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diff --git a/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h
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+/*
+ *  Copyright (c) 2019 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 TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
+#define TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
+
+#include <stdio.h>
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/test/video/function_video_decoder_factory.h"
+#include "api/video_codecs/video_decoder.h"
+#include "call/call.h"
+#include "media/engine/internal_decoder_factory.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/time_utils.h"
+#include "test/null_transport.h"
+#include "test/rtp_file_reader.h"
+#include "test/test_video_capturer.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace test {
+
+// The RtpReplayer is a utility for fuzzing the RTP/RTCP receiver stack in
+// WebRTC. It achieves this by accepting a set of Receiver configurations and
+// an RtpDump (consisting of both RTP and RTCP packets). The `rtp_dump` is
+// passed in as a buffer to allow simple mutation fuzzing directly on the dump.
+class RtpReplayer final {
+ public:
+  // Holds all the important stream information required to emulate the WebRTC
+  // rtp receival code path.
+  struct StreamState {
+    test::NullTransport transport;
+    std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
+    std::vector<VideoReceiveStreamInterface*> receive_streams;
+    std::unique_ptr<VideoDecoderFactory> decoder_factory;
+  };
+
+  // Construct an RtpReplayer from a JSON replay configuration file.
+  static void Replay(const std::string& replay_config_filepath,
+                     const uint8_t* rtp_dump_data,
+                     size_t rtp_dump_size);
+
+  // Construct an RtpReplayer from  a set of
+  // VideoReceiveStreamInterface::Configs. Note the stream_state.transport must
+  // be set for each receiver stream.
+  static void Replay(
+      std::unique_ptr<StreamState> stream_state,
+      std::vector<VideoReceiveStreamInterface::Config> receive_stream_config,
+      const uint8_t* rtp_dump_data,
+      size_t rtp_dump_size);
+
+ private:
+  // Reads the replay configuration from Json.
+  static std::vector<VideoReceiveStreamInterface::Config> ReadConfigFromFile(
+      const std::string& replay_config,
+      Transport* transport);
+
+  // Configures the stream state based on the receiver configurations.
+  static void SetupVideoStreams(
+      std::vector<VideoReceiveStreamInterface::Config>* receive_stream_configs,
+      StreamState* stream_state,
+      Call* call);
+
+  // Creates a new RtpReader which can read the RtpDump
+  static std::unique_ptr<test::RtpFileReader> CreateRtpReader(
+      const uint8_t* rtp_dump_data,
+      size_t rtp_dump_size);
+
+  // Replays each packet to from the RtpDump.
+  static void ReplayPackets(rtc::FakeClock* clock,
+                            Call* call,
+                            test::RtpFileReader* rtp_reader,
+                            const RtpHeaderExtensionMap& extensions);
+};  // class RtpReplayer
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FUZZERS_UTILS_RTP_REPLAYER_H_