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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 #ifdef ENABLE_RTC_EVENT_LOG |
| 12 |
| 13 #include "webrtc/call/rtc_event_log_unittest_helper.h" |
| 14 |
| 15 #include <cstring> |
| 16 #include <string> |
| 17 |
| 18 #include "testing/gtest/include/gtest/gtest.h" |
| 19 #include "webrtc/base/checks.h" |
| 20 #include "webrtc/test/test_suite.h" |
| 21 #include "webrtc/test/testsupport/fileutils.h" |
| 22 |
| 23 // Files generated at build-time by the protobuf compiler. |
| 24 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| 25 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h" |
| 26 #else |
| 27 #include "webrtc/call/rtc_event_log.pb.h" |
| 28 #endif |
| 29 |
| 30 namespace webrtc { |
| 31 |
| 32 namespace { |
| 33 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { |
| 34 switch (media_type) { |
| 35 case rtclog::MediaType::ANY: |
| 36 return MediaType::ANY; |
| 37 case rtclog::MediaType::AUDIO: |
| 38 return MediaType::AUDIO; |
| 39 case rtclog::MediaType::VIDEO: |
| 40 return MediaType::VIDEO; |
| 41 case rtclog::MediaType::DATA: |
| 42 return MediaType::DATA; |
| 43 } |
| 44 RTC_NOTREACHED(); |
| 45 return MediaType::ANY; |
| 46 } |
| 47 } // namespace |
| 48 |
| 49 // Checks that the event has a timestamp, a type and exactly the data field |
| 50 // corresponding to the type. |
| 51 ::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) { |
| 52 if (!event.has_timestamp_us()) |
| 53 return ::testing::AssertionFailure() << "Event has no timestamp"; |
| 54 if (!event.has_type()) |
| 55 return ::testing::AssertionFailure() << "Event has no event type"; |
| 56 rtclog::Event_EventType type = event.type(); |
| 57 if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet()) |
| 58 return ::testing::AssertionFailure() |
| 59 << "Event of type " << type << " has " |
| 60 << (event.has_rtp_packet() ? "" : "no ") << "RTP packet"; |
| 61 if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet()) |
| 62 return ::testing::AssertionFailure() |
| 63 << "Event of type " << type << " has " |
| 64 << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet"; |
| 65 if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) != |
| 66 event.has_audio_playout_event()) |
| 67 return ::testing::AssertionFailure() |
| 68 << "Event of type " << type << " has " |
| 69 << (event.has_audio_playout_event() ? "" : "no ") |
| 70 << "audio_playout event"; |
| 71 if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) != |
| 72 event.has_video_receiver_config()) |
| 73 return ::testing::AssertionFailure() |
| 74 << "Event of type " << type << " has " |
| 75 << (event.has_video_receiver_config() ? "" : "no ") |
| 76 << "receiver config"; |
| 77 if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) != |
| 78 event.has_video_sender_config()) |
| 79 return ::testing::AssertionFailure() |
| 80 << "Event of type " << type << " has " |
| 81 << (event.has_video_sender_config() ? "" : "no ") << "sender config"; |
| 82 if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) != |
| 83 event.has_audio_receiver_config()) { |
| 84 return ::testing::AssertionFailure() |
| 85 << "Event of type " << type << " has " |
| 86 << (event.has_audio_receiver_config() ? "" : "no ") |
| 87 << "audio receiver config"; |
| 88 } |
| 89 if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) != |
| 90 event.has_audio_sender_config()) { |
| 91 return ::testing::AssertionFailure() |
| 92 << "Event of type " << type << " has " |
| 93 << (event.has_audio_sender_config() ? "" : "no ") |
| 94 << "audio sender config"; |
| 95 } |
| 96 return ::testing::AssertionSuccess(); |
