Attachment 348670 Details for Bug 189180 – Patch for landing

[patch] Patch for landing

bug-189180-20180831134652.patch (text/plain), 47.55 KB, created by youenn fablet on 2018-08-31 13:46:53 PDT

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Creator: youenn fablet

Created: 2018-08-31 13:46:53 PDT

Size: 47.55 KB

patch

obsolete

>Subversion Revision: 235563
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 0df6234b102ea5d0256590046645c4cc47e2f2f3..6eb68cd8c3ffed65f3457f7f38d11938db22402b 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,40 @@
>+2018-08-31  Youenn Fablet  <youenn@apple.com>
>+
>+        Move stats gathering out of LibWebRTCMediaEndpoint
>+        https://bugs.webkit.org/show_bug.cgi?id=189180
>+
>+        Reviewed by Alejandro G. Castro.
>+
>+        Move stats gathering in LibWebRTCStatsCollector.
>+        Make sure that the lambda given to the collector is always called and destroyed from the main thread.
>+        This allows capturing the promise here instead of storing it into the peer connection backend.
>+        No change of behavior.
>+
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::getStats):
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp:
>+        (WebCore::LibWebRTCPeerConnectionBackend::getStats):
>+        (WebCore::LibWebRTCPeerConnectionBackend::doStop):
>+        * Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp: Added.
>+        (WebCore::LibWebRTCStatsCollector::LibWebRTCStatsCollector):
>+        (WebCore::LibWebRTCStatsCollector::~LibWebRTCStatsCollector):
>+        (WebCore::fromStdString):
>+        (WebCore::fillRTCStats):
>+        (WebCore::fillRTCRTPStreamStats):
>+        (WebCore::fillInboundRTPStreamStats):
>+        (WebCore::fillOutboundRTPStreamStats):
>+        (WebCore::fillRTCMediaStreamTrackStats):
>+        (WebCore::fillRTCDataChannelStats):
>+        (WebCore::iceCandidatePairState):
>+        (WebCore::fillRTCIceCandidatePairStats):
>+        (WebCore::fillRTCCertificateStats):
>+        (WebCore::LibWebRTCStatsCollector::OnStatsDelivered):
>+        * Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.h: Added.
>+        (WebCore::LibWebRTCStatsCollector::create):
>+        * WebCore.xcodeproj/project.pbxproj:
>+
> 2018-08-31  Youenn Fablet  <youenn@apple.com>
> 
>         IDBRequest and IDBCursor should not hold strong references to JSValue
>diff --git a/Source/WebCore/CMakeLists.txt b/Source/WebCore/CMakeLists.txt
>index fef997632a68c410af098f01eb2c58cdfcdab58c..29bc004149387c176881aa971aecc3ec4a86bf80 100644
>--- a/Source/WebCore/CMakeLists.txt
>+++ b/Source/WebCore/CMakeLists.txt
>@@ -1560,6 +1560,7 @@ if (USE_LIBWEBRTC)
>       "${THIRDPARTY_DIR}/libwebrtc/Source/third_party/abseil-cpp")
>   list(APPEND WebCore_LIBRARIES webrtc)
>   list(APPEND WebCore_SOURCES
>+      Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp
>       Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.cpp
>       Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>       Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index 7f675f39ee09089d4213e7304a4df0fc38d29e9c..6a54faaad1930786ce7c28fa8d6dc0ec18f1d676 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -1,5 +1,5 @@
> /*
>- * Copyright (C) 2017 Apple Inc. All rights reserved.
>+ * Copyright (C) 2017-2018 Apple Inc. All rights reserved.
>  *
>  * Redistribution and use in source and binary forms, with or without
>  * modification, are permitted provided that the following conditions
>@@ -32,6 +32,7 @@
> #include "LibWebRTCDataChannelHandler.h"
> #include "LibWebRTCPeerConnectionBackend.h"
> #include "LibWebRTCProvider.h"
>+#include "LibWebRTCStatsCollector.h"
> #include "Logging.h"
> #include "NotImplemented.h"
> #include "Performance.h"
>@@ -305,299 +306,19 @@ void LibWebRTCMediaEndpoint::doCreateAnswer()
>     m_backend->CreateAnswer(&m_createSessionDescriptionObserver, nullptr);
> }
> 
>-void LibWebRTCMediaEndpoint::getStats(MediaStreamTrack* track, const DeferredPromise& promise)
>+void LibWebRTCMediaEndpoint::getStats(MediaStreamTrack*, Ref<DeferredPromise>&& promise)
> {
>-    auto collector = StatsCollector::create(*this, promise, track);
>-    LibWebRTCProvider::callOnWebRTCSignalingThread([protectedThis = makeRef(*this), collector = WTFMove(collector)] {
>-        if (protectedThis->m_backend)
>-            protectedThis->m_backend->GetStats(collector.get());
>-    });
>-}
>-
>-LibWebRTCMediaEndpoint::StatsCollector::StatsCollector(Ref<LibWebRTCMediaEndpoint>&& endpoint, const DeferredPromise& promise, MediaStreamTrack* track)
>-    : m_endpoint(WTFMove(endpoint))
>-    , m_promise(promise)
>-{
>-    if (track)
>-        m_id = track->id();
>-}
>-
>-static inline void fillRTCStats(RTCStatsReport::Stats& stats, const webrtc::RTCStats& rtcStats)
>-{
>-    stats.timestamp = Performance::reduceTimeResolution(Seconds::fromMicroseconds(rtcStats.timestamp_us())).milliseconds();
>-    stats.id = fromStdString(rtcStats.id());
>-}
>-
>-static inline void fillRTCRTPStreamStats(RTCStatsReport::RTCRTPStreamStats& stats, const webrtc::RTCRTPStreamStats& rtcStats)
>-{
>-    fillRTCStats(stats, rtcStats);
>-
>-    if (rtcStats.ssrc.is_defined())
>-        stats.ssrc = *rtcStats.ssrc;
>-    if (rtcStats.associate_stats_id.is_defined())
>-        stats.associateStatsId = fromStdString(*rtcStats.associate_stats_id);
>-    if (rtcStats.is_remote.is_defined())
>-        stats.isRemote = *rtcStats.is_remote;
>-    if (rtcStats.media_type.is_defined())
>-        stats.mediaType = fromStdString(*rtcStats.media_type);
>-    if (rtcStats.track_id.is_defined())
>-        stats.mediaTrackId = fromStdString(*rtcStats.track_id);
>-    if (rtcStats.transport_id.is_defined())
>-        stats.transportId = fromStdString(*rtcStats.transport_id);
>-    if (rtcStats.codec_id.is_defined())
>-        stats.codecId = fromStdString(*rtcStats.codec_id);
>-    if (rtcStats.fir_count.is_defined())
>-        stats.firCount = *rtcStats.fir_count;
