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| 1 /* | |
| 2 * Copyright (c) 2015 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 <stdio.h> | |
| 14 #include <string> | |
| 15 #include <vector> | |
| 16 | |
| 17 #include "testing/gtest/include/gtest/gtest.h" | |
| 18 #include "webrtc/base/checks.h" | |
| 19 #include "webrtc/base/scoped_ptr.h" | |
| 20 #include "webrtc/call.h" | |
| 21 #include "webrtc/system_wrappers/interface/clock.h" | |
| 22 #include "webrtc/test/test_suite.h" | |
| 23 #include "webrtc/test/testsupport/fileutils.h" | |
| 24 #include "webrtc/test/testsupport/gtest_disable.h" | |
| 25 #include "webrtc/video/rtc_event_log.h" | |
| 26 | |
| 27 // Files generated at build-time by the protobuf compiler. | |
| 28 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | |
| 29 #include "external/webrtc/webrtc/video/rtc_event_log.pb.h" | |
| 30 #else | |
| 31 #include "webrtc/video/rtc_event_log.pb.h" | |
| 32 #endif | |
| 33 | |
| 34 namespace webrtc { | |
| 35 | |
| 36 // TODO(terelius): Place this definition with other parsing functions? | |
| 37 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { | |
| 38 switch (media_type) { | |
| 39 case rtclog::MediaType::ANY: | |
| 40 return MediaType::ANY; | |
| 41 case rtclog::MediaType::AUDIO: | |
| 42 return MediaType::AUDIO; | |
| 43 case rtclog::MediaType::VIDEO: | |
| 44 return MediaType::VIDEO; | |
| 45 case rtclog::MediaType::DATA: | |
| 46 return MediaType::DATA; | |
| 47 } | |
| 48 RTC_NOTREACHED(); | |
| 49 return MediaType::ANY; | |
| 50 } | |
| 51 | |
| 52 // Checks that the event has a timestamp, a type and exactly the data field | |
| 53 // corresponding to the type. | |
| 54 ::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) { | |
| 55 if (!event.has_timestamp_us()) | |
| 56 return ::testing::AssertionFailure() << "Event has no timestamp"; | |
| 57 if (!event.has_type()) | |
| 58 return ::testing::AssertionFailure() << "Event has no event type"; | |
| 59 rtclog::Event_EventType type = event.type(); | |
| 60 if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet()) | |
| 61 return ::testing::AssertionFailure() | |
| 62 << "Event of type " << type << " has " | |
| 63 << (event.has_rtp_packet() ? "" : "no ") << "RTP packet"; | |
| 64 if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet()) | |
| 65 return ::testing::AssertionFailure() | |
| 66 << "Event of type " << type << " has " | |
| 67 << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet"; | |
| 68 if ((type == rtclog::Event::DEBUG_EVENT) != event.has_debug_event()) | |
| 69 return ::testing::AssertionFailure() | |
| 70 << "Event of type " << type << " has " | |
| 71 << (event.has_debug_event() ? "" : "no ") << "debug event"; | |
| 72 if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) != | |
| 73 event.has_video_receiver_config()) | |
| 74 return ::testing::AssertionFailure() | |
| 75 << "Event of type " << type << " has " | |
| 76 << (event.has_video_receiver_config() ? "" : "no ") | |
| 77 << "receiver config"; | |
| 78 if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) != | |
| 79 event.has_video_sender_config()) | |
| 80 return ::testing::AssertionFailure() | |
| 81 << "Event of type " << type << " has " | |
| 82 << (event.has_video_sender_config() ? "" : "no ") << "sender config"; | |
| 83 if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) != | |
