Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 1209 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/call_perf_tests.cc b/third_party/libwebrtc/call/call_perf_tests.cc
new file mode 100644
index 0000000000..0ba6d05b19
--- /dev/null
+++ b/third_party/libwebrtc/call/call_perf_tests.cc
@@ -0,0 +1,1209 @@
+/*
+ *  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 "absl/flags/flag.h"
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.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 "call/call.h"
+#include "call/fake_network_pipe.h"
+#include "call/simulated_network.h"
+#include "media/engine/internal_encoder_factory.h"
+#include "media/engine/simulcast_encoder_adapter.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.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_rtcp_observer.h"
+#include "test/test_flags.h"
+#include "test/testsupport/file_utils.h"
+#include "test/video_encoder_proxy_factory.h"
+#include "test/video_test_constants.h"
+#include "video/config/video_encoder_config.h"
+#include "video/transport_adapter.h"
+
+using webrtc::test::DriftingClock;
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+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,
+                          absl::string_view 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);
+  void TestEncodeFramerate(VideoEncoderFactory* encoder_factory,
+                           absl::string_view payload_name,
+                           const std::vector<int>& max_framerates);
+};
+
+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(TaskQueueBase* task_queue,
+                                    Clock* clock,
+                                    absl::string_view test_label)
+      : test::RtpRtcpObserver(test::VideoTestConstants::kLongTimeout),
+        clock_(clock),
+        test_label_(test_label),
+        creation_time_ms_(clock_->TimeInMilliseconds()),
+        task_queue_(task_queue) {}
+
+  void OnFrame(const VideoFrame& video_frame) override {
+    task_queue_->PostTask([this]() { CheckStats(); });
+  }
+
+  void CheckStats() {
+    if (!receive_stream_)
+      return;
+
+    VideoReceiveStreamInterface::Stats 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;
+        GetGlobalMetricsLogger()->LogSingleValueMetric(
+            "sync_convergence_time" + test_label_, "synchronization",
+            time_since_creation, Unit::kMilliseconds,
+            ImprovementDirection::kSmallerIsBetter);
+      }
+      if (time_since_creation > kMinRunTimeMs)
+        observation_complete_.Set();
+    }
+    if (first_time_in_sync_ != -1)
+      sync_offset_ms_list_.AddSample(stats.sync_offset_ms);
+  }
+
+  void set_receive_stream(VideoReceiveStreamInterface* receive_stream) {
+    RTC_DCHECK_EQ(task_queue_, TaskQueueBase::Current());
+    // Note that receive_stream may be nullptr.
+    receive_stream_ = receive_stream;
+  }
+
+  void PrintResults() {
+    GetGlobalMetricsLogger()->LogMetric(
+        "stream_offset" + test_label_, "synchronization", sync_offset_ms_list_,
+        Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter);
+  }
+
+ private:
+  Clock* const clock_;
+  const std::string test_label_;
+  const int64_t creation_time_ms_;
+  int64_t first_time_in_sync_ = -1;
+  VideoReceiveStreamInterface* receive_stream_ = nullptr;
+  SamplesStatsCounter sync_offset_ms_list_;
+  TaskQueueBase* const task_queue_;
+};
+
+void CallPerfTest::TestAudioVideoSync(FecMode fec,
+                                      CreateOrder create_first,
+                                      float video_ntp_speed,
+                                      float video_rtp_speed,
+                                      float audio_rtp_speed,
+                                      absl::string_view 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;
+
+  auto observer = std::make_unique<VideoRtcpAndSyncObserver>(
+      task_queue(), 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;
+
+  AudioSendStream* audio_send_stream;
+  AudioReceiveStreamInterface* audio_receive_stream;
+  std::unique_ptr<DriftingClock> drifting_clock;
+
+  SendTask(task_queue(), [&]() {
+    metrics::Reset();
+    rtc::scoped_refptr<AudioDeviceModule> 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;
+    CallConfig 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;
+    CallConfig 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.get(),
+        test::PacketTransport::kSender, audio_pt_map,
+        std::make_unique<FakeNetworkPipe>(
+            Clock::GetRealTimeClock(),
+            std::make_unique<SimulatedNetwork>(audio_net_config)),
+        GetRegisteredExtensions(), GetRegisteredExtensions());
+    audio_send_transport->SetReceiver(receiver_call_->Receiver());
+
+    video_send_transport = std::make_unique<test::PacketTransport>(
+        task_queue(), sender_call_.get(), observer.get(),
+        test::PacketTransport::kSender, video_pt_map,
+        std::make_unique<FakeNetworkPipe>(
+            Clock::GetRealTimeClock(),
+            std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
+        GetRegisteredExtensions(), GetRegisteredExtensions());
+    video_send_transport->SetReceiver(receiver_call_->Receiver());
+
+    receive_transport = std::make_unique<test::PacketTransport>(
+        task_queue(), receiver_call_.get(), observer.get(),
+        test::PacketTransport::kReceiver, payload_type_map_,
+        std::make_unique<FakeNetworkPipe>(
+            Clock::GetRealTimeClock(),
+            std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
+        GetRegisteredExtensions(), GetRegisteredExtensions());
+    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;
+    // TODO(bugs.webrtc.org/14683): Let the tests fail with invalid config.