>-    if (rtcStats.pli_count.is_defined())
>-        stats.pliCount = *rtcStats.pli_count;
>-    if (rtcStats.nack_count.is_defined())
>-        stats.nackCount = *rtcStats.nack_count;
>-    if (rtcStats.sli_count.is_defined())
>-        stats.sliCount = *rtcStats.sli_count;
>-    if (rtcStats.qp_sum.is_defined())
>-        stats.qpSum = *rtcStats.qp_sum;
>-    stats.qpSum = 0;
>-}
>-
>-static inline void fillInboundRTPStreamStats(RTCStatsReport::InboundRTPStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
>-{
>-    fillRTCRTPStreamStats(stats, rtcStats);
>-
>-    if (rtcStats.packets_received.is_defined())
>-        stats.packetsReceived = *rtcStats.packets_received;
>-    if (rtcStats.bytes_received.is_defined())
>-        stats.bytesReceived = *rtcStats.bytes_received;
>-    if (rtcStats.packets_lost.is_defined())
>-        stats.packetsLost = *rtcStats.packets_lost;
>-    if (rtcStats.jitter.is_defined())
>-        stats.jitter = *rtcStats.jitter;
>-    if (rtcStats.fraction_lost.is_defined())
>-        stats.fractionLost = *rtcStats.fraction_lost;
>-    if (rtcStats.packets_discarded.is_defined())
>-        stats.packetsDiscarded = *rtcStats.packets_discarded;
>-    if (rtcStats.packets_repaired.is_defined())
>-        stats.packetsRepaired = *rtcStats.packets_repaired;
>-    if (rtcStats.burst_packets_lost.is_defined())
>-        stats.burstPacketsLost = *rtcStats.burst_packets_lost;
>-    if (rtcStats.burst_packets_discarded.is_defined())
>-        stats.burstPacketsDiscarded = *rtcStats.burst_packets_discarded;
>-    if (rtcStats.burst_loss_count.is_defined())
>-        stats.burstLossCount = *rtcStats.burst_loss_count;
>-    if (rtcStats.burst_discard_count.is_defined())
>-        stats.burstDiscardCount = *rtcStats.burst_discard_count;
>-    if (rtcStats.burst_loss_rate.is_defined())
>-        stats.burstLossRate = *rtcStats.burst_loss_rate;
>-    if (rtcStats.burst_discard_rate.is_defined())
>-        stats.burstDiscardRate = *rtcStats.burst_discard_rate;
>-    if (rtcStats.gap_loss_rate.is_defined())
>-        stats.gapLossRate = *rtcStats.gap_loss_rate;
>-    if (rtcStats.gap_discard_rate.is_defined())
>-        stats.gapDiscardRate = *rtcStats.gap_discard_rate;
>-    if (rtcStats.frames_decoded.is_defined())
>-        stats.framesDecoded = *rtcStats.frames_decoded;
>-}
>+    auto collector = LibWebRTCStatsCollector::create([promise = WTFMove(promise), protectedThis = makeRef(*this)](auto&& report) mutable {
>+        ASSERT(isMainThread());
>+        if (protectedThis->isStopped() || !report)
>+            return false;
> 
>-static inline void fillOutboundRTPStreamStats(RTCStatsReport::OutboundRTPStreamStats& stats, const webrtc::RTCOutboundRTPStreamStats& rtcStats)
>-{
>-    fillRTCRTPStreamStats(stats, rtcStats);
>-
>-    if (rtcStats.packets_sent.is_defined())
>-        stats.packetsSent = *rtcStats.packets_sent;
>-    if (rtcStats.bytes_sent.is_defined())
>-        stats.bytesSent = *rtcStats.bytes_sent;
>-    if (rtcStats.target_bitrate.is_defined())
>-        stats.targetBitrate = *rtcStats.target_bitrate;
>-    if (rtcStats.frames_encoded.is_defined())
>-        stats.framesEncoded = *rtcStats.frames_encoded;
>-}
>-
>-static inline void fillRTCMediaStreamTrackStats(RTCStatsReport::MediaStreamTrackStats& stats, const webrtc::RTCMediaStreamTrackStats& rtcStats)
>-{
>-    fillRTCStats(stats, rtcStats);
>-
>-    if (rtcStats.track_identifier.is_defined())
>-        stats.trackIdentifier = fromStdString(*rtcStats.track_identifier);
>-    if (rtcStats.remote_source.is_defined())
>-        stats.remoteSource = *rtcStats.remote_source;
>-    if (rtcStats.ended.is_defined())
>-        stats.ended = *rtcStats.ended;
>-    if (rtcStats.detached.is_defined())
>-        stats.detached = *rtcStats.detached;
>-    if (rtcStats.frame_width.is_defined())
>-        stats.frameWidth = *rtcStats.frame_width;
>-    if (rtcStats.frame_height.is_defined())
>-        stats.frameHeight = *rtcStats.frame_height;
>-    if (rtcStats.frames_per_second.is_defined())
>-        stats.framesPerSecond = *rtcStats.frames_per_second;
>-    if (rtcStats.frames_sent.is_defined())
>-        stats.framesSent = *rtcStats.frames_sent;
>-    if (rtcStats.frames_received.is_defined())
>-        stats.framesReceived = *rtcStats.frames_received;
>-    if (rtcStats.frames_decoded.is_defined())
>-        stats.framesDecoded = *rtcStats.frames_decoded;
>-    if (rtcStats.frames_dropped.is_defined())
>-        stats.framesDropped = *rtcStats.frames_dropped;
>-    if (rtcStats.partial_frames_lost.is_defined())
>-        stats.partialFramesLost = *rtcStats.partial_frames_lost;
>-    if (rtcStats.full_frames_lost.is_defined())
>-        stats.fullFramesLost = *rtcStats.full_frames_lost;
>-    if (rtcStats.audio_level.is_defined())
>-        stats.audioLevel = *rtcStats.audio_level;
>-    if (rtcStats.echo_return_loss.is_defined())
>-        stats.echoReturnLoss = *rtcStats.echo_return_loss;
>-    if (rtcStats.echo_return_loss_enhancement.is_defined())
>-        stats.echoReturnLossEnhancement = *rtcStats.echo_return_loss_enhancement;
>-}
>-
>-static inline void fillRTCDataChannelStats(RTCStatsReport::DataChannelStats& stats, const webrtc::RTCDataChannelStats& rtcStats)
>-{
>-    fillRTCStats(stats, rtcStats);
>-
>-    if (rtcStats.label.is_defined())
>-        stats.label = fromStdString(*rtcStats.label);
>-    if (rtcStats.protocol.is_defined())
>-        stats.protocol = fromStdString(*rtcStats.protocol);
>-    if (rtcStats.datachannelid.is_defined())
>-        stats.datachannelid = *rtcStats.datachannelid;
>-    if (rtcStats.state.is_defined())
>-        stats.state = fromStdString(*rtcStats.state);
>-    if (rtcStats.messages_sent.is_defined())
>-        stats.messagesSent = *rtcStats.messages_sent;
>-    if (rtcStats.bytes_sent.is_defined())
>-        stats.bytesSent = *rtcStats.bytes_sent;
>-    if (rtcStats.messages_received.is_defined())
>-        stats.messagesReceived = *rtcStats.messages_received;
