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webrtc_m130/stats/rtcstats_objects.cc
Fredrik Hernqvist efbe753617 Add RTCAudioPlayoutStats to GetStats(). 
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.

Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
2023-01-16 13:19:45 +00:00

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/*
* 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(std::string id, int64_t timestamp_us)
: RTCCertificateStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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(std::string id, int64_t timestamp_us)
: RTCCodecStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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(std::string id, int64_t timestamp_us)
: RTCDataChannelStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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(std::string id,
int64_t timestamp_us)
: RTCIceCandidatePairStats(std::move(id), Timestamp::Micros(timestamp_us)) {
}
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(std::string id,
int64_t timestamp_us,
bool is_remote)
: RTCIceCandidateStats(std::move(id),
Timestamp::Micros(timestamp_us),
is_remote) {}
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) {}
RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(std::string id,
int64_t timestamp_us)
: RTCLocalIceCandidateStats(std::move(id),
Timestamp::Micros(timestamp_us)) {}
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) {}
RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(std::string id,
int64_t timestamp_us)
: RTCRemoteIceCandidateStats(std::move(id),
Timestamp::Micros(timestamp_us)) {}
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(DEPRECATED_RTCMediaStreamStats, RTCStats, "stream",
&stream_identifier,
&track_ids)
// clang-format on
DEPRECATED_RTCMediaStreamStats::DEPRECATED_RTCMediaStreamStats(
std::string id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
stream_identifier("streamIdentifier"),
track_ids("trackIds") {}
DEPRECATED_RTCMediaStreamStats::DEPRECATED_RTCMediaStreamStats(
std::string id,
int64_t timestamp_us)
: DEPRECATED_RTCMediaStreamStats(std::move(id),
Timestamp::Micros(timestamp_us)) {}
DEPRECATED_RTCMediaStreamStats::DEPRECATED_RTCMediaStreamStats(
const DEPRECATED_RTCMediaStreamStats& other) = default;
DEPRECATED_RTCMediaStreamStats::~DEPRECATED_RTCMediaStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(DEPRECATED_RTCMediaStreamTrackStats, RTCStats, "track",
&track_identifier,
&media_source_id,
&remote_source,
&ended,
&detached,
&kind,
&jitter_buffer_delay,
&jitter_buffer_emitted_count,
&frame_width,
&frame_height,
&frames_sent,
&huge_frames_sent,
&frames_received,
&frames_decoded,
&frames_dropped,
&audio_level,
&total_audio_energy,
&echo_return_loss,
&echo_return_loss_enhancement,
&total_samples_received,
&total_samples_duration,
&concealed_samples,
&silent_concealed_samples,
&concealment_events,
&inserted_samples_for_deceleration,
&removed_samples_for_acceleration)
// clang-format on
DEPRECATED_RTCMediaStreamTrackStats::DEPRECATED_RTCMediaStreamTrackStats(
std::string id,
Timestamp timestamp,
const char* kind)
: RTCStats(std::move(id), timestamp),
track_identifier("trackIdentifier"),
media_source_id("mediaSourceId"),
remote_source("remoteSource"),
ended("ended"),
detached("detached"),
kind("kind", kind),
jitter_buffer_delay("jitterBufferDelay"),
jitter_buffer_emitted_count("jitterBufferEmittedCount"),
frame_width("frameWidth"),
frame_height("frameHeight"),
frames_sent("framesSent"),
huge_frames_sent("hugeFramesSent"),
frames_received("framesReceived"),
frames_decoded("framesDecoded"),
frames_dropped("framesDropped"),
audio_level("audioLevel"),
total_audio_energy("totalAudioEnergy"),
echo_return_loss("echoReturnLoss"),
echo_return_loss_enhancement("echoReturnLossEnhancement"),
total_samples_received("totalSamplesReceived"),
total_samples_duration("totalSamplesDuration"),
concealed_samples("concealedSamples"),
silent_concealed_samples("silentConcealedSamples"),
concealment_events("concealmentEvents"),
inserted_samples_for_deceleration("insertedSamplesForDeceleration"),
removed_samples_for_acceleration("removedSamplesForAcceleration") {
RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
kind == RTCMediaStreamTrackKind::kVideo);
}
DEPRECATED_RTCMediaStreamTrackStats::DEPRECATED_RTCMediaStreamTrackStats(
std::string id,
int64_t timestamp_us,
const char* kind)
: DEPRECATED_RTCMediaStreamTrackStats(std::move(id),
Timestamp::Micros(timestamp_us),
kind) {}
DEPRECATED_RTCMediaStreamTrackStats::DEPRECATED_RTCMediaStreamTrackStats(