+    audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+        test::VideoTestConstants::kAudioSendPayloadType, {"OPUS", 48000, 2});
+    audio_send_config.min_bitrate_bps = 6000;
+    audio_send_config.max_bitrate_bps = 510000;
+    audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
+    audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config);
+
+    GetVideoSendConfig()->rtp.nack.rtp_history_ms =
+        test::VideoTestConstants::kNackRtpHistoryMs;
+    if (fec == FecMode::kOn) {
+      GetVideoSendConfig()->rtp.ulpfec.red_payload_type =
+          test::VideoTestConstants::kRedPayloadType;
+      GetVideoSendConfig()->rtp.ulpfec.ulpfec_payload_type =
+          test::VideoTestConstants::kUlpfecPayloadType;
+      video_receive_configs_[0].rtp.red_payload_type =
+          test::VideoTestConstants::kRedPayloadType;
+      video_receive_configs_[0].rtp.ulpfec_payload_type =
+          test::VideoTestConstants::kUlpfecPayloadType;
+    }
+    video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
+    video_receive_configs_[0].renderer = observer.get();
+    video_receive_configs_[0].sync_group = kSyncGroup;
+
+    AudioReceiveStreamInterface::Config audio_recv_config;
+    audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
+    audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
+    audio_recv_config.rtcp_send_transport = receive_transport.get();
+    audio_recv_config.sync_group = kSyncGroup;
+    audio_recv_config.decoder_factory = audio_decoder_factory_;
+    audio_recv_config.decoder_map = {
+        {test::VideoTestConstants::kAudioSendPayloadType, {"OPUS", 48000, 2}}};
+
+    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,
+        test::VideoTestConstants::kDefaultFramerate,
+        test::VideoTestConstants::kDefaultWidth,
+        test::VideoTestConstants::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(task_queue(), [&]() {
+    // Clear the pointer to the receive stream since it will now be deleted.
+    observer->set_receive_stream(nullptr);
+
+    audio_send_stream->Stop();
+    audio_receive_stream->Stop();
+
+    Stop();
+
+    DestroyStreams();
+
+    sender_call_->DestroyAudioSendStream(audio_send_stream);
+    receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
+
+    DestroyCalls();
+    // Call may post periodic rtcp packet to the transport on the process
+    // thread, thus transport should be destroyed after the call objects.
+    // Though transports keep pointers to the call objects, transports handle
+    // packets on the task_queue() and thus wouldn't create a race while current
+    // destruction happens in the same task as destruction of the call objects.
+    video_send_transport.reset();
+    audio_send_transport.reset();
+    receive_transport.reset();
+  });
+
+  observer->PrintResults();
+
+  // In quick test synchronization may not be achieved in time.
