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webrtc/api/stats/rtcstats_objects.h
Henrik Boström 4e231eedbd Delete deprecated 'track' and 'stream' metrics from WebRTC. 
Track stats are roughly equal in size as the RTP stream stats which
are the largest objects making up the majority of the RTCStatsReport
size and scales with meeting size. Deleting track/stream reduces the
size in approximately half which should reduce performance overhead
and unblock code simplifications.

Blocked on:
- https://chromium-review.googlesource.com/c/chromium/src/+/4517530

# Relevant bots already passed
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Bug: webrtc:14175, webrtc:14419
Change-Id: Ib7bdb84c10459b42b829228d11876498e5227312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40129}
2023-05-24 12:26:56 +00:00

613 lines
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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef API_STATS_RTCSTATS_OBJECTS_H_
#define API_STATS_RTCSTATS_OBJECTS_H_
#include <stdint.h>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "api/stats/rtc_stats.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// https://w3c.github.io/webrtc-pc/#idl-def-rtcdatachannelstate
struct RTCDataChannelState {
static const char* const kConnecting;
static const char* const kOpen;
static const char* const kClosing;
static const char* const kClosed;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcstatsicecandidatepairstate
struct RTCStatsIceCandidatePairState {
static const char* const kFrozen;
static const char* const kWaiting;
static const char* const kInProgress;
static const char* const kFailed;
static const char* const kSucceeded;
};
// https://w3c.github.io/webrtc-pc/#rtcicecandidatetype-enum
struct RTCIceCandidateType {
static const char* const kHost;
static const char* const kSrflx;
static const char* const kPrflx;
static const char* const kRelay;
};
// https://w3c.github.io/webrtc-pc/#idl-def-rtcdtlstransportstate
struct RTCDtlsTransportState {
static const char* const kNew;
static const char* const kConnecting;
static const char* const kConnected;
static const char* const kClosed;
static const char* const kFailed;
};
// `RTCRtpStreamStats::kind` is not an enum in the spec but the only valid
// values are "audio" and "video" as it comes from `MediaStreamTrack::kind`.
// https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-kind
// https://w3c.github.io/mediacapture-main/#dom-mediadeviceinfo-kind
struct RTCMediaStreamTrackKind {
static const char* const kAudio;
static const char* const kVideo;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcnetworktype
struct RTCNetworkType {
static const char* const kBluetooth;
static const char* const kCellular;
static const char* const kEthernet;
static const char* const kWifi;
static const char* const kWimax;
static const char* const kVpn;
static const char* const kUnknown;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcqualitylimitationreason
struct RTCQualityLimitationReason {
static const char* const kNone;
static const char* const kCpu;
static const char* const kBandwidth;
static const char* const kOther;
};
// https://webrtc.org/experiments/rtp-hdrext/video-content-type/
struct RTCContentType {
static const char* const kUnspecified;
static const char* const kScreenshare;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcdtlsrole
struct RTCDtlsRole {
static const char* const kUnknown;
static const char* const kClient;
static const char* const kServer;
};
// https://www.w3.org/TR/webrtc/#rtcicerole
struct RTCIceRole {
static const char* const kUnknown;
static const char* const kControlled;
static const char* const kControlling;
};
// https://www.w3.org/TR/webrtc/#dom-rtcicetransportstate
struct RTCIceTransportState {
static const char* const kNew;
static const char* const kChecking;
static const char* const kConnected;
static const char* const kCompleted;
static const char* const kDisconnected;
static const char* const kFailed;
static const char* const kClosed;
};
// https://w3c.github.io/webrtc-stats/#certificatestats-dict*
class RTC_EXPORT RTCCertificateStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCCertificateStats(std::string id, Timestamp timestamp);
RTCCertificateStats(const RTCCertificateStats& other);
~RTCCertificateStats() override;
RTCStatsMember<std::string> fingerprint;
RTCStatsMember<std::string> fingerprint_algorithm;
