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 "rtc_tools/rtp_generator/rtp_generator.h"
12
13 #include <algorithm>
14 #include <memory>
15 #include <utility>
16
17 #include "api/task_queue/default_task_queue_factory.h"
18 #include "api/test/create_frame_generator.h"
19 #include "api/video_codecs/builtin_video_decoder_factory.h"
20 #include "api/video_codecs/builtin_video_encoder_factory.h"
21 #include "api/video_codecs/video_encoder.h"
22 #include "api/video_codecs/video_encoder_config.h"
23 #include "media/base/media_constants.h"
24 #include "rtc_base/strings/json.h"
25 #include "rtc_base/system/file_wrapper.h"
26 #include "rtc_base/thread.h"
27 #include "test/testsupport/file_utils.h"
28
29 namespace webrtc {
30 namespace {
31
32 // Payload types.
33 constexpr int kPayloadTypeVp8 = 125;
34 constexpr int kPayloadTypeVp9 = 124;
35 constexpr int kPayloadTypeH264 = 123;
36 constexpr int kFakeVideoSendPayloadType = 122;
37
38 // Defaults
39 constexpr int kDefaultSsrc = 1337;
40 constexpr int kMaxConfigBufferSize = 8192;
41
42 // Utility function to validate a correct codec type has been passed in.
IsValidCodecType(const std::string & codec_name)43 bool IsValidCodecType(const std::string& codec_name) {
44 return cricket::kVp8CodecName == codec_name ||
45 cricket::kVp9CodecName == codec_name ||
46 cricket::kH264CodecName == codec_name;
47 }
48
49 // Utility function to return some base payload type for a codec_name.
GetDefaultTypeForPayloadName(const std::string & codec_name)50 int GetDefaultTypeForPayloadName(const std::string& codec_name) {
51 if (cricket::kVp8CodecName == codec_name) {
52 return kPayloadTypeVp8;
53 }
54 if (cricket::kVp9CodecName == codec_name) {
55 return kPayloadTypeVp9;
56 }
57 if (cricket::kH264CodecName == codec_name) {
58 return kPayloadTypeH264;
59 }
60 return kFakeVideoSendPayloadType;
61 }
62
63 // Creates a single VideoSendStream configuration.
64 absl::optional<RtpGeneratorOptions::VideoSendStreamConfig>
ParseVideoSendStreamConfig(const Json::Value & json)65 ParseVideoSendStreamConfig(const Json::Value& json) {
66 RtpGeneratorOptions::VideoSendStreamConfig config;
67
68 // Parse video source settings.
69 if (!rtc::GetIntFromJsonObject(json, "duration_ms", &config.duration_ms)) {
70 RTC_LOG(LS_WARNING) << "duration_ms not specified using default: "
71 << config.duration_ms;
72 }
73 if (!rtc::GetIntFromJsonObject(json, "video_width", &config.video_width)) {
74 RTC_LOG(LS_WARNING) << "video_width not specified using default: "
75 << config.video_width;
76 }
77 if (!rtc::GetIntFromJsonObject(json, "video_height", &config.video_height)) {
78 RTC_LOG(LS_WARNING) << "video_height not specified using default: "
79 << config.video_height;
80 }
81 if (!rtc::GetIntFromJsonObject(json, "video_fps", &config.video_fps)) {
82 RTC_LOG(LS_WARNING) << "video_fps not specified using default: "
83 << config.video_fps;
84 }
85 if (!rtc::GetIntFromJsonObject(json, "num_squares", &config.num_squares)) {
86 RTC_LOG(LS_WARNING) << "num_squares not specified using default: "
87 << config.num_squares;
88 }
89
90 // Parse RTP settings for this configuration.
91 config.rtp.ssrcs.push_back(kDefaultSsrc);
92 Json::Value rtp_json;
93 if (!rtc::GetValueFromJsonObject(json, "rtp", &rtp_json)) {
94 RTC_LOG(LS_ERROR) << "video_streams must have an rtp section";
95 return absl::nullopt;
96 }
97 if (!rtc::GetStringFromJsonObject(rtp_json, "payload_name",
98 &config.rtp.payload_name)) {
99 RTC_LOG(LS_ERROR) << "rtp.payload_name must be specified";
100 return absl::nullopt;
101 }
102 if (!IsValidCodecType(config.rtp.payload_name)) {
103 RTC_LOG(LS_ERROR) << "rtp.payload_name must be VP8,VP9 or H264";
104 return absl::nullopt;
105 }
106
107 config.rtp.payload_type =
108 GetDefaultTypeForPayloadName(config.rtp.payload_name);
109 if (!rtc::GetIntFromJsonObject(rtp_json, "payload_type",
110 &config.rtp.payload_type)) {
111 RTC_LOG(LS_WARNING)
112 << "rtp.payload_type not specified using default for codec type"
113 << config.rtp.payload_type;
114 }
115
116 return config;
117 }
118
119 } // namespace
120
ParseRtpGeneratorOptionsFromFile(const std::string & options_file)121 absl::optional<RtpGeneratorOptions> ParseRtpGeneratorOptionsFromFile(
122 const std::string& options_file) {
123 if (!test::FileExists(options_file)) {
124 RTC_LOG(LS_ERROR) << " configuration file does not exist";
125 return absl::nullopt;
126 }
127
128 // Read the configuration file from disk.
