| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h" |
| |
| #include <string.h> |
| |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/network_state_predictor.h" |
| #include "api/rtp_headers.h" |
| #include "api/rtp_parameters.h" |
| #include "api/transport/network_types.h" |
| #include "logging/rtc_event_log/events/rtc_event_alr_state.h" |
| #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" |
| #include "logging/rtc_event_log/events/rtc_event_audio_playout.h" |
| #include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h" |
| #include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h" |
| #include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h" |
| #include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h" |
| #include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h" |
| #include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h" |
| #include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h" |
| #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h" |
| #include "logging/rtc_event_log/events/rtc_event_probe_result_success.h" |
| #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" |
| #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h" |
| #include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" |
| #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" |
| #include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h" |
| #include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h" |
| #include "logging/rtc_event_log/rtc_stream_config.h" |
| #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/app.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/bye.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/psfb.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/sdes.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/ignore_wundef.h" |
| #include "rtc_base/logging.h" |
| |
| // *.pb.h files are generated at build-time by the protobuf compiler. |
| RTC_PUSH_IGNORING_WUNDEF() |
| #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h" |
| #else |
| #include "logging/rtc_event_log/rtc_event_log.pb.h" |
| #endif |
| RTC_POP_IGNORING_WUNDEF() |
| |
| namespace webrtc { |
| |
| namespace { |
| rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState( |
| BandwidthUsage state) { |
| switch (state) { |
| case BandwidthUsage::kBwNormal: |
| return rtclog::DelayBasedBweUpdate::BWE_NORMAL; |
| case BandwidthUsage::kBwUnderusing: |
| return rtclog::DelayBasedBweUpdate::BWE_UNDERUSING; |
| case BandwidthUsage::kBwOverusing: |
| return rtclog::DelayBasedBweUpdate::BWE_OVERUSING; |
| case BandwidthUsage::kLast: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::DelayBasedBweUpdate::BWE_NORMAL; |
| } |
| |
| rtclog::BweProbeResult::ResultType ConvertProbeResultType( |
| ProbeFailureReason failure_reason) { |
| switch (failure_reason) { |
| case ProbeFailureReason::kInvalidSendReceiveInterval: |
| return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL; |
| case ProbeFailureReason::kInvalidSendReceiveRatio: |
| return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO; |
| case ProbeFailureReason::kTimeout: |
| return rtclog::BweProbeResult::TIMEOUT; |
| case ProbeFailureReason::kLast: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::BweProbeResult::SUCCESS; |
| } |
| |
| rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) { |
| switch (rtcp_mode) { |
| case RtcpMode::kCompound: |
| return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| case RtcpMode::kReducedSize: |
| return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE; |
| case RtcpMode::kOff: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::VideoReceiveConfig::RTCP_COMPOUND; |
| } |
| |
| rtclog::IceCandidatePairConfig::IceCandidatePairConfigType |
| ConvertIceCandidatePairConfigType(IceCandidatePairConfigType type) { |
