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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/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(
&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
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