diff options
Diffstat (limited to 'third_party/libwebrtc/pc/rtc_stats_collector.cc')
| -rw-r--r-- | third_party/libwebrtc/pc/rtc_stats_collector.cc | 2201 |
1 files changed, 2201 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/rtc_stats_collector.cc b/third_party/libwebrtc/pc/rtc_stats_collector.cc new file mode 100644 index 0000000000..0797ba2a76 --- /dev/null +++ b/third_party/libwebrtc/pc/rtc_stats_collector.cc @@ -0,0 +1,2201 @@ +/* + * Copyright 2016 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 "pc/rtc_stats_collector.h" + +#include <stdint.h> +#include <stdio.h> + +#include <cstdint> +#include <map> +#include <memory> +#include <set> +#include <string> +#include <type_traits> +#include <utility> +#include <vector> + +#include "absl/functional/bind_front.h" +#include "absl/strings/string_view.h" +#include "api/array_view.h" +#include "api/candidate.h" +#include "api/dtls_transport_interface.h" +#include "api/media_stream_interface.h" +#include "api/media_types.h" +#include "api/rtp_parameters.h" +#include "api/sequence_checker.h" +#include "api/stats/rtc_stats.h" +#include "api/stats/rtcstats_objects.h" +#include "api/units/time_delta.h" +#include "api/video/video_content_type.h" +#include "api/video_codecs/scalability_mode.h" +#include "common_video/include/quality_limitation_reason.h" +#include "media/base/media_channel.h" +#include "media/base/media_channel_impl.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing_statistics.h" +#include "modules/rtp_rtcp/include/report_block_data.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "p2p/base/connection_info.h" +#include "p2p/base/ice_transport_internal.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/port.h" +#include "pc/channel_interface.h" +#include "pc/data_channel_utils.h" +#include "pc/rtc_stats_traversal.h" +#include "pc/rtp_receiver_proxy.h" +#include "pc/rtp_sender_proxy.h" +#include "pc/webrtc_sdp.h" +#include "rtc_base/checks.h" +#include "rtc_base/ip_address.h" +#include "rtc_base/logging.h" +#include "rtc_base/network_constants.h" +#include "rtc_base/rtc_certificate.h" +#include "rtc_base/socket_address.h" +#include "rtc_base/ssl_stream_adapter.h" +#include "rtc_base/string_encode.h" +#include "rtc_base/strings/string_builder.h" +#include "rtc_base/time_utils.h" +#include "rtc_base/trace_event.h" + +namespace webrtc { + +namespace { + +const char kDirectionInbound = 'I'; +const char kDirectionOutbound = 'O'; + +const char* kAudioPlayoutSingletonId = "AP"; + +// TODO(https://crbug.com/webrtc/10656): Consider making IDs less predictable. +std::string RTCCertificateIDFromFingerprint(const std::string& fingerprint) { + return "CF" + fingerprint; +} + +// `direction` is either kDirectionInbound or kDirectionOutbound. +std::string RTCCodecStatsIDFromTransportAndCodecParameters( + const char direction, + const std::string& transport_id, + const RtpCodecParameters& codec_params) { + char buf[1024]; + rtc::SimpleStringBuilder sb(buf); + sb << 'C' << direction << transport_id << '_' << codec_params.payload_type; + // TODO(https://crbug.com/webrtc/14420): If we stop supporting different FMTP + // lines for the same PT and transport, which should be illegal SDP, then we + // wouldn't need `fmtp` to be part of the ID here. + rtc::StringBuilder fmtp; + if (WriteFmtpParameters(codec_params.parameters, &fmtp)) { + sb << '_' << fmtp.Release(); + } + return sb.str(); +} + +std::string RTCIceCandidatePairStatsIDFromConnectionInfo( + const cricket::ConnectionInfo& info) { + char buf[4096]; + rtc::SimpleStringBuilder sb(buf); + sb << "CP" << info.local_candidate.id() << "_" << info.remote_candidate.id(); + return sb.str(); +} + +std::string RTCTransportStatsIDFromTransportChannel( + const std::string& transport_name, + int channel_component) { + char buf[1024]; + rtc::SimpleStringBuilder sb(buf); + sb << 'T' << transport_name << channel_component; + return sb.str(); +} + +std::string RTCInboundRtpStreamStatsIDFromSSRC(const std::string& transport_id, + cricket::MediaType media_type, + uint32_t ssrc) { + char buf[1024]; + rtc::SimpleStringBuilder sb(buf); + sb << 'I' << transport_id + << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V') << ssrc; + return sb.str(); +} + +std::string RTCOutboundRtpStreamStatsIDFromSSRC(const std::string& transport_id, + cricket::MediaType media_type, + uint32_t ssrc) { + char buf[1024]; + rtc::SimpleStringBuilder sb(buf); + sb << 'O' << transport_id + << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V') << ssrc; + return sb.str(); +} + +std::string RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc( + cricket::MediaType media_type, + uint32_t source_ssrc) { + char buf[1024]; + rtc::SimpleStringBuilder sb(buf); + sb << "RI" << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V') + << source_ssrc; + return sb.str(); +} + +std::string RTCRemoteOutboundRTPStreamStatsIDFromSSRC( + cricket::MediaType media_type, + uint32_t source_ssrc) { + char buf[1024]; + rtc::SimpleStringBuilder sb(buf); + sb << "RO" << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V') + << source_ssrc; + return sb.str(); +} + +std::string RTCMediaSourceStatsIDFromKindAndAttachment( + cricket::MediaType media_type, + int attachment_id) { + char buf[1024]; + rtc::SimpleStringBuilder sb(buf); + sb << 'S' << (media_type == cricket::MEDIA_TYPE_AUDIO ? 'A' : 'V') + << attachment_id; + return sb.str(); +} + +const char* CandidateTypeToRTCIceCandidateType(const std::string& type) { + if (type == cricket::LOCAL_PORT_TYPE) + return "host"; + if (type == cricket::STUN_PORT_TYPE) + return "srflx"; + if (type == cricket::PRFLX_PORT_TYPE) + return "prflx"; + if (type == cricket::RELAY_PORT_TYPE) + return "relay"; + RTC_DCHECK_NOTREACHED(); + return nullptr; +} + +const char* DataStateToRTCDataChannelState( + DataChannelInterface::DataState state) { + switch (state) { + case DataChannelInterface::kConnecting: + return "connecting"; + case DataChannelInterface::kOpen: + return "open"; + case DataChannelInterface::kClosing: + return "closing"; + case DataChannelInterface::kClosed: + return "closed"; + default: + RTC_DCHECK_NOTREACHED(); + return nullptr; + } +} + +const char* IceCandidatePairStateToRTCStatsIceCandidatePairState( + cricket::IceCandidatePairState state) { + switch (state) { + case cricket::IceCandidatePairState::WAITING: + return "waiting"; + case cricket::IceCandidatePairState::IN_PROGRESS: + return "in-progress"; + case cricket::IceCandidatePairState::SUCCEEDED: + return "succeeded"; + case cricket::IceCandidatePairState::FAILED: + return "failed"; + default: + RTC_DCHECK_NOTREACHED(); + return nullptr; + } +} + +const char* IceRoleToRTCIceRole(cricket::IceRole role) { + switch (role) { + case cricket::IceRole::ICEROLE_UNKNOWN: + return "unknown"; + case cricket::IceRole::ICEROLE_CONTROLLED: + return "controlled"; + case cricket::IceRole::ICEROLE_CONTROLLING: + return "controlling"; + default: + RTC_DCHECK_NOTREACHED(); + return nullptr; + } +} + +const char* DtlsTransportStateToRTCDtlsTransportState( + DtlsTransportState state) { + switch (state) { + case DtlsTransportState::kNew: + return "new"; + case DtlsTransportState::kConnecting: + return "connecting"; + case DtlsTransportState::kConnected: + return "connected"; + case DtlsTransportState::kClosed: + return "closed"; + case DtlsTransportState::kFailed: + return "failed"; + default: + RTC_CHECK_NOTREACHED(); + return nullptr; + } +} + +const char* IceTransportStateToRTCIceTransportState(IceTransportState state) { + switch (state) { + case IceTransportState::kNew: + return "new"; + case IceTransportState::kChecking: + return "checking"; + case IceTransportState::kConnected: + return "connected"; + case IceTransportState::kCompleted: + return "completed"; + case IceTransportState::kFailed: + return "failed"; + case IceTransportState::kDisconnected: + return "disconnected"; + case IceTransportState::kClosed: + return "closed"; + default: + RTC_CHECK_NOTREACHED(); + return nullptr; + } +} + +const char* NetworkTypeToStatsType(rtc::AdapterType type) { + switch (type) { + case rtc::ADAPTER_TYPE_CELLULAR: + case rtc::ADAPTER_TYPE_CELLULAR_2G: + case rtc::ADAPTER_TYPE_CELLULAR_3G: + case rtc::ADAPTER_TYPE_CELLULAR_4G: + case rtc::ADAPTER_TYPE_CELLULAR_5G: + return "cellular"; + case rtc::ADAPTER_TYPE_ETHERNET: + return "ethernet"; + case rtc::ADAPTER_TYPE_WIFI: + return "wifi"; + case rtc::ADAPTER_TYPE_VPN: + return "vpn"; + case rtc::ADAPTER_TYPE_UNKNOWN: + case rtc::ADAPTER_TYPE_LOOPBACK: + case rtc::ADAPTER_TYPE_ANY: + return "unknown"; + } + RTC_DCHECK_NOTREACHED(); + return nullptr; +} + +absl::string_view NetworkTypeToStatsNetworkAdapterType(rtc::AdapterType type) { + switch (type) { + case rtc::ADAPTER_TYPE_CELLULAR: + return "cellular"; + case rtc::ADAPTER_TYPE_CELLULAR_2G: + return "cellular2g"; + case rtc::ADAPTER_TYPE_CELLULAR_3G: + return "cellular3g"; + case rtc::ADAPTER_TYPE_CELLULAR_4G: + return "cellular4g"; + case rtc::ADAPTER_TYPE_CELLULAR_5G: + return "cellular5g"; + case rtc::ADAPTER_TYPE_ETHERNET: + return "ethernet"; + case rtc::ADAPTER_TYPE_WIFI: + return "wifi"; + case rtc::ADAPTER_TYPE_UNKNOWN: + return "unknown"; + case rtc::ADAPTER_TYPE_LOOPBACK: + return "loopback"; + case rtc::ADAPTER_TYPE_ANY: + return "any"; + case rtc::ADAPTER_TYPE_VPN: + /* should not be handled here. Vpn is modelled as a bool */ + break; + } + RTC_DCHECK_NOTREACHED(); + return {}; +} + +const char* QualityLimitationReasonToRTCQualityLimitationReason( + QualityLimitationReason reason) { + switch (reason) { + case QualityLimitationReason::kNone: + return "none"; + case QualityLimitationReason::kCpu: + return "cpu"; + case QualityLimitationReason::kBandwidth: + return "bandwidth"; + case QualityLimitationReason::kOther: + return "other"; + } + RTC_CHECK_NOTREACHED(); +} + +std::map<std::string, double> +QualityLimitationDurationToRTCQualityLimitationDuration( + std::map<webrtc::QualityLimitationReason, int64_t> durations_ms) { + std::map<std::string, double> result; + // The internal duration is defined in milliseconds while the spec defines + // the value in seconds: + // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations + for (const auto& elem : durations_ms) { + result[QualityLimitationReasonToRTCQualityLimitationReason(elem.first)] = + elem.second / static_cast<double>(rtc::kNumMillisecsPerSec); + } + return result; +} + +double DoubleAudioLevelFromIntAudioLevel(int audio_level) { + RTC_DCHECK_GE(audio_level, 0); + RTC_DCHECK_LE(audio_level, 32767); + return audio_level / 32767.0; +} + +// Gets the `codecId` identified by `transport_id` and `codec_params`. If no +// such `RTCCodecStats` exist yet, create