modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc - src - Git at Google

 /*
  *  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 "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"

 #include <algorithm>
 #include <cstddef>
 #include <cstdint>
 #include <memory>
 #include <vector>

 #include "absl/types/optional.h"
 #include "api/transport/network_types.h"
 #include "api/units/data_rate.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
 #include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
 #include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
 #include "rtc_base/checks.h"
 #include "test/field_trial.h"

 namespace webrtc {
 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;

 namespace test {

 void TestBitrateObserver::OnReceiveBitrateChanged(uint32_t bitrate) {
   latest_bitrate_ = bitrate;
   updated_ = true;
 }

 RtpStream::RtpStream(int fps, int bitrate_bps)
     : fps_(fps), bitrate_bps_(bitrate_bps), next_rtp_time_(0) {
   RTC_CHECK_GT(fps_, 0);
 }

 // Generates a new frame for this stream. If called too soon after the
 // previous frame, no frame will be generated. The frame is split into
 // packets.
 int64_t RtpStream::GenerateFrame(int64_t time_now_us,
                                  std::vector<PacketResult>* packets) {
   if (time_now_us < next_rtp_time_) {
     return next_rtp_time_;
   }
   RTC_CHECK(packets != NULL);
   size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
   size_t n_packets =
       std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
   size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
   for (size_t i = 0; i < n_packets; ++i) {
     PacketResult packet;
     packet.sent_packet.send_time =
         Timestamp::Micros(time_now_us + kSendSideOffsetUs);
     packet.sent_packet.size = DataSize::Bytes(payload_size);
     packets->push_back(packet);
   }
   next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
   return next_rtp_time_;
 }

 // The send-side time when the next frame can be generated.
 int64_t RtpStream::next_rtp_time() const {
   return next_rtp_time_;
 }

 void RtpStream::set_bitrate_bps(int bitrate_bps) {
   ASSERT_GE(bitrate_bps, 0);
   bitrate_bps_ = bitrate_bps;
 }

 int RtpStream::bitrate_bps() const {
   return bitrate_bps_;
 }

 bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
                         const std::unique_ptr<RtpStream>& rhs) {
   return lhs->next_rtp_time_ < rhs->next_rtp_time_;
 }

 StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
     : capacity_(capacity), prev_arrival_time_us_(time_now) {}

 StreamGenerator::~StreamGenerator() = default;

 // Add a new stream.
 void StreamGenerator::AddStream(RtpStream* stream) {
   streams_.push_back(std::unique_ptr<RtpStream>(stream));
 }

 // Set the link capacity.
 void StreamGenerator::set_capacity_bps(int capacity_bps) {
   ASSERT_GT(capacity_bps, 0);
   capacity_ = capacity_bps;
 }

 // Divides `bitrate_bps` among all streams. The allocated bitrate per stream
 // is decided by the current allocation ratios.
 void StreamGenerator::SetBitrateBps(int bitrate_bps) {
   ASSERT_GE(streams_.size(), 0u);
   int total_bitrate_before = 0;
   for (const auto& stream : streams_) {
     total_bitrate_before += stream->bitrate_bps();
   }
   int64_t bitrate_before = 0;
   int total_bitrate_after = 0;
   for (const auto& stream : streams_) {
     bitrate_before += stream->bitrate_bps();
     int64_t bitrate_after =
         (bitrate_before * bitrate_bps + total_bitrate_before / 2) /
         total_bitrate_before;
     stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
     total_bitrate_after += stream->bitrate_bps();
   }
   ASSERT_EQ(bitrate_before, total_bitrate_before);
   EXPECT_EQ(total_bitrate_after, bitrate_bps);
 }

 // TODO(holmer): Break out the channel simulation part from this class to make
 // it possible to simulate different types of channels.
 int64_t StreamGenerator::GenerateFrame(std::vector<PacketResult>* packets,
                                        int64_t time_now_us) {
   RTC_CHECK(packets != NULL);
   RTC_CHECK(packets->empty());
   RTC_CHECK_GT(capacity_, 0);
   auto it =
       std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
   (*it)->GenerateFrame(time_now_us, packets);
   for (PacketResult& packet : *packets) {
     int capacity_bpus = capacity_ / 1000;
     int64_t required_network_time_us =
         (8 * 1000 * packet.sent_packet.size.bytes() + capacity_bpus / 2) /
         capacity_bpus;
     prev_arrival_time_us_ =
         std::max(time_now_us + required_network_time_us,
                  prev_arrival_time_us_ + required_network_time_us);
     packet.receive_time = Timestamp::Micros(prev_arrival_time_us_);
   }
   it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
   return std::max((*it)->next_rtp_time(), time_now_us);
 }
 }  // namespace test