RTCStatsMember<std::string> base64_certificate;
RTCStatsMember<std::string> issuer_certificate_id;
};
// Non standard extension mapping to rtc::AdapterType
struct RTCNetworkAdapterType {
static constexpr char kUnknown[] = "unknown";
static constexpr char kEthernet[] = "ethernet";
static constexpr char kWifi[] = "wifi";
static constexpr char kCellular[] = "cellular";
static constexpr char kLoopback[] = "loopback";
static constexpr char kAny[] = "any";
static constexpr char kCellular2g[] = "cellular2g";
static constexpr char kCellular3g[] = "cellular3g";
static constexpr char kCellular4g[] = "cellular4g";
static constexpr char kCellular5g[] = "cellular5g";
};
// https://w3c.github.io/webrtc-stats/#codec-dict*
class RTC_EXPORT RTCCodecStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCCodecStats(std::string id, Timestamp timestamp);
RTCCodecStats(const RTCCodecStats& other);
~RTCCodecStats() override;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<uint32_t> payload_type;
RTCStatsMember<std::string> mime_type;
RTCStatsMember<uint32_t> clock_rate;
RTCStatsMember<uint32_t> channels;
RTCStatsMember<std::string> sdp_fmtp_line;
};
// https://w3c.github.io/webrtc-stats/#dcstats-dict*
class RTC_EXPORT RTCDataChannelStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCDataChannelStats(std::string id, Timestamp timestamp);
RTCDataChannelStats(const RTCDataChannelStats& other);
~RTCDataChannelStats() override;
RTCStatsMember<std::string> label;
RTCStatsMember<std::string> protocol;
RTCStatsMember<int32_t> data_channel_identifier;
// Enum type RTCDataChannelState.
RTCStatsMember<std::string> state;
RTCStatsMember<uint32_t> messages_sent;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint32_t> messages_received;
RTCStatsMember<uint64_t> bytes_received;
};
// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCIceCandidatePairStats(std::string id, Timestamp timestamp);
RTCIceCandidatePairStats(const RTCIceCandidatePairStats& other);
~RTCIceCandidatePairStats() override;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> local_candidate_id;
RTCStatsMember<std::string> remote_candidate_id;
// Enum type RTCStatsIceCandidatePairState.
RTCStatsMember<std::string> state;
// Obsolete: priority
RTCStatsMember<uint64_t> priority;
RTCStatsMember<bool> nominated;
// `writable` does not exist in the spec and old comments suggest it used to
// exist but was incorrectly implemented.
// TODO(https://crbug.com/webrtc/14171): Standardize and/or modify
// implementation.
RTCStatsMember<bool> writable;
RTCStatsMember<uint64_t> packets_sent;
RTCStatsMember<uint64_t> packets_received;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<double> total_round_trip_time;
RTCStatsMember<double> current_round_trip_time;
RTCStatsMember<double> available_outgoing_bitrate;
RTCStatsMember<double> available_incoming_bitrate;
RTCStatsMember<uint64_t> requests_received;
RTCStatsMember<uint64_t> requests_sent;
RTCStatsMember<uint64_t> responses_received;
RTCStatsMember<uint64_t> responses_sent;
RTCStatsMember<uint64_t> consent_requests_sent;
RTCStatsMember<uint64_t> packets_discarded_on_send;
RTCStatsMember<uint64_t> bytes_discarded_on_send;
RTCStatsMember<double> last_packet_received_timestamp;
RTCStatsMember<double> last_packet_sent_timestamp;
};
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCIceCandidateStats(const RTCIceCandidateStats& other);
~RTCIceCandidateStats() override;
RTCStatsMember<std::string> transport_id;
// Obsolete: is_remote
RTCStatsMember<bool> is_remote;
RTCStatsMember<std::string> network_type;
RTCStatsMember<std::string> ip;
RTCStatsMember<std::string> address;
RTCStatsMember<int32_t> port;
RTCStatsMember<std::string> protocol;
RTCStatsMember<std::string> relay_protocol;
// Enum type RTCIceCandidateType.
RTCStatsMember<std::string> candidate_type;
RTCStatsMember<int32_t> priority;
RTCStatsMember<std::string> url;
RTCStatsMember<std::string> foundation;
RTCStatsMember<std::string> related_address;
RTCStatsMember<int32_t> related_port;
RTCStatsMember<std::string> username_fragment;
// Enum type RTCIceTcpCandidateType.
RTCStatsMember<std::string> tcp_type;
RTCNonStandardStatsMember<bool> vpn;
RTCNonStandardStatsMember<std::string> network_adapter_type;
protected:
RTCIceCandidateStats(std::string id, Timestamp timestamp, bool is_remote);
};
// In the spec both local and remote varieties are of type RTCIceCandidateStats.
