682 lines
27 KiB
C++
682 lines
27 KiB
C++
/*
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* Copyright 2016 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_STATS_RTCSTATS_OBJECTS_H_
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#define API_STATS_RTCSTATS_OBJECTS_H_
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#include <stdint.h>
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/stats/rtc_stats.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// https://w3c.github.io/webrtc-pc/#idl-def-rtcdatachannelstate
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struct RTCDataChannelState {
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static const char* const kConnecting;
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static const char* const kOpen;
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static const char* const kClosing;
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static const char* const kClosed;
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};
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// https://w3c.github.io/webrtc-stats/#dom-rtcstatsicecandidatepairstate
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struct RTCStatsIceCandidatePairState {
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static const char* const kFrozen;
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static const char* const kWaiting;
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static const char* const kInProgress;
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static const char* const kFailed;
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static const char* const kSucceeded;
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};
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// https://w3c.github.io/webrtc-pc/#rtcicecandidatetype-enum
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struct RTCIceCandidateType {
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static const char* const kHost;
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static const char* const kSrflx;
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static const char* const kPrflx;
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static const char* const kRelay;
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};
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// https://w3c.github.io/webrtc-pc/#idl-def-rtcdtlstransportstate
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struct RTCDtlsTransportState {
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static const char* const kNew;
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static const char* const kConnecting;
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static const char* const kConnected;
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static const char* const kClosed;
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static const char* const kFailed;
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};
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// `RTCMediaStreamTrackStats::kind` is not an enum in the spec but the only
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// valid values are "audio" and "video".
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// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-kind
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struct RTCMediaStreamTrackKind {
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static const char* const kAudio;
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static const char* const kVideo;
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};
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// https://w3c.github.io/webrtc-stats/#dom-rtcnetworktype
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struct RTCNetworkType {
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static const char* const kBluetooth;
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static const char* const kCellular;
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static const char* const kEthernet;
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static const char* const kWifi;
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static const char* const kWimax;
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static const char* const kVpn;
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static const char* const kUnknown;
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};
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// https://w3c.github.io/webrtc-stats/#dom-rtcqualitylimitationreason
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struct RTCQualityLimitationReason {
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static const char* const kNone;
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static const char* const kCpu;
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static const char* const kBandwidth;
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static const char* const kOther;
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};
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// https://webrtc.org/experiments/rtp-hdrext/video-content-type/
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struct RTCContentType {
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static const char* const kUnspecified;
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static const char* const kScreenshare;
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};
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// https://w3c.github.io/webrtc-stats/#certificatestats-dict*
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class RTC_EXPORT RTCCertificateStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCCertificateStats(const std::string& id, int64_t timestamp_us);
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RTCCertificateStats(std::string&& id, int64_t timestamp_us);
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RTCCertificateStats(const RTCCertificateStats& other);
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~RTCCertificateStats() override;
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RTCStatsMember<std::string> fingerprint;
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RTCStatsMember<std::string> fingerprint_algorithm;
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RTCStatsMember<std::string> base64_certificate;
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RTCStatsMember<std::string> issuer_certificate_id;
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};
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// https://w3c.github.io/webrtc-stats/#codec-dict*
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class RTC_EXPORT RTCCodecStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCCodecStats(const std::string& id, int64_t timestamp_us);
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RTCCodecStats(std::string&& id, int64_t timestamp_us);
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RTCCodecStats(const RTCCodecStats& other);
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~RTCCodecStats() override;
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RTCStatsMember<std::string> transport_id;
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RTCStatsMember<uint32_t> payload_type;
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RTCStatsMember<std::string> mime_type;
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RTCStatsMember<uint32_t> clock_rate;
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RTCStatsMember<uint32_t> channels;
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RTCStatsMember<std::string> sdp_fmtp_line;
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};
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// https://w3c.github.io/webrtc-stats/#dcstats-dict*
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class RTC_EXPORT RTCDataChannelStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCDataChannelStats(const std::string& id, int64_t timestamp_us);
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RTCDataChannelStats(std::string&& id, int64_t timestamp_us);
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RTCDataChannelStats(const RTCDataChannelStats& other);
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~RTCDataChannelStats() override;
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RTCStatsMember<std::string> label;
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RTCStatsMember<std::string> protocol;
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RTCStatsMember<int32_t> data_channel_identifier;
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// TODO(hbos): Support enum types? "RTCStatsMember<RTCDataChannelState>"?