| 97 } |
| 98 |
| 99 void RtcEventLogTestHelper::VerifyReceiveStreamConfig( |
| 100 const ParsedRtcEventLog& parsed_log, |
| 101 size_t index, |
| 102 const VideoReceiveStream::Config& config) { |
| 103 const rtclog::Event& event = parsed_log.stream_[index]; |
| 104 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 105 ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type()); |
| 106 const rtclog::VideoReceiveConfig& receiver_config = |
| 107 event.video_receiver_config(); |
| 108 // Check SSRCs. |
| 109 ASSERT_TRUE(receiver_config.has_remote_ssrc()); |
| 110 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc()); |
| 111 ASSERT_TRUE(receiver_config.has_local_ssrc()); |
| 112 EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc()); |
| 113 // Check RTCP settings. |
| 114 ASSERT_TRUE(receiver_config.has_rtcp_mode()); |
| 115 if (config.rtp.rtcp_mode == RtcpMode::kCompound) |
| 116 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND, |
| 117 receiver_config.rtcp_mode()); |
| 118 else |
| 119 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE, |
| 120 receiver_config.rtcp_mode()); |
| 121 ASSERT_TRUE(receiver_config.has_remb()); |
| 122 EXPECT_EQ(config.rtp.remb, receiver_config.remb()); |
| 123 // Check RTX map. |
| 124 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()), |
| 125 receiver_config.rtx_map_size()); |
| 126 for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) { |
| 127 ASSERT_TRUE(rtx_map.has_payload_type()); |
| 128 ASSERT_TRUE(rtx_map.has_config()); |
| 129 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type())); |
| 130 const rtclog::RtxConfig& rtx_config = rtx_map.config(); |
| 131 const VideoReceiveStream::Config::Rtp::Rtx& rtx = |
| 132 config.rtp.rtx.at(rtx_map.payload_type()); |
| 133 ASSERT_TRUE(rtx_config.has_rtx_ssrc()); |
| 134 ASSERT_TRUE(rtx_config.has_rtx_payload_type()); |
| 135 EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc()); |
| 136 EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type()); |
| 137 } |
| 138 // Check header extensions. |
| 139 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()), |
| 140 receiver_config.header_extensions_size()); |
| 141 for (int i = 0; i < receiver_config.header_extensions_size(); i++) { |
| 142 ASSERT_TRUE(receiver_config.header_extensions(i).has_name()); |
| 143 ASSERT_TRUE(receiver_config.header_extensions(i).has_id()); |
| 144 const std::string& name = receiver_config.header_extensions(i).name(); |
| 145 int id = receiver_config.header_extensions(i).id(); |
| 146 EXPECT_EQ(config.rtp.extensions[i].id, id); |
| 147 EXPECT_EQ(config.rtp.extensions[i].name, name); |
| 148 } |
| 149 // Check decoders. |
| 150 ASSERT_EQ(static_cast<int>(config.decoders.size()), |
| 151 receiver_config.decoders_size()); |
| 152 for (int i = 0; i < receiver_config.decoders_size(); i++) { |
| 153 ASSERT_TRUE(receiver_config.decoders(i).has_name()); |
| 154 ASSERT_TRUE(receiver_config.decoders(i).has_payload_type()); |
| 155 const std::string& decoder_name = receiver_config.decoders(i).name(); |
| 156 int decoder_type = receiver_config.decoders(i).payload_type(); |
| 157 EXPECT_EQ(config.decoders[i].payload_name, decoder_name); |
| 158 EXPECT_EQ(config.decoders[i].payload_type, decoder_type); |
| 159 } |
| 160 |
| 161 // Check consistency of the parser. |
| 162 VideoReceiveStream::Config parsed_config(nullptr); |
| 163 parsed_log.GetVideoReceiveConfig(index, &parsed_config); |
| 164 EXPECT_EQ(config.rtp.remote_ssrc, parsed_config.rtp.remote_ssrc); |
| 165 EXPECT_EQ(config.rtp.local_ssrc, parsed_config.rtp.local_ssrc); |
| 166 // Check RTCP settings. |