>-    if (rtcStats.bytes_received.is_defined())
>-        stats.bytesReceived = *rtcStats.bytes_received;
>-}
>-
>-static inline RTCStatsReport::IceCandidatePairState iceCandidatePairState(const std::string& state)
>-{
>-    if (state == "frozen")
>-        return RTCStatsReport::IceCandidatePairState::Frozen;
>-    if (state == "waiting")
>-        return RTCStatsReport::IceCandidatePairState::Waiting;
>-    if (state == "in-progress")
>-        return RTCStatsReport::IceCandidatePairState::Inprogress;
>-    if (state == "failed")
>-        return RTCStatsReport::IceCandidatePairState::Failed;
>-    if (state == "succeeded")
>-        return RTCStatsReport::IceCandidatePairState::Succeeded;
>-    if (state == "cancelled")
>-        return RTCStatsReport::IceCandidatePairState::Cancelled;
>-    ASSERT_NOT_REACHED();
>-    return RTCStatsReport::IceCandidatePairState::Frozen;
>-}
>-
>-static inline void fillRTCIceCandidatePairStats(RTCStatsReport::IceCandidatePairStats& stats, const webrtc::RTCIceCandidatePairStats& rtcStats)
>-{
>-    fillRTCStats(stats, rtcStats);
>-
>-    if (rtcStats.transport_id.is_defined())
>-        stats.transportId = fromStdString(*rtcStats.transport_id);
>-    if (rtcStats.local_candidate_id.is_defined())
>-        stats.localCandidateId = fromStdString(*rtcStats.local_candidate_id);
>-    if (rtcStats.remote_candidate_id.is_defined())
>-        stats.remoteCandidateId = fromStdString(*rtcStats.remote_candidate_id);
>-    if (rtcStats.state.is_defined())
>-        stats.state = iceCandidatePairState(*rtcStats.state);
>-
>-    if (rtcStats.priority.is_defined())
>-        stats.priority = *rtcStats.priority;
>-    if (rtcStats.nominated.is_defined())
>-        stats.nominated = *rtcStats.nominated;
>-    if (rtcStats.writable.is_defined())
>-        stats.writable = *rtcStats.writable;
>-    if (rtcStats.readable.is_defined())
>-        stats.readable = *rtcStats.readable;
>-
>-    if (rtcStats.bytes_sent.is_defined())
>-        stats.bytesSent = *rtcStats.bytes_sent;
>-    if (rtcStats.bytes_received.is_defined())
>-        stats.bytesReceived = *rtcStats.bytes_received;
>-    if (rtcStats.total_round_trip_time.is_defined())
>-        stats.totalRoundTripTime = *rtcStats.total_round_trip_time;
>-    if (rtcStats.current_round_trip_time.is_defined())
>-        stats.currentRoundTripTime = *rtcStats.current_round_trip_time;
>-    if (rtcStats.available_outgoing_bitrate.is_defined())
>-        stats.availableOutgoingBitrate = *rtcStats.available_outgoing_bitrate;
>-    if (rtcStats.available_incoming_bitrate.is_defined())
>-        stats.availableIncomingBitrate = *rtcStats.available_incoming_bitrate;
>-
>-    if (rtcStats.requests_received.is_defined())
>-        stats.requestsReceived = *rtcStats.requests_received;
>-    if (rtcStats.requests_sent.is_defined())
>-        stats.requestsSent = *rtcStats.requests_sent;
>-    if (rtcStats.responses_received.is_defined())
>-        stats.responsesReceived = *rtcStats.responses_received;
>-    if (rtcStats.responses_sent.is_defined())
>-        stats.responsesSent = *rtcStats.responses_sent;
>-
>-    if (rtcStats.requests_received.is_defined())
>-        stats.retransmissionsReceived = *rtcStats.requests_received;
>-    if (rtcStats.requests_sent.is_defined())
>-        stats.retransmissionsSent = *rtcStats.requests_sent;
>-    if (rtcStats.responses_received.is_defined())
>-        stats.consentRequestsReceived = *rtcStats.responses_received;
>-    if (rtcStats.responses_sent.is_defined())
>-        stats.consentRequestsSent = *rtcStats.responses_sent;
>-    if (rtcStats.responses_received.is_defined())
>-        stats.consentResponsesReceived = *rtcStats.responses_received;
>-    if (rtcStats.responses_sent.is_defined())
>-        stats.consentResponsesSent = *rtcStats.responses_sent;
>-}
>-
>-static inline void fillRTCCertificateStats(RTCStatsReport::CertificateStats& stats, const webrtc::RTCCertificateStats& rtcStats)
>-{
>-    fillRTCStats(stats, rtcStats);
>-
>-    if (rtcStats.fingerprint.is_defined())
>-        stats.fingerprint = fromStdString(*rtcStats.fingerprint);
>-    if (rtcStats.fingerprint_algorithm.is_defined())
>-        stats.fingerprintAlgorithm = fromStdString(*rtcStats.fingerprint_algorithm);
>-    if (rtcStats.base64_certificate.is_defined())
>-        stats.base64Certificate = fromStdString(*rtcStats.base64_certificate);
>-    if (rtcStats.issuer_certificate_id.is_defined())
>-        stats.issuerCertificateId = fromStdString(*rtcStats.issuer_certificate_id);
>-}
>-
>-void LibWebRTCMediaEndpoint::StatsCollector::OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>& rtcReport)
>-{
>-    callOnMainThread([protectedThis = rtc::scoped_refptr<LibWebRTCMediaEndpoint::StatsCollector>(this), rtcReport] {
>-        if (protectedThis->m_endpoint->isStopped())
>-            return;
>-
>-        auto report = RTCStatsReport::create();
>-        protectedThis->m_endpoint->m_peerConnectionBackend.getStatsSucceeded(protectedThis->m_promise, report.copyRef());
>-        ASSERT(report->backingMap());
>-
>-        for (const auto& rtcStats : *rtcReport) {
>-            if (rtcStats.type() == webrtc::RTCInboundRTPStreamStats::kType) {
>-                RTCStatsReport::InboundRTPStreamStats stats;
>-                fillInboundRTPStreamStats(stats, static_cast<const webrtc::RTCInboundRTPStreamStats&>(rtcStats));
>-                report->addStats<IDLDictionary<RTCStatsReport::InboundRTPStreamStats>>(WTFMove(stats));
>-            } else if (rtcStats.type() == webrtc::RTCOutboundRTPStreamStats::kType) {
>-                RTCStatsReport::OutboundRTPStreamStats stats;
>-                fillOutboundRTPStreamStats(stats, static_cast<const webrtc::RTCOutboundRTPStreamStats&>(rtcStats));
>-                report->addStats<IDLDictionary<RTCStatsReport::OutboundRTPStreamStats>>(WTFMove(stats));
>-            } else if (rtcStats.type() == webrtc::RTCMediaStreamTrackStats::kType) {
