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webrtc/api/stats/rtcstats_objects.h
Gustaf Ullberg b0a0207838 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 
Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay

Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}
2017-10-02 10:47:00 +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 <string>
#include <vector>
#include "api/stats/rtcstats.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/#certificatestats-dict*
class 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 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 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 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
class 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> ip;
RTCStatsMember<int32_t> port;
RTCStatsMember<std::string> 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*
class 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);
const char* type() const override;
};
class 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);
const char* type() const override;
};
// https://w3c.github.io/webrtc-stats/#msstats-dict*
// TODO(hbos): Tracking bug crbug.com/660827
class 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 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<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;
// 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> 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> concealment_events;
};
// https://w3c.github.io/webrtc-stats/#pcstats-dict*
class 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 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;
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 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> bytes_received;
RTCStatsMember<uint32_t> packets_lost;
// 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://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
// TODO(hbos): Support the remote case |is_remote = true|.
// https://bugs.webrtc.org/7066
class 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<uint32_t> packets_sent;
RTCStatsMember<uint64_t> bytes_sent;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7066
RTCStatsMember<double> target_bitrate;
RTCStatsMember<uint32_t> frames_encoded;
};
// https://w3c.github.io/webrtc-stats/#transportstats-dict*
class 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_
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