pc/rtc_stats_collector.h - src - Git at Google

 /*
  *  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 PC_RTC_STATS_COLLECTOR_H_
 #define PC_RTC_STATS_COLLECTOR_H_

 #include <stdint.h>

 #include <cstdint>
 #include <map>
 #include <memory>
 #include <set>
 #include <string>
 #include <vector>

 #include "absl/types/optional.h"
 #include "api/data_channel_interface.h"
 #include "api/media_types.h"
 #include "api/scoped_refptr.h"
 #include "api/stats/rtc_stats_collector_callback.h"
 #include "api/stats/rtc_stats_report.h"
 #include "api/stats/rtcstats_objects.h"
 #include "call/call.h"
 #include "media/base/media_channel.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "pc/data_channel_utils.h"
 #include "pc/peer_connection_internal.h"
 #include "pc/rtp_receiver.h"
 #include "pc/rtp_sender.h"
 #include "pc/rtp_transceiver.h"
 #include "pc/sctp_data_channel.h"
 #include "pc/track_media_info_map.h"
 #include "pc/transport_stats.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/time_utils.h"

 namespace webrtc {

 class RtpSenderInternal;
 class RtpReceiverInternal;

 // All public methods of the collector are to be called on the signaling thread.
 // Stats are gathered on the signaling, worker and network threads
 // asynchronously. The callback is invoked on the signaling thread. Resulting
 // reports are cached for `cache_lifetime_` ms.
 class RTCStatsCollector : public rtc::RefCountInterface {
  public:
   static rtc::scoped_refptr<RTCStatsCollector> Create(
       PeerConnectionInternal* pc,
       int64_t cache_lifetime_us = 50 * rtc::kNumMicrosecsPerMillisec);

   // Gets a recent stats report. If there is a report cached that is still fresh
   // it is returned, otherwise new stats are gathered and returned. A report is
   // considered fresh for `cache_lifetime_` ms. const RTCStatsReports are safe
   // to use across multiple threads and may be destructed on any thread.
   // If the optional selector argument is used, stats are filtered according to
   // stats selection algorithm before delivery.
   // https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
   void GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
   // If `selector` is null the selection algorithm is still applied (interpreted
   // as: no RTP streams are sent by selector). The result is empty.
   void GetStatsReport(rtc::scoped_refptr<RtpSenderInternal> selector,
                       rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
   // If `selector` is null the selection algorithm is still applied (interpreted
   // as: no RTP streams are received by selector). The result is empty.
   void GetStatsReport(rtc::scoped_refptr<RtpReceiverInternal> selector,
                       rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
   // Clears the cache's reference to the most recent stats report. Subsequently
   // calling `GetStatsReport` guarantees fresh stats. This method must be called
   // any time the PeerConnection visibly changes as a result of an API call as
   // per
   // https://w3c.github.io/webrtc-stats/#guidelines-for-getstats-results-caching-throttling
   // and it must be called any time negotiation happens.
   void ClearCachedStatsReport();

   // If there is a `GetStatsReport` requests in-flight, waits until it has been
   // completed. Must be called on the signaling thread.
   void WaitForPendingRequest();

   // Called by the PeerConnection instance when data channel states change.
   void OnSctpDataChannelStateChanged(DataChannelInterface* channel,
                                      DataChannelInterface::DataState state);

  protected:
   RTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime_us);
   ~RTCStatsCollector();

   struct CertificateStatsPair {
     std::unique_ptr<rtc::SSLCertificateStats> local;
     std::unique_ptr<rtc::SSLCertificateStats> remote;

     CertificateStatsPair Copy() const;
   };

   // Stats gathering on a particular thread. Virtual for the sake of testing.
   virtual void ProducePartialResultsOnSignalingThreadImpl(
       Timestamp timestamp,
       RTCStatsReport* partial_report);
   virtual void ProducePartialResultsOnNetworkThreadImpl(
       Timestamp timestamp,
       const std::map<std::string, cricket::TransportStats>&
           transport_stats_by_name,
       const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
       RTCStatsReport* partial_report);

  private:
   class RequestInfo {
    public:
     enum class FilterMode { kAll, kSenderSelector, kReceiverSelector };

