[Stats] Migrate from the RTCStatsMember type alias to absl::optional.
With this CL, the only usage of RTCStatsMember within WebRTC is the
actual type alias declaration. It's not referenced anywhere anymore.
This allows us to deleting the type alias, but let's do that in a
standalone CL in case it gets reverted.
Bug: webrtc:15164
Change-Id: I766d07abb62b5ddd524859b8ed749394fc439e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335621
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41612}
diff --git a/api/stats/rtc_stats.h b/api/stats/rtc_stats.h
index 6781081..74e7fc4 100644
--- a/api/stats/rtc_stats.h
+++ b/api/stats/rtc_stats.h
@@ -20,6 +20,7 @@
#include <utility>
#include <vector>
+#include "absl/types/optional.h"
#include "api/stats/attribute.h"
#include "api/units/timestamp.h"
#include "rtc_base/checks.h"
@@ -38,8 +39,8 @@
// Use the `WEBRTC_RTCSTATS_IMPL` macro when implementing subclasses, see macro
// for details.
//
-// Derived classes list their dictionary attributes (RTCStatsMember<T> to soon
-// be replaced by absl::optional<T>) as public fields, allowing the following:
+// Derived classes list their dictionary attributes, absl::optional<T>, as
+// public fields, allowing the following:
//
// RTCFooStats foo("fooId", Timestamp::Micros(GetCurrentTime()));
// foo.bar = 42;
@@ -51,7 +52,7 @@
// iteration:
//
// for (const auto& attribute : foo.Attributes()) {
-// printf("%s = %s\n", attribute.name(), attribute.ValueToString().c_str());
+// printf("%s = %s\n", attribute.name(), attribute.ToString().c_str());
// }
class RTC_EXPORT RTCStats {
public:
@@ -72,12 +73,12 @@
// metrics as viewed via the Attribute wrapper.
std::vector<Attribute> Attributes() const;
template <typename T>
- Attribute GetAttribute(const RTCStatsMember<T>& stat) const {
+ Attribute GetAttribute(const absl::optional<T>& stat) const {
for (const auto& attribute : Attributes()) {
if (!attribute.holds_alternative<T>()) {
continue;
}
- if (absl::get<const RTCStatsMember<T>*>(attribute.as_variant()) ==
+ if (absl::get<const absl::optional<T>*>(attribute.as_variant()) ==
&stat) {
return attribute;
}
@@ -135,8 +136,8 @@
//
// RTCFooStats(const std::string& id, Timestamp timestamp);
//
-// RTCStatsMember<int32_t> foo;
-// RTCStatsMember<int32_t> bar;
+// absl::optional<int32_t> foo;
+// absl::optional<int32_t> bar;
// };
//
// rtcfoostats.cc:
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 351c2cb..9f51f56 100644
--- a/api/stats/rtcstats_objects.h
+++ b/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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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;
+ absl::optional<uint64_t> retransmitted_packets_received;
+ absl::optional<uint64_t> retransmitted_bytes_received;
+ absl::optional<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<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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 2edbf9d..1f814cb 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -970,6 +970,7 @@
"../api:scoped_refptr",
"../rtc_base:checks",
]
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_source_set("sdp_offer_answer") {
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 61b3bca..03512bb 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -23,6 +23,7 @@
#include <vector>
#include "absl/strings/str_replace.h"
+#include "absl/types/optional.h"
#include "api/candidate.h"
#include "api/dtls_transport_interface.h"
#include "api/media_stream_interface.h"
@@ -3723,7 +3724,7 @@
RTCTestStats(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- RTCStatsMember<int32_t> dummy_stat;
+ absl::optional<int32_t> dummy_stat;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats,