>-                RTCStatsReport::MediaStreamTrackStats stats;
>-                fillRTCMediaStreamTrackStats(stats, static_cast<const webrtc::RTCMediaStreamTrackStats&>(rtcStats));
>-                report->addStats<IDLDictionary<RTCStatsReport::MediaStreamTrackStats>>(WTFMove(stats));
>-            } else if (rtcStats.type() == webrtc::RTCDataChannelStats::kType) {
>-                RTCStatsReport::DataChannelStats stats;
>-                fillRTCDataChannelStats(stats, static_cast<const webrtc::RTCDataChannelStats&>(rtcStats));
>-                report->addStats<IDLDictionary<RTCStatsReport::DataChannelStats>>(WTFMove(stats));
>-            } else if (rtcStats.type() == webrtc::RTCIceCandidatePairStats::kType) {
>-                RTCStatsReport::IceCandidatePairStats stats;
>-                fillRTCIceCandidatePairStats(stats, static_cast<const webrtc::RTCIceCandidatePairStats&>(rtcStats));
>-                report->addStats<IDLDictionary<RTCStatsReport::IceCandidatePairStats>>(WTFMove(stats));
>-            } else if (rtcStats.type() == webrtc::RTCCertificateStats::kType) {
>-                RTCStatsReport::CertificateStats stats;
>-                fillRTCCertificateStats(stats, static_cast<const webrtc::RTCCertificateStats&>(rtcStats));
>-                report->addStats<IDLDictionary<RTCStatsReport::CertificateStats>>(WTFMove(stats));
>-            }
>-        }
>+        promise->resolve<IDLInterface<RTCStatsReport>>(report.releaseNonNull());
>+        return true;
>+    });
>+    LibWebRTCProvider::callOnWebRTCSignalingThread([this, collector = WTFMove(collector)] {
>+        if (m_backend)
>+            m_backend->GetStats(collector.get());
>     });
> }
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>index 8ae48de6304a49b4447f43bfc2905c72c2fafc77..b717495f18c5cb3c78c22bfab9eadeee4eaf73d8 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>@@ -1,5 +1,5 @@
> /*
>- * Copyright (C) 2017 Apple Inc. All rights reserved.
>+ * Copyright (C) 2017-2018 Apple Inc. All rights reserved.
>  *
>  * Redistribution and use in source and binary forms, with or without
>  * modification, are permitted provided that the following conditions
>@@ -83,7 +83,7 @@ public:
>     void doSetRemoteDescription(RTCSessionDescription&);
>     void doCreateOffer(const RTCOfferOptions&);
>     void doCreateAnswer();
>-    void getStats(MediaStreamTrack*, const DeferredPromise&);
>+    void getStats(MediaStreamTrack*, Ref<DeferredPromise>&&);
>     std::unique_ptr<RTCDataChannelHandler> createDataChannel(const String&, const RTCDataChannelInit&);
>     bool addIceCandidate(webrtc::IceCandidateInterface& candidate) { return m_backend->AddIceCandidate(&candidate); }
> 
>@@ -154,20 +154,6 @@ private:
>     Seconds statsLogInterval(int64_t) const;
> #endif
> 
>-    class StatsCollector : public webrtc::RTCStatsCollectorCallback {
>-    public:
>-        static rtc::scoped_refptr<StatsCollector> create(Ref<LibWebRTCMediaEndpoint>&& endpoint, const DeferredPromise& promise, MediaStreamTrack* track) { return new rtc::RefCountedObject<StatsCollector>(WTFMove(endpoint), promise, track); }
>-
>-        StatsCollector(Ref<LibWebRTCMediaEndpoint>&&, const DeferredPromise&, MediaStreamTrack*);
>-
>-    private:
>-        void OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>&) final;
>-
>-        Ref<LibWebRTCMediaEndpoint> m_endpoint;
>-        const DeferredPromise& m_promise;
>-        String m_id;
>-    };
>-
>     LibWebRTCPeerConnectionBackend& m_peerConnectionBackend;
>     webrtc::PeerConnectionFactoryInterface& m_peerConnectionFactory;
>     rtc::scoped_refptr<webrtc::PeerConnectionInterface> m_backend;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>index 22ba150c4b464dc29abf622e79b19f6d8fc3217c..4580a4476bb45565109537278614cdd55eb65ec6 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.cpp
>@@ -1,5 +1,5 @@
> /*
>- * Copyright (C) 2017 Apple Inc.
>+ * Copyright (C) 2017-2018 Apple Inc.
>  *
>  * Redistribution and use in source and binary forms, with or without
>  * modification, are permitted provided that the following conditions
>@@ -29,7 +29,6 @@
> 
> #include "Document.h"
> #include "IceCandidate.h"
>-#include "JSRTCStatsReport.h"
> #include "LibWebRTCDataChannelHandler.h"
> #include "LibWebRTCMediaEndpoint.h"
> #include "MediaEndpointConfiguration.h"
>@@ -128,26 +127,7 @@ bool LibWebRTCPeerConnectionBackend::setConfiguration(MediaEndpointConfiguration
> 
> void LibWebRTCPeerConnectionBackend::getStats(MediaStreamTrack* track, Ref<DeferredPromise>&& promise)
> {
>-    if (m_endpoint->isStopped())
>-        return;
>-
>-    auto& statsPromise = promise.get();
>-    m_statsPromises.add(&statsPromise, WTFMove(promise));
>-    m_endpoint->getStats(track, statsPromise);
>-}
>-
>-void LibWebRTCPeerConnectionBackend::getStatsSucceeded(const DeferredPromise& promise, Ref<RTCStatsReport>&& report)
>-{
>-    auto statsPromise = m_statsPromises.take(&promise);
>-    ASSERT(statsPromise);
>-    statsPromise.value()->resolve<IDLInterface<RTCStatsReport>>(WTFMove(report));
>-}
>-
>-void LibWebRTCPeerConnectionBackend::getStatsFailed(const DeferredPromise& promise, Exception&& exception)
>-{
>-    auto statsPromise = m_statsPromises.take(&promise);
>-    ASSERT(statsPromise);
>-    statsPromise.value()->reject(WTFMove(exception));
>+    m_endpoint->getStats(track, WTFMove(promise));
> }
> 
> void LibWebRTCPeerConnectionBackend::doSetLocalDescription(RTCSessionDescription& description)
>@@ -200,7 +180,6 @@ void LibWebRTCPeerConnectionBackend::doStop()
> 
>     m_audioSources.clear();
>     m_videoSources.clear();
>-    m_statsPromises.clear();
>     m_pendingReceivers.clear();
> }
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>index fa0f0137095252612f4b0eb6116484d14fa33101..437b7e60c004ff476598d2936ba67eebd05c6f0c 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCPeerConnectionBackend.h
>@@ -1,5 +1,5 @@
> /*
>- * Copyright (C) 2017 Apple Inc. All rights reserved.