+  if (!absl::GetFlag(FLAGS_webrtc_quick_perf_test)) {
+// TODO(bugs.webrtc.org/10417): Reenable this for iOS
+#if !defined(WEBRTC_IOS)
+    EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
+#endif
+  }
+
+  task_queue()->PostTask(
+      [to_delete = observer.release()]() { delete to_delete; });
+}
+
+TEST_F(CallPerfTest, Synchronization_PlaysOutAudioAndVideoWithoutClockDrift) {
+  TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
+                     DriftingClock::kNoDrift, DriftingClock::kNoDrift,
+                     DriftingClock::kNoDrift, "_video_no_drift");
+}
+
+TEST_F(CallPerfTest, Synchronization_PlaysOutAudioAndVideoWithVideoNtpDrift) {
+  TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
+                     DriftingClock::PercentsFaster(10.0f),
+                     DriftingClock::kNoDrift, DriftingClock::kNoDrift,
+                     "_video_ntp_drift");
+}
+
+TEST_F(CallPerfTest,
+       Synchronization_PlaysOutAudioAndVideoWithAudioFasterThanVideoDrift) {
+  TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
+                     DriftingClock::kNoDrift,
+                     DriftingClock::PercentsSlower(30.0f),
+                     DriftingClock::PercentsFaster(30.0f), "_audio_faster");
+}
+
+TEST_F(CallPerfTest,
+       Synchronization_PlaysOutAudioAndVideoWithVideoFasterThanAudioDrift) {
+  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(test::VideoTestConstants::kLongTimeout),
+          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:
+    BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
+      return net_config_;
+    }
+
+    BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const override {
+      return net_config_;
+    }
+
+    void OnFrame(const VideoFrame& video_frame) override {
+      MutexLock lock(&mutex_);
+      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_.AddSample(time_offset_ms);
+
+      EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
+    }
+
+    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
+      MutexLock lock(&mutex_);
+      RtpPacket rtp_packet;
+      EXPECT_TRUE(rtp_packet.Parse(packet));
+
+      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_ = rtp_packet.Timestamp() - first_capture_timestamp;
+        rtp_start_timestamp_set_ = true;
+      }
+
+      uint32_t capture_timestamp =
+          rtp_packet.Timestamp() - rtp_start_timestamp_;
+      capture_time_list_.insert(
+          capture_time_list_.end(),
+          std::make_pair(rtp_packet.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<VideoReceiveStreamInterface::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.";
+      GetGlobalMetricsLogger()->LogMetric(
+          "capture_ntp_time", "real - estimated", time_offset_ms_list_,
+          Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter);
+    }
+
+    Mutex mutex_;
+    const BuiltInNetworkBehaviorConfig net_config_;
+    Clock* const clock_;
+    const int threshold_ms_;
+    const int start_time_ms_;
+    const int run_time_ms_;
+    const 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(&mutex_);
+    SamplesStatsCounter 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, Real_Estimated_CaptureNtpTimeWithNetworkDelay) {
+  BuiltInNetworkBehaviorConfig net_config;
+  net_config.queue_delay_ms = 100;
+  // TODO(wu): lower the threshold as the calculation/estimation becomes more
+  // accurate.
+  const int kThresholdMs = 100;
+  const int kStartTimeMs = 10000;
+  const int kRunTimeMs = 20000;
+  TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
+}
+
+TEST_F(CallPerfTest, Real_Estimated_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/estimation 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(test::VideoTestConstants::kLongTimeout),
+          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 {
+      // The sink wants can change either because an adaptation happened (i.e.
+      // the pixels or frame rate changed) or for other reasons, such as encoded
+      // resolutions being communicated (happens whenever we capture a new frame
+      // size). In this test, we only care about adaptations.
+      bool did_adapt =
+          last_wants_.max_pixel_count != wants.max_pixel_count ||
+          last_wants_.target_pixel_count != wants.target_pixel_count ||
+          last_wants_.max_framerate_fps != wants.max_framerate_fps;
+      last_wants_ = wants;
+      if (!did_adapt) {
+        return;
+      }
+      // 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<VideoReceiveStreamInterface::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_;
+
+   private:
+    rtc::VideoSinkWants last_wants_;
+  } 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,
+                             TaskQueueBase* task_queue)
+        : EndToEndTest(test::VideoTestConstants::kLongTimeout),
+          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),
+          task_queue_(task_queue),
+          task_safety_flag_(PendingTaskSafetyFlag::CreateDetached()) {}
+
+   private:
+    // TODO(holmer): Run this with a timer instead of once per packet.