     // Constructs with FilterMode::kAll.
     explicit RequestInfo(
         rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
     // Constructs with FilterMode::kSenderSelector. The selection algorithm is
     // applied even if `selector` is null, resulting in an empty report.
     RequestInfo(rtc::scoped_refptr<RtpSenderInternal> selector,
                 rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
     // Constructs with FilterMode::kReceiverSelector. The selection algorithm is
     // applied even if `selector` is null, resulting in an empty report.
     RequestInfo(rtc::scoped_refptr<RtpReceiverInternal> selector,
                 rtc::scoped_refptr<RTCStatsCollectorCallback> callback);

     FilterMode filter_mode() const { return filter_mode_; }
     rtc::scoped_refptr<RTCStatsCollectorCallback> callback() const {
       return callback_;
     }
     rtc::scoped_refptr<RtpSenderInternal> sender_selector() const {
       RTC_DCHECK(filter_mode_ == FilterMode::kSenderSelector);
       return sender_selector_;
     }
     rtc::scoped_refptr<RtpReceiverInternal> receiver_selector() const {
       RTC_DCHECK(filter_mode_ == FilterMode::kReceiverSelector);
       return receiver_selector_;
     }

    private:
     RequestInfo(FilterMode filter_mode,
                 rtc::scoped_refptr<RTCStatsCollectorCallback> callback,
                 rtc::scoped_refptr<RtpSenderInternal> sender_selector,
                 rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);

     FilterMode filter_mode_;
     rtc::scoped_refptr<RTCStatsCollectorCallback> callback_;
     rtc::scoped_refptr<RtpSenderInternal> sender_selector_;
     rtc::scoped_refptr<RtpReceiverInternal> receiver_selector_;
   };

   void GetStatsReportInternal(RequestInfo request);

   // Structure for tracking stats about each RtpTransceiver managed by the
   // PeerConnection. This can either by a Plan B style or Unified Plan style
   // transceiver (i.e., can have 0 or many senders and receivers).
   // Some fields are copied from the RtpTransceiver/BaseChannel object so that
   // they can be accessed safely on threads other than the signaling thread.
   // If a BaseChannel is not available (e.g., if signaling has not started),
   // then `mid` and `transport_name` will be null.
   struct RtpTransceiverStatsInfo {
     rtc::scoped_refptr<RtpTransceiver> transceiver;
     cricket::MediaType media_type;
     absl::optional<std::string> mid;
     absl::optional<std::string> transport_name;
     TrackMediaInfoMap track_media_info_map;
     absl::optional<RtpTransceiverDirection> current_direction;
   };

   void DeliverCachedReport(
       rtc::scoped_refptr<const RTCStatsReport> cached_report,
       std::vector<RequestInfo> requests);

