mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 13:50:40 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/api/stats/rtcstats_objects.h
Henrik Boström 646fda0212 Implement RTCMediaSourceStats and friends in standard getStats(). 
This implements RTCAudioSourceStats and RTCVideoSourceStats, both
inheriting from abstract dictionary RTCMediaSourceStats:
https://w3c.github.io/webrtc-stats/#dom-rtcmediasourcestats

All members are implemented except for the total "frames" counter:
- trackIdentifier
- kind
- width
- height
- framesPerSecond

This means to make googFrameWidthInput, googFrameHeightInput and
googFrameRateInput obsolete.

Implemented using the same code path as the goog stats, there are
some minor bugs that should be fixed in the future, but not this CL:
1. We create media-source objects on a per-track attachment basis.
   If the same track is attached multiple times this results in
   multiple media-source objects, but the spec says it should be on a
   per-source basis.
2. framesPerSecond is only calculated after connecting (when we have a
   sender with SSRC), but if collected on a per-source basis the source
   should be able to tell us the FPS whether or not we are sending it.

Bug: webrtc:10453
Change-Id: I23705a79f15075dca2536275934af1904a7f0d39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137804
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28028}
2019-05-22 16:03:41 +00:00

537 lines
22 KiB
C++
Raw Blame History

/*
* Copyright 2016 The WebRTC Project Authors. All rights reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef API_STATS_RTCSTATS_OBJECTS_H_
#define API_STATS_RTCSTATS_OBJECTS_H_
#include <stdint.h>
#include <memory>
#include <string>
#include <vector>
#include "api/stats/rtc_stats.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
// https://w3c.github.io/webrtc-pc/#idl-def-rtcdatachannelstate
struct RTCDataChannelState {
static const char* const kConnecting;
static const char* const kOpen;
static const char* const kClosing;
static const char* const kClosed;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcstatsicecandidatepairstate
struct RTCStatsIceCandidatePairState {
static const char* const kFrozen;
static const char* const kWaiting;
static const char* const kInProgress;
static const char* const kFailed;
static const char* const kSucceeded;
};
// https://w3c.github.io/webrtc-pc/#rtcicecandidatetype-enum
struct RTCIceCandidateType {
static const char* const kHost;
static const char* const kSrflx;
static const char* const kPrflx;
static const char* const kRelay;
};
// https://w3c.github.io/webrtc-pc/#idl-def-rtcdtlstransportstate
struct RTCDtlsTransportState {
static const char* const kNew;
static const char* const kConnecting;
static const char* const kConnected;
static const char* const kClosed;
static const char* const kFailed;
};
// |RTCMediaStreamTrackStats::kind| is not an enum in the spec but the only
// valid values are "audio" and "video".