+    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
+      task_queue_->PostTask(SafeTask(task_safety_flag_, [this]() {
+        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_.AddSample(bitrate_kbps);
+        }
+      }));
+      return SEND_PACKET;
+    }
+
+    void OnVideoStreamsCreated(VideoSendStream* send_stream,
+                               const std::vector<VideoReceiveStreamInterface*>&
+                                   receive_streams) override {
+      send_stream_ = send_stream;
+    }
+
+    void OnStreamsStopped() override { task_safety_flag_->SetNotAlive(); }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStreamInterface::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.";
+      GetGlobalMetricsLogger()->LogMetric(
+          std::string("bitrate_stats_") +
+              (pad_to_min_bitrate_ ? "min_transmit_bitrate"
+                                   : "without_min_transmit_bitrate"),
+          "bitrate_kbps", bitrate_kbps_list_, Unit::kUnitless,
+          ImprovementDirection::kNeitherIsBetter);
+    }
+
+    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_;
+    SamplesStatsCounter bitrate_kbps_list_;
+    TaskQueueBase* task_queue_;
+    rtc::scoped_refptr<PendingTaskSafetyFlag> task_safety_flag_;
+  } test(pad_to_min_bitrate, task_queue());
+
+  fake_encoder_max_bitrate_ = kMaxEncodeBitrateKbps;
+  RunBaseTest(&test);
+}
+
+TEST_F(CallPerfTest, Bitrate_Kbps_PadsToMinTransmitBitrate) {
+  TestMinTransmitBitrate(true);
+}
+
+TEST_F(CallPerfTest, Bitrate_Kbps_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 kInitialBitrateOverheadKpbs = 6;
+  static const uint32_t kReconfigureThresholdKbps = 600;
+
+  class VideoStreamFactory
+      : public VideoEncoderConfig::VideoStreamFactoryInterface {
+   public:
+    VideoStreamFactory() {}
+
+   private:
+    std::vector<VideoStream> CreateEncoderStreams(
+        int frame_width,
+        int frame_height,
+        const webrtc::VideoEncoderConfig& encoder_config) override {
+      std::vector<VideoStream> streams =
+          test::CreateVideoStreams(frame_width, frame_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:
+    explicit BitrateObserver(TaskQueueBase* task_queue)
+        : EndToEndTest(test::VideoTestConstants::kDefaultTimeout),
+          FakeEncoder(Clock::GetRealTimeClock()),
+          encoder_inits_(0),
+          last_set_bitrate_kbps_(0),
+          send_stream_(nullptr),
+          frame_generator_(nullptr),
+          encoder_factory_(this),
+          bitrate_allocator_factory_(
+              CreateBuiltinVideoBitrateAllocatorFactory()),
+          task_queue_(task_queue) {}
+
+    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 - kInitialBitrateOverheadKpbs;
+        EXPECT_EQ(expected_bitrate, config->startBitrate)
+            << "Encoder not initialized at expected bitrate.";
+        EXPECT_EQ(test::VideoTestConstants::kDefaultWidth, config->width);
+        EXPECT_EQ(test::VideoTestConstants::kDefaultHeight, config->height);
+      } else if (encoder_inits_ == 2) {
+        EXPECT_EQ(2 * test::VideoTestConstants::kDefaultWidth, config->width);
+        EXPECT_EQ(2 * test::VideoTestConstants::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<VideoReceiveStreamInterface::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 =
+          rtc::make_ref_counted<VideoStreamFactory>();
+
+      encoder_config_ = encoder_config->Copy();
+    }
+
+    void OnVideoStreamsCreated(VideoSendStream* send_stream,
+                               const std::vector<VideoReceiveStreamInterface*>&
+                                   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(test::VideoTestConstants::kDefaultTimeout))
+          << "Timed out before receiving an initial high bitrate.";
+      frame_generator_->ChangeResolution(
+          test::VideoTestConstants::kDefaultWidth * 2,
+          test::VideoTestConstants::kDefaultHeight * 2);
+      SendTask(task_queue_, [&]() {
+        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_;
+    TaskQueueBase* task_queue_;
+  } test(task_queue());
+