   // Produces `RTCCertificateStats`.
   void ProduceCertificateStats_n(
       Timestamp timestamp,
       const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
       RTCStatsReport* report) const;
   // Produces `RTCDataChannelStats`.
   void ProduceDataChannelStats_s(Timestamp timestamp,
                                  RTCStatsReport* report) const;
   // Produces `RTCIceCandidatePairStats` and `RTCIceCandidateStats`.
   void ProduceIceCandidateAndPairStats_n(
       Timestamp timestamp,
       const std::map<std::string, cricket::TransportStats>&
           transport_stats_by_name,
       const Call::Stats& call_stats,
       RTCStatsReport* report) const;
   // Produces `RTCMediaStreamStats`.
   void ProduceMediaStreamStats_s(Timestamp timestamp,
                                  RTCStatsReport* report) const;
   // Produces `RTCMediaStreamTrackStats`.
   void ProduceMediaStreamTrackStats_s(Timestamp timestamp,
                                       RTCStatsReport* report) const;
   // Produces RTCMediaSourceStats, including RTCAudioSourceStats and
   // RTCVideoSourceStats.
   void ProduceMediaSourceStats_s(Timestamp timestamp,
                                  RTCStatsReport* report) const;
   // Produces `RTCPeerConnectionStats`.
   void ProducePeerConnectionStats_s(Timestamp timestamp,
                                     RTCStatsReport* report) const;
   // Produces `RTCAudioPlayoutStats`.
   void ProduceAudioPlayoutStats_s(Timestamp timestamp,
                                   RTCStatsReport* report) const;
   // Produces `RTCInboundRTPStreamStats`, `RTCOutboundRTPStreamStats`,
   // `RTCRemoteInboundRtpStreamStats`, `RTCRemoteOutboundRtpStreamStats` and any
   // referenced `RTCCodecStats`. This has to be invoked after transport stats
   // have been created because some metrics are calculated through lookup of
   // other metrics.
   void ProduceRTPStreamStats_n(
       Timestamp timestamp,
       const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
       RTCStatsReport* report) const;
   void ProduceAudioRTPStreamStats_n(Timestamp timestamp,
                                     const RtpTransceiverStatsInfo& stats,
                                     RTCStatsReport* report) const;
   void ProduceVideoRTPStreamStats_n(Timestamp timestamp,
                                     const RtpTransceiverStatsInfo& stats,
                                     RTCStatsReport* report) const;
   // Produces `RTCTransportStats`.
   void ProduceTransportStats_n(
       Timestamp timestamp,
       const std::map<std::string, cricket::TransportStats>&
           transport_stats_by_name,
       const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
       RTCStatsReport* report) const;

   // Helper function to stats-producing functions.
   std::map<std::string, CertificateStatsPair>
   PrepareTransportCertificateStats_n(
       const std::map<std::string, cricket::TransportStats>&
           transport_stats_by_name);
   // The results are stored in `transceiver_stats_infos_` and `call_stats_`.
   void PrepareTransceiverStatsInfosAndCallStats_s_w_n();

   // Stats gathering on a particular thread.
   void ProducePartialResultsOnSignalingThread(Timestamp timestamp);
   void ProducePartialResultsOnNetworkThread(
       Timestamp timestamp,
       absl::optional<std::string> sctp_transport_name);
   // Merges `network_report_` into `partial_report_` and completes the request.
   // This is a NO-OP if `network_report_` is null.
   void MergeNetworkReport_s();

   rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
       bool filter_by_sender_selector,
       rtc::scoped_refptr<const RTCStatsReport> report,
       rtc::scoped_refptr<RtpSenderInternal> sender_selector,
       rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);

   PeerConnectionInternal* const pc_;
   rtc::Thread* const signaling_thread_;
   rtc::Thread* const worker_thread_;
   rtc::Thread* const network_thread_;

   int num_pending_partial_reports_;
   int64_t partial_report_timestamp_us_;
   // Reports that are produced on the signaling thread or the network thread are
   // merged into this report. It is only touched on the signaling thread. Once
   // all partial reports are merged this is the result of a request.
   rtc::scoped_refptr<RTCStatsReport> partial_report_;
   std::vector<RequestInfo> requests_;
   // Holds the result of ProducePartialResultsOnNetworkThread(). It is merged
   // into `partial_report_` on the signaling thread and then nulled by
   // MergeNetworkReport_s(). Thread-safety is ensured by using
   // `network_report_event_`.
   rtc::scoped_refptr<RTCStatsReport> network_report_;
   // If set, it is safe to touch the `network_report_` on the signaling thread.
   // This is reset before async-invoking ProducePartialResultsOnNetworkThread()
   // and set when ProducePartialResultsOnNetworkThread() is complete, after it
   // has updated the value of `network_report_`.
   rtc::Event network_report_event_;

