blob: 67cb1c58f2e905a2c704dcad67e6c73899a4821f [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 "api/stats/rtcstats_objects.h"
#include <utility>
#include "api/stats/rtc_stats.h"
#include "rtc_base/checks.h"
namespace webrtc {
const char* const RTCDataChannelState::kConnecting = "connecting";
const char* const RTCDataChannelState::kOpen = "open";
const char* const RTCDataChannelState::kClosing = "closing";
const char* const RTCDataChannelState::kClosed = "closed";
const char* const RTCStatsIceCandidatePairState::kFrozen = "frozen";
const char* const RTCStatsIceCandidatePairState::kWaiting = "waiting";
const char* const RTCStatsIceCandidatePairState::kInProgress = "in-progress";
const char* const RTCStatsIceCandidatePairState::kFailed = "failed";
const char* const RTCStatsIceCandidatePairState::kSucceeded = "succeeded";
// Strings defined in https://tools.ietf.org/html/rfc5245.
const char* const RTCIceCandidateType::kHost = "host";
const char* const RTCIceCandidateType::kSrflx = "srflx";
const char* const RTCIceCandidateType::kPrflx = "prflx";
const char* const RTCIceCandidateType::kRelay = "relay";
const char* const RTCDtlsTransportState::kNew = "new";
const char* const RTCDtlsTransportState::kConnecting = "connecting";
const char* const RTCDtlsTransportState::kConnected = "connected";
const char* const RTCDtlsTransportState::kClosed = "closed";
const char* const RTCDtlsTransportState::kFailed = "failed";
const char* const RTCMediaStreamTrackKind::kAudio = "audio";
const char* const RTCMediaStreamTrackKind::kVideo = "video";
// https://w3c.github.io/webrtc-stats/#dom-rtcnetworktype
const char* const RTCNetworkType::kBluetooth = "bluetooth";
const char* const RTCNetworkType::kCellular = "cellular";
const char* const RTCNetworkType::kEthernet = "ethernet";
const char* const RTCNetworkType::kWifi = "wifi";
const char* const RTCNetworkType::kWimax = "wimax";
const char* const RTCNetworkType::kVpn = "vpn";
const char* const RTCNetworkType::kUnknown = "unknown";
// https://w3c.github.io/webrtc-stats/#dom-rtcqualitylimitationreason
const char* const RTCQualityLimitationReason::kNone = "none";
const char* const RTCQualityLimitationReason::kCpu = "cpu";
const char* const RTCQualityLimitationReason::kBandwidth = "bandwidth";
const char* const RTCQualityLimitationReason::kOther = "other";
// https://webrtc.org/experiments/rtp-hdrext/video-content-type/
const char* const RTCContentType::kUnspecified = "unspecified";
const char* const RTCContentType::kScreenshare = "screenshare";
// https://w3c.github.io/webrtc-stats/#dom-rtcdtlsrole
const char* const RTCDtlsRole::kUnknown = "unknown";
const char* const RTCDtlsRole::kClient = "client";
const char* const RTCDtlsRole::kServer = "server";
// https://www.w3.org/TR/webrtc/#rtcicerole
const char* const RTCIceRole::kUnknown = "unknown";
const char* const RTCIceRole::kControlled = "controlled";
const char* const RTCIceRole::kControlling = "controlling";
// https://www.w3.org/TR/webrtc/#dom-rtcicetransportstate
const char* const RTCIceTransportState::kNew = "new";
const char* const RTCIceTransportState::kChecking = "checking";
const char* const RTCIceTransportState::kConnected = "connected";
const char* const RTCIceTransportState::kCompleted = "completed";
const char* const RTCIceTransportState::kDisconnected = "disconnected";
const char* const RTCIceTransportState::kFailed = "failed";
const char* const RTCIceTransportState::kClosed = "closed";
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCCertificateStats, RTCStats, "certificate",
&fingerprint,
&fingerprint_algorithm,
&base64_certificate,
&issuer_certificate_id)
// clang-format on
RTCCertificateStats::RTCCertificateStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
