Attachment 340841 Details for Bug 185761 – Patch

[patch] Patch

bug-185761-20180521121034.patch (text/plain), 116.83 KB, created by Alejandro G. Castro on 2018-05-21 03:10:36 PDT

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Creator: Alejandro G. Castro

Created: 2018-05-21 03:10:36 PDT

Size: 116.83 KB

patch

obsolete

>Subversion Revision: 231947
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 385dd61ab755a5d049b333bcb2ce9f0b359a9764..4ccb4f43c75636c95f9ebde9ef6db30a7e9b462f 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,85 @@
>+2018-05-18  Alejandro G. Castro  <alex@igalia.com> and Thibault Saunier  <tsaunier@igalia.com>
>+
>+        [GTK][WPE] Add mediaDevices.enumerateDevices support
>+        https://bugs.webkit.org/show_bug.cgi?id=185761
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        We are adding all the required classes to make the
>+        enumerateDevices work, that means our own
>+        RealtimeMediaSourceCenterLibWebRTC for the platform, the
>+        GStreamerCaptureDeviceManager, the audio/video capturers and their
>+        respective audio/video sources. We are using the GstDeviceMonitor
>+        monitor to request the information from the system.
>+
>+        Unskipped the relevant tests for the enumerateDevices.
>+
>+        * Modules/mediastream/MediaDevicesRequest.cpp:
>+        (WebCore::MediaDevicesRequest::filterDeviceList): Add support for
>+        GTK and WPE platforms when filtering devices if there is no
>+        persistent access to the origin.
>+        * platform/GStreamer.cmake: Added the new files to the compilation.
>+        * platform/audio/AudioStreamDescription.h: Added new GStreamer type.
>+        * platform/audio/PlatformAudioData.h: Added new GStreamer type for
>+        the GStreamerAudioData class.
>+        * platform/graphics/gstreamer/GStreamerCommon.cpp:
>+        (WebCore::simpleBusMessageCallback): This function and the next
>+        one help us connect a bus message callback to a pipeline for
>+        debugging.
>+        (WebCore::connectSimpleBusMessageCallback): Ditto.
>+        * platform/graphics/gstreamer/GStreamerCommon.h: Ditto.
>+        * platform/mediastream/RealtimeMediaSourceCenter.cpp:
>+        (WebCore::RealtimeMediaSourceCenter::singleton): Remove the code
>+        used for compilation for the platform when we do not have a
>+        RealtimeMediaSourceCenterLibWebRTC. Now we return the proper class
>+        for the platform.
>+        * platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp:
>+        Added class representing the RealtimeMediaSource for the Audio
>+        with GStreamer.
>+        * platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h:
>+        Ditto.
>+        * platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp: Added
>+        this class that represents the GStreamer pipeline that captures
>+        audio from the system devices, it inherits from GStreamerCapturer.
>+        * platform/mediastream/gstreamer/GStreamerAudioCapturer.h: Dito.
>+        * platform/mediastream/gstreamer/GStreamerAudioData.h: Added this
>+        class implementing PlatformAudioData for the GStreamer platform,
>+        used to pass the samples information.
>+        * platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h:
>+        Added this class implementing AudioStreamDescription to export the
>+        information about the audio stream to libwebrtc.
>+        * platform/mediastream/gstreamer/GStreamerCaptureDevice.h: Added
>+        this base class for the audio and video capturing devices, it
>+        implements general WebKit CaptureDevice class.
>+        * platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.cpp:
>+        Added this class that implements the system monitor to get the
>+        list of available devices in the system. It uses GstDeviceMonitor
>+        to handle the operation. It uses two singleton device managers one
>+        for audio and another one for video, the design required by the
>+        RealtimeMediaSourceCenter.
>+        * platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.h: Ditto.
>+        * platform/mediastream/gstreamer/GStreamerCapturer.cpp: Added this
>+        base class representing how GStreamer captures the media from the
>+        input devices in the system. Two classes inherit from this one to
>+        capture audio and video. It setups the GStreamer pipeline and adds
>+        functions to control it.
>+        * platform/mediastream/gstreamer/GStreamerCapturer.h: Ditto.
>+        * platform/mediastream/gstreamer/GStreamerVideoCaptureSource.cpp:
>+        Added this RealtimeMediaSource representing the source of the
>+        video data for the GStreamer platform. It handles the settings and
>+        capabilities of the source and creates the capturer used to
>+        control the operation of the stream.
>+        * platform/mediastream/gstreamer/GStreamerVideoCaptureSource.h: Ditto.
>+        * platform/mediastream/gstreamer/GStreamerVideoCapturer.cpp: Added
>+        this class that inherits from the GStreamerCapturer and controls
>+        the GStreamer pipelines of the video streams of the system.
>+        * platform/mediastream/gstreamer/GStreamerVideoCapturer.h: Ditto.
>+        * platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.cpp:
>+        Added this class that implements the key RealtimeMediaSourceCenter
>+        functions to configure the base class for the platform, using the
>+        other GStreamer classes.
>+        * platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.h: Ditto
>+
> 2018-05-17  David Kilzer  <ddkilzer@apple.com>
> 
>         Lazily create WebCore::Timer for WebCore::Image
>diff --git a/Source/WebKit/ChangeLog b/Source/WebKit/ChangeLog
>index 15471e939d0e099108ee92aeabb105e22b74ce54..3f61fb65bf7c56cfc200fc1697f4ff66544cd42e 100644
>--- a/Source/WebKit/ChangeLog
>+++ b/Source/WebKit/ChangeLog
>@@ -1,3 +1,73 @@
>+2018-05-18  Alejandro G. Castro  <alex@igalia.com> and Thibault Saunier  <tsaunier@igalia.com>
>+
>+        [GTK][WPE] Add mediaDevices.enumerateDevices support
>+        https://bugs.webkit.org/show_bug.cgi?id=185761
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        In order to implement the checkUserMediaPermissionForOrigin
>+        function that is called when the webpage wants to enumerate the
>+        list of devices in the system we added some new APIs for the GTK
>+        and WPE ports. We have extended the
>+        WebKitUserMediaPermissionRequest class to support a new kind of
>+        request, the checkRequest. With this addition WebKit can
>+        create a WebKitUserMediaPermissionRequest with a permission check
>+        proxy, if that is the case the WebKitWebView widget launches
>+        WebKitWebView::permission-check signal and the browser can
>+        connnect to it to resolve it with the new
>+        webkit_permission_request_resolve_check API, adding the hash salt
>+        and the decision about allowing access to the devices. Main point
>+        of this API is to make sure that the user can control the webpages
>+        and that they don't get smart trying to fingerprint the system. There is a
>+        default implementation in the widget that basically always
>+        denies access and generates a random hashsalt.
>+
>+        * UIProcess/API/glib/WebKitPermissionRequest.cpp:
>+        (webkit_permission_request_resolve_check): Added the API to
>+        resolve the permission-check requested by the engine.
>+        * UIProcess/API/glib/WebKitUIClient.cpp:
>+        (checkUserMediaPermissionForOrigin): Added the implementation for
>+        the platform used when the webpage calls the enumerateDevices API.
>+        * UIProcess/API/glib/WebKitUserMediaPermissionRequest.cpp:
>+        (webkitUserMediaPermissionResolveCheck): Added the function that
>+        resolves the permission check passing the hash salt and the
>+        boolean specifying if the operation is allowed to the completion
>+        handler in the request.
>+        (webkit_permission_request_interface_init): Added the new class
>+        function.
>+        (webkitUserMediaPermissionRequestDispose): Make sure we dispose
>+        correctly now that we have a new checkRequest, in that situation
>+        request is null.
>+        (webkit_user_media_permission_is_for_audio_device): Control the
>+        browser does not call this API when we are doing a permission check.
>+        (webkit_user_media_permission_is_for_video_device): Ditto.
>+        (webkitUserMediaPermissionCheckCreate): Added new builder for the
>+        WebKitPermissionRequest class using a UserMediaPermissionCheckProxy.
>+        * UIProcess/API/glib/WebKitUserMediaPermissionRequestPrivate.h:
>+        Added the new API for the engine to create a WebKitPermissionRequest.
>+        (webkitUserMediaPermissionCheckCreate):
>+        * UIProcess/API/glib/WebKitWebView.cpp:
>+        (webkitWebViewPermissionCheck): Added default implementation for
>+        the signal callback.
>+        (webkit_web_view_class_init): Set the default implementation for
>+        the permission-check signal.
>+        (webkitWebViewMakePermissionCheck): Added new API to launch ask
>+        for permission check, this is private and used by the engine.
>+        * UIProcess/API/glib/WebKitWebViewPrivate.h: Define the new
>+        private API webkitWebViewMakePermissionCheck.
>+        * UIProcess/API/gtk/WebKitPermissionRequest.h: Added the new API
>+        to resolve the permission check in the engine. Two minor style
>+        modifications.
>+        * UIProcess/API/gtk/WebKitWebView.h: Added the new class function
>+        for the permission-check signal.
>+        * UIProcess/API/gtk/docs/webkit2gtk-4.0-sections.txt: Added the
>+        new symbol to the documentation:
>+        webkit_permission_request_resolve_check.
>+        * UIProcess/API/wpe/WebKitPermissionRequest.h: Added the new API
>+        to resolve the permission check in the engine.
>+        * UIProcess/API/wpe/WebKitWebView.h: Added the new class function
>+        for the permission-check signal.
>+
> 2018-05-18  Fujii Hironori  <Hironori.Fujii@sony.com>
> 
>         [Curl] Remove unused SystemProxyWin.cpp
>diff --git a/Source/WebCore/Modules/mediastream/MediaDevicesRequest.cpp b/Source/WebCore/Modules/mediastream/MediaDevicesRequest.cpp
>index 12400dcf2deed8fbc25e1bfa42398822326f4183..4bb0626c205f44566e69312285b21da25b8d99dd 100644
>--- a/Source/WebCore/Modules/mediastream/MediaDevicesRequest.cpp
>+++ b/Source/WebCore/Modules/mediastream/MediaDevicesRequest.cpp
>@@ -81,9 +81,7 @@ void MediaDevicesRequest::contextDestroyed()
> 
> void MediaDevicesRequest::filterDeviceList(Vector<Ref<MediaDeviceInfo>>& devices)
> {
>-#if !PLATFORM(COCOA)
>-    UNUSED_PARAM(devices);
>-#else
>+#if PLATFORM(COCOA) || PLATFORM(GTK) || PLATFORM(WPE)
> 
> #if PLATFORM(IOS)
>     static const int defaultCameraCount = 2;
>@@ -91,6 +89,10 @@ void MediaDevicesRequest::filterDeviceList(Vector<Ref<MediaDeviceInfo>>& devices
> #if PLATFORM(MAC)
>     static const int defaultCameraCount = 1;
> #endif
>+#if PLATFORM(GTK) || PLATFORM(WPE)
>+    static const int defaultCameraCount = 1;
>+#endif
>+
>     static const int defaultMicrophoneCount = 1;
> 
>     int cameraCount = 0;
>@@ -103,7 +105,8 @@ void MediaDevicesRequest::filterDeviceList(Vector<Ref<MediaDeviceInfo>>& devices
> 
>         return false;
>     });
>-
>+#else
>+    UNUSED_PARAM(devices);
> #endif
> }
> 
>diff --git a/Source/WebCore/platform/GStreamer.cmake b/Source/WebCore/platform/GStreamer.cmake
>index 3d5a6e0025a52bb7b7a6f71d34719f2bdc2c8e5d..d087be7542a9974d5eb84917c1db0b41ee59f0fd 100644
>--- a/Source/WebCore/platform/GStreamer.cmake
>+++ b/Source/WebCore/platform/GStreamer.cmake
>@@ -33,6 +33,19 @@ if (ENABLE_VIDEO OR ENABLE_WEB_AUDIO)
>         platform/graphics/gstreamer/mse/PlaybackPipeline.cpp
>         platform/graphics/gstreamer/mse/SourceBufferPrivateGStreamer.cpp
>         platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp
>+
>+        platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp
>+        platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp
>+        platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.cpp
>+        platform/mediastream/gstreamer/GStreamerCapturer.cpp
>+        platform/mediastream/gstreamer/GStreamerVideoCaptureSource.cpp
>+        platform/mediastream/gstreamer/GStreamerVideoCapturer.cpp
>+        platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.cpp
>+
>+        platform/mock/MockRealtimeAudioSource.cpp
>+        platform/mock/MockRealtimeMediaSource.cpp
>+        platform/mock/MockRealtimeMediaSourceCenter.cpp