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-kind
struct RTCMediaStreamTrackKind {
static const char* const kAudio;
static const char* const kVideo;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcnetworktype
struct RTCNetworkType {
static const char* const kBluetooth;
static const char* const kCellular;
static const char* const kEthernet;
static const char* const kWifi;
static const char* const kWimax;
static const char* const kVpn;
static const char* const kUnknown;
};
// https://webrtc.org/experiments/rtp-hdrext/video-content-type/
struct RTCContentType {
static const char* const kUnspecified;
static const char* const kScreenshare;
};
// https://w3c.github.io/webrtc-stats/#certificatestats-dict*
class RTC_EXPORT RTCCertificateStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCCertificateStats(const std::string& id, int64_t timestamp_us);
RTCCertificateStats(std::string&& id, int64_t timestamp_us);
RTCCertificateStats(const RTCCertificateStats& other);
~RTCCertificateStats() override;
RTCStatsMember<std::string> fingerprint;
RTCStatsMember<std::string> fingerprint_algorithm;
RTCStatsMember<std::string> base64_certificate;
RTCStatsMember<std::string> issuer_certificate_id;
};
// https://w3c.github.io/webrtc-stats/#codec-dict*
class RTC_EXPORT RTCCodecStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCCodecStats(const std::string& id, int64_t timestamp_us);
RTCCodecStats(std::string&& id, int64_t timestamp_us);
RTCCodecStats(const RTCCodecStats& other);
~RTCCodecStats() override;
RTCStatsMember<uint32_t> payload_type;
RTCStatsMember<std::string> mime_type;
RTCStatsMember<uint32_t> clock_rate;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
RTCStatsMember<uint32_t> channels;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
RTCStatsMember<std::string> sdp_fmtp_line;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7061
RTCStatsMember<std::string> implementation;
};
// https://w3c.github.io/webrtc-stats/#dcstats-dict*
class RTC_EXPORT RTCDataChannelStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCDataChannelStats(const std::string& id, int64_t timestamp_us);
RTCDataChannelStats(std::string&& id, int64_t timestamp_us);
RTCDataChannelStats(const RTCDataChannelStats& other);
~RTCDataChannelStats() override;
RTCStatsMember<std::string> label;
RTCStatsMember<std::string> protocol;
RTCStatsMember<int32_t> datachannelid;
// TODO(hbos): Support enum types? "RTCStatsMember<RTCDataChannelState>"?
RTCStatsMember<std::string> state;
RTCStatsMember<uint32_t> messages_sent;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint32_t> messages_received;
RTCStatsMember<uint64_t> bytes_received;
};
// https://w3c.github.io/webrtc-stats/#candidatepair-dict*
// TODO(hbos): Tracking bug https://bugs.webrtc.org/7062
class RTC_EXPORT RTCIceCandidatePairStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCIceCandidatePairStats(const std::string& id, int64_t timestamp_us);
RTCIceCandidatePairStats(std::string&& id, int64_t timestamp_us);
RTCIceCandidatePairStats(const RTCIceCandidatePairStats& other);
~RTCIceCandidatePairStats() override;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> local_candidate_id;
RTCStatsMember<std::string> remote_candidate_id;
// TODO(hbos): Support enum types?
// "RTCStatsMember<RTCStatsIceCandidatePairState>"?
RTCStatsMember<std::string> state;
RTCStatsMember<uint64_t> priority;
RTCStatsMember<bool> nominated;
// TODO(hbos): Collect this the way the spec describes it. We have a value for
// it but it is not spec-compliant. https://bugs.webrtc.org/7062
RTCStatsMember<bool> writable;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<bool> readable;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<double> total_round_trip_time;
RTCStatsMember<double> current_round_trip_time;
RTCStatsMember<double> available_outgoing_bitrate;
// TODO(hbos): Populate this value. It is wired up and collected the same way
// "VideoBwe.googAvailableReceiveBandwidth" is, but that value is always
// undefined. https://bugs.webrtc.org/7062
RTCStatsMember<double> available_incoming_bitrate;
RTCStatsMember<uint64_t> requests_received;
RTCStatsMember<uint64_t> requests_sent;
RTCStatsMember<uint64_t> responses_received;
RTCStatsMember<uint64_t> responses_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> retransmissions_received;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> retransmissions_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> consent_requests_received;
RTCStatsMember<uint64_t> consent_requests_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> consent_responses_received;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7062
RTCStatsMember<uint64_t> consent_responses_sent;
};
// https://w3c.github.io/webrtc-stats/#icecandidate-dict*
// TODO(hbos): |RTCStatsCollector| only collects candidates that are part of
// ice candidate pairs, but there could be candidates not paired with anything.
// crbug.com/632723
// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
// them in the new PeerConnection::GetStats.
class RTC_EXPORT RTCIceCandidateStats : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCIceCandidateStats(const RTCIceCandidateStats& other);
~RTCIceCandidateStats() override;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<bool> is_remote;
RTCStatsMember<std::string> network_type;
RTCStatsMember<std::string> ip;
RTCStatsMember<int32_t> port;
RTCStatsMember<std::string> protocol;
RTCStatsMember<std::string> relay_protocol;
// TODO(hbos): Support enum types? "RTCStatsMember<RTCIceCandidateType>"?
