Attachment 356726 Details for Bug 186933 – Patch

[patch] Patch

bug-186933-20181206123451.patch (text/plain), 30.79 KB, created by Thibault Saunier on 2018-12-06 07:34:52 PST

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Creator: Thibault Saunier

Created: 2018-12-06 07:34:52 PST

Size: 30.79 KB

patch

obsolete

>Subversion Revision: 238666
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 0f2b337981efbb85889ff67fc1bdb56516af2658..6774fc04754dc775dab64961c266f1409f454594 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,47 @@
>+2018-12-05  Thibault Saunier  <tsaunier@igalia.com>
>+
>+        [WPE][GTK] Implement WebAudioSourceProviderGStreamer to allow bridging MediaStream and the WebAudio APIs
>+        https://bugs.webkit.org/show_bug.cgi?id=186933
>+
>+        Reusing the AudioSourceProviderGStreamer itself as it was doing almost everything we needed,
>+        just added a constructor to be able to create it from a MediaStreamTrackPrivate and made it a
>+        WebAudioSourceProvider which only means it is now a ThreadSafeRefCounted.
>+
>+        Sensibily refactored GStreamerMediaStreamSource so that we could reuse it to track a single
>+        MediaStreamTrack.
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Enabled all tests depending on that feature.
>+
>+        * platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp:
>+        (WebCore::AudioSourceProviderGStreamer::AudioSourceProviderGStreamer):
>+        (WebCore::AudioSourceProviderGStreamer::~AudioSourceProviderGStreamer):
>+        (WebCore::AudioSourceProviderGStreamer::setClient):
>+        * platform/audio/gstreamer/AudioSourceProviderGStreamer.h:
>+        * platform/mediastream/MediaStreamTrackPrivate.cpp:
>+        (WebCore::MediaStreamTrackPrivate::audioSourceProvider):
>+        * platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp:
>+        (WebCore::GStreamerAudioCapturer::GStreamerAudioCapturer):
>+        * platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h:
>+        * platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
>+        (WebCore::webkitMediaStreamSrcSetupSrc):
>+        (WebCore::webkitMediaStreamSrcSetupAppSrc):
>+        (WebCore::webkitMediaStreamSrcAddTrack):
>+        (WebCore::webkitMediaStreamSrcSetStream):
>+        (WebCore::webkitMediaStreamSrcNew):
>+        * platform/mediastream/gstreamer/GStreamerMediaStreamSource.h:
>+        * platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp:
>+        (WebCore::WrappedMockRealtimeAudioSource::WrappedMockRealtimeAudioSource):
>+        (WebCore::WrappedMockRealtimeAudioSource::start):
>+        (WebCore::WrappedMockRealtimeAudioSource::addHum):
>+        (WebCore::WrappedMockRealtimeAudioSource::render):
>+        (WebCore::WrappedMockRealtimeAudioSource::settingsDidChange):
>+        (WebCore::MockGStreamerAudioCaptureSource::startProducingData):
>+        * platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
>+        (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData): Handle the case where input buffers
>+          are "big" and process all the data we can for each runs of the method.