| switch (type) { |
| case IceCandidatePairConfigType::kAdded: |
| return rtclog::IceCandidatePairConfig::ADDED; |
| case IceCandidatePairConfigType::kUpdated: |
| return rtclog::IceCandidatePairConfig::UPDATED; |
| case IceCandidatePairConfigType::kDestroyed: |
| return rtclog::IceCandidatePairConfig::DESTROYED; |
| case IceCandidatePairConfigType::kSelected: |
| return rtclog::IceCandidatePairConfig::SELECTED; |
| case IceCandidatePairConfigType::kNumValues: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::IceCandidatePairConfig::ADDED; |
| } |
| |
| rtclog::IceCandidatePairConfig::IceCandidateType ConvertIceCandidateType( |
| IceCandidateType type) { |
| switch (type) { |
| case IceCandidateType::kUnknown: |
| return rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE; |
| case IceCandidateType::kLocal: |
| return rtclog::IceCandidatePairConfig::LOCAL; |
| case IceCandidateType::kStun: |
| return rtclog::IceCandidatePairConfig::STUN; |
| case IceCandidateType::kPrflx: |
| return rtclog::IceCandidatePairConfig::PRFLX; |
| case IceCandidateType::kRelay: |
| return rtclog::IceCandidatePairConfig::RELAY; |
| case IceCandidateType::kNumValues: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE; |
| } |
| |
| rtclog::IceCandidatePairConfig::Protocol ConvertIceCandidatePairProtocol( |
| IceCandidatePairProtocol protocol) { |
| switch (protocol) { |
| case IceCandidatePairProtocol::kUnknown: |
| return rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL; |
| case IceCandidatePairProtocol::kUdp: |
| return rtclog::IceCandidatePairConfig::UDP; |
| case IceCandidatePairProtocol::kTcp: |
| return rtclog::IceCandidatePairConfig::TCP; |
| case IceCandidatePairProtocol::kSsltcp: |
| return rtclog::IceCandidatePairConfig::SSLTCP; |
| case IceCandidatePairProtocol::kTls: |
| return rtclog::IceCandidatePairConfig::TLS; |
| case IceCandidatePairProtocol::kNumValues: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL; |
| } |
| |
| rtclog::IceCandidatePairConfig::AddressFamily |
| ConvertIceCandidatePairAddressFamily( |
| IceCandidatePairAddressFamily address_family) { |
| switch (address_family) { |
| case IceCandidatePairAddressFamily::kUnknown: |
| return rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY; |
| case IceCandidatePairAddressFamily::kIpv4: |
| return rtclog::IceCandidatePairConfig::IPV4; |
| case IceCandidatePairAddressFamily::kIpv6: |
| return rtclog::IceCandidatePairConfig::IPV6; |
| case IceCandidatePairAddressFamily::kNumValues: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY; |
| } |
| |
| rtclog::IceCandidatePairConfig::NetworkType ConvertIceCandidateNetworkType( |
| IceCandidateNetworkType network_type) { |
| switch (network_type) { |
| case IceCandidateNetworkType::kUnknown: |
| return rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE; |
| case IceCandidateNetworkType::kEthernet: |
| return rtclog::IceCandidatePairConfig::ETHERNET; |
| case IceCandidateNetworkType::kLoopback: |
| return rtclog::IceCandidatePairConfig::LOOPBACK; |
| case IceCandidateNetworkType::kWifi: |
| return rtclog::IceCandidatePairConfig::WIFI; |
| case IceCandidateNetworkType::kVpn: |
| return rtclog::IceCandidatePairConfig::VPN; |
| case IceCandidateNetworkType::kCellular: |
| return rtclog::IceCandidatePairConfig::CELLULAR; |
| case IceCandidateNetworkType::kNumValues: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE; |
| } |
| |
| rtclog::IceCandidatePairEvent::IceCandidatePairEventType |
| ConvertIceCandidatePairEventType(IceCandidatePairEventType type) { |
| switch (type) { |
| case IceCandidatePairEventType::kCheckSent: |
| return rtclog::IceCandidatePairEvent::CHECK_SENT; |
| case IceCandidatePairEventType::kCheckReceived: |
| return rtclog::IceCandidatePairEvent::CHECK_RECEIVED; |
