1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
|
/*
* 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 <memory>
#include "modules/rtp_rtcp/include/flexfec_sender.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_to_send.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace {
constexpr int kFlexfecPayloadType = 123;
constexpr uint32_t kMediaSsrc = 1234;
constexpr uint32_t kFlexfecSsrc = 5678;
const char kNoMid[] = "";
const std::vector<RtpExtension> kNoRtpHeaderExtensions;
const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
} // namespace
void FuzzOneInput(const uint8_t* data, size_t size) {
size_t i = 0;
if (size < 5 || size > 200) {
return;
}
SimulatedClock clock(1 + data[i++]);
FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
nullptr /* rtp_state */, &clock);
FecProtectionParams params = {
data[i++], static_cast<int>(data[i++] % 100),
data[i++] <= 127 ? kFecMaskRandom : kFecMaskBursty};
sender.SetProtectionParameters(params, params);
uint16_t seq_num = data[i++];
while (i + 1 < size) {
// Everything past the base RTP header (12 bytes) is payload,
// from the perspective of FlexFEC.
size_t payload_size = data[i++];
if (i + kRtpHeaderSize + payload_size >= size)
break;
std::unique_ptr<uint8_t[]> packet(
new uint8_t[kRtpHeaderSize + payload_size]);
memcpy(packet.get(), &data[i], kRtpHeaderSize + payload_size);
i += kRtpHeaderSize + payload_size;
ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++);
ByteWriter<uint32_t>::WriteBigEndian(&packet[8], kMediaSsrc);
RtpPacketToSend rtp_packet(nullptr);
if (!rtp_packet.Parse(packet.get(), kRtpHeaderSize + payload_size))
break;
sender.AddPacketAndGenerateFec(rtp_packet);
sender.GetFecPackets();
}
}
} // namespace webrtc
|
