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302 lines
13 KiB
302 lines
13 KiB
/*
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* Copyright 2016 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_RTC_STATS_COLLECTOR_H_
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#define PC_RTC_STATS_COLLECTOR_H_
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/scoped_refptr.h"
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#include "api/stats/rtc_stats_collector_callback.h"
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#include "api/stats/rtc_stats_report.h"
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#include "api/stats/rtcstats_objects.h"
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#include "call/call.h"
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#include "media/base/media_channel.h"
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#include "pc/data_channel_utils.h"
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#include "pc/peer_connection_internal.h"
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#include "pc/track_media_info_map.h"
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#include "rtc_base/event.h"
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#include "rtc_base/ref_count.h"
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#include "rtc_base/ssl_identity.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/time_utils.h"
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namespace webrtc {
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class RtpSenderInternal;
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class RtpReceiverInternal;
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// All public methods of the collector are to be called on the signaling thread.
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// Stats are gathered on the signaling, worker and network threads
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// asynchronously. The callback is invoked on the signaling thread. Resulting
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// reports are cached for |cache_lifetime_| ms.
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class RTCStatsCollector : public virtual rtc::RefCountInterface,
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public sigslot::has_slots<> {
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public:
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static rtc::scoped_refptr<RTCStatsCollector> Create(
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PeerConnectionInternal* pc,
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int64_t cache_lifetime_us = 50 * rtc::kNumMicrosecsPerMillisec);
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// Gets a recent stats report. If there is a report cached that is still fresh
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// it is returned, otherwise new stats are gathered and returned. A report is
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// considered fresh for |cache_lifetime_| ms. const RTCStatsReports are safe
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// to use across multiple threads and may be destructed on any thread.
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// If the optional selector argument is used, stats are filtered according to
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// stats selection algorithm before delivery.
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// https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
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void GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// If |selector| is null the selection algorithm is still applied (interpreted
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// as: no RTP streams are sent by selector). The result is empty.
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void GetStatsReport(rtc::scoped_refptr<RtpSenderInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// If |selector| is null the selection algorithm is still applied (interpreted
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// as: no RTP streams are received by selector). The result is empty.
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void GetStatsReport(rtc::scoped_refptr<RtpReceiverInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// Clears the cache's reference to the most recent stats report. Subsequently
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// calling |GetStatsReport| guarantees fresh stats.
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void ClearCachedStatsReport();
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// If there is a |GetStatsReport| requests in-flight, waits until it has been
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// completed. Must be called on the signaling thread.
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void WaitForPendingRequest();
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protected:
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RTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime_us);
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~RTCStatsCollector();
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struct CertificateStatsPair {
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std::unique_ptr<rtc::SSLCertificateStats> local;
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std::unique_ptr<rtc::SSLCertificateStats> remote;
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};
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// Stats gathering on a particular thread. Virtual for the sake of testing.
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virtual void ProducePartialResultsOnSignalingThreadImpl(
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int64_t timestamp_us,
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RTCStatsReport* partial_report);
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virtual void ProducePartialResultsOnNetworkThreadImpl(
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int64_t timestamp_us,
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name,
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const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
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RTCStatsReport* partial_report);
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private:
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class RequestInfo {
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public:
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enum class FilterMode { kAll, kSenderSelector, kReceiverSelector };
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// Constructs with FilterMode::kAll.
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explicit RequestInfo(
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// Constructs with FilterMode::kSenderSelector. The selection algorithm is
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// applied even if |selector| is null, resulting in an empty report.
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RequestInfo(rtc::scoped_refptr<RtpSenderInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// Constructs with FilterMode::kReceiverSelector. The selection algorithm is
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// applied even if |selector| is null, resulting in an empty report.
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RequestInfo(rtc::scoped_refptr<RtpReceiverInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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FilterMode filter_mode() const { return filter_mode_; }
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback() const {
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return callback_;
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}
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rtc::scoped_refptr<RtpSenderInternal> sender_selector() const {
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RTC_DCHECK(filter_mode_ == FilterMode::kSenderSelector);
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return sender_selector_;
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}
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rtc::scoped_refptr<RtpReceiverInternal> receiver_selector() const {
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RTC_DCHECK(filter_mode_ == FilterMode::kReceiverSelector);
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return receiver_selector_;
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}
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private:
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RequestInfo(FilterMode filter_mode,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback,
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rtc::scoped_refptr<RtpSenderInternal> sender_selector,
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rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);
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FilterMode filter_mode_;
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback_;
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rtc::scoped_refptr<RtpSenderInternal> sender_selector_;
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rtc::scoped_refptr<RtpReceiverInternal> receiver_selector_;
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};
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void GetStatsReportInternal(RequestInfo request);
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// Structure for tracking stats about each RtpTransceiver managed by the
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// PeerConnection. This can either by a Plan B style or Unified Plan style
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// transceiver (i.e., can have 0 or many senders and receivers).
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// Some fields are copied from the RtpTransceiver/BaseChannel object so that
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// they can be accessed safely on threads other than the signaling thread.
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// If a BaseChannel is not available (e.g., if signaling has not started),
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// then |mid| and |transport_name| will be null.
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struct RtpTransceiverStatsInfo {
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rtc::scoped_refptr<RtpTransceiver> transceiver;
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cricket::MediaType media_type;
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absl::optional<std::string> mid;
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absl::optional<std::string> transport_name;
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std::unique_ptr<TrackMediaInfoMap> track_media_info_map;
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};
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void DeliverCachedReport(
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rtc::scoped_refptr<const RTCStatsReport> cached_report,
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std::vector<RequestInfo> requests);
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// Produces |RTCCertificateStats|.