>+
> 2018-11-29  Rob Buis  <rbuis@igalia.com>
> 
>         Remove some superfluous code in ContentSecurityPolicy::upgradeInsecureRequestIfNeeded
>diff --git a/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp
>index d4a0ca2bf8a6e6be502aeb277adee2a8a6c35993..e7111f3c023ae615f0195302979116c3577b649f 100644
>--- a/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp
>+++ b/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp
>@@ -27,6 +27,10 @@
> #include <gst/audio/audio-info.h>
> #include <gst/base/gstadapter.h>
> 
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+#include "GStreamerAudioData.h"
>+#include "GStreamerMediaStreamSource.h"
>+#endif
> 
> namespace WebCore {
> 
>@@ -94,6 +98,31 @@ AudioSourceProviderGStreamer::AudioSourceProviderGStreamer()
>     m_frontRightAdapter = gst_adapter_new();
> }
> 
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+AudioSourceProviderGStreamer::AudioSourceProviderGStreamer(MediaStreamTrackPrivate& source)
>+    : m_notifier(MainThreadNotifier<MainThreadNotification>::create())
>+    , m_client(nullptr)
>+    , m_deinterleaveSourcePads(0)
>+    , m_deinterleavePadAddedHandlerId(0)
>+    , m_deinterleaveNoMorePadsHandlerId(0)
>+    , m_deinterleavePadRemovedHandlerId(0)
>+{
>+    m_frontLeftAdapter = gst_adapter_new();
>+    m_frontRightAdapter = gst_adapter_new();
>+    auto pipelineName = String::format("WebAudioProvider_MediaStreamTrack_%s", source.id().utf8().data());
>+    m_pipeline = adoptGRef(GST_ELEMENT(g_object_ref_sink(gst_element_factory_make("pipeline", pipelineName.utf8().data()))));
>+    auto src = webkitMediaStreamSrcNew();
>+    webkitMediaStreamSrcAddTrack(WEBKIT_MEDIA_STREAM_SRC(src), &source, true);
>+
>+    m_audioSinkBin = adoptGRef(GST_ELEMENT(g_object_ref_sink(gst_parse_bin_from_description("tee name=audioTee", true, nullptr))));
>+
>+    gst_bin_add_many(GST_BIN(m_pipeline.get()), src, m_audioSinkBin.get(), nullptr);
>+    gst_element_link(src, m_audioSinkBin.get());
>+
>+    connectSimpleBusMessageCallback(m_pipeline.get());
>+}
>+#endif
>+
> AudioSourceProviderGStreamer::~AudioSourceProviderGStreamer()
> {
>     m_notifier->invalidate();
>@@ -105,6 +134,9 @@ AudioSourceProviderGStreamer::~AudioSourceProviderGStreamer()
>         g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadRemovedHandlerId);
>     }
> 
>+    if (m_pipeline)
>+        gst_element_set_state(m_pipeline.get(), GST_STATE_NULL);
>+
>     g_object_unref(m_frontLeftAdapter);
>     g_object_unref(m_frontRightAdapter);
> }
>@@ -205,12 +237,17 @@ void AudioSourceProviderGStreamer::setClient(AudioSourceProviderClient* client)
>     ASSERT(client);
>     m_client = client;
> 
>+    if (m_pipeline)
>+        gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
>+
>     // The volume element is used to mute audio playback towards the
>     // autoaudiosink. This is needed to avoid double playback of audio
>     // from our audio sink and from the WebAudio AudioDestination node
>     // supposedly configured already by application side.
>     GRefPtr<GstElement> volumeElement = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "volume"));
>-    g_object_set(volumeElement.get(), "mute", TRUE, nullptr);
>+
>+    if (volumeElement)
>+        g_object_set(volumeElement.get(), "mute", TRUE, nullptr);
> 
>     // The audioconvert and audioresample elements are needed to
>     // ensure deinterleave and the sinks downstream receive buffers in
>diff --git a/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.h b/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.h
>index ab60198c880fdb94463bd7192dedfd6be07349f7..dfdadca02331f8f1656354166dff2ff3d7be8363 100644
>--- a/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.h
>+++ b/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.h
>@@ -28,14 +28,31 @@