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RTCStatsMember<std::string> state;
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RTCStatsMember<uint32_t> messages_sent;
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RTCStatsMember<uint64_t> bytes_sent;
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RTCStatsMember<uint32_t> messages_received;
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RTCStatsMember<uint64_t> bytes_received;
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};
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// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
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// TODO(hbos): Tracking bug https://bugs.webrtc.org/7062
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class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCIceCandidatePairStats(const std::string& id, int64_t timestamp_us);
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RTCIceCandidatePairStats(std::string&& id, int64_t timestamp_us);
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RTCIceCandidatePairStats(const RTCIceCandidatePairStats& other);
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~RTCIceCandidatePairStats() override;
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RTCStatsMember<std::string> transport_id;
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RTCStatsMember<std::string> local_candidate_id;
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RTCStatsMember<std::string> remote_candidate_id;
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// TODO(hbos): Support enum types?
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// "RTCStatsMember<RTCStatsIceCandidatePairState>"?
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RTCStatsMember<std::string> state;
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// Obsolete: priority
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RTCStatsMember<uint64_t> priority;
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RTCStatsMember<bool> nominated;
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// TODO(hbos): Collect this the way the spec describes it. We have a value for
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// it but it is not spec-compliant. https://bugs.webrtc.org/7062
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RTCStatsMember<bool> writable;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<bool> readable;
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RTCStatsMember<uint64_t> packets_sent;
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RTCStatsMember<uint64_t> packets_received;
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RTCStatsMember<uint64_t> bytes_sent;
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RTCStatsMember<uint64_t> bytes_received;
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RTCStatsMember<double> total_round_trip_time;
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RTCStatsMember<double> current_round_trip_time;
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RTCStatsMember<double> available_outgoing_bitrate;
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// TODO(hbos): Populate this value. It is wired up and collected the same way
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// "VideoBwe.googAvailableReceiveBandwidth" is, but that value is always
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// undefined. https://bugs.webrtc.org/7062
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RTCStatsMember<double> available_incoming_bitrate;
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RTCStatsMember<uint64_t> requests_received;
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RTCStatsMember<uint64_t> requests_sent;
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RTCStatsMember<uint64_t> responses_received;
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RTCStatsMember<uint64_t> responses_sent;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> retransmissions_received;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> retransmissions_sent;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> consent_requests_received;
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RTCStatsMember<uint64_t> consent_requests_sent;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> consent_responses_received;
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// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
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RTCStatsMember<uint64_t> consent_responses_sent;
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RTCStatsMember<uint64_t> packets_discarded_on_send;
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RTCStatsMember<uint64_t> bytes_discarded_on_send;
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};
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// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
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// TODO(hbos): `RTCStatsCollector` only collects candidates that are part of
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// ice candidate pairs, but there could be candidates not paired with anything.
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// crbug.com/632723
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// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
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// them in the new PeerConnection::GetStats.
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class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCIceCandidateStats(const RTCIceCandidateStats& other);
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~RTCIceCandidateStats() override;
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RTCStatsMember<std::string> transport_id;
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// Obsolete: is_remote
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RTCStatsMember<bool> is_remote;
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RTCStatsMember<std::string> network_type;
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RTCStatsMember<std::string> ip;
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RTCStatsMember<std::string> address;
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RTCStatsMember<int32_t> port;
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RTCStatsMember<std::string> protocol;
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RTCStatsMember<std::string> relay_protocol;
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// TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
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RTCStatsMember<std::string> candidate_type;
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RTCStatsMember<int32_t> priority;
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RTCStatsMember<std::string> url;
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protected:
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RTCIceCandidateStats(const std::string& id,
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int64_t timestamp_us,
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bool is_remote);
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RTCIceCandidateStats(std::string&& id, int64_t timestamp_us, bool is_remote);
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};
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// In the spec both local and remote varieties are of type RTCIceCandidateStats.
