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Done in preparation for some threading changes that would be quite messy if implemented with the class as-is. This results in some code duplication, but is preferable to one class having two completely different modes of operation. RTP data channels are in the process of being removed anyway, so the duplicated code won't last forever. Bug: webrtc:9883 Change-Id: Idfd41a669b56a4bb4819572e4a264a4ffaaba9c0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178940 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31691}
301 lines
13 KiB
C++
301 lines
13 KiB
C++
/*
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* Copyright 2016 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef PC_RTC_STATS_COLLECTOR_H_
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#define PC_RTC_STATS_COLLECTOR_H_
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#include <map>
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#include <memory>
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#include <set>
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/scoped_refptr.h"
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#include "api/stats/rtc_stats_collector_callback.h"
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#include "api/stats/rtc_stats_report.h"
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#include "api/stats/rtcstats_objects.h"
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#include "call/call.h"
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#include "media/base/media_channel.h"
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#include "pc/data_channel_utils.h"
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#include "pc/peer_connection_internal.h"
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#include "pc/track_media_info_map.h"
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#include "rtc_base/event.h"
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#include "rtc_base/ref_count.h"
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#include "rtc_base/ssl_identity.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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#include "rtc_base/time_utils.h"
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namespace webrtc {
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class RtpSenderInternal;
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class RtpReceiverInternal;
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// All public methods of the collector are to be called on the signaling thread.
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// Stats are gathered on the signaling, worker and network threads
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// asynchronously. The callback is invoked on the signaling thread. Resulting
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// reports are cached for |cache_lifetime_| ms.
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class RTCStatsCollector : public virtual rtc::RefCountInterface,
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public sigslot::has_slots<> {
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public:
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static rtc::scoped_refptr<RTCStatsCollector> Create(
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PeerConnectionInternal* pc,
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int64_t cache_lifetime_us = 50 * rtc::kNumMicrosecsPerMillisec);
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// Gets a recent stats report. If there is a report cached that is still fresh
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// it is returned, otherwise new stats are gathered and returned. A report is
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// considered fresh for |cache_lifetime_| ms. const RTCStatsReports are safe
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// to use across multiple threads and may be destructed on any thread.
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// If the optional selector argument is used, stats are filtered according to
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// stats selection algorithm before delivery.
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// https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
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void GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// If |selector| is null the selection algorithm is still applied (interpreted
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// as: no RTP streams are sent by selector). The result is empty.
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void GetStatsReport(rtc::scoped_refptr<RtpSenderInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// If |selector| is null the selection algorithm is still applied (interpreted
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// as: no RTP streams are received by selector). The result is empty.
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void GetStatsReport(rtc::scoped_refptr<RtpReceiverInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// Clears the cache's reference to the most recent stats report. Subsequently
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// calling |GetStatsReport| guarantees fresh stats.
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void ClearCachedStatsReport();
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// If there is a |GetStatsReport| requests in-flight, waits until it has been
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// completed. Must be called on the signaling thread.
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void WaitForPendingRequest();
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protected:
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RTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime_us);
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~RTCStatsCollector();
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struct CertificateStatsPair {
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std::unique_ptr<rtc::SSLCertificateStats> local;
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std::unique_ptr<rtc::SSLCertificateStats> remote;
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};
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// Stats gathering on a particular thread. Virtual for the sake of testing.
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virtual void ProducePartialResultsOnSignalingThreadImpl(
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int64_t timestamp_us,
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RTCStatsReport* partial_report);
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virtual void ProducePartialResultsOnNetworkThreadImpl(
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int64_t timestamp_us,
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name,
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const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
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RTCStatsReport* partial_report);
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private:
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class RequestInfo {
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public:
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enum class FilterMode { kAll, kSenderSelector, kReceiverSelector };
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// Constructs with FilterMode::kAll.
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explicit RequestInfo(
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// Constructs with FilterMode::kSenderSelector. The selection algorithm is
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// applied even if |selector| is null, resulting in an empty report.
