cc

 /*
  *  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.
  */
 #include <stdio.h>
@@ skipping 120 matching lines @@
       rtc::CritScope lock(&crit_);
       if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0)
         first_send_frame_ = copy;
 
       frames_.push_back(copy);
     }
 
     input_->IncomingCapturedFrame(video_frame);
   }
 
-  bool SendRtp(const uint8_t* packet, size_t length) override {
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override {
     rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     RTPHeader header;
     parser->Parse(packet, length, &header);
 
     {
       rtc::CritScope lock(&crit_);
       if (rtp_timestamp_delta_ == 0) {
         rtp_timestamp_delta_ = header.timestamp - first_send_frame_.timestamp();
         first_send_frame_.Reset();
       }
       uint32_t timestamp = header.timestamp - rtp_timestamp_delta_;
       send_times_[timestamp] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
       encoded_frame_sizes_[timestamp] +=
           length - (header.headerLength + header.paddingLength);
     }
 
-    return transport_->SendRtp(packet, length);
+    return transport_->SendRtp(packet, length, options);
   }
 
   bool SendRtcp(const uint8_t* packet, size_t length) override {
     return transport_->SendRtcp(packet, length);
   }
 
   void EncodedFrameCallback(const EncodedFrame& frame) override {
     rtc::CritScope lock(&comparison_lock_);
     if (frames_recorded_ < frames_to_process_)
       encoded_frame_size_.AddSample(frame.length_);
@@ skipping 660 matching lines @@
   send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc