text: /*
   *  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.
   */
  
  #include <algorithm>
  #include <limits>
  #include <memory>
  #include <string>
  
  #include "api/audio_codecs/builtin_audio_encoder_factory.h"
  #include "api/rtc_event_log/rtc_event_log.h"
  #include "api/task_queue/task_queue_base.h"
  #include "api/test/simulated_network.h"
  #include "api/video/builtin_video_bitrate_allocator_factory.h"
  #include "api/video/video_bitrate_allocation.h"
  #include "api/video_codecs/video_encoder.h"
  #include "api/video_codecs/video_encoder_config.h"
  #include "call/call.h"
  #include "call/fake_network_pipe.h"
  #include "call/simulated_network.h"
  #include "modules/audio_coding/include/audio_coding_module.h"
  #include "modules/audio_device/include/test_audio_device.h"
  #include "modules/audio_mixer/audio_mixer_impl.h"
  #include "rtc_base/checks.h"
  #include "rtc_base/task_queue_for_test.h"
  #include "rtc_base/thread.h"
  #include "rtc_base/thread_annotations.h"
  #include "system_wrappers/include/metrics.h"
  #include "test/call_test.h"
  #include "test/direct_transport.h"
  #include "test/drifting_clock.h"
  #include "test/encoder_settings.h"
  #include "test/fake_encoder.h"
  #include "test/field_trial.h"
  #include "test/frame_generator_capturer.h"
  #include "test/gtest.h"
  #include "test/null_transport.h"
  #include "test/rtp_header_parser.h"
  #include "test/rtp_rtcp_observer.h"
  #include "test/testsupport/file_utils.h"
  #include "test/testsupport/perf_test.h"
  #include "test/video_encoder_proxy_factory.h"
  #include "video/transport_adapter.h"
  
  using webrtc::test::DriftingClock;
  
  namespace webrtc {
  namespace {
  enum : int {  // The first valid value is 1.
    kTransportSequenceNumberExtensionId = 1,
  };
  }  // namespace
  
  class CallPerfTest : public test::CallTest {
   public:
    CallPerfTest() {
      RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                                        kTransportSequenceNumberExtensionId));
    }
  
   protected:
    enum class FecMode { kOn, kOff };
    enum class CreateOrder { kAudioFirst, kVideoFirst };
    void TestAudioVideoSync(FecMode fec,
                            CreateOrder create_first,
                            float video_ntp_speed,
                            float video_rtp_speed,
                            float audio_rtp_speed,
                            const std::string& test_label);
  
    void TestMinTransmitBitrate(bool pad_to_min_bitrate);
  
    void TestCaptureNtpTime(const BuiltInNetworkBehaviorConfig& net_config,
                            int threshold_ms,
                            int start_time_ms,
                            int run_time_ms);
    void TestMinAudioVideoBitrate(int test_bitrate_from,
                                  int test_bitrate_to,
                                  int test_bitrate_step,
                                  int min_bwe,
                                  int start_bwe,
                                  int max_bwe);
  };
  
  class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
                                   public rtc::VideoSinkInterface<VideoFrame> {
    static const int kInSyncThresholdMs = 50;
    static const int kStartupTimeMs = 2000;
    static const int kMinRunTimeMs = 30000;
  
   public:
    explicit VideoRtcpAndSyncObserver(Clock* clock, const std::string& test_label)
        : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
          clock_(clock),
          test_label_(test_label),
          creation_time_ms_(clock_->TimeInMilliseconds()),
          first_time_in_sync_(-1),
          receive_stream_(nullptr) {}
  
    void OnFrame(const VideoFrame& video_frame) override {
      VideoReceiveStream::Stats stats;
      {
        rtc::CritScope lock(&crit_);
        if (receive_stream_)
          stats = receive_stream_->GetStats();
      }
      if (stats.sync_offset_ms == std::numeric_limits<int>::max())
        return;
  
      int64_t now_ms = clock_->TimeInMilliseconds();
      int64_t time_since_creation = now_ms - creation_time_ms_;
      // During the first couple of seconds audio and video can falsely be
      // estimated as being synchronized. We don't want to trigger on those.
      if (time_since_creation < kStartupTimeMs)
        return;
      if (std::abs(stats.sync_offset_ms) < kInSyncThresholdMs) {
        if (first_time_in_sync_ == -1) {
          first_time_in_sync_ = now_ms;
          webrtc::test::PrintResult("sync_convergence_time", test_label_,
                                    "synchronization", time_since_creation, "ms",
                                    false);
        }
        if (time_since_creation > kMinRunTimeMs)
          observation_complete_.Set();
      }
      if (first_time_in_sync_ != -1)
        sync_offset_ms_list_.push_back(stats.sync_offset_ms);
    }
  
    void set_receive_stream(VideoReceiveStream* receive_stream) {
      rtc::CritScope lock(&crit_);
      receive_stream_ = receive_stream;
    }
  
    void PrintResults() {
      test::PrintResultList("stream_offset", test_label_, "synchronization",
                            sync_offset_ms_list_, "ms", false);
    }
  
   private:
    Clock* const clock_;
    std::string test_label_;
    const int64_t creation_time_ms_;
    int64_t first_time_in_sync_;
    rtc::CriticalSection crit_;
    VideoReceiveStream* receive_stream_ RTC_GUARDED_BY(crit_);
    std::vector<double> sync_offset_ms_list_;
  };
  
  void CallPerfTest::TestAudioVideoSync(FecMode fec,
                                        CreateOrder create_first,
                                        float video_ntp_speed,
                                        float video_rtp_speed,
                                        float audio_rtp_speed,
                                        const std::string& test_label) {
    const char* kSyncGroup = "av_sync";
    const uint32_t kAudioSendSsrc = 1234;
    const uint32_t kAudioRecvSsrc = 5678;
  
    BuiltInNetworkBehaviorConfig audio_net_config;
    audio_net_config.queue_delay_ms = 500;
    audio_net_config.loss_percent = 5;
  
    VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), test_label);
  
    std::map<uint8_t, MediaType> audio_pt_map;
    std::map<uint8_t, MediaType> video_pt_map;
  
    std::unique_ptr<test::PacketTransport> audio_send_transport;
    std::unique_ptr<test::PacketTransport> video_send_transport;
    std::unique_ptr<test::PacketTransport> receive_transport;
    test::NullTransport rtcp_send_transport;
  
    AudioSendStream* audio_send_stream;
    AudioReceiveStream* audio_receive_stream;
    std::unique_ptr<DriftingClock> drifting_clock;
  
    SendTask(RTC_FROM_HERE, task_queue(), [&]() {
      metrics::Reset();
      rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device =
          TestAudioDeviceModule::Create(
              task_queue_factory_.get(),
              TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000),
              TestAudioDeviceModule::CreateDiscardRenderer(48000),
              audio_rtp_speed);
      EXPECT_EQ(0, fake_audio_device->Init());
  
      AudioState::Config send_audio_state_config;
      send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
      send_audio_state_config.audio_processing =
          AudioProcessingBuilder().Create();
      send_audio_state_config.audio_device_module = fake_audio_device;
      Call::Config sender_config(send_event_log_.get());
  
      auto audio_state = AudioState::Create(send_audio_state_config);
      fake_audio_device->RegisterAudioCallback(audio_state->audio_transport());
      sender_config.audio_state = audio_state;
      Call::Config receiver_config(recv_event_log_.get());
      receiver_config.audio_state = audio_state;
      CreateCalls(sender_config, receiver_config);
  
      std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
                   std::inserter(audio_pt_map, audio_pt_map.end()),
                   [](const std::pair<const uint8_t, MediaType>& pair) {
                     return pair.second == MediaType::AUDIO;
                   });
      std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
                   std::inserter(video_pt_map, video_pt_map.end()),
                   [](const std::pair<const uint8_t, MediaType>& pair) {
                     return pair.second == MediaType::VIDEO;
                   });
  
      audio_send_transport = std::make_unique<test::PacketTransport>(
          task_queue(), sender_call_.get(), &observer,
          test::PacketTransport::kSender, audio_pt_map,
          std::make_unique<FakeNetworkPipe>(
              Clock::GetRealTimeClock(),
              std::make_unique<SimulatedNetwork>(audio_net_config)));
      audio_send_transport->SetReceiver(receiver_call_->Receiver());
  
      video_send_transport = std::make_unique<test::PacketTransport>(
          task_queue(), sender_call_.get(), &observer,
          test::PacketTransport::kSender, video_pt_map,
          std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
                                            std::make_unique<SimulatedNetwork>(
                                                BuiltInNetworkBehaviorConfig())));
      video_send_transport->SetReceiver(receiver_call_->Receiver());
  
      receive_transport = std::make_unique<test::PacketTransport>(
          task_queue(), receiver_call_.get(), &observer,
          test::PacketTransport::kReceiver, payload_type_map_,
          std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
                                            std::make_unique<SimulatedNetwork>(
                                                BuiltInNetworkBehaviorConfig())));
      receive_transport->SetReceiver(sender_call_->Receiver());
  
      CreateSendConfig(1, 0, 0, video_send_transport.get());
      CreateMatchingReceiveConfigs(receive_transport.get());
  
      AudioSendStream::Config audio_send_config(audio_send_transport.get());
      audio_send_config.rtp.ssrc = kAudioSendSsrc;
      audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
          kAudioSendPayloadType, {"ISAC", 16000, 1});
      audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
      audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config);
  
      GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
      if (fec == FecMode::kOn) {
        GetVideoSendConfig()->rtp.ulpfec.red_payload_type = kRedPayloadType;
        GetVideoSendConfig()->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
        video_receive_configs_[0].rtp.red_payload_type = kRedPayloadType;
        video_receive_configs_[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
      }
      video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
      video_receive_configs_[0].renderer = &observer;
      video_receive_configs_[0].sync_group = kSyncGroup;
  
      AudioReceiveStream::Config audio_recv_config;
      audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
      audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
      audio_recv_config.rtcp_send_transport = &rtcp_send_transport;
      audio_recv_config.sync_group = kSyncGroup;
      audio_recv_config.decoder_factory = audio_decoder_factory_;
      audio_recv_config.decoder_map = {
          {kAudioSendPayloadType, {"ISAC", 16000, 1}}};
  
      if (create_first == CreateOrder::kAudioFirst) {
        audio_receive_stream =
            receiver_call_->CreateAudioReceiveStream(audio_recv_config);
        CreateVideoStreams();
      } else {
        CreateVideoStreams();
        audio_receive_stream =
            receiver_call_->CreateAudioReceiveStream(audio_recv_config);
      }
      EXPECT_EQ(1u, video_receive_streams_.size());
      observer.set_receive_stream(video_receive_streams_[0]);
      drifting_clock = std::make_unique<DriftingClock>(clock_, video_ntp_speed);
      CreateFrameGeneratorCapturerWithDrift(drifting_clock.get(), video_rtp_speed,
                                            kDefaultFramerate, kDefaultWidth,
                                            kDefaultHeight);
  
      Start();
  
      audio_send_stream->Start();
      audio_receive_stream->Start();
    });
  
    EXPECT_TRUE(observer.Wait())
        << "Timed out while waiting for audio and video to be synchronized.";
  
    SendTask(RTC_FROM_HERE, task_queue(), [&]() {
      audio_send_stream->Stop();
      audio_receive_stream->Stop();
  
      Stop();
  
      DestroyStreams();
  
      video_send_transport.reset();
      audio_send_transport.reset();
      receive_transport.reset();
  
      sender_call_->DestroyAudioSendStream(audio_send_stream);
      receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
  
      DestroyCalls();
    });
  
    observer.PrintResults();
  
    // In quick test synchronization may not be achieved in time.
    if (!field_trial::IsEnabled("WebRTC-QuickPerfTest")) {
  // TODO(bugs.webrtc.org/10417): Reenable this for iOS
  #if !defined(WEBRTC_IOS)
      EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
  #endif
    }
  }
  
  TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithoutClockDrift) {
    TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
                       DriftingClock::kNoDrift, DriftingClock::kNoDrift,
                       DriftingClock::kNoDrift, "_video_no_drift");
  }
  
  TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
    TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
                       DriftingClock::PercentsFaster(10.0f),
                       DriftingClock::kNoDrift, DriftingClock::kNoDrift,
                       "_video_ntp_drift");
  }
  
  TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
    TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
                       DriftingClock::kNoDrift,
                       DriftingClock::PercentsSlower(30.0f),
                       DriftingClock::PercentsFaster(30.0f), "_audio_faster");
  }
  
  TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
    TestAudioVideoSync(FecMode::kOn, CreateOrder::kVideoFirst,
                       DriftingClock::kNoDrift,
                       DriftingClock::PercentsFaster(30.0f),
                       DriftingClock::PercentsSlower(30.0f), "_video_faster");
  }
  
  void CallPerfTest::TestCaptureNtpTime(
      const BuiltInNetworkBehaviorConfig& net_config,
      int threshold_ms,
      int start_time_ms,
      int run_time_ms) {
    class CaptureNtpTimeObserver : public test::EndToEndTest,
                                   public rtc::VideoSinkInterface<VideoFrame> {
     public:
      CaptureNtpTimeObserver(const BuiltInNetworkBehaviorConfig& net_config,
                             int threshold_ms,
                             int start_time_ms,
                             int run_time_ms)
          : EndToEndTest(kLongTimeoutMs),
            net_config_(net_config),
            clock_(Clock::GetRealTimeClock()),
            threshold_ms_(threshold_ms),
            start_time_ms_(start_time_ms),
            run_time_ms_(run_time_ms),
            creation_time_ms_(clock_->TimeInMilliseconds()),
            capturer_(nullptr),
            rtp_start_timestamp_set_(false),
            rtp_start_timestamp_(0) {}
  
     private:
      std::unique_ptr<test::PacketTransport> CreateSendTransport(
          TaskQueueBase* task_queue,
          Call* sender_call) override {
        return std::make_unique<test::PacketTransport>(
            task_queue, sender_call, this, test::PacketTransport::kSender,
            payload_type_map_,
            std::make_unique<FakeNetworkPipe>(
                Clock::GetRealTimeClock(),
                std::make_unique<SimulatedNetwork>(net_config_)));
      }
  
      std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
          TaskQueueBase* task_queue) override {
        return std::make_unique<test::PacketTransport>(
            task_queue, nullptr, this, test::PacketTransport::kReceiver,
            payload_type_map_,
            std::make_unique<FakeNetworkPipe>(
                Clock::GetRealTimeClock(),
                std::make_unique<SimulatedNetwork>(net_config_)));
      }
  
      void OnFrame(const VideoFrame& video_frame) override {
        rtc::CritScope lock(&crit_);
        if (video_frame.ntp_time_ms() <= 0) {
          // Haven't got enough RTCP SR in order to calculate the capture ntp
          // time.
          return;
        }
  
        int64_t now_ms = clock_->TimeInMilliseconds();
        int64_t time_since_creation = now_ms - creation_time_ms_;
        if (time_since_creation < start_time_ms_) {
          // Wait for |start_time_ms_| before start measuring.
          return;
        }
  
        if (time_since_creation > run_time_ms_) {
          observation_complete_.Set();
        }
  
        FrameCaptureTimeList::iterator iter =
            capture_time_list_.find(video_frame.timestamp());
        EXPECT_TRUE(iter != capture_time_list_.end());
  
        // The real capture time has been wrapped to uint32_t before converted
        // to rtp timestamp in the sender side. So here we convert the estimated
        // capture time to a uint32_t 90k timestamp also for comparing.
        uint32_t estimated_capture_timestamp =
            90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
        uint32_t real_capture_timestamp = iter->second;
        int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
        time_offset_ms = time_offset_ms / 90;
        time_offset_ms_list_.push_back(time_offset_ms);
  
        EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
      }
  
      Action OnSendRtp(const uint8_t* packet, size_t length) override {
        rtc::CritScope lock(&crit_);
        RTPHeader header;
        EXPECT_TRUE(parser_->Parse(packet, length, &header));
  
        if (!rtp_start_timestamp_set_) {
          // Calculate the rtp timestamp offset in order to calculate the real
          // capture time.
          uint32_t first_capture_timestamp =
              90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
          rtp_start_timestamp_ = header.timestamp - first_capture_timestamp;
          rtp_start_timestamp_set_ = true;
        }
  
        uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_;
        capture_time_list_.insert(
            capture_time_list_.end(),
            std::make_pair(header.timestamp, capture_timestamp));
        return SEND_PACKET;
      }
  
      void OnFrameGeneratorCapturerCreated(
          test::FrameGeneratorCapturer* frame_generator_capturer) override {
        capturer_ = frame_generator_capturer;
      }
  
      void ModifyVideoConfigs(
          VideoSendStream::Config* send_config,
          std::vector<VideoReceiveStream::Config>* receive_configs,
          VideoEncoderConfig* encoder_config) override {
        (*receive_configs)[0].renderer = this;
        // Enable the receiver side rtt calculation.
        (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
      }
  
      void PerformTest() override {
        EXPECT_TRUE(Wait()) << "Timed out while waiting for "
                               "estimated capture NTP time to be "
                               "within bounds.";
        test::PrintResultList("capture_ntp_time", "", "real - estimated",
                              time_offset_ms_list_, "ms", true);
      }
  
      rtc::CriticalSection crit_;
      const BuiltInNetworkBehaviorConfig net_config_;
      Clock* const clock_;
      int threshold_ms_;
      int start_time_ms_;
      int run_time_ms_;
      int64_t creation_time_ms_;
      test::FrameGeneratorCapturer* capturer_;
      bool rtp_start_timestamp_set_;
      uint32_t rtp_start_timestamp_;
      typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
      FrameCaptureTimeList capture_time_list_ RTC_GUARDED_BY(&crit_);
      std::vector<double> time_offset_ms_list_;
    } test(net_config, threshold_ms, start_time_ms, run_time_ms);
  
    RunBaseTest(&test);
  }
  
  // Flaky tests, disabled on Mac and Windows due to webrtc:8291.
  #if !(defined(WEBRTC_MAC) || defined(WEBRTC_WIN))
  TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
    BuiltInNetworkBehaviorConfig net_config;
    net_config.queue_delay_ms = 100;
    // TODO(wu): lower the threshold as the calculation/estimatation becomes more
    // accurate.
    const int kThresholdMs = 100;
    const int kStartTimeMs = 10000;
    const int kRunTimeMs = 20000;
    TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
  }
  
  TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
    BuiltInNetworkBehaviorConfig net_config;
    net_config.queue_delay_ms = 100;
    net_config.delay_standard_deviation_ms = 10;
    // TODO(wu): lower the threshold as the calculation/estimatation becomes more
    // accurate.
    const int kThresholdMs = 100;
    const int kStartTimeMs = 10000;
    const int kRunTimeMs = 20000;
    TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
  }
  #endif
  
  TEST_F(CallPerfTest, ReceivesCpuOveruseAndUnderuse) {
    // Minimal normal usage at the start, then 30s overuse to allow filter to
    // settle, and then 80s underuse to allow plenty of time for rampup again.
    test::ScopedFieldTrials fake_overuse_settings(
        "WebRTC-ForceSimulatedOveruseIntervalMs/1-30000-80000/");
  
    class LoadObserver : public test::SendTest,
                         public test::FrameGeneratorCapturer::SinkWantsObserver {
     public:
      LoadObserver() : SendTest(kLongTimeoutMs), test_phase_(TestPhase::kInit) {}
  
      void OnFrameGeneratorCapturerCreated(
          test::FrameGeneratorCapturer* frame_generator_capturer) override {
        frame_generator_capturer->SetSinkWantsObserver(this);
        // Set a high initial resolution to be sure that we can scale down.
        frame_generator_capturer->ChangeResolution(1920, 1080);
      }
  
      // OnSinkWantsChanged is called when FrameGeneratorCapturer::AddOrUpdateSink
      // is called.
      // TODO(sprang): Add integration test for maintain-framerate mode?
      void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
                              const rtc::VideoSinkWants& wants) override {
        // At kStart expect CPU overuse. Then expect CPU underuse when the encoder
        // delay has been decreased.
        switch (test_phase_) {
          case TestPhase::kInit:
            // Max framerate should be set initially.
            if (wants.max_framerate_fps != std::numeric_limits<int>::max() &&
                wants.max_pixel_count == std::numeric_limits<int>::max()) {
              test_phase_ = TestPhase::kStart;
            } else {
              ADD_FAILURE() << "Got unexpected adaptation request, max res = "
                            << wants.max_pixel_count << ", target res = "
                            << wants.target_pixel_count.value_or(-1)
                            << ", max fps = " << wants.max_framerate_fps;
            }
            break;
          case TestPhase::kStart:
            if (wants.max_pixel_count < std::numeric_limits<int>::max()) {
              // On adapting down, VideoStreamEncoder::VideoSourceProxy will set
              // only the max pixel count, leaving the target unset.
              test_phase_ = TestPhase::kAdaptedDown;
            } else {
              ADD_FAILURE() << "Got unexpected adaptation request, max res = "
                            << wants.max_pixel_count << ", target res = "
                            << wants.target_pixel_count.value_or(-1)
                            << ", max fps = " << wants.max_framerate_fps;
            }
            break;
          case TestPhase::kAdaptedDown:
            // On adapting up, the adaptation counter will again be at zero, and
            // so all constraints will be reset.
            if (wants.max_pixel_count == std::numeric_limits<int>::max() &&
                !wants.target_pixel_count) {
              test_phase_ = TestPhase::kAdaptedUp;
              observation_complete_.Set();
            } else {
              ADD_FAILURE() << "Got unexpected adaptation request, max res = "
                            << wants.max_pixel_count << ", target res = "
                            << wants.target_pixel_count.value_or(-1)
                            << ", max fps = " << wants.max_framerate_fps;
            }
            break;
          case TestPhase::kAdaptedUp:
            ADD_FAILURE() << "Got unexpected adaptation request, max res = "
                          << wants.max_pixel_count << ", target res = "
                          << wants.target_pixel_count.value_or(-1)
                          << ", max fps = " << wants.max_framerate_fps;
        }
      }
  
      void ModifyVideoConfigs(
          VideoSendStream::Config* send_config,
          std::vector<VideoReceiveStream::Config>* receive_configs,
          VideoEncoderConfig* encoder_config) override {}
  
      void PerformTest() override {
        EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
      }
  
      enum class TestPhase {
        kInit,
        kStart,
        kAdaptedDown,
        kAdaptedUp
      } test_phase_;
    } test;
  
    RunBaseTest(&test);
  }
  
  void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
    static const int kMaxEncodeBitrateKbps = 30;
    static const int kMinTransmitBitrateBps = 150000;
    static const int kMinAcceptableTransmitBitrate = 130;
    static const int kMaxAcceptableTransmitBitrate = 170;
    static const int kNumBitrateObservationsInRange = 100;
    static const int kAcceptableBitrateErrorMargin = 15;  // +- 7
    class BitrateObserver : public test::EndToEndTest {
     public:
      explicit BitrateObserver(bool using_min_transmit_bitrate)
          : EndToEndTest(kLongTimeoutMs),
            send_stream_(nullptr),
            converged_(false),
            pad_to_min_bitrate_(using_min_transmit_bitrate),
            min_acceptable_bitrate_(using_min_transmit_bitrate
                                        ? kMinAcceptableTransmitBitrate
                                        : (kMaxEncodeBitrateKbps -
                                           kAcceptableBitrateErrorMargin / 2)),
            max_acceptable_bitrate_(using_min_transmit_bitrate
                                        ? kMaxAcceptableTransmitBitrate
                                        : (kMaxEncodeBitrateKbps +
                                           kAcceptableBitrateErrorMargin / 2)),
            num_bitrate_observations_in_range_(0) {}
  
     private:
      // TODO(holmer): Run this with a timer instead of once per packet.
      Action OnSendRtp(const uint8_t* packet, size_t length) override {
        VideoSendStream::Stats stats = send_stream_->GetStats();
        if (!stats.substreams.empty()) {
          RTC_DCHECK_EQ(1, stats.substreams.size());
          int bitrate_kbps =
              stats.substreams.begin()->second.total_bitrate_bps / 1000;
          if (bitrate_kbps > min_acceptable_bitrate_ &&
              bitrate_kbps < max_acceptable_bitrate_) {
            converged_ = true;
            ++num_bitrate_observations_in_range_;
            if (num_bitrate_observations_in_range_ ==
                kNumBitrateObservationsInRange)
              observation_complete_.Set();
          }
          if (converged_)
            bitrate_kbps_list_.push_back(bitrate_kbps);
        }
        return SEND_PACKET;
      }
  
      void OnVideoStreamsCreated(
          VideoSendStream* send_stream,
          const std::vector<VideoReceiveStream*>& receive_streams) override {
        send_stream_ = send_stream;
      }
  
      void ModifyVideoConfigs(
          VideoSendStream::Config* send_config,
          std::vector<VideoReceiveStream::Config>* receive_configs,
          VideoEncoderConfig* encoder_config) override {
        if (pad_to_min_bitrate_) {
          encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
        } else {
          RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
        }
      }
  
      void PerformTest() override {
        EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
        test::PrintResultList(
            "bitrate_stats_",
            (pad_to_min_bitrate_ ? "min_transmit_bitrate"
                                 : "without_min_transmit_bitrate"),
            "bitrate_kbps", bitrate_kbps_list_, "kbps", false);
      }
  
      VideoSendStream* send_stream_;
      bool converged_;
      const bool pad_to_min_bitrate_;
      const int min_acceptable_bitrate_;
      const int max_acceptable_bitrate_;
      int num_bitrate_observations_in_range_;
      std::vector<double> bitrate_kbps_list_;
    } test(pad_to_min_bitrate);
  
    fake_encoder_max_bitrate_ = kMaxEncodeBitrateKbps;
    RunBaseTest(&test);
  }
  
  TEST_F(CallPerfTest, PadsToMinTransmitBitrate) {
    TestMinTransmitBitrate(true);
  }
  
  TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) {
    TestMinTransmitBitrate(false);
  }
  
  // TODO(bugs.webrtc.org/8878)
  #if defined(WEBRTC_MAC)
  #define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
    DISABLED_KeepsHighBitrateWhenReconfiguringSender
  #else
  #define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
    KeepsHighBitrateWhenReconfiguringSender
  #endif
  TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
    static const uint32_t kInitialBitrateKbps = 400;
    static const uint32_t kReconfigureThresholdKbps = 600;
  
    class VideoStreamFactory
        : public VideoEncoderConfig::VideoStreamFactoryInterface {
     public:
      VideoStreamFactory() {}
  
     private:
      std::vector<VideoStream> CreateEncoderStreams(
          int width,
          int height,
          const VideoEncoderConfig& encoder_config) override {
        std::vector<VideoStream> streams =
            test::CreateVideoStreams(width, height, encoder_config);
        streams[0].min_bitrate_bps = 50000;
        streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000;
        return streams;
      }
    };
  
    class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder {
     public:
      BitrateObserver()
          : EndToEndTest(kDefaultTimeoutMs),
            FakeEncoder(Clock::GetRealTimeClock()),
            encoder_inits_(0),
            last_set_bitrate_kbps_(0),
            send_stream_(nullptr),
            frame_generator_(nullptr),
            encoder_factory_(this),
            bitrate_allocator_factory_(
                CreateBuiltinVideoBitrateAllocatorFactory()) {}
  