>+ * Copyright (C) 2017-2018 Apple Inc. All rights reserved.
>  *
>  * Redistribution and use in source and binary forms, with or without
>  * modification, are permitted provided that the following conditions
>@@ -86,7 +86,6 @@ private:
>     void addVideoSource(Ref<RealtimeOutgoingVideoSource>&&);
> 
>     void getStatsSucceeded(const DeferredPromise&, Ref<RTCStatsReport>&&);
>-    void getStatsFailed(const DeferredPromise&, Exception&&);
> 
>     bool notifyAddedTrack(RTCRtpSender&) final;
>     void notifyRemovedTrack(RTCRtpSender&) final;
>@@ -112,7 +111,6 @@ private:
>     Vector<std::unique_ptr<webrtc::IceCandidateInterface>> m_pendingCandidates;
>     Vector<Ref<RealtimeOutgoingAudioSource>> m_audioSources;
>     Vector<Ref<RealtimeOutgoingVideoSource>> m_videoSources;
>-    HashMap<const DeferredPromise*, Ref<DeferredPromise>> m_statsPromises;
>     Vector<Ref<RTCRtpReceiver>> m_pendingReceivers;
> };
> 
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..47df7bedb51a92cb1e71933136169b80b1763df8
>--- /dev/null
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.cpp
>@@ -0,0 +1,336 @@
>+/*
>+ * Copyright (C) 2018 Apple Inc. All rights reserved.
>+ *
>+ * Redistribution and use in source and binary forms, with or without
>+ * modification, are permitted provided that the following conditions
>+ * are met:
>+ * 1.  Redistributions of source code must retain the above copyright
>+ *     notice, this list of conditions and the following disclaimer.
>+ * 2.  Redistributions in binary form must reproduce the above copyright
>+ *     notice, this list of conditions and the following disclaimer in the
>+ *     documentation and/or other materials provided with the distribution.
>+ *
>+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
>+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
>+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
>+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
>+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
>+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
>+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
>+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
>+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
>+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
>+ */
>+
>+#include "config.h"
>+#include "LibWebRTCStatsCollector.h"
>+
>+#if USE(LIBWEBRTC)
>+
>+#include "JSRTCStatsReport.h"
>+#include "Performance.h"
>+#include <wtf/MainThread.h>
>+
>+namespace WebCore {
>+
>+LibWebRTCStatsCollector::LibWebRTCStatsCollector(CollectorCallback&& callback)
>+    : m_callback(WTFMove(callback))
>+{
>+}
>+
>+LibWebRTCStatsCollector::~LibWebRTCStatsCollector()
>+{
>+    if (!m_callback)
>+        return;
>+
>+    callOnMainThread([callback = WTFMove(m_callback)]() mutable {
>+        callback({ });
>+    });
>+}
>+
>+static inline String fromStdString(const std::string& value)
>+{
>+    return String::fromUTF8(value.data(), value.length());
>+}
>+
>+static inline void fillRTCStats(RTCStatsReport::Stats& stats, const webrtc::RTCStats& rtcStats)
>+{
>+    stats.timestamp = Performance::reduceTimeResolution(Seconds::fromMicroseconds(rtcStats.timestamp_us())).milliseconds();
>+    stats.id = fromStdString(rtcStats.id());
>+}
>+
>+static inline void fillRTCRTPStreamStats(RTCStatsReport::RTCRTPStreamStats& stats, const webrtc::RTCRTPStreamStats& rtcStats)
>+{
>+    fillRTCStats(stats, rtcStats);
>+
>+    if (rtcStats.ssrc.is_defined())
>+        stats.ssrc = *rtcStats.ssrc;
>+    if (rtcStats.associate_stats_id.is_defined())
>+        stats.associateStatsId = fromStdString(*rtcStats.associate_stats_id);
>+    if (rtcStats.is_remote.is_defined())
>+        stats.isRemote = *rtcStats.is_remote;
>+    if (rtcStats.media_type.is_defined())
>+        stats.mediaType = fromStdString(*rtcStats.media_type);
>+    if (rtcStats.track_id.is_defined())
>+        stats.mediaTrackId = fromStdString(*rtcStats.track_id);
>+    if (rtcStats.transport_id.is_defined())
>+        stats.transportId = fromStdString(*rtcStats.transport_id);
>+    if (rtcStats.codec_id.is_defined())
>+        stats.codecId = fromStdString(*rtcStats.codec_id);
>+    if (rtcStats.fir_count.is_defined())
>+        stats.firCount = *rtcStats.fir_count;
>+    if (rtcStats.pli_count.is_defined())
>+        stats.pliCount = *rtcStats.pli_count;
>+    if (rtcStats.nack_count.is_defined())
>+        stats.nackCount = *rtcStats.nack_count;
>+    if (rtcStats.sli_count.is_defined())
>+        stats.sliCount = *rtcStats.sli_count;
>+    if (rtcStats.qp_sum.is_defined())
>+        stats.qpSum = *rtcStats.qp_sum;
>+    stats.qpSum = 0;
>+}
>+
>+static inline void fillInboundRTPStreamStats(RTCStatsReport::InboundRTPStreamStats& stats, const webrtc::RTCInboundRTPStreamStats& rtcStats)
>+{
>+    fillRTCRTPStreamStats(stats, rtcStats);
>+
>+    if (rtcStats.packets_received.is_defined())
>+        stats.packetsReceived = *rtcStats.packets_received;
>+    if (rtcStats.bytes_received.is_defined())
>+        stats.bytesReceived = *rtcStats.bytes_received;
>+    if (rtcStats.packets_lost.is_defined())
>+        stats.packetsLost = *rtcStats.packets_lost;
>+    if (rtcStats.jitter.is_defined())
>+        stats.jitter = *rtcStats.jitter;
>+    if (rtcStats.fraction_lost.is_defined())
>+        stats.fractionLost = *rtcStats.fraction_lost;
>+    if (rtcStats.packets_discarded.is_defined())
>+        stats.packetsDiscarded = *rtcStats.packets_discarded;
>+    if (rtcStats.packets_repaired.is_defined())
>+        stats.packetsRepaired = *rtcStats.packets_repaired;
>+    if (rtcStats.burst_packets_lost.is_defined())
>+        stats.burstPacketsLost = *rtcStats.burst_packets_lost;
>+    if (rtcStats.burst_packets_discarded.is_defined())
>+        stats.burstPacketsDiscarded = *rtcStats.burst_packets_discarded;
>+    if (rtcStats.burst_loss_count.is_defined())
