File webkit2gtk3-old-gstreamer.patch of Package webkit2gtk3

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File webkit2gtk3-old-gstreamer.patch of Package webkit2gtk3 (Revision f98564207ddb9d84e1e510cea3b26949)

Currently displaying revision f98564207ddb9d84e1e510cea3b26949 , Show latest

diff -urp webkitgtk-2.38.2.orig/Source/cmake/GStreamerChecks.cmake webkitgtk-2.38.2.gstreamer/Source/cmake/GStreamerChecks.cmake
--- webkitgtk-2.38.2.orig/Source/cmake/GStreamerChecks.cmake	2022-08-31 02:59:57.338525800 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/cmake/GStreamerChecks.cmake	2022-11-07 16:23:45.651052138 -0600
@@ -36,7 +36,7 @@ if (ENABLE_VIDEO OR ENABLE_WEB_AUDIO)
               list(APPEND GSTREAMER_COMPONENTS webrtc)
           endif ()
 
-          find_package(GStreamer 1.14.0 REQUIRED COMPONENTS ${GSTREAMER_COMPONENTS})
+          find_package(GStreamer 1.10.0 REQUIRED COMPONENTS ${GSTREAMER_COMPONENTS})
 
           if (ENABLE_WEB_AUDIO)
               if (NOT PC_GSTREAMER_AUDIO_FOUND OR NOT PC_GSTREAMER_FFT_FOUND)
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp	2022-08-31 02:59:56.166519000 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -79,6 +79,7 @@ struct _WebKitWebAudioSrcPrivate {
 
     GRefPtr<GstBufferPool> pool;
 
+    bool enableGapBufferSupport;
     bool hasRenderedAudibleFrame { false };
 
     Lock dispatchToRenderThreadLock;
@@ -93,6 +94,11 @@ struct _WebKitWebAudioSrcPrivate {
         sourcePad = webkitGstGhostPadFromStaticTemplate(&srcTemplate, "src", nullptr);
 
         g_rec_mutex_init(&mutex);
+
+        // GAP buffer support is enabled only for GStreamer 1.12.5 because of a
+        // memory leak that was fixed in that version.
+        // https://bugzilla.gnome.org/show_bug.cgi?id=793067
+        enableGapBufferSupport = webkitGstCheckVersion(1, 12, 5);
     }
 
     ~_WebKitWebAudioSrcPrivate()
@@ -378,7 +384,7 @@ static void webKitWebAudioSrcRenderAndPu
         GST_BUFFER_TIMESTAMP(buffer.get()) = outputTimestamp.position.nanoseconds();
         GST_BUFFER_DURATION(buffer.get()) = duration;
 
-        if (priv->bus->channel(i)->isSilent())
+        if (priv->enableGapBufferSupport && priv->bus->channel(i)->isSilent())
             GST_BUFFER_FLAG_SET(buffer.get(), GST_BUFFER_FLAG_GAP);
 
         if (failed)
@@ -437,7 +443,9 @@ static GstStateChangeReturn webKitWebAud
     auto* src = WEBKIT_WEB_AUDIO_SRC(element);
     auto* priv = src->priv;
 
+#if GST_CHECK_VERSION(1, 14, 0)
     GST_DEBUG_OBJECT(element, "%s", gst_state_change_get_name(transition));
+#endif
 
     switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp	2022-08-31 02:59:56.270519500 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -159,7 +159,11 @@ std::optional<GRefPtr<GstContext>> reque
     if (!g_strcmp0(contextType, "gst.gl.app_context")) {
         GRefPtr<GstContext> appContext = adoptGRef(gst_context_new("gst.gl.app_context", TRUE));
         GstStructure* structure = gst_context_writable_structure(appContext.get());
+#if GST_CHECK_VERSION(1, 12, 0)
         gst_structure_set(structure, "context", GST_TYPE_GL_CONTEXT, gstGLContext, nullptr);
+#else
+        gst_structure_set(structure, "context", GST_GL_TYPE_CONTEXT, gstGLContext, nullptr);
+#endif
         return appContext;
     }
 
