rtcstats_objects.h

/*

 *  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 "absl/types/optional.h"

#include "api/stats/rtc_stats.h"

#include "rtc_base/system/rtc_export.h"

namespace webrtc {

// 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() override;

  absl::optional<std::string> fingerprint;

  absl::optional<std::string> fingerprint_algorithm;

  absl::optional<std::string> base64_certificate;

  absl::optional<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(std::string id, Timestamp timestamp);

  ~RTCCodecStats() override;

  absl::optional<std::string> transport_id;

  absl::optional<uint32_t> payload_type;

  absl::optional<std::string> mime_type;

  absl::optional<uint32_t> clock_rate;

  absl::optional<uint32_t> channels;

  absl::optional<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() override;

  absl::optional<std::string> label;

  absl::optional<std::string> protocol;

  absl::optional<int32_t> data_channel_identifier;

  absl::optional<std::string> state;

  absl::optional<uint32_t> messages_sent;

  absl::optional<uint64_t> bytes_sent;

  absl::optional<uint32_t> messages_received;

  absl::optional<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() override;

  absl::optional<std::string> transport_id;

  absl::optional<std::string> local_candidate_id;

  absl::optional<std::string> remote_candidate_id;

  absl::optional<std::string> state;

  // Obsolete: priority

  absl::optional<uint64_t> priority;

  absl::optional<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.

  absl::optional<bool> writable;

  absl::optional<uint64_t> packets_sent;

  absl::optional<uint64_t> packets_received;

  absl::optional<uint64_t> bytes_sent;

  absl::optional<uint64_t> bytes_received;

  absl::optional<double> total_round_trip_time;

  absl::optional<double> current_round_trip_time;

  absl::optional<double> available_outgoing_bitrate;

  absl::optional<double> available_incoming_bitrate;

  absl::optional<uint64_t> requests_received;

  absl::optional<uint64_t> requests_sent;

  absl::optional<uint64_t> responses_received;

  absl::optional<uint64_t> responses_sent;

  absl::optional<uint64_t> consent_requests_sent;

  absl::optional<uint64_t> packets_discarded_on_send;

  absl::optional<uint64_t> bytes_discarded_on_send;

  absl::optional<double> last_packet_received_timestamp;

  absl::optional<double> last_packet_sent_timestamp;

};

// https://w3c.github.io/webrtc-stats/#icecandidate-dict*

class RTC_EXPORT RTCIceCandidateStats : public RTCStats {

 public:

  WEBRTC_RTCSTATS_DECL();

  ~RTCIceCandidateStats() override;

  absl::optional<std::string> transport_id;

  // Obsolete: is_remote

  absl::optional<bool> is_remote;

  absl::optional<std::string> network_type;

  absl::optional<std::string> ip;

  absl::optional<std::string> address;

  absl::optional<int32_t> port;

  absl::optional<std::string> protocol;

  absl::optional<std::string> relay_protocol;

  absl::optional<std::string> candidate_type;

  absl::optional<int32_t> priority;

  absl::optional<std::string> url;

  absl::optional<std::string> foundation;

  absl::optional<std::string> related_address;

  absl::optional<int32_t> related_port;

  absl::optional<std::string> username_fragment;

  absl::optional<std::string> tcp_type;

  // The following metrics are NOT exposed to JavaScript. We should consider

  // standardizing or removing them.

  absl::optional<bool> vpn;

  absl::optional<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() override;

  absl::optional<uint32_t> data_channels_opened;

  absl::optional<uint32_t> data_channels_closed;

};

// https://w3c.github.io/webrtc-stats/#streamstats-dict*

class RTC_EXPORT RTCRtpStreamStats : public RTCStats {

 public:

  WEBRTC_RTCSTATS_DECL();

  ~RTCRtpStreamStats() override;

  absl::optional<uint32_t> ssrc;

  absl::optional<std::string> kind;

  absl::optional<std::string> transport_id;

  absl::optional<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() override;

  absl::optional<double> jitter;

  absl::optional<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() override;

  absl::optional<uint64_t> packets_sent;

  absl::optional<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() override;

  absl::optional<std::string> playout_id;

  absl::optional<std::string> track_identifier;

  absl::optional<std::string> mid;

  absl::optional<std::string> remote_id;

  absl::optional<uint32_t> packets_received;

  absl::optional<uint64_t> packets_discarded;

  absl::optional<uint64_t> fec_packets_received;

  absl::optional<uint64_t> fec_bytes_received;

  absl::optional<uint64_t> fec_packets_discarded;