| case IceCandidatePairEventType::kCheckResponseSent: |
| return rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT; |
| case IceCandidatePairEventType::kCheckResponseReceived: |
| return rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED; |
| case IceCandidatePairEventType::kNumValues: |
| RTC_NOTREACHED(); |
| } |
| RTC_NOTREACHED(); |
| return rtclog::IceCandidatePairEvent::CHECK_SENT; |
| } |
| |
| } // namespace |
| |
| std::string RtcEventLogEncoderLegacy::EncodeLogStart(int64_t timestamp_us, |
| int64_t utc_time_us) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(timestamp_us); |
| rtclog_event.set_type(rtclog::Event::LOG_START); |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeLogEnd(int64_t timestamp_us) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(timestamp_us); |
| rtclog_event.set_type(rtclog::Event::LOG_END); |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeBatch( |
| std::deque<std::unique_ptr<RtcEvent>>::const_iterator begin, |
| std::deque<std::unique_ptr<RtcEvent>>::const_iterator end) { |
| std::string encoded_output; |
| for (auto it = begin; it != end; ++it) { |
| // TODO(terelius): Can we avoid the slight inefficiency of reallocating the |
| // string? |
| RTC_CHECK(it->get() != nullptr); |
| encoded_output += Encode(**it); |
| } |
| return encoded_output; |
| } |
| |
| std::string RtcEventLogEncoderLegacy::Encode(const RtcEvent& event) { |
| switch (event.GetType()) { |
| case RtcEvent::Type::AudioNetworkAdaptation: { |
| auto& rtc_event = |
| static_cast<const RtcEventAudioNetworkAdaptation&>(event); |
| return EncodeAudioNetworkAdaptation(rtc_event); |
| } |
| |
| case RtcEvent::Type::AlrStateEvent: { |
| auto& rtc_event = static_cast<const RtcEventAlrState&>(event); |
| return EncodeAlrState(rtc_event); |
| } |
| |
| case RtcEvent::Type::AudioPlayout: { |
| auto& rtc_event = static_cast<const RtcEventAudioPlayout&>(event); |
| return EncodeAudioPlayout(rtc_event); |
| } |
| |
| case RtcEvent::Type::AudioReceiveStreamConfig: { |
| auto& rtc_event = |
| static_cast<const RtcEventAudioReceiveStreamConfig&>(event); |
| return EncodeAudioReceiveStreamConfig(rtc_event); |
| } |
| |
| case RtcEvent::Type::AudioSendStreamConfig: { |
| auto& rtc_event = |
| static_cast<const RtcEventAudioSendStreamConfig&>(event); |
| return EncodeAudioSendStreamConfig(rtc_event); |
| } |
| |
| case RtcEvent::Type::BweUpdateDelayBased: { |
| auto& rtc_event = static_cast<const RtcEventBweUpdateDelayBased&>(event); |
| return EncodeBweUpdateDelayBased(rtc_event); |
| } |
| |
| case RtcEvent::Type::BweUpdateLossBased: { |
| auto& rtc_event = static_cast<const RtcEventBweUpdateLossBased&>(event); |
| return EncodeBweUpdateLossBased(rtc_event); |
| } |
| |
| case RtcEvent::Type::DtlsTransportState: { |
| return ""; |
| } |
| |
| case RtcEvent::Type::DtlsWritableState: { |
| return ""; |
| } |
| |
| case RtcEvent::Type::IceCandidatePairConfig: { |
| auto& rtc_event = |
| static_cast<const RtcEventIceCandidatePairConfig&>(event); |
| return EncodeIceCandidatePairConfig(rtc_event); |
| } |
| |
| case RtcEvent::Type::IceCandidatePairEvent: { |
| auto& rtc_event = static_cast<const RtcEventIceCandidatePair&>(event); |
| return EncodeIceCandidatePairEvent(rtc_event); |
| } |
| |
| case RtcEvent::Type::ProbeClusterCreated: { |
| auto& rtc_event = static_cast<const RtcEventProbeClusterCreated&>(event); |
| return EncodeProbeClusterCreated(rtc_event); |
| } |
| |
| case RtcEvent::Type::ProbeResultFailure: { |
| auto& rtc_event = static_cast<const RtcEventProbeResultFailure&>(event); |
| return EncodeProbeResultFailure(rtc_event); |
| } |
| |
| case RtcEvent::Type::ProbeResultSuccess: { |
| auto& rtc_event = static_cast<const RtcEventProbeResultSuccess&>(event); |
| return EncodeProbeResultSuccess(rtc_event); |
| } |
| |
| case RtcEvent::Type::RtcpPacketIncoming: { |