| 84 event.has_audio_receiver_config()) { | |
| 85 return ::testing::AssertionFailure() | |
| 86 << "Event of type " << type << " has " | |
| 87 << (event.has_audio_receiver_config() ? "" : "no ") | |
| 88 << "audio receiver config"; | |
| 89 } | |
| 90 if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) != | |
| 91 event.has_audio_sender_config()) { | |
| 92 return ::testing::AssertionFailure() | |
| 93 << "Event of type " << type << " has " | |
| 94 << (event.has_audio_sender_config() ? "" : "no ") | |
| 95 << "audio sender config"; | |
| 96 } | |
| 97 return ::testing::AssertionSuccess(); | |
| 98 } | |
| 99 | |
| 100 void VerifyReceiveStreamConfig(const rtclog::Event& event, | |
| 101 const VideoReceiveStream::Config& config) { | |
| 102 ASSERT_TRUE(IsValidBasicEvent(event)); | |
| 103 ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type()); | |
| 104 const rtclog::VideoReceiveConfig& receiver_config = | |
| 105 event.video_receiver_config(); | |
| 106 // Check SSRCs. | |
| 107 ASSERT_TRUE(receiver_config.has_remote_ssrc()); | |
| 108 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc()); | |
| 109 ASSERT_TRUE(receiver_config.has_local_ssrc()); | |
| 110 EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc()); | |
| 111 // Check RTCP settings. | |
| 112 ASSERT_TRUE(receiver_config.has_rtcp_mode()); | |
| 113 if (config.rtp.rtcp_mode == newapi::kRtcpCompound) | |
| 114 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND, | |
| 115 receiver_config.rtcp_mode()); | |
| 116 else | |
| 117 EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE, | |
| 118 receiver_config.rtcp_mode()); | |
| 119 ASSERT_TRUE(receiver_config.has_receiver_reference_time_report()); | |
| 120 EXPECT_EQ(config.rtp.rtcp_xr.receiver_reference_time_report, | |
| 121 receiver_config.receiver_reference_time_report()); | |
| 122 ASSERT_TRUE(receiver_config.has_remb()); | |
| 123 EXPECT_EQ(config.rtp.remb, receiver_config.remb()); | |
| 124 // Check RTX map. | |
| 125 ASSERT_EQ(static_cast<int>(config.rtp.rtx.size()), | |
| 126 receiver_config.rtx_map_size()); | |
| 127 for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) { | |
| 128 ASSERT_TRUE(rtx_map.has_payload_type()); | |
| 129 ASSERT_TRUE(rtx_map.has_config()); | |
| 130 EXPECT_EQ(1u, config.rtp.rtx.count(rtx_map.payload_type())); | |
| 131 const rtclog::RtxConfig& rtx_config = rtx_map.config(); | |
| 132 const VideoReceiveStream::Config::Rtp::Rtx& rtx = | |
| 133 config.rtp.rtx.at(rtx_map.payload_type()); | |
| 134 ASSERT_TRUE(rtx_config.has_rtx_ssrc()); | |
| 135 ASSERT_TRUE(rtx_config.has_rtx_payload_type()); | |
| 136 EXPECT_EQ(rtx.ssrc, rtx_config.rtx_ssrc()); | |
| 137 EXPECT_EQ(rtx.payload_type, rtx_config.rtx_payload_type()); | |
| 138 } | |
| 139 // Check header extensions. | |
| 140 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()), | |
| 141 receiver_config.header_extensions_size()); | |
| 142 for (int i = 0; i < receiver_config.header_extensions_size(); i++) { | |
| 143 ASSERT_TRUE(receiver_config.header_extensions(i).has_name()); | |
| 144 ASSERT_TRUE(receiver_config.header_extensions(i).has_id()); | |
| 145 const std::string& name = receiver_config.header_extensions(i).name(); | |
| 146 int id = receiver_config.header_extensions(i).id(); | |
| 147 EXPECT_EQ(config.rtp.extensions[i].id, id); | |
| 148 EXPECT_EQ(config.rtp.extensions[i].name, name); | |
| 149 } | |
| 150 // Check decoders. | |