it and add it to `report`. +std::string GetCodecIdAndMaybeCreateCodecStats( + Timestamp timestamp, + const char direction, + const std::string& transport_id, + const RtpCodecParameters& codec_params, + RTCStatsReport* report) { + RTC_DCHECK_GE(codec_params.payload_type, 0); + RTC_DCHECK_LE(codec_params.payload_type, 127); + RTC_DCHECK(codec_params.clock_rate); + uint32_t payload_type = static_cast<uint32_t>(codec_params.payload_type); + std::string codec_id = RTCCodecStatsIDFromTransportAndCodecParameters( + direction, transport_id, codec_params); + if (report->Get(codec_id) != nullptr) { + // The `RTCCodecStats` already exists. + return codec_id; + } + // Create the `RTCCodecStats` that we want to reference. + auto codec_stats = std::make_unique<RTCCodecStats>(codec_id, timestamp); + codec_stats->payload_type = payload_type; + codec_stats->mime_type = codec_params.mime_type(); + if (codec_params.clock_rate.has_value()) { + codec_stats->clock_rate = static_cast<uint32_t>(*codec_params.clock_rate); + } + if (codec_params.num_channels) { + codec_stats->channels = *codec_params.num_channels; + } + + rtc::StringBuilder fmtp; + if (WriteFmtpParameters(codec_params.parameters, &fmtp)) { + codec_stats->sdp_fmtp_line = fmtp.Release(); + } + codec_stats->transport_id = transport_id; + report->AddStats(std::move(codec_stats)); + return codec_id; +} + +// Provides the media independent counters (both audio and video). +void SetInboundRTPStreamStatsFromMediaReceiverInfo( + const cricket::MediaReceiverInfo& media_receiver_info, + RTCInboundRtpStreamStats* inbound_stats) { + RTC_DCHECK(inbound_stats); + inbound_stats->ssrc = media_receiver_info.ssrc(); + inbound_stats->packets_received = + static_cast<uint32_t>(media_receiver_info.packets_received); + inbound_stats->bytes_received = + static_cast<uint64_t>(media_receiver_info.payload_bytes_received); + inbound_stats->header_bytes_received = static_cast<uint64_t>( + media_receiver_info.header_and_padding_bytes_received); + if (media_receiver_info.retransmitted_bytes_received.has_value()) { + inbound_stats->retransmitted_bytes_received = + *media_receiver_info.retransmitted_bytes_received; + } + if (media_receiver_info.retransmitted_packets_received.has_value()) { + inbound_stats->retransmitted_packets_received = + *media_receiver_info.retransmitted_packets_received; + } + inbound_stats->packets_lost = + static_cast<int32_t>(media_receiver_info.packets_lost); + inbound_stats->jitter_buffer_delay = + media_receiver_info.jitter_buffer_delay_seconds; + inbound_stats->jitter_buffer_target_delay = + media_receiver_info.jitter_buffer_target_delay_seconds; + inbound_stats->jitter_buffer_minimum_delay = + media_receiver_info.jitter_buffer_minimum_delay_seconds; + inbound_stats->jitter_buffer_emitted_count = + media_receiver_info.jitter_buffer_emitted_count; + if (media_receiver_info.nacks_sent.has_value()) { + inbound_stats->nack_count = *media_receiver_info.nacks_sent; + } + if (media_receiver_info.fec_packets_received.has_value()) { + inbound_stats->fec_packets_received = + *media_receiver_info.fec_packets_received; + } + if (media_receiver_info.fec_packets_discarded.has_value()) { + inbound_stats->fec_packets_discarded = + *media_receiver_info.fec_packets_discarded; + } + if (media_receiver_info.fec_bytes_received.has_value()) { + inbound_stats->fec_bytes_received = *media_receiver_info.fec_bytes_received; + } +} + +std::unique_ptr<RTCInboundRtpStreamStats> CreateInboundAudioStreamStats( + const cricket::VoiceMediaInfo& voice_media_info, + const cricket::VoiceReceiverInfo& voice_receiver_info, + const std::string& transport_id, + const std::string& mid, + Timestamp timestamp, + RTCStatsReport* report) { + auto inbound_audio = std::make_unique<RTCInboundRtpStreamStats>( + /*id=*/RTCInboundRtpStreamStatsIDFromSSRC( + transport_id, cricket::MEDIA_TYPE_AUDIO, voice_receiver_info.ssrc()), + timestamp); + SetInboundRTPStreamStatsFromMediaReceiverInfo(voice_receiver_info, + inbound_audio.get()); + inbound_audio->transport_id = transport_id; + inbound_audio->mid = mid; + inbound_audio->kind = "audio"; + if (voice_receiver_info.codec_payload_type.has_value()) { + auto codec_param_it = voice_media_info.receive_codecs.find( + *voice_receiver_info.codec_payload_type); + RTC_DCHECK(codec_param_it != voice_media_info.receive_codecs.end()); + if (codec_param_it != voice_media_info.receive_codecs.end()) { + inbound_audio->codec_id = GetCodecIdAndMaybeCreateCodecStats( + inbound_audio->timestamp(), kDirectionInbound, transport_id, + codec_param_it->second, report); + } + } + inbound_audio->jitter = static_cast<double>(voice_receiver_info.jitter_ms) / + rtc::kNumMillisecsPerSec; + inbound_audio->total_samples_received = + voice_receiver_info.total_samples_received; + inbound_audio->concealed_samples = voice_receiver_info.concealed_samples; + inbound_audio->silent_concealed_samples = + voice_receiver_info.silent_concealed_samples; + inbound_audio->concealment_events = voice_receiver_info.concealment_events; + inbound_audio->inserted_samples_for_deceleration = + voice_receiver_info.inserted_samples_for_deceleration; + inbound_audio->removed_samples_for_acceleration = + voice_receiver_info.removed_samples_for_acceleration; + if (voice_receiver_info.audio_level >= 0) { + inbound_audio->audio_level = + DoubleAudioLevelFromIntAudioLevel(voice_receiver_info.audio_level); + } + inbound_audio->total_audio_energy = voice_receiver_info.total_output_energy; + inbound_audio->total_samples_duration = + voice_receiver_info.total_output_duration; + // `fir_count` and `pli_count` are only valid for video and are + // purposefully left undefined for audio. + if (voice_receiver_info.last_packet_received.has_value()) { + inbound_audio->last_packet_received_timestamp = + voice_receiver_info.last_packet_received->ms<double>(); + } + if (voice_receiver_info.estimated_playout_ntp_timestamp_ms.has_value()) { + // TODO(bugs.webrtc.org/10529): Fix time origin. + inbound_audio->estimated_playout_timestamp = static_cast<double>( + *voice_receiver_info.estimated_playout_ntp_timestamp_ms); + } + inbound_audio->packets_discarded = voice_receiver_info.packets_discarded; + inbound_audio->jitter_buffer_flushes = + voice_receiver_info.jitter_buffer_flushes; + inbound_audio->delayed_packet_outage_samples = + voice_receiver_info.delayed_packet_outage_samples; + inbound_audio->relative_packet_arrival_delay = + voice_receiver_info.relative_packet_arrival_delay_seconds; + inbound_audio->interruption_count = + voice_receiver_info.interruption_count >= 0 + ? voice_receiver_info.interruption_count + : 0; + inbound_audio->total_interruption_duration = + static_cast<double>(voice_receiver_info.total_interruption_duration_ms) / + rtc::kNumMillisecsPerSec; + return inbound_audio; +} + +std::unique_ptr<RTCAudioPlayoutStats> CreateAudioPlayoutStats( + const AudioDeviceModule::Stats& audio_device_stats, + webrtc::Timestamp timestamp) { + auto stats = std::make_unique<RTCAudioPlayoutStats>( + /*id=*/kAudioPlayoutSingletonId, timestamp); + stats->synthesized_samples_duration = + audio_device_stats.synthesized_samples_duration_s; + stats->synthesized_samples_events = + audio_device_stats.synthesized_samples_events; + stats->total_samples_count = audio_device_stats.total_samples_count; + stats->total_samples_duration = audio_device_stats.total_samples_duration_s; + stats->total_playout_delay = audio_device_stats.total_playout_delay_s; + return stats; +} + +std::unique_ptr<RTCRemoteOutboundRtpStreamStats> +CreateRemoteOutboundAudioStreamStats( + const cricket::VoiceReceiverInfo& voice_receiver_info, + const std::string& mid, + const RTCInboundRtpStreamStats& inbound_audio_stats, + const std::string& transport_id) { + if (!voice_receiver_info.last_sender_report_timestamp_ms.has_value()) { + // Cannot create `RTCRemoteOutboundRtpStreamStats` when the RTCP SR arrival + // timestamp is not available - i.e., until the first sender report is + // received. + return nullptr; + } + RTC_DCHECK_GT(voice_receiver_info.sender_reports_reports_count, 0); + + // Create. + auto stats = std::make_unique<RTCRemoteOutboundRtpStreamStats>( + /*id=*/RTCRemoteOutboundRTPStreamStatsIDFromSSRC( + cricket::MEDIA_TYPE_AUDIO, voice_receiver_info.ssrc()), + Timestamp::Millis(*voice_receiver_info.last_sender_report_timestamp_ms)); + + // Populate. + // - RTCRtpStreamStats. + stats->ssrc = voice_receiver_info.ssrc(); + stats->kind = "audio"; + stats->transport_id = transport_id; + if (inbound_audio_stats.codec_id.is_defined()) { + stats->codec_id = *inbound_audio_stats.codec_id; + } + // - RTCSentRtpStreamStats. + stats->packets_sent = voice_receiver_info.sender_reports_packets_sent; + stats->bytes_sent = voice_receiver_info.sender_reports_bytes_sent; + // - RTCRemoteOutboundRtpStreamStats. + stats->local_id = inbound_audio_stats.id(); + // last_sender_report_remote_timestamp_ms is set together with + // last_sender_report_timestamp_ms. + RTC_DCHECK( + voice_receiver_info.last_sender_report_remote_timestamp_ms.has_value()); + stats->remote_timestamp = static_cast<double>( + *voice_receiver_info.last_sender_report_remote_timestamp_ms); + stats->reports_sent = voice_receiver_info.sender_reports_reports_count; + if (voice_receiver_info.round_trip_time.has_value()) { + stats->round_trip_time = + voice_receiver_info.round_trip_time->seconds<double>(); + } + stats->round_trip_time_measurements = + voice_receiver_info.round_trip_time_measurements; + stats->total_round_trip_time = + voice_receiver_info.total_round_trip_time.seconds<double>(); + + return stats; +} + +std::unique_ptr<RTCInboundRtpStreamStats> +CreateInboundRTPStreamStatsFromVideoReceiverInfo( + const std::string& transport_id, + const std::string& mid, + const cricket::VideoMediaInfo& video_media_info, + const cricket::VideoReceiverInfo& video_receiver_info, + Timestamp timestamp, + RTCStatsReport* report) { + auto inbound_video = std::make_unique<RTCInboundRtpStreamStats>( + RTCInboundRtpStreamStatsIDFromSSRC( + transport_id, cricket::MEDIA_TYPE_VIDEO, video_receiver_info.ssrc()), + timestamp); + SetInboundRTPStreamStatsFromMediaReceiverInfo(video_receiver_info, + inbound_video.get()); + inbound_video->transport_id = transport_id; + inbound_video->mid = mid; + inbound_video->kind = "video"; + if (video_receiver_info.codec_payload_type.has_value()) { + auto codec_param_it = video_media_info.receive_codecs.find( + *video_receiver_info.codec_payload_type); + RTC_DCHECK(codec_param_it != video_media_info.receive_codecs.end()); + if (codec_param_it != video_media_info.receive_codecs.end()) { + inbound_video->codec_id = GetCodecIdAndMaybeCreateCodecStats( + inbound_video->timestamp(), kDirectionInbound, transport_id, + codec_param_it->second, report); + } + } + inbound_video->jitter = static_cast<double>(video_receiver_info.jitter_ms) / + rtc::kNumMillisecsPerSec; + inbound_video->fir_count = + static_cast<uint32_t>(video_receiver_info.firs_sent); + inbound_video->pli_count = + static_cast<uint32_t>(video_receiver_info.plis_sent); + inbound_video->frames_received = video_receiver_info.frames_received; + inbound_video->frames_decoded = video_receiver_info.frames_decoded; + inbound_video->frames_dropped = video_receiver_info.frames_dropped; + inbound_video->key_frames_decoded = video_receiver_info.key_frames_decoded; + if (video_receiver_info.frame_width > 0) { + inbound_video->frame_width = + static_cast<uint32_t>(video_receiver_info.frame_width); + } + if (video_receiver_info.frame_height > 0) { + inbound_video->frame_height = + static_cast<uint32_t>(video_receiver_info.frame_height); + } + if (video_receiver_info.framerate_decoded > 0) { + inbound_video->frames_per_second = video_receiver_info.framerate_decoded; + } + if (video_receiver_info.qp_sum.has_value()) { + inbound_video->qp_sum = *video_receiver_info.qp_sum; + } + if (video_receiver_info.timing_frame_info.has_value()) { + inbound_video->goog_timing_frame_info = + video_receiver_info.timing_frame_info->ToString(); + } + inbound_video->total_decode_time = + video_receiver_info.total_decode_time.seconds<double>(); + inbound_video->total_processing_delay = + video_receiver_info.total_processing_delay.seconds<double>(); + inbound_video->total_assembly_time = + video_receiver_info.total_assembly_time.seconds<double>(); + inbound_video->frames_assembled_from_multiple_packets = + video_receiver_info.frames_assembled_from_multiple_packets; + inbound_video->total_inter_frame_delay = + video_receiver_info.total_inter_frame_delay; + inbound_video->total_squared_inter_frame_delay = + video_receiver_info.total_squared_inter_frame_delay; + inbound_video->pause_count = video_receiver_info.pause_count; + inbound_video->total_pauses_duration = + static_cast<double>(video_receiver_info.total_pauses_duration_ms) / + rtc::kNumMillisecsPerSec; + inbound_video->freeze_count = video_receiver_info.freeze_count; + inbound_video->total_freezes_duration = + static_cast<double>(video_receiver_info.total_freezes_duration_ms) / + rtc::kNumMillisecsPerSec; + inbound_video->min_playout_delay = + static_cast<double>(video_receiver_info.min_playout_delay_ms) / + rtc::kNumMillisecsPerSec; + if (video_receiver_info.last_packet_received.has_value()) { + inbound_video->last_packet_received_timestamp = + video_receiver_info.last_packet_received->ms<double>(); + } + if (video_receiver_info.estimated_playout_ntp_timestamp_ms.has_value()) { + // TODO(bugs.webrtc.org/10529): Fix time origin if needed. + inbound_video->estimated_playout_timestamp = static_cast<double>( + *video_receiver_info.estimated_playout_ntp_timestamp_ms); + } + // TODO(bugs.webrtc.org/10529): When info's `content_info` is optional + // support the "unspecified" value. + if (videocontenttypehelpers::IsScreenshare(video_receiver_info.content_type)) + inbound_video->content_type = "screenshare"; + if (video_receiver_info.decoder_implementation_name.has_value()) { + inbound_video->decoder_implementation = + *video_receiver_info.decoder_implementation_name; + } + if (video_receiver_info.power_efficient_decoder.has_value()) { + inbound_video->power_efficient_decoder = + *video_receiver_info.power_efficient_decoder; + } + for (const auto& ssrc_group : video_receiver_info.ssrc_groups) { + if (ssrc_group.semantics == cricket::kFidSsrcGroupSemantics && + ssrc_group.ssrcs.size() == 2) { + inbound_video->rtx_ssrc = ssrc_group.ssrcs[1]; + } else if (ssrc_group.semantics == cricket::kFecFrSsrcGroupSemantics && + ssrc_group.ssrcs.size() == 2) { + // TODO(bugs.webrtc.org/15002): the ssrc-group might be >= 2 with + // multistream support. + inbound_video->fec_ssrc = ssrc_group.ssrcs[1]; + } + } + + return inbound_video; +} + +// Provides the media independent counters and information (both audio and +// video). +void SetOutboundRTPStreamStatsFromMediaSenderInfo( + const cricket::MediaSenderInfo& media_sender_info, + RTCOutboundRtpStreamStats* outbound_stats) { + RTC_DCHECK(outbound_stats); + outbound_stats->ssrc = media_sender_info.ssrc(); + outbound_stats->packets_sent = + static_cast<uint32_t>(media_sender_info.packets_sent); + outbound_stats->total_packet_send_delay = + media_sender_info.total_packet_send_delay.seconds<double>(); + outbound_stats->retransmitted_packets_sent = + media_sender_info.retransmitted_packets_sent; + outbound_stats->bytes_sent = + static_cast<uint64_t>(media_sender_info.payload_bytes_sent); + outbound_stats->header_bytes_sent = + static_cast<uint64_t>(media_sender_info.header_and_padding_bytes_sent); + outbound_stats->retransmitted_bytes_sent = + media_sender_info.retransmitted_bytes_sent; + outbound_stats->nack_count = media_sender_info.nacks_received; + if (media_sender_info.active.has_value()) { + outbound_stats->active = *media_sender_info.active; + } +} + +std::unique_ptr<RTCOutboundRtpStreamStats> +CreateOutboundRTPStreamStatsFromVoiceSenderInfo( + const std::string& transport_id, + const std::string& mid, + const cricket::VoiceMediaInfo& voice_media_info, + const cricket::VoiceSenderInfo& voice_sender_info, + Timestamp timestamp, + RTCStatsReport* report) { + auto outbound_audio = std::make_unique<RTCOutboundRtpStreamStats>( + RTCOutboundRtpStreamStatsIDFromSSRC( + transport_id, cricket::MEDIA_TYPE_AUDIO, voice_sender_info.ssrc()), + timestamp); + SetOutboundRTPStreamStatsFromMediaSenderInfo(voice_sender_info, + outbound_audio.get()); + outbound_audio->transport_id = transport_id; + outbound_audio->mid = mid; + outbound_audio->kind = "audio"; + if (voice_sender_info.target_bitrate.has_value() && + *voice_sender_info.target_bitrate > 0) { + outbound_audio->target_bitrate = *voice_sender_info.target_bitrate; + } + if (voice_sender_info.codec_payload_type.has_value()) { + auto codec_param_it = voice_media_info.send_codecs.find( + *voice_sender_info.codec_payload_type); + RTC_DCHECK(codec_param_it != voice_media_info.send_codecs.end()); + if (codec_param_it != voice_media_info.send_codecs.end()) { + outbound_audio->codec_id = GetCodecIdAndMaybeCreateCodecStats( + outbound_audio->timestamp(), kDirectionOutbound, transport_id, + codec_param_it->second, report); + } + } + // `fir_count` and `pli_count` are only valid for video and are + // purposefully left undefined for audio. + return outbound_audio; +} + +std::unique_ptr<RTCOutboundRtpStreamStats> +CreateOutboundRTPStreamStatsFromVideoSenderInfo( + const std::string& transport_id, + const std::string& mid, + const cricket::VideoMediaInfo& video_media_info, + const cricket::VideoSenderInfo& video_sender_info, + Timestamp timestamp, + RTCStatsReport* report) { + auto outbound_video = std::make_unique<RTCOutboundRtpStreamStats>( + RTCOutboundRtpStreamStatsIDFromSSRC( + transport_id, cricket::MEDIA_TYPE_VIDEO, video_sender_info.ssrc()), + timestamp); + SetOutboundRTPStreamStatsFromMediaSenderInfo(video_sender_info, + outbound_video.get()); + outbound_video->transport_id = transport_id; + outbound_video->mid = mid; + outbound_video->kind = "video"; + if (video_sender_info.codec_payload_type.has_value()) { + auto codec_param_it = video_media_info.send_codecs.find( + *video_sender_info.codec_payload_type); + RTC_DCHECK(codec_param_it != video_media_info.send_codecs.end()); + if (codec_param_it != video_media_info.send_codecs.end()) { + outbound_video->codec_id = GetCodecIdAndMaybeCreateCodecStats( + outbound_video->timestamp(), kDirectionOutbound, transport_id, + codec_param_it->second, report); + } + } + outbound_video->fir_count = + static_cast<uint32_t>(video_sender_info.firs_received); + outbound_video->pli_count = + static_cast<uint32_t>(video_sender_info.plis_received); + if (video_sender_info.qp_sum.has_value()) + outbound_video->qp_sum = *video_sender_info.qp_sum; + if (video_sender_info.target_bitrate.has_value() && + *video_sender_info.target_bitrate > 0) { + outbound_video->target_bitrate = *video_sender_info.target_bitrate; + } + outbound_video->frames_encoded = video_sender_info.frames_encoded; + outbound_video->key_frames_encoded = video_sender_info.key_frames_encoded; + outbound_video->total_encode_time = + static_cast<double>(video_sender_info.total_encode_time_ms) / + rtc::kNumMillisecsPerSec; + outbound_video->total_encoded_bytes_target = + video_sender_info.total_encoded_bytes_target; + if (video_sender_info.send_frame_width > 0) { + outbound_video->frame_width = + static_cast<uint32_t>(video_sender_info.send_frame_width); + } + if (video_sender_info.send_frame_height > 0) { + outbound_video->frame_height = + static_cast<uint32_t>(video_sender_info.send_frame_height); + } + if (video_sender_info.framerate_sent > 0) { + outbound_video->frames_per_second = video_sender_info.framerate_sent; + } + outbound_video->frames_sent = video_sender_info.frames_sent; + outbound_video->huge_frames_sent = video_sender_info.huge_frames_sent; + outbound_video->quality_limitation_reason = + QualityLimitationReasonToRTCQualityLimitationReason( + video_sender_info.quality_limitation_reason); + outbound_video->quality_limitation_durations = + QualityLimitationDurationToRTCQualityLimitationDuration( + video_sender_info.quality_limitation_durations_ms); + outbound_video->quality_limitation_resolution_changes = + video_sender_info.quality_limitation_resolution_changes; + // TODO(https://crbug.com/webrtc/10529): When info's `content_info` is + // optional, support the "unspecified" value. + if (videocontenttypehelpers::IsScreenshare(video_sender_info.content_type)) + outbound_video->content_type = "screenshare"; + if (video_sender_info.encoder_implementation_name.has_value()) { + outbound_video->encoder_implementation = + *video_sender_info.encoder_implementation_name; + } + if (video_sender_info.rid.has_value()) { + outbound_video->rid = *video_sender_info.rid; + } + if (video_sender_info.power_efficient_encoder.has_value()) { + outbound_video->power_efficient_encoder = + *video_sender_info.power_efficient_encoder; + } + if (video_sender_info.scalability_mode) { + outbound_video->scalability_mode = std::string( + ScalabilityModeToString(*video_sender_info.scalability_mode)); + } + for (const auto& ssrc_group : video_sender_info.ssrc_groups) { + if (ssrc_group.semantics == cricket::kFidSsrcGroupSemantics && + ssrc_group.ssrcs.size() == 2 && + video_sender_info.ssrc() == ssrc_group.ssrcs[0]) { + outbound_video->rtx_ssrc = ssrc_group.ssrcs[1]; + } + } + return