>+        stats.burstLossCount = *rtcStats.burst_loss_count;
>+    if (rtcStats.burst_discard_count.is_defined())
>+        stats.burstDiscardCount = *rtcStats.burst_discard_count;
>+    if (rtcStats.burst_loss_rate.is_defined())
>+        stats.burstLossRate = *rtcStats.burst_loss_rate;
>+    if (rtcStats.burst_discard_rate.is_defined())
>+        stats.burstDiscardRate = *rtcStats.burst_discard_rate;
>+    if (rtcStats.gap_loss_rate.is_defined())
>+        stats.gapLossRate = *rtcStats.gap_loss_rate;
>+    if (rtcStats.gap_discard_rate.is_defined())
>+        stats.gapDiscardRate = *rtcStats.gap_discard_rate;
>+    if (rtcStats.frames_decoded.is_defined())
>+        stats.framesDecoded = *rtcStats.frames_decoded;
>+}
>+
>+static inline void fillOutboundRTPStreamStats(RTCStatsReport::OutboundRTPStreamStats& stats, const webrtc::RTCOutboundRTPStreamStats& rtcStats)
>+{
>+    fillRTCRTPStreamStats(stats, rtcStats);
>+
>+    if (rtcStats.packets_sent.is_defined())
>+        stats.packetsSent = *rtcStats.packets_sent;
>+    if (rtcStats.bytes_sent.is_defined())
>+        stats.bytesSent = *rtcStats.bytes_sent;
>+    if (rtcStats.target_bitrate.is_defined())
>+        stats.targetBitrate = *rtcStats.target_bitrate;
>+    if (rtcStats.frames_encoded.is_defined())
>+        stats.framesEncoded = *rtcStats.frames_encoded;
>+}
>+
>+static inline void fillRTCMediaStreamTrackStats(RTCStatsReport::MediaStreamTrackStats& stats, const webrtc::RTCMediaStreamTrackStats& rtcStats)
>+{
>+    fillRTCStats(stats, rtcStats);
>+
>+    if (rtcStats.track_identifier.is_defined())
>+        stats.trackIdentifier = fromStdString(*rtcStats.track_identifier);
>+    if (rtcStats.remote_source.is_defined())
>+        stats.remoteSource = *rtcStats.remote_source;
>+    if (rtcStats.ended.is_defined())
>+        stats.ended = *rtcStats.ended;
>+    if (rtcStats.detached.is_defined())
>+        stats.detached = *rtcStats.detached;
>+    if (rtcStats.frame_width.is_defined())
>+        stats.frameWidth = *rtcStats.frame_width;
>+    if (rtcStats.frame_height.is_defined())
>+        stats.frameHeight = *rtcStats.frame_height;
>+    if (rtcStats.frames_per_second.is_defined())
>+        stats.framesPerSecond = *rtcStats.frames_per_second;
>+    if (rtcStats.frames_sent.is_defined())
>+        stats.framesSent = *rtcStats.frames_sent;
>+    if (rtcStats.frames_received.is_defined())
>+        stats.framesReceived = *rtcStats.frames_received;
>+    if (rtcStats.frames_decoded.is_defined())
>+        stats.framesDecoded = *rtcStats.frames_decoded;
>+    if (rtcStats.frames_dropped.is_defined())
>+        stats.framesDropped = *rtcStats.frames_dropped;
>+    if (rtcStats.partial_frames_lost.is_defined())
>+        stats.partialFramesLost = *rtcStats.partial_frames_lost;
>+    if (rtcStats.full_frames_lost.is_defined())
>+        stats.fullFramesLost = *rtcStats.full_frames_lost;
>+    if (rtcStats.audio_level.is_defined())
>+        stats.audioLevel = *rtcStats.audio_level;
>+    if (rtcStats.echo_return_loss.is_defined())
>+        stats.echoReturnLoss = *rtcStats.echo_return_loss;
>+    if (rtcStats.echo_return_loss_enhancement.is_defined())
>+        stats.echoReturnLossEnhancement = *rtcStats.echo_return_loss_enhancement;
>+}
>+
>+static inline void fillRTCDataChannelStats(RTCStatsReport::DataChannelStats& stats, const webrtc::RTCDataChannelStats& rtcStats)
>+{
>+    fillRTCStats(stats, rtcStats);
>+
>+    if (rtcStats.label.is_defined())
>+        stats.label = fromStdString(*rtcStats.label);
>+    if (rtcStats.protocol.is_defined())
>+        stats.protocol = fromStdString(*rtcStats.protocol);
>+    if (rtcStats.datachannelid.is_defined())
>+        stats.datachannelid = *rtcStats.datachannelid;
>+    if (rtcStats.state.is_defined())
>+        stats.state = fromStdString(*rtcStats.state);
>+    if (rtcStats.messages_sent.is_defined())
>+        stats.messagesSent = *rtcStats.messages_sent;
>+    if (rtcStats.bytes_sent.is_defined())
>+        stats.bytesSent = *rtcStats.bytes_sent;
>+    if (rtcStats.messages_received.is_defined())
>+        stats.messagesReceived = *rtcStats.messages_received;
>+    if (rtcStats.bytes_received.is_defined())
>+        stats.bytesReceived = *rtcStats.bytes_received;
>+}
>+
>+static inline RTCStatsReport::IceCandidatePairState iceCandidatePairState(const std::string& state)
>+{
>+    if (state == "frozen")
>+        return RTCStatsReport::IceCandidatePairState::Frozen;
>+    if (state == "waiting")
>+        return RTCStatsReport::IceCandidatePairState::Waiting;
>+    if (state == "in-progress")
>+        return RTCStatsReport::IceCandidatePairState::Inprogress;
>+    if (state == "failed")
>+        return RTCStatsReport::IceCandidatePairState::Failed;
>+    if (state == "succeeded")
>+        return RTCStatsReport::IceCandidatePairState::Succeeded;
>+    if (state == "cancelled")
>+        return RTCStatsReport::IceCandidatePairState::Cancelled;
>+    ASSERT_NOT_REACHED();
>+    return RTCStatsReport::IceCandidatePairState::Frozen;
>+}
>+
>+static inline void fillRTCIceCandidatePairStats(RTCStatsReport::IceCandidatePairStats& stats, const webrtc::RTCIceCandidatePairStats& rtcStats)
>+{
>+    fillRTCStats(stats, rtcStats);
>+
>+    if (rtcStats.transport_id.is_defined())
>+        stats.transportId = fromStdString(*rtcStats.transport_id);
>+    if (rtcStats.local_candidate_id.is_defined())
>+        stats.localCandidateId = fromStdString(*rtcStats.local_candidate_id);
>+    if (rtcStats.remote_candidate_id.is_defined())
>+        stats.remoteCandidateId = fromStdString(*rtcStats.remote_candidate_id);
>+    if (rtcStats.state.is_defined())
>+        stats.state = iceCandidatePairState(*rtcStats.state);
>+
>+    if (rtcStats.priority.is_defined())
>+        stats.priority = *rtcStats.priority;
>+    if (rtcStats.nominated.is_defined())