 DelayBasedBweTest::DelayBasedBweTest()
     : field_trial(std::make_unique<test::ScopedFieldTrials>(
           "WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/")),
       clock_(100000000),
       acknowledged_bitrate_estimator_(
           AcknowledgedBitrateEstimatorInterface::Create(&field_trial_config_)),
       probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
       bitrate_estimator_(
           new DelayBasedBwe(&field_trial_config_, nullptr, nullptr)),
       stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
                                                   clock_.TimeInMicroseconds())),
       arrival_time_offset_ms_(0),
       next_sequence_number_(0),
       first_update_(true) {}

 DelayBasedBweTest::~DelayBasedBweTest() {}

 void DelayBasedBweTest::AddDefaultStream() {
   stream_generator_->AddStream(new test::RtpStream(30, 3e5));
 }

 const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;

 void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
                                          int64_t send_time_ms,
                                          size_t payload_size) {
   IncomingFeedback(arrival_time_ms, send_time_ms, payload_size,
                    PacedPacketInfo());
 }

 void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
                                          int64_t send_time_ms,
                                          size_t payload_size,
                                          const PacedPacketInfo& pacing_info) {
   RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
   IncomingFeedback(Timestamp::Millis(arrival_time_ms + arrival_time_offset_ms_),
                    Timestamp::Millis(send_time_ms), payload_size, pacing_info);
 }

 void DelayBasedBweTest::IncomingFeedback(Timestamp receive_time,
                                          Timestamp send_time,
                                          size_t payload_size,
                                          const PacedPacketInfo& pacing_info) {
   PacketResult packet;
   packet.receive_time = receive_time;
   packet.sent_packet.send_time = send_time;
   packet.sent_packet.size = DataSize::Bytes(payload_size);
   packet.sent_packet.pacing_info = pacing_info;
   packet.sent_packet.sequence_number = next_sequence_number_++;
   if (packet.sent_packet.pacing_info.probe_cluster_id !=
       PacedPacketInfo::kNotAProbe)
     probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);

   TransportPacketsFeedback msg;
   msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
   msg.packet_feedbacks.push_back(packet);
   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
       msg.SortedByReceiveTime());
   DelayBasedBwe::Result result =
       bitrate_estimator_->IncomingPacketFeedbackVector(
           msg, acknowledged_bitrate_estimator_->bitrate(),
           probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
           /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
   if (result.updated) {
     bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
   }
 }

 // Generates a frame of packets belonging to a stream at a given bitrate and
 // with a given ssrc. The stream is pushed through a very simple simulated
 // network, and is then given to the receive-side bandwidth estimator.
 // Returns true if an over-use was seen, false otherwise.
 // The StreamGenerator::updated() should be used to check for any changes in
 // target bitrate after the call to this function.
 bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
                                                 uint32_t bitrate_bps) {
   stream_generator_->SetBitrateBps(bitrate_bps);
   std::vector<PacketResult> packets;

   int64_t next_time_us =
       stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
   if (packets.empty())
     return false;

   bool overuse = false;
   bitrate_observer_.Reset();
   clock_.AdvanceTimeMicroseconds(packets.back().receive_time.us() -
                                  clock_.TimeInMicroseconds());
   for (auto& packet : packets) {
     RTC_CHECK_GE(packet.receive_time.ms() + arrival_time_offset_ms_, 0);
     packet.receive_time += TimeDelta::Millis(arrival_time_offset_ms_);

     if (packet.sent_packet.pacing_info.probe_cluster_id !=
         PacedPacketInfo::kNotAProbe)
       probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
   }

   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
   TransportPacketsFeedback msg;
   msg.packet_feedbacks = packets;
   msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());

   DelayBasedBwe::Result result =
       bitrate_estimator_->IncomingPacketFeedbackVector(
           msg, acknowledged_bitrate_estimator_->bitrate(),
           probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
           /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
   if (result.updated) {
     bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
     if (!first_update_ && result.target_bitrate.bps() < bitrate_bps)
       overuse = true;
     first_update_ = false;
   }

   clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
   return overuse;
 }

 // Run the bandwidth estimator with a stream of `number_of_frames` frames, or
 // until it reaches `target_bitrate`.
 // Can for instance be used to run the estimator for some time to get it
 // into a steady state.
 uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
                                            int max_number_of_frames,
                                            uint32_t start_bitrate,
                                            uint32_t min_bitrate,
                                            uint32_t max_bitrate,
                                            uint32_t target_bitrate) {
   uint32_t bitrate_bps = start_bitrate;
   bool bitrate_update_seen = false;
   // Produce `number_of_frames` frames and give them to the estimator.
   for (int i = 0; i < max_number_of_frames; ++i) {
     bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
     if (overuse) {
       EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate);
       EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate);
       bitrate_bps = bitrate_observer_.latest_bitrate();
       bitrate_update_seen = true;
     } else if (bitrate_observer_.updated()) {
       bitrate_bps = bitrate_observer_.latest_bitrate();
       bitrate_observer_.Reset();
     }
     if (bitrate_update_seen && bitrate_bps > target_bitrate) {
       break;
     }
   }
   EXPECT_TRUE(bitrate_update_seen);
   return bitrate_bps;
 }

 void DelayBasedBweTest::InitialBehaviorTestHelper(
     uint32_t expected_converge_bitrate) {
   const int kFramerate = 50;  // 50 fps to avoid rounding errors.
   const int kFrameIntervalMs = 1000 / kFramerate;
   const PacedPacketInfo kPacingInfo(0, 5, 5000);
   DataRate bitrate = DataRate::Zero();
   int64_t send_time_ms = 0;
   std::vector<uint32_t> ssrcs;
   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
   EXPECT_EQ(0u, ssrcs.size());
   clock_.AdvanceTimeMilliseconds(1000);
   EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
   EXPECT_FALSE(bitrate_observer_.updated());
   bitrate_observer_.Reset();
   clock_.AdvanceTimeMilliseconds(1000);
   // Inserting packets for 5 seconds to get a valid estimate.
   for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
     // NOTE!!! If the following line is moved under the if case then this test
     //         wont work on windows realease bots.
     PacedPacketInfo pacing_info =
         i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();

     if (i == kNumInitialPackets) {
       EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
       EXPECT_EQ(0u, ssrcs.size());
       EXPECT_FALSE(bitrate_observer_.updated());
       bitrate_observer_.Reset();
     }
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
                      pacing_info);
     clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
     send_time_ms += kFrameIntervalMs;
   }
   EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
   ASSERT_EQ(1u, ssrcs.size());
   EXPECT_EQ(kDefaultSsrc, ssrcs.front());
   EXPECT_NEAR(expected_converge_bitrate, bitrate.bps(),
               kAcceptedBitrateErrorBps);
   EXPECT_TRUE(bitrate_observer_.updated());
   bitrate_observer_.Reset();
   EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate.bps());
 }

 void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
     uint32_t expected_bitrate_bps) {
   const int kFramerate = 50;  // 50 fps to avoid rounding errors.
   const int kFrameIntervalMs = 1000 / kFramerate;
   const PacedPacketInfo kPacingInfo(0, 5, 5000);
   int64_t send_time_ms = 0;
   // Inserting packets for five seconds to get a valid estimate.
   for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
     // NOTE!!! If the following line is moved under the if case then this test
     //         wont work on windows realease bots.
     PacedPacketInfo pacing_info =
         i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();

     // TODO(sprang): Remove this hack once the single stream estimator is gone,
     // as it doesn't do anything in Process().
     if (i == kNumInitialPackets) {
       // Process after we have enough frames to get a valid input rate estimate.

       EXPECT_FALSE(bitrate_observer_.updated());  // No valid estimate.
     }
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
                      pacing_info);
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   EXPECT_TRUE(bitrate_observer_.updated());
   EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
               kAcceptedBitrateErrorBps);
   for (int i = 0; i < 10; ++i) {
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     send_time_ms += 2 * kFrameIntervalMs;
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
     IncomingFeedback(clock_.TimeInMilliseconds(),
                      send_time_ms - kFrameIntervalMs, 1000);
   }
   EXPECT_TRUE(bitrate_observer_.updated());
   EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
               kAcceptedBitrateErrorBps);
 }

 // Make sure we initially increase the bitrate as expected.
 void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
     int expected_iterations) {
   // This threshold corresponds approximately to increasing linearly with
   // bitrate(i) = 1.04 * bitrate(i-1) + 1000
   // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
   uint32_t bitrate_bps = 30000;
   int iterations = 0;
   AddDefaultStream();
   // Feed the estimator with a stream of packets and verify that it reaches
   // 500 kbps at the expected time.
   while (bitrate_bps < 5e5) {
     bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
     if (overuse) {
       EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
       bitrate_bps = bitrate_observer_.latest_bitrate();
       bitrate_observer_.Reset();
     } else if (bitrate_observer_.updated()) {
       bitrate_bps = bitrate_observer_.latest_bitrate();
       bitrate_observer_.Reset();
     }
     ++iterations;
   }
   ASSERT_EQ(expected_iterations, iterations);
 }