  // Inbound FEC SSRC. Only present if a mechanism like FlexFEC is negotiated.

  absl::optional<uint32_t> fec_ssrc;

  absl::optional<uint64_t> bytes_received;

  absl::optional<uint64_t> header_bytes_received;

  // Inbound RTX stats. Only defined when RTX is used and it is therefore

  // possible to distinguish retransmissions.

  absl::optional<uint64_t> retransmitted_packets_received;

  absl::optional<uint64_t> retransmitted_bytes_received;

  absl::optional<uint32_t> rtx_ssrc;

  absl::optional<double> last_packet_received_timestamp;

  absl::optional<double> jitter_buffer_delay;

  absl::optional<double> jitter_buffer_target_delay;

  absl::optional<double> jitter_buffer_minimum_delay;

  absl::optional<uint64_t> jitter_buffer_emitted_count;

  absl::optional<uint64_t> total_samples_received;

  absl::optional<uint64_t> concealed_samples;

  absl::optional<uint64_t> silent_concealed_samples;

  absl::optional<uint64_t> concealment_events;

  absl::optional<uint64_t> inserted_samples_for_deceleration;

  absl::optional<uint64_t> removed_samples_for_acceleration;

  absl::optional<double> audio_level;

  absl::optional<double> total_audio_energy;

  absl::optional<double> total_samples_duration;

  // Stats below are only implemented or defined for video.

  absl::optional<uint32_t> frames_received;

  absl::optional<uint32_t> frame_width;

  absl::optional<uint32_t> frame_height;

  absl::optional<double> frames_per_second;

  absl::optional<uint32_t> frames_decoded;

  absl::optional<uint32_t> key_frames_decoded;

  absl::optional<uint32_t> frames_dropped;

  absl::optional<double> total_decode_time;

  absl::optional<double> total_processing_delay;

  absl::optional<double> total_assembly_time;

  absl::optional<uint32_t> frames_assembled_from_multiple_packets;

  // TODO(https://crbug.com/webrtc/15600): Implement framesRendered, which is

  // incremented at the same time that totalInterFrameDelay and

  // totalSquaredInterFrameDelay is incremented. (Dividing inter-frame delay by

  // framesDecoded is slightly wrong.)

  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesrendered

//

  // TODO(https://crbug.com/webrtc/15601): Inter-frame, pause and freeze metrics

  // all related to when the frame is rendered, but our implementation measures

  // at delivery to sink, not at actual render time. When we have an actual

  // frame rendered callback, move the calculating of these metrics to there in

  // order to make them more accurate.

  absl::optional<double> total_inter_frame_delay;

  absl::optional<double> total_squared_inter_frame_delay;

  absl::optional<uint32_t> pause_count;

  absl::optional<double> total_pauses_duration;

  absl::optional<uint32_t> freeze_count;

  absl::optional<double> total_freezes_duration;

  // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype

  absl::optional<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?

  absl::optional<double> estimated_playout_timestamp;

  // Only defined for video.

  // In JavaScript, this is only exposed if HW exposure is allowed.

  absl::optional<std::string> decoder_implementation;

  // FIR and PLI counts are only defined for |kind == "video"|.

  absl::optional<uint32_t> fir_count;

  absl::optional<uint32_t> pli_count;

  absl::optional<uint32_t> nack_count;

  absl::optional<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.

  absl::optional<std::string> goog_timing_frame_info;

  // In JavaScript, this is only exposed if HW exposure is allowed.

  absl::optional<bool> power_efficient_decoder;

  // The following metrics are NOT exposed to JavaScript. We should consider

  // standardizing or removing them.