@@ -180,11 +184,15 @@ static bool setGLContext(GstElement* ele
 
 static GstStateChangeReturn webKitGLVideoSinkChangeState(GstElement* element, GstStateChange transition)
 {
+#if GST_CHECK_VERSION(1, 14, 0)
     GST_DEBUG_OBJECT(element, "%s", gst_state_change_get_name(transition));
+#endif
 
     switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
+#if GST_CHECK_VERSION(1, 14, 0)
     case GST_STATE_CHANGE_READY_TO_READY:
+#endif
     case GST_STATE_CHANGE_READY_TO_PAUSED: {
         if (!setGLContext(element, GST_GL_DISPLAY_CONTEXT_TYPE))
             return GST_STATE_CHANGE_FAILURE;
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp	2022-08-31 02:59:56.270519500 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -57,8 +57,9 @@ GStreamerAudioMixer::GStreamerAudioMixer
 
 void GStreamerAudioMixer::ensureState(GstStateChange stateChange)
 {
+#if GST_CHECK_VERSION(1, 14, 0)
     GST_DEBUG_OBJECT(m_pipeline.get(), "Handling %s transition (%u mixer pads)", gst_state_change_get_name(stateChange), m_mixer->numsinkpads);
-
+#endif
     switch (stateChange) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
         gst_element_set_state(m_pipeline.get(), GST_STATE_PAUSED);
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp	2022-10-19 03:11:34.526284500 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -42,6 +42,7 @@
 #include <wtf/Scope.h>
 #include <wtf/glib/GUniquePtr.h>
 #include <wtf/glib/RunLoopSourcePriority.h>
+#include <wtf/PrintStream.h>
 
 #if USE(GSTREAMER_MPEGTS)
 #define GST_USE_UNSTABLE_API
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp	2022-11-03 04:40:39.032952500 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp	2022-11-08 13:45:25.869062043 -0600
@@ -51,7 +51,6 @@
 #include "InbandTextTrackPrivateGStreamer.h"
 #include "TextCombinerGStreamer.h"
 #include "TextSinkGStreamer.h"
-#include "VideoFrameMetadataGStreamer.h"
 #include "VideoTrackPrivateGStreamer.h"
 
 #if ENABLE(MEDIA_STREAM)
@@ -148,6 +147,14 @@ using namespace std;
 static const FloatSize s_holePunchDefaultFrameSize(1280, 720);
 #endif
 
+static void convertToInternalProtocol(URL& url)
+{
+    if (webkitGstCheckVersion(1, 12, 0))
+        return;
+    if (url.protocolIsInHTTPFamily() || url.protocolIsBlob())
+        url.setProtocol(makeString("webkit+", url.protocol()));
+}
+
 static void initializeDebugCategory()
 {
     static std::once_flag onceFlag;
@@ -828,15 +835,20 @@ bool MediaPlayerPrivateGStreamer::hasSin
 
 std::optional<bool> MediaPlayerPrivateGStreamer::wouldTaintOrigin(const SecurityOrigin& origin) const
 {
-    GST_TRACE_OBJECT(pipeline(), "Checking %u origins", m_origins.size());
-    for (auto& responseOrigin : m_origins) {
-        if (!origin.isSameOriginDomain(*responseOrigin)) {
-            GST_DEBUG_OBJECT(pipeline(), "Found reachable response origin");
-            return true;
+    if (webkitGstCheckVersion(1, 12, 0)) {
+        GST_TRACE_OBJECT(pipeline(), "Checking %u origins", m_origins.size());
+        for (auto& responseOrigin : m_origins) {
+            if (!origin.isSameOriginDomain(*responseOrigin)) {
+                GST_DEBUG_OBJECT(pipeline(), "Found reachable response origin");
+                return true;
+            }
         }
     }
-    GST_DEBUG_OBJECT(pipeline(), "No valid response origin found");
-    return false;
+
+    // GStreamer < 1.12 has an incomplete uridownloader implementation so we
+    // can't use WebKitWebSrc for adaptive fragments downloading if this
+    // version is detected.
+    return m_hasTaintedOrigin;
 }
 
 void MediaPlayerPrivateGStreamer::simulateAudioInterruption()
@@ -934,6 +946,7 @@ void MediaPlayerPrivateGStreamer::setPla
         cleanURLString = cleanURLString.left(url.pathEnd());
 