>+        platform/mock/MockRealtimeVideoSource.cpp
>     )
> 
>     list(APPEND WebCore_SYSTEM_INCLUDE_DIRECTORIES
>diff --git a/Source/WebCore/platform/audio/AudioStreamDescription.h b/Source/WebCore/platform/audio/AudioStreamDescription.h
>index e440dc90a3e84dfd3b9ed5876046dba71c97fe3b..2ba585c9c0567111d75709d897cd7cb980363240 100644
>--- a/Source/WebCore/platform/audio/AudioStreamDescription.h
>+++ b/Source/WebCore/platform/audio/AudioStreamDescription.h
>@@ -35,6 +35,7 @@ struct PlatformDescription {
>     enum {
>         None,
>         CAAudioStreamBasicType,
>+        GStreamerAudioStreamDescription,
>     } type;
>     Variant<std::nullptr_t, const AudioStreamBasicDescription*> description;
> };
>diff --git a/Source/WebCore/platform/audio/PlatformAudioData.h b/Source/WebCore/platform/audio/PlatformAudioData.h
>index e4544f63254a87876d99d8c3061203aa81679d98..cda72e8ab3c4cdc0a868faac4f05ce61c44402d0 100644
>--- a/Source/WebCore/platform/audio/PlatformAudioData.h
>+++ b/Source/WebCore/platform/audio/PlatformAudioData.h
>@@ -34,6 +34,7 @@ public:
>     enum class Kind {
>         None,
>         WebAudioBufferList,
>+        GStreamerAudioData,
>     };
> 
>     virtual Kind kind() const { return Kind::None; }
>diff --git a/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp b/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
>index e514fe3428de52898d4a07f6bc68a0c1d96ede36..435057a9efbaf2ae26888e6bbbe69685e9fc846c 100644
>--- a/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
>+++ b/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
>@@ -30,6 +30,7 @@
> #include <mutex>
> #include <wtf/glib/GLibUtilities.h>
> #include <wtf/glib/GUniquePtr.h>
>+#include <wtf/glib/RunLoopSourcePriority.h>
> 
> #if ENABLE(VIDEO_TRACK) && USE(GSTREAMER_MPEGTS)
> #define GST_USE_UNSTABLE_API
>@@ -320,6 +321,46 @@ bool gstRegistryHasElementForMediaType(GList* elementFactories, const char* caps
>     return result;
> }
> 
>+static void simpleBusMessageCallback(GstBus*, GstMessage* message, GstBin* pipeline)
>+{
>+    switch (GST_MESSAGE_TYPE(message)) {
>+    case GST_MESSAGE_ERROR:
>+        GST_ERROR_OBJECT(pipeline, "Got message: %" GST_PTR_FORMAT, message);
>+        {
>+            WTF::String dotFileName = String::format("%s_error", GST_OBJECT_NAME(pipeline));
>+            GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(pipeline, GST_DEBUG_GRAPH_SHOW_ALL, dotFileName.utf8().data());
>+        }
>+        break;
>+    case GST_MESSAGE_STATE_CHANGED:
>+        if (GST_MESSAGE_SRC(message) == GST_OBJECT(pipeline)) {
>+            GstState oldState, newState, pending;
>+            gst_message_parse_state_changed(message, &oldState, &newState, &pending);
>+
>+            GST_INFO_OBJECT(pipeline, "State changed (old: %s, new: %s, pending: %s)",
>+                gst_element_state_get_name(oldState),
>+                gst_element_state_get_name(newState),
>+                gst_element_state_get_name(pending));
>+
>+            WTF::String dotFileName = String::format("%s_%s_%s",
>+                GST_OBJECT_NAME(pipeline),
>+                gst_element_state_get_name(oldState),
>+                gst_element_state_get_name(newState));
>+
>+            GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(pipeline), GST_DEBUG_GRAPH_SHOW_ALL, dotFileName.utf8().data());
>+        }
>+        break;
>+    default:
>+        break;
>+    }
>+}
>+
>+void connectSimpleBusMessageCallback(GstElement *pipeline)
>+{
>+    GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(pipeline)));
>+    gst_bus_add_signal_watch_full(bus.get(), RunLoopSourcePriority::RunLoopDispatcher);
>+    g_signal_connect(bus.get(), "message", G_CALLBACK(simpleBusMessageCallback), pipeline);
>+}
>+
> }
> 
> #endif // USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h b/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h
>index 259255e741e415a314a123d34d0188ad4690e64d..3b258586fe4a1add888c0bb1ce45edc1970abdfc 100644
>--- a/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h
>+++ b/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h
>@@ -82,6 +82,8 @@ inline GstClockTime toGstClockTime(const MediaTime &mediaTime)
> }
> 
> bool gstRegistryHasElementForMediaType(GList* elementFactories, const char* capsString);
>+void connectSimpleBusMessageCallback(GstElement *pipeline);
>+
> }
> 
> #ifndef GST_BUFFER_DTS_OR_PTS
>diff --git a/Source/WebCore/platform/mediastream/RealtimeMediaSourceCenter.cpp b/Source/WebCore/platform/mediastream/RealtimeMediaSourceCenter.cpp
>index 2efb82d1e58c71d9da98ebca402db81d254bef45..3886ff875ce89f5c108c437bb9caa2f4d7f1602a 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeMediaSourceCenter.cpp
>+++ b/Source/WebCore/platform/mediastream/RealtimeMediaSourceCenter.cpp
>@@ -64,14 +64,7 @@ RealtimeMediaSourceCenter& RealtimeMediaSourceCenter::singleton()
>     RealtimeMediaSourceCenter* override = mediaStreamCenterOverride();
>     if (override)
>         return *override;
>-#if PLATFORM(GTK)
>-    ASSERT(isMainThread());
>-    notImplemented(); // Return MockRealtimeMediaSourceCenter to avoid crash.
>-    static NeverDestroyed<MockRealtimeMediaSourceCenter> center;
>-    return center;
>-#else
>     return RealtimeMediaSourceCenter::platformCenter();
>-#endif
> }
> 
> void RealtimeMediaSourceCenter::setSharedStreamCenterOverride(RealtimeMediaSourceCenter* center)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..97aa87a26093c052df7a6128f021babed8ac191d
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp
>@@ -0,0 +1,205 @@
>+/*
>+ * Copyright (C) 2017 Apple Inc. All rights reserved.
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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"
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "GStreamerAudioCaptureSource.h"
>+
>+#include "GStreamerAudioData.h"
>+#include "GStreamerAudioStreamDescription.h"
>+#include "GStreamerCaptureDeviceManager.h"
>+
>+#include <gst/app/gstappsink.h>
>+#include <gst/gst.h>
>+#include <wtf/NeverDestroyed.h>
>+
>+namespace WebCore {
>+
>+const static CapabilityValueOrRange defaultVolumeCapability = CapabilityValueOrRange(0.0, 1.0);
>+const static RealtimeMediaSourceCapabilities::EchoCancellation defaultEchoCancellationCapability = RealtimeMediaSourceCapabilities::EchoCancellation::ReadWrite;
>+
>+GST_DEBUG_CATEGORY(webkit_audio_capture_source_debug);
>+#define GST_CAT_DEFAULT webkit_audio_capture_source_debug
>+
>+static void initializeGStreamerDebug()
>+{
>+    static std::once_flag debugRegisteredFlag;
>+    std::call_once(debugRegisteredFlag, [] {
>+        GST_DEBUG_CATEGORY_INIT(webkit_audio_capture_source_debug, "webkitaudiocapturesource", 0, "WebKit Audio Capture Source.");
>+    });
>+}
>+
>+class GStreamerAudioCaptureSourceFactory : public RealtimeMediaSource::AudioCaptureFactory {
>+public:
>+    CaptureSourceOrError createAudioCaptureSource(const CaptureDevice& device, const MediaConstraints* constraints) final
>+    {
>+        return GStreamerAudioCaptureSource::create(device.persistentId(), constraints);
>+    }
>+};
>+
>+static GStreamerAudioCaptureSourceFactory& libWebRTCAudioCaptureSourceFactory()
>+{
>+    static NeverDestroyed<GStreamerAudioCaptureSourceFactory> factory;
>+    return factory.get();
>+}
>+
>+CaptureSourceOrError GStreamerAudioCaptureSource::create(const String& deviceID, const MediaConstraints* constraints)
>+{
>+    auto device = GStreamerAudioCaptureDeviceManager::singleton().gstreamerDeviceWithUID(deviceID);
>+    if (!device) {
>+        auto errorMessage = String::format("GStreamerAudioCaptureSource::create(): GStreamer did not find the device: %s.", deviceID.utf8().data());
>+        return CaptureSourceOrError(WTFMove(errorMessage));
>+    }
>+
>+    auto source = adoptRef(*new GStreamerAudioCaptureSource(device.value()));
>+
>+    if (constraints) {
>+        auto result = source->applyConstraints(*constraints);
>+        if (result)
>+            return WTFMove(result.value().first);
>+    }
>+    return CaptureSourceOrError(WTFMove(source));
>+}
>+
>+RealtimeMediaSource::AudioCaptureFactory& GStreamerAudioCaptureSource::factory()
>+{
>+    return libWebRTCAudioCaptureSourceFactory();
>+}
>+
>+GStreamerAudioCaptureSource::GStreamerAudioCaptureSource(GStreamerCaptureDevice device)
>+    : RealtimeMediaSource(device.persistentId(), RealtimeMediaSource::Type::Audio, device.label())
>+    , m_capturer(std::make_unique<GStreamerAudioCapturer>(device))
>+{
>+    initializeGStreamerDebug();
>+}
>+
>+GStreamerAudioCaptureSource::GStreamerAudioCaptureSource(const String& deviceID, const String& name)
>+    : RealtimeMediaSource(deviceID, RealtimeMediaSource::Type::Audio, name)
>+    , m_capturer(std::make_unique<GStreamerAudioCapturer>())
>+{
>+    initializeGStreamerDebug();
>+}
>+
>+GStreamerAudioCaptureSource::~GStreamerAudioCaptureSource()
>+{
>+}
>+
>+void GStreamerAudioCaptureSource::startProducingData()
>+{
>+    m_capturer->setupPipeline();
>+    m_capturer->setSampleRate(sampleRate());
>+    g_signal_connect(m_capturer->sink(), "new-sample", G_CALLBACK(newSampleCallback), this);
>+    m_capturer->play();
>+}
>+
>+GstFlowReturn GStreamerAudioCaptureSource::newSampleCallback(GstElement* sink, GStreamerAudioCaptureSource* source)
>+{
>+    auto sample = adoptGRef(gst_app_sink_pull_sample(GST_APP_SINK(sink)));
>+
>+    // FIXME - figure out a way to avoid copying (on write) the data.
>+    GstBuffer* buf = gst_sample_get_buffer(sample.get());
>+    auto frames(std::unique_ptr<GStreamerAudioData>(new GStreamerAudioData(WTFMove(sample))));
>+    auto streamDesc(std::unique_ptr<GStreamerAudioStreamDescription>(new GStreamerAudioStreamDescription(frames->getAudioInfo())));
>+
>+    source->audioSamplesAvailable(
>+        MediaTime(GST_TIME_AS_USECONDS(GST_BUFFER_PTS(buf)), G_USEC_PER_SEC),
>+        *frames, *streamDesc, gst_buffer_get_size(buf) / frames->getAudioInfo().bpf);
>+
>+    return GST_FLOW_OK;
>+}
>+
>+void GStreamerAudioCaptureSource::stopProducingData()
>+{
>+    m_capturer->stop();
>+}
>+
>+const RealtimeMediaSourceCapabilities& GStreamerAudioCaptureSource::capabilities() const
>+{
>+    if (!m_capabilities) {
>+        GRefPtr<GstCaps> caps = m_capturer->caps();
>+        int minSampleRate = 0, maxSampleRate = 0;
>+        uint i;
>+
>+        for (i = 0; i < gst_caps_get_size(caps.get()); i++) {
>+            int capabilityMinSampleRate = 0, capabilityMaxSampleRate = 0;
>+            GstStructure* str = gst_caps_get_structure(caps.get(), i);
>+
>+            // Only accept raw audio for now.
>+            if (!gst_structure_has_name(str, "audio/x-raw"))
>+                continue;
>+
>+            gst_structure_get(str, "rate", GST_TYPE_INT_RANGE, &capabilityMinSampleRate, &capabilityMaxSampleRate, nullptr);
>+            if (i > 0) {
>+                minSampleRate = std::min(minSampleRate, capabilityMinSampleRate);
>+                maxSampleRate = std::max(maxSampleRate, capabilityMaxSampleRate);
>+            } else {
>+                minSampleRate = capabilityMinSampleRate;
>+                maxSampleRate = capabilityMaxSampleRate;
>+            }
>+        }
>+
>+        RealtimeMediaSourceCapabilities capabilities(settings().supportedConstraints());
>+        capabilities.setDeviceId(id());
>+        capabilities.setEchoCancellation(defaultEchoCancellationCapability);
>+        capabilities.setVolume(defaultVolumeCapability);
>+        capabilities.setSampleRate(CapabilityValueOrRange(minSampleRate, maxSampleRate));
>+        m_capabilities = WTFMove(capabilities);
>+    }
>+    return m_capabilities.value();
>+}
>+
>+bool GStreamerAudioCaptureSource::applySampleRate(int sampleRate)
>+{
>+    return m_capturer->setSampleRate(sampleRate);
>+}
>+
>+const RealtimeMediaSourceSettings& GStreamerAudioCaptureSource::settings() const
>+{
>+    if (!m_currentSettings) {
>+        RealtimeMediaSourceSettings settings;
>+        settings.setDeviceId(id());
>+
>+        RealtimeMediaSourceSupportedConstraints supportedConstraints;
>+        supportedConstraints.setSupportsDeviceId(true);
>+        supportedConstraints.setSupportsEchoCancellation(true);
>+        supportedConstraints.setSupportsVolume(true);
>+        supportedConstraints.setSupportsSampleRate(true);
>+        settings.setSupportedConstraints(supportedConstraints);
>+
>+        m_currentSettings = WTFMove(settings);
>+    }
>+
>+    m_currentSettings->setVolume(volume());
>+    m_currentSettings->setSampleRate(sampleRate());
>+    m_currentSettings->setEchoCancellation(echoCancellation());
>+
>+    return m_currentSettings.value();
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..af929663abc791cd82d60bf16bc94f62e614d401
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h
>@@ -0,0 +1,66 @@
>+/*
>+ * Copyright (C) 2017 Apple Inc. All rights reserved.