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// But here we define them as subclasses of `RTCIceCandidateStats` because the
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// `kType` need to be different ("RTCStatsType type") in the local/remote case.
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// https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
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// This forces us to have to override copy() and type().
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class RTC_EXPORT RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
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public:
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static const char kType[];
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RTCLocalIceCandidateStats(const std::string& id, int64_t timestamp_us);
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RTCLocalIceCandidateStats(std::string&& id, int64_t timestamp_us);
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std::unique_ptr<RTCStats> copy() const override;
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const char* type() const override;
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};
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class RTC_EXPORT RTCRemoteIceCandidateStats final
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: public RTCIceCandidateStats {
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public:
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static const char kType[];
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RTCRemoteIceCandidateStats(const std::string& id, int64_t timestamp_us);
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RTCRemoteIceCandidateStats(std::string&& id, int64_t timestamp_us);
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std::unique_ptr<RTCStats> copy() const override;
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const char* type() const override;
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};
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// https://w3c.github.io/webrtc-stats/#msstats-dict*
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// TODO(hbos): Tracking bug crbug.com/660827
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class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCMediaStreamStats(const std::string& id, int64_t timestamp_us);
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RTCMediaStreamStats(std::string&& id, int64_t timestamp_us);
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RTCMediaStreamStats(const RTCMediaStreamStats& other);
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~RTCMediaStreamStats() override;
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RTCStatsMember<std::string> stream_identifier;
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RTCStatsMember<std::vector<std::string>> track_ids;
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};
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// https://w3c.github.io/webrtc-stats/#mststats-dict*
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// TODO(hbos): Tracking bug crbug.com/659137
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class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCMediaStreamTrackStats(const std::string& id,
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int64_t timestamp_us,
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const char* kind);
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RTCMediaStreamTrackStats(std::string&& id,
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int64_t timestamp_us,
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const char* kind);
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RTCMediaStreamTrackStats(const RTCMediaStreamTrackStats& other);
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~RTCMediaStreamTrackStats() override;
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RTCStatsMember<std::string> track_identifier;
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RTCStatsMember<std::string> media_source_id;
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RTCStatsMember<bool> remote_source;
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RTCStatsMember<bool> ended;
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// TODO(hbos): `RTCStatsCollector` does not return stats for detached tracks.
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// crbug.com/659137
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RTCStatsMember<bool> detached;
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// See `RTCMediaStreamTrackKind` for valid values.
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RTCStatsMember<std::string> kind;
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RTCStatsMember<double> jitter_buffer_delay;
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RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
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// Video-only members
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RTCStatsMember<uint32_t> frame_width;
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RTCStatsMember<uint32_t> frame_height;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<double> frames_per_second;
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RTCStatsMember<uint32_t> frames_sent;
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RTCStatsMember<uint32_t> huge_frames_sent;
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RTCStatsMember<uint32_t> frames_received;
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RTCStatsMember<uint32_t> frames_decoded;
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RTCStatsMember<uint32_t> frames_dropped;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<uint32_t> frames_corrupted;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<uint32_t> partial_frames_lost;
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// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
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RTCStatsMember<uint32_t> full_frames_lost;
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// Audio-only members
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RTCStatsMember<double> audio_level; // Receive-only
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RTCStatsMember<double> total_audio_energy; // Receive-only
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RTCStatsMember<double> echo_return_loss;
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RTCStatsMember<double> echo_return_loss_enhancement;
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RTCStatsMember<uint64_t> total_samples_received;
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RTCStatsMember<double> total_samples_duration; // Receive-only
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RTCStatsMember<uint64_t> concealed_samples;
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RTCStatsMember<uint64_t> silent_concealed_samples;
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RTCStatsMember<uint64_t> concealment_events;
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RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
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RTCStatsMember<uint64_t> removed_samples_for_acceleration;
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// Non-standard audio-only member
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// TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
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RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
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RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
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RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
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// Non-standard metric showing target delay of jitter buffer.
