Moving overhead counting to bitrate estimators.

webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc

 /*
  *  Copyright (c) 2016 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 "webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h"
 
 #include <algorithm>
 #include <limits>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/modules/congestion_controller/delay_based_bwe.h"
 
-namespace webrtc {
+namespace {
 
 constexpr size_t kMtu = 1200;
 constexpr uint32_t kAcceptedBitrateErrorBps = 50000;
 
 // Number of packets needed before we have a valid estimate.
 constexpr int kNumInitialPackets = 2;
 
 constexpr int kInitialProbingPackets = 5;
 
+constexpr size_t kRtpHeadersSize = 12;
+constexpr size_t kTransportHeadersSize = 20;
+
+}  // namespace
+
+namespace webrtc {
+
 namespace test {
 
 void TestBitrateObserver::OnReceiveBitrateChanged(
     const std::vector<uint32_t>& ssrcs,
     uint32_t bitrate) {
   latest_bitrate_ = bitrate;
   updated_ = true;
 }
 
 RtpStream::RtpStream(int fps, int bitrate_bps)
@@ skipping 132 matching lines @@
                    PacedPacketInfo());
 }
 
 void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
                                          int64_t send_time_ms,
                                          uint16_t sequence_number,
                                          size_t payload_size,
                                          const PacedPacketInfo& pacing_info) {
   RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
   PacketFeedback packet(arrival_time_ms + arrival_time_offset_ms_, send_time_ms,
-                        sequence_number, payload_size, pacing_info);
+                        sequence_number, payload_size, kRtpHeadersSize,
+                        kTransportHeadersSize, pacing_info);
   std::vector<PacketFeedback> packets;
   packets.push_back(packet);
   DelayBasedBwe::Result result =
       bitrate_estimator_->IncomingPacketFeedbackVector(packets);
   const uint32_t kDummySsrc = 0;
   if (result.updated) {
     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
                                               result.target_bitrate_bps);
   }
 }
@@ skipping 304 matching lines @@
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
                      sequence_number++, 1000);
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   uint32_t bitrate_after = 0;
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
   EXPECT_LT(bitrate_after, bitrate_before);
 }
 }  // namespace webrtc