blob: 7742538ca728940cd926b319072d3f9f15624a4f [file] [log] [blame]
/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#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;
};
// `RTCMediaStreamTrackStats::kind` is not an enum in the spec but the only
// valid values are "audio" and "video".
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-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/#certificatestats-dict*
class RTC_EXPORT RTCCertificateStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCCertificateStats(const std::string& id, int64_t timestamp_us);
RTCCertificateStats(std::string&& id, int64_t timestamp_us);
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;
};
// https://w3c.github.io/webrtc-stats/#codec-dict*
class RTC_EXPORT RTCCodecStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCCodecStats(const std::string& id, int64_t timestamp_us);
RTCCodecStats(std::string&& id, int64_t timestamp_us);
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(const std::string& id, int64_t timestamp_us);
RTCDataChannelStats(std::string&& id, int64_t timestamp_us);
RTCDataChannelStats(const RTCDataChannelStats& other);
~RTCDataChannelStats() override;
RTCStatsMember<std::string> label;
RTCStatsMember<std::string> protocol;
RTCStatsMember<int32_t> data_channel_identifier;
// TODO(hbos): Support enum types? "RTCStatsMember<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*
// TODO(hbos): Tracking bug https://bugs.webrtc.org/7062
class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCIceCandidatePairStats(const std::string& id, int64_t timestamp_us);
RTCIceCandidatePairStats(std::string&& id, int64_t timestamp_us);
RTCIceCandidatePairStats(const RTCIceCandidatePairStats& other);
~RTCIceCandidatePairStats() override;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> local_candidate_id;
RTCStatsMember<std::string> remote_candidate_id;
// TODO(hbos): Support enum types?
// "RTCStatsMember<RTCStatsIceCandidatePairState>"?
RTCStatsMember<std::string> state;
// Obsolete: priority
RTCStatsMember<uint64_t> priority;
RTCStatsMember<bool> nominated;
// TODO(hbos): Collect this the way the spec describes it. We have a value for
// it but it is not spec-compliant. https://bugs.webrtc.org/7062
RTCStatsMember<bool> writable;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<bool> readable;
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;
// TODO(hbos): Populate this value. It is wired up and collected the same way
// "VideoBwe.googAvailableReceiveBandwidth" is, but that value is always
// undefined. https://bugs.webrtc.org/7062
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;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> retransmissions_received;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> retransmissions_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> consent_requests_received;
RTCStatsMember<uint64_t> consent_requests_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> consent_responses_received;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> consent_responses_sent;
RTCStatsMember<uint64_t> packets_discarded_on_send;
RTCStatsMember<uint64_t> bytes_discarded_on_send;
};
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
// TODO(hbos): `RTCStatsCollector` only collects candidates that are part of
// ice candidate pairs, but there could be candidates not paired with anything.
// crbug.com/632723
// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
// them in the new PeerConnection::GetStats.
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;
// TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
RTCStatsMember<std::string> candidate_type;
RTCStatsMember<int32_t> priority;
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/632723
RTCStatsMember<std::string> url;
protected:
RTCIceCandidateStats(const std::string& id,
int64_t timestamp_us,
bool is_remote);
RTCIceCandidateStats(std::string&& id, int64_t timestamp_us, 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(const std::string& id, int64_t timestamp_us);
RTCLocalIceCandidateStats(std::string&& id, int64_t timestamp_us);
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(const std::string& id, int64_t timestamp_us);
RTCRemoteIceCandidateStats(std::string&& id, int64_t timestamp_us);
std::unique_ptr<RTCStats> copy() const override;
const char* type() const override;
};
// https://w3c.github.io/webrtc-stats/#msstats-dict*
// TODO(hbos): Tracking bug crbug.com/660827
class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCMediaStreamStats(const std::string& id, int64_t timestamp_us);
RTCMediaStreamStats(std::string&& id, int64_t timestamp_us);
RTCMediaStreamStats(const RTCMediaStreamStats& other);
~RTCMediaStreamStats() override;
RTCStatsMember<std::string> stream_identifier;
RTCStatsMember<std::vector<std::string>> track_ids;
};
// https://w3c.github.io/webrtc-stats/#mststats-dict*
// TODO(hbos): Tracking bug crbug.com/659137
class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCMediaStreamTrackStats(const std::string& id,
int64_t timestamp_us,
const char* kind);
RTCMediaStreamTrackStats(std::string&& id,
int64_t timestamp_us,
const char* kind);
RTCMediaStreamTrackStats(const RTCMediaStreamTrackStats& other);
~RTCMediaStreamTrackStats() override;
RTCStatsMember<std::string> track_identifier;
RTCStatsMember<std::string> media_source_id;
RTCStatsMember<bool> remote_source;
RTCStatsMember<bool> ended;
// TODO(hbos): `RTCStatsCollector` does not return stats for detached tracks.