129 FileWrapper config_file = FileWrapper::OpenReadOnly(options_file);
130 std::vector<char> raw_json_buffer(kMaxConfigBufferSize, 0);
131 size_t bytes_read =
132 config_file.Read(raw_json_buffer.data(), raw_json_buffer.size() - 1);
133 if (bytes_read == 0) {
134 RTC_LOG(LS_ERROR) << "Unable to read the configuration file.";
135 return absl::nullopt;
136 }
137
138 // Parse the file as JSON
139 Json::Reader json_reader;
140 Json::Value json;
141 if (!json_reader.parse(raw_json_buffer.data(), json)) {
142 RTC_LOG(LS_ERROR) << "Unable to parse the corpus config json file";
143 return absl::nullopt;
144 }
145
146 RtpGeneratorOptions gen_options;
147 for (const auto& video_stream_json : json["video_streams"]) {
148 absl::optional<RtpGeneratorOptions::VideoSendStreamConfig>
149 video_stream_config = ParseVideoSendStreamConfig(video_stream_json);
150 if (!video_stream_config.has_value()) {
151 RTC_LOG(LS_ERROR) << "Unable to parse the corpus config json file";
152 return absl::nullopt;
153 }
154 gen_options.video_streams.push_back(*video_stream_config);
155 }
156 return gen_options;
157 }
158
RtpGenerator(const RtpGeneratorOptions & options)159 RtpGenerator::RtpGenerator(const RtpGeneratorOptions& options)
160 : options_(options),
161 video_encoder_factory_(CreateBuiltinVideoEncoderFactory()),
162 video_decoder_factory_(CreateBuiltinVideoDecoderFactory()),
163 video_bitrate_allocator_factory_(
164 CreateBuiltinVideoBitrateAllocatorFactory()),
165 event_log_(std::make_unique<RtcEventLogNull>()),
166 call_(Call::Create(CallConfig(event_log_.get()))),
167 task_queue_(CreateDefaultTaskQueueFactory()) {
168 constexpr int kMinBitrateBps = 30000; // 30 Kbps
169 constexpr int kMaxBitrateBps = 2500000; // 2.5 Mbps
170
171 int stream_count = 0;
172 for (const auto& send_config : options.video_streams) {
173 webrtc::VideoSendStream::Config video_config(this);
174 video_config.encoder_settings.encoder_factory =
175 video_encoder_factory_.get();
176 video_config.encoder_settings.bitrate_allocator_factory =
177 video_bitrate_allocator_factory_.get();
178 video_config.rtp = send_config.rtp;
179 // Update some required to be unique values.
180 stream_count++;
181 video_config.rtp.mid = "mid-" + std::to_string(stream_count);
182
183 // Configure the video encoder configuration.
184 VideoEncoderConfig encoder_config;
185 encoder_config.content_type =
186 VideoEncoderConfig::ContentType::kRealtimeVideo;
187 encoder_config.codec_type =
188 PayloadStringToCodecType(video_config.rtp.payload_name);
189 if (video_config.rtp.payload_name == cricket::kVp8CodecName) {
190 VideoCodecVP8 settings = VideoEncoder::GetDefaultVp8Settings();
191 encoder_config.encoder_specific_settings = new rtc::RefCountedObject<
192 VideoEncoderConfig::Vp8EncoderSpecificSettings>(settings);
193 } else if (video_config.rtp.payload_name == cricket::kVp9CodecName) {
194 VideoCodecVP9 settings = VideoEncoder::GetDefaultVp9Settings();
195 encoder_config.encoder_specific_settings = new rtc::RefCountedObject<
196 VideoEncoderConfig::Vp9EncoderSpecificSettings>(settings);
197 } else if (video_config.rtp.payload_name == cricket::kH264CodecName) {
198 VideoCodecH264 settings = VideoEncoder::GetDefaultH264Settings();
199 encoder_config.encoder_specific_settings = new rtc::RefCountedObject<
200 VideoEncoderConfig::H264EncoderSpecificSettings>(settings);
201 }
202 encoder_config.video_format.name = video_config.rtp.payload_name;
203 encoder_config.min_transmit_bitrate_bps = 0;
204 encoder_config.max_bitrate_bps = kMaxBitrateBps;
205 encoder_config.content_type =
206 VideoEncoderConfig::ContentType::kRealtimeVideo;
207
208 // Configure the simulcast layers.