outbound_video; +} + +std::unique_ptr<RTCRemoteInboundRtpStreamStats> +ProduceRemoteInboundRtpStreamStatsFromReportBlockData( + const std::string& transport_id, + const ReportBlockData& report_block, + cricket::MediaType media_type, + const std::map<std::string, RTCOutboundRtpStreamStats*>& outbound_rtps, + const RTCStatsReport& report) { + // RTCStats' timestamp generally refers to when the metric was sampled, but + // for "remote-[outbound/inbound]-rtp" it refers to the local time when the + // Report Block was received. + auto remote_inbound = std::make_unique<RTCRemoteInboundRtpStreamStats>( + RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc( + media_type, report_block.source_ssrc()), + report_block.report_block_timestamp_utc()); + remote_inbound->ssrc = report_block.source_ssrc(); + remote_inbound->kind = + media_type == cricket::MEDIA_TYPE_AUDIO ? "audio" : "video"; + remote_inbound->packets_lost = report_block.cumulative_lost(); + remote_inbound->fraction_lost = report_block.fraction_lost(); + if (report_block.num_rtts() > 0) { + remote_inbound->round_trip_time = report_block.last_rtt().seconds<double>(); + } + remote_inbound->total_round_trip_time = + report_block.sum_rtts().seconds<double>(); + remote_inbound->round_trip_time_measurements = report_block.num_rtts(); + + std::string local_id = RTCOutboundRtpStreamStatsIDFromSSRC( + transport_id, media_type, report_block.source_ssrc()); + // Look up local stat from `outbound_rtps` where the pointers are non-const. + auto local_id_it = outbound_rtps.find(local_id); + if (local_id_it != outbound_rtps.end()) { + remote_inbound->local_id = local_id; + auto& outbound_rtp = *local_id_it->second; + outbound_rtp.remote_id = remote_inbound->id(); + // The RTP/RTCP transport is obtained from the + // RTCOutboundRtpStreamStats's transport. + const auto* transport_from_id = report.Get(transport_id); + if (transport_from_id) { + const auto& transport = transport_from_id->cast_to<RTCTransportStats>(); + // If RTP and RTCP are not multiplexed, there is a separate RTCP + // transport paired with the RTP transport, otherwise the same + // transport is used for RTCP and RTP. + remote_inbound->transport_id = + transport.rtcp_transport_stats_id.is_defined() + ? *transport.rtcp_transport_stats_id + : *outbound_rtp.transport_id; + } + // We're assuming the same codec is used on both ends. However if the + // codec is switched out on the fly we may have received a Report Block + // based on the previous codec and there is no way to tell which point in + // time the codec changed for the remote end. + const auto* codec_from_id = outbound_rtp.codec_id.is_defined() + ? report.Get(*outbound_rtp.codec_id) + : nullptr; + if (codec_from_id) { + remote_inbound->codec_id = *outbound_rtp.codec_id; + const auto& codec = codec_from_id->cast_to<RTCCodecStats>(); + if (codec.clock_rate.is_defined()) { + remote_inbound->jitter = + report_block.jitter(*codec.clock_rate).seconds<double>(); + } + } + } + return remote_inbound; +} + +void ProduceCertificateStatsFromSSLCertificateStats( + Timestamp timestamp, + const rtc::SSLCertificateStats& certificate_stats, + RTCStatsReport* report) { + RTCCertificateStats* prev_certificate_stats = nullptr; + for (const rtc::SSLCertificateStats* s = &certificate_stats; s; + s = s->issuer.get()) { + std::string certificate_stats_id = + RTCCertificateIDFromFingerprint(s->fingerprint); + // It is possible for the same certificate to show up multiple times, e.g. + // if local and remote side use the same certificate in a loopback call. + // If the report already contains stats for this certificate, skip it. + if (report->Get(certificate_stats_id)) { + RTC_DCHECK_EQ(s, &certificate_stats); + break; + } + RTCCertificateStats* certificate_stats = + new RTCCertificateStats(certificate_stats_id, timestamp); + certificate_stats->fingerprint = s->fingerprint; + certificate_stats->fingerprint_algorithm = s->fingerprint_algorithm; + certificate_stats->base64_certificate = s->base64_certificate; + if (prev_certificate_stats) + prev_certificate_stats->issuer_certificate_id = certificate_stats->id(); + report->AddStats(std::unique_ptr<RTCCertificateStats>(certificate_stats)); + prev_certificate_stats = certificate_stats; + } +} + +const std::string& ProduceIceCandidateStats(Timestamp timestamp, + const cricket::Candidate& candidate, + bool is_local, + const std::string& transport_id, + RTCStatsReport* report) { + std::string id = "I" + candidate.id(); + const RTCStats* stats = report->Get(id); + if (!stats) { + std::unique_ptr<RTCIceCandidateStats> candidate_stats; + if (is_local) { + candidate_stats = + std::make_unique<RTCLocalIceCandidateStats>(std::move(id), timestamp); + } else { + candidate_stats = std::make_unique<RTCRemoteIceCandidateStats>( + std::move(id), timestamp); + } + candidate_stats->transport_id = transport_id; + if (is_local) { + candidate_stats->network_type = + NetworkTypeToStatsType(candidate.network_type()); + const std::string& candidate_type = candidate.type(); + const std::string& relay_protocol = candidate.relay_protocol(); + const std::string& url = candidate.url(); + if (candidate_type == cricket::RELAY_PORT_TYPE || + (candidate_type == cricket::PRFLX_PORT_TYPE && + !relay_protocol.empty())) { + RTC_DCHECK(relay_protocol.compare("udp") == 0 || + relay_protocol.compare("tcp") == 0 || + relay_protocol.compare("tls") == 0); + candidate_stats->relay_protocol = relay_protocol; + if (!url.empty()) { + candidate_stats->url = url; + } + } else if (candidate_type == cricket::STUN_PORT_TYPE) { + if (!url.empty()) { + candidate_stats->url = url; + } + } + if (candidate.network_type() == rtc::ADAPTER_TYPE_VPN) { + candidate_stats->vpn = true; + candidate_stats->network_adapter_type = + std::string(NetworkTypeToStatsNetworkAdapterType( + candidate.underlying_type_for_vpn())); + } else { + candidate_stats->vpn = false; + candidate_stats->network_adapter_type = std::string( + NetworkTypeToStatsNetworkAdapterType(candidate.network_type())); + } + } else { + // We don't expect to know the adapter type of remote candidates. + RTC_DCHECK_EQ(rtc::ADAPTER_TYPE_UNKNOWN, candidate.network_type()); + RTC_DCHECK_EQ(0, candidate.relay_protocol().compare("")); + RTC_DCHECK_EQ(rtc::ADAPTER_TYPE_UNKNOWN, + candidate.underlying_type_for_vpn()); + } + candidate_stats->ip = candidate.address().ipaddr().ToString(); + candidate_stats->address = candidate.address().ipaddr().ToString(); + candidate_stats->port = static_cast<int32_t>(candidate.address().port()); + candidate_stats->protocol = candidate.protocol(); + candidate_stats->candidate_type = + CandidateTypeToRTCIceCandidateType(candidate.type()); + candidate_stats->priority = static_cast<int32_t>(candidate.priority()); + candidate_stats->foundation = candidate.foundation(); + auto related_address = candidate.related_address(); + if (related_address.port() != 0) { + candidate_stats->related_address = related_address.ipaddr().ToString(); + candidate_stats->related_port = + static_cast<int32_t>(related_address.port()); + } + candidate_stats->username_fragment = candidate.username(); + if (candidate.protocol() == "tcp") { + candidate_stats->tcp_type = candidate.tcptype(); + } + + stats = candidate_stats.get(); + report->AddStats(std::move(candidate_stats)); + } + RTC_DCHECK_EQ(stats->type(), is_local ? RTCLocalIceCandidateStats::kType + : RTCRemoteIceCandidateStats::kType); + return stats->id(); +} + +template <typename StatsType> +void SetAudioProcessingStats(StatsType* stats, + const AudioProcessingStats& apm_stats) { + if (apm_stats.echo_return_loss.has_value()) { + stats->echo_return_loss = *apm_stats.echo_return_loss; + } + if (apm_stats.echo_return_loss_enhancement.has_value()) { + stats->echo_return_loss_enhancement = + *apm_stats.echo_return_loss_enhancement; + } +} + +} // namespace + +rtc::scoped_refptr<RTCStatsReport> +RTCStatsCollector::CreateReportFilteredBySelector( + bool filter_by_sender_selector, + rtc::scoped_refptr<const RTCStatsReport> report, + rtc::scoped_refptr<RtpSenderInternal> sender_selector, + rtc::scoped_refptr<RtpReceiverInternal> receiver_selector) { + std::vector<std::string> rtpstream_ids; + if (filter_by_sender_selector) { + // Filter mode: RTCStatsCollector::RequestInfo::kSenderSelector + if (sender_selector) { + // Find outbound-rtp(s) of the sender using ssrc lookup. + auto encodings = sender_selector->GetParametersInternal().encodings; + for (const auto* outbound_rtp : + report->GetStatsOfType<RTCOutboundRtpStreamStats>()) { + RTC_DCHECK(outbound_rtp->ssrc.is_defined()); + auto it = std::find_if(encodings.begin(), encodings.end(), + [ssrc = *outbound_rtp->ssrc]( + const RtpEncodingParameters& encoding) { + return encoding.ssrc == ssrc; + }); + if (it != encodings.end()) { + rtpstream_ids.push_back(outbound_rtp->id()); + } + } + } + } else { + // Filter mode: RTCStatsCollector::RequestInfo::kReceiverSelector + if (receiver_selector) { + // Find the inbound-rtp of the receiver using ssrc lookup. + absl::optional<uint32_t> ssrc; + worker_thread_->BlockingCall([&] { ssrc = receiver_selector->ssrc(); }); + if (ssrc.has_value()) { + for (const auto* inbound_rtp : + report->GetStatsOfType<RTCInboundRtpStreamStats>()) { + RTC_DCHECK(inbound_rtp->ssrc.is_defined()); + if (*inbound_rtp->ssrc == *ssrc) { + rtpstream_ids.push_back(inbound_rtp->id()); + } + } + } + } + } + if (rtpstream_ids.empty()) + return RTCStatsReport::Create(report->timestamp()); + return TakeReferencedStats(report->Copy(), rtpstream_ids); +} + +RTCStatsCollector::CertificateStatsPair +RTCStatsCollector::CertificateStatsPair::Copy() const { + CertificateStatsPair copy; + copy.local = local ? local->Copy() : nullptr; + copy.remote = remote ? remote->Copy() : nullptr; + return copy; +} + +RTCStatsCollector::RequestInfo::RequestInfo( + rtc::scoped_refptr<RTCStatsCollectorCallback> callback) + : RequestInfo(FilterMode::kAll, std::move(callback), nullptr, nullptr) {} + +RTCStatsCollector::RequestInfo::RequestInfo( + rtc::scoped_refptr<RtpSenderInternal> selector, + rtc::scoped_refptr<RTCStatsCollectorCallback> callback) + : RequestInfo(FilterMode::kSenderSelector, + std::move(callback), + std::move(selector), + nullptr) {} + +RTCStatsCollector::RequestInfo::RequestInfo( + rtc::scoped_refptr<RtpReceiverInternal> selector, + rtc::scoped_refptr<RTCStatsCollectorCallback> callback) + : RequestInfo(FilterMode::kReceiverSelector, + std::move(callback), + nullptr, + std::move(selector)) {} + +RTCStatsCollector::RequestInfo::RequestInfo( + RTCStatsCollector::RequestInfo::FilterMode