   // Cleared and set in `PrepareTransceiverStatsInfosAndCallStats_s_w_n`,
   // starting out on the signaling thread, then network. Later read on the
   // network and signaling threads as part of collecting stats and finally
   // reset when the work is done. Initially this variable was added and not
   // passed around as an arguments to avoid copies. This is thread safe due to
   // how operations are sequenced and we don't start the stats collection
   // sequence if one is in progress. As a future improvement though, we could
   // now get rid of the variable and keep the data scoped within a stats
   // collection sequence.
   std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos_;
   // This cache avoids having to call rtc::SSLCertChain::GetStats(), which can
   // relatively expensive. ClearCachedStatsReport() needs to be called on
   // negotiation to ensure the cache is not obsolete.
   Mutex cached_certificates_mutex_;
   std::map<std::string, CertificateStatsPair> cached_certificates_by_transport_
       RTC_GUARDED_BY(cached_certificates_mutex_);

   Call::Stats call_stats_;

   absl::optional<AudioDeviceModule::Stats> audio_device_stats_;

   // A timestamp, in microseconds, that is based on a timer that is
   // monotonically increasing. That is, even if the system clock is modified the
   // difference between the timer and this timestamp is how fresh the cached
   // report is.
   int64_t cache_timestamp_us_;
   int64_t cache_lifetime_us_;
   rtc::scoped_refptr<const RTCStatsReport> cached_report_;

   // Data recorded and maintained by the stats collector during its lifetime.
   // Some stats are produced from this record instead of other components.
   struct InternalRecord {
     InternalRecord() : data_channels_opened(0), data_channels_closed(0) {}

     // The opened count goes up when a channel is fully opened and the closed
     // count goes up if a previously opened channel has fully closed. The opened
     // count does not go down when a channel closes, meaning (opened - closed)
     // is the number of channels currently opened. A channel that is closed
     // before reaching the open state does not affect these counters.
     uint32_t data_channels_opened;
     uint32_t data_channels_closed;
     // Identifies by address channels that have been opened, which remain in the
     // set until they have been fully closed.
     std::set<uintptr_t> opened_data_channels;
   };
   InternalRecord internal_record_;
 };

 const char* CandidateTypeToRTCIceCandidateTypeForTesting(
     const std::string& type);
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state);

 }  // namespace webrtc

 #endif  // PC_RTC_STATS_COLLECTOR_H_
	/*
	* 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 PC_RTC_STATS_COLLECTOR_H_
	#define PC_RTC_STATS_COLLECTOR_H_

	#include <stdint.h>

	#include <cstdint>
	#include <map>
	#include <memory>
	#include <set>
	#include <string>
	#include <vector>

	#include "absl/types/optional.h"
	#include "api/data_channel_interface.h"
	#include "api/media_types.h"
	#include "api/scoped_refptr.h"
	#include "api/stats/rtc_stats_collector_callback.h"
	#include "api/stats/rtc_stats_report.h"
	#include "api/stats/rtcstats_objects.h"
	#include "call/call.h"
	#include "media/base/media_channel.h"
	#include "modules/audio_device/include/audio_device.h"
	#include "pc/data_channel_utils.h"
	#include "pc/peer_connection_internal.h"
	#include "pc/rtp_receiver.h"
	#include "pc/rtp_sender.h"
	#include "pc/rtp_transceiver.h"
	#include "pc/sctp_data_channel.h"
	#include "pc/track_media_info_map.h"
	#include "pc/transport_stats.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/event.h"
	#include "rtc_base/ref_count.h"
	#include "rtc_base/ssl_certificate.h"
	#include "rtc_base/ssl_identity.h"
	#include "rtc_base/synchronization/mutex.h"
	#include "rtc_base/thread.h"
	#include "rtc_base/time_utils.h"

	namespace webrtc {

	class RtpSenderInternal;
	class RtpReceiverInternal;

	// All public methods of the collector are to be called on the signaling thread.
	// Stats are gathered on the signaling, worker and network threads
	// asynchronously. The callback is invoked on the signaling thread. Resulting
	// reports are cached for `cache_lifetime_` ms.
	class RTCStatsCollector : public rtc::RefCountInterface {
	public:
	static rtc::scoped_refptr<RTCStatsCollector> Create(
	PeerConnectionInternal* pc,
	int64_t cache_lifetime_us = 50 * rtc::kNumMicrosecsPerMillisec);