| 151 ASSERT_EQ(static_cast<int>(config.decoders.size()), | |
| 152 receiver_config.decoders_size()); | |
| 153 for (int i = 0; i < receiver_config.decoders_size(); i++) { | |
| 154 ASSERT_TRUE(receiver_config.decoders(i).has_name()); | |
| 155 ASSERT_TRUE(receiver_config.decoders(i).has_payload_type()); | |
| 156 const std::string& decoder_name = receiver_config.decoders(i).name(); | |
| 157 int decoder_type = receiver_config.decoders(i).payload_type(); | |
| 158 EXPECT_EQ(config.decoders[i].payload_name, decoder_name); | |
| 159 EXPECT_EQ(config.decoders[i].payload_type, decoder_type); | |
| 160 } | |
| 161 } | |
| 162 | |
| 163 void VerifySendStreamConfig(const rtclog::Event& event, | |
| 164 const VideoSendStream::Config& config) { | |
| 165 ASSERT_TRUE(IsValidBasicEvent(event)); | |
| 166 ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type()); | |
| 167 const rtclog::VideoSendConfig& sender_config = event.video_sender_config(); | |
| 168 // Check SSRCs. | |
| 169 ASSERT_EQ(static_cast<int>(config.rtp.ssrcs.size()), | |
| 170 sender_config.ssrcs_size()); | |
| 171 for (int i = 0; i < sender_config.ssrcs_size(); i++) { | |
| 172 EXPECT_EQ(config.rtp.ssrcs[i], sender_config.ssrcs(i)); | |
| 173 } | |
| 174 // Check header extensions. | |
| 175 ASSERT_EQ(static_cast<int>(config.rtp.extensions.size()), | |
| 176 sender_config.header_extensions_size()); | |
| 177 for (int i = 0; i < sender_config.header_extensions_size(); i++) { | |
| 178 ASSERT_TRUE(sender_config.header_extensions(i).has_name()); | |
| 179 ASSERT_TRUE(sender_config.header_extensions(i).has_id()); | |
| 180 const std::string& name = sender_config.header_extensions(i).name(); | |
| 181 int id = sender_config.header_extensions(i).id(); | |
| 182 EXPECT_EQ(config.rtp.extensions[i].id, id); | |
| 183 EXPECT_EQ(config.rtp.extensions[i].name, name); | |
| 184 } | |
| 185 // Check RTX settings. | |
| 186 ASSERT_EQ(static_cast<int>(config.rtp.rtx.ssrcs.size()), | |
| 187 sender_config.rtx_ssrcs_size()); | |
| 188 for (int i = 0; i < sender_config.rtx_ssrcs_size(); i++) { | |
| 189 EXPECT_EQ(config.rtp.rtx.ssrcs[i], sender_config.rtx_ssrcs(i)); | |
| 190 } | |
| 191 if (sender_config.rtx_ssrcs_size() > 0) { | |
| 192 ASSERT_TRUE(sender_config.has_rtx_payload_type()); | |
| 193 EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type()); | |
| 194 } | |
| 195 // Check CNAME. | |
| 196 ASSERT_TRUE(sender_config.has_c_name()); | |
| 197 EXPECT_EQ(config.rtp.c_name, sender_config.c_name()); | |
| 198 // Check encoder. | |
| 199 ASSERT_TRUE(sender_config.has_encoder()); | |
| 200 ASSERT_TRUE(sender_config.encoder().has_name()); | |
| 201 ASSERT_TRUE(sender_config.encoder().has_payload_type()); | |
| 202 EXPECT_EQ(config.encoder_settings.payload_name, | |
| 203 sender_config.encoder().name()); | |
| 204 EXPECT_EQ(config.encoder_settings.payload_type, | |
| 205 sender_config.encoder().payload_type()); | |
| 206 } | |
| 207 | |
| 208 void VerifyRtpEvent(const rtclog::Event& event, | |
| 209 bool incoming, | |
| 210 MediaType media_type, | |
| 211 uint8_t* header, | |
| 212 size_t header_size, | |
| 213 size_t total_size) { | |
| 214 ASSERT_TRUE(IsValidBasicEvent(event)); | |
| 215 ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type()); | |
| 216 const rtclog::RtpPacket& rtp_packet = event.rtp_packet(); | |
| 217 ASSERT_TRUE(rtp_packet.has_incoming()); | |
