webrtc/pc/rtc_stats_collector.h
Olov Brändström 4baeed3b97 Use environment monotonic timestamps (i.e. not UTC) in RTCStats.
Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.

Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
2024-10-03 09:07:17 +00:00

332 lines
15 KiB
C++

/*
* 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 <optional>
#include <string>
#include <vector>
#include "api/audio/audio_device.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/containers/flat_set.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 RefCountInterface {
public:
static rtc::scoped_refptr<RTCStatsCollector> Create(
PeerConnectionInternal* pc,
const Environment& env,
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(int channel_id,
DataChannelInterface::DataState state);
protected:
RTCStatsCollector(PeerConnectionInternal* pc,
const Environment& env,
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;
std::optional<std::string> mid;
std::optional<std::string> transport_name;
TrackMediaInfoMap track_media_info_map;
std::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_n(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 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,
std::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_;
const Environment env_;
const bool stats_timestamp_with_environment_clock_;
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_;
std::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 channels that have been opened, whose internal id is stored in
// the set until they have been fully closed.
flat_set<int> opened_data_channels;
};
InternalRecord internal_record_;
};
} // namespace webrtc
#endif // PC_RTC_STATS_COLLECTOR_H_