filter_mode, + rtc::scoped_refptr<RTCStatsCollectorCallback> callback, + rtc::scoped_refptr<RtpSenderInternal> sender_selector, + rtc::scoped_refptr<RtpReceiverInternal> receiver_selector) + : filter_mode_(filter_mode), + callback_(std::move(callback)), + sender_selector_(std::move(sender_selector)), + receiver_selector_(std::move(receiver_selector)) { + RTC_DCHECK(callback_); + RTC_DCHECK(!sender_selector_ || !receiver_selector_); +} + +rtc::scoped_refptr<RTCStatsCollector> RTCStatsCollector::Create( + PeerConnectionInternal* pc, + int64_t cache_lifetime_us) { + return rtc::make_ref_counted<RTCStatsCollector>(pc, cache_lifetime_us); +} + +RTCStatsCollector::RTCStatsCollector(PeerConnectionInternal* pc, + int64_t cache_lifetime_us) + : pc_(pc), + signaling_thread_(pc->signaling_thread()), + worker_thread_(pc->worker_thread()), + network_thread_(pc->network_thread()), + num_pending_partial_reports_(0), + partial_report_timestamp_us_(0), + network_report_event_(true /* manual_reset */, + true /* initially_signaled */), + cache_timestamp_us_(0), + cache_lifetime_us_(cache_lifetime_us) { + RTC_DCHECK(pc_); + RTC_DCHECK(signaling_thread_); + RTC_DCHECK(worker_thread_); + RTC_DCHECK(network_thread_); + RTC_DCHECK_GE(cache_lifetime_us_, 0); +} + +RTCStatsCollector::~RTCStatsCollector() { + RTC_DCHECK_EQ(num_pending_partial_reports_, 0); +} + +void RTCStatsCollector::GetStatsReport( + rtc::scoped_refptr<RTCStatsCollectorCallback> callback) { + GetStatsReportInternal(RequestInfo(std::move(callback))); +} + +void RTCStatsCollector::GetStatsReport( + rtc::scoped_refptr<RtpSenderInternal> selector, + rtc::scoped_refptr<RTCStatsCollectorCallback> callback) { + GetStatsReportInternal(RequestInfo(std::move(selector), std::move(callback))); +} + +void RTCStatsCollector::GetStatsReport( + rtc::scoped_refptr<RtpReceiverInternal> selector, + rtc::scoped_refptr<RTCStatsCollectorCallback> callback) { + GetStatsReportInternal(RequestInfo(std::move(selector), std::move(callback))); +} + +void RTCStatsCollector::GetStatsReportInternal( + RTCStatsCollector::RequestInfo request) { + RTC_DCHECK_RUN_ON(signaling_thread_); + requests_.push_back(std::move(request)); + + // "Now" using a monotonically increasing timer. + int64_t cache_now_us = rtc::TimeMicros(); + if (cached_report_ && + cache_now_us - cache_timestamp_us_ <= cache_lifetime_us_) { + // We have a fresh cached report to deliver. Deliver asynchronously, since + // the caller may not be expecting a synchronous callback, and it avoids + // reentrancy problems. + signaling_thread_->PostTask( + absl::bind_front(&RTCStatsCollector::DeliverCachedReport, + rtc::scoped_refptr<RTCStatsCollector>(this), + cached_report_, std::move(requests_))); + } else if (!num_pending_partial_reports_) { + // Only start gathering stats if we're not already gathering stats. In the + // case of already gathering stats, `callback_` will be invoked when there + // are no more pending partial reports. + + // "Now" using a system clock, relative to the UNIX epoch (Jan 1, 1970, + // UTC), in microseconds. The system clock could be modified and is not + // necessarily monotonically increasing. + Timestamp timestamp = Timestamp::Micros(rtc::TimeUTCMicros()); + + num_pending_partial_reports_ = 2; + partial_report_timestamp_us_ = cache_now_us; + + // Prepare `transceiver_stats_infos_` and `call_stats_` for use in + // `ProducePartialResultsOnNetworkThread` and + // `ProducePartialResultsOnSignalingThread`. + PrepareTransceiverStatsInfosAndCallStats_s_w_n(); + // Don't touch `network_report_` on the signaling thread until + // ProducePartialResultsOnNetworkThread() has signaled the + // `network_report_event_`. + network_report_event_.Reset(); + rtc::scoped_refptr<RTCStatsCollector> collector(this); + network_thread_->PostTask([collector, + sctp_transport_name = pc_->sctp_transport_name(), + timestamp]() mutable { + collector->ProducePartialResultsOnNetworkThread( + timestamp, std::move(sctp_transport_name)); + }); + ProducePartialResultsOnSignalingThread(timestamp); + } +} + +void RTCStatsCollector::ClearCachedStatsReport() { + RTC_DCHECK_RUN_ON(signaling_thread_); + cached_report_ = nullptr; + MutexLock lock(&cached_certificates_mutex_); + cached_certificates_by_transport_.clear(); +} + +void RTCStatsCollector::WaitForPendingRequest() { + RTC_DCHECK_RUN_ON(signaling_thread_); + // If a request is pending, blocks until the `network_report_event_` is + // signaled and then delivers the result. Otherwise this is a NO-OP. + MergeNetworkReport_s(); +} + +void RTCStatsCollector::ProducePartialResultsOnSignalingThread( + Timestamp timestamp) { + RTC_DCHECK_RUN_ON(signaling_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + partial_report_ = RTCStatsReport::Create(timestamp); + + ProducePartialResultsOnSignalingThreadImpl(timestamp, partial_report_.get()); + + // ProducePartialResultsOnSignalingThread() is running synchronously on the + // signaling thread, so it is always the first partial result delivered on the + // signaling thread. The request is not complete until MergeNetworkReport_s() + // happens; we don't have to do anything here. + RTC_DCHECK_GT(num_pending_partial_reports_, 1); + --num_pending_partial_reports_; +} + +void RTCStatsCollector::ProducePartialResultsOnSignalingThreadImpl( + Timestamp timestamp, + RTCStatsReport* partial_report) { + RTC_DCHECK_RUN_ON(signaling_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + ProduceMediaSourceStats_s(timestamp, partial_report); + ProducePeerConnectionStats_s(timestamp, partial_report); + ProduceAudioPlayoutStats_s(timestamp, partial_report); +} + +void RTCStatsCollector::ProducePartialResultsOnNetworkThread( + Timestamp timestamp, + absl::optional<std::string> sctp_transport_name) { + TRACE_EVENT0("webrtc", + "RTCStatsCollector::ProducePartialResultsOnNetworkThread"); + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + // Touching `network_report_` on this thread is safe by this method because + // `network_report_event_` is reset before this method is invoked. + network_report_ = RTCStatsReport::Create(timestamp); + + ProduceDataChannelStats_n(timestamp, network_report_.get()); + + std::set<std::string> transport_names; + if (sctp_transport_name) { + transport_names.emplace(std::move(*sctp_transport_name)); + } + + for (const auto& info : transceiver_stats_infos_) { + if (info.transport_name) + transport_names.insert(*info.transport_name); + } + + std::map<std::string, cricket::TransportStats> transport_stats_by_name = + pc_->GetTransportStatsByNames(transport_names); + std::map<std::string, CertificateStatsPair> transport_cert_stats = + PrepareTransportCertificateStats_n(transport_stats_by_name); + + ProducePartialResultsOnNetworkThreadImpl(timestamp, transport_stats_by_name, + transport_cert_stats, + network_report_.get()); + + // Signal that it is now safe to touch `network_report_` on the signaling + // thread, and post a task to merge it into the final results. + network_report_event_.Set(); + rtc::scoped_refptr<RTCStatsCollector> collector(this); + signaling_thread_->PostTask( + [collector] { collector->MergeNetworkReport_s(); }); +} + +void RTCStatsCollector::ProducePartialResultsOnNetworkThreadImpl( + Timestamp timestamp, + const std::map<std::string, cricket::TransportStats>& + transport_stats_by_name, + const std::map<std::string, CertificateStatsPair>& transport_cert_stats, + RTCStatsReport* partial_report) { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + ProduceCertificateStats_n(timestamp, transport_cert_stats, partial_report); + ProduceIceCandidateAndPairStats_n(timestamp, transport_stats_by_name, + call_stats_, partial_report); + ProduceTransportStats_n(timestamp, transport_stats_by_name, + transport_cert_stats, partial_report); + ProduceRTPStreamStats_n(timestamp, transceiver_stats_infos_, partial_report); +} + +void RTCStatsCollector::MergeNetworkReport_s() { + RTC_DCHECK_RUN_ON(signaling_thread_); + // The `network_report_event_` must be signaled for it to be safe to touch + // `network_report_`. This is normally not blocking, but if + // WaitForPendingRequest() is called while a request is pending, we might have + // to wait until the network thread is done touching `network_report_`. + network_report_event_.Wait(rtc::Event::kForever); + if (!network_report_) { + // Normally, MergeNetworkReport_s() is executed because it is posted from + // the network thread. But if WaitForPendingRequest() is called while a + // request is pending, an early call to MergeNetworkReport_s() is made, + // merging the report and setting `network_report_` to null. If so, when the + // previously posted MergeNetworkReport_s() is later executed, the report is + // already null and nothing needs to be done here. + return; + } + RTC_DCHECK_GT(num_pending_partial_reports_, 0); + RTC_DCHECK(partial_report_); + partial_report_->TakeMembersFrom(network_report_); + network_report_ = nullptr; + --num_pending_partial_reports_; + // `network_report_` is currently the only partial report collected + // asynchronously, so `num_pending_partial_reports_` must now be 0 and we are + // ready to deliver the result. + RTC_DCHECK_EQ(num_pending_partial_reports_, 0); + cache_timestamp_us_ = partial_report_timestamp_us_; + cached_report_ = partial_report_; + partial_report_ = nullptr; + transceiver_stats_infos_.clear(); + // Trace WebRTC Stats when getStats is called on Javascript. + // This allows access to WebRTC stats from trace logs. To enable them, + // select the "webrtc_stats" category when recording traces. + TRACE_EVENT_INSTANT1("webrtc_stats", "webrtc_stats", "report", + cached_report_->ToJson()); + + // Deliver report and clear `requests_`. + std::vector<RequestInfo> requests; + requests.swap(requests_); + DeliverCachedReport(cached_report_, std::move(requests)); +} + +void RTCStatsCollector::DeliverCachedReport( + rtc::scoped_refptr<const RTCStatsReport> cached_report, + std::vector<RTCStatsCollector::RequestInfo> requests) { + RTC_DCHECK_RUN_ON(signaling_thread_); + RTC_DCHECK(!requests.empty()); + RTC_DCHECK(cached_report); + + for (const RequestInfo& request : requests) { + if (request.filter_mode() == RequestInfo::FilterMode::kAll) { + request.callback()->OnStatsDelivered(cached_report); + } else { + bool filter_by_sender_selector; + rtc::scoped_refptr<RtpSenderInternal> sender_selector; + rtc::scoped_refptr<RtpReceiverInternal> receiver_selector; + if (request.filter_mode() == RequestInfo::FilterMode::kSenderSelector) { + filter_by_sender_selector = true; + sender_selector = request.sender_selector(); + } else { + RTC_DCHECK(request.filter_mode() == + RequestInfo::FilterMode::kReceiverSelector); + filter_by_sender_selector = false; + receiver_selector = request.receiver_selector(); + } + request.callback()->OnStatsDelivered(CreateReportFilteredBySelector( + filter_by_sender_selector, cached_report, sender_selector, + receiver_selector)); + } + } +} + +void RTCStatsCollector::ProduceCertificateStats_n( + Timestamp timestamp, + const std::map<std::string, CertificateStatsPair>& transport_cert_stats, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + for (const auto& transport_cert_stats_pair : transport_cert_stats) { + if (transport_cert_stats_pair.second.local) { + ProduceCertificateStatsFromSSLCertificateStats( + timestamp, *transport_cert_stats_pair.second.local.get(), report); + } + if (transport_cert_stats_pair.second.remote) { + ProduceCertificateStatsFromSSLCertificateStats( + timestamp, *transport_cert_stats_pair.second.remote.get(), report); + } + } +} + +void RTCStatsCollector::ProduceDataChannelStats_n( + Timestamp timestamp, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + std::vector<DataChannelStats> data_stats = pc_->GetDataChannelStats(); + for (const auto& stats : data_stats) { + auto data_channel_stats = std::make_unique<RTCDataChannelStats>( + "D" + rtc::ToString(stats.internal_id), timestamp); + data_channel_stats->label = std::move(stats.label); + data_channel_stats->protocol = std::move(stats.protocol); + if (stats.id >= 0) { + // Do not set this value before the DTLS handshake is finished + // and filter out the magic value -1. + data_channel_stats->data_channel_identifier = stats.id; + } + data_channel_stats->state = DataStateToRTCDataChannelState(stats.state); + data_channel_stats->messages_sent = stats.messages_sent; + data_channel_stats->bytes_sent = stats.bytes_sent; + data_channel_stats->messages_received = stats.messages_received; + data_channel_stats->bytes_received = stats.bytes_received; + report->AddStats(std::move(data_channel_stats)); + } +} + +void RTCStatsCollector::ProduceIceCandidateAndPairStats_n( + Timestamp timestamp, + const std::map<std::string, cricket::TransportStats>& + transport_stats_by_name, + const Call::Stats& call_stats, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + for (const auto& entry : transport_stats_by_name) { + const std::string& transport_name = entry.first; + const cricket::TransportStats& transport_stats = entry.second; + for (const auto& channel_stats : transport_stats.channel_stats) { + std::string transport_id = RTCTransportStatsIDFromTransportChannel( + transport_name, channel_stats.component); + for (const auto& info : + channel_stats.ice_transport_stats.connection_infos) { + auto candidate_pair_stats = std::make_unique<RTCIceCandidatePairStats>( + RTCIceCandidatePairStatsIDFromConnectionInfo(info), timestamp); + + candidate_pair_stats->transport_id = transport_id; + candidate_pair_stats->local_candidate_id = ProduceIceCandidateStats( + timestamp, info.local_candidate, true, transport_id, report); + candidate_pair_stats->remote_candidate_id = ProduceIceCandidateStats( + timestamp, info.remote_candidate, false, transport_id, report); + candidate_pair_stats->state = + IceCandidatePairStateToRTCStatsIceCandidatePairState(info.state); + candidate_pair_stats->priority = info.priority; + candidate_pair_stats->nominated = info.nominated; + // TODO(hbos): This writable is different than the spec. It goes to + // false after a certain amount of time without a response passes. + // https://crbug.com/633550 + candidate_pair_stats->writable = info.writable; + // Note that sent_total_packets includes discarded packets but + // sent_total_bytes does not. + candidate_pair_stats->packets_sent = static_cast<uint64_t>( + info.sent_total_packets - info.sent_discarded_packets); + candidate_pair_stats->packets_discarded_on_send = + static_cast<uint64_t>(info.sent_discarded_packets); + candidate_pair_stats->packets_received = + static_cast<uint64_t>(info.packets_received); + candidate_pair_stats->bytes_sent = + static_cast<uint64_t>(info.sent_total_bytes); + candidate_pair_stats->bytes_discarded_on_send = + static_cast<uint64_t>(info.sent_discarded_bytes); + candidate_pair_stats->bytes_received = + static_cast<uint64_t>(info.recv_total_bytes); + candidate_pair_stats->total_round_trip_time = + static_cast<double>(info.total_round_trip_time_ms) / + rtc::kNumMillisecsPerSec; + if (info.current_round_trip_time_ms.has_value()) { + candidate_pair_stats->current_round_trip_time = + static_cast<double>(*info.current_round_trip_time_ms) / + rtc::kNumMillisecsPerSec; + } + if (info.best_connection) { + // The bandwidth estimations we have are for the selected candidate + // pair ("info.best_connection"). + RTC_DCHECK_GE(call_stats.send_bandwidth_bps, 0); + RTC_DCHECK_GE(call_stats.recv_bandwidth_bps, 0); + if (call_stats.send_bandwidth_bps > 0) { + candidate_pair_stats->available_outgoing_bitrate = + static_cast<double>(call_stats.send_bandwidth_bps); + } + if (call_stats.recv_bandwidth_bps > 0) { + candidate_pair_stats->available_incoming_bitrate = + static_cast<double>(call_stats.recv_bandwidth_bps); + } + } + candidate_pair_stats->requests_received = + static_cast<uint64_t>(info.recv_ping_requests); + candidate_pair_stats->requests_sent = + static_cast<uint64_t>(info.sent_ping_requests_total); + candidate_pair_stats->responses_received = + static_cast<uint64_t>(info.recv_ping_responses); + candidate_pair_stats->responses_sent = + static_cast<uint64_t>(info.sent_ping_responses); + RTC_DCHECK_GE(info.sent_ping_requests_total, + info.sent_ping_requests_before_first_response); + candidate_pair_stats->consent_requests_sent = static_cast<uint64_t>( + info.sent_ping_requests_total - + info.sent_ping_requests_before_first_response); + + if (info.last_data_received.has_value()) { + candidate_pair_stats->last_packet_received_timestamp = + static_cast<double>(info.last_data_received->ms()); + } + if (info.last_data_sent) { + candidate_pair_stats->last_packet_sent_timestamp = + static_cast<double>(info.last_data_sent->ms()); + } + + report->AddStats(std::move(candidate_pair_stats)); + } + + // Produce local candidate stats. If a transport exists these will already + // have been produced. + for (const auto& candidate_stats : + channel_stats.ice_transport_stats.candidate_stats_list) { + const auto& candidate = candidate_stats.candidate(); + ProduceIceCandidateStats(timestamp, candidate, true, transport_id, + report); + } + } + } +} + +void RTCStatsCollector::ProduceMediaSourceStats_s( + Timestamp timestamp, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(signaling_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + for (const RtpTransceiverStatsInfo& transceiver_stats_info : + transceiver_stats_infos_) { + const auto& track_media_info_map = + transceiver_stats_info.track_media_info_map; + for (const auto& sender : transceiver_stats_info.transceiver->senders()) { + const auto& sender_internal = sender->internal(); + const auto& track = sender_internal->track(); + if (!track) + continue; + // TODO(https://crbug.com/webrtc/10771): The same track could be attached + // to multiple senders which should result in multiple senders referencing + // the same media-source stats. When all media source related metrics are + // moved to the track's source (e.g. input frame rate is moved from + // cricket::VideoSenderInfo to VideoTrackSourceInterface::Stats and audio + // levels are moved to the corresponding audio track/source object), don't + // create separate media source stats objects on a per-attachment basis. + std::unique_ptr<RTCMediaSourceStats> media_source_stats; + if (track->kind() == MediaStreamTrackInterface::kAudioKind) { + AudioTrackInterface* audio_track = + static_cast<AudioTrackInterface*>(track.get()); + auto audio_source_stats = std::make_unique<RTCAudioSourceStats>( + RTCMediaSourceStatsIDFromKindAndAttachment( + cricket::MEDIA_TYPE_AUDIO, sender_internal->AttachmentId()), + timestamp); + // TODO(https://crbug.com/webrtc/10771): We shouldn't need to have an + // SSRC assigned (there shouldn't need to exist a send-stream, created + // by an O/A exchange) in order to read audio media-source stats. + // TODO(https://crbug.com/webrtc/8694): SSRC 0 shouldn't be a magic + // value indicating no SSRC. + if (sender_internal->ssrc() != 0) { + auto* voice_sender_info = + track_media_info_map.GetVoiceSenderInfoBySsrc( + sender_internal->ssrc()); + if (voice_sender_info) { + audio_source_stats->audio_level = DoubleAudioLevelFromIntAudioLevel( + voice_sender_info->audio_level); + audio_source_stats->total_audio_energy = + voice_sender_info->total_input_energy; + audio_source_stats->total_samples_duration = + voice_sender_info->total_input_duration; + SetAudioProcessingStats(audio_source_stats.get(), + voice_sender_info->apm_statistics); + } + } + // Audio processor may be attached to either the track or the send + // stream, so look in both places. + auto audio_processor(audio_track->GetAudioProcessor()); + if (audio_processor.get()) { + // The `has_remote_tracks` argument is obsolete; makes no difference + // if it's set to true or false. + AudioProcessorInterface::AudioProcessorStatistics ap_stats = + audio_processor->GetStats(/*has_remote_tracks=*/false); + SetAudioProcessingStats(audio_source_stats.get(), + ap_stats.apm_statistics); + } + media_source_stats = std::move(audio_source_stats); + } else { + RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind()); + auto video_source_stats = std::make_unique<RTCVideoSourceStats>( + RTCMediaSourceStatsIDFromKindAndAttachment( + cricket::MEDIA_TYPE_VIDEO, sender_internal->AttachmentId()), + timestamp); + auto* video_track = static_cast<VideoTrackInterface*>(track.get()); + auto* video_source = video_track->GetSource(); + VideoTrackSourceInterface::Stats source_stats; + if (video_source && video_source->GetStats(&source_stats)) { + video_source_stats->width = source_stats.input_width; + video_source_stats->height = source_stats.input_height; + } + // TODO(https://crbug.com/webrtc/10771): We shouldn't need to have an + // SSRC assigned (there shouldn't need to exist a send-stream, created + // by an O/A exchange) in order to get framesPerSecond. + // TODO(https://crbug.com/webrtc/8694): SSRC 0 shouldn't be a magic + // value indicating no SSRC. + if (sender_internal->ssrc() != 0) { + auto* video_sender_info = + track_media_info_map.GetVideoSenderInfoBySsrc( + sender_internal->ssrc()); + if (video_sender_info) { + video_source_stats->frames_per_second = + video_sender_info->framerate_input; + video_source_stats->frames = video_sender_info->frames; + } + } + media_source_stats = std::move(video_source_stats); + } + media_source_stats->track_identifier = track->id(); + media_source_stats->kind = track->kind(); + report->AddStats(std::move(media_source_stats)); + } + } +} + +void