diff options
Diffstat (limited to 'third_party/libwebrtc/api/stats/rtcstats_objects.h')
-rw-r--r-- | third_party/libwebrtc/api/stats/rtcstats_objects.h | 417 |
1 files changed, 209 insertions, 208 deletions
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 <string> #include <vector> +#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<std::string> fingerprint; - RTCStatsMember<std::string> fingerprint_algorithm; - RTCStatsMember<std::string> base64_certificate; - RTCStatsMember<std::string> issuer_certificate_id; + 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* @@ -43,12 +44,12 @@ class RTC_EXPORT RTCCodecStats final : public RTCStats { RTCCodecStats(std::string id, Timestamp timestamp); ~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; + 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* @@ -58,14 +59,14 @@ class RTC_EXPORT RTCDataChannelStats final : public RTCStats { RTCDataChannelStats(std::string id, Timestamp timestamp); ~RTCDataChannelStats() override; - RTCStatsMember<std::string> label; - RTCStatsMember<std::string> protocol; - RTCStatsMember<int32_t> data_channel_identifier; - RTCStatsMember<std::string> state; - RTCStatsMember<uint32_t> messages_sent; - RTCStatsMember<uint64_t> bytes_sent; - RTCStatsMember<uint32_t> messages_received; - RTCStatsMember<uint64_t> bytes_received; + 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* @@ -75,35 +76,35 @@ class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats { RTCIceCandidatePairStats(std::string id, Timestamp timestamp); ~RTCIceCandidatePairStats() override; - RTCStatsMember<std::string> transport_id; - RTCStatsMember<std::string> local_candidate_id; - RTCStatsMember<std::string> remote_candidate_id; - RTCStatsMember<std::string> state; + 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 - RTCStatsMember<uint64_t> priority; - RTCStatsMember<bool> nominated; + 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. - RTCStatsMember<bool> writable; - RTCStatsMember<uint64_t> packets_sent; - RTCStatsMember<uint64_t> packets_received; - RTCStatsMember<uint64_t> bytes_sent; - RTCStatsMember<uint64_t> bytes_received; - RTCStatsMember<double> total_round_trip_time; - RTCStatsMember<double> current_round_trip_time; - RTCStatsMember<double> available_outgoing_bitrate; - RTCStatsMember<double> available_incoming_bitrate; - RTCStatsMember<uint64_t> requests_received; - RTCStatsMember<uint64_t> requests_sent; - RTCStatsMember<uint64_t> responses_received; - RTCStatsMember<uint64_t> responses_sent; - RTCStatsMember<uint64_t> consent_requests_sent; - RTCStatsMember<uint64_t> packets_discarded_on_send; - RTCStatsMember<uint64_t> bytes_discarded_on_send; - RTCStatsMember<double> last_packet_received_timestamp; - RTCStatsMember<double> last_packet_sent_timestamp; + 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* @@ -112,28 +113,28 @@ class RTC_EXPORT RTCIceCandidateStats : public RTCStats { WEBRTC_RTCSTATS_DECL(); ~RTCIceCandidateStats() override; - RTCStatsMember<std::string> transport_id; + absl::optional<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; - RTCStatsMember<std::string> candidate_type; - RTCStatsMember<int32_t> priority; - RTCStatsMember<std::string> url; - RTCStatsMember<std::string> foundation; - RTCStatsMember<std::string> related_address; - RTCStatsMember<int32_t> related_port; - RTCStatsMember<std::string> username_fragment; - RTCStatsMember<std::string> tcp_type; + 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. - RTCStatsMember<bool> vpn; - RTCStatsMember<std::string> network_adapter_type; + absl::optional<bool> vpn; + absl::optional<std::string> 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<uint32_t> data_channels_opened; - RTCStatsMember<uint32_t> data_channels_closed; + absl::optional<uint32_t> data_channels_opened; + absl::optional<uint32_t> 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<uint32_t> ssrc; - RTCStatsMember<std::string> kind; - RTCStatsMember<std::string> transport_id; - RTCStatsMember<std::string> codec_id; + 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); @@ -193,8 +194,8 @@ class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRtpStreamStats { WEBRTC_RTCSTATS_DECL(); ~RTCReceivedRtpStreamStats() override; - RTCStatsMember<double> jitter; - RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550 + absl::optional<double> jitter; + absl::optional<int32_t> 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<uint64_t> packets_sent; - RTCStatsMember<uint64_t> bytes_sent; + absl::optional<uint64_t> packets_sent; + absl::optional<uint64_t> 