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 002f9d3..de8e6e3 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -18,6 +18,7 @@
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
+#include "absl/types/optional.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_options.h"
@@ -212,14 +213,14 @@
}
template <typename T>
- void MarkAttributeTested(const RTCStatsMember<T>& field,
+ void MarkAttributeTested(const absl::optional<T>& field,
bool test_successful) {
untested_attribute_names_.erase(stats_->GetAttribute(field).name());
all_tests_successful_ &= test_successful;
}
template <typename T>
- void TestAttributeIsDefined(const RTCStatsMember<T>& field) {
+ void TestAttributeIsDefined(const absl::optional<T>& field) {
EXPECT_TRUE(field.has_value())
<< stats_->type() << "." << stats_->GetAttribute(field).name() << "["
<< stats_->id() << "] was undefined.";
@@ -227,7 +228,7 @@
}
template <typename T>
- void TestAttributeIsUndefined(const RTCStatsMember<T>& field) {
+ void TestAttributeIsUndefined(const absl::optional<T>& field) {
Attribute attribute = stats_->GetAttribute(field);
EXPECT_FALSE(field.has_value())
<< stats_->type() << "." << attribute.name() << "[" << stats_->id()
@@ -236,7 +237,7 @@
}
template <typename T>
- void TestAttributeIsPositive(const RTCStatsMember<T>& field) {
+ void TestAttributeIsPositive(const absl::optional<T>& field) {
Attribute attribute = stats_->GetAttribute(field);
EXPECT_TRUE(field.has_value()) << stats_->type() << "." << attribute.name()
<< "[" << stats_->id() << "] was undefined.";
@@ -252,7 +253,7 @@
}
template <typename T>
- void TestAttributeIsNonNegative(const RTCStatsMember<T>& field) {
+ void TestAttributeIsNonNegative(const absl::optional<T>& field) {
Attribute attribute = stats_->GetAttribute(field);
EXPECT_TRUE(field.has_value()) << stats_->type() << "." << attribute.name()
<< "[" << stats_->id() << "] was undefined.";
@@ -268,13 +269,13 @@
}
template <typename T>
- void TestAttributeIsIDReference(const RTCStatsMember<T>& field,
+ void TestAttributeIsIDReference(const absl::optional<T>& field,
const char* expected_type) {
TestAttributeIsIDReference(field, expected_type, false);
}
template <typename T>
- void TestAttributeIsOptionalIDReference(const RTCStatsMember<T>& field,
+ void TestAttributeIsOptionalIDReference(const absl::optional<T>& field,
const char* expected_type) {
TestAttributeIsIDReference(field, expected_type, true);
}
@@ -291,7 +292,7 @@
private:
template <typename T>
- void TestAttributeIsIDReference(const RTCStatsMember<T>& field,
+ void TestAttributeIsIDReference(const absl::optional<T>& field,
const char* expected_type,
bool optional) {
if (optional && !field.has_value()) {
diff --git a/pc/rtc_stats_traversal.cc b/pc/rtc_stats_traversal.cc
index dfd0570..e9d11b5 100644
--- a/pc/rtc_stats_traversal.cc
+++ b/pc/rtc_stats_traversal.cc
@@ -15,6 +15,7 @@
#include <utility>
#include <vector>
+#include "absl/types/optional.h"
#include "api/stats/rtcstats_objects.h"
#include "rtc_base/checks.h"
@@ -42,7 +43,7 @@
}
}
-void AddIdIfDefined(const RTCStatsMember<std::string>& id,
+void AddIdIfDefined(const absl::optional<std::string>& id,
std::vector<const std::string*>* neighbor_ids) {
if (id.has_value())
neighbor_ids->push_back(&(*id));
diff --git a/stats/BUILD.gn b/stats/BUILD.gn
index 76edc44..1ca584d 100644
--- a/stats/BUILD.gn
+++ b/stats/BUILD.gn
@@ -45,6 +45,7 @@
"../rtc_base:checks",
"../rtc_base/system:rtc_export",
]
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
if (rtc_include_tests && !build_with_chromium) {
@@ -74,5 +75,7 @@
"//testing/android/native_test:native_test_support",
]
}
+
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
}
diff --git a/stats/rtc_stats_report_unittest.cc b/stats/rtc_stats_report_unittest.cc
index b3ac0a2..f11ff52 100644