RTCStatsMember<std::string> candidate_type;
RTCStatsMember<int32_t> priority;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/632723
RTCStatsMember<std::string> url;
// TODO(hbos): |deleted = true| case is not supported by |RTCStatsCollector|.
// crbug.com/632723
RTCStatsMember<bool> deleted; // = false
protected:
RTCIceCandidateStats(const std::string& id,
int64_t timestamp_us,
bool is_remote);
RTCIceCandidateStats(std::string&& id, int64_t timestamp_us, bool is_remote);
};
// In the spec both local and remote varieties are of type RTCIceCandidateStats.
// But here we define them as subclasses of |RTCIceCandidateStats| because the
// |kType| need to be different ("RTCStatsType type") in the local/remote case.
// https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
// This forces us to have to override copy() and type().
class RTC_EXPORT RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
public:
static const char kType[];
RTCLocalIceCandidateStats(const std::string& id, int64_t timestamp_us);
RTCLocalIceCandidateStats(std::string&& id, int64_t timestamp_us);
std::unique_ptr<RTCStats> copy() const override;
const char* type() const override;
};
class RTC_EXPORT RTCRemoteIceCandidateStats final
: public RTCIceCandidateStats {
public:
static const char kType[];
RTCRemoteIceCandidateStats(const std::string& id, int64_t timestamp_us);
RTCRemoteIceCandidateStats(std::string&& id, int64_t timestamp_us);
std::unique_ptr<RTCStats> copy() const override;
const char* type() const override;
};
// https://w3c.github.io/webrtc-stats/#msstats-dict*
// TODO(hbos): Tracking bug crbug.com/660827
class RTC_EXPORT RTCMediaStreamStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCMediaStreamStats(const std::string& id, int64_t timestamp_us);
RTCMediaStreamStats(std::string&& id, int64_t timestamp_us);
RTCMediaStreamStats(const RTCMediaStreamStats& other);
~RTCMediaStreamStats() override;
RTCStatsMember<std::string> stream_identifier;
RTCStatsMember<std::vector<std::string>> track_ids;
};
// https://w3c.github.io/webrtc-stats/#mststats-dict*
// TODO(hbos): Tracking bug crbug.com/659137
class RTC_EXPORT RTCMediaStreamTrackStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCMediaStreamTrackStats(const std::string& id,
int64_t timestamp_us,
const char* kind);
RTCMediaStreamTrackStats(std::string&& id,
int64_t timestamp_us,
const char* kind);
RTCMediaStreamTrackStats(const RTCMediaStreamTrackStats& other);
~RTCMediaStreamTrackStats() override;
RTCStatsMember<std::string> track_identifier;
RTCStatsMember<std::string> media_source_id;
RTCStatsMember<bool> remote_source;
RTCStatsMember<bool> ended;
// TODO(hbos): |RTCStatsCollector| does not return stats for detached tracks.
// crbug.com/659137
RTCStatsMember<bool> detached;
// See |RTCMediaStreamTrackKind| for valid values.
RTCStatsMember<std::string> kind;
// TODO(gustaf): Implement jitter_buffer_delay for video (currently
// implemented for audio only).