fingerprint("fingerprint"),
fingerprint_algorithm("fingerprintAlgorithm"),
base64_certificate("base64Certificate"),
issuer_certificate_id("issuerCertificateId") {}
RTCCertificateStats::RTCCertificateStats(const RTCCertificateStats& other) =
default;
RTCCertificateStats::~RTCCertificateStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCCodecStats, RTCStats, "codec",
&transport_id,
&payload_type,
&mime_type,
&clock_rate,
&channels,
&sdp_fmtp_line)
// clang-format on
RTCCodecStats::RTCCodecStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
transport_id("transportId"),
payload_type("payloadType"),
mime_type("mimeType"),
clock_rate("clockRate"),
channels("channels"),
sdp_fmtp_line("sdpFmtpLine") {}
RTCCodecStats::RTCCodecStats(const RTCCodecStats& other) = default;
RTCCodecStats::~RTCCodecStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCDataChannelStats, RTCStats, "data-channel",
&label,
&protocol,
&data_channel_identifier,
&state,
&messages_sent,
&bytes_sent,
&messages_received,
&bytes_received)
// clang-format on
RTCDataChannelStats::RTCDataChannelStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
label("label"),
protocol("protocol"),
data_channel_identifier("dataChannelIdentifier"),
state("state"),
messages_sent("messagesSent"),
bytes_sent("bytesSent"),
messages_received("messagesReceived"),
bytes_received("bytesReceived") {}
RTCDataChannelStats::RTCDataChannelStats(const RTCDataChannelStats& other) =
default;
RTCDataChannelStats::~RTCDataChannelStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair",
&transport_id,
&local_candidate_id,
&remote_candidate_id,
&state,
&priority,
&nominated,
&writable,
&packets_sent,
&packets_received,
&bytes_sent,
&bytes_received,
&total_round_trip_time,
&current_round_trip_time,
&available_outgoing_bitrate,
&available_incoming_bitrate,
&requests_received,
&requests_sent,
&responses_received,
&responses_sent,
&consent_requests_sent,
&packets_discarded_on_send,
&bytes_discarded_on_send,
&last_packet_received_timestamp,
&last_packet_sent_timestamp)
// clang-format on
RTCIceCandidatePairStats::RTCIceCandidatePairStats(std::string id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
transport_id("transportId"),
local_candidate_id("localCandidateId"),
remote_candidate_id("remoteCandidateId"),
state("state"),
priority("priority"),
nominated("nominated"),
writable("writable"),
packets_sent("packetsSent"),
packets_received("packetsReceived"),
bytes_sent("bytesSent"),
bytes_received("bytesReceived"),
total_round_trip_time("totalRoundTripTime"),
current_round_trip_time("currentRoundTripTime"),
available_outgoing_bitrate("availableOutgoingBitrate"),
available_incoming_bitrate("availableIncomingBitrate"),
requests_received("requestsReceived"),
requests_sent("requestsSent"),
responses_received("responsesReceived"),
responses_sent("responsesSent"),
consent_requests_sent("consentRequestsSent"),
packets_discarded_on_send("packetsDiscardedOnSend"),
bytes_discarded_on_send("bytesDiscardedOnSend"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
last_packet_sent_timestamp("lastPacketSentTimestamp") {}
RTCIceCandidatePairStats::RTCIceCandidatePairStats(
const RTCIceCandidatePairStats& other) = default;
RTCIceCandidatePairStats::~RTCIceCandidatePairStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCIceCandidateStats, RTCStats, "abstract-ice-candidate",
&transport_id,
&is_remote,
&network_type,
&ip,
&address,
&port,
&protocol,
&relay_protocol,
&candidate_type,
&priority,
&url,
&foundation,
&related_address,
&related_port,
&username_fragment,
&tcp_type,
&vpn,
&network_adapter_type)
// clang-format on
RTCIceCandidateStats::RTCIceCandidateStats(std::string id,
Timestamp timestamp,