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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 ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "GStreamerAudioCapturer.h"
>+#include "GStreamerCaptureDevice.h"
>+#include "RealtimeMediaSource.h"
>+
>+namespace WebCore {
>+
>+class GStreamerAudioCaptureSource : public RealtimeMediaSource {
>+public:
>+    static CaptureSourceOrError create(const String& deviceID, const MediaConstraints*);
>+    WEBCORE_EXPORT static AudioCaptureFactory& factory();
>+
>+    const RealtimeMediaSourceCapabilities& capabilities() const final;
>+    const RealtimeMediaSourceSettings& settings() const final;
>+
>+private:
>+    GStreamerAudioCaptureSource(GStreamerCaptureDevice);
>+    GStreamerAudioCaptureSource(const String& deviceID, const String& name);
>+    virtual ~GStreamerAudioCaptureSource();
>+
>+    bool applySampleRate(int) final;
>+    bool isCaptureSource() const final { return true; }
>+    void startProducingData() final;
>+    void stopProducingData() final;
>+    bool applyVolume(double) final { return true; }
>+
>+    std::unique_ptr<GStreamerAudioCapturer> m_capturer;
>+
>+    static GstFlowReturn newSampleCallback(GstElement*, GStreamerAudioCaptureSource*);
>+    void triggerSampleAvailable(GstSample*);
>+
>+    mutable std::optional<RealtimeMediaSourceCapabilities> m_capabilities;
>+    mutable std::optional<RealtimeMediaSourceSettings> m_currentSettings;
>+};
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..0f860736ec61dc7634306a1d6f7f26ef0a249f33
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp
>@@ -0,0 +1,74 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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"
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "GStreamerAudioCapturer.h"
>+
>+#include "LibWebRTCAudioFormat.h"
>+
>+#include <gst/app/gstappsink.h>
>+
>+namespace WebCore {
>+
>+GStreamerAudioCapturer::GStreamerAudioCapturer(GStreamerCaptureDevice device)
>+    : GStreamerCapturer(device, adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
>+{
>+}
>+
>+GStreamerAudioCapturer::GStreamerAudioCapturer()
>+    : GStreamerCapturer("audiotestsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
>+{
>+}
>+
>+GstElement* GStreamerAudioCapturer::createConverter()
>+{
>+    // FIXME Handle errors.
>+    return gst_parse_bin_from_description("audioconvert ! audioresample", TRUE, nullptr);
>+}
>+
>+bool GStreamerAudioCapturer::setSampleRate(int sampleRate)
>+{
>+
>+    if (sampleRate > 0) {
>+        GST_INFO_OBJECT(m_pipeline.get(), "Setting SampleRate %d", sampleRate);
>+        m_caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, sampleRate, nullptr));
>+    } else {
>+        GST_INFO_OBJECT(m_pipeline.get(), "Not forcing sample rate");
>+        m_caps = adoptGRef(gst_caps_new_empty_simple("audio/x-raw"));
>+    }
>+
>+    if (!m_capsfilter.get())
>+        return false;
>+
>+    g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
>+
>+    return true;
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..c2de444a883f4b6063cc020175e1bf9e6ddd94ae
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.h
>@@ -0,0 +1,47 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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 ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+
>+#include "GStreamerCapturer.h"
>+
>+namespace WebCore {
>+
>+class GStreamerAudioCapturer : public GStreamerCapturer {
>+public:
>+    GStreamerAudioCapturer(GStreamerCaptureDevice);
>+    GStreamerAudioCapturer();
>+
>+    GstElement* createConverter() final;
>+    const char* name() final { return "Audio"; }
>+
>+    bool setSampleRate(int);
>+};
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..706d448aa6347dab04122fd4e21f3fdffc30707f
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioData.h
>@@ -0,0 +1,57 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L
>+ *
>+ * This library is free software; you can redistribute it and/or
>+ * modify it under the terms of the GNU Library General Public
>+ * License as published by the Free Software Foundation; either
>+ * version 2 of the License, or (at your option) any later version.
>+ *
>+ * This library is distributed in the hope that it will be useful,
>+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
>+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
>+ * Library General Public License for more details.
>+ *
>+ * You should have received a copy of the GNU Library General Public License
>+ * aint with this library; see the file COPYING.LIB.  If not, write to
>+ * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
>+ * Boston, MA 02110-1301, USA.
>+ */
>+
>+#pragma once
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+
>+#include "GRefPtrGStreamer.h"
>+#include "PlatformAudioData.h"
>+
>+#include <gst/audio/audio.h>
>+
>+namespace WebCore {
>+class GStreamerAudioData : public PlatformAudioData {
>+public:
>+    GStreamerAudioData(GRefPtr<GstSample>&& sample, GstAudioInfo info)
>+        : m_sample(WTFMove(sample))
>+        , m_audioInfo(info)
>+    {
>+    }
>+
>+    GStreamerAudioData(GRefPtr<GstSample>&& sample)
>+        : m_sample(WTFMove(sample))
>+    {
>+        gst_audio_info_from_caps(&m_audioInfo, gst_sample_get_caps(m_sample.get()));
>+    }
>+
>+    GstSample* getSample() { return m_sample.get(); }
>+    GstAudioInfo getAudioInfo() {return m_audioInfo; }
>+
>+private:
>+    Kind kind() const { return Kind::GStreamerAudioData; }
>+    GRefPtr<GstSample> m_sample;
>+    GRefPtr<GstCaps> m_caps;
>+
>+    GstAudioInfo m_audioInfo;
>+};
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..1aff44923454c18c9d76ccfb5660ff7ad53e076d
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h
>@@ -0,0 +1,109 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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 ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+
>+#include "AudioStreamDescription.h"
>+#include <gst/audio/audio.h>
>+
>+namespace WebCore {
>+
>+class GStreamerAudioStreamDescription final: public AudioStreamDescription {
>+public:
>+    GStreamerAudioStreamDescription(GstAudioInfo info)
>+        : m_info(info)
>+        , m_caps(adoptGRef(gst_audio_info_to_caps(&m_info)))
>+    {
>+    }
>+
>+    GStreamerAudioStreamDescription(GstAudioInfo *info)
>+        : m_info(*info)
>+        , m_caps(adoptGRef(gst_audio_info_to_caps(&m_info)))
>+    {
>+    }
>+
>+    GStreamerAudioStreamDescription()
>+    {
>+        gst_audio_info_init(&m_info);
>+    }
>+
>+    WEBCORE_EXPORT ~GStreamerAudioStreamDescription() { };
>+
>+    const PlatformDescription& platformDescription() const
>+    {
>+        m_platformDescription = { PlatformDescription::GStreamerAudioStreamDescription, (AudioStreamBasicDescription*) &m_info };
>+
>+        return m_platformDescription;
>+    }
>+
>+    WEBCORE_EXPORT PCMFormat format() const final {
>+        switch (GST_AUDIO_INFO_FORMAT(&m_info)) {
>+        case GST_AUDIO_FORMAT_S16LE:
>+        case GST_AUDIO_FORMAT_S16BE:
>+            return Int16;
>+        case GST_AUDIO_FORMAT_S32LE:
>+        case GST_AUDIO_FORMAT_S32BE:
>+            return Int32;
>+        case GST_AUDIO_FORMAT_F32LE:
>+        case GST_AUDIO_FORMAT_F32BE:
>+            return Float32;
>+        case GST_AUDIO_FORMAT_F64LE:
>+        case GST_AUDIO_FORMAT_F64BE:
>+            return Float64;
>+        default:
>+            break;
>+        }
>+        return None;
>+    }
>+
>+    double sampleRate() const final { return GST_AUDIO_INFO_RATE (&m_info); }
>+    bool isPCM() const final { return format() != None; }
>+    bool isInterleaved() const final { return GST_AUDIO_INFO_LAYOUT (&m_info) == GST_AUDIO_LAYOUT_INTERLEAVED; }
>+    bool isSignedInteger() const final { return GST_AUDIO_INFO_IS_INTEGER (&m_info); }
>+    bool isNativeEndian() const final { return GST_AUDIO_INFO_ENDIANNESS (&m_info) == G_BYTE_ORDER; }
>+    bool isFloat() const final { return GST_AUDIO_INFO_IS_FLOAT (&m_info); }
>+
>+    uint32_t numberOfInterleavedChannels() const final { return isInterleaved() ? GST_AUDIO_INFO_CHANNELS (&m_info) : TRUE; }
>+    uint32_t numberOfChannelStreams() const final { return GST_AUDIO_INFO_CHANNELS (&m_info); }
>+    uint32_t numberOfChannels() const final { return GST_AUDIO_INFO_CHANNELS (&m_info); }
>+    uint32_t sampleWordSize() const final { return GST_AUDIO_INFO_BPS (&m_info); }
>+
>+    bool operator==(const GStreamerAudioStreamDescription& other) { return gst_audio_info_is_equal (&m_info, &other.m_info); }
>+    bool operator!=(const GStreamerAudioStreamDescription& other) { return !operator == (other); }
>+
>+    GstCaps* caps() { return m_caps.get(); }
>+    GstAudioInfo* getInfo() { return &m_info; }
>+
>+private:
>+    GstAudioInfo m_info;
>+    GRefPtr<GstCaps> m_caps;
>+    mutable PlatformDescription m_platformDescription;
>+};
>+
>+} // WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDevice.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDevice.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..e418aead73f970b6b8df468fee80d363123d874b
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDevice.h
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (C) 2017 Igalia S.L. All rights reserved.
>+ *
>+ *  This library is free software; you can redistribute it and/or
>+ *  modify it under the terms of the GNU Lesser General Public
>+ *  License as published by the Free Software Foundation; either
>+ *  version 2 of the License, or (at your option) any later version.
>+ *
>+ *  This library is distributed in the hope that it will be useful,
>+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
>+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
>+ *  Lesser General Public License for more details.
>+ *
>+ *  You should have received a copy of the GNU Lesser General Public
>+ *  License along with this library; if not, write to the Free Software
>+ *  Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
>+ */
>+
>+#pragma once
>+
>+#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
>+
>+#include "CaptureDevice.h"
>+#include "GRefPtrGStreamer.h"
>+
>+#include <gst/gst.h>
>+
>+namespace WebCore {
>+
>+class GStreamerCaptureDevice : public CaptureDevice {
>+public:
>+    GStreamerCaptureDevice(GRefPtr<GstDevice>&& device, const String& persistentId, DeviceType type, const String& label, const String& groupId = emptyString())
>+        : CaptureDevice(persistentId, type, label, groupId)
>+        , m_device(device)
>+    {
>+    }
>+
>+    GstCaps* caps() const { return gst_device_get_caps(m_device.get()); }
>+    GstDevice * device() { return m_device.get(); }
>+
>+private:
>+    GRefPtr<GstDevice> m_device;
>+};
>+
>+}
>+
>+#endif // ENABLE(MEDIA_STREAM)  && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..df26b647db7569b9ff9b5080f93a2788aeb79117
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.cpp
>@@ -0,0 +1,138 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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"
>+
>+#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
>+#include "GStreamerCaptureDeviceManager.h"
>+
>+#include "GStreamerCommon.h"
>+
>+namespace WebCore {
>+
>+GStreamerAudioCaptureDeviceManager& GStreamerAudioCaptureDeviceManager::singleton()
>+{
>+    static NeverDestroyed<GStreamerAudioCaptureDeviceManager> manager;
>+    return manager;
>+}
>+
>+GStreamerVideoCaptureDeviceManager& GStreamerVideoCaptureDeviceManager::singleton()
>+{
>+    static NeverDestroyed<GStreamerVideoCaptureDeviceManager> manager;
>+    return manager;
>+}
>+
>+GStreamerDisplayCaptureDeviceManager& GStreamerDisplayCaptureDeviceManager::singleton()
>+{
>+    static NeverDestroyed<GStreamerDisplayCaptureDeviceManager> manager;
>+    return manager;
>+}
>+
>+std::optional<GStreamerCaptureDevice> GStreamerCaptureDeviceManager::gstreamerDeviceWithUID(const String& deviceID)
>+{
>+    captureDevices();
>+
>+    for (auto& device : m_gstreamerDevices) {
>+        if (device.persistentId() == deviceID)
>+            return device;
>+    }
>+    return std::nullopt;
>+}
>+
>+const Vector<CaptureDevice>& GStreamerCaptureDeviceManager::captureDevices()
>+{
>+    initializeGStreamer();
>+    if (m_devices.isEmpty())
>+        refreshCaptureDevices();
>+
>+    return m_devices;
>+}
>+
>+void GStreamerCaptureDeviceManager::deviceAdded(GRefPtr<GstDevice>&& device)
>+{
>+    GUniquePtr<GstStructure> properties(gst_device_get_properties(device.get()));
>+    const char* klass = gst_structure_get_string(properties.get(), "device.class");
>+
>+    if (klass && !g_strcmp0(klass, "monitor"))
>+        return;
>+
>+    CaptureDevice::DeviceType type = deviceType();
>+    GUniquePtr<char> deviceClassChar(gst_device_get_device_class(device.get()));
>+    String deviceClass(String(deviceClassChar.get()));
>+    if (type == CaptureDevice::DeviceType::Microphone && !deviceClass.startsWith("Audio"))
>+        return;
>+    if (type == CaptureDevice::DeviceType::Camera && !deviceClass.startsWith("Video"))
>+        return;
>+
>+    // This isn't really a UID but should be good enough (libwebrtc
>+    // itself does that at least for pulseaudio devices).
>+    GUniquePtr<char> deviceName(gst_device_get_display_name(device.get()));
>+    String identifier = String::fromUTF8(deviceName.get());
>+
>+    auto gstCaptureDevice = GStreamerCaptureDevice(WTFMove(device), identifier, type, identifier);
>+    gstCaptureDevice.setEnabled(true);
>+    m_gstreamerDevices.append(WTFMove(gstCaptureDevice));
>+    // FIXME: We need a CaptureDevice copy in other vector just for captureDevices API.