> #include <wtf/Forward.h>
> #include <wtf/Noncopyable.h>
> 
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+#include "GStreamerAudioStreamDescription.h"
>+#include "MediaStreamTrackPrivate.h"
>+#include "WebAudioSourceProvider.h"
>+#endif
>+
> typedef struct _GstAdapter GstAdapter;
> typedef struct _GstAppSink GstAppSink;
> 
> namespace WebCore {
> 
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+class AudioSourceProviderGStreamer final : public WebAudioSourceProvider {
>+public:
>+    static Ref<AudioSourceProviderGStreamer> create(MediaStreamTrackPrivate& source)
>+    {
>+        return adoptRef(*new AudioSourceProviderGStreamer(source));
>+    }
>+    AudioSourceProviderGStreamer(MediaStreamTrackPrivate&);
>+#else
> class AudioSourceProviderGStreamer : public AudioSourceProvider {
>     WTF_MAKE_NONCOPYABLE(AudioSourceProviderGStreamer);
> public:
>+#endif
>+
>     AudioSourceProviderGStreamer();
>     ~AudioSourceProviderGStreamer();
> 
>@@ -54,6 +71,11 @@ public:
>     void clearAdapters();
> 
> private:
>+
>+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>+    GRefPtr<GstElement> m_pipeline;
>+#endif
>+
>     enum MainThreadNotification {
>         DeinterleavePadsConfigured = 1 << 0,
>     };
>diff --git a/Source/WebCore/platform/mediastream/MediaStreamTrackPrivate.cpp b/Source/WebCore/platform/mediastream/MediaStreamTrackPrivate.cpp
>index 5166de6f7fa1caa3a51fecda38fcdee046a359e1..95fa95ed4f94b84cd09d6f18bbfdf61d53f588f9 100644
>--- a/Source/WebCore/platform/mediastream/MediaStreamTrackPrivate.cpp
>+++ b/Source/WebCore/platform/mediastream/MediaStreamTrackPrivate.cpp
>@@ -35,6 +35,8 @@
> 
> #if PLATFORM(COCOA)
> #include "WebAudioSourceProviderAVFObjC.h"
>+#elif ENABLE(WEB_AUDIO) && ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
>+#include "AudioSourceProviderGStreamer.h"
> #else
> #include "WebAudioSourceProvider.h"
> #endif
>@@ -178,6 +180,9 @@ AudioSourceProvider* MediaStreamTrackPrivate::audioSourceProvider()
> #if PLATFORM(COCOA)
>     if (!m_audioSourceProvider)
>         m_audioSourceProvider = WebAudioSourceProviderAVFObjC::create(*this);
>+#elif USE(LIBWEBRTC) && USE(GSTREAMER)
>+    if (!m_audioSourceProvider)
>+        m_audioSourceProvider = AudioSourceProviderGStreamer::create(*this);
> #endif
>     return m_audioSourceProvider.get();
> }
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp
>index 8df758a109fee65e823993ea842418b4b3218a68..3c972b8a1a1592de3e746951c8ff2cda39bda3f2 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp
>@@ -36,7 +36,7 @@ GStreamerAudioCapturer::GStreamerAudioCapturer(GStreamerCaptureDevice device)
> }
> 
> GStreamerAudioCapturer::GStreamerAudioCapturer()
>-    : GStreamerCapturer("audiotestsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
>+    : GStreamerCapturer("appsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
> {
> }
> 
>@@ -44,9 +44,6 @@ GstElement* GStreamerAudioCapturer::createSource()
> {
>     GstElement* source = GStreamerCapturer::createSource();
> 
>-    if (!m_device)
>-        gst_util_set_object_arg(G_OBJECT(m_src.get()), "wave", "ticks");
>-
>     return source;
> }
> 
>@@ -62,14 +59,15 @@ GstElement* GStreamerAudioCapturer::createConverter()
> 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 {
>+    if (sampleRate <= 0) {
>         GST_INFO_OBJECT(m_pipeline.get(), "Not forcing sample rate");
>-        m_caps = adoptGRef(gst_caps_new_empty_simple("audio/x-raw"));
>+
>+        return false;
>     }
> 
>+    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));
>+
>     if (!m_capsfilter.get())
>         return false;
> 
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h
>index 861adef0108211d0c43b4ba25762bc6008f620b1..64fced400333d6b5294c7c6db9dc8570df2c3cc8 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h
>@@ -82,6 +82,7 @@ public:
>     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); }
>+    int bytesPerFrame() { return GST_AUDIO_INFO_BPF(&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); }
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
>index 2513ca94be9c7e000c46608ca110dcae1378ac2f..927d986c205bb3f0cb21d9ffceb00c30286a1e8c 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
>@@ -41,7 +41,6 @@ namespace WebCore {
> static void webkitMediaStreamSrcPushVideoSample(WebKitMediaStreamSrc* self, GstSample* gstsample);
> static void webkitMediaStreamSrcPushAudioSample(WebKitMediaStreamSrc* self, GstSample* gstsample);
> static void webkitMediaStreamSrcTrackEnded(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate&);
>-static void webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate*);
> static void webkitMediaStreamSrcRemoveTrackByType(WebKitMediaStreamSrc* self, RealtimeMediaSource::Type trackType);
> 
> static GstStaticPadTemplate videoSrcTemplate = GST_STATIC_PAD_TEMPLATE("video_src",
>@@ -156,7 +155,7 @@ public:
> 
>     void didAddTrack(MediaStreamTrackPrivate& track) final
>     {
>-        webkitMediaStreamSrcAddTrack(m_mediaStreamSrc.get(), &track);
>+        webkitMediaStreamSrcAddTrack(m_mediaStreamSrc.get(), &track, false);
>     }
> 
>     void didRemoveTrack(MediaStreamTrackPrivate& track) final
>@@ -182,8 +181,8 @@ struct _WebKitMediaStreamSrc {
>     std::unique_ptr<WebKitMediaStreamTrackObserver> mediaStreamTrackObserver;
>     std::unique_ptr<WebKitMediaStreamObserver> mediaStreamObserver;
>     volatile gint npads;
>-    gulong probeid;
>     RefPtr<MediaStreamPrivate> stream;
>+    RefPtr<MediaStreamTrackPrivate> track;
> 
>     GstFlowCombiner* flowCombiner;
>     GRefPtr<GstStreamCollection> streamCollection;
>@@ -314,8 +313,11 @@ static GstStateChangeReturn webkitMediaStreamSrcChangeState(GstElement* element,
>     if (transition == GST_STATE_CHANGE_PAUSED_TO_READY) {
> 
>         GST_OBJECT_LOCK(self);
>-        for (auto& track : self->stream->tracks())
>-            track->removeObserver(*self->mediaStreamTrackObserver.get());
>+        if (self->stream) {
>+            for (auto& track : self->stream->tracks())
>+                track->removeObserver(*self->mediaStreamTrackObserver.get());
>+        } else if (self->track)
>+            self->track->removeObserver(*self->mediaStreamTrackObserver.get());
>         GST_OBJECT_UNLOCK(self);
>     }
> 
>@@ -435,22 +437,26 @@ static GstPadProbeReturn webkitMediaStreamSrcPadProbeCb(GstPad* pad, GstPadProbe
> 
> static gboolean webkitMediaStreamSrcSetupSrc(WebKitMediaStreamSrc* self,
>     MediaStreamTrackPrivate* track, GstElement* element,
>-    GstStaticPadTemplate* pad_template, gboolean observe_track)
>+    GstStaticPadTemplate* pad_template, gboolean observe_track,
>+    bool onlyTrack)
> {
>     auto pad = adoptGRef(gst_element_get_static_pad(element, "src"));
> 
>     gst_bin_add(GST_BIN(self), element);
> 
>-    ProbeData* data = new ProbeData;
>-    data->self = WEBKIT_MEDIA_STREAM_SRC(self);
>-    data->pad_template = pad_template;
>-    data->track = track;
>-
>-    self->probeid = gst_pad_add_probe(pad.get(), (GstPadProbeType)GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
>-        (GstPadProbeCallback)webkitMediaStreamSrcPadProbeCb, data,
>-        [](gpointer data) {
>-            delete (ProbeData*)data;
>-        });
>+    if (!onlyTrack) {
>+        ProbeData* data = new ProbeData;
>+        data->self = WEBKIT_MEDIA_STREAM_SRC(self);
>+        data->pad_template = pad_template;
>+        data->track = track;
>+
>+        gst_pad_add_probe(pad.get(), (GstPadProbeType)GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
>+            (GstPadProbeCallback)webkitMediaStreamSrcPadProbeCb, data,
>+            [](gpointer data) {
>+                delete (ProbeData*)data;