+  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 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() const {
+      BuiltInNetworkBehaviorConfig pipe_config;
+      pipe_config.link_capacity_kbps = test_bitrate_from_;
+      return pipe_config;
+    }
+
+    BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
+      return GetFakeNetworkPipeConfig();
+    }
+    BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const override {
+      return GetFakeNetworkPipeConfig();
+    }
+
+    void OnTransportCreated(
+        test::PacketTransport* to_receiver,
+        SimulatedNetworkInterface* sender_network,
+        test::PacketTransport* to_sender,
+        SimulatedNetworkInterface* receiver_network) override {
+      send_simulated_network_ = sender_network;
+      receive_simulated_network_ = receiver_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 = absl::GetFlag(FLAGS_webrtc_quick_perf_test);
+
+      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(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) {
+          RTC_LOG(LS_WARNING)
+              << "Failed test bitrate: " << test_bitrate << " RTT: " << avg_rtt;
+          break;
+        } else {
+          RTC_LOG(LS_INFO) << "Passed test bitrate: " << test_bitrate
+                           << " RTT: " << avg_rtt;
+          last_passed_test_bitrate = test_bitrate;
+        }
+      }
+      EXPECT_GT(last_passed_test_bitrate, -1)
+          << "Minimum supported bitrate out of the test scope";
+      GetGlobalMetricsLogger()->LogSingleValueMetric(
+          "min_test_bitrate_", "min_bitrate", last_passed_test_bitrate,
+          Unit::kUnitless, ImprovementDirection::kNeitherIsBetter);
+    }
+
+    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; }
+
+   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_;
+    SimulatedNetworkInterface* send_simulated_network_;
+    SimulatedNetworkInterface* 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);
+}
+
+TEST_F(CallPerfTest, Min_Bitrate_VideoAndAudio) {
+  TestMinAudioVideoBitrate(110, 40, -10, 10000, 70000, 200000);
+}
+
+void CallPerfTest::TestEncodeFramerate(VideoEncoderFactory* encoder_factory,
+                                       absl::string_view payload_name,
+                                       const std::vector<int>& max_framerates) {
+  static constexpr double kAllowedFpsDiff = 1.5;
+  static constexpr TimeDelta kMinGetStatsInterval = TimeDelta::Millis(400);
+  static constexpr TimeDelta kMinRunTime = TimeDelta::Seconds(15);
+  static constexpr DataRate kMaxBitrate = DataRate::KilobitsPerSec(1000);
+
+  class FramerateObserver
+      : public test::EndToEndTest,
+        public test::FrameGeneratorCapturer::SinkWantsObserver {
+   public:
+    FramerateObserver(VideoEncoderFactory* encoder_factory,
+                      absl::string_view payload_name,
+                      const std::vector<int>& max_framerates,
+                      TaskQueueBase* task_queue)
+        : EndToEndTest(test::VideoTestConstants::kDefaultTimeout),
+          clock_(Clock::GetRealTimeClock()),
+          encoder_factory_(encoder_factory),
+          payload_name_(payload_name),
+          max_framerates_(max_framerates),
+          task_queue_(task_queue),
+          start_time_(clock_->CurrentTime()),
+          last_getstats_time_(start_time_),
+          send_stream_(nullptr) {}
+
+    void OnFrameGeneratorCapturerCreated(
+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
+      frame_generator_capturer->ChangeResolution(640, 360);
+    }
+
+    void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
+                            const rtc::VideoSinkWants& wants) override {}
+
+    void ModifySenderBitrateConfig(
+        BitrateConstraints* bitrate_config) override {
+      bitrate_config->start_bitrate_bps = kMaxBitrate.bps() / 2;
+    }
+
+    void OnVideoStreamsCreated(VideoSendStream* send_stream,
+                               const std::vector<VideoReceiveStreamInterface*>&
+                                   receive_streams) override {
+      send_stream_ = send_stream;
+    }
+
+    size_t GetNumVideoStreams() const override {