| 218 EXPECT_EQ(incoming, rtp_packet.incoming()); | |
| 219 ASSERT_TRUE(rtp_packet.has_type()); | |
| 220 EXPECT_EQ(media_type, GetRuntimeMediaType(rtp_packet.type())); | |
| 221 ASSERT_TRUE(rtp_packet.has_packet_length()); | |
| 222 EXPECT_EQ(total_size, rtp_packet.packet_length()); | |
| 223 ASSERT_TRUE(rtp_packet.has_header()); | |
| 224 ASSERT_EQ(header_size, rtp_packet.header().size()); | |
| 225 for (size_t i = 0; i < header_size; i++) { | |
| 226 EXPECT_EQ(header[i], static_cast<uint8_t>(rtp_packet.header()[i])); | |
| 227 } | |
| 228 } | |
| 229 | |
| 230 void VerifyRtcpEvent(const rtclog::Event& event, | |
| 231 bool incoming, | |
| 232 MediaType media_type, | |
| 233 uint8_t* packet, | |
| 234 size_t total_size) { | |
| 235 ASSERT_TRUE(IsValidBasicEvent(event)); | |
| 236 ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type()); | |
| 237 const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet(); | |
| 238 ASSERT_TRUE(rtcp_packet.has_incoming()); | |
| 239 EXPECT_EQ(incoming, rtcp_packet.incoming()); | |
| 240 ASSERT_TRUE(rtcp_packet.has_type()); | |
| 241 EXPECT_EQ(media_type, GetRuntimeMediaType(rtcp_packet.type())); | |
| 242 ASSERT_TRUE(rtcp_packet.has_packet_data()); | |
| 243 ASSERT_EQ(total_size, rtcp_packet.packet_data().size()); | |
| 244 for (size_t i = 0; i < total_size; i++) { | |
| 245 EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i])); | |
| 246 } | |
| 247 } | |
| 248 | |
| 249 void VerifyLogStartEvent(const rtclog::Event& event) { | |
| 250 ASSERT_TRUE(IsValidBasicEvent(event)); | |
| 251 ASSERT_EQ(rtclog::Event::DEBUG_EVENT, event.type()); | |
| 252 const rtclog::DebugEvent& debug_event = event.debug_event(); | |
| 253 ASSERT_TRUE(debug_event.has_type()); | |
| 254 EXPECT_EQ(rtclog::DebugEvent::LOG_START, debug_event.type()); | |
| 255 } | |
| 256 | |
| 257 void GenerateVideoReceiveConfig(VideoReceiveStream::Config* config) { | |
| 258 // Create a map from a payload type to an encoder name. | |
| 259 VideoReceiveStream::Decoder decoder; | |
| 260 decoder.payload_type = rand(); | |
| 261 decoder.payload_name = (rand() % 2 ? "VP8" : "H264"); | |
| 262 config->decoders.push_back(decoder); | |
| 263 // Add SSRCs for the stream. | |
| 264 config->rtp.remote_ssrc = rand(); | |
| 265 config->rtp.local_ssrc = rand(); | |
| 266 // Add extensions and settings for RTCP. | |
| 267 config->rtp.rtcp_mode = rand() % 2 ? newapi::kRtcpCompound | |
| 268 : newapi::kRtcpReducedSize; | |
| 269 config->rtp.rtcp_xr.receiver_reference_time_report = | |
| 270 static_cast<bool>(rand() % 2); | |
| 271 config->rtp.remb = static_cast<bool>(rand() % 2); | |
| 272 // Add a map from a payload type to a new ssrc and a new payload type for RTX. | |
| 273 VideoReceiveStream::Config::Rtp::Rtx rtx_pair; | |
| 274 rtx_pair.ssrc = rand(); | |
| 275 rtx_pair.payload_type = rand(); | |
| 276 config->rtp.rtx.insert(std::make_pair(rand(), rtx_pair)); | |
| 277 // Add two random header extensions. | |
| 278 const char* extension_name = rand() % 2 ? RtpExtension::kTOffset | |
| 279 : RtpExtension::kVideoRotation; | |
| 280 config->rtp.extensions.push_back(RtpExtension(extension_name, rand())); | |
| 281 extension_name = rand() % 2 ? RtpExtension::kAudioLevel | |
| 282 : RtpExtension::kAbsSendTime; | |
| 283 config->rtp.extensions.push_back(RtpExtension(extension_name, rand())); | |
| 284 } | |
| 285 | |
| 286 void GenerateVideoSendConfig(VideoSendStream::Config* config) { | |
