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webrtc_m130/api/stats/rtcstats_objects.h
Philipp Hancke 0359ba2225 stats: add frame assembly time stats 
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)

This is similar to totalProcessingDelay
  https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.

This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.

Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as

totalAssemblyTime of type double
	Only exists for video. 	The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
    Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.

    This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.

framesAssembledFromMultiplePacket of type unsigned long
	Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
	For such frames the totalAssemblyTime is incremented.

BUG=webrtc:13986

Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36922}
2022-05-18 09:16:10 +00:00

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/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef API_STATS_RTCSTATS_OBJECTS_H_
#define API_STATS_RTCSTATS_OBJECTS_H_
#include <stdint.h>
#include <map>
#include <memory>
#include <string>
#include <vector>
#include "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/#dom-rtcdtlsrole
struct RTCDtlsRole {
static const char* const kUnknown;
static const char* const kClient;
static const char* const kServer;
};
// https://www.w3.org/TR/webrtc/#rtcicerole
struct RTCIceRole {
static const char* const kUnknown;
static const char* const kControlled;
static const char* const kControlling;
};
// https://www.w3.org/TR/webrtc/#dom-rtcicetransportstate
struct RTCIceTransportState {
static const char* const kNew;
static const char* const kChecking;
static const char* const kConnected;
static const char* const kCompleted;
static const char* const kDisconnected;
static const char* const kFailed;
static const char* const kClosed;
};
// 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;
};
// Non standard extension mapping to rtc::AdapterType
struct RTCNetworkAdapterType {
static constexpr char kUnknown[] = "unknown";
static constexpr char kEthernet[] = "ethernet";
static constexpr char kWifi[] = "wifi";
static constexpr char kCellular[] = "cellular";
static constexpr char kLoopback[] = "loopback";
static constexpr char kAny[] = "any";
static constexpr char kCellular2g[] = "cellular2g";
static constexpr char kCellular3g[] = "cellular3g";
static constexpr char kCellular4g[] = "cellular4g";
static constexpr char kCellular5g[] = "cellular5g";
};
// 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<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;
};
// 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> data_channel_identifier;
// 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;
// Obsolete: priority
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> 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;
// 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;
RTCStatsMember<uint64_t> packets_discarded_on_send;
RTCStatsMember<uint64_t> bytes_discarded_on_send;
};
// 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;
// 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;
// TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
RTCStatsMember<std::string> candidate_type;
RTCStatsMember<int32_t> priority;
RTCStatsMember<std::string> url;
RTCNonStandardStatsMember<bool> vpn;
RTCNonStandardStatsMember<std::string> network_adapter_type;
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;
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;
// Non-standard metric showing target delay of jitter buffer.
// This value is increased by the target jitter buffer delay every time a
// sample is emitted by the jitter buffer. The added target is the target
// delay, in seconds, at the time that the sample was emitted from the jitter
// buffer. (https://github.com/w3c/webrtc-provisional-stats/pull/20)
// Currently it is implemented only for audio.
// TODO(titovartem) implement for video streams when will be requested.
RTCNonStandardStatsMember<double> jitter_buffer_target_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;
RTCStatsMember<std::string> kind;
// Obsolete: track_id
RTCStatsMember<std::string> track_id;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> codec_id;
// Obsolete
RTCStatsMember<std::string> media_type; // renamed to kind.
protected:
RTCRTPStreamStats(const std::string& id, int64_t timestamp_us);
RTCRTPStreamStats(std::string&& id, int64_t timestamp_us);
};
// https://www.w3.org/TR/webrtc-stats/#receivedrtpstats-dict*
class RTC_EXPORT RTCReceivedRtpStreamStats : public RTCRTPStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCReceivedRtpStreamStats(const RTCReceivedRtpStreamStats& other);
~RTCReceivedRtpStreamStats() override;
// TODO(hbos) The following fields need to be added and migrated
// both from RTCInboundRtpStreamStats and RTCRemoteInboundRtpStreamStats:
// packetsReceived, packetsRepaired, burstPacketsLost,
// burstPacketDiscarded, burstLossCount, burstDiscardCount, burstLossRate,
// burstDiscardRate, gapLossRate, gapDiscardRate, framesDropped,
// partialFramesLost, fullFramesLost
// crbug.com/webrtc/12532
RTCStatsMember<double> jitter;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<uint64_t> packets_discarded;
protected:
RTCReceivedRtpStreamStats(const std::string&& id, int64_t timestamp_us);
RTCReceivedRtpStreamStats(std::string&& id, int64_t timestamp_us);
};
// https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict*
class RTC_EXPORT RTCSentRtpStreamStats : public RTCRTPStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCSentRtpStreamStats(const RTCSentRtpStreamStats& other);
~RTCSentRtpStreamStats() override;
RTCStatsMember<uint32_t> packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
protected:
RTCSentRtpStreamStats(const std::string&& id, int64_t timestamp_us);
RTCSentRtpStreamStats(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 RTCReceivedRtpStreamStats {
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<std::string> remote_id;
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<double> last_packet_received_timestamp;
RTCStatsMember<double> jitter_buffer_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;
// 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_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;
// Stats below are only implemented or defined for video.
RTCStatsMember<int32_t> frames_received;
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;
RTCStatsMember<uint32_t> frame_bit_depth;
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;
// TODO(bugs.webrtc.org/13986): standardize
RTCNonStandardStatsMember<double> total_assembly_time;
RTCNonStandardStatsMember<uint32_t> frames_assembled_from_multiple_packets;
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;
// 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;
};
// 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<std::string> rid;
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(https://crbug.com/webrtc/13394): Also collect this metric for video.
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;
// 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
RTCStatsMember<std::string> quality_limitation_reason;
RTCStatsMember<std::map<std::string, double>> quality_limitation_durations;
// 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;
// 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;
};
// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
class RTC_EXPORT RTCRemoteInboundRtpStreamStats final
: public RTCReceivedRtpStreamStats {
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;
// TODO(hbos): The following RTCReceivedRtpStreamStats metrics should also be
// implemented: packetsReceived, packetsRepaired,
// burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount,
// burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate.
// RTCRemoteInboundRtpStreamStats
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;
};
// https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
class RTC_EXPORT RTCRemoteOutboundRtpStreamStats final
: public RTCSentRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCRemoteOutboundRtpStreamStats(const std::string& id, int64_t timestamp_us);
RTCRemoteOutboundRtpStreamStats(std::string&& id, int64_t timestamp_us);
RTCRemoteOutboundRtpStreamStats(const RTCRemoteOutboundRtpStreamStats& other);
~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;
};
// 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;
RTCStatsMember<double> echo_return_loss;
RTCStatsMember<double> echo_return_loss_enhancement;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats
class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCVideoSourceStats(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;
RTCStatsMember<uint32_t> frames;
RTCStatsMember<double> 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> packets_sent;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<uint64_t> packets_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> 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;
};
} // namespace webrtc
#endif // API_STATS_RTCSTATS_OBJECTS_H_
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