--- a/stats/rtc_stats_report_unittest.cc
+++ b/stats/rtc_stats_report_unittest.cc
@@ -10,6 +10,7 @@
#include "api/stats/rtc_stats_report.h"
+#include "absl/types/optional.h"
#include "api/stats/attribute.h"
#include "api/stats/rtc_stats.h"
#include "rtc_base/checks.h"
@@ -24,7 +25,7 @@
RTCTestStats1(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- RTCStatsMember<int32_t> integer;
+ absl::optional<int32_t> integer;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats1,
@@ -39,7 +40,7 @@
RTCTestStats2(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- RTCStatsMember<double> number;
+ absl::optional<double> number;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats2,
@@ -54,7 +55,7 @@
RTCTestStats3(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- RTCStatsMember<std::string> string;
+ absl::optional<std::string> string;
};
WEBRTC_RTCSTATS_IMPL(RTCTestStats3,
diff --git a/stats/rtc_stats_unittest.cc b/stats/rtc_stats_unittest.cc
index 1098f04..555360f 100644
--- a/stats/rtc_stats_unittest.cc
+++ b/stats/rtc_stats_unittest.cc
@@ -15,6 +15,7 @@
#include <cstring>
#include <iostream>
+#include "absl/types/optional.h"
#include "rtc_base/checks.h"
#include "rtc_base/strings/json.h"
#include "stats/test/rtc_test_stats.h"
@@ -46,7 +47,7 @@
RTCChildStats(const std::string& id, Timestamp timestamp)
: RTCStats(id, timestamp) {}
- RTCStatsMember<int32_t> child_int;
+ absl::optional<int32_t> child_int;
};
WEBRTC_RTCSTATS_IMPL(RTCChildStats,
@@ -61,7 +62,7 @@
RTCGrandChildStats(const std::string& id, Timestamp timestamp)
: RTCChildStats(id, timestamp) {}
- RTCStatsMember<int32_t> grandchild_int;
+ absl::optional<int32_t> grandchild_int;
};
WEBRTC_RTCSTATS_IMPL(RTCGrandChildStats,
diff --git a/stats/test/rtc_test_stats.h b/stats/test/rtc_test_stats.h
index 05c0904..ff1a06f 100644
--- a/stats/test/rtc_test_stats.h
+++ b/stats/test/rtc_test_stats.h
@@ -16,6 +16,7 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
#include "api/stats/rtc_stats.h"
#include "rtc_base/system/rtc_export.h"
@@ -27,22 +28,22 @@
RTCTestStats(const std::string& id, Timestamp timestamp);
~RTCTestStats() override;
- RTCStatsMember<bool> m_bool;
- RTCStatsMember<int32_t> m_int32;
- RTCStatsMember<uint32_t> m_uint32;
- RTCStatsMember<int64_t> m_int64;
- RTCStatsMember<uint64_t> m_uint64;
- RTCStatsMember<double> m_double;
- RTCStatsMember<std::string> m_string;
- RTCStatsMember<std::vector<bool>> m_sequence_bool;
- RTCStatsMember<std::vector<int32_t>> m_sequence_int32;
- RTCStatsMember<std::vector<uint32_t>> m_sequence_uint32;
- RTCStatsMember<std::vector<int64_t>> m_sequence_int64;
- RTCStatsMember<std::vector<uint64_t>> m_sequence_uint64;
- RTCStatsMember<std::vector<double>> m_sequence_double;
- RTCStatsMember<std::vector<std::string>> m_sequence_string;
- RTCStatsMember<std::map<std::string, uint64_t>> m_map_string_uint64;
- RTCStatsMember<std::map<std::string, double>> m_map_string_double;
+ absl::optional<bool> m_bool;
+ absl::optional<int32_t> m_int32;
+ absl::optional<uint32_t> m_uint32;
+ absl::optional<int64_t> m_int64;
+ absl::optional<uint64_t> m_uint64;
+ absl::optional<double> m_double;
+ absl::optional<std::string> m_string;
+ absl::optional<std::vector<bool>> m_sequence_bool;
+ absl::optional<std::vector<int32_t>> m_sequence_int32;
+ absl::optional<std::vector<uint32_t>> m_sequence_uint32;
+ absl::optional<std::vector<int64_t>> m_sequence_int64;
+ absl::optional<std::vector<uint64_t>> m_sequence_uint64;
+ absl::optional<std::vector<double>> m_sequence_double;
+ absl::optional<std::vector<std::string>> m_sequence_string;
+ absl::optional<std::map<std::string, uint64_t>> m_map_string_uint64;
+ absl::optional<std::map<std::string, double>> m_map_string_double;
};
} // namespace webrtc