     m_url = URL { cleanURLString };
+    convertToInternalProtocol(m_url);
     GST_INFO_OBJECT(pipeline(), "Load %s", m_url.string().utf8().data());
     g_object_set(m_pipeline.get(), "uri", m_url.string().utf8().data(), nullptr);
 }
@@ -1915,6 +1928,7 @@ void MediaPlayerPrivateGStreamer::handle
             GST_DEBUG_OBJECT(pipeline(), "Processing HTTP headers: %" GST_PTR_FORMAT, structure);
             if (const char* uri = gst_structure_get_string(structure, "uri")) {
                 URL url { String::fromLatin1(uri) };
+                convertToInternalProtocol(m_url);
                 m_origins.add(SecurityOrigin::create(url));
 
                 if (url != m_url) {
@@ -1953,6 +1967,11 @@ void MediaPlayerPrivateGStreamer::handle
         } else if (gst_structure_has_name(structure, "webkit-network-statistics")) {
             if (gst_structure_get(structure, "read-position", G_TYPE_UINT64, &m_networkReadPosition, "size", G_TYPE_UINT64, &m_httpResponseTotalSize, nullptr))
                 GST_DEBUG_OBJECT(pipeline(), "Updated network read position %" G_GUINT64_FORMAT ", size: %" G_GUINT64_FORMAT, m_networkReadPosition, m_httpResponseTotalSize);
+        } else if (gst_structure_has_name(structure, "adaptive-streaming-statistics")) {
+            if (WEBKIT_IS_WEB_SRC(m_source.get()) && !webkitGstCheckVersion(1, 12, 0)) {
+                if (const char* uri = gst_structure_get_string(structure, "uri"))
+                    m_hasTaintedOrigin = webKitSrcWouldTaintOrigin(WEBKIT_WEB_SRC_CAST(m_source.get()), SecurityOrigin::create(URL(URL(), String::fromLatin1(uri))));
+            }
         } else if (gst_structure_has_name(structure, "GstCacheDownloadComplete")) {
             GST_INFO_OBJECT(pipeline(), "Stream is fully downloaded, stopping monitoring downloading progress.");
             m_fillTimer.stop();
@@ -2163,6 +2182,7 @@ void MediaPlayerPrivateGStreamer::proces
         processTableOfContentsEntry(static_cast<GstTocEntry*>(i->data));
 }
 
+#if 0
 void MediaPlayerPrivateGStreamer::configureElement(GstElement* element)
 {
     GUniquePtr<char> elementName(gst_element_get_name(element));
@@ -2198,6 +2218,7 @@ void MediaPlayerPrivateGStreamer::config
     if (!g_strcmp0(G_OBJECT_TYPE_NAME(G_OBJECT(element)), "GstQueue2"))
         g_object_set(G_OBJECT(element), "high-watermark", 0.10, nullptr);
 }
+#endif
 
 void MediaPlayerPrivateGStreamer::configureDownloadBuffer(GstElement* element)
 {
@@ -2803,8 +2824,34 @@ void MediaPlayerPrivateGStreamer::create
 
     g_object_set(m_pipeline.get(), "mute", m_player->muted(), nullptr);
 
-    g_signal_connect(GST_BIN_CAST(m_pipeline.get()), "element-setup", G_CALLBACK(+[](GstBin*, GstElement* element, MediaPlayerPrivateGStreamer* player) {
-        player->configureElement(element);
+    g_signal_connect(GST_BIN_CAST(m_pipeline.get()), "deep-element-added", G_CALLBACK(+[](GstBin*, GstBin* subBin, GstElement* element, MediaPlayerPrivateGStreamer* player) {
+        GUniquePtr<char> binName(gst_element_get_name(GST_ELEMENT_CAST(subBin)));
+        GUniquePtr<char> elementName(gst_element_get_name(element));
+
+        if (g_str_has_prefix(elementName.get(), "downloadbuffer")) {
+            player->configureDownloadBuffer(element);
+            return;
+        }
+
+        if (g_str_has_prefix(elementName.get(), "uridecodebin")) {
+            // This will set the multiqueue size to the default value.
+            g_object_set(element, "buffer-size", 2 * MB, nullptr);
+            return;
+        }
+
+        if (!g_str_has_prefix(binName.get(), "decodebin"))
+            return;
+
+        if (g_str_has_prefix(elementName.get(), "v4l2"))
+            player->m_videoDecoderPlatform = GstVideoDecoderPlatform::Video4Linux;
+        else if (g_str_has_prefix(elementName.get(), "imxvpudec"))
+            player->m_videoDecoderPlatform = GstVideoDecoderPlatform::ImxVPU;
+        else if (g_str_has_prefix(elementName.get(), "omx"))
+            player->m_videoDecoderPlatform = GstVideoDecoderPlatform::OpenMAX;
+
+#if USE(TEXTURE_MAPPER_GL)
+        player->updateTextureMapperFlags();
+#endif
     }), this);
 
