| /* |
| * 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 "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_logging_started.h" |
| #include "logging/rtc_event_log/events/rtc_event_logging_stopped.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/remote_bitrate_estimator/include/bwe_defines.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" |
| |
| #ifdef ENABLE_RTC_EVENT_LOG |
| |
| // *.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; |
| } |
| } // namespace |
| |
| 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::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::LoggingStarted: { |
| auto& rtc_event = static_cast<const RtcEventLoggingStarted&>(event); |
| return EncodeLoggingStarted(rtc_event); |
| } |
| |
| case RtcEvent::Type::LoggingStopped: { |
| auto& rtc_event = static_cast<const RtcEventLoggingStopped&>(event); |
| return EncodeLoggingStopped(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); |
| } |
| } |
| |
| int event_type = static_cast<int>(event.GetType()); |
| RTC_NOTREACHED() << "Unknown event type (" << event_type << ")"; |
| return ""; |
| } |
| |
| 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::EncodeLoggingStarted( |
| const RtcEventLoggingStarted& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us_); |
| rtclog_event.set_type(rtclog::Event::LOG_START); |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeLoggingStopped( |
| const RtcEventLoggingStopped& event) { |
| rtclog::Event rtclog_event; |
| rtclog_event.set_timestamp_us(event.timestamp_us_); |
| rtclog_event.set_type(rtclog::Event::LOG_END); |
| return Serialize(&rtclog_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.header_, |
| event.packet_length_, PacedPacketInfo::kNotAProbe, |
| true); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtpPacketOutgoing( |
| const RtcEventRtpPacketOutgoing& event) { |
| return EncodeRtpPacket(event.timestamp_us_, event.header_, |
| 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(); |
| RTC_DCHECK(packet.size() <= IP_PACKET_SIZE); |
| uint8_t buffer[IP_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 + 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, buffer_length); |
| |
| return Serialize(&rtclog_event); |
| } |
| |
| std::string RtcEventLogEncoderLegacy::EncodeRtpPacket( |
| int64_t timestamp_us, |
| const webrtc::RtpPacket& 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 |
| |
| #endif // ENABLE_RTC_EVENT_LOG |