>+            });
>+    } else
>+        webkitMediaStreamSrcAddPad(self, pad.get(), pad_template);
> 
>     if (observe_track)
>         track->addObserver(*self->mediaStreamTrackObserver.get());
>@@ -461,12 +467,12 @@ static gboolean webkitMediaStreamSrcSetupSrc(WebKitMediaStreamSrc* self,
> 
> static gboolean webkitMediaStreamSrcSetupAppSrc(WebKitMediaStreamSrc* self,
>     MediaStreamTrackPrivate* track, GstElement** element,
>-    GstStaticPadTemplate* pad_template)
>+    GstStaticPadTemplate* pad_template, bool onlyTrack)
> {
>     *element = gst_element_factory_make("appsrc", nullptr);
>     g_object_set(*element, "is-live", true, "format", GST_FORMAT_TIME, nullptr);
> 
>-    return webkitMediaStreamSrcSetupSrc(self, track, *element, pad_template, TRUE);
>+    return webkitMediaStreamSrcSetupSrc(self, track, *element, pad_template, TRUE, onlyTrack);
> }
> 
> static void webkitMediaStreamSrcPostStreamCollection(WebKitMediaStreamSrc* self, MediaStreamPrivate* stream)
>@@ -484,14 +490,20 @@ static void webkitMediaStreamSrcPostStreamCollection(WebKitMediaStreamSrc* self,
>         gst_message_new_stream_collection(GST_OBJECT(self), self->streamCollection.get()));
> }
> 
>-static void webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate* track)
>+bool webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate* track, bool onlyTrack)
> {
>+    bool res = false;
>     if (track->type() == RealtimeMediaSource::Type::Audio)
>-        webkitMediaStreamSrcSetupAppSrc(self, track, &self->audioSrc, &audioSrcTemplate);
>+        res = webkitMediaStreamSrcSetupAppSrc(self, track, &self->audioSrc, &audioSrcTemplate, onlyTrack);
>     else if (track->type() == RealtimeMediaSource::Type::Video)
>-        webkitMediaStreamSrcSetupAppSrc(self, track, &self->videoSrc, &videoSrcTemplate);
>+        res = webkitMediaStreamSrcSetupAppSrc(self, track, &self->videoSrc, &videoSrcTemplate, onlyTrack);
>     else
>         GST_INFO("Unsupported track type: %d", static_cast<int>(track->type()));
>+
>+    if (onlyTrack && res)
>+        self->track = track;
>+
>+    return false;
> }
> 
> static void webkitMediaStreamSrcRemoveTrackByType(WebKitMediaStreamSrc* self, RealtimeMediaSource::Type trackType)
>@@ -524,7 +536,7 @@ bool webkitMediaStreamSrcSetStream(WebKitMediaStreamSrc* self, MediaStreamPrivat
>     self->stream = stream;
>     self->stream->addObserver(*self->mediaStreamObserver.get());
>     for (auto& track : stream->tracks())
>-        webkitMediaStreamSrcAddTrack(self, track.get());
>+        webkitMediaStreamSrcAddTrack(self, track.get(), false);
> 
>     return TRUE;
> }
>@@ -593,6 +605,11 @@ static void webkitMediaStreamSrcTrackEnded(WebKitMediaStreamSrc* self,
>     gst_pad_push_event(pad.get(), gst_event_new_eos());
> }
> 
>+GstElement* webkitMediaStreamSrcNew(void)
>+{
>+    return GST_ELEMENT(g_object_new(webkit_media_stream_src_get_type(), nullptr));
>+}
>+
> } // WebCore
> #endif // GST_CHECK_VERSION(1, 10, 0)
> #endif // ENABLE(VIDEO) && ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.h
>index 1599bae7862714a7bb9d342ce4a3208026434632..c33bd0285b6111941540eea1f47e5f9c89e14042 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.h
>+++ b/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.h
>@@ -41,6 +41,8 @@ typedef struct _WebKitMediaStreamSrc WebKitMediaStreamSrc;
> #define WEBKIT_TYPE_MEDIA_STREAM_SRC (webkit_media_stream_src_get_type())
> GType webkit_media_stream_src_get_type(void) G_GNUC_CONST;
> bool webkitMediaStreamSrcSetStream(WebKitMediaStreamSrc*, MediaStreamPrivate*);
>+bool webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc*, MediaStreamTrackPrivate*, bool onlyTrack);
>+GstElement * webkitMediaStreamSrcNew(void);
> } // WebCore
> 
> #endif // ENABLE(VIDEO) && ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp b/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp
>index fd52ebf496e7b1e697696b85daac61576a061adf..04899ade4b37d38aa1cdcfa19597868af1878b92 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp
>+++ b/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp
>@@ -24,16 +24,110 @@
> #if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
> #include "MockGStreamerAudioCaptureSource.h"
> 
>+#include "GStreamerAudioStreamDescription.h"
> #include "MockRealtimeAudioSource.h"
> 
>+#include <gst/app/gstappsrc.h>
>+
> namespace WebCore {
> 
>+static const double Tau = 2 * M_PI;
>+static const double BipBopDuration = 0.07;
>+static const double BipBopVolume = 0.5;
>+static const double BipFrequency = 1500;
>+static const double BopFrequency = 500;
>+static const double HumFrequency = 150;
>+static const double HumVolume = 0.1;
>+
> class WrappedMockRealtimeAudioSource : public MockRealtimeAudioSource {
> public:
>     WrappedMockRealtimeAudioSource(String&& deviceID, String&& name, String&& hashSalt)
>         : MockRealtimeAudioSource(WTFMove(deviceID), WTFMove(name), WTFMove(hashSalt))
>+        , m_src(nullptr)
>+    {
>+    }
>+
>+    void start(GRefPtr<GstElement> src)
>+    {
>+        m_src = src;
>+        if (m_streamFormat)
>+            gst_app_src_set_caps(GST_APP_SRC(m_src.get()), m_streamFormat->caps());
>+        MockRealtimeAudioSource::start();
>+    }
>+
>+    void addHum(float amplitude, float frequency, float sampleRate, uint64_t start, float *p, uint64_t count)
>+    {
>+        float humPeriod = sampleRate / frequency;
>+        for (uint64_t i = start, end = start + count; i < end; ++i) {
>+            float a = amplitude * sin(i * Tau / humPeriod);
>+            a += *p;
>+            *p++ = a;
>+        }
>+    }
>+
>+    void render(Seconds delta)
>     {
>+        ASSERT(m_src);
>+
>+        uint32_t totalFrameCount = GST_ROUND_UP_16(static_cast<size_t>(delta.seconds() * sampleRate()));
>+        uint32_t frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
>+        while (frameCount) {
>+            uint32_t bipBopStart = m_samplesRendered % m_bipBopBuffer.size();
>+            uint32_t bipBopRemain = m_bipBopBuffer.size() - bipBopStart;
>+            uint32_t bipBopCount = std::min(frameCount, bipBopRemain);
>+
>+            GstBuffer* buffer = gst_buffer_new_allocate(nullptr, bipBopCount * m_streamFormat->bytesPerFrame(), nullptr);
>+            {
>+                GstMappedBuffer map(buffer, GST_MAP_WRITE);
>+
>+                if (!muted()) {
>+                    memcpy(map.data(), &m_bipBopBuffer[bipBopStart], sizeof(float) * bipBopCount);
>+                    addHum(HumVolume, HumFrequency, sampleRate(), m_samplesRendered, (float*)map.data(), bipBopCount);
>+                } else
>+                    memset(map.data(), 0, sizeof(float) * bipBopCount);
>+            }
>+
>+            gst_app_src_push_buffer(GST_APP_SRC(m_src.get()), buffer);
>+            m_samplesRendered += bipBopCount;
>+            totalFrameCount -= bipBopCount;
>+            frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
>+        }
>     }
>+
>+    void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag> settings)
>+    {
>+        if (settings.contains(RealtimeMediaSourceSettings::Flag::SampleRate)) {
>+            GstAudioInfo info;
>+            auto rate = sampleRate();
>+            size_t sampleCount = 2 * rate;
>+
>+            m_maximiumFrameCount = WTF::roundUpToPowerOfTwo(renderInterval().seconds() * sampleRate());
>+            gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_F32LE, rate, 1, nullptr);
>+            m_streamFormat = GStreamerAudioStreamDescription(info);
>+
>+            if (m_src)
>+                gst_app_src_set_caps(GST_APP_SRC(m_src.get()), m_streamFormat->caps());
>+