     g_signal_connect_swapped(m_pipeline.get(), "source-setup", G_CALLBACK(sourceSetupCallback), this);
@@ -2859,6 +2906,7 @@ void MediaPlayerPrivateGStreamer::config
         g_object_set(depayloader, "wait-for-keyframe", TRUE, nullptr);
 }
 
+#if 0
 void MediaPlayerPrivateGStreamer::configureVideoDecoder(GstElement* decoder)
 {
     auto decoderHasProperty = [&decoder](const char* name) -> bool {
@@ -2895,6 +2943,7 @@ void MediaPlayerPrivateGStreamer::config
     if (decoderHasProperty("max-errors"))
         g_object_set(decoder, "max-errors", -1, nullptr);
 }
+#endif
 
 bool MediaPlayerPrivateGStreamer::didPassCORSAccessCheck() const
 {
@@ -3007,8 +3056,6 @@ void MediaPlayerPrivateGStreamer::pushTe
     if (!GST_IS_SAMPLE(m_sample.get()))
         return;
 
-    ++m_sampleCount;
-
     auto internalCompositingOperation = [this](TextureMapperPlatformLayerProxyGL& proxy, std::unique_ptr<GstVideoFrameHolder>&& frameHolder) {
         std::unique_ptr<TextureMapperPlatformLayerBuffer> layerBuffer;
         if (frameHolder->hasMappedTextures()) {
@@ -3576,7 +3623,7 @@ void MediaPlayerPrivateGStreamer::flushC
 {
     Locker sampleLocker { m_sampleMutex };
 
-    if (m_sample && gst_sample_get_buffer(m_sample.get())) {
+    if (m_sample) {
         // Allocate a new copy of the sample which has to be released. The copy is necessary so that
         // the video dimensions can still be fetched and also for canvas rendering. The release is
         // necessary because the sample might have been allocated by a hardware decoder and memory
@@ -4181,31 +4228,6 @@ WTFLogChannel& MediaPlayerPrivateGStream
 }
 #endif
 
-std::optional<VideoFrameMetadata> MediaPlayerPrivateGStreamer::videoFrameMetadata()
-{
-    if (m_sampleCount == m_lastVideoFrameMetadataSampleCount)
-        return { };
-
-    m_lastVideoFrameMetadataSampleCount = m_sampleCount;
-
-    Locker sampleLocker { m_sampleMutex };
-    if (!GST_IS_SAMPLE(m_sample.get()))
-        return { };
-
-    auto* buffer = gst_sample_get_buffer(m_sample.get());
-    auto metadata = webkitGstBufferGetVideoFrameMetadata(buffer);
-    auto size = naturalSize();
-    metadata.width = size.width();
-    metadata.height = size.height();
-    metadata.presentedFrames = m_sampleCount;
-
-    // FIXME: presentationTime and expectedDisplayTime might not always have the same value, we should try getting more precise values.
-    metadata.presentationTime = MonotonicTime::now().secondsSinceEpoch().seconds();
-    metadata.expectedDisplayTime = metadata.presentationTime;
-
-    return metadata;
-}
-
 }
 
