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diff --git a/third_party/libwebrtc/call/rampup_tests.h b/third_party/libwebrtc/call/rampup_tests.h
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+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef CALL_RAMPUP_TESTS_H_
+#define CALL_RAMPUP_TESTS_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/metrics/metric.h"
+#include "api/test/simulated_network.h"
+#include "call/call.h"
+#include "call/simulated_network.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "test/call_test.h"
+
+namespace webrtc {
+
+static const int kTransmissionTimeOffsetExtensionId = 6;
+static const int kAbsSendTimeExtensionId = 7;
+static const int kTransportSequenceNumberExtensionId = 8;
+static const unsigned int kSingleStreamTargetBps = 1000000;
+
+class Clock;
+
+class RampUpTester : public test::EndToEndTest {
+ public:
+  RampUpTester(size_t num_video_streams,
+               size_t num_audio_streams,
+               size_t num_flexfec_streams,
+               unsigned int start_bitrate_bps,
+               int64_t min_run_time_ms,
+               bool rtx,
+               bool red,
+               bool report_perf_stats,
+               TaskQueueBase* task_queue);
+  ~RampUpTester() override;
+
+  size_t GetNumVideoStreams() const override;
+  size_t GetNumAudioStreams() const override;
+  size_t GetNumFlexfecStreams() const override;
+
+  void PerformTest() override;
+
+ protected:
+  virtual void PollStats();
+
+  void AccumulateStats(const VideoSendStream::StreamStats& stream,
+                       size_t* total_packets_sent,
+                       size_t* total_sent,
+                       size_t* padding_sent,
+                       size_t* media_sent) const;
+
+  void ReportResult(absl::string_view measurement,
+                    size_t value,
+                    test::Unit unit,
+                    test::ImprovementDirection improvement_direction) const;
+  void TriggerTestDone();
+
+  Clock* const clock_;
+  BuiltInNetworkBehaviorConfig forward_transport_config_;
+  const size_t num_video_streams_;
+  const size_t num_audio_streams_;
+  const size_t num_flexfec_streams_;
+  const bool rtx_;
+  const bool red_;
+  const bool report_perf_stats_;
+  Call* sender_call_;
+  VideoSendStream* send_stream_;
+  test::PacketTransport* send_transport_;
+  SimulatedNetworkInterface* send_simulated_network_;
+
+ private:
+  typedef std::map<uint32_t, uint32_t> SsrcMap;
+  class VideoStreamFactory;
+
+  void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) override;
+  void OnVideoStreamsCreated(VideoSendStream* send_stream,
+                             const std::vector<VideoReceiveStreamInterface*>&
+                                 receive_streams) override;
+  BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override;
+  void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config) override;
+  void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                          std::vector<AudioReceiveStreamInterface::Config>*
+                              receive_configs) override;
+  void ModifyFlexfecConfigs(
+      std::vector<FlexfecReceiveStream::Config>* receive_configs) override;
+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
+  void OnTransportCreated(test::PacketTransport* to_receiver,
+                          SimulatedNetworkInterface* sender_network,
+                          test::PacketTransport* to_sender,
+                          SimulatedNetworkInterface* receiver_network) override;
+
+  const int start_bitrate_bps_;
+  const int64_t min_run_time_ms_;
+  int expected_bitrate_bps_;
+  int64_t test_start_ms_;
+  int64_t ramp_up_finished_ms_;
+
+  std::vector<uint32_t> video_ssrcs_;
+  std::vector<uint32_t> video_rtx_ssrcs_;
+  std::vector<uint32_t> audio_ssrcs_;
+
+ protected:
+  TaskQueueBase* const task_queue_;
+  RepeatingTaskHandle pending_task_;
+};
+
+class RampUpDownUpTester : public RampUpTester {
+ public:
+  RampUpDownUpTester(size_t num_video_streams,
+                     size_t num_audio_streams,
+                     size_t num_flexfec_streams,
+                     unsigned int start_bitrate_bps,
+                     bool rtx,
+                     bool red,
+                     const std::vector<int>& loss_rates,
+                     bool report_perf_stats,
+                     TaskQueueBase* task_queue);
+  ~RampUpDownUpTester() override;
+
+ protected:
+  void PollStats() override;
+
+ private:
+  enum TestStates {
+    kFirstRampup = 0,
+    kLowRate,
+    kSecondRampup,
+    kTestEnd,
+    kTransitionToNextState,
+  };
+
+  void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) override;
+
+  std::string GetModifierString() const;
+  int GetExpectedHighBitrate() const;
+  int GetHighLinkCapacity() const;
+  size_t GetFecBytes() const;
+  bool ExpectingFec() const;
+  void EvolveTestState(int bitrate_bps, bool suspended);
+
+  const std::vector<int> link_rates_;
+  TestStates test_state_;
+  TestStates next_state_;
+  int64_t state_start_ms_;
+  int64_t interval_start_ms_;
+  int sent_bytes_;
+  std::vector<int> loss_rates_;
+};
+
+}  // namespace webrtc
+#endif  // CALL_RAMPUP_TESTS_H_