| /* |
| * 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 <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<uint32_t> payload_type; |
| RTCStatsMember<std::string> mime_type; |
| RTCStatsMember<uint32_t> clock_rate; |
| // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061 |
| RTCStatsMember<uint32_t> channels; |
| // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061 |
| 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> datachannelid; |
| // 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; |
| 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> 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; |
| }; |
| |
| // 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; |
| RTCStatsMember<bool> is_remote; |
| RTCStatsMember<std::string> network_type; |
| RTCStatsMember<std::string> ip; |
| 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; |
| // TODO(hbos): |deleted = true| case is not supported by |RTCStatsCollector|. |
| // crbug.com/632723 |
| RTCStatsMember<bool> deleted; // = false |
| |
| 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; |
| // TODO(gustaf): Implement jitter_buffer_delay for video (currently |
| // implemented for audio only). |
| // https://crbug.com/webrtc/8318 |
| 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; |
| // 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; |
| // TODO(hbos): Remote case not supported by |RTCStatsCollector|. |
| // crbug.com/657855, 657856 |
| RTCStatsMember<bool> is_remote; // = false |
| RTCStatsMember<std::string> media_type; // renamed to kind. |
| RTCStatsMember<std::string> kind; |
| RTCStatsMember<std::string> track_id; |
| RTCStatsMember<std::string> transport_id; |
| RTCStatsMember<std::string> codec_id; |
| // FIR and PLI counts are only defined for |media_type == "video"|. |
| RTCStatsMember<uint32_t> fir_count; |
| RTCStatsMember<uint32_t> pli_count; |
| // TODO(hbos): NACK count should be collected by |RTCStatsCollector| for both |
| // audio and video but is only defined in the "video" case. crbug.com/657856 |
| RTCStatsMember<uint32_t> nack_count; |
| // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657854 |
| // SLI count is only defined for |media_type == "video"|. |
| RTCStatsMember<uint32_t> sli_count; |
| RTCStatsMember<uint64_t> qp_sum; |
| |
| protected: |
| RTCRTPStreamStats(const std::string& id, int64_t timestamp_us); |
| RTCRTPStreamStats(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 RTCRTPStreamStats { |
| 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<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<int32_t> packets_lost; // Signed per RFC 3550 |
| RTCStatsMember<double> last_packet_received_timestamp; |
| // TODO(hbos): Collect and populate this value for both "audio" and "video", |
| // currently not collected for "video". https://bugs.webrtc.org/7065 |
| RTCStatsMember<double> jitter; |
| // 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_discarded; |
| // 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> frames_decoded; |
| RTCStatsMember<uint32_t> key_frames_decoded; |
| 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; |
| }; |
| |
| // 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<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; |
| // 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 |
| // TODO(https://crbug.com/webrtc/10686): Also expose |
| // qualityLimitationDurations. Requires RTCStatsMember support for |
| // "record<DOMString, double>", see https://crbug.com/webrtc/10685. |
| RTCStatsMember<std::string> quality_limitation_reason; |
| // 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; |
| }; |
| |
| // TODO(https://crbug.com/webrtc/10671): Refactor the stats dictionaries to have |
| // the same hierarchy as in the spec; implement RTCReceivedRtpStreamStats. |
| // Several metrics are shared between "outbound-rtp", "remote-inbound-rtp", |
| // "inbound-rtp" and "remote-outbound-rtp". In the spec there is a hierarchy of |
| // dictionaries that minimizes defining the same metrics in multiple places. |
| // From JavaScript this hierarchy is not observable and the spec's hierarchy is |
| // purely editorial. In C++ non-final classes in the hierarchy could be used to |
| // refer to different stats objects within the hierarchy. |
| // https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict* |
| class RTC_EXPORT RTCRemoteInboundRtpStreamStats final : public RTCStats { |
| 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; |
| |
| // In the spec RTCRemoteInboundRtpStreamStats inherits from RTCRtpStreamStats |
| // and RTCReceivedRtpStreamStats. The members here are listed based on where |
| // they are defined in the spec. |
| // RTCRtpStreamStats |
| RTCStatsMember<uint32_t> ssrc; |
| RTCStatsMember<std::string> kind; |
| RTCStatsMember<std::string> transport_id; |
| RTCStatsMember<std::string> codec_id; |
| // RTCReceivedRtpStreamStats |
| RTCStatsMember<int32_t> packets_lost; |
| RTCStatsMember<double> jitter; |
| // TODO(hbos): The following RTCReceivedRtpStreamStats metrics should also be |
| // implemented: packetsReceived, packetsDiscarded, packetsRepaired, |
| // burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount, |
| // burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate. |
| // RTCRemoteInboundRtpStreamStats |
| RTCStatsMember<std::string> local_id; |
| RTCStatsMember<double> round_trip_time; |
| // TODO(hbos): The following RTCRemoteInboundRtpStreamStats metric should also |
| // be implemented: fractionLost. |
| }; |
| |
| // 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; |
| }; |
| |
| // 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; |
| // TODO(hbos): Implement this metric. |
| 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> bytes_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_ |