From 8dd16259287f58f9273002717ec4d27e97127719 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 12 Jun 2024 07:43:14 +0200 Subject: Merging upstream version 127.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/api/stats/rtcstats_objects.h | 417 +++++++++++---------- 1 file changed, 209 insertions(+), 208 deletions(-) (limited to 'third_party/libwebrtc/api/stats/rtcstats_objects.h') diff --git a/third_party/libwebrtc/api/stats/rtcstats_objects.h b/third_party/libwebrtc/api/stats/rtcstats_objects.h index 351c2cbefe..9f51f56cc5 100644 --- a/third_party/libwebrtc/api/stats/rtcstats_objects.h +++ b/third_party/libwebrtc/api/stats/rtcstats_objects.h @@ -18,6 +18,7 @@ #include #include +#include "absl/types/optional.h" #include "api/stats/rtc_stats.h" #include "rtc_base/system/rtc_export.h" @@ -30,10 +31,10 @@ class RTC_EXPORT RTCCertificateStats final : public RTCStats { RTCCertificateStats(std::string id, Timestamp timestamp); ~RTCCertificateStats() override; - RTCStatsMember fingerprint; - RTCStatsMember fingerprint_algorithm; - RTCStatsMember base64_certificate; - RTCStatsMember issuer_certificate_id; + absl::optional fingerprint; + absl::optional fingerprint_algorithm; + absl::optional base64_certificate; + absl::optional issuer_certificate_id; }; // https://w3c.github.io/webrtc-stats/#codec-dict* @@ -43,12 +44,12 @@ class RTC_EXPORT RTCCodecStats final : public RTCStats { RTCCodecStats(std::string id, Timestamp timestamp); ~RTCCodecStats() override; - RTCStatsMember transport_id; - RTCStatsMember payload_type; - RTCStatsMember mime_type; - RTCStatsMember clock_rate; - RTCStatsMember channels; - RTCStatsMember sdp_fmtp_line; + absl::optional transport_id; + absl::optional payload_type; + absl::optional mime_type; + absl::optional clock_rate; + absl::optional channels; + absl::optional sdp_fmtp_line; }; // https://w3c.github.io/webrtc-stats/#dcstats-dict* @@ -58,14 +59,14 @@ class RTC_EXPORT RTCDataChannelStats final : public RTCStats { RTCDataChannelStats(std::string id, Timestamp timestamp); ~RTCDataChannelStats() override; - RTCStatsMember label; - RTCStatsMember protocol; - RTCStatsMember data_channel_identifier; - RTCStatsMember state; - RTCStatsMember messages_sent; - RTCStatsMember bytes_sent; - RTCStatsMember messages_received; - RTCStatsMember bytes_received; + absl::optional label; + absl::optional protocol; + absl::optional data_channel_identifier; + absl::optional state; + absl::optional messages_sent; + absl::optional bytes_sent; + absl::optional messages_received; + absl::optional bytes_received; }; // https://w3c.github.io/webrtc-stats/#candidatepair-dict* @@ -75,35 +76,35 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats { RTCIceCandidatePairStats(std::string id, Timestamp timestamp); ~RTCIceCandidatePairStats() override; - RTCStatsMember transport_id; - RTCStatsMember local_candidate_id; - RTCStatsMember remote_candidate_id; - RTCStatsMember state; + absl::optional transport_id; + absl::optional local_candidate_id; + absl::optional remote_candidate_id; + absl::optional state; // Obsolete: priority - RTCStatsMember priority; - RTCStatsMember nominated; + absl::optional priority; + absl::optional 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 writable; - RTCStatsMember packets_sent; - RTCStatsMember packets_received; - RTCStatsMember bytes_sent; - RTCStatsMember bytes_received; - RTCStatsMember total_round_trip_time; - RTCStatsMember current_round_trip_time; - RTCStatsMember available_outgoing_bitrate; - RTCStatsMember available_incoming_bitrate; - RTCStatsMember requests_received; - RTCStatsMember requests_sent; - RTCStatsMember responses_received; - RTCStatsMember responses_sent; - RTCStatsMember consent_requests_sent; - RTCStatsMember packets_discarded_on_send; - RTCStatsMember bytes_discarded_on_send; - RTCStatsMember last_packet_received_timestamp; - RTCStatsMember last_packet_sent_timestamp; + absl::optional writable; + absl::optional packets_sent; + absl::optional packets_received; + absl::optional bytes_sent; + absl::optional bytes_received; + absl::optional total_round_trip_time; + absl::optional current_round_trip_time; + absl::optional