| auto& rtc_event = static_cast<const RtcEventRtcpPacketIncoming&>(event); |
| return EncodeRtcpPacketIncoming(rtc_event); |
| } |
| |
| case RtcEvent::Type::RtcpPacketOutgoing: { |
| auto& rtc_event = static_cast<const RtcEventRtcpPacketOutgoing&>(event); |
| return EncodeRtcpPacketOutgoing(rtc_event); |
| } |
| |
| case RtcEvent::Type::RtpPacketIncoming: { |
| auto& rtc_event = static_cast<const RtcEventRtpPacketIncoming&>(event); |
| return EncodeRtpPacketIncoming(rtc_event); |
| } |
| |
| case RtcEvent::Type::RtpPacketOutgoing: { |
| auto& rtc_event = static_cast<const RtcEventRtpPacketOutgoing&>(event); |
| return EncodeRtpPacketOutgoing(rtc_event); |
| } |
| |
| case RtcEvent::Type::VideoReceiveStreamConfig: { |
| auto& rtc_event = |
| static_cast<const RtcEventVideoReceiveStreamConfig&>(event); |
| return EncodeVideoReceiveStreamConfig(rtc_event); |
| } |
| |
| case RtcEvent::Type::VideoSendStreamConfig: { |
| auto& rtc_event = |
| static_cast<const RtcEventVideoSendStreamConfig&>(event); |
| return EncodeVideoSendStreamConfig(rtc_event); |
| } |
| case RtcEvent::Type::RouteChangeEvent: |
| case RtcEvent::Type::RemoteEstimateEvent: |
| case RtcEvent::Type::GenericPacketReceived: |
| case RtcEvent::Type::GenericPacketSent: |
| case RtcEvent::Type::GenericAckReceived: |
| case RtcEvent::Type::FrameDecoded: |
| // These are unsupported in the old format, but shouldn't crash. |
| return ""; |
| } |
| |
| int event_type = static_cast<int>(event.GetType()); |
| RTC_NOTREACHED() << "Unknown event type (" << event_type << ")"; |
| return ""; |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeAlrState( |
| const RtcEventAlrState& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::ALR_STATE_EVENT); |
| |
| auto* alr_state = rtclog_event.mutable_alr_state(); |
| alr_state->set_in_alr(event.in_alr()); |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeAudioNetworkAdaptation( |
| const RtcEventAudioNetworkAdaptation& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT); |
| |
| auto* audio_network_adaptation = |
| rtclog_event.mutable_audio_network_adaptation(); |
| if (event.config().bitrate_bps) |
| audio_network_adaptation->set_bitrate_bps(*event.config().bitrate_bps); |
| if (event.config().frame_length_ms) |
| audio_network_adaptation->set_frame_length_ms( |
| *event.config().frame_length_ms); |
| if (event.config().uplink_packet_loss_fraction) { |
| audio_network_adaptation->set_uplink_packet_loss_fraction( |
| *event.config().uplink_packet_loss_fraction); |
| } |
| if (event.config().enable_fec) |
| audio_network_adaptation->set_enable_fec(*event.config().enable_fec); |
| if (event.config().enable_dtx) |
| audio_network_adaptation->set_enable_dtx(*event.config().enable_dtx); |
| if (event.config().num_channels) |
| audio_network_adaptation->set_num_channels(*event.config().num_channels); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeAudioPlayout( |
| const RtcEventAudioPlayout& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT); |
| |
| auto* playout_event = rtclog_event.mutable_audio_playout_event(); |
| playout_event->set_local_ssrc(event.ssrc()); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeAudioReceiveStreamConfig( |
| const RtcEventAudioReceiveStreamConfig& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT); |
| |
| rtclog::AudioReceiveConfig* receiver_config = |
| rtclog_event.mutable_audio_receiver_config(); |
| receiver_config->set_remote_ssrc(event.config().remote_ssrc); |
| receiver_config->set_local_ssrc(event.config().local_ssrc); |
| |
| for (const auto& e : event.config().rtp_extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| receiver_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeAudioSendStreamConfig( |
| const RtcEventAudioSendStreamConfig& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT); |
