1 /*
2 * Copyright (c) 2019 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 "test/fuzzers/utils/rtp_replayer.h"
12
13 #include <algorithm>
14 #include <memory>
15 #include <string>
16 #include <utility>
17
18 #include "api/task_queue/default_task_queue_factory.h"
19 #include "api/transport/field_trial_based_config.h"
20 #include "rtc_base/strings/json.h"
21 #include "system_wrappers/include/clock.h"
22 #include "test/call_config_utils.h"
23 #include "test/encoder_settings.h"
24 #include "test/fake_decoder.h"
25 #include "test/rtp_file_reader.h"
26 #include "test/rtp_header_parser.h"
27 #include "test/run_loop.h"
28
29 namespace webrtc {
30 namespace test {
31
Replay(const std::string & replay_config_filepath,const uint8_t * rtp_dump_data,size_t rtp_dump_size)32 void RtpReplayer::Replay(const std::string& replay_config_filepath,
33 const uint8_t* rtp_dump_data,
34 size_t rtp_dump_size) {
35 auto stream_state = std::make_unique<StreamState>();
36 std::vector<VideoReceiveStream::Config> receive_stream_configs =
37 ReadConfigFromFile(replay_config_filepath, &(stream_state->transport));
38 return Replay(std::move(stream_state), std::move(receive_stream_configs),
39 rtp_dump_data, rtp_dump_size);
40 }
41
Replay(std::unique_ptr<StreamState> stream_state,std::vector<VideoReceiveStream::Config> receive_stream_configs,const uint8_t * rtp_dump_data,size_t rtp_dump_size)42 void RtpReplayer::Replay(
43 std::unique_ptr<StreamState> stream_state,
44 std::vector<VideoReceiveStream::Config> receive_stream_configs,
45 const uint8_t* rtp_dump_data,
46 size_t rtp_dump_size) {
47 RunLoop loop;
48 rtc::ScopedBaseFakeClock fake_clock;
49
50 // Work around: webrtc calls webrtc::Random(clock.TimeInMicroseconds())
51 // everywhere and Random expects non-zero seed. Let's set the clock non-zero
52 // to make them happy.
53 fake_clock.SetTime(webrtc::Timestamp::Millis(1));
54
55 // Attempt to create an RtpReader from the input file.
56 auto rtp_reader = CreateRtpReader(rtp_dump_data, rtp_dump_size);
57 if (rtp_reader == nullptr) {
58 RTC_LOG(LS_ERROR) << "Failed to create the rtp_reader";
59 return;
60 }
61
62 // Setup the video streams based on the configuration.
63 webrtc::RtcEventLogNull event_log;
64 std::unique_ptr<TaskQueueFactory> task_queue_factory =
65 CreateDefaultTaskQueueFactory();
66 Call::Config call_config(&event_log);
67 call_config.task_queue_factory = task_queue_factory.get();
68 FieldTrialBasedConfig field_trials;
69 call_config.trials = &field_trials;
70 std::unique_ptr<Call> call(Call::Create(call_config));
71 SetupVideoStreams(&receive_stream_configs, stream_state.get(), call.get());
72
73 // Start replaying the provided stream now that it has been configured.
74 for (const auto& receive_stream : stream_state->receive_streams) {
75 receive_stream->Start();
76 }
77
78 ReplayPackets(&fake_clock, call.get(), rtp_reader.get());
79
80 for (const auto& receive_stream : stream_state->receive_streams) {
81 call->DestroyVideoReceiveStream(receive_stream);
82 }
83 }
84
ReadConfigFromFile(const std::string & replay_config,Transport * transport)85 std::vector<VideoReceiveStream::Config> RtpReplayer::ReadConfigFromFile(
86 const std::string& replay_config,
87 Transport* transport) {
88 Json::Reader json_reader;
89 Json::Value json_configs;
90 if (!json_reader.parse(replay_config, json_configs)) {
91 RTC_LOG(LS_ERROR)
92 << "Error parsing JSON replay configuration for the fuzzer"
93 << json_reader.getFormatedErrorMessages();
94 return {};
95 }
96
97 std::vector<VideoReceiveStream::Config> receive_stream_configs;
98 receive_stream_configs.reserve(json_configs.size());
99 for (const auto& json : json_configs) {
100 receive_stream_configs.push_back(
101 ParseVideoReceiveStreamJsonConfig(transport, json));
102 }
103 return receive_stream_configs;
104 }
105
SetupVideoStreams(std::vector<VideoReceiveStream::Config> * receive_stream_configs,StreamState * stream_state,Call * call)106 void RtpReplayer::SetupVideoStreams(
107 std::vector<VideoReceiveStream::Config>* receive_stream_configs,
108 StreamState* stream_state,
109 Call* call) {
110 stream_state->decoder_factory = std::make_unique<InternalDecoderFactory>();
111 for (auto& receive_config : *receive_stream_configs) {
112 // Attach the decoder for the corresponding payload type in the config.
