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diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
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+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2012 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 MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_UNITTEST_HELPER_H_
+#define MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_UNITTEST_HELPER_H_
+
+#include <list>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace testing {
+
+class TestBitrateObserver : public RemoteBitrateObserver {
+ public:
+  TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
+  virtual ~TestBitrateObserver() {}
+
+  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                               uint32_t bitrate) override;
+
+  void Reset() { updated_ = false; }
+
+  bool updated() const { return updated_; }
+
+  uint32_t latest_bitrate() const { return latest_bitrate_; }
+
+ private:
+  bool updated_;
+  uint32_t latest_bitrate_;
+};
+
+class RtpStream {
+ public:
+  struct RtpPacket {
+    int64_t send_time;
+    int64_t arrival_time;
+    uint32_t rtp_timestamp;
+    size_t size;
+    uint32_t ssrc;
+  };
+
+  struct RtcpPacket {
+    uint32_t ntp_secs;
+    uint32_t ntp_frac;
+    uint32_t timestamp;
+    uint32_t ssrc;
+  };
+
+  typedef std::list<RtpPacket*> PacketList;
+
+  enum { kSendSideOffsetUs = 1000000 };
+
+  RtpStream(int fps,
+            int bitrate_bps,
+            uint32_t ssrc,
+            uint32_t frequency,
+            uint32_t timestamp_offset,
+            int64_t rtcp_receive_time);
+
+  RtpStream(const RtpStream&) = delete;
+  RtpStream& operator=(const RtpStream&) = delete;
+
+  void set_rtp_timestamp_offset(uint32_t offset);
+
+  // Generates a new frame for this stream. If called too soon after the
+  // previous frame, no frame will be generated. The frame is split into
+  // packets.
+  int64_t GenerateFrame(int64_t time_now_us, PacketList* packets);
+
+  // The send-side time when the next frame can be generated.
+  int64_t next_rtp_time() const;
+
+  // Generates an RTCP packet.
+  RtcpPacket* Rtcp(int64_t time_now_us);
+
+  void set_bitrate_bps(int bitrate_bps);
+
+  int bitrate_bps() const;
+
+  uint32_t ssrc() const;
+
+  static bool Compare(const std::pair<uint32_t, RtpStream*>& left,
+                      const std::pair<uint32_t, RtpStream*>& right);
+
+ private:
+  enum { kRtcpIntervalUs = 1000000 };
+
+  int fps_;
+  int bitrate_bps_;
+  uint32_t ssrc_;
+  uint32_t frequency_;
+  int64_t next_rtp_time_;
+  int64_t next_rtcp_time_;
+  uint32_t rtp_timestamp_offset_;
+  const double kNtpFracPerMs;
+};
+
+class StreamGenerator {
+ public:
+  typedef std::list<RtpStream::RtcpPacket*> RtcpList;
+
+  StreamGenerator(int capacity, int64_t time_now);
+
+  ~StreamGenerator();
+
+  StreamGenerator(const StreamGenerator&) = delete;
+  StreamGenerator& operator=(const StreamGenerator&) = delete;
+
+  // Add a new stream.
+  void AddStream(RtpStream* stream);
+
+  // Set the link capacity.
+  void set_capacity_bps(int capacity_bps);
+
+  // Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+  // is decided by the initial allocation ratios.
+  void SetBitrateBps(int bitrate_bps);
+
+  // Set the RTP timestamp offset for the stream identified by `ssrc`.
+  void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
+
+  // TODO(holmer): Break out the channel simulation part from this class to make
+  // it possible to simulate different types of channels.
+  int64_t GenerateFrame(RtpStream::PacketList* packets, int64_t time_now_us);
+
+ private:
+  typedef std::map<uint32_t, RtpStream*> StreamMap;
+
+  // Capacity of the simulated channel in bits per second.
+  int capacity_;
+  // The time when the last packet arrived.
+  int64_t prev_arrival_time_us_;
+  // All streams being transmitted on this simulated channel.
+  StreamMap streams_;
+};
+}  // namespace testing
+
+class RemoteBitrateEstimatorTest : public ::testing::Test {
+ public:
+  RemoteBitrateEstimatorTest();
+  virtual ~RemoteBitrateEstimatorTest();
+
+  RemoteBitrateEstimatorTest(const RemoteBitrateEstimatorTest&) = delete;
+  RemoteBitrateEstimatorTest& operator=(const RemoteBitrateEstimatorTest&) =
+      delete;
+
+ protected:
+  virtual void SetUp() = 0;
+
+  void AddDefaultStream();
+
+  // Helper to convert some time format to resolution used in absolute send time
+  // header extension, rounded upwards. `t` is the time to convert, in some
+  // resolution. `denom` is the value to divide `t` by to get whole seconds,
+  // e.g. `denom` = 1000 if `t` is in milliseconds.
+  static uint32_t AbsSendTime(int64_t t, int64_t denom);
+
+  // Helper to add two absolute send time values and keep it less than 1<<24.
+  static uint32_t AddAbsSendTime(uint32_t t1, uint32_t t2);
+
+  // Helper to create an RTPHeader containing the relevant data for the
+  // estimator (all other fields are cleared) and call IncomingPacket on the
+  // estimator.
+  void IncomingPacket(uint32_t ssrc,
+                      size_t payload_size,
+                      int64_t arrival_time,
+                      uint32_t rtp_timestamp,
+                      uint32_t absolute_send_time);
+
+  // Generates a frame of packets belonging to a stream at a given bitrate and
+  // with a given ssrc. The stream is pushed through a very simple simulated
+  // network, and is then given to the receive-side bandwidth estimator.
+  // Returns true if an over-use was seen, false otherwise.
+  // The StreamGenerator::updated() should be used to check for any changes in
+  // target bitrate after the call to this function.
+  bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);
+
+  // Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+  // until it reaches `target_bitrate`.
+  // Can for instance be used to run the estimator for some time to get it
+  // into a steady state.
+  uint32_t SteadyStateRun(uint32_t ssrc,
+                          int number_of_frames,
+                          uint32_t start_bitrate,
+                          uint32_t min_bitrate,
+                          uint32_t max_bitrate,
+                          uint32_t target_bitrate);
+
+  void TestTimestampGroupingTestHelper();
+
+  void TestWrappingHelper(int silence_time_s);
+
+  void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
+  void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
+  void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
+  void CapacityDropTestHelper(int number_of_streams,
+                              bool wrap_time_stamp,
+                              uint32_t expected_bitrate_drop_delta,
+                              int64_t receiver_clock_offset_change_ms);
+
+  static const uint32_t kDefaultSsrc;
+
+  SimulatedClock clock_;  // Time at the receiver.
+  std::unique_ptr<testing::TestBitrateObserver> bitrate_observer_;
+  std::unique_ptr<RemoteBitrateEstimator> bitrate_estimator_;
+  std::unique_ptr<testing::StreamGenerator> stream_generator_;
+  int64_t arrival_time_offset_ms_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_REMOTE_BITRATE_ESTIMATOR_UNITTEST_HELPER_H_