// But here we define them as subclasses of `RTCIceCandidateStats` because the
// `kType` need to be different ("RTCStatsType type") in the local/remote case.
// https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
// This forces us to have to override copy() and type().
class RTC_EXPORT RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
public:
static const char kType[];
RTCLocalIceCandidateStats(std::string id, Timestamp timestamp);
std::unique_ptr<RTCStats> copy() const override;
const char* type() const override;
};
class RTC_EXPORT RTCRemoteIceCandidateStats final
: public RTCIceCandidateStats {
public:
static const char kType[];
RTCRemoteIceCandidateStats(std::string id, Timestamp timestamp);
std::unique_ptr<RTCStats> copy() const override;
const char* type() const override;
};
// https://w3c.github.io/webrtc-stats/#pcstats-dict*
class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCPeerConnectionStats(std::string id, Timestamp timestamp);
RTCPeerConnectionStats(const RTCPeerConnectionStats& other);
~RTCPeerConnectionStats() override;
RTCStatsMember<uint32_t> data_channels_opened;
RTCStatsMember<uint32_t> data_channels_closed;
};
// https://w3c.github.io/webrtc-stats/#streamstats-dict*
class RTC_EXPORT RTCRtpStreamStats : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCRtpStreamStats(const RTCRtpStreamStats& other);
~RTCRtpStreamStats() override;
RTCStatsMember<uint32_t> ssrc;
RTCStatsMember<std::string> kind;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> codec_id;
protected:
RTCRtpStreamStats(std::string id, Timestamp timestamp);
};
// https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict*
class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCReceivedRtpStreamStats(const RTCReceivedRtpStreamStats& other);
~RTCReceivedRtpStreamStats() override;
RTCStatsMember<double> jitter;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
protected:
RTCReceivedRtpStreamStats(std::string id, Timestamp timestamp);
};
// https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict*
class RTC_EXPORT RTCSentRtpStreamStats : public RTCRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCSentRtpStreamStats(const RTCSentRtpStreamStats& other);
~RTCSentRtpStreamStats() override;
RTCStatsMember<uint64_t> packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
protected:
RTCSentRtpStreamStats(std::string id, Timestamp timestamp);
};
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
class RTC_EXPORT RTCInboundRtpStreamStats final
: public RTCReceivedRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCInboundRtpStreamStats(std::string id, Timestamp timestamp);
RTCInboundRtpStreamStats(const RTCInboundRtpStreamStats& other);
~RTCInboundRtpStreamStats() override;
RTCStatsMember<std::string> playout_id;
RTCStatsMember<std::string> track_identifier;
RTCStatsMember<std::string> mid;
RTCStatsMember<std::string> remote_id;
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> packets_discarded;
RTCStatsMember<uint64_t> fec_packets_received;
RTCStatsMember<uint64_t> fec_packets_discarded;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<uint64_t> header_bytes_received;
// Inbound RTX stats. Only defined when RTX is used and it is therefore
// possible to distinguish retransmissions.
RTCStatsMember<uint64_t> retransmitted_packets_received;
RTCStatsMember<uint64_t> retransmitted_bytes_received;
RTCStatsMember<double> last_packet_received_timestamp;
RTCStatsMember<double> jitter_buffer_delay;
RTCStatsMember<double> jitter_buffer_target_delay;
RTCStatsMember<double> jitter_buffer_minimum_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;
// Stats below are only implemented or defined for video.
RTCStatsMember<uint32_t> frames_received;
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;
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_processing_delay;
RTCStatsMember<double> total_assembly_time;
RTCStatsMember<uint32_t> frames_assembled_from_multiple_packets;
RTCStatsMember<double> total_inter_frame_delay;
RTCStatsMember<double> total_squared_inter_frame_delay;
RTCStatsMember<uint32_t> pause_count;
RTCStatsMember<double> total_pauses_duration;
RTCStatsMember<uint32_t> freeze_count;
RTCStatsMember<double> total_freezes_duration;
// https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
RTCStatsMember<std::string> content_type;
// Only populated if audio/video sync is enabled.
// TODO(https://crbug.com/webrtc/14177): Expose even if A/V sync is off?
RTCStatsMember<double> estimated_playout_timestamp;
// Only defined for video.