  absl::optional<uint64_t> jitter_buffer_flushes;

  absl::optional<uint64_t> delayed_packet_outage_samples;

  absl::optional<double> relative_packet_arrival_delay;

  absl::optional<uint32_t> interruption_count;

  absl::optional<double> total_interruption_duration;

  absl::optional<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() override;

  absl::optional<std::string> media_source_id;

  absl::optional<std::string> remote_id;

  absl::optional<std::string> mid;

  absl::optional<std::string> rid;

  absl::optional<uint64_t> retransmitted_packets_sent;

  absl::optional<uint64_t> header_bytes_sent;

  absl::optional<uint64_t> retransmitted_bytes_sent;

  absl::optional<double> target_bitrate;

  absl::optional<uint32_t> frames_encoded;

  absl::optional<uint32_t> key_frames_encoded;

  absl::optional<double> total_encode_time;

  absl::optional<uint64_t> total_encoded_bytes_target;

  absl::optional<uint32_t> frame_width;

  absl::optional<uint32_t> frame_height;

  absl::optional<double> frames_per_second;

  absl::optional<uint32_t> frames_sent;

  absl::optional<uint32_t> huge_frames_sent;

  absl::optional<double> total_packet_send_delay;

  absl::optional<std::string> quality_limitation_reason;

  absl::optional<std::map<std::string, double>> quality_limitation_durations;

  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges

  absl::optional<uint32_t> quality_limitation_resolution_changes;

  // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype

  absl::optional<std::string> content_type;

  // In JavaScript, this is only exposed if HW exposure is allowed.

  // Only implemented for video.

  // TODO(https://crbug.com/webrtc/14178): Implement for audio as well.

  absl::optional<std::string> encoder_implementation;

  // FIR and PLI counts are only defined for |kind == "video"|.

  absl::optional<uint32_t> fir_count;

  absl::optional<uint32_t> pli_count;

  absl::optional<uint32_t> nack_count;

  absl::optional<uint64_t> qp_sum;

  absl::optional<bool> active;

  // In JavaScript, this is only exposed if HW exposure is allowed.

  absl::optional<bool> power_efficient_encoder;

  absl::optional<std::string> scalability_mode;

  // RTX ssrc. Only present if RTX is negotiated.

  absl::optional<uint32_t> rtx_ssrc;

};

// 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() override;

  absl::optional<std::string> local_id;

  absl::optional<double> round_trip_time;

  absl::optional<double> fraction_lost;

  absl::optional<double> total_round_trip_time;

  absl::optional<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() override;

  absl::optional<std::string> local_id;

  absl::optional<double> remote_timestamp;

  absl::optional<uint64_t> reports_sent;

  absl::optional<double> round_trip_time;

  absl::optional<uint64_t> round_trip_time_measurements;

  absl::optional<double> total_round_trip_time;

};

// https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats

class RTC_EXPORT RTCMediaSourceStats : public RTCStats {

 public:

  WEBRTC_RTCSTATS_DECL();

  ~RTCMediaSourceStats() override;

  absl::optional<std::string> track_identifier;

  absl::optional<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() override;

  absl::optional<double> audio_level;

  absl::optional<double> total_audio_energy;

  absl::optional<double> total_samples_duration;

  absl::optional<double> echo_return_loss;

  absl::optional<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() override;

  absl::optional<uint32_t> width;

  absl::optional<uint32_t> height;

  absl::optional<uint32_t> frames;

  absl::optional<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() override;

  absl::optional<uint64_t> bytes_sent;

  absl::optional<uint64_t> packets_sent;

  absl::optional<uint64_t> bytes_received;

  absl::optional<uint64_t> packets_received;

  absl::optional<std::string> rtcp_transport_stats_id;

  absl::optional<std::string> dtls_state;

  absl::optional<std::string> selected_candidate_pair_id;

  absl::optional<std::string> local_certificate_id;

  absl::optional<std::string> remote_certificate_id;

  absl::optional<std::string> tls_version;

  absl::optional<std::string> dtls_cipher;

  absl::optional<std::string> dtls_role;

  absl::optional<std::string> srtp_cipher;

  absl::optional<uint32_t> selected_candidate_pair_changes;

  absl::optional<std::string> ice_role;

  absl::optional<std::string> ice_local_username_fragment;

  absl::optional<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() override;

  absl::optional<std::string> kind;

  absl::optional<double> synthesized_samples_duration;

  absl::optional<uint64_t> synthesized_samples_events;

  absl::optional<double> total_samples_duration;

  absl::optional<double> total_playout_delay;

  absl::optional<uint64_t> total_samples_count;

};

}  // namespace webrtc

#endif  // API_STATS_RTCSTATS_OBJECTS_H_

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