RTCStatsCollector relies on PeerConnection and its WebRtcSession. If the PeerConnection is destroyed, reference counting keeps the RTCStatsCollector alive until the request has completed. But the request is using PeerConnection/WebRtcSession resources that are destroyed in ~PeerConnection(). To get around this problem, RTCStatsCollector::WaitForPendingRequest() is added, which is invoked at ~PeerConnection(). Integration test added, it caused a segmentation fault before this change / EXPECT failure. BUG=chromium:627816 Review-Url: https://codereview.webrtc.org/2583613003 Cr-Commit-Position: refs/heads/master@{#15674}
202 lines
7.9 KiB
C++
202 lines
7.9 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 WEBRTC_API_RTCSTATSCOLLECTOR_H_
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#define WEBRTC_API_RTCSTATSCOLLECTOR_H_
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#include <map>
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#include <memory>
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#include <set>
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#include <vector>
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#include "webrtc/api/datachannel.h"
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#include "webrtc/api/datachannelinterface.h"
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#include "webrtc/api/stats/rtcstats_objects.h"
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#include "webrtc/api/stats/rtcstatsreport.h"
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#include "webrtc/base/asyncinvoker.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/base/refcount.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/base/sigslot.h"
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#include "webrtc/base/sslidentity.h"
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#include "webrtc/base/timeutils.h"
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#include "webrtc/media/base/mediachannel.h"
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namespace cricket {
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class Candidate;
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} // namespace cricket
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namespace rtc {
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class SSLCertificate;
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} // namespace rtc
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namespace webrtc {
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class PeerConnection;
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struct SessionStats;
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struct ChannelNamePairs;
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class RTCStatsCollectorCallback : public virtual rtc::RefCountInterface {
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public:
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virtual ~RTCStatsCollectorCallback() {}
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virtual void OnStatsDelivered(
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const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;
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};
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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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PeerConnection* 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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void GetStatsReport(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(PeerConnection* pc, int64_t cache_lifetime_us);
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~RTCStatsCollector();
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// Stats gathering on a particular thread. Calls |AddPartialResults| before
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// returning. Virtual for the sake of testing.
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virtual void ProducePartialResultsOnSignalingThread(int64_t timestamp_us);
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virtual void ProducePartialResultsOnWorkerThread(int64_t timestamp_us);
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virtual void ProducePartialResultsOnNetworkThread(int64_t timestamp_us);
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// Can be called on any thread.
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void AddPartialResults(
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const rtc::scoped_refptr<RTCStatsReport>& partial_report);
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private:
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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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struct MediaInfo {
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rtc::Optional<cricket::VoiceMediaInfo> voice;
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rtc::Optional<cricket::VideoMediaInfo> video;
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};
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void AddPartialResults_s(rtc::scoped_refptr<RTCStatsReport> partial_report);
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void DeliverCachedReport();
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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, const MediaInfo& media_info,
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RTCStatsReport* report) const;
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// Produces |RTCDataChannelStats|.
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void ProduceDataChannelStats_s(
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int64_t timestamp_us, 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, const SessionStats& session_stats,
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RTCStatsReport* report) const;
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// Produces |RTCMediaStreamStats| and |RTCMediaStreamTrackStats|.
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void ProduceMediaStreamAndTrackStats_s(
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int64_t timestamp_us, RTCStatsReport* report) const;
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// Produces |RTCPeerConnectionStats|.
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void ProducePeerConnectionStats_s(
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int64_t timestamp_us, RTCStatsReport* report) const;
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// Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|.
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void ProduceRTPStreamStats_n(
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int64_t timestamp_us, const SessionStats& session_stats,
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const MediaInfo& media_info, RTCStatsReport* report) const;
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// Produces |RTCTransportStats|.
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void ProduceTransportStats_n(
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int64_t timestamp_us, const SessionStats& session_stats,
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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(const SessionStats& session_stats) const;
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std::unique_ptr<MediaInfo> PrepareMediaInfo_s() const;
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// Slots for signals (sigslot) that are wired up to |pc_|.
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void OnDataChannelCreated(DataChannel* channel);
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// Slots for signals (sigslot) that are wired up to |channel|.
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void OnDataChannelOpened(DataChannel* channel);
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void OnDataChannelClosed(DataChannel* channel);
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PeerConnection* 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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rtc::AsyncInvoker invoker_;
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int num_pending_partial_reports_;
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int64_t partial_report_timestamp_us_;
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rtc::scoped_refptr<RTCStatsReport> partial_report_;
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std::vector<rtc::scoped_refptr<RTCStatsCollectorCallback>> callbacks_;
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// Set in |GetStatsReport|, used in |ProducePartialResultsOnNetworkThread|
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// (not passed as arguments to avoid copies). This is thread safe - it is set
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// at the start of |GetStatsReport| after making sure there are no pending
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// stats requests in progress.
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std::unique_ptr<ChannelNamePairs> channel_name_pairs_;
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std::unique_ptr<MediaInfo> media_info_;
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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),
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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 // WEBRTC_API_RTCSTATSCOLLECTOR_H_
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