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<std::string> playout_id; - RTCStatsMember<std::string> track_identifier; - RTCStatsMember<std::string> mid; - RTCStatsMember<std::string> remote_id; - RTCStatsMember<uint32_t> packets_received; - RTCStatsMember<uint64_t> packets_discarded; - RTCStatsMember<uint64_t> fec_packets_received; - RTCStatsMember<uint64_t> fec_bytes_received; - RTCStatsMember<uint64_t> fec_packets_discarded; + 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. - RTCStatsMember<uint32_t> fec_ssrc; - RTCStatsMember<uint64_t> bytes_received; - RTCStatsMember<uint64_t> header_bytes_received; + 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. - RTCStatsMember<uint64_t> retransmitted_packets_received; - RTCStatsMember<uint64_t> retransmitted_bytes_received; - RTCStatsMember<uint32_t> rtx_ssrc; - - RTCStatsMember<double> last_packet_received_timestamp; - RTCStatsMember<double> jitter_buffer_delay; - RTCStatsMember<double> jitter_buffer_target_delay; - RTCStatsMember<double> jitter_buffer_minimum_delay; - RTCStatsMember<uint64_t> jitter_buffer_emitted_count; - RTCStatsMember<uint64_t> total_samples_received; - RTCStatsMember<uint64_t> concealed_samples; - RTCStatsMember<uint64_t> silent_concealed_samples; - RTCStatsMember<uint64_t> concealment_events; - RTCStatsMember<uint64_t> inserted_samples_for_deceleration; - RTCStatsMember<uint64_t> removed_samples_for_acceleration; - RTCStatsMember<double> audio_level; - RTCStatsMember<double> total_audio_energy; - RTCStatsMember<double> total_samples_duration; + 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. - RTCStatsMember<uint32_t> frames_received; - RTCStatsMember<uint32_t> frame_width; - RTCStatsMember<uint32_t> frame_height; - RTCStatsMember<double> frames_per_second; - RTCStatsMember<uint32_t> frames_decoded; - RTCStatsMember<uint32_t> key_frames_decoded; - RTCStatsMember<uint32_t> frames_dropped; - RTCStatsMember<double> total_decode_time; - RTCStatsMember<double> total_processing_delay; - RTCStatsMember<double> total_assembly_time; - RTCStatsMember<uint32_t> frames_assembled_from_multiple_packets; + 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 @@ -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<double> total_inter_frame_delay; - RTCStatsMember<double> total_squared_inter_frame_delay; - RTCStatsMember<uint32_t> pause_count; - RTCStatsMember<double> total_pauses_duration; - RTCStatsMember<uint32_t> freeze_count; - RTCStatsMember<double> total_freezes_duration; + 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 - RTCStatsMember<std::string> content_type; + 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? - RTCStatsMember<double> estimated_playout_timestamp; + absl::optional<double> estimated_playout_timestamp; // Only defined for video. // In JavaScript, this is only exposed if HW exposure is allowed. - RTCStatsMember<std::string> decoder_implementation; + absl::optional<std::string> decoder_implementation; // FIR and PLI counts are only defined for |kind == "video"|. - RTCStatsMember<uint32_t> fir_count; - RTCStatsMember<uint32_t> pli_count; - RTCStatsMember<uint32_t> nack_count; - RTCStatsMember<uint64_t> qp_sum; + 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. - RTCStatsMember<std::string> goog_timing_frame_info; + absl::optional<std::string> goog_timing_frame_info; // In JavaScript, this is only exposed if HW exposure is allowed. - RTCStatsMember<bool> power_efficient_decoder; + absl::optional<bool> power_efficient_decoder; // The following metrics are NOT exposed to JavaScript. We should consider // standardizing or removing them. - RTCStatsMember<uint64_t> jitter_buffer_flushes; - RTCStatsMember<uint64_t> delayed_packet_outage_samples; - RTCStatsMember<double> relative_packet_arrival_delay; - RTCStatsMember<uint32_t> interruption_count; - RTCStatsMember<double> total_interruption_duration; - RTCStatsMember<double> min_playout_delay; + 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* @@ -324,46 +325,46 @@ class RTC_EXPORT RTCOutboundRtpStreamStats final RTCOutboundRtpStreamStats(std::string id, Timestamp