// https://crbug.com/webrtc/8318
RTCStatsMember<double> jitter_buffer_delay;
RTCStatsMember<uint64_t> jitter_buffer_emitted_count;
// Video-only members
RTCStatsMember<uint32_t> frame_width;
RTCStatsMember<uint32_t> frame_height;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
RTCStatsMember<double> frames_per_second;
RTCStatsMember<uint32_t> frames_sent;
RTCStatsMember<uint32_t> huge_frames_sent;
RTCStatsMember<uint32_t> frames_received;
RTCStatsMember<uint32_t> frames_decoded;
RTCStatsMember<uint32_t> frames_dropped;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
RTCStatsMember<uint32_t> frames_corrupted;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
RTCStatsMember<uint32_t> partial_frames_lost;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
RTCStatsMember<uint32_t> full_frames_lost;
// Audio-only members
RTCStatsMember<double> audio_level;
RTCStatsMember<double> total_audio_energy;
RTCStatsMember<double> echo_return_loss;
RTCStatsMember<double> echo_return_loss_enhancement;
RTCStatsMember<uint64_t> total_samples_received;
RTCStatsMember<double> total_samples_duration;
RTCStatsMember<uint64_t> concealed_samples;
RTCStatsMember<uint64_t> silent_concealed_samples;
RTCStatsMember<uint64_t> concealment_events;
RTCStatsMember<uint64_t> inserted_samples_for_deceleration;
RTCStatsMember<uint64_t> removed_samples_for_acceleration;
// Non-standard audio-only member
// TODO(kuddai): Add description to standard. crbug.com/webrtc/10042
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
// TODO(henrik.lundin): Add description of the interruption metrics at
// https://github.com/henbos/webrtc-provisional-stats/issues/17
RTCNonStandardStatsMember<uint32_t> interruption_count;
RTCNonStandardStatsMember<double> total_interruption_duration;
// Non-standard video-only members.
// https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
RTCNonStandardStatsMember<uint32_t> freeze_count;
RTCNonStandardStatsMember<uint32_t> pause_count;
RTCNonStandardStatsMember<double> total_freezes_duration;
RTCNonStandardStatsMember<double> total_pauses_duration;
RTCNonStandardStatsMember<double> total_frames_duration;
RTCNonStandardStatsMember<double> sum_squared_frame_durations;
};
// https://w3c.github.io/webrtc-stats/#pcstats-dict*
class RTC_EXPORT RTCPeerConnectionStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCPeerConnectionStats(const std::string& id, int64_t timestamp_us);
RTCPeerConnectionStats(std::string&& id, int64_t timestamp_us);
RTCPeerConnectionStats(const RTCPeerConnectionStats& other);
~RTCPeerConnectionStats() override;
RTCStatsMember<uint32_t> data_channels_opened;
RTCStatsMember<uint32_t> data_channels_closed;
};
// https://w3c.github.io/webrtc-stats/#streamstats-dict*
// TODO(hbos): Tracking bug crbug.com/657854
class RTC_EXPORT RTCRTPStreamStats : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCRTPStreamStats(const RTCRTPStreamStats& other);
~RTCRTPStreamStats() override;
RTCStatsMember<uint32_t> ssrc;
// TODO(hbos): When the remote case is supported |RTCStatsCollector| needs to
// set this. crbug.com/657855, 657856
RTCStatsMember<std::string> associate_stats_id;
// TODO(hbos): Remote case not supported by |RTCStatsCollector|.
// crbug.com/657855, 657856
RTCStatsMember<bool> is_remote; // = false
RTCStatsMember<std::string> media_type; // renamed to kind.
RTCStatsMember<std::string> kind;
RTCStatsMember<std::string> track_id;
RTCStatsMember<std::string> transport_id;
RTCStatsMember<std::string> codec_id;
// FIR and PLI counts are only defined for |media_type == "video"|.
RTCStatsMember<uint32_t> fir_count;
RTCStatsMember<uint32_t> pli_count;
// TODO(hbos): NACK count should be collected by |RTCStatsCollector| for both
// audio and video but is only defined in the "video" case. crbug.com/657856
RTCStatsMember<uint32_t> nack_count;
// TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/657854
// SLI count is only defined for |media_type == "video"|.