RTCRestrictedStatsMember<std::string,
StatExposureCriteria::kHardwareCapability>
decoder_implementation;
// FIR and PLI counts are only defined for |kind == "video"|.
RTCStatsMember<uint32_t> fir_count;
RTCStatsMember<uint32_t> pli_count;
RTCStatsMember<uint32_t> nack_count;
RTCStatsMember<uint64_t> qp_sum;
// This is a remnant of the legacy getStats() API. When the "video-timing"
// header extension is used,
// https://webrtc.github.io/webrtc-org/experiments/rtp-hdrext/video-timing/,
// `googTimingFrameInfo` is exposed with the value of
// TimingFrameInfo::ToString().
// TODO(https://crbug.com/webrtc/14586): Unship or standardize this metric.
RTCStatsMember<std::string> goog_timing_frame_info;
RTCRestrictedStatsMember<bool, StatExposureCriteria::kHardwareCapability>
power_efficient_decoder;
// Non-standard audio metrics.
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// The former googMinPlayoutDelayMs (in seconds).
RTCNonStandardStatsMember<double> min_playout_delay;
};
// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
class RTC_EXPORT RTCOutboundRtpStreamStats final
: public RTCSentRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCOutboundRtpStreamStats(std::string id, Timestamp timestamp);
RTCOutboundRtpStreamStats(const RTCOutboundRtpStreamStats& other);
~RTCOutboundRtpStreamStats() override;
RTCStatsMember<std::string> media_source_id;
RTCStatsMember<std::string> remote_id;
RTCStatsMember<std::string> mid;
RTCStatsMember<std::string> rid;
RTCStatsMember<uint64_t> retransmitted_packets_sent;
RTCStatsMember<uint64_t> header_bytes_sent;
RTCStatsMember<uint64_t> retransmitted_bytes_sent;
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;
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://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
RTCStatsMember<std::string> content_type;
// Only implemented for video.
// TODO(https://crbug.com/webrtc/14178): Implement for audio as well.
RTCRestrictedStatsMember<std::string,
StatExposureCriteria::kHardwareCapability>
encoder_implementation;
// FIR and PLI counts are only defined for |kind == "video"|.
RTCStatsMember<uint32_t> fir_count;
RTCStatsMember<uint32_t> pli_count;
RTCStatsMember<uint32_t> nack_count;
RTCStatsMember<uint64_t> qp_sum;
RTCStatsMember<bool> active;
RTCRestrictedStatsMember<bool, StatExposureCriteria::kHardwareCapability>
power_efficient_encoder;
RTCStatsMember<std::string> scalability_mode;
};
// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
class RTC_EXPORT RTCRemoteInboundRtpStreamStats final
: public RTCReceivedRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCRemoteInboundRtpStreamStats(std::string id, Timestamp timestamp);
RTCRemoteInboundRtpStreamStats(const RTCRemoteInboundRtpStreamStats& other);
~RTCRemoteInboundRtpStreamStats() override;
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(std::string id, Timestamp timestamp);
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(std::string id, Timestamp timestamp);
};
// https://w3c.github.io/webrtc-stats/#dom-rtcaudiosourcestats
class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCAudioSourceStats(std::string id, Timestamp timestamp);
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(std::string id, Timestamp timestamp);
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(std::string id, Timestamp timestamp);
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;
// Enum type 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> dtls_role;
RTCStatsMember<std::string> srtp_cipher;
RTCStatsMember<uint32_t> selected_candidate_pair_changes;
RTCStatsMember<std::string> ice_role;
RTCStatsMember<std::string> ice_local_username_fragment;
RTCStatsMember<std::string> ice_state;
};
// https://w3c.github.io/webrtc-stats/#playoutstats-dict*
class RTC_EXPORT RTCAudioPlayoutStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCAudioPlayoutStats(const std::string& id, Timestamp timestamp);
RTCAudioPlayoutStats(const RTCAudioPlayoutStats& other);
~RTCAudioPlayoutStats() override;
RTCStatsMember<std::string> kind;
RTCStatsMember<double> synthesized_samples_duration;
RTCStatsMember<uint64_t> synthesized_samples_events;
RTCStatsMember<double> total_samples_duration;
RTCStatsMember<double> total_playout_delay;
RTCStatsMember<uint64_t> total_samples_count;
};
} // namespace webrtc
#endif // API_STATS_RTCSTATS_OBJECTS_H_
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