 #endif // USE(GSTREAMER)
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h	2022-11-03 04:40:39.032952500 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h	2022-11-07 16:45:14.997859264 -0600
@@ -54,6 +54,16 @@ typedef struct _GstMpegtsSection GstMpeg
 #if USE(LIBEPOXY)
 // Include the <epoxy/gl.h> header before <gst/gl/gl.h>.
 #include <epoxy/gl.h>
+
+// Workaround build issue with RPi userland GLESv2 headers and libepoxy <https://webkit.org/b/185639>
+#if !GST_CHECK_VERSION(1, 14, 0)
+#include <gst/gl/gstglconfig.h>
+#if defined(GST_GL_HAVE_WINDOW_DISPMANX) && GST_GL_HAVE_WINDOW_DISPMANX
+#define __gl2_h_
+#undef GST_GL_HAVE_GLSYNC
+#define GST_GL_HAVE_GLSYNC 1
+#endif
+#endif // !GST_CHECK_VERSION(1, 14, 0)
 #endif // USE(LIBEPOXY)
 
 #define GST_USE_UNSTABLE_API
@@ -488,7 +498,6 @@ private:
     static void downloadBufferFileCreatedCallback(MediaPlayerPrivateGStreamer*);
 
     void configureDepayloader(GstElement*);
-    void configureVideoDecoder(GstElement*);
     void configureElement(GstElement*);
 
     void setPlaybinURL(const URL& urlString);
@@ -590,9 +599,6 @@ private:
     DataMutex<TaskAtMediaTimeScheduler> m_TaskAtMediaTimeSchedulerDataMutex;
 
 private:
-    std::optional<VideoFrameMetadata> videoFrameMetadata() final;
-    uint64_t m_sampleCount { 0 };
-    uint64_t m_lastVideoFrameMetadataSampleCount { 0 };
 #if USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)
     GUniquePtr<struct wpe_video_plane_display_dmabuf_source> m_wpeVideoPlaneDisplayDmaBuf;
 #endif
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp	2022-08-31 02:59:56.274519400 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -98,13 +98,21 @@ bool PlatformDisplay::tryEnsureGstGLCont
     if (!contextHandle)
         return false;
 
-    m_gstGLDisplay = adoptGRef(createGstGLDisplay(*this));
+    bool shouldAdoptRef = webkitGstCheckVersion(1, 14, 0);
+
+    if (shouldAdoptRef)
+        m_gstGLDisplay = adoptGRef(createGstGLDisplay(*this));
+    else
+        m_gstGLDisplay = createGstGLDisplay(*this);
     if (!m_gstGLDisplay)
         return false;
 
     GstGLPlatform glPlatform = sharedContext->isEGLContext() ? GST_GL_PLATFORM_EGL : GST_GL_PLATFORM_GLX;
 
-    m_gstGLContext = adoptGRef(gst_gl_context_new_wrapped(m_gstGLDisplay.get(), reinterpret_cast<guintptr>(contextHandle), glPlatform, glAPI));
+    if (shouldAdoptRef)
+        m_gstGLContext = adoptGRef(gst_gl_context_new_wrapped(m_gstGLDisplay.get(), reinterpret_cast<guintptr>(contextHandle), glPlatform, glAPI));
+    else
+        m_gstGLContext = gst_gl_context_new_wrapped(m_gstGLDisplay.get(), reinterpret_cast<guintptr>(contextHandle), glPlatform, glAPI);
 
     // Activate and fill the GStreamer wrapped context with the Webkit's shared one.
     auto* previousActiveContext = GLContext::current();
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp	2022-08-31 02:59:56.274519400 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -256,7 +256,9 @@ static GstStateChangeReturn webKitAudioS
     auto* sink = WEBKIT_AUDIO_SINK(element);
     auto* priv = sink->priv;
 