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// This value is increased by the target jitter buffer delay every time a
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// sample is emitted by the jitter buffer. The added target is the target
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// delay, in seconds, at the time that the sample was emitted from the jitter
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// buffer. (https://github.com/w3c/webrtc-provisional-stats/pull/20)
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// Currently it is implemented only for audio.
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// TODO(titovartem) implement for video streams when will be requested.
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RTCNonStandardStatsMember<double> jitter_buffer_target_delay;
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// TODO(henrik.lundin): Add description of the interruption metrics at
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// https://github.com/henbos/webrtc-provisional-stats/issues/17
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RTCNonStandardStatsMember<uint32_t> interruption_count;
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RTCNonStandardStatsMember<double> total_interruption_duration;
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// Non-standard video-only members.
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// https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
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RTCNonStandardStatsMember<uint32_t> freeze_count;
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RTCNonStandardStatsMember<uint32_t> pause_count;
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RTCNonStandardStatsMember<double> total_freezes_duration;
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RTCNonStandardStatsMember<double> total_pauses_duration;
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RTCNonStandardStatsMember<double> total_frames_duration;
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RTCNonStandardStatsMember<double> sum_squared_frame_durations;
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};
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// https://w3c.github.io/webrtc-stats/#pcstats-dict*
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class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCPeerConnectionStats(const std::string& id, int64_t timestamp_us);
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RTCPeerConnectionStats(std::string&& id, int64_t timestamp_us);
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RTCPeerConnectionStats(const RTCPeerConnectionStats& other);
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~RTCPeerConnectionStats() override;
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RTCStatsMember<uint32_t> data_channels_opened;
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RTCStatsMember<uint32_t> data_channels_closed;
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};
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// https://w3c.github.io/webrtc-stats/#streamstats-dict*
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// TODO(hbos): Tracking bug crbug.com/657854
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class RTC_EXPORT RTCRTPStreamStats : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCRTPStreamStats(const RTCRTPStreamStats& other);
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~RTCRTPStreamStats() override;
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RTCStatsMember<uint32_t> ssrc;
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RTCStatsMember<std::string> kind;
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// Obsolete: track_id
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RTCStatsMember<std::string> track_id;
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RTCStatsMember<std::string> transport_id;
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RTCStatsMember<std::string> codec_id;
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// Obsolete
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RTCStatsMember<std::string> media_type; // renamed to kind.
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protected:
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RTCRTPStreamStats(const std::string& id, int64_t timestamp_us);
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RTCRTPStreamStats(std::string&& id, int64_t timestamp_us);
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};
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// https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict*
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class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRTPStreamStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCReceivedRtpStreamStats(const RTCReceivedRtpStreamStats& other);
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~RTCReceivedRtpStreamStats() override;
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// TODO(hbos) The following fields need to be added and migrated
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// both from RTCInboundRtpStreamStats and RTCRemoteInboundRtpStreamStats:
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// packetsReceived, packetsRepaired, burstPacketsLost,
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// burstPacketDiscarded, burstLossCount, burstDiscardCount, burstLossRate,
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// burstDiscardRate, gapLossRate, gapDiscardRate, framesDropped,
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// partialFramesLost, fullFramesLost
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// crbug.com/webrtc/12532
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RTCStatsMember<double> jitter;
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RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
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RTCStatsMember<uint64_t> packets_discarded;
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protected:
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RTCReceivedRtpStreamStats(const std::string&& id, int64_t timestamp_us);
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RTCReceivedRtpStreamStats(std::string&& id, int64_t timestamp_us);
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};
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// https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict*
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class RTC_EXPORT RTCSentRtpStreamStats : public RTCRTPStreamStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCSentRtpStreamStats(const RTCSentRtpStreamStats& other);
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~RTCSentRtpStreamStats() override;
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RTCStatsMember<uint32_t> packets_sent;
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RTCStatsMember<uint64_t> bytes_sent;
|
|
|
|
protected:
|
|
RTCSentRtpStreamStats(const std::string&& id, int64_t timestamp_us);
|
|
RTCSentRtpStreamStats(std::string&& id, int64_t timestamp_us);
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
|
|
// TODO(hbos): Support the remote case |is_remote = true|.