	// Gets a recent stats report. If there is a report cached that is still fresh
	// it is returned, otherwise new stats are gathered and returned. A report is
	// considered fresh for `cache_lifetime_` ms. const RTCStatsReports are safe
	// to use across multiple threads and may be destructed on any thread.
	// If the optional selector argument is used, stats are filtered according to
	// stats selection algorithm before delivery.
	// https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
	void GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
	// If `selector` is null the selection algorithm is still applied (interpreted
	// as: no RTP streams are sent by selector). The result is empty.
	void GetStatsReport(rtc::scoped_refptr<RtpSenderInternal> selector,
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
	// If `selector` is null the selection algorithm is still applied (interpreted
	// as: no RTP streams are received by selector). The result is empty.
	void GetStatsReport(rtc::scoped_refptr<RtpReceiverInternal> selector,
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
	// Clears the cache's reference to the most recent stats report. Subsequently
	// calling `GetStatsReport` guarantees fresh stats. This method must be called
	// any time the PeerConnection visibly changes as a result of an API call as
	// per
	// https://w3c.github.io/webrtc-stats/#guidelines-for-getstats-results-caching-throttling
	// and it must be called any time negotiation happens.
	void ClearCachedStatsReport();

	// If there is a `GetStatsReport` requests in-flight, waits until it has been
	// completed. Must be called on the signaling thread.
	void WaitForPendingRequest();

	// Called by the PeerConnection instance when data channel states change.
	void OnSctpDataChannelStateChanged(DataChannelInterface* channel,
	DataChannelInterface::DataState state);

	protected:
	RTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime_us);
	~RTCStatsCollector();

	struct CertificateStatsPair {
	std::unique_ptr<rtc::SSLCertificateStats> local;
	std::unique_ptr<rtc::SSLCertificateStats> remote;

	CertificateStatsPair Copy() const;
	};

	// Stats gathering on a particular thread. Virtual for the sake of testing.
	virtual void ProducePartialResultsOnSignalingThreadImpl(
	Timestamp timestamp,
	RTCStatsReport* partial_report);
	virtual void ProducePartialResultsOnNetworkThreadImpl(
	Timestamp timestamp,
	const std::map<std::string, cricket::TransportStats>&
	transport_stats_by_name,
	const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
	RTCStatsReport* partial_report);

	private:
	class RequestInfo {
	public:
	enum class FilterMode { kAll, kSenderSelector, kReceiverSelector };

	// Constructs with FilterMode::kAll.
	explicit RequestInfo(
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
	// Constructs with FilterMode::kSenderSelector. The selection algorithm is
	// applied even if `selector` is null, resulting in an empty report.
	RequestInfo(rtc::scoped_refptr<RtpSenderInternal> selector,
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
	// Constructs with FilterMode::kReceiverSelector. The selection algorithm is
	// applied even if `selector` is null, resulting in an empty report.
	RequestInfo(rtc::scoped_refptr<RtpReceiverInternal> selector,
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback);

	FilterMode filter_mode() const { return filter_mode_; }
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback() const {
	return callback_;
	}
	rtc::scoped_refptr<RtpSenderInternal> sender_selector() const {
	RTC_DCHECK(filter_mode_ == FilterMode::kSenderSelector);
	return sender_selector_;
	}
	rtc::scoped_refptr<RtpReceiverInternal> receiver_selector() const {
	RTC_DCHECK(filter_mode_ == FilterMode::kReceiverSelector);
	return receiver_selector_;
	}

	private:
	RequestInfo(FilterMode filter_mode,
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback,
	rtc::scoped_refptr<RtpSenderInternal> sender_selector,
	rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);

	FilterMode filter_mode_;
	rtc::scoped_refptr<RTCStatsCollectorCallback> callback_;
	rtc::scoped_refptr<RtpSenderInternal> sender_selector_;
	rtc::scoped_refptr<RtpReceiverInternal> receiver_selector_;
	};

	void GetStatsReportInternal(RequestInfo request);