available_outgoing_bitrate; + absl::optional available_incoming_bitrate; + absl::optional requests_received; + absl::optional requests_sent; + absl::optional responses_received; + absl::optional responses_sent; + absl::optional consent_requests_sent; + absl::optional packets_discarded_on_send; + absl::optional bytes_discarded_on_send; + absl::optional last_packet_received_timestamp; + absl::optional last_packet_sent_timestamp; }; // https://w3c.github.io/webrtc-stats/#icecandidate-dict* @@ -112,28 +113,28 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats { WEBRTC_RTCSTATS_DECL(); ~RTCIceCandidateStats() override; - RTCStatsMember transport_id; + absl::optional transport_id; // Obsolete: is_remote - RTCStatsMember is_remote; - RTCStatsMember network_type; - RTCStatsMember ip; - RTCStatsMember address; - RTCStatsMember port; - RTCStatsMember protocol; - RTCStatsMember relay_protocol; - RTCStatsMember candidate_type; - RTCStatsMember priority; - RTCStatsMember url; - RTCStatsMember foundation; - RTCStatsMember related_address; - RTCStatsMember related_port; - RTCStatsMember username_fragment; - RTCStatsMember tcp_type; + absl::optional is_remote; + absl::optional network_type; + absl::optional ip; + absl::optional address; + absl::optional port; + absl::optional protocol; + absl::optional relay_protocol; + absl::optional candidate_type; + absl::optional priority; + absl::optional url; + absl::optional foundation; + absl::optional related_address; + absl::optional related_port; + absl::optional username_fragment; + absl::optional tcp_type; // The following metrics are NOT exposed to JavaScript. We should consider // standardizing or removing them. - RTCStatsMember vpn; - RTCStatsMember network_adapter_type; + absl::optional vpn; + absl::optional network_adapter_type; protected: RTCIceCandidateStats(std::string id, Timestamp timestamp, bool is_remote); @@ -168,8 +169,8 @@ class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats { RTCPeerConnectionStats(std::string id, Timestamp timestamp); ~RTCPeerConnectionStats() override; - RTCStatsMember data_channels_opened; - RTCStatsMember data_channels_closed; + absl::optional data_channels_opened; + absl::optional data_channels_closed; }; // https://w3c.github.io/webrtc-stats/#streamstats-dict* @@ -178,10 +179,10 @@ class RTC_EXPORT RTCRtpStreamStats : public RTCStats { WEBRTC_RTCSTATS_DECL(); ~RTCRtpStreamStats() override; - RTCStatsMember ssrc; - RTCStatsMember kind; - RTCStatsMember transport_id; - RTCStatsMember codec_id; + absl::optional ssrc; + absl::optional kind; + absl::optional transport_id; + absl::optional codec_id; protected: RTCRtpStreamStats(std::string id, Timestamp timestamp); @@ -193,8 +194,8 @@ class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRtpStreamStats { WEBRTC_RTCSTATS_DECL(); ~RTCReceivedRtpStreamStats() override; - RTCStatsMember jitter; - RTCStatsMember packets_lost; // Signed per RFC 3550 + absl::optional jitter; + absl::optional packets_lost; // Signed per RFC 3550 protected: RTCReceivedRtpStreamStats(std::string id, Timestamp timestamp); @@ -206,8 +207,8 @@ class RTC_EXPORT RTCSentRtpStreamStats : public RTCRtpStreamStats { WEBRTC_RTCSTATS_DECL(); ~RTCSentRtpStreamStats() override; - RTCStatsMember packets_sent; - RTCStatsMember bytes_sent; + absl::optional packets_sent; + absl::optional bytes_sent; protected: RTCSentRtpStreamStats(std::string id, Timestamp timestamp); @@ -221,51 +222,51 @@ class RTC_EXPORT RTCInboundRtpStreamStats final RTCInboundRtpStreamStats(std::string id, Timestamp timestamp); ~RTCInboundRtpStreamStats() override; - RTCStatsMember playout_id; - RTCStatsMember track_identifier; - RTCStatsMember mid; - RTCStatsMember remote_id; - RTCStatsMember packets_received; - RTCStatsMember packets_discarded; - RTCStatsMember fec_packets_received; - RTCStatsMember fec_bytes_received; - RTCStatsMember fec_packets_discarded; + absl::optional playout_id; + absl::optional track_identifier; + absl::optional mid; + absl::optional remote_id; + absl::optional packets_received; + absl::optional packets_discarded; + absl::optional fec_packets_received; + absl::optional fec_bytes_received; + absl::optional