| 287 // Create a map from a payload type to an encoder name. | |
| 288 config->encoder_settings.payload_type = rand(); | |
| 289 config->encoder_settings.payload_name = (rand() % 2 ? "VP8" : "H264"); | |
| 290 // Add SSRCs for the stream. | |
| 291 config->rtp.ssrcs.push_back(rand()); | |
| 292 // Add a map from a payload type to new ssrcs and a new payload type for RTX. | |
| 293 config->rtp.rtx.ssrcs.push_back(rand()); | |
| 294 config->rtp.rtx.payload_type = rand(); | |
| 295 // Add a CNAME. | |
| 296 config->rtp.c_name = "some.user@some.host"; | |
| 297 // Add two random header extensions. | |
| 298 const char* extension_name = rand() % 2 ? RtpExtension::kTOffset | |
| 299 : RtpExtension::kVideoRotation; | |
| 300 config->rtp.extensions.push_back(RtpExtension(extension_name, rand())); | |
| 301 extension_name = rand() % 2 ? RtpExtension::kAudioLevel | |
| 302 : RtpExtension::kAbsSendTime; | |
| 303 config->rtp.extensions.push_back(RtpExtension(extension_name, rand())); | |
| 304 } | |
| 305 | |
| 306 // Test for the RtcEventLog class. Dumps some RTP packets to disk, then reads | |
| 307 // them back to see if they match. | |
| 308 void LogSessionAndReadBack(size_t rtp_count, unsigned random_seed) { | |
| 309 std::vector<std::vector<uint8_t>> rtp_packets; | |
| 310 std::vector<uint8_t> incoming_rtcp_packet; | |
| 311 std::vector<uint8_t> outgoing_rtcp_packet; | |
| 312 | |
| 313 VideoReceiveStream::Config receiver_config; | |
| 314 VideoSendStream::Config sender_config; | |
| 315 | |
| 316 srand(random_seed); | |
| 317 | |
| 318 // Create rtp_count RTP packets containing random data. | |
| 319 const size_t rtp_header_size = 20; | |
| 320 for (size_t i = 0; i < rtp_count; i++) { | |
| 321 size_t packet_size = 1000 + rand() % 30; | |
| 322 rtp_packets.push_back(std::vector<uint8_t>()); | |
| 323 rtp_packets[i].reserve(packet_size); | |
| 324 for (size_t j = 0; j < packet_size; j++) { | |
| 325 rtp_packets[i].push_back(rand()); | |
| 326 } | |
| 327 } | |
| 328 // Create two RTCP packets containing random data. | |
| 329 size_t packet_size = 1000 + rand() % 30; | |
| 330 outgoing_rtcp_packet.reserve(packet_size); | |
| 331 for (size_t j = 0; j < packet_size; j++) { | |
| 332 outgoing_rtcp_packet.push_back(rand()); | |
| 333 } | |
| 334 packet_size = 1000 + rand() % 30; | |
| 335 incoming_rtcp_packet.reserve(packet_size); | |
| 336 for (size_t j = 0; j < packet_size; j++) { | |
| 337 incoming_rtcp_packet.push_back(rand()); | |
| 338 } | |
| 339 // Create configurations for the video streams. | |
| 340 GenerateVideoReceiveConfig(&receiver_config); | |
| 341 GenerateVideoSendConfig(&sender_config); | |
| 342 | |
| 343 // Find the name of the current test, in order to use it as a temporary | |
| 344 // filename. | |
| 345 auto test_info = ::testing::UnitTest::GetInstance()->current_test_info(); | |
| 346 const std::string temp_filename = | |
| 347 test::OutputPath() + test_info->test_case_name() + test_info->name(); | |
| 348 | |
| 349 // When log_dumper goes out of scope, it causes the log file to be flushed | |
| 350 // to disk. | |
| 351 { | |
| 352 rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create()); | |
| 353 log_dumper->LogVideoReceiveStreamConfig(receiver_config); | |
| 354 log_dumper->LogVideoSendStreamConfig(sender_config); | |
| 355 size_t i = 0; | |
| 356 for (; i < rtp_count / 2; i++) { | |
| 357 log_dumper->LogRtpHeader( | |
