2fb83072db
They may be non-standard, but they shouldn't be on a stats dictionary that is deprecated (track is going away soon-ish). By moving them to inbound-rtp they can continue to exist beyond track deprecation and live in the right place in case we decide to standardize them later. To help downstream projects transitions, the metrics are temporarily available in both old and new locations. Delete of old location will happen in a follow-up CL. TODOs added. Bug: webrtc:14524 Change-Id: I2008060fa4ba76cde859d9144d2bb9648c7ff9af Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278200 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38315}
683 lines
26 KiB
C++
683 lines
26 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/#dom-rtcdtlsrole
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struct RTCDtlsRole {
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static const char* const kUnknown;
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static const char* const kClient;
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static const char* const kServer;
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};
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// https://www.w3.org/TR/webrtc/#rtcicerole
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struct RTCIceRole {
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static const char* const kUnknown;
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static const char* const kControlled;
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static const char* const kControlling;
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};
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// https://www.w3.org/TR/webrtc/#dom-rtcicetransportstate
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struct RTCIceTransportState {
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static const char* const kNew;
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static const char* const kChecking;
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static const char* const kConnected;
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static const char* const kCompleted;
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static const char* const kDisconnected;
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static const char* const kFailed;
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static const char* const kClosed;
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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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// Non standard extension mapping to rtc::AdapterType
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struct RTCNetworkAdapterType {
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static constexpr char kUnknown[] = "unknown";
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static constexpr char kEthernet[] = "ethernet";
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static constexpr char kWifi[] = "wifi";
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static constexpr char kCellular[] = "cellular";
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static constexpr char kLoopback[] = "loopback";
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static constexpr char kAny[] = "any";
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static constexpr char kCellular2g[] = "cellular2g";
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static constexpr char kCellular3g[] = "cellular3g";
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static constexpr char kCellular4g[] = "cellular4g";
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static constexpr char kCellular5g[] = "cellular5g";
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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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// Enum type 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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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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// Enum type 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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// `writable` does not exist in the spec and old comments suggest it used to
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// exist but was incorrectly implemented.
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// TODO(https://crbug.com/webrtc/14171): Standardize and/or modify
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// implementation.
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RTCStatsMember<bool> writable;
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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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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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RTCStatsMember<uint64_t> consent_requests_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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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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// Enum type 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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RTCStatsMember<std::string> foundation;
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RTCStatsMember<std::string> related_address;
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RTCStatsMember<int32_t> related_port;
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RTCStatsMember<std::string> username_fragment;
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// Enum type RTCIceTcpCandidateType.
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RTCStatsMember<std::string> tcp_type;
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RTCNonStandardStatsMember<bool> vpn;
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RTCNonStandardStatsMember<std::string> network_adapter_type;
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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/#dom-rtcmediastreamstats
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// TODO(https://crbug.com/webrtc/14172): Deprecate and remove.
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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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// TODO(https://crbug.com/webrtc/14175): Deprecate and remove in favor of
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// RTCMediaSourceStats/RTCOutboundRtpStreamStats and RTCInboundRtpStreamStats.
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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(https://crbug.com/webrtc/14173): Remove this obsolete metric.
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RTCStatsMember<bool> detached;
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// Enum type RTCMediaStreamTrackKind.
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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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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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// 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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// TODO(crbug.com/webrtc/14524): These metrics have been moved, delete them.
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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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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://w3c.github.io/webrtc-provisional-stats/#dom-rtcvideoreceiverstats
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RTCNonStandardStatsMember<double> total_frames_duration;
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RTCNonStandardStatsMember<double> sum_squared_frame_durations;
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// TODO(crbug.com/webrtc/14521): These metrics have been moved, delete them.
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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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};
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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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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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RTCStatsMember<double> jitter;
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RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
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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;
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protected:
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RTCSentRtpStreamStats(const std::string&& id, int64_t timestamp_us);
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RTCSentRtpStreamStats(std::string&& id, int64_t timestamp_us);
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};
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// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
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class RTC_EXPORT RTCInboundRTPStreamStats final
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: public RTCReceivedRtpStreamStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCInboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
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RTCInboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
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RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
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~RTCInboundRTPStreamStats() override;
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// TODO(https://crbug.com/webrtc/14174): Implement trackIdentifier and kind.
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RTCStatsMember<std::string> track_identifier;
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RTCStatsMember<std::string> mid;
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RTCStatsMember<std::string> remote_id;
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RTCStatsMember<uint32_t> packets_received;
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RTCStatsMember<uint64_t> packets_discarded;
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RTCStatsMember<uint64_t> fec_packets_received;
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RTCStatsMember<uint64_t> fec_packets_discarded;
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RTCStatsMember<uint64_t> bytes_received;
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RTCStatsMember<uint64_t> header_bytes_received;
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RTCStatsMember<double> last_packet_received_timestamp;
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RTCStatsMember<double> jitter_buffer_delay;
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RTCStatsMember<double> jitter_buffer_target_delay;
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RTCStatsMember<double> jitter_buffer_minimum_delay;
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RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
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RTCStatsMember<uint64_t> total_samples_received;
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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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RTCStatsMember<double> audio_level;
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RTCStatsMember<double> total_audio_energy;
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RTCStatsMember<double> total_samples_duration;
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// Stats below are only implemented or defined for video.
