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webrtc / src / 24992e95183d4c5a853b6e791485851a0cf55cbd / . / pc / rtc_stats_collector.cc
blob: 451590fced9558311d84e5fdf414164130bb6b4f [file] [log] [blame]
/*
* 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/audio/audio_device.h"
#include "api/audio/audio_processing_statistics.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/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';
static constexpr 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* 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<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;
}
inbound_stats->total_processing_delay =
media_receiver_info.total_processing_delay_seconds;
}
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,
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>
CreateRemoteOutboundMediaStreamStats(
const cricket::MediaReceiverInfo& media_receiver_info,
const std::string& mid,
cricket::MediaType media_type,
const RTCInboundRtpStreamStats& inbound_audio_stats,
const std::string& transport_id,
const bool stats_timestamp_with_environment_clock) {
std::optional<Timestamp> last_sender_report_timestamp =
stats_timestamp_with_environment_clock
? media_receiver_info.last_sender_report_timestamp
: media_receiver_info.last_sender_report_utc_timestamp;
if (!last_sender_report_timestamp.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(media_receiver_info.sender_reports_reports_count, 0);
// Create.
auto stats = std::make_unique<RTCRemoteOutboundRtpStreamStats>(
/*id=*/RTCRemoteOutboundRTPStreamStatsIDFromSSRC(
media_type, media_receiver_info.ssrc()),
*last_sender_report_timestamp);
// Populate.
// - RTCRtpStreamStats.
stats->ssrc = media_receiver_info.ssrc();
stats->kind = cricket::MediaTypeToString(media_type);
stats->transport_id = transport_id;
if (inbound_audio_stats.codec_id.has_value()) {
stats->codec_id = *inbound_audio_stats.codec_id;
}
// - RTCSentRtpStreamStats.
stats->packets_sent = media_receiver_info.sender_reports_packets_sent;
stats->bytes_sent = media_receiver_info.sender_reports_bytes_sent;
// - RTCRemoteOutboundRtpStreamStats.
stats->local_id = inbound_audio_stats.id();
// last_sender_report_remote_utc_timestamp_ms is set together with
// last_sender_report_utc_timestamp_ms.
RTC_DCHECK(
media_receiver_info.last_sender_report_remote_utc_timestamp.has_value());
stats->remote_timestamp =
media_receiver_info.last_sender_report_remote_utc_timestamp->ms<double>();
stats->reports_sent = media_receiver_info.sender_reports_reports_count;
if (media_receiver_info.round_trip_time.has_value()) {
stats->round_trip_time =
media_receiver_info.round_trip_time->seconds<double>();
}
stats->round_trip_time_measurements =
media_receiver_info.round_trip_time_measurements;
stats->total_round_trip_time =
media_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.corruption_score_sum.has_value()) {
RTC_CHECK(video_receiver_info.corruption_score_squared_sum.has_value());
RTC_CHECK_GT(video_receiver_info.corruption_score_count, 0);
inbound_video->total_corruption_probability =
*video_receiver_info.corruption_score_sum;
inbound_video->total_squared_corruption_probability =
*video_receiver_info.corruption_score_squared_sum;
inbound_video->corruption_measurements =
video_receiver_info.corruption_score_count;
}
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,
const bool stats_timestamp_with_environment_clock) {
// 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.
Timestamp arrival_timestamp = stats_timestamp_with_environment_clock
? report_block.report_block_timestamp()
: report_block.report_block_timestamp_utc();
auto remote_inbound = std::make_unique<RTCRemoteInboundRtpStreamStats>(
RTCRemoteInboundRtpStreamStatsIdFromSourceSsrc(
media_type, report_block.source_ssrc()),
arrival_timestamp);
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.has_value()
? *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.has_value()
? 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.has_value()) {
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& relay_protocol = candidate.relay_protocol();
const std::string& url = candidate.url();
if (candidate.is_relay() ||
(candidate.is_prflx() && !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.is_stun()) {
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 = candidate.type_name();
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.has_value());
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.
std::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.has_value());
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,
const Environment& env,
int64_t cache_lifetime_us) {
return rtc::make_ref_counted<RTCStatsCollector>(pc, env, cache_lifetime_us);
}
RTCStatsCollector::RTCStatsCollector(PeerConnectionInternal* pc,
const Environment& env,
int64_t cache_lifetime_us)
: pc_(pc),
env_(env),
stats_timestamp_with_environment_clock_(
pc->GetConfiguration().stats_timestamp_with_environment_clock()),
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.
Timestamp timestamp =
stats_timestamp_with_environment_clock_
?
// "Now" using a monotonically increasing timer.
env_.clock().CurrentTime()
:
// "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::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,
std::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", TRACE_EVENT_SCOPE_GLOBAL,
"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 = CreateRemoteOutboundMediaStreamStats(
voice_receiver_info, mid, cricket::MEDIA_TYPE_AUDIO, *inbound_audio_ptr,
transport_id, stats_timestamp_with_environment_clock_);
// 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,
stats_timestamp_with_environment_clock_));
}
}
}
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 and remote-outbound.
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();
}
auto* inbound_video_ptr = report->TryAddStats(std::move(inbound_video));
if (!inbound_video_ptr) {
RTC_LOG(LS_ERROR)
<< "Unable to add video 'inbound-rtp' to report, ID is not unique.";
continue;
}
// Remote-outbound.
auto remote_outbound_video = CreateRemoteOutboundMediaStreamStats(
video_receiver_info, mid, cricket::MEDIA_TYPE_VIDEO, *inbound_video_ptr,
transport_id, stats_timestamp_with_environment_clock_);
// Add stats.
if (remote_outbound_video) {
// When the remote outbound stats are available, the remote ID for the
// local inbound stats is set.
auto* remote_outbound_video_ptr =
report->TryAddStats(std::move(remote_outbound_video));
if (remote_outbound_video_ptr) {
inbound_video_ptr->remote_id = remote_outbound_video_ptr->id();
} else {
RTC_LOG(LS_ERROR) << "Unable to add video 'remote-outbound-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,
stats_timestamp_with_environment_clock_));
}
}
}
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";
}
transport_stats->dtls_cipher = channel_stats.tls_cipher_suite_name;
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
// and keep track of whether we have at least one audio receiver.
bool has_audio_receiver = false;
for (auto& stats : transceiver_stats_infos_) {
auto transceiver = stats.transceiver;
std::optional<cricket::VoiceMediaInfo> voice_media_info;
std::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);
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
has_audio_receiver |= !receivers.empty();
}
}
call_stats_ = pc_->GetCallStats();
audio_device_stats_ =
has_audio_receiver ? pc_->GetAudioDeviceStats() : std::nullopt;
});
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;
}
}
}
} // namespace webrtc