timestamp); ~RTCOutboundRtpStreamStats() override; - RTCStatsMember<std::string> media_source_id; - RTCStatsMember<std::string> remote_id; - RTCStatsMember<std::string> mid; - RTCStatsMember<std::string> rid; - RTCStatsMember<uint64_t> retransmitted_packets_sent; - RTCStatsMember<uint64_t> header_bytes_sent; - RTCStatsMember<uint64_t> retransmitted_bytes_sent; - RTCStatsMember<double> target_bitrate; - RTCStatsMember<uint32_t> frames_encoded; - RTCStatsMember<uint32_t> key_frames_encoded; - RTCStatsMember<double> total_encode_time; - RTCStatsMember<uint64_t> total_encoded_bytes_target; - RTCStatsMember<uint32_t> frame_width; - RTCStatsMember<uint32_t> frame_height; - RTCStatsMember<double> frames_per_second; - RTCStatsMember<uint32_t> frames_sent; - RTCStatsMember<uint32_t> huge_frames_sent; - RTCStatsMember<double> total_packet_send_delay; - RTCStatsMember<std::string> quality_limitation_reason; - RTCStatsMember<std::map<std::string, double>> quality_limitation_durations; + 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 - RTCStatsMember<uint32_t> quality_limitation_resolution_changes; + absl::optional<uint32_t> quality_limitation_resolution_changes; // https://w3c.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype - RTCStatsMember<std::string> content_type; + 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. - RTCStatsMember<std::string> encoder_implementation; + absl::optional<std::string> encoder_implementation; // FIR and PLI counts are only defined for |kind == "video"|. - RTCStatsMember<uint32_t> fir_count; - RTCStatsMember<uint32_t> pli_count; - RTCStatsMember<uint32_t> nack_count; - RTCStatsMember<uint64_t> qp_sum; - RTCStatsMember<bool> active; + 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. - RTCStatsMember<bool> power_efficient_encoder; - RTCStatsMember<std::string> scalability_mode; + absl::optional<bool> power_efficient_encoder; + absl::optional<std::string> scalability_mode; // RTX ssrc. Only present if RTX is negotiated. - RTCStatsMember<uint32_t> rtx_ssrc; + absl::optional<uint32_t> 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<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; + 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* @@ -389,12 +390,12 @@ class RTC_EXPORT RTCRemoteOutboundRtpStreamStats final RTCRemoteOutboundRtpStreamStats(std::string id, Timestamp timestamp); ~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; + 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 @@ -403,8 +404,8 @@ class RTC_EXPORT RTCMediaSourceStats : public RTCStats { WEBRTC_RTCSTATS_DECL(); ~RTCMediaSourceStats() override; - RTCStatsMember<std::string> track_identifier; - RTCStatsMember<std::string> kind; + absl::optional<std::string> track_identifier; + absl::optional<std::string> 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<double> audio_level; - RTCStatsMember<double> total_audio_energy; - RTCStatsMember<double> total_samples_duration; - RTCStatsMember<double> echo_return_loss; - RTCStatsMember<double> echo_return_loss_enhancement; + 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 @@ -431,10 +432,10 @@ class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats { RTCVideoSourceStats(std::string id, Timestamp timestamp); ~RTCVideoSourceStats() override; - RTCStatsMember<uint32_t> width; - RTCStatsMember<uint32_t> height; - RTCStatsMember<uint32_t> frames; - RTCStatsMember<double> frames_per_second; + 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* @@ -444,23 +445,23 @@ class RTC_EXPORT RTCTransportStats final : public RTCStats { RTCTransportStats(std::string id, Timestamp timestamp); ~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; - RTCStatsMember<std::string> dtls_state; - RTCStatsMember<std::string> selected_candidate_pair_id; - RTCStatsMember<std::string> local_certificate_id; - RTCStatsMember<std::string> remote_certificate_id; - RTCStatsMember<std::string> tls_version; - RTCStatsMember<std::string> dtls_cipher; - RTCStatsMember<std::string> dtls_role; - RTCStatsMember<std::string> srtp_cipher; - RTCStatsMember<uint32_t> selected_candidate_pair_changes; - RTCStatsMember<std::string> ice_role; - RTCStatsMember<std::string> ice_local_username_fragment; - RTCStatsMember<std::string> ice_state; + 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* @@ -470,12 +471,12 @@ class RTC_EXPORT RTCAudioPlayoutStats final : public RTCStats { RTCAudioPlayoutStats(const std::string& id, Timestamp timestamp); ~RTCAudioPlayoutStats() override; - RTCStatsMember<std::string> kind; - RTCStatsMember<double> synthesized_samples_duration; - RTCStatsMember<uint64_t> synthesized_samples_events; - RTCStatsMember<double> total_samples_duration; - RTCStatsMember<double> total_playout_delay; - RTCStatsMember<uint64_t> total_samples_count; + 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 |