+#if GST_CHECK_VERSION(1, 14, 0)
     GST_DEBUG_OBJECT(sink, "Handling %s transition", gst_state_change_get_name(stateChange));
+#endif
 
     auto& mixer = GStreamerAudioMixer::singleton();
     if (priv->interAudioSink && stateChange == GST_STATE_CHANGE_NULL_TO_READY)
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp	2022-08-31 02:59:56.274519400 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -467,8 +467,12 @@ static GstFlowReturn webKitWebSrcCreate(
     // 1) webKitWebSrcSetMediaPlayer() is called by MediaPlayerPrivateGStreamer by means of hooking playbin's
     //    "source-setup" event. This doesn't work for additional WebKitWebSrc elements created by adaptivedemux.
     //
-    // 2) A GstContext query made here.
-    if (!members->player) {
+    // 2) A GstContext query made here. Because of a bug, this only works in GStreamer >= 1.12.
+    //
+    // As a compatibility workaround, the http: URI protocol is only registered for gst>=1.12; otherwise using
+    // webkit+http:, which is used by MediaPlayerPrivateGStreamer but not by adaptivedemux's additional source
+    // elements, therefore using souphttpsrc instead and not routing traffic through the NetworkProcess.
+    if (webkitGstCheckVersion(1, 12, 0) && !members->player) {
         members.runUnlocked([src, baseSrc]() {
             GRefPtr<GstQuery> query = adoptGRef(gst_query_new_context(WEBKIT_WEB_SRC_PLAYER_CONTEXT_TYPE_NAME));
             if (gst_pad_peer_query(GST_BASE_SRC_PAD(baseSrc), query.get())) {
@@ -872,16 +876,27 @@ static GstURIType webKitWebSrcUriGetType
 const gchar* const* webKitWebSrcGetProtocols(GType)
 {
     static const char* protocols[4];
-    protocols[0] = "http";
-    protocols[1] = "https";
-    protocols[2] = "blob";
+    if (webkitGstCheckVersion(1, 12, 0)) {
+        protocols[0] = "http";
+        protocols[1] = "https";
+        protocols[2] = "blob";
+    } else {
+        protocols[0] = "webkit+http";
+        protocols[1] = "webkit+https";
+        protocols[2] = "webkit+blob";
+    }
     protocols[3] = nullptr;
     return protocols;
 }
 
 static URL convertPlaybinURI(const char* uriString)
 {
-    return URL { String::fromLatin1(uriString) };
+    URL url { String::fromLatin1(uriString) };
+    if (!webkitGstCheckVersion(1, 12, 0)) {
+        ASSERT(url.protocol().substring(0, 7) == "webkit+");
+        url.setProtocol(url.protocol().substring(7).toString());
+    }
+     return url;
 }
 
 static gchar* webKitWebSrcGetUri(GstURIHandler* handler)
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp	2022-09-14 06:58:10.508863200 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -23,14 +23,11 @@
 #include "config.h"
 
 #if ENABLE(VIDEO) && ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
-
 #include "GStreamerCapturer.h"
-#include "VideoFrameMetadataGStreamer.h"
 
 #include <gst/app/gstappsink.h>
 #include <gst/app/gstappsrc.h>
 #include <mutex>
-#include <wtf/MonotonicTime.h>
 
 GST_DEBUG_CATEGORY(webkit_capturer_debug);
 #define GST_CAT_DEFAULT webkit_capturer_debug
@@ -102,20 +99,9 @@ GstElement* GStreamerCapturer::createSou
         if (GST_IS_APP_SRC(m_src.get()))
             g_object_set(m_src.get(), "is-live", true, "format", GST_FORMAT_TIME, "do-timestamp", true, nullptr);
 