>+    auto captureDevice = CaptureDevice(identifier, type, identifier);
>+    captureDevice.setEnabled(true);
>+    m_devices.append(WTFMove(captureDevice));
>+}
>+
>+void GStreamerCaptureDeviceManager::refreshCaptureDevices()
>+{
>+    if (!m_deviceMonitor) {
>+        m_deviceMonitor = adoptGRef(gst_device_monitor_new());
>+
>+        CaptureDevice::DeviceType type = deviceType();
>+        if (type == CaptureDevice::DeviceType::Camera) {
>+            GRefPtr<GstCaps> caps = adoptGRef(gst_caps_new_empty_simple("video/x-raw"));
>+            gst_device_monitor_add_filter(m_deviceMonitor.get(), "Video/Source", caps.get());
>+        } else if (type == CaptureDevice::DeviceType::Microphone) {
>+            GRefPtr<GstCaps> caps = adoptGRef(gst_caps_new_empty_simple("audio/x-raw"));
>+            gst_device_monitor_add_filter(m_deviceMonitor.get(), "Audio/Source", caps.get());
>+        }
>+        // FIXME: Add monitor for added/removed messages on the bus.
>+    }
>+
>+    if (!gst_device_monitor_start(m_deviceMonitor.get())) {
>+        GST_WARNING_OBJECT(m_deviceMonitor.get(), "Could not start device monitor");
>+        m_deviceMonitor = nullptr;
>+
>+        return;
>+    }
>+
>+    GList* devices = gst_device_monitor_get_devices(m_deviceMonitor.get());
>+    while (devices) {
>+        GRefPtr<GstDevice> device = adoptGRef(GST_DEVICE_CAST(devices->data));
>+        deviceAdded(WTFMove(device));
>+        devices = g_list_delete_link(devices, devices);
>+    }
>+
>+    gst_device_monitor_stop(m_deviceMonitor.get());
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..a185ed7e50c15a1d25128d318db4b1a5d004ce6c
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCaptureDeviceManager.h
>@@ -0,0 +1,78 @@
>+/*
>+ *  Copyright (C) 2017 Igalia S.L. All rights reserved.
>+ *
>+ *  This library is free software; you can redistribute it and/or
>+ *  modify it under the terms of the GNU Lesser General Public
>+ *  License as published by the Free Software Foundation; either
>+ *  version 2 of the License, or (at your option) any later version.
>+ *
>+ *  This library is distributed in the hope that it will be useful,
>+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
>+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
>+ *  Lesser General Public License for more details.
>+ *
>+ *  You should have received a copy of the GNU Lesser General Public
>+ *  License along with this library; if not, write to the Free Software
>+ *  Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
>+ */
>+
>+#pragma once
>+
>+#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
>+
>+#include "CaptureDeviceManager.h"
>+#include "GRefPtrGStreamer.h"
>+#include "GStreamerCaptureDevice.h"
>+
>+namespace WebCore {
>+
>+class GStreamerCaptureDeviceManager : public CaptureDeviceManager {
>+public:
>+    std::optional<GStreamerCaptureDevice> gstreamerDeviceWithUID(const String&);
>+
>+    void refreshCaptureDevices() final;
>+    const Vector<CaptureDevice>& captureDevices() final;
>+    virtual CaptureDevice::DeviceType deviceType() = 0;
>+
>+private:
>+    void deviceAdded(GRefPtr<GstDevice>&&);
>+
>+    GRefPtr<GstDeviceMonitor> m_deviceMonitor;
>+    Vector<GStreamerCaptureDevice> m_gstreamerDevices;
>+    Vector<CaptureDevice> m_devices;
>+};
>+
>+class GStreamerAudioCaptureDeviceManager final : public GStreamerCaptureDeviceManager {
>+    friend class NeverDestroyed<GStreamerAudioCaptureDeviceManager>;
>+public:
>+    static GStreamerAudioCaptureDeviceManager& singleton();
>+    CaptureDevice::DeviceType deviceType() final { return CaptureDevice::DeviceType::Microphone; }
>+private:
>+    GStreamerAudioCaptureDeviceManager() = default;
>+    ~GStreamerAudioCaptureDeviceManager() = default;
>+};
>+
>+class GStreamerVideoCaptureDeviceManager final : public GStreamerCaptureDeviceManager {
>+    friend class NeverDestroyed<GStreamerVideoCaptureDeviceManager>;
>+public:
>+    static GStreamerVideoCaptureDeviceManager& singleton();
>+    static RealtimeMediaSource::VideoCaptureFactory& videoFactory();
>+    CaptureDevice::DeviceType deviceType() final { return CaptureDevice::DeviceType::Camera; }
>+private:
>+    GStreamerVideoCaptureDeviceManager() = default;
>+    ~GStreamerVideoCaptureDeviceManager() = default;
>+};
>+
>+class GStreamerDisplayCaptureDeviceManager final : public GStreamerCaptureDeviceManager {
>+    friend class NeverDestroyed<GStreamerDisplayCaptureDeviceManager>;
>+public:
>+    static GStreamerDisplayCaptureDeviceManager& singleton();
>+    CaptureDevice::DeviceType deviceType() final { return CaptureDevice::DeviceType::Screen; }
>+private:
>+    GStreamerDisplayCaptureDeviceManager() = default;
>+    ~GStreamerDisplayCaptureDeviceManager() = default;
>+};
>+
>+}
>+
>+#endif // ENABLE(MEDIA_STREAM)  && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..9ca52b51180f7b1aca00eabf86585d5387a02839
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp
>@@ -0,0 +1,181 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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"
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "GStreamerCapturer.h"
>+
>+#include <gst/app/gstappsink.h>
>+#include <mutex>
>+#include <webrtc/api/mediastreaminterface.h>
>+
>+GST_DEBUG_CATEGORY(webkit_capturer_debug);
>+#define GST_CAT_DEFAULT webkit_capturer_debug
>+
>+namespace WebCore {
>+
>+static void initializeGStreamerAndDebug()
>+{
>+    initializeGStreamer();
>+
>+    static std::once_flag debugRegisteredFlag;
>+    std::call_once(debugRegisteredFlag, [] {
>+        GST_DEBUG_CATEGORY_INIT(webkit_capturer_debug, "webkitcapturer", 0, "WebKit Capturer");
>+    });
>+}
>+
>+GStreamerCapturer::GStreamerCapturer(GStreamerCaptureDevice device, GRefPtr<GstCaps> caps)
>+    : m_device(device.device())
>+    , m_caps(caps)
>+    , m_sourceFactory(nullptr)
>+{
>+    initializeGStreamerAndDebug();
>+}
>+
>+GStreamerCapturer::GStreamerCapturer(const char* sourceFactory, GRefPtr<GstCaps> caps)
>+    : m_device(nullptr)
>+    , m_caps(caps)
>+    , m_sourceFactory(sourceFactory)
>+{
>+    initializeGStreamerAndDebug();
>+}
>+
>+GstElement* GStreamerCapturer::createSource()
>+{
>+    if (m_sourceFactory) {
>+        m_src = adoptGRef(makeElement(m_sourceFactory));
>+        ASSERT(m_src);
>+
>+        return m_src.get();
>+    }
>+
>+    ASSERT(m_device);
>+    GUniquePtr<char> sourceName(g_strdup_printf("%s_%p", name(), this));
>+    m_src = gst_device_create_element(m_device.get(), sourceName.get());
>+    ASSERT(m_src);
>+
>+    return m_src.get();
>+}
>+
>+GstCaps* GStreamerCapturer::caps()
>+{
>+    if (m_sourceFactory) {
>+        GRefPtr<GstElement> element = makeElement(m_sourceFactory);
>+        auto pad = adoptGRef(gst_element_get_static_pad(element.get(), "src"));
>+
>+        return gst_pad_query_caps(pad.get(), nullptr);
>+    }
>+
>+    ASSERT(m_device);
>+    return gst_device_get_caps(m_device.get());
>+}
>+
>+void GStreamerCapturer::setupPipeline()
>+{
>+    m_pipeline = makeElement("pipeline");
>+
>+    GRefPtr<GstElement> source = createSource();
>+    GRefPtr<GstElement> converter = createConverter();
>+
>+    m_capsfilter = makeElement("capsfilter");
>+    m_tee = makeElement("tee");
>+    m_sink = makeElement("appsink");
>+
>+    gst_app_sink_set_emit_signals(GST_APP_SINK(m_sink.get()), TRUE);
>+    g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
>+
>+    gst_bin_add_many(GST_BIN(m_pipeline.get()), source.get(), converter.get(), m_capsfilter.get(), m_tee.get(), nullptr);
>+    gst_element_link_many(source.get(), converter.get(), m_capsfilter.get(), m_tee.get(), NULL);
>+
>+    addSink(m_sink.get());
>+
>+    connectSimpleBusMessageCallback(pipeline());
>+}
>+
>+GstElement* GStreamerCapturer::makeElement(const char* factoryName)
>+{
>+    auto element = gst_element_factory_make(factoryName, nullptr);
>+    GUniquePtr<char> capturerName(g_strdup_printf("%s_capturer_%s_%p", name(), GST_OBJECT_NAME(element), this));
>+    gst_object_set_name(GST_OBJECT(element), capturerName.get());
>+
>+    return element;
>+}
>+
>+void GStreamerCapturer::addSink(GstElement *newSink)
>+{
>+    if (!m_pipeline || !m_tee)
>+        return;
>+
>+    auto queue = makeElement("queue");
>+    gst_bin_add_many(GST_BIN(pipeline()), queue, newSink, nullptr);
>+    gst_element_sync_state_with_parent(queue);
>+    gst_element_sync_state_with_parent(newSink);
>+
>+    if (!gst_element_link_pads(m_tee.get(), "src_%u", queue, "sink"))
>+        ASSERT_NOT_REACHED();
>+
>+    if (!gst_element_link(queue, newSink))
>+        ASSERT_NOT_REACHED();
>+
>+    if (newSink == sink()) {
>+        GST_INFO_OBJECT(pipeline(), "Setting queue as leaky upstream",
>+            " so that the player can set the sink as to PAUSED without "
>+            " setting the whole capturer to PAUSED");
>+        g_object_set(queue, "leaky", 2 /* upstream */, nullptr);
>+    }
>+
>+    GST_INFO_OBJECT(pipeline(), "Adding sink: %" GST_PTR_FORMAT, newSink);
>+
>+    GUniquePtr<char> dumpName(g_strdup_printf("%s_sink_%s_added", GST_OBJECT_NAME(pipeline()), GST_OBJECT_NAME(newSink)));
>+    GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(pipeline()), GST_DEBUG_GRAPH_SHOW_ALL, dumpName.get());
>+}
>+
>+void GStreamerCapturer::play()
>+{
>+    ASSERT(m_pipeline);
>+
>+    GST_INFO_OBJECT((gpointer) pipeline(), "Going to PLAYING!");
>+
>+    gst_element_set_state(pipeline(), GST_STATE_PLAYING);
>+}
>+
>+void GStreamerCapturer::stop()
>+{
>+    ASSERT(m_pipeline);
>+
>+    GST_INFO_OBJECT((gpointer) pipeline(), "Tearing down!");
>+
>+    // Make sure to remove sync handler before tearing down, avoiding
>+    // possible deadlocks.
>+    GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(pipeline())));
>+    gst_bus_set_sync_handler(bus.get(), nullptr, nullptr, nullptr);
>+
>+    gst_element_set_state(pipeline(), GST_STATE_NULL);
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..d734b81d1b83ae0898dffcf8112fd3e185b637c8
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h
>@@ -0,0 +1,77 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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 ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+
>+#include "GStreamerCommon.h"
>+
>+#include "LibWebRTCMacros.h"
>+#include "gstreamer/GStreamerCaptureDevice.h"
>+
>+#include <gst/gst.h>
>+
>+#pragma once
>+
>+namespace WebCore {
>+
>+class GStreamerCapturer {
>+public:
>+    GStreamerCapturer(GStreamerCaptureDevice, GRefPtr<GstCaps>);
>+    GStreamerCapturer(const char* sourceFactory, GRefPtr<GstCaps>);
>+
>+    void setupPipeline();
>+    virtual void play();
>+    virtual void stop();
>+    GstCaps* caps();
>+    void addSink(GstElement *newSink);
>+    GstElement* makeElement(const char* factoryName);
>+    GstElement* createSource();
>+    virtual const char* name() = 0;
>+
>+    GstElement* sink() const { return m_sink.get(); }
>+    void setSink(GstElement* sink) { m_sink = adoptGRef(sink); };
>+
>+    GstElement* pipeline() const { return m_pipeline.get(); }
>+    virtual GstElement* createConverter() = 0;
>+
>+protected:
>+    GRefPtr<GstElement> m_sink;
>+    GRefPtr<GstElement> m_src;
>+    GRefPtr<GstElement> m_tee;
>+    GRefPtr<GstElement> m_capsfilter;
>+    GRefPtr<GstDevice> m_device;
>+    GRefPtr<GstCaps> m_caps;
>+    GRefPtr<GstElement> m_pipeline;
>+
>+private:
>+    const char* m_sourceFactory;
>+
>+};
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..3bd7581a2e4753fd7f00cc90bdff60f8ccd63616
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.cpp
>@@ -0,0 +1,269 @@
>+/*
>+ * Copyright (C) 2017 Apple Inc. All rights reserved.