const DEPRECATED_RTCMediaStreamTrackStats& other) = default;
DEPRECATED_RTCMediaStreamTrackStats::~DEPRECATED_RTCMediaStreamTrackStats() {}
// 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(std::string id,
int64_t timestamp_us)
: RTCPeerConnectionStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
RTCPeerConnectionStats::RTCPeerConnectionStats(
const RTCPeerConnectionStats& other) = default;
RTCPeerConnectionStats::~RTCPeerConnectionStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(RTCRTPStreamStats, RTCStats, "rtp",
&ssrc,
&kind,
&track_id,
&transport_id,
&codec_id,
&media_type)
// clang-format on
RTCRTPStreamStats::RTCRTPStreamStats(std::string id, Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
ssrc("ssrc"),
kind("kind"),
track_id("trackId"),
transport_id("transportId"),
codec_id("codecId"),
media_type("mediaType") {}
RTCRTPStreamStats::RTCRTPStreamStats(std::string id, int64_t timestamp_us)
: RTCRTPStreamStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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(std::string id,
int64_t timestamp_us)
: RTCReceivedRtpStreamStats(std::move(id),
Timestamp::Micros(timestamp_us)) {}
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(std::string(id),
int64_t timestamp_us)
: RTCSentRtpStreamStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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,
&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,
&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),
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"),
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",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets}),
delayed_packet_outage_samples(
"delayedPacketOutageSamples",
{NonStandardGroupId::kRtcAudioJitterBufferMaxPackets,
NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
relative_packet_arrival_delay(
"relativePacketArrivalDelay",
{NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
interruption_count("interruptionCount"),
total_interruption_duration("totalInterruptionDuration"),
min_playout_delay("minPlayoutDelay") {}
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string id,
int64_t timestamp_us)
: RTCInboundRTPStreamStats(std::move(id), Timestamp::Micros(timestamp_us)) {
}
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& other) = default;
RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCOutboundRTPStreamStats, RTCRTPStreamStats, "outbound-rtp",
&media_source_id,
&remote_id,
&mid,
&rid,
&packets_sent,
&retransmitted_packets_sent,
&bytes_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)
: RTCRTPStreamStats(std::move(id), timestamp),
media_source_id("mediaSourceId"),
remote_id("remoteId"),
mid("mid"),
rid("rid"),
packets_sent("packetsSent"),
retransmitted_packets_sent("retransmittedPacketsSent"),
bytes_sent("bytesSent"),
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(std::string id,
int64_t timestamp_us)
: RTCOutboundRTPStreamStats(std::move(id),
Timestamp::Micros(timestamp_us)) {}
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(
std::string id,
int64_t timestamp_us)
: RTCRemoteInboundRtpStreamStats(std::move(id),
Timestamp::Micros(timestamp_us)) {}
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(
std::string id,
int64_t timestamp)
: RTCRemoteOutboundRtpStreamStats(std::move(id),
Timestamp::Micros(timestamp)) {}
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(std::string id, int64_t timestamp_us)
: RTCMediaSourceStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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(std::string id, int64_t timestamp_us)
: RTCAudioSourceStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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(std::string id, int64_t timestamp_us)
: RTCVideoSourceStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
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(std::string id, int64_t timestamp_us)
: RTCTransportStats(std::move(id), Timestamp::Micros(timestamp_us)) {}
RTCTransportStats::RTCTransportStats(const RTCTransportStats& other) = default;
RTCTransportStats::~RTCTransportStats() {}
RTCAudioPlayoutStats::RTCAudioPlayoutStats(const std::string& id,
Timestamp timestamp)
: RTCStats(std::move(id), timestamp),
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, "audio-playout",
&synthesized_samples_duration,
&synthesized_samples_events,
&total_samples_duration,
&total_playout_delay,
&total_samples_count)
// clang-format on
} // namespace webrtc
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