| |
| rtclog::AudioSendConfig* sender_config = |
| rtclog_event.mutable_audio_sender_config(); |
| |
| sender_config->set_ssrc(event.config().local_ssrc); |
| |
| for (const auto& e : event.config().rtp_extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| sender_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeBweUpdateDelayBased( |
| const RtcEventBweUpdateDelayBased& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE); |
| |
| auto* bwe_event = rtclog_event.mutable_delay_based_bwe_update(); |
| bwe_event->set_bitrate_bps(event.bitrate_bps()); |
| bwe_event->set_detector_state(ConvertDetectorState(event.detector_state())); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeBweUpdateLossBased( |
| const RtcEventBweUpdateLossBased& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::LOSS_BASED_BWE_UPDATE); |
| |
| auto* bwe_event = rtclog_event.mutable_loss_based_bwe_update(); |
| bwe_event->set_bitrate_bps(event.bitrate_bps()); |
| bwe_event->set_fraction_loss(event.fraction_loss()); |
| bwe_event->set_total_packets(event.total_packets()); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeIceCandidatePairConfig( |
| const RtcEventIceCandidatePairConfig& event) { |
| rtclog::Event encoded_rtc_event; |
| encoded_rtc_event.set_timestamp_us(event.timestamp_us()); |
| encoded_rtc_event.set_type(rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG); |
| |
| auto* encoded_ice_event = |
| encoded_rtc_event.mutable_ice_candidate_pair_config(); |
| encoded_ice_event->set_config_type( |
| ConvertIceCandidatePairConfigType(event.type())); |
| encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id()); |
| const auto& desc = event.candidate_pair_desc(); |
| encoded_ice_event->set_local_candidate_type( |
| ConvertIceCandidateType(desc.local_candidate_type)); |
| encoded_ice_event->set_local_relay_protocol( |
| ConvertIceCandidatePairProtocol(desc.local_relay_protocol)); |
| encoded_ice_event->set_local_network_type( |
| ConvertIceCandidateNetworkType(desc.local_network_type)); |
| encoded_ice_event->set_local_address_family( |
| ConvertIceCandidatePairAddressFamily(desc.local_address_family)); |
| encoded_ice_event->set_remote_candidate_type( |
| ConvertIceCandidateType(desc.remote_candidate_type)); |
| encoded_ice_event->set_remote_address_family( |
| ConvertIceCandidatePairAddressFamily(desc.remote_address_family)); |
| encoded_ice_event->set_candidate_pair_protocol( |
| ConvertIceCandidatePairProtocol(desc.candidate_pair_protocol)); |
| return Serialize(&encoded_rtc_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeIceCandidatePairEvent( |
| const RtcEventIceCandidatePair& event) { |
| rtclog::Event encoded_rtc_event; |
| encoded_rtc_event.set_timestamp_us(event.timestamp_us()); |
| encoded_rtc_event.set_type(rtclog::Event::ICE_CANDIDATE_PAIR_EVENT); |
| |
| auto* encoded_ice_event = |
| encoded_rtc_event.mutable_ice_candidate_pair_event(); |
| encoded_ice_event->set_event_type( |
| ConvertIceCandidatePairEventType(event.type())); |
| encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id()); |
| return Serialize(&encoded_rtc_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeProbeClusterCreated( |
| const RtcEventProbeClusterCreated& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT); |
| |
| auto* probe_cluster = rtclog_event.mutable_probe_cluster(); |
| probe_cluster->set_id(event.id()); |
| probe_cluster->set_bitrate_bps(event.bitrate_bps()); |
| probe_cluster->set_min_packets(event.min_probes()); |
| probe_cluster->set_min_bytes(event.min_bytes()); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeProbeResultFailure( |
| const RtcEventProbeResultFailure& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT); |
| |
| auto* probe_result = rtclog_event.mutable_probe_result(); |
| probe_result->set_id(event.id()); |
