Wire up PacketTime to ReceiveStreams.

webrtc/video/rampup_tests.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 247 matching lines @@
           last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000;
     }
     EvolveTestState(average_rate_bps);
     interval_start_ms_ = now_ms;
     sent_bytes_ = 0;
   }
   return test::DirectTransport::SendRtp(data, length);
 }
 
 PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket(
-    MediaType media_type, const uint8_t* packet, size_t length) {
+    MediaType media_type,
+    const uint8_t* packet,
+    size_t length,
+    const PacketTime& packet_time) {
   rtc::CritScope lock(&crit_);
   RTPHeader header;
   EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));
   receive_stats_->IncomingPacket(header, length, false);
   remote_bitrate_estimator_->IncomingPacket(
       clock_->TimeInMilliseconds(), length - header.headerLength, header, true);
   if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) {
     remote_bitrate_estimator_->Process();
   }
   suspended_in_stats_ = send_stream_->GetStats().suspended;
@@ skipping 275 matching lines @@
 
 TEST_F(RampUpTest, AbsSendTimeSimulcastByRedWithRtx) {
   RunRampUpTest(3, 0, RtpExtension::kAbsSendTime, true, true);
 }
 
 TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
   RunRampUpTest(1, 0.9 * kSingleStreamTargetBps, RtpExtension::kAbsSendTime,
                 false, false);
 }
 }  // namespace webrtc