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
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
|
/*
* Copyright (c) 2019 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.
*/
#ifndef TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
#define TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
#include <stdio.h>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/test/video/function_video_decoder_factory.h"
#include "api/video_codecs/video_decoder.h"
#include "call/call.h"
#include "media/engine/internal_decoder_factory.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/time_utils.h"
#include "test/null_transport.h"
#include "test/rtp_file_reader.h"
#include "test/test_video_capturer.h"
#include "test/video_renderer.h"
namespace webrtc {
namespace test {
// The RtpReplayer is a utility for fuzzing the RTP/RTCP receiver stack in
// WebRTC. It achieves this by accepting a set of Receiver configurations and
// an RtpDump (consisting of both RTP and RTCP packets). The `rtp_dump` is
// passed in as a buffer to allow simple mutation fuzzing directly on the dump.
class RtpReplayer final {
public:
// Holds all the important stream information required to emulate the WebRTC
// rtp receival code path.
struct StreamState {
test::NullTransport transport;
std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
std::vector<VideoReceiveStreamInterface*> receive_streams;
std::unique_ptr<VideoDecoderFactory> decoder_factory;
};
// Construct an RtpReplayer from a JSON replay configuration file.
static void Replay(const std::string& replay_config_filepath,
const uint8_t* rtp_dump_data,
size_t rtp_dump_size);
// Construct an RtpReplayer from a set of
// VideoReceiveStreamInterface::Configs. Note the stream_state.transport must
// be set for each receiver stream.
static void Replay(
std::unique_ptr<StreamState> stream_state,
std::vector<VideoReceiveStreamInterface::Config> receive_stream_config,
const uint8_t* rtp_dump_data,
size_t rtp_dump_size);
private:
// Reads the replay configuration from Json.
static std::vector<VideoReceiveStreamInterface::Config> ReadConfigFromFile(
const std::string& replay_config,
Transport* transport);
// Configures the stream state based on the receiver configurations.
static void SetupVideoStreams(
std::vector<VideoReceiveStreamInterface::Config>* receive_stream_configs,
StreamState* stream_state,
Call* call);
// Creates a new RtpReader which can read the RtpDump
static std::unique_ptr<test::RtpFileReader> CreateRtpReader(
const uint8_t* rtp_dump_data,
size_t rtp_dump_size);
// Replays each packet to from the RtpDump.
static void ReplayPackets(rtc::FakeClock* clock,
Call* call,
test::RtpFileReader* rtp_reader,
const RtpHeaderExtensionMap& extensions);
}; // class RtpReplayer
} // namespace test
} // namespace webrtc
#endif // TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
|