 void DelayBasedBweTest::CapacityDropTestHelper(
     int number_of_streams,
     bool wrap_time_stamp,
     uint32_t expected_bitrate_drop_delta,
     int64_t receiver_clock_offset_change_ms) {
   const int kFramerate = 30;
   const int kStartBitrate = 900e3;
   const int kMinExpectedBitrate = 800e3;
   const int kMaxExpectedBitrate = 1100e3;
   const uint32_t kInitialCapacityBps = 1000e3;
   const uint32_t kReducedCapacityBps = 500e3;

   int steady_state_time = 0;
   if (number_of_streams <= 1) {
     steady_state_time = 10;
     AddDefaultStream();
   } else {
     steady_state_time = 10 * number_of_streams;
     int bitrate_sum = 0;
     int kBitrateDenom = number_of_streams * (number_of_streams - 1);
     for (int i = 0; i < number_of_streams; i++) {
       // First stream gets half available bitrate, while the rest share the
       // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
       int bitrate = kStartBitrate / 2;
       if (i > 0) {
         bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
       }
       stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
       bitrate_sum += bitrate;
     }
     ASSERT_EQ(bitrate_sum, kStartBitrate);
   }

   // Run in steady state to make the estimator converge.
   stream_generator_->set_capacity_bps(kInitialCapacityBps);
   uint32_t bitrate_bps = SteadyStateRun(
       kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
       kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
   EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u);
   bitrate_observer_.Reset();

   // Add an offset to make sure the BWE can handle it.
   arrival_time_offset_ms_ += receiver_clock_offset_change_ms;

   // Reduce the capacity and verify the decrease time.
   stream_generator_->set_capacity_bps(kReducedCapacityBps);
   int64_t overuse_start_time = clock_.TimeInMilliseconds();
   int64_t bitrate_drop_time = -1;
   for (int i = 0; i < 100 * number_of_streams; ++i) {
     GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
     if (bitrate_drop_time == -1 &&
         bitrate_observer_.latest_bitrate() <= kReducedCapacityBps) {
       bitrate_drop_time = clock_.TimeInMilliseconds();
     }
     if (bitrate_observer_.updated())
       bitrate_bps = bitrate_observer_.latest_bitrate();
   }

   EXPECT_NEAR(expected_bitrate_drop_delta,
               bitrate_drop_time - overuse_start_time, 33);
 }

 void DelayBasedBweTest::TestTimestampGroupingTestHelper() {
   const int kFramerate = 50;  // 50 fps to avoid rounding errors.
   const int kFrameIntervalMs = 1000 / kFramerate;
   int64_t send_time_ms = 0;
   // Initial set of frames to increase the bitrate. 6 seconds to have enough
   // time for the first estimate to be generated and for Process() to be called.
   for (int i = 0; i <= 6 * kFramerate; ++i) {
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);

     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   EXPECT_TRUE(bitrate_observer_.updated());
   EXPECT_GE(bitrate_observer_.latest_bitrate(), 400000u);

   // Insert batches of frames which were sent very close in time. Also simulate
   // capacity over-use to see that we back off correctly.
   const int kTimestampGroupLength = 15;
   for (int i = 0; i < 100; ++i) {
     for (int j = 0; j < kTimestampGroupLength; ++j) {
       // Insert `kTimestampGroupLength` frames with just 1 timestamp ticks in
       // between. Should be treated as part of the same group by the estimator.
       IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 100);
       clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
       send_time_ms += 1;
     }
     // Increase time until next batch to simulate over-use.
     clock_.AdvanceTimeMilliseconds(10);
     send_time_ms += kFrameIntervalMs - kTimestampGroupLength;
   }
   EXPECT_TRUE(bitrate_observer_.updated());
   // Should have reduced the estimate.
   EXPECT_LT(bitrate_observer_.latest_bitrate(), 400000u);
 }

 void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
   const int kFramerate = 100;
   const int kFrameIntervalMs = 1000 / kFramerate;
   int64_t send_time_ms = 0;

   for (size_t i = 0; i < 3000; ++i) {
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   DataRate bitrate_before = DataRate::Zero();
   std::vector<uint32_t> ssrcs;
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);

   clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
   send_time_ms += silence_time_s * 1000;

   for (size_t i = 0; i < 24; ++i) {
     IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
     send_time_ms += kFrameIntervalMs;
   }
   DataRate bitrate_after = DataRate::Zero();
   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
   EXPECT_LT(bitrate_after, bitrate_before);
 }
 }  // namespace webrtc
	/*
	* 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 "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"

	#include <algorithm>
	#include <cstddef>
	#include <cstdint>
	#include <memory>
	#include <vector>