| probe_result->set_result(ConvertProbeResultType(event.failure_reason())); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeProbeResultSuccess( |
| const RtcEventProbeResultSuccess& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT); |
| |
| auto* probe_result = rtclog_event.mutable_probe_result(); |
| probe_result->set_id(event.id()); |
| probe_result->set_result(rtclog::BweProbeResult::SUCCESS); |
| probe_result->set_bitrate_bps(event.bitrate_bps()); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtcpPacketIncoming( |
| const RtcEventRtcpPacketIncoming& event) { |
| return EncodeRtcpPacket(event.timestamp_us(), event.packet(), true); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtcpPacketOutgoing( |
| const RtcEventRtcpPacketOutgoing& event) { |
| return EncodeRtcpPacket(event.timestamp_us(), event.packet(), false); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtpPacketIncoming( |
| const RtcEventRtpPacketIncoming& event) { |
| return EncodeRtpPacket(event.timestamp_us(), event.RawHeader(), |
| event.packet_length(), PacedPacketInfo::kNotAProbe, |
| true); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtpPacketOutgoing( |
| const RtcEventRtpPacketOutgoing& event) { |
| return EncodeRtpPacket(event.timestamp_us(), event.RawHeader(), |
| event.packet_length(), event.probe_cluster_id(), |
| false); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeVideoReceiveStreamConfig( |
| const RtcEventVideoReceiveStreamConfig& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT); |
| |
| rtclog::VideoReceiveConfig* receiver_config = |
| rtclog_event.mutable_video_receiver_config(); |
| receiver_config->set_remote_ssrc(event.config().remote_ssrc); |
| receiver_config->set_local_ssrc(event.config().local_ssrc); |
| |
| // TODO(perkj): Add field for rsid. |
| receiver_config->set_rtcp_mode(ConvertRtcpMode(event.config().rtcp_mode)); |
| receiver_config->set_remb(event.config().remb); |
| |
| for (const auto& e : event.config().rtp_extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| receiver_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| |
| for (const auto& d : event.config().codecs) { |
| rtclog::DecoderConfig* decoder = receiver_config->add_decoders(); |
| decoder->set_name(d.payload_name); |
| decoder->set_payload_type(d.payload_type); |
| if (d.rtx_payload_type != 0) { |
| rtclog::RtxMap* rtx = receiver_config->add_rtx_map(); |
| rtx->set_payload_type(d.payload_type); |
| rtx->mutable_config()->set_rtx_ssrc(event.config().rtx_ssrc); |
| rtx->mutable_config()->set_rtx_payload_type(d.rtx_payload_type); |
| } |
| } |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeVideoSendStreamConfig( |
| const RtcEventVideoSendStreamConfig& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us()); |
| rtclog_event.set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT); |
| |
| rtclog::VideoSendConfig* sender_config = |
| rtclog_event.mutable_video_sender_config(); |
| |
| // TODO(perkj): rtclog::VideoSendConfig should only contain one SSRC. |
| sender_config->add_ssrcs(event.config().local_ssrc); |
| if (event.config().rtx_ssrc != 0) { |
| sender_config->add_rtx_ssrcs(event.config().rtx_ssrc); |
| } |
| |
| for (const auto& e : event.config().rtp_extensions) { |
| rtclog::RtpHeaderExtension* extension = |
| sender_config->add_header_extensions(); |
| extension->set_name(e.uri); |
| extension->set_id(e.id); |
| } |
| |
| // TODO(perkj): rtclog::VideoSendConfig should contain many possible codec |
| // configurations. |
| for (const auto& codec : event.config().codecs) { |
| sender_config->set_rtx_payload_type(codec.rtx_payload_type); |
| rtclog::EncoderConfig* encoder = sender_config->mutable_encoder(); |
| encoder->set_name(codec.payload_name); |
| encoder->set_payload_type(codec.payload_type); |
| |