>+            m_bipBopBuffer.grow(sampleCount);
>+            m_bipBopBuffer.fill(0);
>+
>+            size_t bipBopSampleCount = ceil(BipBopDuration * rate);
>+            size_t bipStart = 0;
>+            size_t bopStart = rate;
>+
>+            addHum(BipBopVolume, BipFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bipStart), bipBopSampleCount);
>+            addHum(BipBopVolume, BopFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bopStart), bipBopSampleCount);
>+        }
>+
>+        MockRealtimeAudioSource::settingsDidChange(settings);
>+    }
>+
>+    GRefPtr<GstElement> m_src;
>+    std::optional<GStreamerAudioStreamDescription> m_streamFormat;
>+    Vector<float> m_bipBopBuffer;
>+    uint32_t m_maximiumFrameCount;
>+    uint64_t m_samplesEmitted { 0 };
>+    uint64_t m_samplesRendered { 0 };
> };
> 
> CaptureSourceOrError MockRealtimeAudioSource::create(String&& deviceID,
>@@ -80,7 +174,7 @@ void MockGStreamerAudioCaptureSource::stopProducingData()
> void MockGStreamerAudioCaptureSource::startProducingData()
> {
>     GStreamerAudioCaptureSource::startProducingData();
>-    m_wrappedSource->start();
>+    static_cast<WrappedMockRealtimeAudioSource*>(m_wrappedSource.get())->start(capturer()->source());
> }
> 
> const RealtimeMediaSourceSettings& MockGStreamerAudioCaptureSource::settings()
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp
>index 534af7701037211273f983e00f4d4b021adc161d..5e7f977a96b2179db618fe800b2d5635db9d6a1f 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp
>+++ b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp
>@@ -108,33 +108,26 @@ void RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData()
>     size_t inChunkSampleCount = gst_audio_converter_get_in_frames(m_sampleConverter.get(), outChunkSampleCount);
>     size_t inBufferSize = inChunkSampleCount * m_inputStreamDescription->getInfo()->bpf;
> 
>-    auto available = gst_adapter_available(m_adapter.get());
>-    if (inBufferSize > available) {
>-        GST_DEBUG("Not enough data: wanted: %ld > %ld available",
>-            inBufferSize, available);
>-
>-        return;
>-    }
>-
>-    auto inBuffer = adoptGRef(gst_adapter_take_buffer(m_adapter.get(), inBufferSize));
>-    auto outBuffer = adoptGRef(gst_buffer_new_allocate(nullptr, outBufferSize, 0));
>-    GstMappedBuffer outMap(outBuffer.get(), GST_MAP_WRITE);
>-    if (isSilenced())
>-        gst_audio_format_fill_silence(m_outputStreamDescription->getInfo()->finfo, outMap.data(), outMap.size());
>-    else {
>-        GstMappedBuffer inMap(inBuffer.get(), GST_MAP_READ);
>-
>-        gpointer in[1] = { inMap.data() };
>-        gpointer out[1] = { outMap.data() };
>-        if (!gst_audio_converter_samples(m_sampleConverter.get(), static_cast<GstAudioConverterFlags>(0), in, inChunkSampleCount, out, outChunkSampleCount)) {
>-            GST_ERROR("Could not convert samples.");
>-
>-            return;
>+    while (gst_adapter_available(m_adapter.get()) > inBufferSize) {
>+        auto inBuffer = adoptGRef(gst_adapter_take_buffer(m_adapter.get(), inBufferSize));
>+        m_audioBuffer.grow(outBufferSize);
>+        if (isSilenced())
>+            gst_audio_format_fill_silence(m_outputStreamDescription->getInfo()->finfo, m_audioBuffer.data(), outBufferSize);
>+        else {
>+            GstMappedBuffer inMap(inBuffer.get(), GST_MAP_READ);
>+
>+            gpointer in[1] = { inMap.data() };
>+            gpointer out[1] = { m_audioBuffer.data() };
>+            if (!gst_audio_converter_samples(m_sampleConverter.get(), static_cast<GstAudioConverterFlags>(0), in, inChunkSampleCount, out, outChunkSampleCount)) {
>+                GST_ERROR("Could not convert samples.");
>+
>+                return;
>+            }
>         }
>-    }
> 
>-    sendAudioFrames(outMap.data(), LibWebRTCAudioFormat::sampleSize, static_cast<int>(m_outputStreamDescription->sampleRate()),