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RequestInfo(rtc::scoped_refptr<RtpSenderInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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// Constructs with FilterMode::kReceiverSelector. The selection algorithm is
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// applied even if |selector| is null, resulting in an empty report.
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RequestInfo(rtc::scoped_refptr<RtpReceiverInternal> selector,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
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FilterMode filter_mode() const { return filter_mode_; }
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback() const {
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return callback_;
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}
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rtc::scoped_refptr<RtpSenderInternal> sender_selector() const {
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RTC_DCHECK(filter_mode_ == FilterMode::kSenderSelector);
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return sender_selector_;
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}
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rtc::scoped_refptr<RtpReceiverInternal> receiver_selector() const {
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RTC_DCHECK(filter_mode_ == FilterMode::kReceiverSelector);
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return receiver_selector_;
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}
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private:
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RequestInfo(FilterMode filter_mode,
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback,
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rtc::scoped_refptr<RtpSenderInternal> sender_selector,
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rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);
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FilterMode filter_mode_;
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rtc::scoped_refptr<RTCStatsCollectorCallback> callback_;
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rtc::scoped_refptr<RtpSenderInternal> sender_selector_;
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rtc::scoped_refptr<RtpReceiverInternal> receiver_selector_;
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};
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void GetStatsReportInternal(RequestInfo request);
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// Structure for tracking stats about each RtpTransceiver managed by the
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// PeerConnection. This can either by a Plan B style or Unified Plan style
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// transceiver (i.e., can have 0 or many senders and receivers).
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// Some fields are copied from the RtpTransceiver/BaseChannel object so that
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// they can be accessed safely on threads other than the signaling thread.
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// If a BaseChannel is not available (e.g., if signaling has not started),
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// then |mid| and |transport_name| will be null.
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struct RtpTransceiverStatsInfo {
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rtc::scoped_refptr<RtpTransceiver> transceiver;
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cricket::MediaType media_type;
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absl::optional<std::string> mid;
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absl::optional<std::string> transport_name;
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std::unique_ptr<TrackMediaInfoMap> track_media_info_map;
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};
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void DeliverCachedReport(
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rtc::scoped_refptr<const RTCStatsReport> cached_report,
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std::vector<RequestInfo> requests);
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// Produces |RTCCertificateStats|.
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void ProduceCertificateStats_n(
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int64_t timestamp_us,
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const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
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RTCStatsReport* report) const;
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// Produces |RTCCodecStats|.
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void ProduceCodecStats_n(
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int64_t timestamp_us,
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const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
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RTCStatsReport* report) const;
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// Produces |RTCDataChannelStats|.
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void ProduceDataChannelStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCIceCandidatePairStats| and |RTCIceCandidateStats|.
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void ProduceIceCandidateAndPairStats_n(
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int64_t timestamp_us,
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name,
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const Call::Stats& call_stats,
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RTCStatsReport* report) const;
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// Produces |RTCMediaStreamStats|.
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void ProduceMediaStreamStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCMediaStreamTrackStats|.
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void ProduceMediaStreamTrackStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces RTCMediaSourceStats, including RTCAudioSourceStats and
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// RTCVideoSourceStats.
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void ProduceMediaSourceStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCPeerConnectionStats|.
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void ProducePeerConnectionStats_s(int64_t timestamp_us,
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RTCStatsReport* report) const;
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// Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|.
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// This has to be invoked after codecs and transport stats have been created
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// because some metrics are calculated through lookup of other metrics.
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void ProduceRTPStreamStats_n(
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int64_t timestamp_us,
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const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
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RTCStatsReport* report) const;
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void ProduceAudioRTPStreamStats_n(int64_t timestamp_us,
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const RtpTransceiverStatsInfo& stats,
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RTCStatsReport* report) const;
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void ProduceVideoRTPStreamStats_n(int64_t timestamp_us,
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const RtpTransceiverStatsInfo& stats,
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RTCStatsReport* report) const;
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// Produces |RTCTransportStats|.
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void ProduceTransportStats_n(
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int64_t timestamp_us,
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name,
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const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
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RTCStatsReport* report) const;
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// Helper function to stats-producing functions.