| 358 (i % 2 == 0), // Every second packet is incoming. | |
| 359 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, | |
| 360 rtp_packets[i].data(), rtp_header_size, rtp_packets[i].size()); | |
| 361 } | |
| 362 log_dumper->LogRtcpPacket(false, MediaType::AUDIO, | |
| 363 outgoing_rtcp_packet.data(), | |
| 364 outgoing_rtcp_packet.size()); | |
| 365 log_dumper->StartLogging(temp_filename, 10000000); | |
| 366 for (; i < rtp_count; i++) { | |
| 367 log_dumper->LogRtpHeader( | |
| 368 (i % 2 == 0), // Every second packet is incoming, | |
| 369 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, | |
| 370 rtp_packets[i].data(), rtp_header_size, rtp_packets[i].size()); | |
| 371 } | |
| 372 log_dumper->LogRtcpPacket(true, MediaType::VIDEO, | |
| 373 incoming_rtcp_packet.data(), | |
| 374 incoming_rtcp_packet.size()); | |
| 375 } | |
| 376 | |
| 377 const int config_count = 2; | |
| 378 const int rtcp_count = 2; | |
| 379 const int debug_count = 1; // Only LogStart event, | |
| 380 const int event_count = config_count + debug_count + rtcp_count + rtp_count; | |
| 381 | |
| 382 // Read the generated file from disk. | |
| 383 rtclog::EventStream parsed_stream; | |
| 384 | |
| 385 ASSERT_TRUE(RtcEventLog::ParseRtcEventLog(temp_filename, &parsed_stream)); | |
| 386 | |
| 387 // Verify the result. | |
| 388 EXPECT_EQ(event_count, parsed_stream.stream_size()); | |
| 389 VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config); | |
| 390 VerifySendStreamConfig(parsed_stream.stream(1), sender_config); | |
| 391 size_t i = 0; | |
| 392 for (; i < rtp_count / 2; i++) { | |
| 393 VerifyRtpEvent(parsed_stream.stream(config_count + i), | |
| 394 (i % 2 == 0), // Every second packet is incoming. | |
| 395 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, | |
| 396 rtp_packets[i].data(), rtp_header_size, | |
| 397 rtp_packets[i].size()); | |
| 398 } | |
| 399 VerifyRtcpEvent(parsed_stream.stream(config_count + rtp_count / 2), | |
| 400 false, // Outgoing RTCP packet. | |
| 401 MediaType::AUDIO, outgoing_rtcp_packet.data(), | |
| 402 outgoing_rtcp_packet.size()); | |
| 403 | |
| 404 VerifyLogStartEvent(parsed_stream.stream(1 + config_count + rtp_count / 2)); | |
| 405 for (; i < rtp_count; i++) { | |
| 406 VerifyRtpEvent(parsed_stream.stream(2 + config_count + i), | |
| 407 (i % 2 == 0), // Every second packet is incoming. | |
| 408 (i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, | |
| 409 rtp_packets[i].data(), rtp_header_size, | |
| 410 rtp_packets[i].size()); | |
| 411 } | |
| 412 VerifyRtcpEvent(parsed_stream.stream(2 + config_count + rtp_count), | |
| 413 true, // Incoming RTCP packet. | |
| 414 MediaType::VIDEO, incoming_rtcp_packet.data(), | |
| 415 incoming_rtcp_packet.size()); | |
| 416 | |
| 417 // Clean up temporary file - can be pretty slow. | |
| 418 remove(temp_filename.c_str()); | |
| 419 } | |
| 420 | |
| 421 TEST(RtcEventLogTest, LogSessionAndReadBack) { | |
| 422 LogSessionAndReadBack(5, 321); | |
| 423 LogSessionAndReadBack(8, 3141592653u); | |
| 424 LogSessionAndReadBack(9, 2718281828u); | |
| 425 } | |
| 426 | |
| 427 } // namespace webrtc | |
| 428 | |
| 429 #endif // ENABLE_RTC_EVENT_LOG | |
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Chromium Code Reviews
Issue 1250383003:
Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific t… (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master