|
|
// https://bugs.webrtc.org/7065
|
|
class RTC_EXPORT RTCInboundRTPStreamStats final
|
|
: public RTCReceivedRtpStreamStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCInboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
|
|
RTCInboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
|
|
RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
|
|
~RTCInboundRTPStreamStats() override;
|
|
|
|
RTCStatsMember<std::string> remote_id;
|
|
RTCStatsMember<uint32_t> packets_received;
|
|
RTCStatsMember<uint64_t> fec_packets_received;
|
|
RTCStatsMember<uint64_t> fec_packets_discarded;
|
|
RTCStatsMember<uint64_t> bytes_received;
|
|
RTCStatsMember<uint64_t> header_bytes_received;
|
|
RTCStatsMember<double> last_packet_received_timestamp;
|
|
RTCStatsMember<double> jitter_buffer_delay;
|
|
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
|
|
RTCStatsMember<uint64_t> total_samples_received;
|
|
RTCStatsMember<uint64_t> concealed_samples;
|
|
RTCStatsMember<uint64_t> silent_concealed_samples;
|
|
RTCStatsMember<uint64_t> concealment_events;
|
|
RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
|
|
RTCStatsMember<uint64_t> removed_samples_for_acceleration;
|
|
RTCStatsMember<double> audio_level;
|
|
RTCStatsMember<double> total_audio_energy;
|
|
RTCStatsMember<double> total_samples_duration;
|
|
RTCStatsMember<int32_t> frames_received;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<double> round_trip_time;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<uint32_t> packets_repaired;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<uint32_t> burst_packets_lost;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<uint32_t> burst_packets_discarded;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<uint32_t> burst_loss_count;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<uint32_t> burst_discard_count;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<double> burst_loss_rate;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<double> burst_discard_rate;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<double> gap_loss_rate;
|
|
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
|
|
RTCStatsMember<double> gap_discard_rate;
|
|
RTCStatsMember<uint32_t> frame_width;
|
|
RTCStatsMember<uint32_t> frame_height;
|
|
RTCStatsMember<uint32_t> frame_bit_depth;
|
|
RTCStatsMember<double> frames_per_second;
|
|
RTCStatsMember<uint32_t> frames_decoded;
|
|
RTCStatsMember<uint32_t> key_frames_decoded;
|
|
RTCStatsMember<uint32_t> frames_dropped;
|
|
RTCStatsMember<double> total_decode_time;
|
|
RTCStatsMember<double> total_inter_frame_delay;
|
|
RTCStatsMember<double> total_squared_inter_frame_delay;
|
|
// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
|
|
RTCStatsMember<std::string> content_type;
|
|
// TODO(asapersson): Currently only populated if audio/video sync is enabled.
|
|
RTCStatsMember<double> estimated_playout_timestamp;
|
|
// TODO(hbos): This is only implemented for video; implement it for audio as
|
|
// well.
|
|
RTCStatsMember<std::string> decoder_implementation;
|
|
// FIR and PLI counts are only defined for |media_type == "video"|.
|
|
RTCStatsMember<uint32_t> fir_count;
|
|
RTCStatsMember<uint32_t> pli_count;
|
|
RTCStatsMember<uint32_t> nack_count;
|
|
RTCStatsMember<uint64_t> qp_sum;
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
|
|
// TODO(hbos): Support the remote case |is_remote = true|.
