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| 1 /* |
| 2 * Copyright (c) 2017 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 "webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h" |
| 12 |
| 13 #include "webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation
.h" |
| 14 #include "webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h" |
| 15 #include "webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_con
fig.h" |
| 16 #include "webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config
.h" |
| 17 #include "webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
" |
| 18 #include "webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h" |
| 19 #include "webrtc/logging/rtc_event_log/events/rtc_event_logging_started.h" |
| 20 #include "webrtc/logging/rtc_event_log/events/rtc_event_logging_stopped.h" |
| 21 #include "webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h" |
| 22 #include "webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h" |
| 23 #include "webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h" |
| 24 #include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" |
| 25 #include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h" |
| 26 #include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" |
| 27 #include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" |
| 28 #include "webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_con
fig.h" |
| 29 #include "webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config
.h" |
| 30 #include "webrtc/logging/rtc_event_log/rtc_stream_config.h" |
| 31 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_networ
k_adaptor_config.h" |
| 32 #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" |
| 33 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| 34 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" |
| 35 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| 36 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
| 37 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
| 38 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| 39 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h" |
| 40 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| 41 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" |
| 42 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h" |
| 43 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| 44 #include "webrtc/modules/rtp_rtcp/source/rtp_packet.h" |
| 45 #include "webrtc/rtc_base/checks.h" |
| 46 #include "webrtc/rtc_base/ignore_wundef.h" |
| 47 #include "webrtc/rtc_base/logging.h" |
| 48 |
| 49 #ifdef ENABLE_RTC_EVENT_LOG |
| 50 // *.pb.h files are generated at build-time by the protobuf compiler. |
| 51 RTC_PUSH_IGNORING_WUNDEF() |
| 52 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| 53 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
| 54 #else |
| 55 #include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
| 56 #endif |
| 57 RTC_POP_IGNORING_WUNDEF() |
| 58 #endif |
| 59 |
| 60 namespace webrtc { |
| 61 |
| 62 namespace { |
| 63 rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState( |
| 64 BandwidthUsage state) { |
| 65 switch (state) { |
| 66 case BandwidthUsage::kBwNormal: |
| 67 return rtclog::DelayBasedBweUpdate::BWE_NORMAL; |
| 68 case BandwidthUsage::kBwUnderusing: |
| 69 return rtclog::DelayBasedBweUpdate::BWE_UNDERUSING; |
| 70 case BandwidthUsage::kBwOverusing: |
| 71 return rtclog::DelayBasedBweUpdate::BWE_OVERUSING; |
| 72 } |
| 73 RTC_NOTREACHED(); |
| 74 return rtclog::DelayBasedBweUpdate::BWE_NORMAL; |
| 75 } |
| 76 |
| 77 rtclog::BweProbeResult::ResultType ConvertProbeResultType( |
| 78 ProbeFailureReason failure_reason) { |
| 79 switch (failure_reason) { |
| 80 case kInvalidSendReceiveInterval: |
| 81 return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL; |
| 82 case kInvalidSendReceiveRatio: |