	#include "absl/types/optional.h"
	#include "api/transport/network_types.h"
	#include "api/units/data_rate.h"
	#include "api/units/data_size.h"
	#include "api/units/time_delta.h"
	#include "api/units/timestamp.h"
	#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
	#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
	#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
	#include "rtc_base/checks.h"
	#include "test/field_trial.h"

	namespace webrtc {
	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;

	namespace test {

	void TestBitrateObserver::OnReceiveBitrateChanged(uint32_t bitrate) {
	latest_bitrate_ = bitrate;
	updated_ = true;
	}

	RtpStream::RtpStream(int fps, int bitrate_bps)
	: fps_(fps), bitrate_bps_(bitrate_bps), next_rtp_time_(0) {
	RTC_CHECK_GT(fps_, 0);
	}

	// Generates a new frame for this stream. If called too soon after the
	// previous frame, no frame will be generated. The frame is split into
	// packets.
	int64_t RtpStream::GenerateFrame(int64_t time_now_us,
	std::vector<PacketResult>* packets) {
	if (time_now_us < next_rtp_time_) {
	return next_rtp_time_;
	}
	RTC_CHECK(packets != NULL);
	size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
	size_t n_packets =
	std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
	size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
	for (size_t i = 0; i < n_packets; ++i) {
	PacketResult packet;
	packet.sent_packet.send_time =
	Timestamp::Micros(time_now_us + kSendSideOffsetUs);
	packet.sent_packet.size = DataSize::Bytes(payload_size);
	packets->push_back(packet);
	}
	next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
	return next_rtp_time_;
	}

	// The send-side time when the next frame can be generated.
	int64_t RtpStream::next_rtp_time() const {
	return next_rtp_time_;
	}

	void RtpStream::set_bitrate_bps(int bitrate_bps) {
	ASSERT_GE(bitrate_bps, 0);
	bitrate_bps_ = bitrate_bps;
	}

	int RtpStream::bitrate_bps() const {
	return bitrate_bps_;
	}

	bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
	const std::unique_ptr<RtpStream>& rhs) {
	return lhs->next_rtp_time_ < rhs->next_rtp_time_;
	}

	StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
	: capacity_(capacity), prev_arrival_time_us_(time_now) {}

	StreamGenerator::~StreamGenerator() = default;

	// Add a new stream.
	void StreamGenerator::AddStream(RtpStream* stream) {
	streams_.push_back(std::unique_ptr<RtpStream>(stream));
	}

	// Set the link capacity.
	void StreamGenerator::set_capacity_bps(int capacity_bps) {
	ASSERT_GT(capacity_bps, 0);
	capacity_ = capacity_bps;
	}

	// Divides `bitrate_bps` among all streams. The allocated bitrate per stream
	// is decided by the current allocation ratios.
	void StreamGenerator::SetBitrateBps(int bitrate_bps) {
	ASSERT_GE(streams_.size(), 0u);
	int total_bitrate_before = 0;
	for (const auto& stream : streams_) {
	total_bitrate_before += stream->bitrate_bps();
	}
	int64_t bitrate_before = 0;
	int total_bitrate_after = 0;
	for (const auto& stream : streams_) {
	bitrate_before += stream->bitrate_bps();
	int64_t bitrate_after =
	(bitrate_before * bitrate_bps + total_bitrate_before / 2) /
	total_bitrate_before;
	stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
	total_bitrate_after += stream->bitrate_bps();
	}
	ASSERT_EQ(bitrate_before, total_bitrate_before);
	EXPECT_EQ(total_bitrate_after, bitrate_bps);
	}

	// TODO(holmer): Break out the channel simulation part from this class to make
	// it possible to simulate different types of channels.
	int64_t StreamGenerator::GenerateFrame(std::vector<PacketResult>* packets,
	int64_t time_now_us) {
	RTC_CHECK(packets != NULL);
	RTC_CHECK(packets->empty());
	RTC_CHECK_GT(capacity_, 0);
	auto it =
	std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
	(*it)->GenerateFrame(time_now_us, packets);
	for (PacketResult& packet : *packets) {
	int capacity_bpus = capacity_ / 1000;
	int64_t required_network_time_us =
	(8 * 1000 * packet.sent_packet.size.bytes() + capacity_bpus / 2) /
	capacity_bpus;
	prev_arrival_time_us_ =
	std::max(time_now_us + required_network_time_us,
	prev_arrival_time_us_ + required_network_time_us);
	packet.receive_time = Timestamp::Micros(prev_arrival_time_us_);
	}
	it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
	return std::max((*it)->next_rtp_time(), time_now_us);
	}
	} // namespace test