fec_packets_discarded; // Inbound FEC SSRC. Only present if a mechanism like FlexFEC is negotiated. - RTCStatsMember fec_ssrc; - RTCStatsMember bytes_received; - RTCStatsMember header_bytes_received; + absl::optional fec_ssrc; + absl::optional bytes_received; + absl::optional header_bytes_received; // Inbound RTX stats. Only defined when RTX is used and it is therefore // possible to distinguish retransmissions. - RTCStatsMember retransmitted_packets_received; - RTCStatsMember retransmitted_bytes_received; - RTCStatsMember rtx_ssrc; - - RTCStatsMember last_packet_received_timestamp; - RTCStatsMember jitter_buffer_delay; - RTCStatsMember jitter_buffer_target_delay; - RTCStatsMember jitter_buffer_minimum_delay; - RTCStatsMember jitter_buffer_emitted_count; - RTCStatsMember total_samples_received; - RTCStatsMember concealed_samples; - RTCStatsMember silent_concealed_samples; - RTCStatsMember concealment_events; - RTCStatsMember inserted_samples_for_deceleration; - RTCStatsMember removed_samples_for_acceleration; - RTCStatsMember audio_level; - RTCStatsMember total_audio_energy; - RTCStatsMember total_samples_duration; + absl::optional retransmitted_packets_received; + absl::optional retransmitted_bytes_received; + absl::optional rtx_ssrc; + + absl::optional last_packet_received_timestamp; + absl::optional jitter_buffer_delay; + absl::optional jitter_buffer_target_delay; + absl::optional jitter_buffer_minimum_delay; + absl::optional jitter_buffer_emitted_count; + absl::optional total_samples_received; + absl::optional concealed_samples; + absl::optional silent_concealed_samples; + absl::optional concealment_events; + absl::optional inserted_samples_for_deceleration; + absl::optional removed_samples_for_acceleration; + absl::optional audio_level; + absl::optional total_audio_energy; + absl::optional total_samples_duration; // Stats below are only implemented or defined for video. - RTCStatsMember frames_received; - RTCStatsMember frame_width; - RTCStatsMember frame_height; - RTCStatsMember frames_per_second; - RTCStatsMember frames_decoded; - RTCStatsMember key_frames_decoded; - RTCStatsMember frames_dropped; - RTCStatsMember total_decode_time; - RTCStatsMember total_processing_delay; - RTCStatsMember total_assembly_time; - RTCStatsMember frames_assembled_from_multiple_packets; + absl::optional frames_received; + absl::optional frame_width; + absl::optional frame_height; + absl::optional frames_per_second; + absl::optional frames_decoded; + absl::optional key_frames_decoded; + absl::optional frames_dropped; + absl::optional total_decode_time; + absl::optional total_processing_delay; + absl::optional total_assembly_time; + absl::optional 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 @@ -277,43 +278,43 @@ class RTC_EXPORT RTCInboundRtpStreamStats final // 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. - RTCStatsMember total_inter_frame_delay; - RTCStatsMember total_squared_inter_frame_delay; - RTCStatsMember pause_count; - RTCStatsMember total_pauses_duration; - RTCStatsMember freeze_count; - RTCStatsMember total_freezes_duration; + absl::optional total_inter_frame_delay; + absl::optional total_squared_inter_frame_delay; + absl::optional pause_count; + absl::optional total_pauses_duration; + absl::optional freeze_count; + absl::optional total_freezes_duration; // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype - RTCStatsMember content_type; + absl::optional 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 estimated_playout_timestamp; + absl::optional estimated_playout_timestamp; // Only defined for video. // In JavaScript, this is only exposed if HW exposure is allowed. - RTCStatsMember decoder_implementation; + absl::optional decoder_implementation; // FIR and PLI counts are only defined for |kind == "video"|. - RTCStatsMember fir_count; - RTCStatsMember pli_count; - RTCStatsMember nack_count; - RTCStatsMember qp_sum; + absl::optional fir_count; + absl::optional pli_count; + absl::optional nack_count; + absl::optional 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 goog_timing_frame_info; + absl::optional goog_timing_frame_info; // In JavaScript, this is only exposed if HW