-        auto srcPad = adoptGRef(gst_element_get_static_pad(m_src.get(), "src"));
-        if (m_deviceType == CaptureDevice::DeviceType::Camera) {
-            gst_pad_add_probe(srcPad.get(), static_cast<GstPadProbeType>(GST_PAD_PROBE_TYPE_PUSH | GST_PAD_PROBE_TYPE_BUFFER), [](GstPad*, GstPadProbeInfo* info, gpointer) -> GstPadProbeReturn {
-                VideoFrameTimeMetadata metadata;
-                metadata.captureTime = MonotonicTime::now().secondsSinceEpoch();
-                auto* buffer = GST_PAD_PROBE_INFO_BUFFER(info);
-                auto* modifiedBuffer = webkitGstBufferSetVideoFrameTimeMetadata(buffer, metadata);
-                gst_buffer_replace(&buffer, modifiedBuffer);
-                return GST_PAD_PROBE_OK;
-            }, nullptr, nullptr);
-        }
-
         if (m_deviceType == CaptureDevice::DeviceType::Screen) {
-            gst_pad_add_probe(srcPad.get(), GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, [](GstPad*, GstPadProbeInfo* info, void* userData) -> GstPadProbeReturn {
++            auto pad = adoptGRef(gst_element_get_static_pad(m_src.get(), "src"));
+            gst_pad_add_probe(pad.get(), GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, [](GstPad*, GstPadProbeInfo* info, void* userData) -> GstPadProbeReturn {
                 auto* event = gst_pad_probe_info_get_event(info);
                 if (GST_EVENT_TYPE(event) != GST_EVENT_CAPS)
                     return GST_PAD_PROBE_OK;
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp	2022-11-03 04:36:42.779971000 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp	2022-11-07 16:23:45.655052159 -0600
@@ -31,7 +31,6 @@
 #include "GStreamerCommon.h"
 #include "MediaStreamPrivate.h"
 #include "VideoFrameGStreamer.h"
-#include "VideoFrameMetadataGStreamer.h"
 #include "VideoTrackPrivateMediaStream.h"
 
 #if USE(GSTREAMER_WEBRTC)
@@ -68,10 +67,8 @@ GRefPtr<GstTagList> mediaStreamTrackPriv
         gst_tag_list_add(tagList.get(), GST_TAG_MERGE_APPEND, WEBKIT_MEDIA_TRACK_TAG_KIND, static_cast<int>(VideoTrackPrivate::Kind::Main), nullptr);
 
         auto& settings = track->settings();
-        if (settings.width())
-            gst_tag_list_add(tagList.get(), GST_TAG_MERGE_APPEND, WEBKIT_MEDIA_TRACK_TAG_WIDTH, settings.width(), nullptr);
-        if (settings.height())
-            gst_tag_list_add(tagList.get(), GST_TAG_MERGE_APPEND, WEBKIT_MEDIA_TRACK_TAG_HEIGHT, settings.height(), nullptr);
+        gst_tag_list_add(tagList.get(), GST_TAG_MERGE_APPEND, WEBKIT_MEDIA_TRACK_TAG_WIDTH, settings.width(),
+            WEBKIT_MEDIA_TRACK_TAG_HEIGHT, settings.height(), nullptr);
     }
 
     GST_DEBUG("Track tags: %" GST_PTR_FORMAT, tagList.get());
@@ -324,17 +321,18 @@ public:
         if (!m_parent || !m_isObserving)
             return;
 
-        auto videoFrameSize = videoFrame.presentationSize();
-        IntSize captureSize(videoFrameSize.width(), videoFrameSize.height());
+        auto* gstSample = static_cast<MediaSampleGStreamer*>(&sample)->platformSample().sample.gstSample;
+        auto* caps = gst_sample_get_caps(gstSample);
+        GstVideoInfo info;
+        gst_video_info_from_caps(&info, caps);
 
-        auto settings = m_track.settings();
-        m_configuredSize.setWidth(settings.width());
-        m_configuredSize.setHeight(settings.height());
-
-        if (!m_configuredSize.width())
-            m_configuredSize.setWidth(captureSize.width());
-        if (!m_configuredSize.height())
-            m_configuredSize.setHeight(captureSize.height());
+        int width = GST_VIDEO_INFO_WIDTH(&info);
+        int height = GST_VIDEO_INFO_HEIGHT(&info);
+        if (m_lastKnownSize != IntSize(width, height)) {
+            m_lastKnownSize.setWidth(width);
+            m_lastKnownSize.setHeight(height);
+            updateBlackFrame(caps);
+        }
 
         auto videoRotation = videoFrame.rotation();
         bool videoMirrored = videoFrame.isMirrored();
@@ -348,13 +346,6 @@ public:
             gst_pad_push_event(pad.get(), gst_event_new_tag(gst_tag_list_new(GST_TAG_IMAGE_ORIENTATION, orientation.utf8().data(), nullptr)));
         }
 