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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"
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "GStreamerVideoCaptureSource.h"
>+
>+#include "GStreamerCaptureDeviceManager.h"
>+#include "MediaSampleGStreamer.h"
>+
>+#include <gst/app/gstappsink.h>
>+#include <webrtc/api/mediastreaminterface.h>
>+#include <webrtc/api/peerconnectioninterface.h>
>+#include <webrtc/media/base/videocommon.h>
>+#include <webrtc/media/engine/webrtcvideocapturer.h>
>+#include <webrtc/media/engine/webrtcvideocapturerfactory.h>
>+#include <webrtc/modules/video_capture/video_capture_defines.h>
>+
>+namespace WebCore {
>+
>+const static int defaultWidth = 640;
>+const static int defaultHeight = 480;
>+
>+GST_DEBUG_CATEGORY(webkit_video_capture_source_debug);
>+#define GST_CAT_DEFAULT webkit_video_capture_source_debug
>+
>+static void initializeGStreamerDebug()
>+{
>+    static std::once_flag debugRegisteredFlag;
>+    std::call_once(debugRegisteredFlag, [] {
>+        GST_DEBUG_CATEGORY_INIT(webkit_video_capture_source_debug, "webkitvideocapturesource", 0,
>+            "WebKit Video Capture Source.");
>+    });
>+}
>+
>+class GStreamerVideoCaptureSourceFactory final : public RealtimeMediaSource::VideoCaptureFactory {
>+public:
>+    CaptureSourceOrError createVideoCaptureSource(const CaptureDevice& device, const MediaConstraints* constraints) final
>+    {
>+        return GStreamerVideoCaptureSource::create(device.persistentId(), constraints);
>+    }
>+};
>+
>+RealtimeMediaSource::VideoCaptureFactory& libWebRTCVideoCaptureSourceFactory()
>+{
>+    static NeverDestroyed<GStreamerVideoCaptureSourceFactory> factory;
>+    return factory.get();
>+}
>+
>+CaptureSourceOrError GStreamerVideoCaptureSource::create(const String& deviceID, const MediaConstraints* constraints)
>+{
>+    auto device = GStreamerVideoCaptureDeviceManager::singleton().gstreamerDeviceWithUID(deviceID);
>+    if (!device) {
>+        auto errorMessage = String::format("GStreamerVideoCaptureSource::create(): GStreamer did not find the device: %s.", deviceID.utf8().data());
>+        return CaptureSourceOrError(WTFMove(errorMessage));
>+    }
>+
>+    auto source = adoptRef(*new GStreamerVideoCaptureSource(device.value()));
>+
>+    if (constraints) {
>+        auto result = source->applyConstraints(*constraints);
>+        if (result)
>+            return WTFMove(result.value().first);
>+    }
>+    return CaptureSourceOrError(WTFMove(source));
>+}
>+
>+RealtimeMediaSource::VideoCaptureFactory& GStreamerVideoCaptureSource::factory()
>+{
>+    return libWebRTCVideoCaptureSourceFactory();
>+}
>+
>+GStreamerVideoCaptureSource::GStreamerVideoCaptureSource(const String& deviceID, const String& name, const gchar *source_factory)
>+    : RealtimeMediaSource(deviceID, RealtimeMediaSource::Type::Video, name)
>+    , m_capturer(std::make_unique<GStreamerVideoCapturer>(source_factory))
>+{
>+    initializeGStreamerDebug();
>+}
>+
>+GStreamerVideoCaptureSource::GStreamerVideoCaptureSource(GStreamerCaptureDevice device)
>+    : RealtimeMediaSource(device.persistentId(), RealtimeMediaSource::Type::Video, device.label())
>+    , m_capturer(std::make_unique<GStreamerVideoCapturer>(device))
>+{
>+    initializeGStreamerDebug();
>+}
>+
>+GStreamerVideoCaptureSource::~GStreamerVideoCaptureSource()
>+{
>+}
>+
>+bool GStreamerVideoCaptureSource::applySize(const IntSize &size)
>+{
>+    m_capturer->setSize(size.width(), size.height());
>+
>+    return true;
>+}
>+
>+bool GStreamerVideoCaptureSource::applyFrameRate(double framerate)
>+{
>+    m_capturer->setFrameRate(framerate);
>+
>+    return true;
>+}
>+
>+void GStreamerVideoCaptureSource::startProducingData()
>+{
>+    m_capturer->setupPipeline();
>+    m_capturer->setSize(size().width(), size().height());
>+    m_capturer->setFrameRate(frameRate());
>+    g_signal_connect(m_capturer->sink(), "new-sample", G_CALLBACK(newSampleCallback), this);
>+    m_capturer->play();
>+}
>+
>+GstFlowReturn GStreamerVideoCaptureSource::newSampleCallback(GstElement* sink, GStreamerVideoCaptureSource* source)
>+{
>+    auto gstSample = adoptGRef(gst_app_sink_pull_sample(GST_APP_SINK(sink)));
>+    auto mediaSample = MediaSampleGStreamer::create(WTFMove(gstSample), WebCore::FloatSize(), String());
>+
>+    // FIXME - Check how presentationSize is supposed to be used here.
>+    callOnMainThread([protectedThis = makeRef(*source), mediaSample = WTFMove(mediaSample)] {
>+        protectedThis->videoSampleAvailable(mediaSample.get());
>+    });
>+
>+    return GST_FLOW_OK;
>+}
>+
>+void GStreamerVideoCaptureSource::stopProducingData()
>+{
>+    m_capturer->stop();
>+
>+    GST_INFO("Reset height and width after stopping source");
>+    setHeight(defaultHeight);
>+    setWidth(defaultWidth);
>+}
>+
>+const RealtimeMediaSourceCapabilities& GStreamerVideoCaptureSource::capabilities() const
>+{
>+    if (!m_capabilities) {
>+        RealtimeMediaSourceCapabilities capabilities(settings().supportedConstraints());
>+        GRefPtr<GstCaps> caps = adoptGRef(m_capturer->caps());
>+        int32_t minWidth = G_MAXINT32, minHeight = G_MAXINT32, minFramerate = G_MAXINT32;
>+        int32_t maxWidth = G_MININT32, maxHeight = G_MININT32, maxFramerate = G_MININT32;
>+
>+        for (guint i = 0; i < gst_caps_get_size(caps.get()); i++) {
>+            GstStructure* str = gst_caps_get_structure(caps.get(), i);
>+
>+            // Only accept raw video for now.
>+            if (!gst_structure_has_name(str, "video/x-raw"))
>+                continue;
>+
>+            int32_t tmpMinWidth, tmpMinHeight, tmpMinFPSNumerator, tmpMinFPSDenominator, tmpMinFramerate;
>+            int32_t tmpMaxWidth, tmpMaxHeight, tmpMaxFPSNumerator, tmpMaxFPSDenominator, tmpMaxFramerate;
>+
>+            if (!gst_structure_get(str, "width", GST_TYPE_INT_RANGE, &tmpMinWidth, &tmpMaxWidth, "height", GST_TYPE_INT_RANGE, &tmpMinHeight, &tmpMaxHeight, nullptr)) {
>+
>+                g_assert(gst_structure_get(str, "width", G_TYPE_INT, &tmpMinWidth, "height", G_TYPE_INT, &tmpMinHeight, nullptr));
>+                tmpMaxWidth = tmpMinWidth;
>+                tmpMaxHeight = tmpMinHeight;
>+            }
>+
>+            if (gst_structure_get(str, "framerate", GST_TYPE_FRACTION_RANGE, &tmpMinFPSNumerator, &tmpMinFPSDenominator, &tmpMaxFPSNumerator, &tmpMaxFPSDenominator, nullptr)) {
>+                tmpMinFramerate = (int)(tmpMinFPSNumerator / tmpMinFPSDenominator);
>+                tmpMaxFramerate = (int)(tmpMaxFPSNumerator / tmpMaxFPSDenominator);
>+            } else if (gst_structure_get(str,
>+                "framerate", GST_TYPE_FRACTION, &tmpMinFPSNumerator, &tmpMinFPSDenominator, nullptr)) {
>+                tmpMaxFPSNumerator = tmpMinFPSNumerator;
>+                tmpMaxFPSDenominator = tmpMinFPSDenominator;
>+                tmpMinFramerate = (int)(tmpMinFPSNumerator / tmpMinFPSDenominator);
>+                tmpMaxFramerate = (int)(tmpMaxFPSNumerator / tmpMaxFPSDenominator);
>+            } else {
>+                const GValue* frameRates(gst_structure_get_value(str, "framerate"));
>+                tmpMinFramerate = G_MAXINT;
>+                tmpMaxFramerate = 0;
>+
>+                guint frameRatesLength = static_cast<guint>(gst_value_list_get_size(frameRates)) - 1;
>+
>+                for (guint i = 0; i < frameRatesLength; i++) {
>+                    const GValue* val = gst_value_list_get_value(frameRates, i);
>+
>+                    g_assert(G_VALUE_TYPE(val) == GST_TYPE_FRACTION);
>+                    gint framerate = (int)(gst_value_get_fraction_numerator(val) / gst_value_get_fraction_denominator(val));
>+
>+                    tmpMinFramerate = std::min(tmpMinFramerate, framerate);
>+                    tmpMaxFramerate = std::max(tmpMaxFramerate, framerate);
>+                }
>+
>+                if (i > 0) {
>+                    minWidth = std::min(tmpMinWidth, minWidth);
>+                    minHeight = std::min(tmpMinHeight, minHeight);
>+                    minFramerate = std::min(tmpMinFramerate, minFramerate);
>+
>+                    maxWidth = std::max(tmpMaxWidth, maxWidth);
>+                    maxHeight = std::max(tmpMaxHeight, maxHeight);
>+                    maxFramerate = std::max(tmpMaxFramerate, maxFramerate);
>+                } else {
>+                    minWidth = tmpMinWidth;
>+                    minHeight = tmpMinHeight;
>+                    minFramerate = tmpMinFramerate;
>+
>+                    maxWidth = tmpMaxWidth;
>+                    maxHeight = tmpMaxHeight;
>+                    maxFramerate = tmpMaxFramerate;
>+                }
>+            }
>+
>+            capabilities.setDeviceId(id());
>+            capabilities.setWidth(CapabilityValueOrRange(minWidth, maxWidth));
>+            capabilities.setHeight(CapabilityValueOrRange(minHeight, maxHeight));
>+            capabilities.setFrameRate(CapabilityValueOrRange(minFramerate, maxFramerate));
>+            m_capabilities = WTFMove(capabilities);
>+        }
>+    }
>+
>+    return m_capabilities.value();
>+}
>+
>+const RealtimeMediaSourceSettings& GStreamerVideoCaptureSource::settings() const
>+{
>+    if (!m_currentSettings) {
>+        RealtimeMediaSourceSettings settings;
>+        settings.setDeviceId(id());
>+
>+        RealtimeMediaSourceSupportedConstraints supportedConstraints;
>+        supportedConstraints.setSupportsDeviceId(true);
>+        supportedConstraints.setSupportsFacingMode(true);
>+        supportedConstraints.setSupportsWidth(true);
>+        supportedConstraints.setSupportsHeight(true);
>+        supportedConstraints.setSupportsAspectRatio(true);
>+        supportedConstraints.setSupportsFrameRate(true);
>+        settings.setSupportedConstraints(supportedConstraints);
>+
>+        m_currentSettings = WTFMove(settings);
>+    }
>+
>+    m_currentSettings->setWidth(size().width());
>+    m_currentSettings->setHeight(size().height());
>+    m_currentSettings->setFrameRate(frameRate());
>+    m_currentSettings->setAspectRatio(aspectRatio());
>+    m_currentSettings->setFacingMode(facingMode());
>+    return m_currentSettings.value();
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..4fae63a696d03256117e9eb9b1d45e296141be16
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCaptureSource.h
>@@ -0,0 +1,66 @@
>+/*
>+ * Copyright (C) 2017 Apple Inc. All rights reserved.
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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 ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "GStreamerVideoCapturer.h"
>+#include "RealtimeMediaSource.h"
>+
>+namespace WebCore {
>+
>+class GStreamerVideoCaptureSource : public RealtimeMediaSource {
>+public:
>+    static CaptureSourceOrError create(const String& deviceID, const MediaConstraints*);
>+    WEBCORE_EXPORT static VideoCaptureFactory& factory();
>+
>+    const RealtimeMediaSourceCapabilities& capabilities() const final;
>+    const RealtimeMediaSourceSettings& settings() const final;
>+
>+private:
>+    GStreamerVideoCaptureSource(const String& deviceID, const String& name, const gchar * source_factory);
>+    virtual ~GStreamerVideoCaptureSource();
>+
>+    GStreamerVideoCaptureSource(GStreamerCaptureDevice);
>+
>+    static GstFlowReturn newSampleCallback(GstElement*, GStreamerVideoCaptureSource*);
>+
>+    void startProducingData() final;
>+    void stopProducingData() final;
>+
>+    bool isCaptureSource() const final { return true; }
>+    bool applySize(const IntSize&) final;
>+    bool applyFrameRate(double) final;
>+    bool applyAspectRatio(double) final { return true; }
>+
>+    mutable std::optional<RealtimeMediaSourceCapabilities> m_capabilities;
>+    mutable std::optional<RealtimeMediaSourceSettings> m_currentSettings;
>+    std::unique_ptr<GStreamerVideoCapturer> m_capturer;
>+};
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..13ede5baaf83f236fd332edb1f1ba1f1ff5cf70b
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.cpp
>@@ -0,0 +1,105 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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"
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "GStreamerVideoCapturer.h"
>+
>+#include <gst/app/gstappsink.h>
>+
>+namespace WebCore {
>+
>+GStreamerVideoCapturer::GStreamerVideoCapturer(GStreamerCaptureDevice device)
>+    : GStreamerCapturer(device, adoptGRef(gst_caps_new_empty_simple("video/x-raw")))
>+{
>+}
>+
>+GStreamerVideoCapturer::GStreamerVideoCapturer(const char* sourceFactory)
>+    : GStreamerCapturer(sourceFactory, adoptGRef(gst_caps_new_empty_simple("video/x-raw")))
>+{
>+}
>+
>+GstElement* GStreamerVideoCapturer::createConverter()
>+{
>+    // FIXME Handle errors.