	// Structure for tracking stats about each RtpTransceiver managed by the
	// PeerConnection. This can either by a Plan B style or Unified Plan style
	// transceiver (i.e., can have 0 or many senders and receivers).
	// Some fields are copied from the RtpTransceiver/BaseChannel object so that
	// they can be accessed safely on threads other than the signaling thread.
	// If a BaseChannel is not available (e.g., if signaling has not started),
	// then `mid` and `transport_name` will be null.
	struct RtpTransceiverStatsInfo {
	rtc::scoped_refptr<RtpTransceiver> transceiver;
	cricket::MediaType media_type;
	absl::optional<std::string> mid;
	absl::optional<std::string> transport_name;
	TrackMediaInfoMap track_media_info_map;
	absl::optional<RtpTransceiverDirection> current_direction;
	};

	void DeliverCachedReport(
	rtc::scoped_refptr<const RTCStatsReport> cached_report,
	std::vector<RequestInfo> requests);

	// Produces `RTCCertificateStats`.
	void ProduceCertificateStats_n(
	Timestamp timestamp,
	const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
	RTCStatsReport* report) const;
	// Produces `RTCDataChannelStats`.
	void ProduceDataChannelStats_s(Timestamp timestamp,
	RTCStatsReport* report) const;
	// Produces `RTCIceCandidatePairStats` and `RTCIceCandidateStats`.
	void ProduceIceCandidateAndPairStats_n(
	Timestamp timestamp,
	const std::map<std::string, cricket::TransportStats>&
	transport_stats_by_name,
	const Call::Stats& call_stats,
	RTCStatsReport* report) const;
	// Produces `RTCMediaStreamStats`.
	void ProduceMediaStreamStats_s(Timestamp timestamp,
	RTCStatsReport* report) const;
	// Produces `RTCMediaStreamTrackStats`.
	void ProduceMediaStreamTrackStats_s(Timestamp timestamp,
	RTCStatsReport* report) const;
	// Produces RTCMediaSourceStats, including RTCAudioSourceStats and
	// RTCVideoSourceStats.
	void ProduceMediaSourceStats_s(Timestamp timestamp,
	RTCStatsReport* report) const;
	// Produces `RTCPeerConnectionStats`.
	void ProducePeerConnectionStats_s(Timestamp timestamp,
	RTCStatsReport* report) const;
	// Produces `RTCAudioPlayoutStats`.
	void ProduceAudioPlayoutStats_s(Timestamp timestamp,
	RTCStatsReport* report) const;
	// Produces `RTCInboundRTPStreamStats`, `RTCOutboundRTPStreamStats`,
	// `RTCRemoteInboundRtpStreamStats`, `RTCRemoteOutboundRtpStreamStats` and any
	// referenced `RTCCodecStats`. This has to be invoked after transport stats
	// have been created because some metrics are calculated through lookup of
	// other metrics.
	void ProduceRTPStreamStats_n(
	Timestamp timestamp,
	const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
	RTCStatsReport* report) const;
	void ProduceAudioRTPStreamStats_n(Timestamp timestamp,
	const RtpTransceiverStatsInfo& stats,
	RTCStatsReport* report) const;
	void ProduceVideoRTPStreamStats_n(Timestamp timestamp,
	const RtpTransceiverStatsInfo& stats,
	RTCStatsReport* report) const;
	// Produces `RTCTransportStats`.
	void ProduceTransportStats_n(
	Timestamp timestamp,
	const std::map<std::string, cricket::TransportStats>&
	transport_stats_by_name,
	const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
	RTCStatsReport* report) const;

	// Helper function to stats-producing functions.
	std::map<std::string, CertificateStatsPair>
	PrepareTransportCertificateStats_n(
	const std::map<std::string, cricket::TransportStats>&
	transport_stats_by_name);
	// The results are stored in `transceiver_stats_infos_` and `call_stats_`.
	void PrepareTransceiverStatsInfosAndCallStats_s_w_n();