113 for (auto& decoder : receive_config.decoders) {
114 decoder = test::CreateMatchingDecoder(decoder.payload_type,
115 decoder.video_format.name);
116 decoder.decoder_factory = stream_state->decoder_factory.get();
117 }
118
119 // Create the window to display the rendered video.
120 stream_state->sinks.emplace_back(
121 test::VideoRenderer::Create("Fuzzing WebRTC Video Config", 640, 480));
122 // Create a receive stream for this config.
123 receive_config.renderer = stream_state->sinks.back().get();
124 stream_state->receive_streams.emplace_back(
125 call->CreateVideoReceiveStream(std::move(receive_config)));
126 }
127 }
128
CreateRtpReader(const uint8_t * rtp_dump_data,size_t rtp_dump_size)129 std::unique_ptr<test::RtpFileReader> RtpReplayer::CreateRtpReader(
130 const uint8_t* rtp_dump_data,
131 size_t rtp_dump_size) {
132 std::unique_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create(
133 test::RtpFileReader::kRtpDump, rtp_dump_data, rtp_dump_size, {}));
134 if (!rtp_reader) {
135 RTC_LOG(LS_ERROR) << "Unable to open input file with any supported format";
136 return nullptr;
137 }
138 return rtp_reader;
139 }
140
ReplayPackets(rtc::FakeClock * clock,Call * call,test::RtpFileReader * rtp_reader)141 void RtpReplayer::ReplayPackets(rtc::FakeClock* clock,
142 Call* call,
143 test::RtpFileReader* rtp_reader) {
144 int64_t replay_start_ms = -1;
145 int num_packets = 0;
146 std::map<uint32_t, int> unknown_packets;
147
148 while (true) {
149 int64_t now_ms = rtc::TimeMillis();
150 if (replay_start_ms == -1) {
151 replay_start_ms = now_ms;
152 }
153
154 test::RtpPacket packet;
155 if (!rtp_reader->NextPacket(&packet)) {
156 break;
157 }
158
159 int64_t deliver_in_ms = replay_start_ms + packet.time_ms - now_ms;
160 if (deliver_in_ms > 0) {
161 // StatsCounter::ReportMetricToAggregatedCounter is O(elapsed time).
162 // Set an upper limit to prevent waste time.
163 clock->AdvanceTime(webrtc::TimeDelta::Millis(
164 std::min(deliver_in_ms, static_cast<int64_t>(100))));
165 }
166
167 ++num_packets;
168 switch (call->Receiver()->DeliverPacket(
169 webrtc::MediaType::VIDEO,
170 rtc::CopyOnWriteBuffer(packet.data, packet.length),
171 /* packet_time_us */ -1)) {
172 case PacketReceiver::DELIVERY_OK:
173 break;
174 case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
175 RTPHeader header;
176 std::unique_ptr<RtpHeaderParser> parser(
177 RtpHeaderParser::CreateForTest());
178
179 parser->Parse(packet.data, packet.length, &header);
180 if (unknown_packets[header.ssrc] == 0) {
181 RTC_LOG(LS_ERROR) << "Unknown SSRC: " << header.ssrc;
182 }
183 ++unknown_packets[header.ssrc];
184 break;
185 }
186 case PacketReceiver::DELIVERY_PACKET_ERROR: {
187 RTC_LOG(LS_ERROR)
188 << "Packet error, corrupt packets or incorrect setup?";
189 RTPHeader header;
190 std::unique_ptr<RtpHeaderParser> parser(
191 RtpHeaderParser::CreateForTest());
192 parser->Parse(packet.data, packet.length, &header);
193 RTC_LOG(LS_ERROR) << "Packet packet_length=" << packet.length
194 << " payload_type=" << header.payloadType
195 << " sequence_number=" << header.sequenceNumber
196 << " time_stamp=" << header.timestamp
197 << " ssrc=" << header.ssrc;
198 break;
199 }
200 }
201 }
202 RTC_LOG(LS_INFO) << "num_packets: " << num_packets;
203
204 for (const auto& unknown_packet : unknown_packets) {
205 RTC_LOG(LS_ERROR) << "Packets for unknown ssrc " << unknown_packet.first
206 << ":" << unknown_packet.second;
207 }
208 }
209
210 } // namespace test
211 } // namespace webrtc
212