| 83 return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO; |
| 84 case kTimeout: |
| 85 return rtclog::BweProbeResult::TIMEOUT; |
| 86 } |
| 87 RTC_NOTREACHED(); |
| 88 return rtclog::BweProbeResult::SUCCESS; |
| 89 } |
| 90 |
| 91 rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) { |
| 92 switch (rtcp_mode) { |
| 93 case RtcpMode::kCompound: |
| 94 return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| 95 case RtcpMode::kReducedSize: |
| 96 return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE; |
| 97 case RtcpMode::kOff: |
| 98 RTC_NOTREACHED(); |
| 99 return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| 100 } |
| 101 RTC_NOTREACHED(); |
| 102 return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| 103 } |
| 104 } // namespace |
| 105 |
| 106 std::string RtcEventLogEncoderLegacy::Encode(const RtcEvent& event) { |
| 107 switch (event.GetType()) { |
| 108 case RtcEvent::Type::AudioNetworkAdaptation: { |
| 109 auto& rtc_event = |
| 110 static_cast<const RtcEventAudioNetworkAdaptation&>(event); |
| 111 return AudioNetworkAdaptation(rtc_event); |
| 112 } |
| 113 |
| 114 case RtcEvent::Type::AudioPlayout: { |
| 115 auto& rtc_event = static_cast<const RtcEventAudioPlayout&>(event); |
| 116 return AudioPlayout(rtc_event); |
| 117 } |
| 118 |
| 119 case RtcEvent::Type::AudioReceiveStreamConfig: { |
| 120 auto& rtc_event = |
| 121 static_cast<const RtcEventAudioReceiveStreamConfig&>(event); |
| 122 return AudioReceiveStreamConfig(rtc_event); |
| 123 } |
| 124 |
| 125 case RtcEvent::Type::AudioSendStreamConfig: { |
| 126 auto& rtc_event = |
| 127 static_cast<const RtcEventAudioSendStreamConfig&>(event); |
| 128 return AudioSendStreamConfig(rtc_event); |
| 129 } |
| 130 |
| 131 case RtcEvent::Type::BweUpdateDelayBased: { |
| 132 auto& rtc_event = static_cast<const RtcEventBweUpdateDelayBased&>(event); |
| 133 return BweUpdateDelayBased(rtc_event); |
| 134 } |
| 135 |
| 136 case RtcEvent::Type::BweUpdateLossBased: { |
| 137 auto& rtc_event = static_cast<const RtcEventBweUpdateLossBased&>(event); |
| 138 return BweUpdateLossBased(rtc_event); |
| 139 } |
| 140 |
| 141 case RtcEvent::Type::LoggingStarted: { |
| 142 auto& rtc_event = static_cast<const RtcEventLoggingStarted&>(event); |
| 143 return LoggingStarted(rtc_event); |
| 144 } |
| 145 |
| 146 case RtcEvent::Type::LoggingStopped: { |
| 147 auto& rtc_event = static_cast<const RtcEventLoggingStopped&>(event); |
| 148 return LoggingStopped(rtc_event); |
| 149 } |
| 150 |
| 151 case RtcEvent::Type::ProbeClusterCreated: { |
| 152 auto& rtc_event = static_cast<const RtcEventProbeClusterCreated&>(event); |
| 153 return ProbeClusterCreated(rtc_event); |
| 154 } |
| 155 |
| 156 case RtcEvent::Type::ProbeResultFailure: { |
| 157 auto& rtc_event = static_cast<const RtcEventProbeResultFailure&>(event); |
| 158 return ProbeResultFailure(rtc_event); |
| 159 } |
| 160 |
| 161 case RtcEvent::Type::ProbeResultSuccess: { |
| 162 auto& rtc_event = static_cast<const RtcEventProbeResultSuccess&>(event); |
| 163 return ProbeResultSuccess(rtc_event); |
| 164 } |
| 165 |
| 166 case RtcEvent::Type::RtcpPacketIncoming: { |
| 167 auto& rtc_event = static_cast<const RtcEventRtcpPacketIncoming&>(event); |
| 168 return RtcpPacketIncoming(rtc_event); |
| 169 } |
| 170 |
| 171 case RtcEvent::Type::RtcpPacketOutgoing: { |
| 172 auto& rtc_event = static_cast<const RtcEventRtcpPacketOutgoing&>(event); |
| 173 return RtcpPacketOutgoing(rtc_event); |
| 174 } |
| 175 |
| 176 case RtcEvent::Type::RtpPacketIncoming: { |
| 177 auto& rtc_event = static_cast<const RtcEventRtpPacketIncoming&>(event); |
| 178 return RtpPacketIncoming(rtc_event); |
| 179 } |
| 180 |
| 181 case RtcEvent::Type::RtpPacketOutgoing: { |
| 182 auto& rtc_event = static_cast<const RtcEventRtpPacketOutgoing&>(event); |
| 183 return RtpPacketOutgoing(rtc_event); |
| 184 } |
| 185 |
| 186 case RtcEvent::Type::VideoReceiveStreamConfig: { |