	DelayBasedBweTest::DelayBasedBweTest()
	: field_trial(std::make_unique<test::ScopedFieldTrials>(
	"WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/")),
	clock_(100000000),
	acknowledged_bitrate_estimator_(
	AcknowledgedBitrateEstimatorInterface::Create(&field_trial_config_)),
	probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
	bitrate_estimator_(
	new DelayBasedBwe(&field_trial_config_, nullptr, nullptr)),
	stream_generator_(new test::StreamGenerator(1e6, // Capacity.
	clock_.TimeInMicroseconds())),
	arrival_time_offset_ms_(0),
	next_sequence_number_(0),
	first_update_(true) {}

	DelayBasedBweTest::~DelayBasedBweTest() {}

	void DelayBasedBweTest::AddDefaultStream() {
	stream_generator_->AddStream(new test::RtpStream(30, 3e5));
	}

	const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;

	void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
	int64_t send_time_ms,
	size_t payload_size) {
	IncomingFeedback(arrival_time_ms, send_time_ms, payload_size,
	PacedPacketInfo());
	}

	void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
	int64_t send_time_ms,
	size_t payload_size,
	const PacedPacketInfo& pacing_info) {
	RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
	IncomingFeedback(Timestamp::Millis(arrival_time_ms + arrival_time_offset_ms_),
	Timestamp::Millis(send_time_ms), payload_size, pacing_info);
	}

	void DelayBasedBweTest::IncomingFeedback(Timestamp receive_time,
	Timestamp send_time,
	size_t payload_size,
	const PacedPacketInfo& pacing_info) {
	PacketResult packet;
	packet.receive_time = receive_time;
	packet.sent_packet.send_time = send_time;
	packet.sent_packet.size = DataSize::Bytes(payload_size);
	packet.sent_packet.pacing_info = pacing_info;
	packet.sent_packet.sequence_number = next_sequence_number_++;
	if (packet.sent_packet.pacing_info.probe_cluster_id !=
	PacedPacketInfo::kNotAProbe)
	probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);

	TransportPacketsFeedback msg;
	msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
	msg.packet_feedbacks.push_back(packet);
	acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
	msg.SortedByReceiveTime());
	DelayBasedBwe::Result result =
	bitrate_estimator_->IncomingPacketFeedbackVector(
	msg, acknowledged_bitrate_estimator_->bitrate(),
	probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
	/network_estimate/ absl::nullopt, /in_alr/ false);
	if (result.updated) {
	bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
	}
	}

	// Generates a frame of packets belonging to a stream at a given bitrate and
	// with a given ssrc. The stream is pushed through a very simple simulated
	// network, and is then given to the receive-side bandwidth estimator.
	// Returns true if an over-use was seen, false otherwise.
	// The StreamGenerator::updated() should be used to check for any changes in
	// target bitrate after the call to this function.
	bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
	uint32_t bitrate_bps) {
	stream_generator_->SetBitrateBps(bitrate_bps);
	std::vector<PacketResult> packets;

	int64_t next_time_us =
	stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
	if (packets.empty())
	return false;

	bool overuse = false;
	bitrate_observer_.Reset();
	clock_.AdvanceTimeMicroseconds(packets.back().receive_time.us() -
	clock_.TimeInMicroseconds());
	for (auto& packet : packets) {
	RTC_CHECK_GE(packet.receive_time.ms() + arrival_time_offset_ms_, 0);
	packet.receive_time += TimeDelta::Millis(arrival_time_offset_ms_);

	if (packet.sent_packet.pacing_info.probe_cluster_id !=
	PacedPacketInfo::kNotAProbe)
	probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
	}

	acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
	TransportPacketsFeedback msg;
	msg.packet_feedbacks = packets;
	msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());

	DelayBasedBwe::Result result =
	bitrate_estimator_->IncomingPacketFeedbackVector(
	msg, acknowledged_bitrate_estimator_->bitrate(),
	probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
	/network_estimate/ absl::nullopt, /in_alr/ false);
	if (result.updated) {
	bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
	if (!first_update_ && result.target_bitrate.bps() < bitrate_bps)
	overuse = true;
	first_update_ = false;
	}

	clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
	return overuse;
	}

	// Run the bandwidth estimator with a stream of `number_of_frames` frames, or
	// until it reaches `target_bitrate`.
	// Can for instance be used to run the estimator for some time to get it
	// into a steady state.
	uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
	int max_number_of_frames,
	uint32_t start_bitrate,
	uint32_t min_bitrate,
	uint32_t max_bitrate,
	uint32_t target_bitrate) {
	uint32_t bitrate_bps = start_bitrate;
	bool bitrate_update_seen = false;
	// Produce `number_of_frames` frames and give them to the estimator.
	for (int i = 0; i < max_number_of_frames; ++i) {
	bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
	if (overuse) {
	EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate);
	EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate);
	bitrate_bps = bitrate_observer_.latest_bitrate();
	bitrate_update_seen = true;
	} else if (bitrate_observer_.updated()) {
	bitrate_bps = bitrate_observer_.latest_bitrate();
	bitrate_observer_.Reset();
	}
	if (bitrate_update_seen && bitrate_bps > target_bitrate) {
	break;
	}
	}
	EXPECT_TRUE(bitrate_update_seen);
	return bitrate_bps;
	}