      int32_t InitEncode(const VideoCodec* config,
                         const VideoEncoder::Settings& settings) override {
        ++encoder_inits_;
        if (encoder_inits_ == 1) {
          // First time initialization. Frame size is known.
          // |expected_bitrate| is affected by bandwidth estimation before the
          // first frame arrives to the encoder.
          uint32_t expected_bitrate = last_set_bitrate_kbps_ > 0
                                          ? last_set_bitrate_kbps_
                                          : kInitialBitrateKbps;
          EXPECT_EQ(expected_bitrate, config->startBitrate)
              << "Encoder not initialized at expected bitrate.";
          EXPECT_EQ(kDefaultWidth, config->width);
          EXPECT_EQ(kDefaultHeight, config->height);
        } else if (encoder_inits_ == 2) {
          EXPECT_EQ(2 * kDefaultWidth, config->width);
          EXPECT_EQ(2 * kDefaultHeight, config->height);
          EXPECT_GE(last_set_bitrate_kbps_, kReconfigureThresholdKbps);
          EXPECT_GT(config->startBitrate, kReconfigureThresholdKbps)
              << "Encoder reconfigured with bitrate too far away from last set.";
          observation_complete_.Set();
        }
        return FakeEncoder::InitEncode(config, settings);
      }
  
      void SetRates(const RateControlParameters& parameters) override {
        last_set_bitrate_kbps_ = parameters.bitrate.get_sum_kbps();
        if (encoder_inits_ == 1 &&
            parameters.bitrate.get_sum_kbps() > kReconfigureThresholdKbps) {
          time_to_reconfigure_.Set();
        }
        FakeEncoder::SetRates(parameters);
      }
  
      void ModifySenderBitrateConfig(
          BitrateConstraints* bitrate_config) override {
        bitrate_config->start_bitrate_bps = kInitialBitrateKbps * 1000;
      }
  
      void ModifyVideoConfigs(
          VideoSendStream::Config* send_config,
          std::vector<VideoReceiveStream::Config>* receive_configs,
          VideoEncoderConfig* encoder_config) override {
        send_config->encoder_settings.encoder_factory = &encoder_factory_;
        send_config->encoder_settings.bitrate_allocator_factory =
            bitrate_allocator_factory_.get();
        encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000;
        encoder_config->video_stream_factory =
            new rtc::RefCountedObject<VideoStreamFactory>();
  
        encoder_config_ = encoder_config->Copy();
      }
  
      void OnVideoStreamsCreated(
          VideoSendStream* send_stream,
          const std::vector<VideoReceiveStream*>& receive_streams) override {
        send_stream_ = send_stream;
      }
  
      void OnFrameGeneratorCapturerCreated(
          test::FrameGeneratorCapturer* frame_generator_capturer) override {
        frame_generator_ = frame_generator_capturer;
      }
  
      void PerformTest() override {
        ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
            << "Timed out before receiving an initial high bitrate.";
        frame_generator_->ChangeResolution(kDefaultWidth * 2, kDefaultHeight * 2);
        send_stream_->ReconfigureVideoEncoder(encoder_config_.Copy());
        EXPECT_TRUE(Wait())
            << "Timed out while waiting for a couple of high bitrate estimates "
               "after reconfiguring the send stream.";
      }
  
     private:
      rtc::Event time_to_reconfigure_;
      int encoder_inits_;
      uint32_t last_set_bitrate_kbps_;
      VideoSendStream* send_stream_;
      test::FrameGeneratorCapturer* frame_generator_;
      test::VideoEncoderProxyFactory encoder_factory_;
      std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
      VideoEncoderConfig encoder_config_;
    } test;
  
    RunBaseTest(&test);
  }
  
  // Discovers the minimal supported audio+video bitrate. The test bitrate is
  // considered supported if Rtt does not go above 400ms with the network
  // contrained to the test bitrate.
  //
  // |test_bitrate_from test_bitrate_to| bitrate constraint range
  // |test_bitrate_step| bitrate constraint update step during the test
  // |min_bwe max_bwe| BWE range
  // |start_bwe| initial BWE
  void CallPerfTest::TestMinAudioVideoBitrate(int test_bitrate_from,
                                              int test_bitrate_to,
                                              int test_bitrate_step,
                                              int min_bwe,
                                              int start_bwe,
                                              int max_bwe) {
    static const std::string kAudioTrackId = "audio_track_0";
    static constexpr int kOpusBitrateFbBps = 32000;
    static constexpr int kBitrateStabilizationMs = 10000;
    static constexpr int kBitrateMeasurements = 10;
    static constexpr int kBitrateMeasurementMs = 1000;
    static constexpr int kShortDelayMs = 10;
    static constexpr int kMinGoodRttMs = 400;
  
    class MinVideoAndAudioBitrateTester : public test::EndToEndTest {
     public:
      MinVideoAndAudioBitrateTester(int test_bitrate_from,
                                    int test_bitrate_to,
                                    int test_bitrate_step,
                                    int min_bwe,
                                    int start_bwe,
                                    int max_bwe,
                                    TaskQueueBase* task_queue)
          : EndToEndTest(),
            test_bitrate_from_(test_bitrate_from),
            test_bitrate_to_(test_bitrate_to),
            test_bitrate_step_(test_bitrate_step),
            min_bwe_(min_bwe),
            start_bwe_(start_bwe),
            max_bwe_(max_bwe),
            task_queue_(task_queue) {}
  