>+    return gst_parse_bin_from_description("videoscale ! videoconvert ! videorate", TRUE, nullptr);
>+}
>+
>+GstVideoInfo GStreamerVideoCapturer::getBestFormat()
>+{
>+    GRefPtr<GstCaps> caps = adoptGRef(gst_caps_fixate(gst_device_get_caps(m_device.get())));
>+    GstVideoInfo info;
>+    gst_video_info_from_caps(&info, caps.get());
>+
>+    return info;
>+}
>+
>+bool GStreamerVideoCapturer::setSize(int width, int height)
>+{
>+    if (!width || !height)
>+        return false;
>+
>+    m_caps = adoptGRef(gst_caps_copy(m_caps.get()));
>+    gst_caps_set_simple(m_caps.get(), "width", G_TYPE_INT, width, "height", G_TYPE_INT, height, nullptr);
>+
>+    if (!m_capsfilter)
>+        return false;
>+
>+    g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
>+
>+    return true;
>+}
>+
>+bool GStreamerVideoCapturer::setFrameRate(double frameRate)
>+{
>+    int numerator, denominator;
>+
>+    gst_util_double_to_fraction(frameRate, &numerator, &denominator);
>+
>+    if (numerator < -G_MAXINT) {
>+        GST_INFO_OBJECT(m_pipeline.get(), "Framerate %f not allowed", frameRate);
>+        return false;
>+    }
>+
>+    if (!numerator) {
>+        GST_ERROR_OBJECT(m_pipeline.get(), "Do not force variable framerate");
>+        return false;
>+    }
>+
>+    m_caps = adoptGRef(gst_caps_copy(m_caps.get()));
>+    gst_caps_set_simple(m_caps.get(), "framerate", GST_TYPE_FRACTION, numerator, denominator, nullptr);
>+
>+    if (!m_capsfilter)
>+        return false;
>+
>+    g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
>+
>+    return true;
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..2475ac017e9c8c51ed7f92c77ed8b771b889f29b
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerVideoCapturer.h
>@@ -0,0 +1,50 @@
>+/*
>+ * Copyright (C) 2017 Igalia S.L. 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. ``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
>+ * 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 ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+
>+#include "GStreamerCapturer.h"
>+
>+#include <gst/video/video.h>
>+
>+namespace WebCore {
>+
>+class GStreamerVideoCapturer : public GStreamerCapturer {
>+public:
>+    GStreamerVideoCapturer(GStreamerCaptureDevice);
>+    GStreamerVideoCapturer(const char* source_factory);
>+
>+    GstElement* createConverter() final;
>+    const char* name() final { return "Video"; }
>+
>+    bool setSize(int width, int height);
>+    bool setFrameRate(double);
>+    GstVideoInfo getBestFormat();
>+};
>+
>+} // namespace WebCore
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.cpp b/Source/WebCore/platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..59de912c51505a29e523a03a66ad1b4b9e6bfb80
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.cpp
>@@ -0,0 +1,98 @@
>+/*
>+ * Copyright (C) 2013-2016 Apple, Inc. All rights reserved.
>+ * Copyright (C) 2017 Igalia S.L. 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 THE COPYRIGHT HOLDERS AND 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 THE COPYRIGHT
>+ * OWNER OR 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"
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+#include "RealtimeMediaSourceCenterLibWebRTC.h"
>+
>+#include "GStreamerAudioCaptureSource.h"
>+#include "GStreamerCaptureDevice.h"
>+#include "GStreamerCaptureDeviceManager.h"
>+#include "GStreamerVideoCaptureSource.h"
>+#include <wtf/MainThread.h>
>+
>+namespace WebCore {
>+
>+RealtimeMediaSource::AudioCaptureFactory& RealtimeMediaSourceCenterLibWebRTC::audioCaptureSourceFactory()
>+{
>+    return RealtimeMediaSourceCenterLibWebRTC::singleton().audioFactory();
>+}
>+
>+RealtimeMediaSourceCenterLibWebRTC& RealtimeMediaSourceCenterLibWebRTC::singleton()
>+{
>+    ASSERT(isMainThread());
>+    static NeverDestroyed<RealtimeMediaSourceCenterLibWebRTC> center;
>+    return center;
>+}
>+
>+RealtimeMediaSourceCenter& RealtimeMediaSourceCenter::platformCenter()
>+{
>+    return RealtimeMediaSourceCenterLibWebRTC::singleton();
>+}
>+
>+RealtimeMediaSourceCenterLibWebRTC::RealtimeMediaSourceCenterLibWebRTC()
>+    : m_libWebRTCProvider(LibWebRTCProvider::create())
>+{
>+}
>+
>+RealtimeMediaSourceCenterLibWebRTC::~RealtimeMediaSourceCenterLibWebRTC()
>+{
>+}
>+
>+RealtimeMediaSource::AudioCaptureFactory& RealtimeMediaSourceCenterLibWebRTC::audioFactory()
>+{
>+    if (m_audioFactoryOverride)
>+        return *m_audioFactoryOverride;
>+
>+    return GStreamerAudioCaptureSource::factory();
>+}
>+
>+RealtimeMediaSource::VideoCaptureFactory& RealtimeMediaSourceCenterLibWebRTC::videoFactory()
>+{
>+    return GStreamerVideoCaptureSource::factory();
>+}
>+
>+CaptureDeviceManager& RealtimeMediaSourceCenterLibWebRTC::audioCaptureDeviceManager()
>+{
>+    return GStreamerAudioCaptureDeviceManager::singleton();
>+}
>+
>+CaptureDeviceManager& RealtimeMediaSourceCenterLibWebRTC::videoCaptureDeviceManager()
>+{
>+    return GStreamerVideoCaptureDeviceManager::singleton();
>+}
>+
>+CaptureDeviceManager& RealtimeMediaSourceCenterLibWebRTC::displayCaptureDeviceManager()
>+{
>+    return GStreamerDisplayCaptureDeviceManager::singleton();
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.h b/Source/WebCore/platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..4b22d83c162b4538e345cdcc38fbb780c59ea882
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/gstreamer/RealtimeMediaSourceCenterLibWebRTC.h
>@@ -0,0 +1,75 @@
>+/*
>+ * Copyright (C) 2013-2016 Apple, Inc. All rights reserved.
>+ * Copyright (C) 2017 Igalia S.L. 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 THE COPYRIGHT HOLDERS AND 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 THE COPYRIGHT
>+ * OWNER OR 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 ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+
>+#include "CaptureDeviceManager.h"
>+#include "LibWebRTCProvider.h"
>+#include "RealtimeMediaSource.h"
>+#include "RealtimeMediaSourceCenter.h"
>+
>+#include <webrtc/api/peerconnectioninterface.h>
>+#include <wtf/RefPtr.h>
>+#include <wtf/text/WTFString.h>
>+
>+namespace WebCore {
>+
>+class RealtimeMediaSourceCenterLibWebRTC final : public RealtimeMediaSourceCenter {
>+public:
>+    WEBCORE_EXPORT static RealtimeMediaSourceCenterLibWebRTC& singleton();
>+
>+    webrtc::PeerConnectionFactoryInterface* factory() { return m_libWebRTCProvider->factory(); }
>+
>+    static RealtimeMediaSource::VideoCaptureFactory& videoCaptureSourceFactory();
>+    static RealtimeMediaSource::AudioCaptureFactory& audioCaptureSourceFactory();
>+
>+private:
>+    friend class NeverDestroyed<RealtimeMediaSourceCenterLibWebRTC>;
>+    RealtimeMediaSourceCenterLibWebRTC();
>+    ~RealtimeMediaSourceCenterLibWebRTC();
>+
>+    void setAudioFactory(RealtimeMediaSource::AudioCaptureFactory& factory) final { m_audioFactoryOverride = &factory; }
>+    void unsetAudioFactory(RealtimeMediaSource::AudioCaptureFactory&) final { m_audioFactoryOverride = nullptr; }
>+
>+    RealtimeMediaSource::AudioCaptureFactory& audioFactory() final;
>+    RealtimeMediaSource::VideoCaptureFactory& videoFactory() final;
>+
>+    CaptureDeviceManager& audioCaptureDeviceManager() final;
>+    CaptureDeviceManager& videoCaptureDeviceManager() final;
>+    CaptureDeviceManager& displayCaptureDeviceManager() final;
>+
>+    RealtimeMediaSource::AudioCaptureFactory* m_audioFactoryOverride { nullptr };
>+    UniqueRef<LibWebRTCProvider> m_libWebRTCProvider;
>+};
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+
>diff --git a/Source/WebKit/UIProcess/API/glib/WebKitPermissionRequest.cpp b/Source/WebKit/UIProcess/API/glib/WebKitPermissionRequest.cpp
>index 92235cef17e563a30b4d3c146d6c4579e9860250..ed20560b7207bb42898be586e3ad6659d1a8064c 100644
>--- a/Source/WebKit/UIProcess/API/glib/WebKitPermissionRequest.cpp
>+++ b/Source/WebKit/UIProcess/API/glib/WebKitPermissionRequest.cpp
>@@ -70,3 +70,31 @@ void webkit_permission_request_deny(WebKitPermissionRequest* request)
>     if (iface->deny)
>         iface->deny(request);
> }
>+
>+/**
>+ * webkit_permission_request_resolve_check:
>+ * @request: a #WebKitPermissionRequest representing a situation where
>+ * the engine needs to check if the user gave permission to do certain
>+ * type of operations for an origin.
>+ * @salt: random data generated by the embedder
>+ * that the engine will use to hash data in case the user does not
>+ * allow the origin to access certain information.
>+ * @allowed: a boolean specifying if the user allows the requested
>+ * operation, if it does the engine will send more information to the
>+ * origin, if it doesn't it will use the salt to hash the information
>+ * to generate
>+ *
>+ * We have to call this function to specify if a request is allowed
>+ * using the @allowed parameter. Also we add the random data that
>+ * would be used as a hash salt in case the user did not allow the
>+ * origin for this kind of requests, using the
>+ * @salt parameter.
>+ */
>+void webkit_permission_request_resolve_check(WebKitPermissionRequest *request, const gchar* mediaIdentifierHashSalt, gboolean allowed)
>+{
>+    g_return_if_fail(WEBKIT_IS_PERMISSION_REQUEST(request));
>+
>+    WebKitPermissionRequestIface* iface = WEBKIT_PERMISSION_REQUEST_GET_IFACE(request);
>+    if (iface->resolve_check)
>+        iface->resolve_check(request, mediaIdentifierHashSalt, allowed);
>+}
>diff --git a/Source/WebKit/UIProcess/API/glib/WebKitUIClient.cpp b/Source/WebKit/UIProcess/API/glib/WebKitUIClient.cpp
>index 42bfa9ff05fe9027e854f7db45f143352b467d5a..ee998e8739a1a22ca8cc9678e822cac9eab8c0a9 100644
>--- a/Source/WebKit/UIProcess/API/glib/WebKitUIClient.cpp
>+++ b/Source/WebKit/UIProcess/API/glib/WebKitUIClient.cpp
>@@ -191,6 +191,13 @@ private:
>         return true;
>     }
> 
>+    bool checkUserMediaPermissionForOrigin(WebPageProxy&, WebFrameProxy&, API::SecurityOrigin& userMediaDocumentOrigin, API::SecurityOrigin& topLevelDocumentOrigin, UserMediaPermissionCheckProxy& permissionRequest) override
>+    {
>+        GRefPtr<WebKitUserMediaPermissionRequest> userMediaPermissionRequest = adoptGRef(webkitUserMediaPermissionCheckCreate(permissionRequest, userMediaDocumentOrigin, topLevelDocumentOrigin));
>+        webkitWebViewMakePermissionCheck(m_webView, WEBKIT_PERMISSION_REQUEST(userMediaPermissionRequest.get()));
>+        return true;
>+    }
>+
>     void decidePolicyForNotificationPermissionRequest(WebPageProxy&, API::SecurityOrigin&, Function<void(bool)>&& completionHandler) final
>     {
>         GRefPtr<WebKitNotificationPermissionRequest> notificationPermissionRequest = adoptGRef(webkitNotificationPermissionRequestCreate(NotificationPermissionRequest::create(WTFMove(completionHandler)).ptr()));
>diff --git a/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequest.cpp b/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequest.cpp
>index f08bfc110f8a087161fe2af921810d1981b84e55..728a81b0cdcadbda9154454070789b2051a8bb5e 100644
>--- a/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequest.cpp
>+++ b/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequest.cpp
>@@ -19,6 +19,7 @@
> #include "config.h"
> #include "WebKitUserMediaPermissionRequest.h"
> 
>+#include "UserMediaPermissionCheckProxy.h"
> #include "UserMediaPermissionRequestProxy.h"
> #include "WebKitPermissionRequest.h"
> #include "WebKitUserMediaPermissionRequestPrivate.h"
>@@ -53,6 +54,7 @@ static void webkit_permission_request_interface_init(WebKitPermissionRequestIfac
> 
> struct _WebKitUserMediaPermissionRequestPrivate {
>     RefPtr<UserMediaPermissionRequestProxy> request;
>+    RefPtr<UserMediaPermissionCheckProxy> checkRequest;
>     bool madeDecision;
> };
> 
>@@ -95,16 +97,28 @@ static void webkitUserMediaPermissionRequestDeny(WebKitPermissionRequest* reques
>     priv->request->deny(UserMediaPermissionRequestProxy::UserMediaAccessDenialReason::PermissionDenied);
> }
> 
>+static void webkitUserMediaPermissionResolveCheck(WebKitPermissionRequest* request, const gchar* salt, gboolean allowed)
>+{
>+    ASSERT(WEBKIT_IS_USER_MEDIA_PERMISSION_REQUEST(request));
>+
>+    WebKitUserMediaPermissionRequestPrivate* priv = WEBKIT_USER_MEDIA_PERMISSION_REQUEST(request)->priv;
>+
>+    priv->checkRequest->setUserMediaAccessInfo(String::fromUTF8(salt), allowed);
>+}
>+
> static void webkit_permission_request_interface_init(WebKitPermissionRequestIface* iface)
> {
>     iface->allow = webkitUserMediaPermissionRequestAllow;
>     iface->deny = webkitUserMediaPermissionRequestDeny;
>+    iface->resolve_check = webkitUserMediaPermissionResolveCheck;
> }
> 
> static void webkitUserMediaPermissionRequestDispose(GObject* object)
> {
>     // Default behaviour when no decision has been made is denying the request.