// crbug.com/659137
RTCStatsMember<bool> detached;
// See `RTCMediaStreamTrackKind` for valid values.
RTCStatsMember<std::string> kind;
RTCStatsMember<double> jitter_buffer_delay;
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
// Video-only members
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<double> frames_per_second;
RTCStatsMember<uint32_t> frames_sent;
RTCStatsMember<uint32_t> huge_frames_sent;
RTCStatsMember<uint32_t> frames_received;
RTCStatsMember<uint32_t> frames_decoded;
RTCStatsMember<uint32_t> frames_dropped;
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<uint32_t> frames_corrupted;
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<uint32_t> partial_frames_lost;
// TODO(hbos): Not collected by `RTCStatsCollector`. crbug.com/659137
RTCStatsMember<uint32_t> full_frames_lost;
// Audio-only members
RTCStatsMember<double> audio_level; // Receive-only
RTCStatsMember<double> total_audio_energy; // Receive-only
RTCStatsMember<double> echo_return_loss;
RTCStatsMember<double> echo_return_loss_enhancement;
RTCStatsMember<uint64_t> total_samples_received;
RTCStatsMember<double> total_samples_duration; // Receive-only
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;
// Non-standard audio-only member
// TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
// Non-standard metric showing target delay of jitter buffer.
// This value is increased by the target jitter buffer delay every time a
// sample is emitted by the jitter buffer. The added target is the target
// delay, in seconds, at the time that the sample was emitted from the jitter
// buffer. (https://github.com/w3c/webrtc-provisional-stats/pull/20)
// Currently it is implemented only for audio.
// TODO(titovartem) implement for video streams when will be requested.
RTCNonStandardStatsMember<double> jitter_buffer_target_delay;
// TODO(henrik.lundin): Add description of the interruption metrics at
// https://github.com/henbos/webrtc-provisional-stats/issues/17
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// Non-standard video-only members.
// https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
RTCNonStandardStatsMember<uint32_t> freeze_count;
RTCNonStandardStatsMember<uint32_t> pause_count;
RTCNonStandardStatsMember<double> total_freezes_duration;
RTCNonStandardStatsMember<double> total_pauses_duration;
RTCNonStandardStatsMember<double> total_frames_duration;
RTCNonStandardStatsMember<double> sum_squared_frame_durations;
};
// https://w3c.github.io/webrtc-stats/#pcstats-dict*
class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCPeerConnectionStats(const std::string& id, int64_t timestamp_us);
RTCPeerConnectionStats(std::string&& id, int64_t timestamp_us);
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*
// TODO(hbos): Tracking bug crbug.com/657854
class RTC_EXPORT RTCRTPStreamStats : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCRTPStreamStats(const RTCRTPStreamStats& other);
~RTCRTPStreamStats() override;
RTCStatsMember<uint32_t> ssrc;
RTCStatsMember<std::string> kind;
// Obsolete: track_id
RTCStatsMember<std::string> track_id;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> codec_id;
// Obsolete
RTCStatsMember<std::string> media_type; // renamed to kind.
protected:
RTCRTPStreamStats(const std::string& id, int64_t timestamp_us);
RTCRTPStreamStats(std::string&& id, int64_t timestamp_us);
};
// 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;
// TODO(hbos) The following fields need to be added and migrated
// both from RTCInboundRtpStreamStats and RTCRemoteInboundRtpStreamStats:
// packetsReceived, packetsRepaired, burstPacketsLost,
// burstPacketDiscarded, burstLossCount, burstDiscardCount, burstLossRate,
// burstDiscardRate, gapLossRate, gapDiscardRate, framesDropped,
// partialFramesLost, fullFramesLost
// crbug.com/webrtc/12532
RTCStatsMember<double> jitter;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<uint64_t> packets_discarded;
protected:
RTCReceivedRtpStreamStats(const std::string&& id, int64_t timestamp_us);
RTCReceivedRtpStreamStats(std::string&& id, int64_t timestamp_us);
};
// 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<uint32_t> packets_sent;
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(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
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<uint32_t> 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_