	// Stats gathering on a particular thread.
	void ProducePartialResultsOnSignalingThread(Timestamp timestamp);
	void ProducePartialResultsOnNetworkThread(
	Timestamp timestamp,
	absl::optional<std::string> sctp_transport_name);
	// Merges `network_report_` into `partial_report_` and completes the request.
	// This is a NO-OP if `network_report_` is null.
	void MergeNetworkReport_s();

	rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
	bool filter_by_sender_selector,
	rtc::scoped_refptr<const RTCStatsReport> report,
	rtc::scoped_refptr<RtpSenderInternal> sender_selector,
	rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);

	PeerConnectionInternal* const pc_;
	rtc::Thread* const signaling_thread_;
	rtc::Thread* const worker_thread_;
	rtc::Thread* const network_thread_;

	int num_pending_partial_reports_;
	int64_t partial_report_timestamp_us_;
	// Reports that are produced on the signaling thread or the network thread are
	// merged into this report. It is only touched on the signaling thread. Once
	// all partial reports are merged this is the result of a request.
	rtc::scoped_refptr<RTCStatsReport> partial_report_;
	std::vector<RequestInfo> requests_;
	// Holds the result of ProducePartialResultsOnNetworkThread(). It is merged
	// into `partial_report_` on the signaling thread and then nulled by
	// MergeNetworkReport_s(). Thread-safety is ensured by using
	// `network_report_event_`.
	rtc::scoped_refptr<RTCStatsReport> network_report_;
	// If set, it is safe to touch the `network_report_` on the signaling thread.
	// This is reset before async-invoking ProducePartialResultsOnNetworkThread()
	// and set when ProducePartialResultsOnNetworkThread() is complete, after it
	// has updated the value of `network_report_`.
	rtc::Event network_report_event_;

	// Cleared and set in `PrepareTransceiverStatsInfosAndCallStats_s_w_n`,
	// starting out on the signaling thread, then network. Later read on the
	// network and signaling threads as part of collecting stats and finally
	// reset when the work is done. Initially this variable was added and not
	// passed around as an arguments to avoid copies. This is thread safe due to
	// how operations are sequenced and we don't start the stats collection
	// sequence if one is in progress. As a future improvement though, we could
	// now get rid of the variable and keep the data scoped within a stats
	// collection sequence.
	std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos_;
	// This cache avoids having to call rtc::SSLCertChain::GetStats(), which can
	// relatively expensive. ClearCachedStatsReport() needs to be called on
	// negotiation to ensure the cache is not obsolete.
	Mutex cached_certificates_mutex_;
	std::map<std::string, CertificateStatsPair> cached_certificates_by_transport_
	RTC_GUARDED_BY(cached_certificates_mutex_);

	Call::Stats call_stats_;

	absl::optional<AudioDeviceModule::Stats> audio_device_stats_;

	// A timestamp, in microseconds, that is based on a timer that is
	// monotonically increasing. That is, even if the system clock is modified the
	// difference between the timer and this timestamp is how fresh the cached
	// report is.
	int64_t cache_timestamp_us_;
	int64_t cache_lifetime_us_;
	rtc::scoped_refptr<const RTCStatsReport> cached_report_;

	// Data recorded and maintained by the stats collector during its lifetime.
	// Some stats are produced from this record instead of other components.
	struct InternalRecord {
	InternalRecord() : data_channels_opened(0), data_channels_closed(0) {}

	// The opened count goes up when a channel is fully opened and the closed
	// count goes up if a previously opened channel has fully closed. The opened
	// count does not go down when a channel closes, meaning (opened - closed)
	// is the number of channels currently opened. A channel that is closed
	// before reaching the open state does not affect these counters.
	uint32_t data_channels_opened;
	uint32_t data_channels_closed;
	// Identifies by address channels that have been opened, which remain in the
	// set until they have been fully closed.
	std::set<uintptr_t> opened_data_channels;
	};
	InternalRecord internal_record_;
	};

	const char* CandidateTypeToRTCIceCandidateTypeForTesting(
	const std::string& type);
	const char* DataStateToRTCDataChannelStateForTesting(
	DataChannelInterface::DataState state);

	} // namespace webrtc

	#endif // PC_RTC_STATS_COLLECTOR_H_