+      return max_framerates_.size();
+    }
+
+    void ModifyVideoConfigs(
+        VideoSendStream::Config* send_config,
+        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
+        VideoEncoderConfig* encoder_config) override {
+      send_config->encoder_settings.encoder_factory = encoder_factory_;
+      send_config->rtp.payload_name = payload_name_;
+      send_config->rtp.payload_type =
+          test::VideoTestConstants::kVideoSendPayloadType;
+      encoder_config->video_format.name = payload_name_;
+      encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
+      encoder_config->max_bitrate_bps = kMaxBitrate.bps();
+      for (size_t i = 0; i < max_framerates_.size(); ++i) {
+        encoder_config->simulcast_layers[i].max_framerate = max_framerates_[i];
+        configured_framerates_[send_config->rtp.ssrcs[i]] = max_framerates_[i];
+      }
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timeout while waiting for framerate stats.";
+    }
+
+    void VerifyStats() const {
+      const bool quick_perf_test = absl::GetFlag(FLAGS_webrtc_quick_perf_test);
+      double input_fps = 0.0;
+      for (const auto& configured_framerate : configured_framerates_) {
+        input_fps = std::max(configured_framerate.second, input_fps);
+      }
+      for (const auto& encode_frame_rate_list : encode_frame_rate_lists_) {
+        const SamplesStatsCounter& values = encode_frame_rate_list.second;
+        GetGlobalMetricsLogger()->LogMetric(
+            "substream_fps", "encode_frame_rate", values, Unit::kUnitless,
+            ImprovementDirection::kNeitherIsBetter);
+        if (values.IsEmpty()) {
+          continue;
+        }
+        double average_fps = values.GetAverage();
+        uint32_t ssrc = encode_frame_rate_list.first;
+        double expected_fps = configured_framerates_.find(ssrc)->second;
+        if (quick_perf_test && expected_fps != input_fps)
+          EXPECT_NEAR(expected_fps, average_fps, kAllowedFpsDiff);
+      }
+    }
+
+    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
+      const Timestamp now = clock_->CurrentTime();
+      if (now - last_getstats_time_ > kMinGetStatsInterval) {
+        last_getstats_time_ = now;
+        task_queue_->PostTask([this, now]() {
+          VideoSendStream::Stats stats = send_stream_->GetStats();
+          for (const auto& stat : stats.substreams) {
+            encode_frame_rate_lists_[stat.first].AddSample(
+                stat.second.encode_frame_rate);
+          }
+          if (now - start_time_ > kMinRunTime) {
+            VerifyStats();
+            observation_complete_.Set();
+          }
+        });
+      }
+      return SEND_PACKET;
+    }
+
+    Clock* const clock_;
+    VideoEncoderFactory* const encoder_factory_;
+    const std::string payload_name_;
+    const std::vector<int> max_framerates_;
+    TaskQueueBase* const task_queue_;
+    const Timestamp start_time_;
+    Timestamp last_getstats_time_;
+    VideoSendStream* send_stream_;
+    std::map<uint32_t, SamplesStatsCounter> encode_frame_rate_lists_;
+    std::map<uint32_t, double> configured_framerates_;
+  } test(encoder_factory, payload_name, max_framerates, task_queue());
+
+  RunBaseTest(&test);
+}
+
+TEST_F(CallPerfTest, TestEncodeFramerateVp8Simulcast) {
+  InternalEncoderFactory internal_encoder_factory;
+  test::FunctionVideoEncoderFactory encoder_factory(
+      [&internal_encoder_factory]() {
+        return std::make_unique<SimulcastEncoderAdapter>(
+            &internal_encoder_factory, SdpVideoFormat("VP8"));
+      });
+
+  TestEncodeFramerate(&encoder_factory, "VP8",
+                      /*max_framerates=*/{20, 30});
+}
+
+TEST_F(CallPerfTest, TestEncodeFramerateVp8SimulcastLowerInputFps) {
+  InternalEncoderFactory internal_encoder_factory;
+  test::FunctionVideoEncoderFactory encoder_factory(
+      [&internal_encoder_factory]() {
+        return std::make_unique<SimulcastEncoderAdapter>(
+            &internal_encoder_factory, SdpVideoFormat("VP8"));
+      });
+
+  TestEncodeFramerate(&encoder_factory, "VP8",
+                      /*max_framerates=*/{14, 20});
+}
+
+}  // namespace webrtc