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diff --git a/third_party/libwebrtc/rtc_tools/rtp_generator/rtp_generator.h b/third_party/libwebrtc/rtc_tools/rtp_generator/rtp_generator.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 RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_
+#define RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/call/transport.h"
+#include "api/media_types.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "call/call.h"
+#include "call/rtp_config.h"
+#include "call/video_send_stream.h"
+#include "media/engine/webrtc_video_engine.h"
+#include "test/frame_generator_capturer.h"
+#include "test/rtp_file_reader.h"
+#include "test/rtp_file_writer.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+
+// Specifies all the configurable options to pass to the corpus generator.
+// If modified please update the JSON parser as well as all.
+struct RtpGeneratorOptions {
+  struct VideoSendStreamConfig {
+    // The time to record the RtpDump for.
+    int duration_ms = 10000;
+    // The video resolution width.
+    int video_width = 640;
+    // The video resolution height.
+    int video_height = 480;
+    // The video fps.
+    int video_fps = 24;
+    // The number of squares to render.
+    int num_squares = 128;
+    // The individual RTP configuration.
+    RtpConfig rtp;
+  };
+  // Multiple senders can be active at once on an rtp channel.
+  std::vector<VideoSendStreamConfig> video_streams;
+};
+
+// Attempts to parse RtpGeneratorOptions from a JSON file. Any failures
+// will result in absl::nullopt.
+absl::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile(
+    const std::string& options_file);
+
+// The RtpGenerator allows generating of corpus material intended to be
+// used by fuzzers. It accepts a simple Json configuration file that allows the
+// user to configure the codec, extensions and error correction mechanisms. It
+// will then proceed to generate an rtpdump for the specified duration using
+// that configuration that can be replayed by the video_replayer. The receiver
+// configuration JSON will also be output and can be replayed as follows:
+// ./rtp_generator --config_file sender_config --output_rtpdump my.rtpdump
+//                 --output_config receiver_config.json
+// ./video_replay --config_file receiver_config.json --output_file my.rtpdump
+//
+// It achieves this by creating a VideoStreamSender, configuring it as requested
+// by the user and then intercepting all outgoing RTP packets and writing them
+// to a file instead of out of the network. It then uses this sender
+// configuration to generate a mirror receiver configuration that can be read by
+// the video_replay program.
+class RtpGenerator final : public webrtc::Transport {
+ public:
+  // Construct a new RtpGenerator using the specified options.
+  explicit RtpGenerator(const RtpGeneratorOptions& options);
+
+  RtpGenerator() = delete;
+  RtpGenerator(const RtpGenerator&) = delete;
+  RtpGenerator& operator=(const RtpGenerator&) = delete;
+
+  // Cleans up the VideoSendStream.
+  ~RtpGenerator() override;
+  // Generates an rtp_dump that is written out to
+  void GenerateRtpDump(const std::string& rtp_dump_path);
+
+ private:
+  // webrtc::Transport implementation
+  // Captured RTP packets are written to the RTPDump file instead of over the
+  // network.
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const webrtc::PacketOptions& options) override;
+  // RTCP packets are ignored for now.
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
+  // Returns the maximum duration
+  int GetMaxDuration() const;
+  // Waits until all video streams have finished.
+  void WaitUntilAllVideoStreamsFinish();
+  // Converts packet data into an RtpPacket.
+  test::RtpPacket DataToRtpPacket(const uint8_t* packet, size_t packet_len);
+
+  const RtpGeneratorOptions options_;
+  std::unique_ptr<VideoEncoderFactory> video_encoder_factory_;
+  std::unique_ptr<VideoDecoderFactory> video_decoder_factory_;
+  std::unique_ptr<VideoBitrateAllocatorFactory>
+      video_bitrate_allocator_factory_;
+  std::unique_ptr<RtcEventLog> event_log_;
+  std::unique_ptr<Call> call_;
+  std::unique_ptr<test::RtpFileWriter> rtp_dump_writer_;
+  std::vector<std::unique_ptr<test::FrameGeneratorCapturer>> frame_generators_;
+  std::vector<VideoSendStream*> video_send_streams_;
+  std::vector<uint32_t> durations_ms_;
+  uint32_t start_ms_ = 0;
+  std::unique_ptr<TaskQueueFactory> task_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // RTC_TOOLS_RTP_GENERATOR_RTP_GENERATOR_H_