>+        stats.nominated = *rtcStats.nominated;
>+    if (rtcStats.writable.is_defined())
>+        stats.writable = *rtcStats.writable;
>+    if (rtcStats.readable.is_defined())
>+        stats.readable = *rtcStats.readable;
>+
>+    if (rtcStats.bytes_sent.is_defined())
>+        stats.bytesSent = *rtcStats.bytes_sent;
>+    if (rtcStats.bytes_received.is_defined())
>+        stats.bytesReceived = *rtcStats.bytes_received;
>+    if (rtcStats.total_round_trip_time.is_defined())
>+        stats.totalRoundTripTime = *rtcStats.total_round_trip_time;
>+    if (rtcStats.current_round_trip_time.is_defined())
>+        stats.currentRoundTripTime = *rtcStats.current_round_trip_time;
>+    if (rtcStats.available_outgoing_bitrate.is_defined())
>+        stats.availableOutgoingBitrate = *rtcStats.available_outgoing_bitrate;
>+    if (rtcStats.available_incoming_bitrate.is_defined())
>+        stats.availableIncomingBitrate = *rtcStats.available_incoming_bitrate;
>+
>+    if (rtcStats.requests_received.is_defined())
>+        stats.requestsReceived = *rtcStats.requests_received;
>+    if (rtcStats.requests_sent.is_defined())
>+        stats.requestsSent = *rtcStats.requests_sent;
>+    if (rtcStats.responses_received.is_defined())
>+        stats.responsesReceived = *rtcStats.responses_received;
>+    if (rtcStats.responses_sent.is_defined())
>+        stats.responsesSent = *rtcStats.responses_sent;
>+
>+    if (rtcStats.requests_received.is_defined())
>+        stats.retransmissionsReceived = *rtcStats.requests_received;
>+    if (rtcStats.requests_sent.is_defined())
>+        stats.retransmissionsSent = *rtcStats.requests_sent;
>+    if (rtcStats.responses_received.is_defined())
>+        stats.consentRequestsReceived = *rtcStats.responses_received;
>+    if (rtcStats.responses_sent.is_defined())
>+        stats.consentRequestsSent = *rtcStats.responses_sent;
>+    if (rtcStats.responses_received.is_defined())
>+        stats.consentResponsesReceived = *rtcStats.responses_received;
>+    if (rtcStats.responses_sent.is_defined())
>+        stats.consentResponsesSent = *rtcStats.responses_sent;
>+}
>+
>+static inline void fillRTCCertificateStats(RTCStatsReport::CertificateStats& stats, const webrtc::RTCCertificateStats& rtcStats)
>+{
>+    fillRTCStats(stats, rtcStats);
>+
>+    if (rtcStats.fingerprint.is_defined())
>+        stats.fingerprint = fromStdString(*rtcStats.fingerprint);
>+    if (rtcStats.fingerprint_algorithm.is_defined())
>+        stats.fingerprintAlgorithm = fromStdString(*rtcStats.fingerprint_algorithm);
>+    if (rtcStats.base64_certificate.is_defined())
>+        stats.base64Certificate = fromStdString(*rtcStats.base64_certificate);
>+    if (rtcStats.issuer_certificate_id.is_defined())
>+        stats.issuerCertificateId = fromStdString(*rtcStats.issuer_certificate_id);
>+}
>+void LibWebRTCStatsCollector::OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>& rtcReport)
>+{
>+    callOnMainThread([protectedThis = rtc::scoped_refptr<LibWebRTCStatsCollector>(this), rtcReport] {
>+        auto report = RTCStatsReport::create();
>+        if (!protectedThis->m_callback(report.copyRef()))
>+            return;
>+
>+        ASSERT(report->backingMap());
>+
>+        for (const auto& rtcStats : *rtcReport) {
>+            if (rtcStats.type() == webrtc::RTCInboundRTPStreamStats::kType) {
>+                RTCStatsReport::InboundRTPStreamStats stats;
>+                fillInboundRTPStreamStats(stats, static_cast<const webrtc::RTCInboundRTPStreamStats&>(rtcStats));
>+                report->addStats<IDLDictionary<RTCStatsReport::InboundRTPStreamStats>>(WTFMove(stats));
>+            } else if (rtcStats.type() == webrtc::RTCOutboundRTPStreamStats::kType) {
>+                RTCStatsReport::OutboundRTPStreamStats stats;
>+                fillOutboundRTPStreamStats(stats, static_cast<const webrtc::RTCOutboundRTPStreamStats&>(rtcStats));
>+                report->addStats<IDLDictionary<RTCStatsReport::OutboundRTPStreamStats>>(WTFMove(stats));
>+            } else if (rtcStats.type() == webrtc::RTCMediaStreamTrackStats::kType) {
>+                RTCStatsReport::MediaStreamTrackStats stats;
>+                fillRTCMediaStreamTrackStats(stats, static_cast<const webrtc::RTCMediaStreamTrackStats&>(rtcStats));
>+                report->addStats<IDLDictionary<RTCStatsReport::MediaStreamTrackStats>>(WTFMove(stats));
>+            } else if (rtcStats.type() == webrtc::RTCDataChannelStats::kType) {
>+                RTCStatsReport::DataChannelStats stats;
>+                fillRTCDataChannelStats(stats, static_cast<const webrtc::RTCDataChannelStats&>(rtcStats));
>+                report->addStats<IDLDictionary<RTCStatsReport::DataChannelStats>>(WTFMove(stats));
>+            } else if (rtcStats.type() == webrtc::RTCIceCandidatePairStats::kType) {
>+                RTCStatsReport::IceCandidatePairStats stats;
>+                fillRTCIceCandidatePairStats(stats, static_cast<const webrtc::RTCIceCandidatePairStats&>(rtcStats));
>+                report->addStats<IDLDictionary<RTCStatsReport::IceCandidatePairStats>>(WTFMove(stats));
>+            } else if (rtcStats.type() == webrtc::RTCCertificateStats::kType) {
>+                RTCStatsReport::CertificateStats stats;
>+                fillRTCCertificateStats(stats, static_cast<const webrtc::RTCCertificateStats&>(rtcStats));
>+                report->addStats<IDLDictionary<RTCStatsReport::CertificateStats>>(WTFMove(stats));
>+            }
>+        }
>+    });
>+}
>+
>+}; // namespace WTF
>+
>+
>+#endif // USE(LIBWEBRTC)
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..c64617f969500826e905b73fccf8b7529d56b5e3
>--- /dev/null
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCStatsCollector.h
>@@ -0,0 +1,60 @@
>+/*
>+ * Copyright (C) 2018 Apple Inc. All rights reserved.