-        auto* gstSample = static_cast<VideoFrameGStreamer*>(&videoFrame)->sample();
-        if (!m_configuredSize.isEmpty() && m_lastKnownSize != m_configuredSize) {
-            GST_DEBUG_OBJECT(m_src.get(), "Video size changed from %dx%d to %dx%d", m_lastKnownSize.width(), m_lastKnownSize.height(), m_configuredSize.width(), m_configuredSize.height());
-            m_lastKnownSize = m_configuredSize;
-            updateBlackFrame();
-        }
-
         if (!m_blackFrame)
             updateBlackFrame();
 
@@ -442,13 +433,6 @@ private:
     void pushBlackFrame()
     {
         GST_TRACE_OBJECT(m_src.get(), "Pushing black video frame");
-        VideoFrameTimeMetadata metadata;
-        metadata.captureTime = MonotonicTime::now().secondsSinceEpoch();
-        auto* buffer = webkitGstBufferSetVideoFrameTimeMetadata(gst_sample_get_buffer(m_blackFrame.get()), metadata);
-        GST_BUFFER_DTS(buffer) = GST_BUFFER_PTS(buffer) = gst_element_get_current_running_time(m_parent);
-        // TODO: Use gst_sample_set_buffer() after bumping GStreamer dependency to 1.16.
-        auto* caps = gst_sample_get_caps(m_blackFrame.get());
-        m_blackFrame = adoptGRef(gst_sample_new(buffer, caps, nullptr, nullptr));
         pushSample(m_blackFrame.get());
     }
 
@@ -463,7 +447,6 @@ private:
     bool m_isObserving { false };
     RefPtr<AudioTrackPrivateMediaStream> m_audioTrack;
     RefPtr<VideoTrackPrivateMediaStream> m_videoTrack;
-    IntSize m_configuredSize;
     IntSize m_lastKnownSize;
     GRefPtr<GstSample> m_blackFrame;
     GRefPtr<GstSample> m_silentSample;
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp	2022-08-31 02:59:56.358520000 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp	2022-11-07 16:23:45.659052181 -0600
@@ -64,7 +64,7 @@ void RealtimeIncomingVideoSourceLibWebRT
         videoFrameAvailable(VideoFrameGStreamer::createWrappedSample(framebuffer->getSample(), presentationTime, static_cast<VideoFrame::Rotation>(frame.rotation())), { });
     } else {
         auto gstSample = convertLibWebRTCVideoFrameToGStreamerSample(frame);
-        auto metadata = std::make_optional(metadataFromVideoFrame(frame));
+        auto sample = MediaSampleGStreamer::create(WTFMove(gstSample), { }, { });
         videoFrameAvailable(VideoFrameGStreamer::create(WTFMove(gstSample), { }, presentationTime, static_cast<VideoFrame::Rotation>(frame.rotation()), false, WTFMove(metadata)), { });
     }
 }
diff -urp webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h
--- webkitgtk-2.38.2.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h	2022-08-31 02:59:56.358520000 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h	2022-11-07 16:23:45.659052181 -0600
@@ -45,6 +45,8 @@ private:
 
     // rtc::VideoSinkInterface
     void OnFrame(const webrtc::VideoFrame&) final;
+    void setCapsFromSettings();
+    GRefPtr<GstCaps> m_caps;
 };
 
 } // namespace WebCore
diff -urp webkitgtk-2.38.2.orig/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml webkitgtk-2.38.2.gstreamer/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml
--- webkitgtk-2.38.2.orig/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml	2022-09-14 06:58:10.404862600 -0500
+++ webkitgtk-2.38.2.gstreamer/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml	2022-11-07 16:23:45.659052181 -0600
@@ -1274,11 +1274,9 @@ RequestVideoFrameCallbackEnabled:
       default: false
     WebKit:
       "PLATFORM(COCOA) && HAVE(AVSAMPLEBUFFERVIDEOOUTPUT)" : true
-      "USE(GSTREAMER)": true
       default: false
     WebCore:
       "PLATFORM(COCOA) && HAVE(AVSAMPLEBUFFERVIDEOOUTPUT)" : true
-      "USE(GSTREAMER)": true
       default: false
 
 ResizeObserverEnabled:

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File webkit2gtk3-old-gstreamer.patch of Package webkit2gtk3 (Revision f98564207ddb9d84e1e510cea3b26949)

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