bool is_remote)
: RTCStats(std::move(id), timestamp),
transport_id("transportId"),
is_remote("isRemote", is_remote),
network_type("networkType"),
ip("ip"),
address("address"),
port("port"),
protocol("protocol"),
relay_protocol("relayProtocol"),
candidate_type("candidateType"),
priority("priority"),
url("url"),
foundation("foundation"),
related_address("relatedAddress"),
related_port("relatedPort"),
username_fragment("usernameFragment"),
tcp_type("tcpType"),
vpn("vpn"),
network_adapter_type("networkAdapterType") {}
RTCIceCandidateStats::RTCIceCandidateStats(const RTCIceCandidateStats& other) =
default;
RTCIceCandidateStats::~RTCIceCandidateStats() {}
const char RTCLocalIceCandidateStats::kType[] = "local-candidate";
RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(std::string id,
Timestamp timestamp)
: RTCIceCandidateStats(std::move(id), timestamp, false) {}
std::unique_ptr<RTCStats> RTCLocalIceCandidateStats::copy() const {
return std::make_unique<RTCLocalIceCandidateStats>(*this);
}
const char* RTCLocalIceCandidateStats::type() const {
return kType;
}
const char RTCRemoteIceCandidateStats::kType[] = "remote-candidate";
RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(std::string id,
Timestamp timestamp)
: RTCIceCandidateStats(std::move(id), timestamp, true) {}
std::unique_ptr<RTCStats> RTCRemoteIceCandidateStats::copy() const {
return std::make_unique<RTCRemoteIceCandidateStats>(*this);
}
const char* RTCRemoteIceCandidateStats::type() const {
return kType;
}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
&data_channels_opened,
&data_channels_closed)
// clang-format on
RTCPeerConnectionStats::RTCPeerConnectionStats(std::string id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
data_channels_opened("dataChannelsOpened"),
data_channels_closed("dataChannelsClosed") {}
RTCPeerConnectionStats::RTCPeerConnectionStats(
const RTCPeerConnectionStats& other) = default;
RTCPeerConnectionStats::~RTCPeerConnectionStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCRtpStreamStats, RTCStats, "rtp",
&ssrc,
&kind,
&transport_id,
&codec_id)
// clang-format on
RTCRtpStreamStats::RTCRtpStreamStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
ssrc("ssrc"),
kind("kind"),
transport_id("transportId"),
codec_id("codecId") {}
RTCRtpStreamStats::RTCRtpStreamStats(const RTCRtpStreamStats& other) = default;
RTCRtpStreamStats::~RTCRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCReceivedRtpStreamStats, RTCRtpStreamStats, "received-rtp",
&jitter,
&packets_lost)
// clang-format on
RTCReceivedRtpStreamStats::RTCReceivedRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCRtpStreamStats(std::move(id), timestamp),
jitter("jitter"),
packets_lost("packetsLost") {}
RTCReceivedRtpStreamStats::RTCReceivedRtpStreamStats(
const RTCReceivedRtpStreamStats& other) = default;
RTCReceivedRtpStreamStats::~RTCReceivedRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCSentRtpStreamStats, RTCRtpStreamStats, "sent-rtp",
&packets_sent,
&bytes_sent)
// clang-format on
RTCSentRtpStreamStats::RTCSentRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCRtpStreamStats(std::move(id), timestamp),
packets_sent("packetsSent"),
bytes_sent("bytesSent") {}
RTCSentRtpStreamStats::RTCSentRtpStreamStats(
const RTCSentRtpStreamStats& other) = default;
RTCSentRtpStreamStats::~RTCSentRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCInboundRtpStreamStats, RTCReceivedRtpStreamStats, "inbound-rtp",
&track_identifier,
&mid,
&remote_id,
&packets_received,
&packets_discarded,
&fec_packets_received,
&fec_packets_discarded,
&bytes_received,
&header_bytes_received,
&retransmitted_packets_received,
&retransmitted_bytes_received,
&last_packet_received_timestamp,
&jitter_buffer_delay,