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RTCStatsMember<int32_t> frames_received;
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RTCStatsMember<uint32_t> frame_width;
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RTCStatsMember<uint32_t> frame_height;
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RTCStatsMember<double> frames_per_second;
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RTCStatsMember<uint32_t> frames_decoded;
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RTCStatsMember<uint32_t> key_frames_decoded;
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RTCStatsMember<uint32_t> frames_dropped;
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RTCStatsMember<double> total_decode_time;
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RTCStatsMember<double> total_processing_delay;
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// TODO(https://crbug.com/webrtc/13986): standardize
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RTCNonStandardStatsMember<double> total_assembly_time;
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RTCNonStandardStatsMember<uint32_t> frames_assembled_from_multiple_packets;
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RTCStatsMember<double> total_inter_frame_delay;
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RTCStatsMember<double> total_squared_inter_frame_delay;
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RTCStatsMember<uint32_t> pause_count;
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RTCStatsMember<double> total_pauses_duration;
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RTCStatsMember<uint32_t> freeze_count;
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RTCStatsMember<double> total_freezes_duration;
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// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
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RTCStatsMember<std::string> content_type;
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// Only populated if audio/video sync is enabled.
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// TODO(https://crbug.com/webrtc/14177): Expose even if A/V sync is off?
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RTCStatsMember<double> estimated_playout_timestamp;
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// Only implemented for video.
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// TODO(https://crbug.com/webrtc/14178): Also implement for audio.
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RTCStatsMember<std::string> decoder_implementation;
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// FIR and PLI counts are only defined for |kind == "video"|.
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RTCStatsMember<uint32_t> fir_count;
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RTCStatsMember<uint32_t> pli_count;
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RTCStatsMember<uint32_t> nack_count;
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RTCStatsMember<uint64_t> qp_sum;
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// Non-standard audio metrics.
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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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RTCNonStandardStatsMember<uint32_t> interruption_count;
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RTCNonStandardStatsMember<double> total_interruption_duration;
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// The former googMinPlayoutDelayMs (in seconds).
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RTCNonStandardStatsMember<double> min_playout_delay;
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};
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// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
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class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCOutboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
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RTCOutboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
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RTCOutboundRTPStreamStats(const RTCOutboundRTPStreamStats& other);
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~RTCOutboundRTPStreamStats() override;
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RTCStatsMember<std::string> media_source_id;
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RTCStatsMember<std::string> remote_id;
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RTCStatsMember<std::string> mid;
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RTCStatsMember<std::string> rid;
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RTCStatsMember<uint32_t> packets_sent;
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RTCStatsMember<uint64_t> retransmitted_packets_sent;
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RTCStatsMember<uint64_t> bytes_sent;
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RTCStatsMember<uint64_t> header_bytes_sent;
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RTCStatsMember<uint64_t> retransmitted_bytes_sent;
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RTCStatsMember<double> target_bitrate;
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RTCStatsMember<uint32_t> frames_encoded;
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RTCStatsMember<uint32_t> key_frames_encoded;
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RTCStatsMember<double> total_encode_time;
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RTCStatsMember<uint64_t> total_encoded_bytes_target;
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RTCStatsMember<uint32_t> frame_width;
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RTCStatsMember<uint32_t> frame_height;
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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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// TODO(https://crbug.com/webrtc/10635): This is only implemented for video;
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// implement it for audio as well.
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RTCStatsMember<double> total_packet_send_delay;
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// Enum type RTCQualityLimitationReason
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RTCStatsMember<std::string> quality_limitation_reason;
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RTCStatsMember<std::map<std::string, double>> quality_limitation_durations;
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// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
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RTCStatsMember<uint32_t> quality_limitation_resolution_changes;
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// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
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RTCStatsMember<std::string> content_type;
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// Only implemented for video.
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// TODO(https://crbug.com/webrtc/14178): Implement for audio as well.
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RTCStatsMember<std::string> encoder_implementation;
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// FIR and PLI counts are only defined for |kind == "video"|.