| 187 auto& rtc_event = |
| 188 static_cast<const RtcEventVideoReceiveStreamConfig&>(event); |
| 189 return VideoReceiveStreamConfig(rtc_event); |
| 190 } |
| 191 |
| 192 case RtcEvent::Type::VideoSendStreamConfig: { |
| 193 auto& rtc_event = |
| 194 static_cast<const RtcEventVideoSendStreamConfig&>(event); |
| 195 return VideoSendStreamConfig(rtc_event); |
| 196 } |
| 197 } |
| 198 |
| 199 int event_type = static_cast<int>(event.GetType()); |
| 200 RTC_NOTREACHED() << "Unknown event type (" << event_type << ")"; |
| 201 } |
| 202 |
| 203 std::string RtcEventLogEncoderLegacy::AudioNetworkAdaptation( |
| 204 const RtcEventAudioNetworkAdaptation& event) { |
| 205 rtclog::Event rtclog_event; |
| 206 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 207 rtclog_event.set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT); |
| 208 |
| 209 auto audio_network_adaptation = |
| 210 rtclog_event.mutable_audio_network_adaptation(); |
| 211 if (event.config_->bitrate_bps) |
| 212 audio_network_adaptation->set_bitrate_bps(*event.config_->bitrate_bps); |
| 213 if (event.config_->frame_length_ms) |
| 214 audio_network_adaptation->set_frame_length_ms( |
| 215 *event.config_->frame_length_ms); |
| 216 if (event.config_->uplink_packet_loss_fraction) { |
| 217 audio_network_adaptation->set_uplink_packet_loss_fraction( |
| 218 *event.config_->uplink_packet_loss_fraction); |
| 219 } |
| 220 if (event.config_->enable_fec) |
| 221 audio_network_adaptation->set_enable_fec(*event.config_->enable_fec); |
| 222 if (event.config_->enable_dtx) |
| 223 audio_network_adaptation->set_enable_dtx(*event.config_->enable_dtx); |
| 224 if (event.config_->num_channels) |
| 225 audio_network_adaptation->set_num_channels(*event.config_->num_channels); |
| 226 |
| 227 return Serialize(&rtclog_event); |
| 228 } |
| 229 |
| 230 std::string RtcEventLogEncoderLegacy::AudioPlayout( |
| 231 const RtcEventAudioPlayout& event) { |
| 232 rtclog::Event rtclog_event; |
| 233 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 234 rtclog_event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT); |
| 235 |
| 236 auto playout_event = rtclog_event.mutable_audio_playout_event(); |
| 237 playout_event->set_local_ssrc(event.ssrc_); |
| 238 |
| 239 return Serialize(&rtclog_event); |
| 240 } |
| 241 |
| 242 std::string RtcEventLogEncoderLegacy::AudioReceiveStreamConfig( |
| 243 const RtcEventAudioReceiveStreamConfig& event) { |
| 244 rtclog::Event rtclog_event; |
| 245 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 246 rtclog_event.set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT); |
| 247 |
| 248 rtclog::AudioReceiveConfig* receiver_config = |
| 249 rtclog_event.mutable_audio_receiver_config(); |
| 250 receiver_config->set_remote_ssrc(event.config_->remote_ssrc); |
| 251 receiver_config->set_local_ssrc(event.config_->local_ssrc); |
| 252 |
| 253 for (const auto& e : event.config_->rtp_extensions) { |
| 254 rtclog::RtpHeaderExtension* extension = |
| 255 receiver_config->add_header_extensions(); |
| 256 extension->set_name(e.uri); |
| 257 extension->set_id(e.id); |
| 258 } |
| 259 |
| 260 return Serialize(&rtclog_event); |
| 261 } |
| 262 |
| 263 std::string RtcEventLogEncoderLegacy::AudioSendStreamConfig( |
| 264 const RtcEventAudioSendStreamConfig& event) { |
| 265 rtclog::Event rtclog_event; |
| 266 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 267 rtclog_event.set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT); |
| 268 |
| 269 rtclog::AudioSendConfig* sender_config = |
| 270 rtclog_event.mutable_audio_sender_config(); |
| 271 |
| 272 sender_config->set_ssrc(event.config_->local_ssrc); |
| 273 |
| 274 for (const auto& e : event.config_->rtp_extensions) { |
| 275 rtclog::RtpHeaderExtension* extension = |
| 276 sender_config->add_header_extensions(); |
| 277 extension->set_name(e.uri); |
| 278 extension->set_id(e.id); |
| 279 } |