| 167 EXPECT_EQ(config.rtp.rtcp_mode, parsed_config.rtp.rtcp_mode); |
| 168 EXPECT_EQ(config.rtp.remb, parsed_config.rtp.remb); |
| 169 // Check RTX map. |
| 170 EXPECT_EQ(config.rtp.rtx.size(), parsed_config.rtp.rtx.size()); |
| 171 for (const auto& kv : config.rtp.rtx) { |
| 172 auto parsed_kv = parsed_config.rtp.rtx.find(kv.first); |
| 173 EXPECT_EQ(kv.first, parsed_kv->first); |
| 174 EXPECT_EQ(kv.second.ssrc, parsed_kv->second.ssrc); |
| 175 EXPECT_EQ(kv.second.payload_type, parsed_kv->second.payload_type); |
| 176 } |
| 177 // Check header extensions. |
| 178 EXPECT_EQ(config.rtp.extensions.size(), parsed_config.rtp.extensions.size()); |
| 179 for (size_t i = 0; i < parsed_config.rtp.extensions.size(); i++) { |
| 180 EXPECT_EQ(config.rtp.extensions[i].name, |
| 181 parsed_config.rtp.extensions[i].name); |
| 182 EXPECT_EQ(config.rtp.extensions[i].id, parsed_config.rtp.extensions[i].id); |
| 183 } |
| 184 // Check decoders. |
| 185 EXPECT_EQ(config.decoders.size(), parsed_config.decoders.size()); |
| 186 for (size_t i = 0; i < parsed_config.decoders.size(); i++) { |
| 187 EXPECT_EQ(config.decoders[i].payload_name, |
| 188 parsed_config.decoders[i].payload_name); |
| 189 EXPECT_EQ(config.decoders[i].payload_type, |
| 190 parsed_config.decoders[i].payload_type); |
| 191 } |
| 192 } |
| 193 |
| 194 void RtcEventLogTestHelper::VerifySendStreamConfig( |
| 195 const ParsedRtcEventLog& parsed_log, |
| 196 size_t index, |
| 197 const VideoSendStream::Config& config) { |
| 198 const rtclog::Event& event = parsed_log.stream_[index]; |
| 199 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 200 ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type()); |
| 201 const rtclog::VideoSendConfig& sender_config = event.video_sender_config(); |
| 202 // Check SSRCs. |
| 203 ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()), |
| 204 sender_config.ssrcs_size()); |
| 205 for (int i = 0; i < sender_config.ssrcs_size(); i++) { |
| 206 EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i)); |
| 207 } |
| 208 // Check header extensions. |
| 209 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()), |
| 210 sender_config.header_extensions_size()); |
| 211 for (int i = 0; i < sender_config.header_extensions_size(); i++) { |
| 212 ASSERT_TRUE(sender_config.header_extensions(i).has_name()); |
| 213 ASSERT_TRUE(sender_config.header_extensions(i).has_id()); |
| 214 const std::string& name = sender_config.header_extensions(i).name(); |
| 215 int id = sender_config.header_extensions(i).id(); |
| 216 EXPECT_EQ(config.rtp.extensions[i].id, id); |
| 217 EXPECT_EQ(config.rtp.extensions[i].name, name); |
| 218 } |
| 219 // Check RTX settings. |
| 220 ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()), |
| 221 sender_config.rtx_ssrcs_size()); |
| 222 for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) { |
| 223 EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i)); |
| 224 } |
| 225 if (sender_config.rtx_ssrcs_size() > 0) { |
| 226 ASSERT_TRUE(sender_config.has_rtx_payload_type()); |
| 227 EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type()); |
| 228 } |
| 229 // Check encoder. |
| 230 ASSERT_TRUE(sender_config.has_encoder()); |
| 231 ASSERT_TRUE(sender_config.encoder().has_name()); |
| 232 ASSERT_TRUE(sender_config.encoder().has_payload_type()); |
| 233 EXPECT_EQ(config.encoder_settings.payload_name, |
| 234 sender_config.encoder().name()); |
| 235 EXPECT_EQ(config.encoder_settings.payload_type, |
| 236 sender_config.encoder().payload_type()); |
| 237 |
| 238 // Check consistency of the parser. |
| 239 VideoSendStream::Config parsed_config(nullptr); |
| 240 parsed_log.GetVideoSendConfig(index, &parsed_config); |
