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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 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 "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/third_party/sigslot/sigslot.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 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. 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();
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(
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;
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 `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`, `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(
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);
// The results are stored in `transceiver_stats_infos_` and `call_stats_`.
void PrepareTransceiverStatsInfosAndCallStats_s_w_n();
// Stats gathering on a particular thread.
void ProducePartialResultsOnSignalingThread(int64_t timestamp_us);
void ProducePartialResultsOnNetworkThread(
int64_t timestamp_us,
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();
// Slots for signals (sigslot) that are wired up to `pc_`.
void OnSctpDataChannelCreated(SctpDataChannel* channel);
// Slots for signals (sigslot) that are wired up to `channel`.
void OnDataChannelOpened(DataChannelInterface* channel);
void OnDataChannelClosed(DataChannelInterface* 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_;
// 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.
std::map<std::string, CertificateStatsPair> cached_certificates_by_transport_;
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_;
};
const char* CandidateTypeToRTCIceCandidateTypeForTesting(
const std::string& type);
const char* DataStateToRTCDataChannelStateForTesting(
DataChannelInterface::DataState state);
} // namespace webrtc
#endif // PC_RTC_STATS_COLLECTOR_H_