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webrtc/api/stats/rtcstats_objects.h
Henrik Boström 1df1bf8551 PeerConnectionInterface::GetStats() with selector argument added. 
This exposes the stats selection algorithm[1] on the PeerConnection.

Per-spec, there are four flavors of getStats():
1. RTCPeerConnection.getStats().
2. RTCPeerConnection.getStats(MediaStreamTrack selector).
3. RTCRtpSender.getStats().
4. RTCRtpReceiver.getStats().

1) is the parameterless getStats() which is already shipped.
2) is the same as 3) and 4) except the track is used to look up the
corresponding sender/receiver to use as the selector.
3) and 4) perform stats collection with a filter, which is implemented
in RTCStatsCollector.GetStatsReport(selector).

For technical reasons, it is easier to place GetStats() on the
PeerConnection where the RTCStatsCollector lives than to place it on the
sender/receiver. Passing the selector as an argument or as a "this"
makes little difference other than style. Wiring Chrome up such that the
JavaScript APIs is like the spec is trivial after GetStats() is added to
PeerConnectionInterface.

This CL also adds comments documenting our intent to deprecate and
remove the legacy GetStats() APIs some time in the future.

[1] https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm

Bug: chromium:680172
Change-Id: I09316ba6f20b25d4f9c11785d0a1a1262d6062a1
Reviewed-on: https://webrtc-review.googlesource.com/62900
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22602}
2018-03-26 12:08:20 +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 <memory>
#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/#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/#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
// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
// them in the new PeerConnection::GetStats.
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> network_type;
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*
// This forces us to have to override copy() and type().
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);
std::unique_ptr<RTCStats> copy() const override;
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);
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 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> 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> 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<int32_t> packets_lost; // Signed per RFC 3550
// 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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