| 241 // Check SSRCs |
| 242 EXPECT_EQ(config.rtp.ssrcs.size(), parsed_config.rtp.ssrcs.size()); |
| 243 for (size_t i = 0; i < config.rtp.ssrcs.size(); i++) { |
| 244 EXPECT_EQ(config.rtp.ssrcs[i], parsed_config.rtp.ssrcs[i]); |
| 245 } |
| 246 // Check header extensions. |
| 247 EXPECT_EQ(config.rtp.extensions.size(), parsed_config.rtp.extensions.size()); |
| 248 for (size_t i = 0; i < parsed_config.rtp.extensions.size(); i++) { |
| 249 EXPECT_EQ(config.rtp.extensions[i].name, |
| 250 parsed_config.rtp.extensions[i].name); |
| 251 EXPECT_EQ(config.rtp.extensions[i].id, parsed_config.rtp.extensions[i].id); |
| 252 } |
| 253 // Check RTX settings. |
| 254 EXPECT_EQ(config.rtp.rtx.ssrcs.size(), parsed_config.rtp.rtx.ssrcs.size()); |
| 255 for (size_t i = 0; i < config.rtp.rtx.ssrcs.size(); i++) { |
| 256 EXPECT_EQ(config.rtp.rtx.ssrcs[i], parsed_config.rtp.rtx.ssrcs[i]); |
| 257 } |
| 258 EXPECT_EQ(config.rtp.rtx.payload_type, parsed_config.rtp.rtx.payload_type); |
| 259 // Check encoder. |
| 260 EXPECT_EQ(config.encoder_settings.payload_name, |
| 261 parsed_config.encoder_settings.payload_name); |
| 262 EXPECT_EQ(config.encoder_settings.payload_type, |
| 263 parsed_config.encoder_settings.payload_type); |
| 264 } |
| 265 |
| 266 void RtcEventLogTestHelper::VerifyRtpEvent(const ParsedRtcEventLog& parsed_log, |
| 267 size_t index, |
| 268 PacketDirection direction, |
| 269 MediaType media_type, |
| 270 const uint8_t* header, |
| 271 size_t header_size, |
| 272 size_t total_size) { |
| 273 const rtclog::Event& event = parsed_log.stream_[index]; |
| 274 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 275 ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type()); |
| 276 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); |
| 277 ASSERT_TRUE(rtp_packet.has_incoming()); |
| 278 EXPECT_EQ(direction == kIncomingPacket, rtp_packet.incoming()); |
| 279 ASSERT_TRUE(rtp_packet.has_type()); |
| 280 EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type())); |
| 281 ASSERT_TRUE(rtp_packet.has_packet_length()); |
| 282 EXPECT_EQ(total_size, rtp_packet.packet_length()); |
| 283 ASSERT_TRUE(rtp_packet.has_header()); |
| 284 ASSERT_EQ(header_size, rtp_packet.header().size()); |
| 285 for (size_t i = 0; i < header_size; i++) { |
| 286 EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i])); |
| 287 } |
| 288 |
| 289 // Check consistency of the parser. |
| 290 PacketDirection parsed_direction; |
| 291 MediaType parsed_media_type; |
| 292 uint8_t parsed_header[1500]; |
| 293 size_t parsed_header_size, parsed_total_size; |
| 294 parsed_log.GetRtpHeader(index, &parsed_direction, &parsed_media_type, |
| 295 parsed_header, &parsed_header_size, |
| 296 &parsed_total_size); |
| 297 EXPECT_EQ(direction, parsed_direction); |
| 298 EXPECT_EQ(media_type, parsed_media_type); |
| 299 ASSERT_EQ(header_size, parsed_header_size); |
| 300 EXPECT_EQ(0, std::memcmp(header, parsed_header, header_size)); |
| 301 EXPECT_EQ(total_size, parsed_total_size); |
| 302 } |
| 303 |
| 304 void RtcEventLogTestHelper::VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log, |
| 305 size_t index, |
| 306 PacketDirection direction, |
| 307 MediaType media_type, |
| 308 const uint8_t* packet, |
| 309 size_t total_size) { |
| 310 const rtclog::Event& event = parsed_log.stream_[index]; |
| 311 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 312 ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type()); |
| 313 const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); |
| 314 ASSERT_TRUE(rtcp_packet.has_incoming()); |
| 315 EXPECT_EQ(direction == kIncomingPacket, rtcp_packet.incoming()); |