	void DelayBasedBweTest::InitialBehaviorTestHelper(
	uint32_t expected_converge_bitrate) {
	const int kFramerate = 50; // 50 fps to avoid rounding errors.
	const int kFrameIntervalMs = 1000 / kFramerate;
	const PacedPacketInfo kPacingInfo(0, 5, 5000);
	DataRate bitrate = DataRate::Zero();
	int64_t send_time_ms = 0;
	std::vector<uint32_t> ssrcs;
	EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
	EXPECT_EQ(0u, ssrcs.size());
	clock_.AdvanceTimeMilliseconds(1000);
	EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
	EXPECT_FALSE(bitrate_observer_.updated());
	bitrate_observer_.Reset();
	clock_.AdvanceTimeMilliseconds(1000);
	// Inserting packets for 5 seconds to get a valid estimate.
	for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
	// NOTE!!! If the following line is moved under the if case then this test
	// wont work on windows realease bots.
	PacedPacketInfo pacing_info =
	i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();

	if (i == kNumInitialPackets) {
	EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
	EXPECT_EQ(0u, ssrcs.size());
	EXPECT_FALSE(bitrate_observer_.updated());
	bitrate_observer_.Reset();
	}
	IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
	pacing_info);
	clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
	send_time_ms += kFrameIntervalMs;
	}
	EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
	ASSERT_EQ(1u, ssrcs.size());
	EXPECT_EQ(kDefaultSsrc, ssrcs.front());
	EXPECT_NEAR(expected_converge_bitrate, bitrate.bps(),
	kAcceptedBitrateErrorBps);
	EXPECT_TRUE(bitrate_observer_.updated());
	bitrate_observer_.Reset();
	EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate.bps());
	}

	void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
	uint32_t expected_bitrate_bps) {
	const int kFramerate = 50; // 50 fps to avoid rounding errors.
	const int kFrameIntervalMs = 1000 / kFramerate;
	const PacedPacketInfo kPacingInfo(0, 5, 5000);
	int64_t send_time_ms = 0;
	// Inserting packets for five seconds to get a valid estimate.
	for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
	// NOTE!!! If the following line is moved under the if case then this test
	// wont work on windows realease bots.
	PacedPacketInfo pacing_info =
	i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();

	// TODO(sprang): Remove this hack once the single stream estimator is gone,
	// as it doesn't do anything in Process().
	if (i == kNumInitialPackets) {
	// Process after we have enough frames to get a valid input rate estimate.

	EXPECT_FALSE(bitrate_observer_.updated()); // No valid estimate.
	}
	IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
	pacing_info);
	clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
	send_time_ms += kFrameIntervalMs;
	}
	EXPECT_TRUE(bitrate_observer_.updated());
	EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
	kAcceptedBitrateErrorBps);
	for (int i = 0; i < 10; ++i) {
	clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
	send_time_ms += 2 * kFrameIntervalMs;
	IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
	IncomingFeedback(clock_.TimeInMilliseconds(),
	send_time_ms - kFrameIntervalMs, 1000);
	}
	EXPECT_TRUE(bitrate_observer_.updated());
	EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
	kAcceptedBitrateErrorBps);
	}

	// Make sure we initially increase the bitrate as expected.
	void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
	int expected_iterations) {
	// This threshold corresponds approximately to increasing linearly with
	// bitrate(i) = 1.04 * bitrate(i-1) + 1000
	// until bitrate(i) > 500000, with bitrate(1) ~= 30000.
	uint32_t bitrate_bps = 30000;
	int iterations = 0;
	AddDefaultStream();
	// Feed the estimator with a stream of packets and verify that it reaches
	// 500 kbps at the expected time.
	while (bitrate_bps < 5e5) {
	bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
	if (overuse) {
	EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
	bitrate_bps = bitrate_observer_.latest_bitrate();
	bitrate_observer_.Reset();
	} else if (bitrate_observer_.updated()) {
	bitrate_bps = bitrate_observer_.latest_bitrate();
	bitrate_observer_.Reset();
	}
	++iterations;
	}
	ASSERT_EQ(expected_iterations, iterations);
	}