| if (event.config().codecs.size() > 1) { |
| RTC_LOG(WARNING) |
| << "LogVideoSendStreamConfig currently only supports one " |
| "codec. Logging codec :" |
| << codec.payload_name; |
| break; |
| } |
| } |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtcpPacket( |
| int64_t timestamp_us, |
| const rtc::Buffer& packet, |
| bool is_incoming) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(timestamp_us); |
| rtclog_event.set_type(rtclog::Event::RTCP_EVENT); |
| rtclog_event.mutable_rtcp_packet()->set_incoming(is_incoming); |
| |
| rtcp::CommonHeader header; |
| const uint8_t* block_begin = packet.data(); |
| const uint8_t* packet_end = packet.data() + packet.size(); |
| std::vector<uint8_t> buffer(packet.size()); |
| uint32_t buffer_length = 0; |
| while (block_begin < packet_end) { |
| if (!header.Parse(block_begin, packet_end - block_begin)) { |
| break; // Incorrect message header. |
| } |
| const uint8_t* next_block = header.NextPacket(); |
| uint32_t block_size = next_block - block_begin; |
| switch (header.type()) { |
| case rtcp::Bye::kPacketType: |
| case rtcp::ExtendedJitterReport::kPacketType: |
| case rtcp::ExtendedReports::kPacketType: |
| case rtcp::Psfb::kPacketType: |
| case rtcp::ReceiverReport::kPacketType: |
| case rtcp::Rtpfb::kPacketType: |
| case rtcp::SenderReport::kPacketType: |
| // We log sender reports, receiver reports, bye messages |
| // inter-arrival jitter, third-party loss reports, payload-specific |
| // feedback and extended reports. |
| memcpy(buffer.data() + buffer_length, block_begin, block_size); |
| buffer_length += block_size; |
| break; |
| case rtcp::App::kPacketType: |
| case rtcp::Sdes::kPacketType: |
| default: |
| // We don't log sender descriptions, application defined messages |
| // or message blocks of unknown type. |
| break; |
| } |
| |
| block_begin += block_size; |
| } |
| rtclog_event.mutable_rtcp_packet()->set_packet_data(buffer.data(), |
| buffer_length); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtpPacket( |
| int64_t timestamp_us, |
| rtc::ArrayView<const uint8_t> header, |
| size_t packet_length, |
| int probe_cluster_id, |
| bool is_incoming) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(timestamp_us); |
| rtclog_event.set_type(rtclog::Event::RTP_EVENT); |
| |
| rtclog_event.mutable_rtp_packet()->set_incoming(is_incoming); |
| rtclog_event.mutable_rtp_packet()->set_packet_length(packet_length); |
| rtclog_event.mutable_rtp_packet()->set_header(header.data(), header.size()); |
| if (probe_cluster_id != PacedPacketInfo::kNotAProbe) { |
| RTC_DCHECK(!is_incoming); |
| rtclog_event.mutable_rtp_packet()->set_probe_cluster_id(probe_cluster_id); |
| } |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::Serialize(rtclog::Event* event) { |
| // Even though we're only serializing a single event during this call, what |
| // we intend to get is a list of events, with a tag and length preceding |
| // each actual event. To produce that, we serialize a list of a single event. |
| // If we later concatenate several results from this function, the result will |
| // be a proper concatenation of all those events. |
| |
| rtclog::EventStream event_stream; |
| event_stream.add_stream(); |
| |
| // As a tweak, we swap the new event into the event-stream, write that to |
| // file, then swap back. This saves on some copying, while making sure that |
| // the caller wouldn't be surprised by Serialize() modifying the object. |
| rtclog::Event* output_event = event_stream.mutable_stream(0); |
| output_event->Swap(event); |
| |
| std::string output_string = event_stream.SerializeAsString(); |
| RTC_DCHECK(!output_string.empty()); |
| |
| // When the function returns, the original Event will be unchanged. |
| output_event->Swap(event); |
| |
| return output_string; |
| } |
| |
| } // namespace webrtc |