| 280 |
| 281 return Serialize(&rtclog_event); |
| 282 } |
| 283 |
| 284 std::string RtcEventLogEncoderLegacy::BweUpdateDelayBased( |
| 285 const RtcEventBweUpdateDelayBased& event) { |
| 286 rtclog::Event rtclog_event; |
| 287 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 288 rtclog_event.set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE); |
| 289 |
| 290 auto bwe_event = rtclog_event.mutable_delay_based_bwe_update(); |
| 291 bwe_event->set_bitrate_bps(event.bitrate_bps_); |
| 292 bwe_event->set_detector_state(ConvertDetectorState(event.detector_state_)); |
| 293 |
| 294 return Serialize(&rtclog_event); |
| 295 } |
| 296 |
| 297 std::string RtcEventLogEncoderLegacy::BweUpdateLossBased( |
| 298 const RtcEventBweUpdateLossBased& event) { |
| 299 rtclog::Event rtclog_event; |
| 300 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 301 rtclog_event.set_type(rtclog::Event::LOSS_BASED_BWE_UPDATE); |
| 302 |
| 303 auto bwe_event = rtclog_event.mutable_loss_based_bwe_update(); |
| 304 bwe_event->set_bitrate_bps(event.bitrate_bps_); |
| 305 bwe_event->set_fraction_loss(event.fraction_loss_); |
| 306 bwe_event->set_total_packets(event.total_packets_); |
| 307 |
| 308 return Serialize(&rtclog_event); |
| 309 } |
| 310 |
| 311 std::string RtcEventLogEncoderLegacy::LoggingStarted( |
| 312 const RtcEventLoggingStarted& event) { |
| 313 rtclog::Event rtclog_event; |
| 314 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 315 rtclog_event.set_type(rtclog::Event::LOG_START); |
| 316 return Serialize(&rtclog_event); |
| 317 } |
| 318 |
| 319 std::string RtcEventLogEncoderLegacy::LoggingStopped( |
| 320 const RtcEventLoggingStopped& event) { |
| 321 rtclog::Event rtclog_event; |
| 322 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 323 rtclog_event.set_type(rtclog::Event::LOG_END); |
| 324 return Serialize(&rtclog_event); |
| 325 } |
| 326 |
| 327 std::string RtcEventLogEncoderLegacy::ProbeClusterCreated( |
| 328 const RtcEventProbeClusterCreated& event) { |
| 329 rtclog::Event rtclog_event; |
| 330 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 331 rtclog_event.set_type(rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT); |
| 332 |
| 333 auto probe_cluster = rtclog_event.mutable_probe_cluster(); |
| 334 probe_cluster->set_id(event.id_); |
| 335 probe_cluster->set_bitrate_bps(event.bitrate_bps_); |
| 336 probe_cluster->set_min_packets(event.min_probes_); |
| 337 probe_cluster->set_min_bytes(event.min_bytes_); |
| 338 |
| 339 return Serialize(&rtclog_event); |
| 340 } |
| 341 |
| 342 std::string RtcEventLogEncoderLegacy::ProbeResultFailure( |
| 343 const RtcEventProbeResultFailure& event) { |
| 344 rtclog::Event rtclog_event; |
| 345 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 346 rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT); |
| 347 |
| 348 auto probe_result = rtclog_event.mutable_probe_result(); |
| 349 probe_result->set_id(event.id_); |
| 350 probe_result->set_result(ConvertProbeResultType(event.failure_reason_)); |
| 351 |
| 352 return Serialize(&rtclog_event); |
| 353 } |
| 354 |
| 355 std::string RtcEventLogEncoderLegacy::ProbeResultSuccess( |
| 356 const RtcEventProbeResultSuccess& event) { |
| 357 rtclog::Event rtclog_event; |
| 358 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 359 rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT); |
| 360 |
| 361 auto probe_result = rtclog_event.mutable_probe_result(); |
| 362 probe_result->set_id(event.id_); |
| 363 probe_result->set_result(rtclog::BweProbeResult::SUCCESS); |
| 364 probe_result->set_bitrate_bps(event.bitrate_bps_); |
| 365 |
| 366 return Serialize(&rtclog_event); |
| 367 } |
| 368 |
| 369 std::string RtcEventLogEncoderLegacy::RtcpPacketIncoming( |
| 370 const RtcEventRtcpPacketIncoming& event) { |
| 371 return RtcpPacket(event.timestamp_us_, event.packet_, true); |
| 372 } |
| 373 |