&jitter_buffer_target_delay,
&jitter_buffer_minimum_delay,
&jitter_buffer_emitted_count,
&total_samples_received,
&concealed_samples,
&silent_concealed_samples,
&concealment_events,
&inserted_samples_for_deceleration,
&removed_samples_for_acceleration,
&audio_level,
&total_audio_energy,
&total_samples_duration,
&playout_id,
&frames_received,
&frame_width,
&frame_height,
&frames_per_second,
&frames_decoded,
&key_frames_decoded,
&frames_dropped,
&total_decode_time,
&total_processing_delay,
&total_assembly_time,
&frames_assembled_from_multiple_packets,
&total_inter_frame_delay,
&total_squared_inter_frame_delay,
&pause_count,
&total_pauses_duration,
&freeze_count,
&total_freezes_duration,
&content_type,
&estimated_playout_timestamp,
&decoder_implementation,
&fir_count,
&pli_count,
&nack_count,
&qp_sum,
&goog_timing_frame_info,
&power_efficient_decoder,
&jitter_buffer_flushes,
&delayed_packet_outage_samples,
&relative_packet_arrival_delay,
&interruption_count,
&total_interruption_duration,
&min_playout_delay)
// clang-format on
RTCInboundRtpStreamStats::RTCInboundRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCReceivedRtpStreamStats(std::move(id), timestamp),
playout_id("playoutId"),
track_identifier("trackIdentifier"),
mid("mid"),
remote_id("remoteId"),
packets_received("packetsReceived"),
packets_discarded("packetsDiscarded"),
fec_packets_received("fecPacketsReceived"),
fec_packets_discarded("fecPacketsDiscarded"),
bytes_received("bytesReceived"),
header_bytes_received("headerBytesReceived"),
retransmitted_packets_received("retransmittedPacketsReceived"),
retransmitted_bytes_received("retransmittedBytesReceived"),
last_packet_received_timestamp("lastPacketReceivedTimestamp"),
jitter_buffer_delay("jitterBufferDelay"),
jitter_buffer_target_delay("jitterBufferTargetDelay"),
jitter_buffer_minimum_delay("jitterBufferMinimumDelay"),
jitter_buffer_emitted_count("jitterBufferEmittedCount"),
total_samples_received("totalSamplesReceived"),
concealed_samples("concealedSamples"),
silent_concealed_samples("silentConcealedSamples"),
concealment_events("concealmentEvents"),
inserted_samples_for_deceleration("insertedSamplesForDeceleration"),
removed_samples_for_acceleration("removedSamplesForAcceleration"),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
total_samples_duration("totalSamplesDuration"),
frames_received("framesReceived"),
frame_width("frameWidth"),
frame_height("frameHeight"),
frames_per_second("framesPerSecond"),
frames_decoded("framesDecoded"),
key_frames_decoded("keyFramesDecoded"),
frames_dropped("framesDropped"),
total_decode_time("totalDecodeTime"),
total_processing_delay("totalProcessingDelay"),
total_assembly_time("totalAssemblyTime"),
frames_assembled_from_multiple_packets(
"framesAssembledFromMultiplePackets"),
total_inter_frame_delay("totalInterFrameDelay"),
total_squared_inter_frame_delay("totalSquaredInterFrameDelay"),
pause_count("pauseCount"),
total_pauses_duration("totalPausesDuration"),
freeze_count("freezeCount"),
total_freezes_duration("totalFreezesDuration"),
content_type("contentType"),
estimated_playout_timestamp("estimatedPlayoutTimestamp"),
decoder_implementation("decoderImplementation"),
fir_count("firCount"),
pli_count("pliCount"),
nack_count("nackCount"),
qp_sum("qpSum"),
goog_timing_frame_info("googTimingFrameInfo"),
power_efficient_decoder("powerEfficientDecoder"),
jitter_buffer_flushes("jitterBufferFlushes"),
delayed_packet_outage_samples("delayedPacketOutageSamples"),
relative_packet_arrival_delay("relativePacketArrivalDelay"),
interruption_count("interruptionCount"),
total_interruption_duration("totalInterruptionDuration"),
min_playout_delay("minPlayoutDelay") {}
RTCInboundRtpStreamStats::RTCInboundRtpStreamStats(
