| /* |
| * 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 "modules/congestion_controller/include/receive_side_congestion_controller.h" |
| #include "modules/pacing/packet_router.h" |
| #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "modules/rtp_rtcp/source/byte_io.h" |
| |
| namespace webrtc { |
| |
| void FuzzOneInput(const uint8_t* data, size_t size) { |
| size_t i = 0; |
| if (size < sizeof(int64_t) + sizeof(uint8_t) + sizeof(uint32_t)) |
| return; |
| SimulatedClock clock(data[i++]); |
| PacketRouter packet_router; |
| ReceiveSideCongestionController cc(&clock, &packet_router); |
| RemoteBitrateEstimator* rbe = cc.GetRemoteBitrateEstimator(true); |
| RTPHeader header; |
| header.ssrc = ByteReader<uint32_t>::ReadBigEndian(&data[i]); |
| i += sizeof(uint32_t); |
| header.extension.hasTransportSequenceNumber = true; |
| int64_t arrival_time_ms = std::min<int64_t>( |
| std::max<int64_t>(ByteReader<int64_t>::ReadBigEndian(&data[i]), 0), |
| std::numeric_limits<int64_t>::max() / 2); |
| i += sizeof(int64_t); |
| const size_t kMinPacketSize = |
| sizeof(size_t) + sizeof(uint16_t) + sizeof(uint8_t); |
| while (i + kMinPacketSize < size) { |
| size_t payload_size = ByteReader<size_t>::ReadBigEndian(&data[i]) % 1500; |
| i += sizeof(size_t); |
| header.extension.transportSequenceNumber = |
| ByteReader<uint16_t>::ReadBigEndian(&data[i]); |
| i += sizeof(uint16_t); |
| rbe->IncomingPacket(arrival_time_ms, payload_size, header); |
| clock.AdvanceTimeMilliseconds(5); |
| arrival_time_ms += ByteReader<uint8_t>::ReadBigEndian(&data[i]); |
| i += sizeof(uint8_t); |
| } |
| rbe->Process(); |
| } |
| } // namespace webrtc |