| 374 std::string RtcEventLogEncoderLegacy::RtcpPacketOutgoing( |
| 375 const RtcEventRtcpPacketOutgoing& event) { |
| 376 return RtcpPacket(event.timestamp_us_, event.packet_, false); |
| 377 } |
| 378 |
| 379 std::string RtcEventLogEncoderLegacy::RtpPacketIncoming( |
| 380 const RtcEventRtpPacketIncoming& event) { |
| 381 return RtpPacket(event.timestamp_us_, event.header_, event.packet_length_, |
| 382 true); |
| 383 } |
| 384 |
| 385 std::string RtcEventLogEncoderLegacy::RtpPacketOutgoing( |
| 386 const RtcEventRtpPacketOutgoing& event) { |
| 387 return RtpPacket(event.timestamp_us_, event.header_, event.packet_length_, |
| 388 false); |
| 389 } |
| 390 |
| 391 std::string RtcEventLogEncoderLegacy::VideoReceiveStreamConfig( |
| 392 const RtcEventVideoReceiveStreamConfig& event) { |
| 393 rtclog::Event rtclog_event; |
| 394 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 395 rtclog_event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT); |
| 396 |
| 397 rtclog::VideoReceiveConfig* receiver_config = |
| 398 rtclog_event.mutable_video_receiver_config(); |
| 399 receiver_config->set_remote_ssrc(event.config_->remote_ssrc); |
| 400 receiver_config->set_local_ssrc(event.config_->local_ssrc); |
| 401 |
| 402 // TODO(perkj): Add field for rsid. |
| 403 receiver_config->set_rtcp_mode(ConvertRtcpMode(event.config_->rtcp_mode)); |
| 404 receiver_config->set_remb(event.config_->remb); |
| 405 |
| 406 for (const auto& e : event.config_->rtp_extensions) { |
| 407 rtclog::RtpHeaderExtension* extension = |
| 408 receiver_config->add_header_extensions(); |
| 409 extension->set_name(e.uri); |
| 410 extension->set_id(e.id); |
| 411 } |
| 412 |
| 413 for (const auto& d : event.config_->codecs) { |
| 414 rtclog::DecoderConfig* decoder = receiver_config->add_decoders(); |
| 415 decoder->set_name(d.payload_name); |
| 416 decoder->set_payload_type(d.payload_type); |
| 417 if (d.rtx_payload_type != 0) { |
| 418 rtclog::RtxMap* rtx = receiver_config->add_rtx_map(); |
| 419 rtx->set_payload_type(d.payload_type); |
| 420 rtx->mutable_config()->set_rtx_ssrc(event.config_->rtx_ssrc); |
| 421 rtx->mutable_config()->set_rtx_payload_type(d.rtx_payload_type); |
| 422 } |
| 423 } |
| 424 |
| 425 return Serialize(&rtclog_event); |
| 426 } |
| 427 |
| 428 std::string RtcEventLogEncoderLegacy::VideoSendStreamConfig( |
| 429 const RtcEventVideoSendStreamConfig& event) { |
| 430 rtclog::Event rtclog_event; |
| 431 rtclog_event.set_timestamp_us(event.timestamp_us_); |
| 432 rtclog_event.set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT); |
| 433 |
| 434 rtclog::VideoSendConfig* sender_config = |
| 435 rtclog_event.mutable_video_sender_config(); |
| 436 |
| 437 // TODO(perkj): rtclog::VideoSendConfig should only contain one SSRC. |
| 438 sender_config->add_ssrcs(event.config_->local_ssrc); |
| 439 if (event.config_->rtx_ssrc != 0) { |
| 440 sender_config->add_rtx_ssrcs(event.config_->rtx_ssrc); |
| 441 } |
| 442 |
| 443 for (const auto& e : event.config_->rtp_extensions) { |
| 444 rtclog::RtpHeaderExtension* extension = |
| 445 sender_config->add_header_extensions(); |
| 446 extension->set_name(e.uri); |
| 447 extension->set_id(e.id); |
| 448 } |
| 449 |
| 450 // TODO(perkj): rtclog::VideoSendConfig should contain many possible codec |
| 451 // configurations. |
| 452 for (const auto& codec : event.config_->codecs) { |
| 453 sender_config->set_rtx_payload_type(codec.rtx_payload_type); |
| 454 rtclog::EncoderConfig* encoder = sender_config->mutable_encoder(); |
| 455 encoder->set_name(codec.payload_name); |
| 456 encoder->set_payload_type(codec.payload_type); |
| 457 |
| 458 if (event.config_->codecs.size() > 1) { |
| 459 LOG(WARNING) << "LogVideoSendStreamConfig currently only supports one " |
| 460 << "codec. Logging codec :" << codec.payload_name; |
| 461 break; |
| 462 } |
| 463 } |
| 464 |
| 465 return Serialize(&rtclog_event); |
| 466 } |
| 467 |