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void ProduceCertificateStats_n(
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int64_t timestamp_us,
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const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
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RTCStatsReport* report) const;
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// Produces |RTCCodecStats|.
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void ProduceCodecStats_n(
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int64_t timestamp_us,
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const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
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RTCStatsReport* report) const;
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// Produces |RTCDataChannelStats|.
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void ProduceDataChannelStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCIceCandidatePairStats| and |RTCIceCandidateStats|.
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void ProduceIceCandidateAndPairStats_n(
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int64_t timestamp_us,
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name,
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const Call::Stats& call_stats,
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RTCStatsReport* report) const;
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// Produces |RTCMediaStreamStats|.
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void ProduceMediaStreamStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCMediaStreamTrackStats|.
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void ProduceMediaStreamTrackStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces RTCMediaSourceStats, including RTCAudioSourceStats and
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// RTCVideoSourceStats.
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void ProduceMediaSourceStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCPeerConnectionStats|.
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void ProducePeerConnectionStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|.
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// This has to be invoked after codecs and transport stats have been created
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// because some metrics are calculated through lookup of other metrics.
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void ProduceRTPStreamStats_n(
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int64_t timestamp_us,
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const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
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RTCStatsReport* report) const;
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void ProduceAudioRTPStreamStats_n(int64_t timestamp_us,
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const RtpTransceiverStatsInfo& stats,
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RTCStatsReport* report) const;
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void ProduceVideoRTPStreamStats_n(int64_t timestamp_us,
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const RtpTransceiverStatsInfo& stats,
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RTCStatsReport* report) const;
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// Produces |RTCTransportStats|.
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void ProduceTransportStats_n(
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int64_t timestamp_us,
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name,
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const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
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RTCStatsReport* report) const;
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// Helper function to stats-producing functions.
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std::map<std::string, CertificateStatsPair>
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PrepareTransportCertificateStats_n(
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name) const;
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std::vector<RtpTransceiverStatsInfo> PrepareTransceiverStatsInfos_s_w() const;
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std::set<std::string> PrepareTransportNames_s() const;
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// Stats gathering on a particular thread.
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void ProducePartialResultsOnSignalingThread(int64_t timestamp_us);
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void ProducePartialResultsOnNetworkThread(int64_t timestamp_us);
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// Merges |network_report_| into |partial_report_| and completes the request.
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// This is a NO-OP if |network_report_| is null.
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void MergeNetworkReport_s();
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// Slots for signals (sigslot) that are wired up to |pc_|.
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void OnRtpDataChannelCreated(RtpDataChannel* channel);
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void OnSctpDataChannelCreated(SctpDataChannel* channel);
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// Slots for signals (sigslot) that are wired up to |channel|.
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void OnDataChannelOpened(DataChannelInterface* channel);
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void OnDataChannelClosed(DataChannelInterface* channel);
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PeerConnectionInternal* const pc_;
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rtc::Thread* const signaling_thread_;
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rtc::Thread* const worker_thread_;
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rtc::Thread* const network_thread_;
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int num_pending_partial_reports_;
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int64_t partial_report_timestamp_us_;
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// Reports that are produced on the signaling thread or the network thread are
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// merged into this report. It is only touched on the signaling thread. Once
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// all partial reports are merged this is the result of a request.
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rtc::scoped_refptr<RTCStatsReport> partial_report_;
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std::vector<RequestInfo> requests_;
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// Holds the result of ProducePartialResultsOnNetworkThread(). It is merged
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// into |partial_report_| on the signaling thread and then nulled by
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// MergeNetworkReport_s(). Thread-safety is ensured by using
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// |network_report_event_|.
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rtc::scoped_refptr<RTCStatsReport> network_report_;
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// If set, it is safe to touch the |network_report_| on the signaling thread.
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// This is reset before async-invoking ProducePartialResultsOnNetworkThread()
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// and set when ProducePartialResultsOnNetworkThread() is complete, after it
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// has updated the value of |network_report_|.
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rtc::Event network_report_event_;
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// Set in |GetStatsReport|, read in |ProducePartialResultsOnNetworkThread| and
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// |ProducePartialResultsOnSignalingThread|, reset after work is complete. Not
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// passed as arguments to avoid copies. This is thread safe - when we
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// set/reset we know there are no pending stats requests in progress.
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std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos_;
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std::set<std::string> transport_names_;
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Call::Stats call_stats_;
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// A timestamp, in microseconds, that is based on a timer that is
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// monotonically increasing. That is, even if the system clock is modified the
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// difference between the timer and this timestamp is how fresh the cached
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// report is.
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int64_t cache_timestamp_us_;
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int64_t cache_lifetime_us_;
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rtc::scoped_refptr<const RTCStatsReport> cached_report_;
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// Data recorded and maintained by the stats collector during its lifetime.
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// Some stats are produced from this record instead of other components.
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struct InternalRecord {
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InternalRecord() : data_channels_opened(0), data_channels_closed(0) {}
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// The opened count goes up when a channel is fully opened and the closed
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// count goes up if a previously opened channel has fully closed. The opened
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// count does not go down when a channel closes, meaning (opened - closed)
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// is the number of channels currently opened. A channel that is closed
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// before reaching the open state does not affect these counters.
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uint32_t data_channels_opened;
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uint32_t data_channels_closed;
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// Identifies by address channels that have been opened, which remain in the
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// set until they have been fully closed.
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std::set<uintptr_t> opened_data_channels;
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};
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InternalRecord internal_record_;
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};
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const char* CandidateTypeToRTCIceCandidateTypeForTesting(
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const std::string& type);
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const char* DataStateToRTCDataChannelStateForTesting(
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DataChannelInterface::DataState state);
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} // namespace webrtc
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#endif // PC_RTC_STATS_COLLECTOR_H_
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