| /* |
| * 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 <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.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 "pc/data_channel.h" |
| #include "pc/peer_connection_internal.h" |
| #include "pc/track_media_info_map.h" |
| #include "rtc_base/event.h" |
| #include "rtc_base/ref_count.h" |
| #include "rtc_base/ssl_identity.h" |
| #include "rtc_base/third_party/sigslot/sigslot.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 virtual rtc::RefCountInterface, |
| public sigslot::has_slots<> { |
| 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. |
| 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(); |
| |
| protected: |
| RTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime_us); |
| ~RTCStatsCollector(); |
| |
| struct CertificateStatsPair { |
| std::unique_ptr<rtc::SSLCertificateStats> local; |
| std::unique_ptr<rtc::SSLCertificateStats> remote; |
| }; |
| |
| // Stats gathering on a particular thread. Virtual for the sake of testing. |
| virtual void ProducePartialResultsOnSignalingThreadImpl( |
| int64_t timestamp_us, |
| RTCStatsReport* partial_report); |
| virtual void ProducePartialResultsOnNetworkThreadImpl( |
| int64_t timestamp_us, |
| 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; |
| std::unique_ptr<TrackMediaInfoMap> track_media_info_map; |
| }; |
| |
| void DeliverCachedReport( |
| rtc::scoped_refptr<const RTCStatsReport> cached_report, |
| std::vector<RequestInfo> requests); |
| |
| // Produces |RTCCertificateStats|. |
| void ProduceCertificateStats_n( |
| int64_t timestamp_us, |
| const std::map<std::string, CertificateStatsPair>& transport_cert_stats, |
| RTCStatsReport* report) const; |
| // Produces |RTCCodecStats|. |
| void ProduceCodecStats_n( |
| int64_t timestamp_us, |
| const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos, |
| RTCStatsReport* report) const; |
| // Produces |RTCDataChannelStats|. |
| void ProduceDataChannelStats_s(int64_t timestamp_us, |
| RTCStatsReport* report) const; |
| // Produces |RTCIceCandidatePairStats| and |RTCIceCandidateStats|. |
| void ProduceIceCandidateAndPairStats_n( |
| int64_t timestamp_us, |
| const std::map<std::string, cricket::TransportStats>& |
| transport_stats_by_name, |
| const Call::Stats& call_stats, |
| RTCStatsReport* report) const; |
| // Produces |RTCMediaStreamStats|. |
| void ProduceMediaStreamStats_s(int64_t timestamp_us, |
| RTCStatsReport* report) const; |
| // Produces |RTCMediaStreamTrackStats|. |
| void ProduceMediaStreamTrackStats_s(int64_t timestamp_us, |
| RTCStatsReport* report) const; |
| // Produces RTCMediaSourceStats, including RTCAudioSourceStats and |
| // RTCVideoSourceStats. |
| void ProduceMediaSourceStats_s(int64_t timestamp_us, |
| RTCStatsReport* report) const; |
| // Produces |RTCPeerConnectionStats|. |
| void ProducePeerConnectionStats_s(int64_t timestamp_us, |
| RTCStatsReport* report) const; |
| // Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|. |
| // This has to be invoked after codecs and transport stats have been created |
| // because some metrics are calculated through lookup of other metrics. |
| void ProduceRTPStreamStats_n( |
| int64_t timestamp_us, |
| const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos, |
| RTCStatsReport* report) const; |
| void ProduceAudioRTPStreamStats_n(int64_t timestamp_us, |
| const RtpTransceiverStatsInfo& stats, |
| RTCStatsReport* report) const; |
| void ProduceVideoRTPStreamStats_n(int64_t timestamp_us, |
| const RtpTransceiverStatsInfo& stats, |
| RTCStatsReport* report) const; |
| // Produces |RTCTransportStats|. |
| void ProduceTransportStats_n( |
| int64_t timestamp_us, |
| 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) const; |
| std::vector<RtpTransceiverStatsInfo> PrepareTransceiverStatsInfos_s() const; |
| std::set<std::string> PrepareTransportNames_s() const; |
| |
| // Stats gathering on a particular thread. |
| void ProducePartialResultsOnSignalingThread(int64_t timestamp_us); |
| void ProducePartialResultsOnNetworkThread(int64_t timestamp_us); |
| // Merges |network_report_| into |partial_report_| and completes the request. |
| // This is a NO-OP if |network_report_| is null. |
| void MergeNetworkReport_s(); |
| |
| // Slots for signals (sigslot) that are wired up to |pc_|. |
| void OnDataChannelCreated(DataChannel* channel); |
| // Slots for signals (sigslot) that are wired up to |channel|. |
| void OnDataChannelOpened(DataChannel* channel); |
| void OnDataChannelClosed(DataChannel* channel); |
| |
| 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_; |
| |
| // Set in |GetStatsReport|, read in |ProducePartialResultsOnNetworkThread| and |
| // |ProducePartialResultsOnSignalingThread|, reset after work is complete. Not |
| // passed as arguments to avoid copies. This is thread safe - when we |
| // set/reset we know there are no pending stats requests in progress. |
| std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos_; |
| std::set<std::string> transport_names_; |
| |
| Call::Stats call_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_; |
| bool enable_simulcast_stats_ = false; |
| }; |
| |
| const char* CandidateTypeToRTCIceCandidateTypeForTesting( |
| const std::string& type); |
| const char* DataStateToRTCDataChannelStateForTesting( |
| DataChannelInterface::DataState state); |
| |
| } // namespace webrtc |
| |
| #endif // PC_RTC_STATS_COLLECTOR_H_ |