>-    webkitUserMediaPermissionRequestDeny(WEBKIT_PERMISSION_REQUEST(object));
>+    WebKitUserMediaPermissionRequestPrivate* priv = WEBKIT_USER_MEDIA_PERMISSION_REQUEST(object)->priv;
>+    if (priv->request)
>+        webkitUserMediaPermissionRequestDeny(WEBKIT_PERMISSION_REQUEST(object));
>     G_OBJECT_CLASS(webkit_user_media_permission_request_parent_class)->dispose(object);
> }
> 
>@@ -118,7 +132,9 @@ static void webkitUserMediaPermissionRequestDispose(GObject* object)
>  */
> gboolean webkit_user_media_permission_is_for_audio_device(WebKitUserMediaPermissionRequest* request)
> {
>-    g_return_val_if_fail(request->priv->request, FALSE);
>+    g_return_val_if_fail(request->priv->request || request->priv->checkRequest, FALSE);
>+    if (request->priv->checkRequest)
>+        return FALSE;
>     return request->priv->request->requiresAudioCapture();
> }
> 
>@@ -132,7 +148,9 @@ gboolean webkit_user_media_permission_is_for_audio_device(WebKitUserMediaPermiss
>  */
> gboolean webkit_user_media_permission_is_for_video_device(WebKitUserMediaPermissionRequest* request)
> {
>-    g_return_val_if_fail(request->priv->request, FALSE);
>+    g_return_val_if_fail(request->priv->request || request->priv->checkRequest, FALSE);
>+    if (request->priv->checkRequest)
>+        return FALSE;
>     return request->priv->request->requiresVideoCapture();
> }
> 
>@@ -196,3 +214,14 @@ WebKitUserMediaPermissionRequest* webkitUserMediaPermissionRequestCreate(UserMed
>     usermediaPermissionRequest->priv->request = &request;
>     return usermediaPermissionRequest;
> }
>+
>+WebKitUserMediaPermissionRequest* webkitUserMediaPermissionCheckCreate(UserMediaPermissionCheckProxy& request, API::SecurityOrigin& userMediaDocumentOrigin, API::SecurityOrigin& topLevelDocumentOrigin)
>+{
>+    WebKitUserMediaPermissionRequest* usermediaPermissionRequest = WEBKIT_USER_MEDIA_PERMISSION_REQUEST(g_object_new(WEBKIT_TYPE_USER_MEDIA_PERMISSION_REQUEST, nullptr));
>+
>+    UNUSED_PARAM(userMediaDocumentOrigin);
>+    UNUSED_PARAM(topLevelDocumentOrigin);
>+
>+    usermediaPermissionRequest->priv->checkRequest = &request;
>+    return usermediaPermissionRequest;
>+}
>diff --git a/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequestPrivate.h b/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequestPrivate.h
>index 904a15be76ee75f6c9685e97cb65c3b969dc186c..29977268d424fbb147b833fa1ea3185399059e4e 100644
>--- a/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequestPrivate.h
>+++ b/Source/WebKit/UIProcess/API/glib/WebKitUserMediaPermissionRequestPrivate.h
>@@ -22,3 +22,4 @@
> #include "WebKitUserMediaPermissionRequest.h"
> 
> WebKitUserMediaPermissionRequest* webkitUserMediaPermissionRequestCreate(WebKit::UserMediaPermissionRequestProxy&, API::SecurityOrigin&, API::SecurityOrigin&);
>+WebKitUserMediaPermissionRequest* webkitUserMediaPermissionCheckCreate(WebKit::UserMediaPermissionCheckProxy&, API::SecurityOrigin&, API::SecurityOrigin&);
>diff --git a/Source/WebKit/UIProcess/API/glib/WebKitWebView.cpp b/Source/WebKit/UIProcess/API/glib/WebKitWebView.cpp
>index 3b7f2871520b634b8f97b7068b1b773705a536f0..ed670479cbef07cf80a174d3e50e8d6833441c9e 100644
>--- a/Source/WebKit/UIProcess/API/glib/WebKitWebView.cpp
>+++ b/Source/WebKit/UIProcess/API/glib/WebKitWebView.cpp
>@@ -69,6 +69,8 @@
> #include <WebCore/RefPtrCairo.h>
> #include <WebCore/URL.h>
> #include <glib/gi18n-lib.h>
>+#include <wtf/CryptographicallyRandomNumber.h>
>+#include <wtf/HexNumber.h>
> #include <wtf/SetForScope.h>
> #include <wtf/glib/GRefPtr.h>
> #include <wtf/glib/WTFGType.h>
>@@ -124,6 +126,7 @@ enum {
> 
>     DECIDE_POLICY,
>     PERMISSION_REQUEST,
>+    PERMISSION_CHECK,
> 
>     MOUSE_TARGET_CHANGED,
> 
>@@ -212,6 +215,7 @@ struct _WebKitWebViewPrivate {
>     CString title;
>     CString customTextEncoding;
>     CString activeURI;
>+    String mediaIdentifierHashSalt;
>     bool isLoading;
>     bool isEphemeral;
>     bool isControlledByAutomation;
>@@ -428,6 +432,26 @@ static gboolean webkitWebViewPermissionRequest(WebKitWebView*, WebKitPermissionR
>     return TRUE;
> }
> 
>+static gboolean webkitWebViewPermissionCheck(WebKitWebView* webView, WebKitPermissionRequest* request)
>+{
>+    WebKitWebViewPrivate* priv = webView->priv;
>+    if (priv->mediaIdentifierHashSalt.isNull()) {
>+        static int hashSaltSize = 128;
>+        static int randomDataSize = hashSaltSize / 16;
>+        uint64_t randomData[randomDataSize];
>+        cryptographicallyRandomValues(reinterpret_cast<unsigned char*>(randomData), sizeof(randomData));
>+
>+        StringBuilder builder;
>+        builder.reserveCapacity(128);
>+        for (int i = 0; i < randomDataSize; i++)
>+            appendUnsigned64AsHex(randomData[0], builder);
>+        priv->mediaIdentifierHashSalt = builder.toString();
>+    }
>+
>+    webkit_permission_request_resolve_check(request, priv->mediaIdentifierHashSalt.utf8().data(), false);
>+    return TRUE;
>+}
>+
> static void allowModalDialogsChanged(WebKitSettings* settings, GParamSpec*, WebKitWebView* webView)
> {
>     getPage(webView).setCanRunModal(webkit_settings_get_allow_modal_dialogs(settings));
>@@ -835,6 +859,7 @@ static void webkit_web_view_class_init(WebKitWebViewClass* webViewClass)
>     webViewClass->script_dialog = webkitWebViewScriptDialog;
>     webViewClass->decide_policy = webkitWebViewDecidePolicy;
>     webViewClass->permission_request = webkitWebViewPermissionRequest;
>+    webViewClass->permission_check = webkitWebViewPermissionCheck;
>     webViewClass->run_file_chooser = webkitWebViewRunFileChooser;
>     webViewClass->authenticate = webkitWebViewAuthenticate;
>     webViewClass->show_notification = webkitWebViewShowNotification;
>@@ -1492,6 +1517,48 @@ static void webkit_web_view_class_init(WebKitWebViewClass* webViewClass)
>         g_cclosure_marshal_generic,
>         G_TYPE_BOOLEAN, 1, /* number of parameters */
>         WEBKIT_TYPE_PERMISSION_REQUEST);
>+
>+    /**
>+     * WebKitWebView::permission-check:
>+     * @web_view: the #WebKitWebView on which the signal is emitted
>+     * @request: the #WebKitPermissionRequest
>+     *
>+     * This signal is emitted when WebKit wants to check the
>+     * permissions already given to an origin by a user, such as
>+     * allowing the browser to use the microphone or the
>+     * camera. Usually the embedder does not ask the user to resolve
>+     * the request but uses what the user answered previously, denying
>+     * by default if the user did not give permissions specifically to
>+     * this origin.
>+     *
>+     * For more information about how to use the @request check
>+     * permission-request signal documentation.
>+     *
>+     * If the signal is not handled, the @request will be denied and a
>+     * random hash salt will be generated. Usually a embedder wants to
>+     * connect to the signal and call
>+     * webkit_permission_request_resolve_check() answering based on
>+     * what the user already decided about some previous
>+     * #WebKitPermissionRequest, that means storing the previous
>+     * decisions of the user about each origin. It also should use a
>+     * random salt that must be stored for each origin to make sure
>+     * the webpage always gets the same ID for the devices, or some
>+     * applications could not work properly.
>+     *
>+     * Returns: %TRUE to stop other handlers from being invoked for the event.
>+     *   %FALSE to propagate the event further.
>+     *
>+     */
>+    signals[PERMISSION_CHECK] = g_signal_new(
>+        "permission-check",
>+        G_TYPE_FROM_CLASS(webViewClass),
>+        G_SIGNAL_RUN_LAST,
>+        G_STRUCT_OFFSET(WebKitWebViewClass, permission_check),
>+        g_signal_accumulator_true_handled, nullptr /* accumulator data */,
>+        g_cclosure_marshal_generic,
>+        G_TYPE_BOOLEAN, 1, /* number of parameters */
>+        WEBKIT_TYPE_PERMISSION_REQUEST);
>+
>     /**
>      * WebKitWebView::mouse-target-changed:
>      * @web_view: the #WebKitWebView on which the signal is emitted
>@@ -2200,6 +2267,12 @@ void webkitWebViewMakePermissionRequest(WebKitWebView* webView, WebKitPermission
>     g_signal_emit(webView, signals[PERMISSION_REQUEST], 0, request, &returnValue);
> }
> 
>+void webkitWebViewMakePermissionCheck(WebKitWebView* webView, WebKitPermissionRequest* request)
>+{
>+    gboolean returnValue;
>+    g_signal_emit(webView, signals[PERMISSION_CHECK], 0, request, &returnValue);
>+}
>+
> void webkitWebViewMouseTargetChanged(WebKitWebView* webView, const WebHitTestResultData& hitTestResult, WebEvent::Modifiers modifiers)
> {
> #if PLATFORM(GTK)
>diff --git a/Source/WebKit/UIProcess/API/glib/WebKitWebViewPrivate.h b/Source/WebKit/UIProcess/API/glib/WebKitWebViewPrivate.h
>index 3605a915ed5010866f592a554325e2f65de0bf55..cc456fae8bc17317d05d307b9f95d29fee14e10b 100644
>--- a/Source/WebKit/UIProcess/API/glib/WebKitWebViewPrivate.h
>+++ b/Source/WebKit/UIProcess/API/glib/WebKitWebViewPrivate.h
>@@ -63,6 +63,7 @@ void webkitWebViewAcceptCurrentScriptDialog(WebKitWebView*);
> void webkitWebViewDismissCurrentScriptDialog(WebKitWebView*);
> std::optional<WebKitScriptDialogType> webkitWebViewGetCurrentScriptDialogType(WebKitWebView*);
> void webkitWebViewMakePermissionRequest(WebKitWebView*, WebKitPermissionRequest*);
>+void webkitWebViewMakePermissionCheck(WebKitWebView*, WebKitPermissionRequest*);
> void webkitWebViewMakePolicyDecision(WebKitWebView*, WebKitPolicyDecisionType, WebKitPolicyDecision*);
> void webkitWebViewMouseTargetChanged(WebKitWebView*, const WebKit::WebHitTestResultData&, WebKit::WebEvent::Modifiers);
> void webkitWebViewHandleDownloadRequest(WebKitWebView*, WebKit::DownloadProxy*);
>diff --git a/Source/WebKit/UIProcess/API/gtk/WebKitPermissionRequest.h b/Source/WebKit/UIProcess/API/gtk/WebKitPermissionRequest.h
>index a62f1f1a5b6d706253b307e26f43afa294891d86..1bdaeab26d345ed53cf3b820cdba677d8c7c4076 100644
>--- a/Source/WebKit/UIProcess/API/gtk/WebKitPermissionRequest.h
>+++ b/Source/WebKit/UIProcess/API/gtk/WebKitPermissionRequest.h
>@@ -42,16 +42,24 @@ struct _WebKitPermissionRequestIface {
> 
>     void (* allow) (WebKitPermissionRequest *request);
>     void (* deny)  (WebKitPermissionRequest *request);
>+    void (* resolve_check)  (WebKitPermissionRequest *request,
>+                             const gchar* salt,
>+                             gboolean allowed);
> };
> 
> WEBKIT_API GType
> webkit_permission_request_get_type (void);
> 
> WEBKIT_API void
>-webkit_permission_request_allow    (WebKitPermissionRequest *request);
>+webkit_permission_request_allow        (WebKitPermissionRequest *request);
> 
> WEBKIT_API void
>-webkit_permission_request_deny     (WebKitPermissionRequest *request);
>+webkit_permission_request_deny         (WebKitPermissionRequest *request);
>+
>+WEBKIT_API void
>+webkit_permission_request_resolve_check (WebKitPermissionRequest *request,
>+                                         const gchar* salt,
>+                                         gboolean allowed);
> 
> G_END_DECLS
> 
>diff --git a/Source/WebKit/UIProcess/API/gtk/WebKitWebView.h b/Source/WebKit/UIProcess/API/gtk/WebKitWebView.h
>index 03dfea538bf179eb45db19251cf4f54dfe183721..107ee3fa7652c14ede5cda9ccedb0fe23663b410 100644
>--- a/Source/WebKit/UIProcess/API/gtk/WebKitWebView.h
>+++ b/Source/WebKit/UIProcess/API/gtk/WebKitWebView.h
>@@ -231,6 +231,8 @@ struct _WebKitWebViewClass {
>                                                 WebKitPolicyDecisionType     type);
>     gboolean   (* permission_request)          (WebKitWebView               *web_view,
>                                                 WebKitPermissionRequest     *permission_request);
