| /* |
| * Copyright (c) 2016 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 "modules/rtp_rtcp/include/flexfec_receiver.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/byte_io.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| class DummyCallback : public RecoveredPacketReceiver { |
| void OnRecoveredPacket(const uint8_t* packet, size_t length) override {} |
| }; |
| } // namespace |
| |
| void FuzzOneInput(const uint8_t* data, size_t size) { |
| constexpr size_t kMinDataNeeded = 12; |
| if (size < kMinDataNeeded) { |
| return; |
| } |
| |
| uint32_t flexfec_ssrc; |
| memcpy(&flexfec_ssrc, data + 0, 4); |
| uint16_t flexfec_seq_num; |
| memcpy(&flexfec_seq_num, data + 4, 2); |
| uint32_t media_ssrc; |
| memcpy(&media_ssrc, data + 6, 4); |
| uint16_t media_seq_num; |
| memcpy(&media_seq_num, data + 10, 2); |
| |
| DummyCallback callback; |
| FlexfecReceiver receiver(flexfec_ssrc, media_ssrc, &callback); |
| |
| std::unique_ptr<uint8_t[]> packet; |
| size_t packet_length; |
| size_t i = kMinDataNeeded; |
| while (i < size) { |
| packet_length = kRtpHeaderSize + data[i++]; |
| packet = std::unique_ptr<uint8_t[]>(new uint8_t[packet_length]); |
| if (i + packet_length >= size) { |
| break; |
| } |
| memcpy(packet.get(), data + i, packet_length); |
| i += packet_length; |
| if (i < size && data[i++] % 2 == 0) { |
| // Simulate FlexFEC packet. |
| ByteWriter<uint16_t>::WriteBigEndian(packet.get() + 2, flexfec_seq_num++); |
| ByteWriter<uint32_t>::WriteBigEndian(packet.get() + 8, flexfec_ssrc); |
| } else { |
| // Simulate media packet. |
| ByteWriter<uint16_t>::WriteBigEndian(packet.get() + 2, media_seq_num++); |
| ByteWriter<uint32_t>::WriteBigEndian(packet.get() + 8, media_ssrc); |
| } |
| RtpPacketReceived parsed_packet; |
| if (parsed_packet.Parse(packet.get(), packet_length)) { |
| receiver.OnRtpPacket(parsed_packet); |
| } |
| } |
| } |
| |
| } // namespace webrtc |