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RTCStatsMember<uint32_t> fir_count;
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RTCStatsMember<uint32_t> pli_count;
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RTCStatsMember<uint32_t> nack_count;
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RTCStatsMember<uint64_t> qp_sum;
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RTCStatsMember<bool> active;
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};
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// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
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class RTC_EXPORT RTCRemoteInboundRtpStreamStats final
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: public RTCReceivedRtpStreamStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCRemoteInboundRtpStreamStats(const std::string& id, int64_t timestamp_us);
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RTCRemoteInboundRtpStreamStats(std::string&& id, int64_t timestamp_us);
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RTCRemoteInboundRtpStreamStats(const RTCRemoteInboundRtpStreamStats& other);
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~RTCRemoteInboundRtpStreamStats() override;
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RTCStatsMember<std::string> local_id;
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RTCStatsMember<double> round_trip_time;
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RTCStatsMember<double> fraction_lost;
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RTCStatsMember<double> total_round_trip_time;
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RTCStatsMember<int32_t> round_trip_time_measurements;
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};
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// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
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class RTC_EXPORT RTCRemoteOutboundRtpStreamStats final
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: public RTCSentRtpStreamStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCRemoteOutboundRtpStreamStats(const std::string& id, int64_t timestamp_us);
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RTCRemoteOutboundRtpStreamStats(std::string&& id, int64_t timestamp_us);
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RTCRemoteOutboundRtpStreamStats(const RTCRemoteOutboundRtpStreamStats& other);
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~RTCRemoteOutboundRtpStreamStats() override;
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RTCStatsMember<std::string> local_id;
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RTCStatsMember<double> remote_timestamp;
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RTCStatsMember<uint64_t> reports_sent;
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RTCStatsMember<double> round_trip_time;
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RTCStatsMember<uint64_t> round_trip_time_measurements;
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RTCStatsMember<double> total_round_trip_time;
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};
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// https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats
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class RTC_EXPORT RTCMediaSourceStats : public RTCStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCMediaSourceStats(const RTCMediaSourceStats& other);
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~RTCMediaSourceStats() override;
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RTCStatsMember<std::string> track_identifier;
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RTCStatsMember<std::string> kind;
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protected:
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RTCMediaSourceStats(const std::string& id, int64_t timestamp_us);
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RTCMediaSourceStats(std::string&& id, int64_t timestamp_us);
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};
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// https://w3c.github.io/webrtc-stats/#dom-rtcaudiosourcestats
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class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats {
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public:
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WEBRTC_RTCSTATS_DECL();
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RTCAudioSourceStats(const std::string& id, int64_t timestamp_us);
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RTCAudioSourceStats(std::string&& id, int64_t timestamp_us);
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RTCAudioSourceStats(const RTCAudioSourceStats& other);
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~RTCAudioSourceStats() override;
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RTCStatsMember<double> audio_level;
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RTCStatsMember<double> total_audio_energy;
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RTCStatsMember<double> total_samples_duration;
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RTCStatsMember<double> echo_return_loss;
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RTCStatsMember<double> echo_return_loss_enhancement;
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};
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// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats
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class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats {
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public:
|
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WEBRTC_RTCSTATS_DECL();
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|
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RTCVideoSourceStats(const std::string& id, int64_t timestamp_us);
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RTCVideoSourceStats(std::string&& id, int64_t timestamp_us);
|
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RTCVideoSourceStats(const RTCVideoSourceStats& other);
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~RTCVideoSourceStats() override;
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RTCStatsMember<uint32_t> width;
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RTCStatsMember<uint32_t> height;
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RTCStatsMember<uint32_t> frames;
|
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RTCStatsMember<double> frames_per_second;
|
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};
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|
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// https://w3c.github.io/webrtc-stats/#transportstats-dict*
|
|
class RTC_EXPORT RTCTransportStats final : public RTCStats {
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|
public:
|
|
WEBRTC_RTCSTATS_DECL();
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|
|
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RTCTransportStats(const std::string& id, int64_t timestamp_us);
|
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RTCTransportStats(std::string&& id, int64_t timestamp_us);
|
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RTCTransportStats(const RTCTransportStats& other);
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~RTCTransportStats() override;
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|
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RTCStatsMember<uint64_t> bytes_sent;
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RTCStatsMember<uint64_t> packets_sent;
|
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RTCStatsMember<uint64_t> bytes_received;
|
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RTCStatsMember<uint64_t> packets_received;
|
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RTCStatsMember<std::string> rtcp_transport_stats_id;
|
|
// Enum type RTCDtlsTransportState.
|
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RTCStatsMember<std::string> dtls_state;
|
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RTCStatsMember<std::string> selected_candidate_pair_id;
|
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RTCStatsMember<std::string> local_certificate_id;
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RTCStatsMember<std::string> remote_certificate_id;
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RTCStatsMember<std::string> tls_version;
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RTCStatsMember<std::string> dtls_cipher;
|
|
RTCStatsMember<std::string> dtls_role;
|
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RTCStatsMember<std::string> srtp_cipher;
|
|
RTCStatsMember<uint32_t> selected_candidate_pair_changes;
|
|
RTCStatsMember<std::string> ice_role;
|
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RTCStatsMember<std::string> ice_local_username_fragment;
|
|
RTCStatsMember<std::string> ice_state;
|
|
};
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} // namespace webrtc
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#endif // API_STATS_RTCSTATS_OBJECTS_H_
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