exposure is allowed. - RTCStatsMember power_efficient_decoder; + absl::optional power_efficient_decoder; // The following metrics are NOT exposed to JavaScript. We should consider // standardizing or removing them. - RTCStatsMember jitter_buffer_flushes; - RTCStatsMember delayed_packet_outage_samples; - RTCStatsMember relative_packet_arrival_delay; - RTCStatsMember interruption_count; - RTCStatsMember total_interruption_duration; - RTCStatsMember min_playout_delay; + absl::optional jitter_buffer_flushes; + absl::optional delayed_packet_outage_samples; + absl::optional relative_packet_arrival_delay; + absl::optional interruption_count; + absl::optional total_interruption_duration; + absl::optional min_playout_delay; }; // https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict* @@ -324,46 +325,46 @@ class RTC_EXPORT RTCOutboundRtpStreamStats final RTCOutboundRtpStreamStats(std::string id, Timestamp timestamp); ~RTCOutboundRtpStreamStats() override; - RTCStatsMember media_source_id; - RTCStatsMember remote_id; - RTCStatsMember mid; - RTCStatsMember rid; - RTCStatsMember retransmitted_packets_sent; - RTCStatsMember header_bytes_sent; - RTCStatsMember retransmitted_bytes_sent; - RTCStatsMember target_bitrate; - RTCStatsMember frames_encoded; - RTCStatsMember key_frames_encoded; - RTCStatsMember total_encode_time; - RTCStatsMember total_encoded_bytes_target; - RTCStatsMember frame_width; - RTCStatsMember frame_height; - RTCStatsMember frames_per_second; - RTCStatsMember frames_sent; - RTCStatsMember huge_frames_sent; - RTCStatsMember total_packet_send_delay; - RTCStatsMember quality_limitation_reason; - RTCStatsMember> quality_limitation_durations; + absl::optional media_source_id; + absl::optional remote_id; + absl::optional mid; + absl::optional rid; + absl::optional retransmitted_packets_sent; + absl::optional header_bytes_sent; + absl::optional retransmitted_bytes_sent; + absl::optional target_bitrate; + absl::optional frames_encoded; + absl::optional key_frames_encoded; + absl::optional total_encode_time; + absl::optional total_encoded_bytes_target; + absl::optional frame_width; + absl::optional frame_height; + absl::optional frames_per_second; + absl::optional frames_sent; + absl::optional huge_frames_sent; + absl::optional total_packet_send_delay; + absl::optional quality_limitation_reason; + absl::optional> quality_limitation_durations; // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges - RTCStatsMember quality_limitation_resolution_changes; + absl::optional quality_limitation_resolution_changes; // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype - RTCStatsMember content_type; + absl::optional 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. - RTCStatsMember encoder_implementation; + absl::optional encoder_implementation; // FIR and PLI counts are only defined for |kind == "video"|. - RTCStatsMember fir_count; - RTCStatsMember pli_count; - RTCStatsMember nack_count; - RTCStatsMember qp_sum; - RTCStatsMember active; + absl::optional fir_count; + absl::optional pli_count; + absl::optional nack_count; + absl::optional qp_sum; + absl::optional active; // In JavaScript, this is only exposed if HW exposure is allowed. - RTCStatsMember power_efficient_encoder; - RTCStatsMember scalability_mode; + absl::optional power_efficient_encoder; + absl::optional scalability_mode; // RTX ssrc. Only present if RTX is negotiated. - RTCStatsMember rtx_ssrc; + absl::optional rtx_ssrc; }; // https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict* @@ -374,11 +375,11 @@ class RTC_EXPORT RTCRemoteInboundRtpStreamStats final RTCRemoteInboundRtpStreamStats(std::string id, Timestamp timestamp); ~RTCRemoteInboundRtpStreamStats() override; - RTCStatsMember local_id; - RTCStatsMember round_trip_time; - RTCStatsMember fraction_lost; - RTCStatsMember total_round_trip_time; - RTCStatsMember round_trip_time_measurements; + absl::optional local_id; + absl::optional round_trip_time; + absl::optional fraction_lost; + absl::optional total_round_trip_time; + absl::optional round_trip_time_measurements; }; // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict* @@ -389,12 +390,12 @@ class RTC_EXPORT