mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00

Bug: webrtc:15874 Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42137}
327 lines
14 KiB
C++
327 lines
14 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 <string>
|
|
#include <vector>
|
|
|
|
#include "absl/types/optional.h"
|
|
#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 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(int channel_id,
|
|
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_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,
|
|
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 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_
|