|
|
// https://bugs.webrtc.org/7066
|
|
class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCOutboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
|
|
RTCOutboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
|
|
RTCOutboundRTPStreamStats(const RTCOutboundRTPStreamStats& other);
|
|
~RTCOutboundRTPStreamStats() override;
|
|
|
|
RTCStatsMember<std::string> media_source_id;
|
|
RTCStatsMember<std::string> remote_id;
|
|
RTCStatsMember<std::string> rid;
|
|
RTCStatsMember<uint32_t> packets_sent;
|
|
RTCStatsMember<uint64_t> retransmitted_packets_sent;
|
|
RTCStatsMember<uint64_t> bytes_sent;
|
|
RTCStatsMember<uint64_t> header_bytes_sent;
|
|
RTCStatsMember<uint64_t> retransmitted_bytes_sent;
|
|
// TODO(https://crbug.com/webrtc/13394): Also collect this metric for video.
|
|
RTCStatsMember<double> target_bitrate;
|
|
RTCStatsMember<uint32_t> frames_encoded;
|
|
RTCStatsMember<uint32_t> key_frames_encoded;
|
|
RTCStatsMember<double> total_encode_time;
|
|
RTCStatsMember<uint64_t> total_encoded_bytes_target;
|
|
RTCStatsMember<uint32_t> frame_width;
|
|
RTCStatsMember<uint32_t> frame_height;
|
|
RTCStatsMember<double> frames_per_second;
|
|
RTCStatsMember<uint32_t> frames_sent;
|
|
RTCStatsMember<uint32_t> huge_frames_sent;
|
|
// TODO(https://crbug.com/webrtc/10635): This is only implemented for video;
|
|
// implement it for audio as well.
|
|
RTCStatsMember<double> total_packet_send_delay;
|
|
// Enum type RTCQualityLimitationReason
|
|
RTCStatsMember<std::string> quality_limitation_reason;
|
|
RTCStatsMember<std::map<std::string, double>> quality_limitation_durations;
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
|
|
RTCStatsMember<uint32_t> quality_limitation_resolution_changes;
|
|
// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
|
|
RTCStatsMember<std::string> content_type;
|
|
// TODO(hbos): This is only implemented for video; implement it for audio as
|
|
// well.
|
|
RTCStatsMember<std::string> encoder_implementation;
|
|
// FIR and PLI counts are only defined for |media_type == "video"|.
|
|
RTCStatsMember<uint32_t> fir_count;
|
|
RTCStatsMember<uint32_t> pli_count;
|
|
RTCStatsMember<uint32_t> nack_count;
|
|
RTCStatsMember<uint64_t> qp_sum;
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
|
|
class RTC_EXPORT RTCRemoteInboundRtpStreamStats final
|
|
: public RTCReceivedRtpStreamStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCRemoteInboundRtpStreamStats(const std::string& id, int64_t timestamp_us);
|
|
RTCRemoteInboundRtpStreamStats(std::string&& id, int64_t timestamp_us);
|
|
RTCRemoteInboundRtpStreamStats(const RTCRemoteInboundRtpStreamStats& other);
|
|
~RTCRemoteInboundRtpStreamStats() override;
|
|
|
|
// TODO(hbos): The following RTCReceivedRtpStreamStats metrics should also be
|
|
// implemented: packetsReceived, packetsRepaired,
|
|
// burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount,
|
|
// burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate.