     protected:
      BuiltInNetworkBehaviorConfig GetFakeNetworkPipeConfig() {
        BuiltInNetworkBehaviorConfig pipe_config;
        pipe_config.link_capacity_kbps = test_bitrate_from_;
        return pipe_config;
      }
  
      std::unique_ptr<test::PacketTransport> CreateSendTransport(
          TaskQueueBase* task_queue,
          Call* sender_call) override {
        auto network =
            std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig());
        send_simulated_network_ = network.get();
        return std::make_unique<test::PacketTransport>(
            task_queue, sender_call, this, test::PacketTransport::kSender,
            test::CallTest::payload_type_map_,
            std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
                                              std::move(network)));
      }
  
      std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
          TaskQueueBase* task_queue) override {
        auto network =
            std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig());
        receive_simulated_network_ = network.get();
        return std::make_unique<test::PacketTransport>(
            task_queue, nullptr, this, test::PacketTransport::kReceiver,
            test::CallTest::payload_type_map_,
            std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
                                              std::move(network)));
      }
  
      void PerformTest() override {
        // Quick test mode, just to exercise all the code paths without actually
        // caring about performance measurements.
        const bool quick_perf_test =
            field_trial::IsEnabled("WebRTC-QuickPerfTest");
        int last_passed_test_bitrate = -1;
        for (int test_bitrate = test_bitrate_from_;
             test_bitrate_from_ < test_bitrate_to_
                 ? test_bitrate <= test_bitrate_to_
                 : test_bitrate >= test_bitrate_to_;
             test_bitrate += test_bitrate_step_) {
          BuiltInNetworkBehaviorConfig pipe_config;
          pipe_config.link_capacity_kbps = test_bitrate;
          send_simulated_network_->SetConfig(pipe_config);
          receive_simulated_network_->SetConfig(pipe_config);
  
          rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs
                                               : kBitrateStabilizationMs);
  
          int64_t avg_rtt = 0;
          for (int i = 0; i < kBitrateMeasurements; i++) {
            Call::Stats call_stats;
            SendTask(RTC_FROM_HERE, task_queue_, [this, &call_stats]() {
              call_stats = sender_call_->GetStats();
            });
            avg_rtt += call_stats.rtt_ms;
            rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs
                                                 : kBitrateMeasurementMs);
          }
          avg_rtt = avg_rtt / kBitrateMeasurements;
          if (avg_rtt > kMinGoodRttMs) {
            break;
          } else {
            last_passed_test_bitrate = test_bitrate;
          }
        }
        EXPECT_GT(last_passed_test_bitrate, -1)
            << "Minimum supported bitrate out of the test scope";
        webrtc::test::PrintResult("min_test_bitrate_", "", "min_bitrate",
                                  last_passed_test_bitrate, "kbps", false);
      }
  
      void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
        sender_call_ = sender_call;
        BitrateConstraints bitrate_config;
        bitrate_config.min_bitrate_bps = min_bwe_;
        bitrate_config.start_bitrate_bps = start_bwe_;
        bitrate_config.max_bitrate_bps = max_bwe_;
        sender_call->GetTransportControllerSend()->SetSdpBitrateParameters(
            bitrate_config);
      }
  
      size_t GetNumVideoStreams() const override { return 1; }
  
      size_t GetNumAudioStreams() const override { return 1; }
  
      void ModifyAudioConfigs(
          AudioSendStream::Config* send_config,
          std::vector<AudioReceiveStream::Config>* receive_configs) override {
        send_config->send_codec_spec->target_bitrate_bps =
            absl::optional<int>(kOpusBitrateFbBps);
      }
  
     private:
      const int test_bitrate_from_;
      const int test_bitrate_to_;
      const int test_bitrate_step_;
      const int min_bwe_;
      const int start_bwe_;
      const int max_bwe_;
      SimulatedNetwork* send_simulated_network_;
      SimulatedNetwork* receive_simulated_network_;
      Call* sender_call_;
      TaskQueueBase* const task_queue_;
    } test(test_bitrate_from, test_bitrate_to, test_bitrate_step, min_bwe,
           start_bwe, max_bwe, task_queue());
  
    RunBaseTest(&test);
  }
  
  // TODO(bugs.webrtc.org/8878)
  #if defined(WEBRTC_MAC)
  #define MAYBE_MinVideoAndAudioBitrate DISABLED_MinVideoAndAudioBitrate
  #else
  #define MAYBE_MinVideoAndAudioBitrate MinVideoAndAudioBitrate
  #endif
  TEST_F(CallPerfTest, MAYBE_MinVideoAndAudioBitrate) {
    TestMinAudioVideoBitrate(110, 40, -10, 10000, 70000, 200000);
  }
  
  }  // namespace webrtc
  