209 encoder_config.number_of_streams = video_config.rtp.ssrcs.size();
210 encoder_config.bitrate_priority = 1.0;
211 encoder_config.simulcast_layers.resize(encoder_config.number_of_streams);
212 for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
213 encoder_config.simulcast_layers[i].active = true;
214 encoder_config.simulcast_layers[i].min_bitrate_bps = kMinBitrateBps;
215 encoder_config.simulcast_layers[i].max_bitrate_bps = kMaxBitrateBps;
216 encoder_config.simulcast_layers[i].max_framerate = send_config.video_fps;
217 }
218
219 encoder_config.video_stream_factory =
220 new rtc::RefCountedObject<cricket::EncoderStreamFactory>(
221 video_config.rtp.payload_name, /*max qp*/ 56, /*screencast*/ false,
222 /*screenshare enabled*/ false);
223
224 // Setup the fake video stream for this.
225 std::unique_ptr<test::FrameGeneratorCapturer> frame_generator =
226 std::make_unique<test::FrameGeneratorCapturer>(
227 Clock::GetRealTimeClock(),
228 test::CreateSquareFrameGenerator(send_config.video_width,
229 send_config.video_height,
230 absl::nullopt, absl::nullopt),
231 send_config.video_fps, *task_queue_);
232 frame_generator->Init();
233
234 VideoSendStream* video_send_stream = call_->CreateVideoSendStream(
235 std::move(video_config), std::move(encoder_config));
236 video_send_stream->SetSource(
237 frame_generator.get(),
238 webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
239 // Store these objects so we can destropy them at the end.
240 frame_generators_.push_back(std::move(frame_generator));
241 video_send_streams_.push_back(video_send_stream);
242 }
243 }
244
~RtpGenerator()245 RtpGenerator::~RtpGenerator() {
246 for (VideoSendStream* send_stream : video_send_streams_) {
247 call_->DestroyVideoSendStream(send_stream);
248 }
249 }
250
GenerateRtpDump(const std::string & rtp_dump_path)251 void RtpGenerator::GenerateRtpDump(const std::string& rtp_dump_path) {
252 rtp_dump_writer_.reset(test::RtpFileWriter::Create(
253 test::RtpFileWriter::kRtpDump, rtp_dump_path));
254
255 call_->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
256 webrtc::kNetworkUp);
257 for (VideoSendStream* send_stream : video_send_streams_) {
258 send_stream->Start();
259 }
260
261 // Spinlock until all the durations end.
262 WaitUntilAllVideoStreamsFinish();
263
264 call_->SignalChannelNetworkState(webrtc::MediaType::VIDEO,
265 webrtc::kNetworkDown);
266 }
267
SendRtp(const uint8_t * packet,size_t length,const webrtc::PacketOptions & options)268 bool RtpGenerator::SendRtp(const uint8_t* packet,
269 size_t length,
270 const webrtc::PacketOptions& options) {
271 test::RtpPacket rtp_packet = DataToRtpPacket(packet, length);
272 rtp_dump_writer_->WritePacket(&rtp_packet);
273 return true;
274 }
275
SendRtcp(const uint8_t * packet,size_t length)276 bool RtpGenerator::SendRtcp(const uint8_t* packet, size_t length) {
277 test::RtpPacket rtcp_packet = DataToRtpPacket(packet, length);
278 rtp_dump_writer_->WritePacket(&rtcp_packet);
279 return true;
280 }
281
GetMaxDuration() const282 int RtpGenerator::GetMaxDuration() const {
283 int max_end_ms = 0;
284 for (const auto& video_stream : options_.video_streams) {
285 max_end_ms = std::max(video_stream.duration_ms, max_end_ms);
286 }
287 return max_end_ms;
288 }
289
WaitUntilAllVideoStreamsFinish()290 void RtpGenerator::WaitUntilAllVideoStreamsFinish() {
291 // Find the maximum duration required by the streams.
292 start_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds();
293 int64_t max_end_ms = start_ms_ + GetMaxDuration();
294
295 int64_t current_time = 0;
296 do {
297 int64_t min_wait_time = 0;
298 current_time = Clock::GetRealTimeClock()->TimeInMilliseconds();
299 // Stop any streams that are no longer active.
300 for (size_t i = 0; i < options_.video_streams.size(); ++i) {
301 const int64_t end_ms = start_ms_ + options_.video_streams[i].duration_ms;
302 if (current_time > end_ms) {
303 video_send_streams_[i]->Stop();
304 } else {
305 min_wait_time = std::min(min_wait_time, end_ms - current_time);
306 }
307 }
308 rtc::Thread::Current()->SleepMs(min_wait_time);
309 } while (current_time < max_end_ms);
310 }
311
DataToRtpPacket(const uint8_t * packet,size_t packet_len)312 test::RtpPacket RtpGenerator::DataToRtpPacket(const uint8_t* packet,
313 size_t packet_len) {
314 webrtc::test::RtpPacket rtp_packet;
315 memcpy(rtp_packet.data, packet, packet_len);
316 rtp_packet.length = packet_len;
317 rtp_packet.original_length = packet_len;
318 rtp_packet.time_ms =
319 webrtc::Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_;
320 return rtp_packet;
321 }
322
323 } // namespace webrtc
324