>+ *
>+ * Redistribution and use in source and binary forms, with or without
>+ * modification, are permitted provided that the following conditions
>+ * are met:
>+ * 1.  Redistributions of source code must retain the above copyright
>+ *     notice, this list of conditions and the following disclaimer.
>+ * 2.  Redistributions in binary form must reproduce the above copyright
>+ *     notice, this list of conditions and the following disclaimer in the
>+ *     documentation and/or other materials provided with the distribution.
>+ *
>+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
>+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
>+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
>+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
>+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
>+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
>+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
>+ * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
>+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
>+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
>+ */
>+
>+#pragma once
>+
>+#if USE(LIBWEBRTC)
>+
>+#include "LibWebRTCMacros.h"
>+#include <wtf/CompletionHandler.h>
>+#include <wtf/RefPtr.h>
>+
>+#pragma GCC diagnostic push
>+#pragma GCC diagnostic ignored "-Wunused-parameter"
>+
>+#include <webrtc/pc/rtcstatscollector.h>
>+
>+#pragma GCC diagnostic pop
>+
>+namespace WebCore {
>+
>+class RTCStatsReport;
>+
>+class LibWebRTCStatsCollector : public webrtc::RTCStatsCollectorCallback {
>+public:
>+    using CollectorCallback = WTF::CompletionHandler<bool(RefPtr<RTCStatsReport>&&)>;
>+    static rtc::scoped_refptr<LibWebRTCStatsCollector> create(CollectorCallback&& callback) { return new rtc::RefCountedObject<LibWebRTCStatsCollector>(WTFMove(callback)); }
>+
>+    explicit LibWebRTCStatsCollector(CollectorCallback&&);
>+    ~LibWebRTCStatsCollector();
>+
>+private:
>+    void OnStatsDelivered(const rtc::scoped_refptr<const webrtc::RTCStatsReport>&) final;
>+
>+    CollectorCallback m_callback;
>+};
>+
>+} // namespace WebCore
>+
>+#endif // USE(LIBWEBRTC)
>diff --git a/Source/WebCore/WebCore.xcodeproj/project.pbxproj b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>index 1c81c31c9c5ea4b9b5a32a07b53c11ca681d5b63..85ccbd8a37ad0c982045e4080e00db77d8b9f17e 100644
>--- a/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>+++ b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>@@ -1126,6 +1126,7 @@
> 		41D129D31F3D0F1600D15E47 /* CacheStorageConnection.h in Headers */ = {isa = PBXBuildFile; fileRef = 41D129CC1F3D0EE300D15E47 /* CacheStorageConnection.h */; settings = {ATTRIBUTES = (Private, ); }; };
> 		41D129D51F3D0F6900D15E47 /* CacheStorageProvider.h in Headers */ = {isa = PBXBuildFile; fileRef = 41D129D41F3D0F6600D15E47 /* CacheStorageProvider.h */; settings = {ATTRIBUTES = (Private, ); }; };
> 		41D129DB1F3D143800D15E47 /* FetchHeaders.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F54F831C50C4F600338488 /* FetchHeaders.h */; settings = {ATTRIBUTES = (Private, ); }; };
>+		41D28D0D2139E05800F4206F /* LibWebRTCStatsCollector.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 41D28D0B2139E01D00F4206F /* LibWebRTCStatsCollector.cpp */; };
> 		41DEFCB61E56C1BD000D9E5F /* JSDOMMapLike.h in Headers */ = {isa = PBXBuildFile; fileRef = 41DEFCB41E56C1B9000D9E5F /* JSDOMMapLike.h */; };
> 		41E1B1D10FF5986900576B3B /* AbstractWorker.h in Headers */ = {isa = PBXBuildFile; fileRef = 41E1B1CB0FF5986900576B3B /* AbstractWorker.h */; };
> 		41F062140F5F192600A07EAC /* InspectorDatabaseResource.h in Headers */ = {isa = PBXBuildFile; fileRef = 41F062120F5F192600A07EAC /* InspectorDatabaseResource.h */; };
>@@ -7335,6 +7336,8 @@
> 		41D129CA1F3D0EE300D15E47 /* CacheStorageRecord.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CacheStorageRecord.h; sourceTree = "<group>"; };
> 		41D129CC1F3D0EE300D15E47 /* CacheStorageConnection.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CacheStorageConnection.h; sourceTree = "<group>"; };
> 		41D129D41F3D0F6600D15E47 /* CacheStorageProvider.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CacheStorageProvider.h; sourceTree = "<group>"; };
>+		41D28D0B2139E01D00F4206F /* LibWebRTCStatsCollector.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = LibWebRTCStatsCollector.cpp; path = libwebrtc/LibWebRTCStatsCollector.cpp; sourceTree = "<group>"; };
>+		41D28D0C2139E01E00F4206F /* LibWebRTCStatsCollector.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = LibWebRTCStatsCollector.h; path = libwebrtc/LibWebRTCStatsCollector.h; sourceTree = "<group>"; };
> 		41D51BB21E4E2E8100131A5B /* LibWebRTCAudioFormat.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = LibWebRTCAudioFormat.h; path = libwebrtc/LibWebRTCAudioFormat.h; sourceTree = "<group>"; };
> 		41DEFCB21E56C1B9000D9E5F /* JSDOMBindingInternals.js */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.javascript; path = JSDOMBindingInternals.js; sourceTree = "<group>"; };
> 		41DEFCB31E56C1B9000D9E5F /* JSDOMMapLike.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSDOMMapLike.cpp; sourceTree = "<group>"; };
>@@ -16828,6 +16831,8 @@
> 				415A3B732138E264001B4BAA /* LibWebRTCObservers.h */,
> 				417612AD1E3A993B00C3D81D /* LibWebRTCPeerConnectionBackend.cpp */,
> 				417612AE1E3A993B00C3D81D /* LibWebRTCPeerConnectionBackend.h */,
>+				41D28D0B2139E01D00F4206F /* LibWebRTCStatsCollector.cpp */,
>+				41D28D0C2139E01E00F4206F /* LibWebRTCStatsCollector.h */,
> 			);
> 			name = libwebrtc;
> 			sourceTree = "<group>";
>@@ -31776,6 +31781,7 @@
> 				DECA7E651F9EBD8300E3B661 /* UnifiedSource234.cpp in Sources */,
> 				DECA7E661F9EBD8300E3B661 /* UnifiedSource235.cpp in Sources */,
> 				DECA7E671F9EBD8300E3B661 /* UnifiedSource236.cpp in Sources */,
>+				41D28D0D2139E05800F4206F /* LibWebRTCStatsCollector.cpp in Sources */,
> 				DECA7E681F9EBD8300E3B661 /* UnifiedSource237.cpp in Sources */,
> 				DECA7E691F9EBD8300E3B661 /* UnifiedSource238.cpp in Sources */,
> 				DECA7E6A1F9EBD8300E3B661 /* UnifiedSource239.cpp in Sources */,

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Attachments on bug 189180: 348598 | 348602 | 348670