receive_side_congestion_controller_fuzzer.cc

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/*

 *  Copyright (c) 2015 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 <cstddef>

#include <cstdint>

#include "api/array_view.h"

#include "api/environment/environment_factory.h"

#include "api/units/time_delta.h"

#include "api/units/timestamp.h"

#include "modules/congestion_controller/include/receive_side_congestion_controller.h"

#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"

#include "modules/rtp_rtcp/source/byte_io.h"

#include "modules/rtp_rtcp/source/rtp_header_extensions.h"

#include "modules/rtp_rtcp/source/rtp_packet_received.h"

#include "system_wrappers/include/clock.h"

namespace webrtc {

void FuzzOneInput(const uint8_t* data, size_t size) {

  Timestamp arrival_time = Timestamp::Micros(123'456'789);

  SimulatedClock clock(arrival_time);

  ReceiveSideCongestionController cc(

      CreateEnvironment(&clock),

      /*feedback_sender=*/[](auto...) {},

      /*remb_sender=*/[](auto...) {},

      /*network_state_estimator=*/nullptr);

  RtpHeaderExtensionMap extensions;

  extensions.Register<TransmissionOffset>(1);

  extensions.Register<AbsoluteSendTime>(2);

  extensions.Register<TransportSequenceNumber>(3);

  extensions.Register<TransportSequenceNumberV2>(4);

  RtpPacketReceived rtp_packet(&extensions);

  constexpr int kMinPacketSize = sizeof(uint16_t) + sizeof(uint8_t) + 12;

  const uint8_t* const end_data = data + size;

  while (end_data - data >= kMinPacketSize) {

    size_t packet_size = ByteReader<uint16_t>::ReadBigEndian(data) % 1500;

    data += sizeof(uint16_t);

    arrival_time += TimeDelta::Millis(ByteReader<uint8_t>::ReadBigEndian(data));

    data += sizeof(uint8_t);

    packet_size = std::min<size_t>(end_data - data, packet_size);

    auto raw_packet = rtc::MakeArrayView(data, packet_size);

    data += packet_size;

    if (!rtp_packet.Parse(raw_packet)) {

      continue;

    rtp_packet.set_arrival_time(arrival_time);

    cc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);

    clock.AdvanceTimeMilliseconds(5);

    cc.MaybeProcess();

}  // namespace webrtc