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| 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #include <memory> |
| 12 |
| 13 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h" |
| 14 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| 15 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
| 16 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| 17 #include "webrtc/system_wrappers/include/clock.h" |
| 18 |
| 19 namespace webrtc { |
| 20 |
| 21 namespace { |
| 22 |
| 23 constexpr int kFlexfecPayloadType = 123; |
| 24 constexpr uint32_t kMediaSsrc = 1234; |
| 25 constexpr uint32_t kFlexfecSsrc = 5678; |
| 26 const std::vector<RtpExtension> kNoRtpHeaderExtensions; |
| 27 |
| 28 } // namespace |
| 29 |
| 30 void FuzzOneInput(const uint8_t* data, size_t size) { |
| 31 size_t i = 0; |
| 32 if (size < 5) { |
| 33 return; |
| 34 } |
| 35 |
| 36 SimulatedClock clock(1 + data[i++]); |
| 37 FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, |
| 38 kNoRtpHeaderExtensions, &clock); |
| 39 FecProtectionParams params = { |
| 40 data[i++], static_cast<int>(data[i++] % 100), |
| 41 data[i++] <= 127 ? kFecMaskRandom : kFecMaskBursty}; |
| 42 sender.SetFecParameters(params); |
| 43 uint16_t seq_num = data[i++]; |
| 44 |
| 45 while (i + 1 < size) { |
| 46 // Everything past the base RTP header (12 bytes) is payload, |
| 47 // from the perspective of FlexFEC. |
| 48 size_t payload_size = data[i++]; |
| 49 if (i + kRtpHeaderSize + payload_size >= size) |
| 50 break; |
| 51 std::unique_ptr<uint8_t[]> packet( |
| 52 new uint8_t[kRtpHeaderSize + payload_size]); |
| 53 memcpy(packet.get(), &data[i], kRtpHeaderSize + payload_size); |
| 54 i += kRtpHeaderSize + payload_size; |
| 55 ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++); |
| 56 ByteWriter<uint32_t>::WriteBigEndian(&packet[8], kMediaSsrc); |
| 57 RtpPacketToSend rtp_packet(nullptr); |
| 58 if (!rtp_packet.Parse(packet.get(), kRtpHeaderSize + payload_size)) |
| 59 break; |
| 60 sender.AddRtpPacketAndGenerateFec(rtp_packet); |
| 61 if (sender.FecAvailable()) { |
| 62 std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets = |
| 63 sender.GetFecPackets(); |
| 64 } |
| 65 } |
| 66 } |
| 67 |
| 68 } // namespace webrtc |
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