const RTCInboundRtpStreamStats& other) = default;
RTCInboundRtpStreamStats::~RTCInboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCOutboundRtpStreamStats, RTCSentRtpStreamStats, "outbound-rtp",
&media_source_id,
&remote_id,
&mid,
&rid,
&retransmitted_packets_sent,
&header_bytes_sent,
&retransmitted_bytes_sent,
&target_bitrate,
&frames_encoded,
&key_frames_encoded,
&total_encode_time,
&total_encoded_bytes_target,
&frame_width,
&frame_height,
&frames_per_second,
&frames_sent,
&huge_frames_sent,
&total_packet_send_delay,
&quality_limitation_reason,
&quality_limitation_durations,
&quality_limitation_resolution_changes,
&content_type,
&encoder_implementation,
&fir_count,
&pli_count,
&nack_count,
&qp_sum,
&active,
&power_efficient_encoder,
&scalability_mode)
// clang-format on
RTCOutboundRtpStreamStats::RTCOutboundRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCSentRtpStreamStats(std::move(id), timestamp),
media_source_id("mediaSourceId"),
remote_id("remoteId"),
mid("mid"),
rid("rid"),
retransmitted_packets_sent("retransmittedPacketsSent"),
header_bytes_sent("headerBytesSent"),
retransmitted_bytes_sent("retransmittedBytesSent"),
target_bitrate("targetBitrate"),
frames_encoded("framesEncoded"),
key_frames_encoded("keyFramesEncoded"),
total_encode_time("totalEncodeTime"),
total_encoded_bytes_target("totalEncodedBytesTarget"),
frame_width("frameWidth"),
frame_height("frameHeight"),
frames_per_second("framesPerSecond"),
frames_sent("framesSent"),
huge_frames_sent("hugeFramesSent"),
total_packet_send_delay("totalPacketSendDelay"),
quality_limitation_reason("qualityLimitationReason"),
quality_limitation_durations("qualityLimitationDurations"),
quality_limitation_resolution_changes(
"qualityLimitationResolutionChanges"),
content_type("contentType"),
encoder_implementation("encoderImplementation"),
fir_count("firCount"),
pli_count("pliCount"),
nack_count("nackCount"),
qp_sum("qpSum"),
active("active"),
power_efficient_encoder("powerEfficientEncoder"),
scalability_mode("scalabilityMode") {}
RTCOutboundRtpStreamStats::RTCOutboundRtpStreamStats(
const RTCOutboundRtpStreamStats& other) = default;
RTCOutboundRtpStreamStats::~RTCOutboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCRemoteInboundRtpStreamStats, RTCReceivedRtpStreamStats,
"remote-inbound-rtp",
&local_id,
&round_trip_time,
&fraction_lost,
&total_round_trip_time,
&round_trip_time_measurements)
// clang-format on
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
std::string id,
Timestamp timestamp)
: RTCReceivedRtpStreamStats(std::move(id), timestamp),
local_id("localId"),
round_trip_time("roundTripTime"),
fraction_lost("fractionLost"),
total_round_trip_time("totalRoundTripTime"),
round_trip_time_measurements("roundTripTimeMeasurements") {}
RTCRemoteInboundRtpStreamStats::RTCRemoteInboundRtpStreamStats(
const RTCRemoteInboundRtpStreamStats& other) = default;
RTCRemoteInboundRtpStreamStats::~RTCRemoteInboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCRemoteOutboundRtpStreamStats, RTCSentRtpStreamStats,
"remote-outbound-rtp",
&local_id,
&remote_timestamp,
&reports_sent,
&round_trip_time,
&round_trip_time_measurements,
&total_round_trip_time)
// clang-format on
RTCRemoteOutboundRtpStreamStats::RTCRemoteOutboundRtpStreamStats(
std::string id,
Timestamp timestamp)
: RTCSentRtpStreamStats(std::move(id), timestamp),
local_id("localId"),
remote_timestamp("remoteTimestamp"),
reports_sent("reportsSent"),
round_trip_time("roundTripTime"),
round_trip_time_measurements("roundTripTimeMeasurements"),
total_round_trip_time("totalRoundTripTime") {}
RTCRemoteOutboundRtpStreamStats::RTCRemoteOutboundRtpStreamStats(
const RTCRemoteOutboundRtpStreamStats& other) = default;