RTCStatsMember<uint32_t> sli_count;
RTCStatsMember<uint64_t> qp_sum;
protected:
RTCRTPStreamStats(const std::string& id, int64_t timestamp_us);
RTCRTPStreamStats(std::string&& id, int64_t timestamp_us);
};
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
// TODO(hbos): Support the remote case |is_remote = true|.
// https://bugs.webrtc.org/7065
class RTC_EXPORT RTCInboundRTPStreamStats final : public RTCRTPStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCInboundRTPStreamStats(const std::string& id, int64_t timestamp_us);
RTCInboundRTPStreamStats(std::string&& id, int64_t timestamp_us);
RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
~RTCInboundRTPStreamStats() override;
RTCStatsMember<uint32_t> packets_received;
RTCStatsMember<uint64_t> fec_packets_received;
RTCStatsMember<uint64_t> fec_packets_discarded;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
RTCStatsMember<double> last_packet_received_timestamp;
// TODO(hbos): Collect and populate this value for both "audio" and "video",
// currently not collected for "video". https://bugs.webrtc.org/7065
RTCStatsMember<double> jitter;
RTCStatsMember<double> fraction_lost;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> round_trip_time;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> packets_discarded;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> packets_repaired;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_packets_lost;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_packets_discarded;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_loss_count;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_discard_count;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> burst_loss_rate;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> burst_discard_rate;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> gap_loss_rate;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> gap_discard_rate;
RTCStatsMember<uint32_t> frames_decoded;
// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcinboundrtpstreamstats-contenttype
RTCStatsMember<std::string> content_type;
};
// 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<uint32_t> packets_sent;
RTCStatsMember<uint64_t> retransmitted_packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint64_t> retransmitted_bytes_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
RTCStatsMember<double> target_bitrate;
RTCStatsMember<uint32_t> frames_encoded;
RTCStatsMember<double> total_encode_time;
RTCStatsMember<uint64_t> total_encoded_bytes_target;
// TODO(https://crbug.com/webrtc/10635): This is only implemented for video;
// implement it for audio as well.
RTCStatsMember<double> total_packet_send_delay;
// https://henbos.github.io/webrtc-provisional-stats/#dom-rtcoutboundrtpstreamstats-contenttype
RTCStatsMember<std::string> content_type;
};
// 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;
};
// https://w3c.github.io/webrtc-stats/#dom-rtcvideosourcestats
class RTC_EXPORT RTCVideoSourceStats final : public RTCMediaSourceStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCVideoSourceStats(const std::string& id, int64_t timestamp_us);
RTCVideoSourceStats(std::string&& id, int64_t timestamp_us);
RTCVideoSourceStats(const RTCVideoSourceStats& other);
~RTCVideoSourceStats() override;
RTCStatsMember<uint32_t> width;
RTCStatsMember<uint32_t> height;
// TODO(hbos): Implement this metric.
RTCStatsMember<uint32_t> frames;
RTCStatsMember<uint32_t> frames_per_second;
};
// https://w3c.github.io/webrtc-stats/#transportstats-dict*
class RTC_EXPORT RTCTransportStats final : public RTCStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCTransportStats(const std::string& id, int64_t timestamp_us);
RTCTransportStats(std::string&& id, int64_t timestamp_us);
RTCTransportStats(const RTCTransportStats& other);
~RTCTransportStats() override;
RTCStatsMember<uint64_t> bytes_sent;
RTCStatsMember<uint64_t> bytes_received;
RTCStatsMember<std::string> rtcp_transport_stats_id;
// TODO(hbos): Support enum types? "RTCStatsMember<RTCDtlsTransportState>"?
RTCStatsMember<std::string> dtls_state;
RTCStatsMember<std::string> selected_candidate_pair_id;
RTCStatsMember<std::string> local_certificate_id;
RTCStatsMember<std::string> remote_certificate_id;
};
} // namespace webrtc
#endif // API_STATS_RTCSTATS_OBJECTS_H_
Reference in a new issue View git blame Copy permalink