	void DelayBasedBweTest::CapacityDropTestHelper(
	int number_of_streams,
	bool wrap_time_stamp,
	uint32_t expected_bitrate_drop_delta,
	int64_t receiver_clock_offset_change_ms) {
	const int kFramerate = 30;
	const int kStartBitrate = 900e3;
	const int kMinExpectedBitrate = 800e3;
	const int kMaxExpectedBitrate = 1100e3;
	const uint32_t kInitialCapacityBps = 1000e3;
	const uint32_t kReducedCapacityBps = 500e3;

	int steady_state_time = 0;
	if (number_of_streams <= 1) {
	steady_state_time = 10;
	AddDefaultStream();
	} else {
	steady_state_time = 10 * number_of_streams;
	int bitrate_sum = 0;
	int kBitrateDenom = number_of_streams * (number_of_streams - 1);
	for (int i = 0; i < number_of_streams; i++) {
	// First stream gets half available bitrate, while the rest share the
	// remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
	int bitrate = kStartBitrate / 2;
	if (i > 0) {
	bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
	}
	stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
	bitrate_sum += bitrate;
	}
	ASSERT_EQ(bitrate_sum, kStartBitrate);
	}

	// Run in steady state to make the estimator converge.
	stream_generator_->set_capacity_bps(kInitialCapacityBps);
	uint32_t bitrate_bps = SteadyStateRun(
	kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
	kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
	EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u);
	bitrate_observer_.Reset();

	// Add an offset to make sure the BWE can handle it.
	arrival_time_offset_ms_ += receiver_clock_offset_change_ms;

	// Reduce the capacity and verify the decrease time.
	stream_generator_->set_capacity_bps(kReducedCapacityBps);
	int64_t overuse_start_time = clock_.TimeInMilliseconds();
	int64_t bitrate_drop_time = -1;
	for (int i = 0; i < 100 * number_of_streams; ++i) {
	GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
	if (bitrate_drop_time == -1 &&
	bitrate_observer_.latest_bitrate() <= kReducedCapacityBps) {
	bitrate_drop_time = clock_.TimeInMilliseconds();
	}
	if (bitrate_observer_.updated())
	bitrate_bps = bitrate_observer_.latest_bitrate();
	}

	EXPECT_NEAR(expected_bitrate_drop_delta,
	bitrate_drop_time - overuse_start_time, 33);
	}

	void DelayBasedBweTest::TestTimestampGroupingTestHelper() {
	const int kFramerate = 50; // 50 fps to avoid rounding errors.
	const int kFrameIntervalMs = 1000 / kFramerate;
	int64_t send_time_ms = 0;
	// Initial set of frames to increase the bitrate. 6 seconds to have enough
	// time for the first estimate to be generated and for Process() to be called.
	for (int i = 0; i <= 6 * kFramerate; ++i) {
	IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);

	clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
	send_time_ms += kFrameIntervalMs;
	}
	EXPECT_TRUE(bitrate_observer_.updated());
	EXPECT_GE(bitrate_observer_.latest_bitrate(), 400000u);

	// Insert batches of frames which were sent very close in time. Also simulate
	// capacity over-use to see that we back off correctly.
	const int kTimestampGroupLength = 15;
	for (int i = 0; i < 100; ++i) {
	for (int j = 0; j < kTimestampGroupLength; ++j) {
	// Insert `kTimestampGroupLength` frames with just 1 timestamp ticks in
	// between. Should be treated as part of the same group by the estimator.
	IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 100);
	clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
	send_time_ms += 1;
	}
	// Increase time until next batch to simulate over-use.
	clock_.AdvanceTimeMilliseconds(10);
	send_time_ms += kFrameIntervalMs - kTimestampGroupLength;
	}
	EXPECT_TRUE(bitrate_observer_.updated());
	// Should have reduced the estimate.
	EXPECT_LT(bitrate_observer_.latest_bitrate(), 400000u);
	}

	void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
	const int kFramerate = 100;
	const int kFrameIntervalMs = 1000 / kFramerate;
	int64_t send_time_ms = 0;

	for (size_t i = 0; i < 3000; ++i) {
	IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
	clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
	send_time_ms += kFrameIntervalMs;
	}
	DataRate bitrate_before = DataRate::Zero();
	std::vector<uint32_t> ssrcs;
	bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);

	clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
	send_time_ms += silence_time_s * 1000;

	for (size_t i = 0; i < 24; ++i) {
	IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
	clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
	send_time_ms += kFrameIntervalMs;
	}
	DataRate bitrate_after = DataRate::Zero();
	bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
	EXPECT_LT(bitrate_after, bitrate_before);
	}
	} // namespace webrtc