| 468 std::string RtcEventLogEncoderLegacy::RtcpPacket(int64_t timestamp_us, |
| 469 const rtc::Buffer& packet, |
| 470 bool is_incoming) { |
| 471 rtclog::Event rtclog_event; |
| 472 rtclog_event.set_timestamp_us(timestamp_us); |
| 473 rtclog_event.set_type(rtclog::Event::RTCP_EVENT); |
| 474 rtclog_event.mutable_rtcp_packet()->set_incoming(is_incoming); |
| 475 |
| 476 rtcp::CommonHeader header; |
| 477 const uint8_t* block_begin = packet.data(); |
| 478 const uint8_t* packet_end = packet.data() + packet.size(); |
| 479 RTC_DCHECK(packet.size() <= IP_PACKET_SIZE); |
| 480 uint8_t buffer[IP_PACKET_SIZE]; |
| 481 uint32_t buffer_length = 0; |
| 482 while (block_begin < packet_end) { |
| 483 if (!header.Parse(block_begin, packet_end - block_begin)) { |
| 484 break; // Incorrect message header. |
| 485 } |
| 486 const uint8_t* next_block = header.NextPacket(); |
| 487 uint32_t block_size = next_block - block_begin; |
| 488 switch (header.type()) { |
| 489 case rtcp::Bye::kPacketType: |
| 490 case rtcp::ExtendedJitterReport::kPacketType: |
| 491 case rtcp::ExtendedReports::kPacketType: |
| 492 case rtcp::Psfb::kPacketType: |
| 493 case rtcp::ReceiverReport::kPacketType: |
| 494 case rtcp::Rtpfb::kPacketType: |
| 495 case rtcp::SenderReport::kPacketType: |
| 496 // We log sender reports, receiver reports, bye messages |
| 497 // inter-arrival jitter, third-party loss reports, payload-specific |
| 498 // feedback and extended reports. |
| 499 memcpy(buffer + buffer_length, block_begin, block_size); |
| 500 buffer_length += block_size; |
| 501 break; |
| 502 case rtcp::App::kPacketType: |
| 503 case rtcp::Sdes::kPacketType: |
| 504 default: |
| 505 // We don't log sender descriptions, application defined messages |
| 506 // or message blocks of unknown type. |
| 507 break; |
| 508 } |
| 509 |
| 510 block_begin += block_size; |
| 511 } |
| 512 rtclog_event.mutable_rtcp_packet()->set_packet_data(buffer, buffer_length); |
| 513 |
| 514 return Serialize(&rtclog_event); |
| 515 } |
| 516 |
| 517 std::string RtcEventLogEncoderLegacy::RtpPacket(int64_t timestamp_us, |
| 518 const rtp::Packet& header, |
| 519 size_t packet_length, |
| 520 bool is_incoming) { |
| 521 rtclog::Event rtclog_event; |
| 522 rtclog_event.set_timestamp_us(timestamp_us); |
| 523 rtclog_event.set_type(rtclog::Event::RTP_EVENT); |
| 524 |
| 525 rtclog_event.mutable_rtp_packet()->set_incoming(is_incoming); |
| 526 rtclog_event.mutable_rtp_packet()->set_packet_length(packet_length); |
| 527 rtclog_event.mutable_rtp_packet()->set_header(header.data(), header.size()); |
| 528 |
| 529 return Serialize(&rtclog_event); |
| 530 } |
| 531 |
| 532 std::string RtcEventLogEncoderLegacy::Serialize(rtclog::Event* event) { |
| 533 // Even though we're only serializing a single event during this call, what |
| 534 // we intend to get is a list of events, with a tag and length preceding |
| 535 // each actual event. To produce that, we serialize a list of a single event. |
| 536 // If we later concatenate several results from this function, the result will |
| 537 // be a proper concatenation of all those events. |
| 538 |
| 539 rtclog::EventStream event_stream; |
| 540 event_stream.add_stream(); |
| 541 |
| 542 // As a tweak, we swap the new event into the event-stream, write that to |
| 543 // file, then swap back. This saves on some copying, while making sure that |
| 544 // the caller wouldn't be surprised by Serialize() modifying the object. |
| 545 rtclog::Event* output_event = event_stream.mutable_stream(0); |
| 546 output_event->Swap(event); |
| 547 |
| 548 std::string output_string; |
| 549 event_stream.AppendToString(&output_string); |
| 550 |
| 551 // When the function returns, the original Event will be unchanged. |
| 552 output_event->Swap(event); |
| 553 |
| 554 return output_string; |
| 555 } |
| 556 |
| 557 } // namespace webrtc |
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