|
|
// RTCRemoteInboundRtpStreamStats
|
|
RTCStatsMember<std::string> local_id;
|
|
RTCStatsMember<double> round_trip_time;
|
|
RTCStatsMember<double> fraction_lost;
|
|
RTCStatsMember<double> total_round_trip_time;
|
|
RTCStatsMember<int32_t> round_trip_time_measurements;
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
|
|
class RTC_EXPORT RTCRemoteOutboundRtpStreamStats final
|
|
: public RTCSentRtpStreamStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCRemoteOutboundRtpStreamStats(const std::string& id, int64_t timestamp_us);
|
|
RTCRemoteOutboundRtpStreamStats(std::string&& id, int64_t timestamp_us);
|
|
RTCRemoteOutboundRtpStreamStats(const RTCRemoteOutboundRtpStreamStats& other);
|
|
~RTCRemoteOutboundRtpStreamStats() override;
|
|
|
|
RTCStatsMember<std::string> local_id;
|
|
RTCStatsMember<double> remote_timestamp;
|
|
RTCStatsMember<uint64_t> reports_sent;
|
|
RTCStatsMember<double> round_trip_time;
|
|
RTCStatsMember<uint64_t> round_trip_time_measurements;
|
|
RTCStatsMember<double> total_round_trip_time;
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
|
|
class RTC_EXPORT RTCMediaSourceStats : public RTCStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCMediaSourceStats(const RTCMediaSourceStats& other);
|
|
~RTCMediaSourceStats() override;
|
|
|
|
RTCStatsMember<std::string> track_identifier;
|
|
RTCStatsMember<std::string> kind;
|
|
|
|
protected:
|
|
RTCMediaSourceStats(const std::string& id, int64_t timestamp_us);
|
|
RTCMediaSourceStats(std::string&& id, int64_t timestamp_us);
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcaudiosourcestats
|
|
class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCAudioSourceStats(const std::string& id, int64_t timestamp_us);
|
|
RTCAudioSourceStats(std::string&& id, int64_t timestamp_us);
|
|
RTCAudioSourceStats(const RTCAudioSourceStats& other);
|
|
~RTCAudioSourceStats() override;
|
|
|
|
RTCStatsMember<double> audio_level;
|
|
RTCStatsMember<double> total_audio_energy;
|
|
RTCStatsMember<double> total_samples_duration;
|
|
RTCStatsMember<double> echo_return_loss;
|
|
RTCStatsMember<double> echo_return_loss_enhancement;
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats
|
|
class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCVideoSourceStats(const std::string& id, int64_t timestamp_us);
|
|
RTCVideoSourceStats(std::string&& id, int64_t timestamp_us);
|
|
RTCVideoSourceStats(const RTCVideoSourceStats& other);
|
|
~RTCVideoSourceStats() override;
|
|
|
|
RTCStatsMember<uint32_t> width;
|
|
RTCStatsMember<uint32_t> height;
|
|
RTCStatsMember<uint32_t> frames;
|
|
RTCStatsMember<double> frames_per_second;
|
|
};
|
|
|
|
// https://w3c.github.io/webrtc-stats/#transportstats-dict*
|
|
class RTC_EXPORT RTCTransportStats final : public RTCStats {
|
|
public:
|
|
WEBRTC_RTCSTATS_DECL();
|
|
|
|
RTCTransportStats(const std::string& id, int64_t timestamp_us);
|
|
RTCTransportStats(std::string&& id, int64_t timestamp_us);
|
|
RTCTransportStats(const RTCTransportStats& other);
|
|
~RTCTransportStats() override;
|
|
|
|
RTCStatsMember<uint64_t> bytes_sent;
|
|
RTCStatsMember<uint64_t> packets_sent;
|
|
RTCStatsMember<uint64_t> bytes_received;
|
|
RTCStatsMember<uint64_t> packets_received;
|
|
RTCStatsMember<std::string> rtcp_transport_stats_id;
|
|
// TODO(hbos): Support enum types? "RTCStatsMember<RTCDtlsTransportState>"?
|
|
RTCStatsMember<std::string> dtls_state;
|
|
RTCStatsMember<std::string> selected_candidate_pair_id;
|
|
RTCStatsMember<std::string> local_certificate_id;
|
|
RTCStatsMember<std::string> remote_certificate_id;
|
|
RTCStatsMember<std::string> tls_version;
|
|
RTCStatsMember<std::string> dtls_cipher;
|
|
RTCStatsMember<std::string> srtp_cipher;
|
|
RTCStatsMember<uint32_t> selected_candidate_pair_changes;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_STATS_RTCSTATS_OBJECTS_H_
|