RTCRemoteOutboundRtpStreamStats final RTCRemoteOutboundRtpStreamStats(std::string id, Timestamp timestamp); ~RTCRemoteOutboundRtpStreamStats() override; - RTCStatsMember local_id; - RTCStatsMember remote_timestamp; - RTCStatsMember reports_sent; - RTCStatsMember round_trip_time; - RTCStatsMember round_trip_time_measurements; - RTCStatsMember total_round_trip_time; + absl::optional local_id; + absl::optional remote_timestamp; + absl::optional reports_sent; + absl::optional round_trip_time; + absl::optional round_trip_time_measurements; + absl::optional total_round_trip_time; }; // https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats @@ -403,8 +404,8 @@ class RTC_EXPORT RTCMediaSourceStats : public RTCStats { WEBRTC_RTCSTATS_DECL(); ~RTCMediaSourceStats() override; - RTCStatsMember track_identifier; - RTCStatsMember kind; + absl::optional track_identifier; + absl::optional kind; protected: RTCMediaSourceStats(std::string id, Timestamp timestamp); @@ -417,11 +418,11 @@ class RTC_EXPORT RTCAudioSourceStats final : public RTCMediaSourceStats { RTCAudioSourceStats(std::string id, Timestamp timestamp); ~RTCAudioSourceStats() override; - RTCStatsMember audio_level; - RTCStatsMember total_audio_energy; - RTCStatsMember total_samples_duration; - RTCStatsMember echo_return_loss; - RTCStatsMember echo_return_loss_enhancement; + absl::optional audio_level; + absl::optional total_audio_energy; + absl::optional total_samples_duration; + absl::optional echo_return_loss; + absl::optional echo_return_loss_enhancement; }; // https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats @@ -431,10 +432,10 @@ class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats { RTCVideoSourceStats(std::string id, Timestamp timestamp); ~RTCVideoSourceStats() override; - RTCStatsMember width; - RTCStatsMember height; - RTCStatsMember frames; - RTCStatsMember frames_per_second; + absl::optional width; + absl::optional height; + absl::optional frames; + absl::optional frames_per_second; }; // https://w3c.github.io/webrtc-stats/#transportstats-dict* @@ -444,23 +445,23 @@ class RTC_EXPORT RTCTransportStats final : public RTCStats { RTCTransportStats(std::string id, Timestamp timestamp); ~RTCTransportStats() override; - RTCStatsMember bytes_sent; - RTCStatsMember packets_sent; - RTCStatsMember bytes_received; - RTCStatsMember packets_received; - RTCStatsMember rtcp_transport_stats_id; - RTCStatsMember dtls_state; - RTCStatsMember selected_candidate_pair_id; - RTCStatsMember local_certificate_id; - RTCStatsMember remote_certificate_id; - RTCStatsMember tls_version; - RTCStatsMember dtls_cipher; - RTCStatsMember dtls_role; - RTCStatsMember srtp_cipher; - RTCStatsMember selected_candidate_pair_changes; - RTCStatsMember ice_role; - RTCStatsMember ice_local_username_fragment; - RTCStatsMember ice_state; + absl::optional bytes_sent; + absl::optional packets_sent; + absl::optional bytes_received; + absl::optional packets_received; + absl::optional rtcp_transport_stats_id; + absl::optional dtls_state; + absl::optional selected_candidate_pair_id; + absl::optional local_certificate_id; + absl::optional remote_certificate_id; + absl::optional tls_version; + absl::optional dtls_cipher; + absl::optional dtls_role; + absl::optional srtp_cipher; + absl::optional selected_candidate_pair_changes; + absl::optional ice_role; + absl::optional ice_local_username_fragment; + absl::optional ice_state; }; // https://w3c.github.io/webrtc-stats/#playoutstats-dict* @@ -470,12 +471,12 @@ class RTC_EXPORT RTCAudioPlayoutStats final : public RTCStats { RTCAudioPlayoutStats(const std::string& id, Timestamp timestamp); ~RTCAudioPlayoutStats() override; - RTCStatsMember kind; - RTCStatsMember synthesized_samples_duration; - RTCStatsMember synthesized_samples_events; - RTCStatsMember total_samples_duration; - RTCStatsMember total_playout_delay; - RTCStatsMember total_samples_count; + absl::optional kind; + absl::optional synthesized_samples_duration; + absl::optional synthesized_samples_events; + absl::optional total_samples_duration; + absl::optional total_playout_delay; + absl::optional total_samples_count; }; } // namespace webrtc -- cgit v1.2.3