| 316 ASSERT_TRUE(rtcp_packet.has_type()); |
| 317 EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type())); |
| 318 ASSERT_TRUE(rtcp_packet.has_packet_data()); |
| 319 ASSERT_EQ(total_size, rtcp_packet.packet_data().size()); |
| 320 for (size_t i = 0; i < total_size; i++) { |
| 321 EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i])); |
| 322 } |
| 323 |
| 324 // Check consistency of the parser. |
| 325 PacketDirection parsed_direction; |
| 326 MediaType parsed_media_type; |
| 327 uint8_t parsed_packet[1500]; |
| 328 size_t parsed_total_size; |
| 329 parsed_log.GetRtcpPacket(index, &parsed_direction, &parsed_media_type, |
| 330 parsed_packet, &parsed_total_size); |
| 331 EXPECT_EQ(direction, parsed_direction); |
| 332 EXPECT_EQ(media_type, parsed_media_type); |
| 333 ASSERT_EQ(total_size, parsed_total_size); |
| 334 EXPECT_EQ(0, std::memcmp(packet, parsed_packet, total_size)); |
| 335 } |
| 336 |
| 337 void RtcEventLogTestHelper::VerifyPlayoutEvent( |
| 338 const ParsedRtcEventLog& parsed_log, |
| 339 size_t index, |
| 340 uint32_t ssrc) { |
| 341 const rtclog::Event& event = parsed_log.stream_[index]; |
| 342 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 343 ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type()); |
| 344 const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event(); |
| 345 ASSERT_TRUE(playout_event.has_local_ssrc()); |
| 346 EXPECT_EQ(ssrc, playout_event.local_ssrc()); |
| 347 |
| 348 // Check consistency of the parser. |
| 349 uint32_t parsed_ssrc; |
| 350 parsed_log.GetAudioPlayout(index, &parsed_ssrc); |
| 351 EXPECT_EQ(ssrc, parsed_ssrc); |
| 352 } |
| 353 |
| 354 void RtcEventLogTestHelper::VerifyBweLossEvent( |
| 355 const ParsedRtcEventLog& parsed_log, |
| 356 size_t index, |
| 357 int32_t bitrate, |
| 358 uint8_t fraction_loss, |
| 359 int32_t total_packets) { |
| 360 const rtclog::Event& event = parsed_log.stream_[index]; |
| 361 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 362 ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type()); |
| 363 const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event(); |
| 364 ASSERT_TRUE(bwe_event.has_bitrate()); |
| 365 EXPECT_EQ(bitrate, bwe_event.bitrate()); |
| 366 ASSERT_TRUE(bwe_event.has_fraction_loss()); |
| 367 EXPECT_EQ(fraction_loss, bwe_event.fraction_loss()); |
| 368 ASSERT_TRUE(bwe_event.has_total_packets()); |
| 369 EXPECT_EQ(total_packets, bwe_event.total_packets()); |
| 370 |
| 371 // Check consistency of the parser. |
| 372 int32_t parsed_bitrate; |
| 373 uint8_t parsed_fraction_loss; |
| 374 int32_t parsed_total_packets; |
| 375 parsed_log.GetBwePacketLossEvent( |
| 376 index, &parsed_bitrate, &parsed_fraction_loss, &parsed_total_packets); |
| 377 EXPECT_EQ(bitrate, parsed_bitrate); |
| 378 EXPECT_EQ(fraction_loss, parsed_fraction_loss); |
| 379 EXPECT_EQ(total_packets, parsed_total_packets); |
| 380 } |
| 381 |
| 382 void RtcEventLogTestHelper::VerifyLogStartEvent( |
| 383 const ParsedRtcEventLog& parsed_log, |
| 384 size_t index) { |
| 385 const rtclog::Event& event = parsed_log.stream_[index]; |
| 386 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 387 EXPECT_EQ(rtclog::Event::LOG_START, event.type()); |
| 388 } |
| 389 |
| 390 void RtcEventLogTestHelper::VerifyLogEndEvent( |
| 391 const ParsedRtcEventLog& parsed_log, |
| 392 size_t index) { |
| 393 const rtclog::Event& event = parsed_log.stream_[index]; |
| 394 ASSERT_TRUE(IsValidBasicEvent(event)); |
| 395 EXPECT_EQ(rtclog::Event::LOG_END, event.type()); |
| 396 } |
| 397 |
| 398 } // namespace webrtc |
| 399 |
| 400 #endif // ENABLE_RTC_EVENT_LOG |
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