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std::map<std::string, CertificateStatsPair>
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PrepareTransportCertificateStats_n(
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const std::map<std::string, cricket::TransportStats>&
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transport_stats_by_name) const;
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std::vector<RtpTransceiverStatsInfo> PrepareTransceiverStatsInfos_s_w() const;
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std::set<std::string> PrepareTransportNames_s() const;
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// Stats gathering on a particular thread.
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void ProducePartialResultsOnSignalingThread(int64_t timestamp_us);
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void ProducePartialResultsOnNetworkThread(int64_t timestamp_us);
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// Merges |network_report_| into |partial_report_| and completes the request.
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// This is a NO-OP if |network_report_| is null.
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void MergeNetworkReport_s();
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// Slots for signals (sigslot) that are wired up to |pc_|.
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void OnRtpDataChannelCreated(RtpDataChannel* channel);
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void OnSctpDataChannelCreated(SctpDataChannel* channel);
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// Slots for signals (sigslot) that are wired up to |channel|.
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void OnDataChannelOpened(DataChannelInterface* channel);
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void OnDataChannelClosed(DataChannelInterface* channel);
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PeerConnectionInternal* const pc_;
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rtc::Thread* const signaling_thread_;
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rtc::Thread* const worker_thread_;
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rtc::Thread* const network_thread_;
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int num_pending_partial_reports_;
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int64_t partial_report_timestamp_us_;
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// Reports that are produced on the signaling thread or the network thread are
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// merged into this report. It is only touched on the signaling thread. Once
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// all partial reports are merged this is the result of a request.
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rtc::scoped_refptr<RTCStatsReport> partial_report_;
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std::vector<RequestInfo> requests_;
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// Holds the result of ProducePartialResultsOnNetworkThread(). It is merged
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// into |partial_report_| on the signaling thread and then nulled by
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// MergeNetworkReport_s(). Thread-safety is ensured by using
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// |network_report_event_|.
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rtc::scoped_refptr<RTCStatsReport> network_report_;
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// If set, it is safe to touch the |network_report_| on the signaling thread.
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// This is reset before async-invoking ProducePartialResultsOnNetworkThread()
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// and set when ProducePartialResultsOnNetworkThread() is complete, after it
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// has updated the value of |network_report_|.
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rtc::Event network_report_event_;
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// Set in |GetStatsReport|, read in |ProducePartialResultsOnNetworkThread| and
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// |ProducePartialResultsOnSignalingThread|, reset after work is complete. Not
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// passed as arguments to avoid copies. This is thread safe - when we
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// set/reset we know there are no pending stats requests in progress.
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std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos_;
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std::set<std::string> transport_names_;
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Call::Stats call_stats_;
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// A timestamp, in microseconds, that is based on a timer that is
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// monotonically increasing. That is, even if the system clock is modified the
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// difference between the timer and this timestamp is how fresh the cached
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// report is.
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int64_t cache_timestamp_us_;
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int64_t cache_lifetime_us_;
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rtc::scoped_refptr<const RTCStatsReport> cached_report_;
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// Data recorded and maintained by the stats collector during its lifetime.
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// Some stats are produced from this record instead of other components.
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struct InternalRecord {
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InternalRecord() : data_channels_opened(0), data_channels_closed(0) {}
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// The opened count goes up when a channel is fully opened and the closed
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// count goes up if a previously opened channel has fully closed. The opened
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// count does not go down when a channel closes, meaning (opened - closed)
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// is the number of channels currently opened. A channel that is closed
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// before reaching the open state does not affect these counters.
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uint32_t data_channels_opened;
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uint32_t data_channels_closed;
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// Identifies by address channels that have been opened, which remain in the
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// set until they have been fully closed.
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std::set<uintptr_t> opened_data_channels;
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};
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InternalRecord internal_record_;
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};
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const char* CandidateTypeToRTCIceCandidateTypeForTesting(
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const std::string& type);
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const char* DataStateToRTCDataChannelStateForTesting(
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DataChannelInterface::DataState state);
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} // namespace webrtc
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#endif // PC_RTC_STATS_COLLECTOR_H_
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