RTCRemoteOutboundRtpStreamStats::~RTCRemoteOutboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCMediaSourceStats, RTCStats, "parent-media-source",
&track_identifier,
&kind)
// clang-format on
RTCMediaSourceStats::RTCMediaSourceStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
track_identifier("trackIdentifier"),
kind("kind") {}
RTCMediaSourceStats::RTCMediaSourceStats(const RTCMediaSourceStats& other) =
default;
RTCMediaSourceStats::~RTCMediaSourceStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCAudioSourceStats, RTCMediaSourceStats, "media-source",
&audio_level,
&total_audio_energy,
&total_samples_duration,
&echo_return_loss,
&echo_return_loss_enhancement)
// clang-format on
RTCAudioSourceStats::RTCAudioSourceStats(std::string id, Timestamp timestamp)
: RTCMediaSourceStats(std::move(id), timestamp),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
total_samples_duration("totalSamplesDuration"),
echo_return_loss("echoReturnLoss"),
echo_return_loss_enhancement("echoReturnLossEnhancement") {}
RTCAudioSourceStats::RTCAudioSourceStats(const RTCAudioSourceStats& other) =
default;
RTCAudioSourceStats::~RTCAudioSourceStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCVideoSourceStats, RTCMediaSourceStats, "media-source",
&width,
&height,
&frames,
&frames_per_second)
// clang-format on
RTCVideoSourceStats::RTCVideoSourceStats(std::string id, Timestamp timestamp)
: RTCMediaSourceStats(std::move(id), timestamp),
width("width"),
height("height"),
frames("frames"),
frames_per_second("framesPerSecond") {}
RTCVideoSourceStats::RTCVideoSourceStats(const RTCVideoSourceStats& other) =
default;
RTCVideoSourceStats::~RTCVideoSourceStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCTransportStats, RTCStats, "transport",
&bytes_sent,
&packets_sent,
&bytes_received,
&packets_received,
&rtcp_transport_stats_id,
&dtls_state,
&selected_candidate_pair_id,
&local_certificate_id,
&remote_certificate_id,
&tls_version,
&dtls_cipher,
&dtls_role,
&srtp_cipher,
&selected_candidate_pair_changes,
&ice_role,
&ice_local_username_fragment,
&ice_state)
// clang-format on
RTCTransportStats::RTCTransportStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
bytes_sent("bytesSent"),
packets_sent("packetsSent"),
bytes_received("bytesReceived"),
packets_received("packetsReceived"),
rtcp_transport_stats_id("rtcpTransportStatsId"),
dtls_state("dtlsState"),
selected_candidate_pair_id("selectedCandidatePairId"),
local_certificate_id("localCertificateId"),
remote_certificate_id("remoteCertificateId"),
tls_version("tlsVersion"),
dtls_cipher("dtlsCipher"),
dtls_role("dtlsRole"),
srtp_cipher("srtpCipher"),
selected_candidate_pair_changes("selectedCandidatePairChanges"),
ice_role("iceRole"),
ice_local_username_fragment("iceLocalUsernameFragment"),
ice_state("iceState") {}
RTCTransportStats::RTCTransportStats(const RTCTransportStats& other) = default;
RTCTransportStats::~RTCTransportStats() {}
RTCAudioPlayoutStats::RTCAudioPlayoutStats(const std::string& id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
kind("kind", "audio"),
synthesized_samples_duration("synthesizedSamplesDuration"),
synthesized_samples_events("synthesizedSamplesEvents"),
total_samples_duration("totalSamplesDuration"),
total_playout_delay("totalPlayoutDelay"),
total_samples_count("totalSamplesCount") {}
RTCAudioPlayoutStats::RTCAudioPlayoutStats(const RTCAudioPlayoutStats& other) =
default;
RTCAudioPlayoutStats::~RTCAudioPlayoutStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCAudioPlayoutStats, RTCStats, "media-playout",
&kind,
&synthesized_samples_duration,
&synthesized_samples_events,
&total_samples_duration,
&total_playout_delay,
&total_samples_count)
// clang-format on
} // namespace webrtc