>+    gboolean   (* permission_check)            (WebKitWebView               *web_view,
>+                                                WebKitPermissionRequest     *permission_request);
>     void       (* mouse_target_changed)        (WebKitWebView               *web_view,
>                                                 WebKitHitTestResult         *hit_test_result,
>                                                 guint                        modifiers);
>diff --git a/Source/WebKit/UIProcess/API/gtk/docs/webkit2gtk-4.0-sections.txt b/Source/WebKit/UIProcess/API/gtk/docs/webkit2gtk-4.0-sections.txt
>index 8ac042007860d6045e35a7d50487e1652480f9b0..bf2b8c2c15a6cd053f752d2e0c5644cca87ed936 100644
>--- a/Source/WebKit/UIProcess/API/gtk/docs/webkit2gtk-4.0-sections.txt
>+++ b/Source/WebKit/UIProcess/API/gtk/docs/webkit2gtk-4.0-sections.txt
>@@ -596,6 +596,7 @@ webkit_download_get_type
> WebKitPermissionRequest
> webkit_permission_request_allow
> webkit_permission_request_deny
>+webkit_permission_request_resolve_check
> 
> <SUBSECTION Standard>
> WebKitPermissionRequestIface
>diff --git a/Source/WebKit/UIProcess/API/wpe/WebKitPermissionRequest.h b/Source/WebKit/UIProcess/API/wpe/WebKitPermissionRequest.h
>index 8ae7598e2ec86c548f10391993cb6c03d8414c0c..6609beeefa4bfe5cd8f8806d5df5bf167f8a8992 100644
>--- a/Source/WebKit/UIProcess/API/wpe/WebKitPermissionRequest.h
>+++ b/Source/WebKit/UIProcess/API/wpe/WebKitPermissionRequest.h
>@@ -42,6 +42,9 @@ struct _WebKitPermissionRequestIface {
> 
>     void (* allow) (WebKitPermissionRequest *request);
>     void (* deny)  (WebKitPermissionRequest *request);
>+    void (* resolve_check)  (WebKitPermissionRequest *request,
>+                             const gchar* salt,
>+                             gboolean allowed);
> };
> 
> WEBKIT_API GType
>@@ -53,6 +56,11 @@ webkit_permission_request_allow    (WebKitPermissionRequest *request);
> WEBKIT_API void
> webkit_permission_request_deny     (WebKitPermissionRequest *request);
> 
>+WEBKIT_API void
>+webkit_permission_request_resolve_check (WebKitPermissionRequest *request,
>+                                         const gchar* salt,
>+                                         gboolean allowed);
>+
> G_END_DECLS
> 
> #endif
>diff --git a/Source/WebKit/UIProcess/API/wpe/WebKitWebView.h b/Source/WebKit/UIProcess/API/wpe/WebKitWebView.h
>index 3c02f0d907327295314fe2f009c5b20e05bf9710..a3fc3a47bfb9ad5fc112ce3b605b86821a697c9a 100644
>--- a/Source/WebKit/UIProcess/API/wpe/WebKitWebView.h
>+++ b/Source/WebKit/UIProcess/API/wpe/WebKitWebView.h
>@@ -197,6 +197,8 @@ struct _WebKitWebViewClass {
>                                                     WebKitPolicyDecisionType     type);
>     gboolean       (* permission_request)          (WebKitWebView               *web_view,
>                                                     WebKitPermissionRequest     *permission_request);
>+    gboolean       (* permission_check)            (WebKitWebView               *web_view,
>+                                                    WebKitPermissionRequest     *permission_request);
>     void           (* mouse_target_changed)        (WebKitWebView               *web_view,
>                                                     WebKitHitTestResult         *hit_test_result,
>                                                     guint                        modifiers);
>diff --git a/Tools/ChangeLog b/Tools/ChangeLog
>index 7f293dcefe335367ace20b2ebcf26a67b50a625a..85b81813e4d116a0a5a6e6edf24bb798ca0c4e27 100644
>--- a/Tools/ChangeLog
>+++ b/Tools/ChangeLog
>@@ -1,3 +1,20 @@
>+2018-05-18  Alejandro G. Castro  <alex@igalia.com> and Thibault Saunier  <tsaunier@igalia.com>
>+
>+        [GTK][WPE] Add mediaDevices.enumerateDevices support
>+        https://bugs.webkit.org/show_bug.cgi?id=185761
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Added new API test for the mediaDevices.enumerateDevices:
>+        usermedia-enumeratedevices-permission-check.
>+
>+        * TestWebKitAPI/Tests/WebKitGLib/TestUIClient.cpp:
>+        (permissionCheck): Implemented the permission check resolution.
>+        (testWebViewUserMediaEnumerateDevicesPermissionCheck): Added the
>+        new test checking the enumerateDevices API when permission is
>+        denied and when permission is allowed for the origin.
>+        (beforeAll): Defined the new test.
>+
> 2018-05-17  Nan Wang  <n_wang@apple.com>
> 
>         AX: [macOS] Expose the primary screen height through AX API
>diff --git a/Tools/TestWebKitAPI/Tests/WebKitGLib/TestUIClient.cpp b/Tools/TestWebKitAPI/Tests/WebKitGLib/TestUIClient.cpp
>index 6939102fd318dfbe40e493be3c7a375e7cd5cc49..cfa254f24d32e3cc3a6b6cb8600a86fdd36099d6 100644
>--- a/Tools/TestWebKitAPI/Tests/WebKitGLib/TestUIClient.cpp
>+++ b/Tools/TestWebKitAPI/Tests/WebKitGLib/TestUIClient.cpp
>@@ -231,6 +231,25 @@ public:
>         return TRUE;
>     }
> 
>+    static gboolean permissionCheck(WebKitWebView*, WebKitPermissionRequest* request, UIClientTest* test)
>+    {
>+        g_assert(WEBKIT_IS_PERMISSION_REQUEST(request));
>+        test->assertObjectIsDeletedWhenTestFinishes(G_OBJECT(request));
>+
>+        if (test->m_verifyMediaTypes && WEBKIT_IS_USER_MEDIA_PERMISSION_REQUEST(request)) {
>+            WebKitUserMediaPermissionRequest* userMediaRequest = WEBKIT_USER_MEDIA_PERMISSION_REQUEST(request);
>+            g_assert(!webkit_user_media_permission_is_for_audio_device(userMediaRequest));
>+            g_assert(!webkit_user_media_permission_is_for_video_device(userMediaRequest));
>+        }
>+
>+        if (test->m_allowPermissionRequests)
>+            webkit_permission_request_resolve_check(request, "randomHashSalt", true);
>+        else
>+            webkit_permission_request_resolve_check(request, "randomHashSalt", false);
>+
>+        return TRUE;
>+    }
>+
>     static void permissionResultMessageReceivedCallback(WebKitUserContentManager* userContentManager, WebKitJavascriptResult* javascriptResult, UIClientTest* test)
>     {
>         test->m_permissionResult.reset(WebViewTest::javascriptResultToCString(javascriptResult));
>@@ -251,6 +270,7 @@ public:
>         g_signal_connect(m_webView, "script-dialog", G_CALLBACK(scriptDialog), this);
>         g_signal_connect(m_webView, "mouse-target-changed", G_CALLBACK(mouseTargetChanged), this);
>         g_signal_connect(m_webView, "permission-request", G_CALLBACK(permissionRequested), this);
>+        g_signal_connect(m_webView, "permission-check", G_CALLBACK(permissionCheck), this);
>         webkit_user_content_manager_register_script_message_handler(m_userContentManager.get(), "permission");
>         g_signal_connect(m_userContentManager.get(), "script-message-received::permission", G_CALLBACK(permissionResultMessageReceivedCallback), this);
>     }
>@@ -818,6 +838,47 @@ static void testWebViewGeolocationPermissionRequests(UIClientTest* test, gconstp
> #endif // ENABLE(GEOLOCATION)
> 
> #if ENABLE(MEDIA_STREAM)
>+static void testWebViewUserMediaEnumerateDevicesPermissionCheck(UIClientTest* test, gconstpointer)
>+{
>+    WebKitSettings* settings = webkit_web_view_get_settings(test->m_webView);
>+    gboolean enabled = webkit_settings_get_enable_media_stream(settings);
>+    webkit_settings_set_enable_media_stream(settings, TRUE);
>+
>+#if PLATFORM(GTK)
>+    test->showInWindowAndWaitUntilMapped();
>+#endif
>+    static const char* userMediaRequestHTML =
>+        "<html>"
>+        "  <script>"
>+        "  function runTest()"
>+        "  {"
>+        "    navigator.mediaDevices.enumerateDevices().then("
>+        "        function(devices) { "
>+        "            devices.forEach(function(device) {"
>+        "                                if (device.label) document.title = \"OK\";"
>+        "                                             else document.title = \"Permission denied\";"
>+        "            })"
>+        "    })"
>+        "  }"
>+        "  </script>"
>+        "  <body onload='runTest();'></body>"
>+        "</html>";
>+
>+    test->m_verifyMediaTypes = TRUE;
>+
>+    // Test denying a permission request.
>+    test->m_allowPermissionRequests = false;
>+    test->loadHtml(userMediaRequestHTML, nullptr);
>+    test->waitUntilTitleChangedTo("Permission denied");
>+
>+    // Test allowing a permission request.
>+    test->m_allowPermissionRequests = true;
>+    test->loadHtml(userMediaRequestHTML, nullptr);
>+    test->waitUntilTitleChangedTo("OK");
>+
>+    webkit_settings_set_enable_media_stream(settings, enabled);
>+}
>+
> static void testWebViewUserMediaPermissionRequests(UIClientTest* test, gconstpointer)
> {
>     WebKitSettings* settings = webkit_web_view_get_settings(test->m_webView);
>@@ -1140,6 +1201,7 @@ void beforeAll()
>     UIClientTest::add("WebKitWebView", "geolocation-permission-requests", testWebViewGeolocationPermissionRequests);
> #endif
> #if ENABLE(MEDIA_STREAM)
>+    UIClientTest::add("WebKitWebView", "usermedia-enumeratedevices-permission-check", testWebViewUserMediaEnumerateDevicesPermissionCheck);
>     UIClientTest::add("WebKitWebView", "usermedia-permission-requests", testWebViewUserMediaPermissionRequests);
>     UIClientTest::add("WebKitWebView", "audio-usermedia-permission-request", testWebViewAudioOnlyUserMediaPermissionRequests);
> #endif
>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index 29cac6a6ca048553fa419ab7aad7e0939a0609b2..34310d620d8cf09236b4fcdbf594739cd874df66 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,23 @@
>+2018-05-18  Alejandro G. Castro  <alex@igalia.com> and Thibault Saunier  <tsaunier@igalia.com>
>+
>+        [GTK][WPE] Add mediaDevices.enumerateDevices support
>+        https://bugs.webkit.org/show_bug.cgi?id=185761
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Remove the Failure expectation for the test
>+        media-devices-enumerate-devices.html. Just for the record we have
>+        also solved the following tests, but we are not removing the
>+        general skip of the mediastream directory until the next patch:
>+
>+        fast/mediastream/media-devices-enumerate-devices.html
>+        fast/mediastream/resources/enumerate-devices-frame.html
>+        http/tests/media/media-stream/enumerate-devices-source-id.html
>+        http/tests/media/media-stream/enumerate-devices-source-id-persistent.html
>+        http/tests/media/media-stream/resources/enumerate-devices-source-id-frame.html
>+
>+        * platform/gtk/TestExpectations:
>+
> 2018-05-17  Michael Saboff  <msaboff@apple.com>
> 
>         We don't throw SyntaxErrors for runtime generated regular expressions with errors
>diff --git a/LayoutTests/platform/gtk/TestExpectations b/LayoutTests/platform/gtk/TestExpectations
>index 8c62f2c8fe98752d41e8a9e01e2805ca4b812343..b7818a8e49acb79271718300b4f54d8c3b892e59 100644
>--- a/LayoutTests/platform/gtk/TestExpectations
>+++ b/LayoutTests/platform/gtk/TestExpectations
>@@ -636,7 +636,6 @@ webkit.org/b/79203 imported/w3c/web-platform-tests/mediacapture-streams/MediaStr
> webkit.org/b/151344 fast/mediastream/MediaStream-add-ended-tracks.html [ Timeout ]
> # Crash is bug #176801
> webkit.org/b/160996 fast/mediastream/MediaStream-video-element-video-tracks-disabled.html [ ImageOnlyFailure Crash ]
>-webkit.org/b/172269 fast/mediastream/media-devices-enumerate-devices.html [ Failure ]
> webkit.org/b/173257 fast/mediastream/getUserMedia-grant-persistency3.html [ Pass Failure ]
> 
> # Canvas captureStream support is not implemented

Flags:

mcatanzaro: review-

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Attachments on bug 185761: 340841 | 346900 | 351077 | 351080 | 351081 | 351526 | 351531 | 351533 | 351591 | 351952 | 352030