RTCStatsCollector::ProducePeerConnectionStats_s( + Timestamp timestamp, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(signaling_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + auto stats(std::make_unique<RTCPeerConnectionStats>("P", timestamp)); + stats->data_channels_opened = internal_record_.data_channels_opened; + stats->data_channels_closed = internal_record_.data_channels_closed; + report->AddStats(std::move(stats)); +} + +void RTCStatsCollector::ProduceAudioPlayoutStats_s( + Timestamp timestamp, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(signaling_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + if (audio_device_stats_) { + report->AddStats(CreateAudioPlayoutStats(*audio_device_stats_, timestamp)); + } +} + +void RTCStatsCollector::ProduceRTPStreamStats_n( + Timestamp timestamp, + const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos) { + if (stats.media_type == cricket::MEDIA_TYPE_AUDIO) { + ProduceAudioRTPStreamStats_n(timestamp, stats, report); + } else if (stats.media_type == cricket::MEDIA_TYPE_VIDEO) { + ProduceVideoRTPStreamStats_n(timestamp, stats, report); + } else { + RTC_DCHECK_NOTREACHED(); + } + } +} + +void RTCStatsCollector::ProduceAudioRTPStreamStats_n( + Timestamp timestamp, + const RtpTransceiverStatsInfo& stats, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + if (!stats.mid || !stats.transport_name) { + return; + } + RTC_DCHECK(stats.track_media_info_map.voice_media_info().has_value()); + std::string mid = *stats.mid; + std::string transport_id = RTCTransportStatsIDFromTransportChannel( + *stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP); + // Inbound and remote-outbound. + // The remote-outbound stats are based on RTCP sender reports sent from the + // remote endpoint providing metrics about the remote outbound streams. + for (const cricket::VoiceReceiverInfo& voice_receiver_info : + stats.track_media_info_map.voice_media_info()->receivers) { + if (!voice_receiver_info.connected()) + continue; + // Inbound. + auto inbound_audio = CreateInboundAudioStreamStats( + *stats.track_media_info_map.voice_media_info(), voice_receiver_info, + transport_id, mid, timestamp, report); + // TODO(hta): This lookup should look for the sender, not the track. + rtc::scoped_refptr<AudioTrackInterface> audio_track = + stats.track_media_info_map.GetAudioTrack(voice_receiver_info); + if (audio_track) { + inbound_audio->track_identifier = audio_track->id(); + } + if (audio_device_stats_ && stats.media_type == cricket::MEDIA_TYPE_AUDIO && + stats.current_direction && + (*stats.current_direction == RtpTransceiverDirection::kSendRecv || + *stats.current_direction == RtpTransceiverDirection::kRecvOnly)) { + inbound_audio->playout_id = kAudioPlayoutSingletonId; + } + auto* inbound_audio_ptr = report->TryAddStats(std::move(inbound_audio)); + if (!inbound_audio_ptr) { + RTC_LOG(LS_ERROR) + << "Unable to add audio 'inbound-rtp' to report, ID is not unique."; + continue; + } + // Remote-outbound. + auto remote_outbound_audio = CreateRemoteOutboundAudioStreamStats( + voice_receiver_info, mid, *inbound_audio_ptr, transport_id); + // Add stats. + if (remote_outbound_audio) { + // When the remote outbound stats are available, the remote ID for the + // local inbound stats is set. + auto* remote_outbound_audio_ptr = + report->TryAddStats(std::move(remote_outbound_audio)); + if (remote_outbound_audio_ptr) { + inbound_audio_ptr->remote_id = remote_outbound_audio_ptr->id(); + } else { + RTC_LOG(LS_ERROR) << "Unable to add audio 'remote-outbound-rtp' to " + << "report, ID is not unique."; + } + } + } + // Outbound. + std::map<std::string, RTCOutboundRtpStreamStats*> audio_outbound_rtps; + for (const cricket::VoiceSenderInfo& voice_sender_info : + stats.track_media_info_map.voice_media_info()->senders) { + if (!voice_sender_info.connected()) + continue; + auto outbound_audio = CreateOutboundRTPStreamStatsFromVoiceSenderInfo( + transport_id, mid, *stats.track_media_info_map.voice_media_info(), + voice_sender_info, timestamp, report); + rtc::scoped_refptr<AudioTrackInterface> audio_track = + stats.track_media_info_map.GetAudioTrack(voice_sender_info); + if (audio_track) { + int attachment_id = + stats.track_media_info_map.GetAttachmentIdByTrack(audio_track.get()) + .value(); + outbound_audio->media_source_id = + RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_AUDIO, + attachment_id); + } + auto audio_outbound_pair = + std::make_pair(outbound_audio->id(), outbound_audio.get()); + if (report->TryAddStats(std::move(outbound_audio))) { + audio_outbound_rtps.insert(std::move(audio_outbound_pair)); + } else { + RTC_LOG(LS_ERROR) + << "Unable to add audio 'outbound-rtp' to report, ID is not unique."; + } + } + // Remote-inbound. + // These are Report Block-based, information sent from the remote endpoint, + // providing metrics about our Outbound streams. We take advantage of the fact + // that RTCOutboundRtpStreamStats, RTCCodecStats and RTCTransport have already + // been added to the report. + for (const cricket::VoiceSenderInfo& voice_sender_info : + stats.track_media_info_map.voice_media_info()->senders) { + for (const auto& report_block_data : voice_sender_info.report_block_datas) { + report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData( + transport_id, report_block_data, cricket::MEDIA_TYPE_AUDIO, + audio_outbound_rtps, *report)); + } + } +} + +void RTCStatsCollector::ProduceVideoRTPStreamStats_n( + Timestamp timestamp, + const RtpTransceiverStatsInfo& stats, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + if (!stats.mid || !stats.transport_name) { + return; + } + RTC_DCHECK(stats.track_media_info_map.video_media_info().has_value()); + std::string mid = *stats.mid; + std::string transport_id = RTCTransportStatsIDFromTransportChannel( + *stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP); + // Inbound + for (const cricket::VideoReceiverInfo& video_receiver_info : + stats.track_media_info_map.video_media_info()->receivers) { + if (!video_receiver_info.connected()) + continue; + auto inbound_video = CreateInboundRTPStreamStatsFromVideoReceiverInfo( + transport_id, mid, *stats.track_media_info_map.video_media_info(), + video_receiver_info, timestamp, report); + rtc::scoped_refptr<VideoTrackInterface> video_track = + stats.track_media_info_map.GetVideoTrack(video_receiver_info); + if (video_track) { + inbound_video->track_identifier = video_track->id(); + } + if (!report->TryAddStats(std::move(inbound_video))) { + RTC_LOG(LS_ERROR) + << "Unable to add video 'inbound-rtp' to report, ID is not unique."; + } + } + // Outbound + std::map<std::string, RTCOutboundRtpStreamStats*> video_outbound_rtps; + for (const cricket::VideoSenderInfo& video_sender_info : + stats.track_media_info_map.video_media_info()->senders) { + if (!video_sender_info.connected()) + continue; + auto outbound_video = CreateOutboundRTPStreamStatsFromVideoSenderInfo( + transport_id, mid, *stats.track_media_info_map.video_media_info(), + video_sender_info, timestamp, report); + rtc::scoped_refptr<VideoTrackInterface> video_track = + stats.track_media_info_map.GetVideoTrack(video_sender_info); + if (video_track) { + int attachment_id = + stats.track_media_info_map.GetAttachmentIdByTrack(video_track.get()) + .value(); + outbound_video->media_source_id = + RTCMediaSourceStatsIDFromKindAndAttachment(cricket::MEDIA_TYPE_VIDEO, + attachment_id); + } + auto video_outbound_pair = + std::make_pair(outbound_video->id(), outbound_video.get()); + if (report->TryAddStats(std::move(outbound_video))) { + video_outbound_rtps.insert(std::move(video_outbound_pair)); + } else { + RTC_LOG(LS_ERROR) + << "Unable to add video 'outbound-rtp' to report, ID is not unique."; + } + } + // Remote-inbound + // These are Report Block-based, information sent from the remote endpoint, + // providing metrics about our Outbound streams. We take advantage of the fact + // that RTCOutboundRtpStreamStats, RTCCodecStats and RTCTransport have already + // been added to the report. + for (const cricket::VideoSenderInfo& video_sender_info : + stats.track_media_info_map.video_media_info()->senders) { + for (const auto& report_block_data : video_sender_info.report_block_datas) { + report->AddStats(ProduceRemoteInboundRtpStreamStatsFromReportBlockData( + transport_id, report_block_data, cricket::MEDIA_TYPE_VIDEO, + video_outbound_rtps, *report)); + } + } +} + +void RTCStatsCollector::ProduceTransportStats_n( + Timestamp timestamp, + const std::map<std::string, cricket::TransportStats>& + transport_stats_by_name, + const std::map<std::string, CertificateStatsPair>& transport_cert_stats, + RTCStatsReport* report) const { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + for (const auto& entry : transport_stats_by_name) { + const std::string& transport_name = entry.first; + const cricket::TransportStats& transport_stats = entry.second; + + // Get reference to RTCP channel, if it exists. + std::string rtcp_transport_stats_id; + for (const cricket::TransportChannelStats& channel_stats : + transport_stats.channel_stats) { + if (channel_stats.component == cricket::ICE_CANDIDATE_COMPONENT_RTCP) { + rtcp_transport_stats_id = RTCTransportStatsIDFromTransportChannel( + transport_name, channel_stats.component); + break; + } + } + + // Get reference to local and remote certificates of this transport, if they + // exist. + const auto& certificate_stats_it = + transport_cert_stats.find(transport_name); + std::string local_certificate_id, remote_certificate_id; + RTC_DCHECK(certificate_stats_it != transport_cert_stats.cend()); + if (certificate_stats_it != transport_cert_stats.cend()) { + if (certificate_stats_it->second.local) { + local_certificate_id = RTCCertificateIDFromFingerprint( + certificate_stats_it->second.local->fingerprint); + } + if (certificate_stats_it->second.remote) { + remote_certificate_id = RTCCertificateIDFromFingerprint( + certificate_stats_it->second.remote->fingerprint); + } + } + + // There is one transport stats for each channel. + for (const cricket::TransportChannelStats& channel_stats : + transport_stats.channel_stats) { + auto transport_stats = std::make_unique<RTCTransportStats>( + RTCTransportStatsIDFromTransportChannel(transport_name, + channel_stats.component), + timestamp); + transport_stats->packets_sent = + channel_stats.ice_transport_stats.packets_sent; + transport_stats->packets_received = + channel_stats.ice_transport_stats.packets_received; + transport_stats->bytes_sent = + channel_stats.ice_transport_stats.bytes_sent; + transport_stats->bytes_received = + channel_stats.ice_transport_stats.bytes_received; + transport_stats->dtls_state = + DtlsTransportStateToRTCDtlsTransportState(channel_stats.dtls_state); + transport_stats->selected_candidate_pair_changes = + channel_stats.ice_transport_stats.selected_candidate_pair_changes; + transport_stats->ice_role = + IceRoleToRTCIceRole(channel_stats.ice_transport_stats.ice_role); + transport_stats->ice_local_username_fragment = + channel_stats.ice_transport_stats.ice_local_username_fragment; + transport_stats->ice_state = IceTransportStateToRTCIceTransportState( + channel_stats.ice_transport_stats.ice_state); + for (const cricket::ConnectionInfo& info : + channel_stats.ice_transport_stats.connection_infos) { + if (info.best_connection) { + transport_stats->selected_candidate_pair_id = + RTCIceCandidatePairStatsIDFromConnectionInfo(info); + } + } + if (channel_stats.component != cricket::ICE_CANDIDATE_COMPONENT_RTCP && + !rtcp_transport_stats_id.empty()) { + transport_stats->rtcp_transport_stats_id = rtcp_transport_stats_id; + } + if (!local_certificate_id.empty()) + transport_stats->local_certificate_id = local_certificate_id; + if (!remote_certificate_id.empty()) + transport_stats->remote_certificate_id = remote_certificate_id; + // Crypto information + if (channel_stats.ssl_version_bytes) { + char bytes[5]; + snprintf(bytes, sizeof(bytes), "%04X", channel_stats.ssl_version_bytes); + transport_stats->tls_version = bytes; + } + + if (channel_stats.dtls_role) { + transport_stats->dtls_role = + *channel_stats.dtls_role == rtc::SSL_CLIENT ? "client" : "server"; + } else { + transport_stats->dtls_role = "unknown"; + } + + if (channel_stats.ssl_cipher_suite != rtc::kTlsNullWithNullNull && + rtc::SSLStreamAdapter::SslCipherSuiteToName( + channel_stats.ssl_cipher_suite) + .length()) { + transport_stats->dtls_cipher = + rtc::SSLStreamAdapter::SslCipherSuiteToName( + channel_stats.ssl_cipher_suite); + } + if (channel_stats.srtp_crypto_suite != rtc::kSrtpInvalidCryptoSuite && + rtc::SrtpCryptoSuiteToName(channel_stats.srtp_crypto_suite) + .length()) { + transport_stats->srtp_cipher = + rtc::SrtpCryptoSuiteToName(channel_stats.srtp_crypto_suite); + } + report->AddStats(std::move(transport_stats)); + } + } +} + +std::map<std::string, RTCStatsCollector::CertificateStatsPair> +RTCStatsCollector::PrepareTransportCertificateStats_n( + const std::map<std::string, cricket::TransportStats>& + transport_stats_by_name) { + RTC_DCHECK_RUN_ON(network_thread_); + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + std::map<std::string, CertificateStatsPair> transport_cert_stats; + { + MutexLock lock(&cached_certificates_mutex_); + // Copy the certificate info from the cache, avoiding expensive + // rtc::SSLCertChain::GetStats() calls. + for (const auto& pair : cached_certificates_by_transport_) { + transport_cert_stats.insert( + std::make_pair(pair.first, pair.second.Copy())); + } + } + if (transport_cert_stats.empty()) { + // Collect certificate info. + for (const auto& entry : transport_stats_by_name) { + const std::string& transport_name = entry.first; + + CertificateStatsPair certificate_stats_pair; + rtc::scoped_refptr<rtc::RTCCertificate> local_certificate; + if (pc_->GetLocalCertificate(transport_name, &local_certificate)) { + certificate_stats_pair.local = + local_certificate->GetSSLCertificateChain().GetStats(); + } + + auto remote_cert_chain = pc_->GetRemoteSSLCertChain(transport_name); + if (remote_cert_chain) { + certificate_stats_pair.remote = remote_cert_chain->GetStats(); + } + + transport_cert_stats.insert( + std::make_pair(transport_name, std::move(certificate_stats_pair))); + } + // Copy the result into the certificate cache for future reference. + MutexLock lock(&cached_certificates_mutex_); + for (const auto& pair : transport_cert_stats) { + cached_certificates_by_transport_.insert( + std::make_pair(pair.first, pair.second.Copy())); + } + } + return transport_cert_stats; +} + +void RTCStatsCollector::PrepareTransceiverStatsInfosAndCallStats_s_w_n() { + RTC_DCHECK_RUN_ON(signaling_thread_); + + transceiver_stats_infos_.clear(); + // These are used to invoke GetStats for all the media channels together in + // one worker thread hop. + std::map<cricket::VoiceMediaSendChannelInterface*, + cricket::VoiceMediaSendInfo> + voice_send_stats; + std::map<cricket::VideoMediaSendChannelInterface*, + cricket::VideoMediaSendInfo> + video_send_stats; + std::map<cricket::VoiceMediaReceiveChannelInterface*, + cricket::VoiceMediaReceiveInfo> + voice_receive_stats; + std::map<cricket::VideoMediaReceiveChannelInterface*, + cricket::VideoMediaReceiveInfo> + video_receive_stats; + + auto transceivers = pc_->GetTransceiversInternal(); + + // TODO(tommi): See if we can avoid synchronously blocking the signaling + // thread while we do this (or avoid the BlockingCall at all). + network_thread_->BlockingCall([&] { + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + for (const auto& transceiver_proxy : transceivers) { + RtpTransceiver* transceiver = transceiver_proxy->internal(); + cricket::MediaType media_type = transceiver->media_type(); + + // Prepare stats entry. The TrackMediaInfoMap will be filled in after the + // stats have been fetched on the worker thread. + transceiver_stats_infos_.emplace_back(); + RtpTransceiverStatsInfo& stats = transceiver_stats_infos_.back(); + stats.transceiver = transceiver; + stats.media_type = media_type; + + cricket::ChannelInterface* channel = transceiver->channel(); + if (!channel) { + // The remaining fields require a BaseChannel. + continue; + } + + stats.mid = channel->mid(); + stats.transport_name = std::string(channel->transport_name()); + + if (media_type == cricket::MEDIA_TYPE_AUDIO) { + auto voice_send_channel = channel->voice_media_send_channel(); + RTC_DCHECK(voice_send_stats.find(voice_send_channel) == + voice_send_stats.end()); + voice_send_stats.insert( + std::make_pair(voice_send_channel, cricket::VoiceMediaSendInfo())); + + auto voice_receive_channel = channel->voice_media_receive_channel(); + RTC_DCHECK(voice_receive_stats.find(voice_receive_channel) == + voice_receive_stats.end()); + voice_receive_stats.insert(std::make_pair( + voice_receive_channel, cricket::VoiceMediaReceiveInfo())); + } else if (media_type == cricket::MEDIA_TYPE_VIDEO) { + auto video_send_channel = channel->video_media_send_channel(); + RTC_DCHECK(video_send_stats.find(video_send_channel) == + video_send_stats.end()); + video_send_stats.insert( + std::make_pair(video_send_channel, cricket::VideoMediaSendInfo())); + auto video_receive_channel = channel->video_media_receive_channel(); + RTC_DCHECK(video_receive_stats.find(video_receive_channel) == + video_receive_stats.end()); + video_receive_stats.insert(std::make_pair( + video_receive_channel, cricket::VideoMediaReceiveInfo())); + } else { + RTC_DCHECK_NOTREACHED(); + } + } + }); + + // We jump to the worker thread and call GetStats() on each media channel as + // well as GetCallStats(). At the same time we construct the + // TrackMediaInfoMaps, which also needs info from the worker thread. This + // minimizes the number of thread jumps. + worker_thread_->BlockingCall([&] { + rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls; + + for (auto& pair : voice_send_stats) { + if (!pair.first->GetStats(&pair.second)) { + RTC_LOG(LS_WARNING) << "Failed to get voice send stats."; + } + } + for (auto& pair : voice_receive_stats) { + if (!pair.first->GetStats(&pair.second, + /*get_and_clear_legacy_stats=*/false)) { + RTC_LOG(LS_WARNING) << "Failed to get voice receive stats."; + } + } + for (auto& pair : video_send_stats) { + if (!pair.first->GetStats(&pair.second)) { + RTC_LOG(LS_WARNING) << "Failed to get video send stats."; + } + } + for (auto& pair : video_receive_stats) { + if (!pair.first->GetStats(&pair.second)) { + RTC_LOG(LS_WARNING) << "Failed to get video receive stats."; + } + } + + // Create the TrackMediaInfoMap for each transceiver stats object. + for (auto& stats : transceiver_stats_infos_) { + auto transceiver = stats.transceiver; + absl::optional<cricket::VoiceMediaInfo> voice_media_info; + absl::optional<cricket::VideoMediaInfo> video_media_info; + auto channel = transceiver->channel(); + if (channel) { + cricket::MediaType media_type = transceiver->media_type(); + if (media_type == cricket::MEDIA_TYPE_AUDIO) { + auto voice_send_channel = channel->voice_media_send_channel(); + auto voice_receive_channel = channel->voice_media_receive_channel(); + voice_media_info = cricket::VoiceMediaInfo( + std::move(voice_send_stats[voice_send_channel]), + std::move(voice_receive_stats[voice_receive_channel])); + } else if (media_type == cricket::MEDIA_TYPE_VIDEO) { + auto video_send_channel = channel->video_media_send_channel(); + auto video_receive_channel = channel->video_media_receive_channel(); + video_media_info = cricket::VideoMediaInfo( + std::move(video_send_stats[video_send_channel]), + std::move(video_receive_stats[video_receive_channel])); + } + } + std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders; + for (const auto& sender : transceiver->senders()) { + senders.push_back( + rtc::scoped_refptr<RtpSenderInternal>(sender->internal())); + } + std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers; + for (const auto& receiver : transceiver->receivers()) { + receivers.push_back( + rtc::scoped_refptr<RtpReceiverInternal>(receiver->internal())); + } + stats.track_media_info_map.Initialize(std::move(voice_media_info), + std::move(video_media_info), + senders, receivers); + } + + call_stats_ = pc_->GetCallStats(); + audio_device_stats_ = pc_->GetAudioDeviceStats(); + }); + + for (auto& stats : transceiver_stats_infos_) { + stats.current_direction = stats.transceiver->current_direction(); + } +} + +void RTCStatsCollector::OnSctpDataChannelStateChanged( + int channel_id, + DataChannelInterface::DataState state) { + RTC_DCHECK_RUN_ON(signaling_thread_); + if (state == DataChannelInterface::DataState::kOpen) { + bool result = + internal_record_.opened_data_channels.insert(channel_id).second; + RTC_DCHECK(result); + ++internal_record_.data_channels_opened; + } else if (state == DataChannelInterface::DataState::kClosed) { + // Only channels that have been fully opened (and have increased the + // `data_channels_opened_` counter) increase the closed counter. + if (internal_record_.opened_data_channels.erase(channel_id)) { + ++internal_record_.data_channels_closed; + } + } +} + +const char* CandidateTypeToRTCIceCandidateTypeForTesting( + const std::string& type) { + return CandidateTypeToRTCIceCandidateType(type); +} + +const char* DataStateToRTCDataChannelStateForTesting( + DataChannelInterface::DataState state) { + return DataStateToRTCDataChannelState(state); +} + +} // namespace webrtc |