>-        static_cast<int>(m_outputStreamDescription->numberOfChannels()), outChunkSampleCount);
>+        sendAudioFrames(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, static_cast<int>(m_outputStreamDescription->sampleRate()),
>+            static_cast<int>(m_outputStreamDescription->numberOfChannels()), outChunkSampleCount);
>+    }
> }
> 
> bool RealtimeOutgoingAudioSourceLibWebRTC::isReachingBufferedAudioDataHighLimit()
>diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h
>index 4609e9d9bad25f6e92844241f8f1555564087fc0..090405e1848facf8a87f08f565b18b747becb162 100644
>--- a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h
>+++ b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h
>@@ -54,6 +54,7 @@ private:
> 
>     Lock m_adapterMutex;
>     GRefPtr<GstAdapter> m_adapter;
>+    Vector<uint8_t> m_audioBuffer;
> };
> 
> } // namespace WebCore
>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index 7257553e71bfa3a1436fa80c98be72c58f810a34..b5b8be49ab7d634d7af3f07722e93eb920611d9c 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,15 @@
>+2018-12-05  Thibault Saunier  <tsaunier@igalia.com>
>+
>+        [WPE][GTK] Implement WebAudioSourceProviderGStreamer to allow bridging MediaStream and the WebAudio APIs
>+        https://bugs.webkit.org/show_bug.cgi?id=186933
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Enabled all tests depending on that feature.
>+
>+        * platform/gtk/TestExpectations:
>+        * webrtc/clone-audio-track.html:
>+
> 2018-11-28  Said Abou-Hallawa  <sabouhallawa@apple.com>
> 
>         Updating href on linearGradient and radialGradient doesn't update its rendering
>diff --git a/LayoutTests/platform/gtk/TestExpectations b/LayoutTests/platform/gtk/TestExpectations
>index 4f6e27d2820b6d89f97b0ca5ce95684bec024b1b..7feb17c2aef4cd943a3304580879e178dc625496 100644
>--- a/LayoutTests/platform/gtk/TestExpectations
>+++ b/LayoutTests/platform/gtk/TestExpectations
>@@ -586,16 +586,6 @@ webkit.org/b/187064 webrtc/libwebrtc/descriptionGetters.html
> webkit.org/b/177533 webrtc/video-interruption.html
> 
> webkit.org/b/186933 webrtc/peer-connection-createMediaStreamDestination.html
>-webkit.org/b/186933 webrtc/peer-connection-remote-audio-mute2.html
>-webkit.org/b/186933 webrtc/peer-connection-audio-unmute.html
>-webkit.org/b/186933 webrtc/peer-connection-audio-mute2.html
>-webkit.org/b/186933 webrtc/peer-connection-audio-mute.html
>-webkit.org/b/186933 webrtc/peer-connection-remote-audio-mute.html
>-webkit.org/b/186933 webrtc/clone-audio-track.html
>-webkit.org/b/186933 webrtc/audio-replace-track.html
>-webkit.org/b/186933 webrtc/audio-peer-connection-webaudio.html
>-webkit.org/b/186933 webrtc/audio-muted-stats.html
>-webkit.org/b/186933 webrtc/getUserMedia-webaudio-autoplay.html
> 
> imported/w3c/web-platform-tests/webrtc/ [ Skip ]
> http/tests/webrtc [ Skip ]
>diff --git a/LayoutTests/webrtc/clone-audio-track.html b/LayoutTests/webrtc/clone-audio-track.html
>index 649b446e8584253b513ab7293a090c15f0dbf543..0f9374849083f4ade02766574a153b582d91bbdb 100644
>--- a/LayoutTests/webrtc/clone-audio-track.html
>+++ b/LayoutTests/webrtc/clone-audio-track.html
>@@ -33,7 +33,7 @@
>                 });
>             }).then(() => {
>                 return analyseAudio(remoteStream, 200, context).then((results) => {
>-                    assert_false(results.heardHum, "Did not heard hum from remote enabled track");
>+                    assert_false(results.heardHum, "Did not hear hum from remote disabled track");
>                 });
>             }).then(() => {
>                 return analyseAudio(new MediaStream([clonedTrack]), 200, context).then((results) => {

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Attachments on bug 186933: 356726 | 356735