File webkit2gtk3-old-gstreamer.patch of Package webkit2gtk3.19288
diff -urp webkitgtk-2.32.0.orig/Source/cmake/GStreamerChecks.cmake webkitgtk-2.32.0/Source/cmake/GStreamerChecks.cmake
--- webkitgtk-2.32.0.orig/Source/cmake/GStreamerChecks.cmake 2021-02-26 03:57:17.000000000 -0600
+++ webkitgtk-2.32.0/Source/cmake/GStreamerChecks.cmake 2021-04-12 13:55:46.974577043 -0500
@@ -28,7 +28,7 @@ if (ENABLE_VIDEO OR ENABLE_WEB_AUDIO)
list(APPEND GSTREAMER_COMPONENTS audio fft)
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)
@@ -52,7 +52,14 @@ if (ENABLE_VIDEO OR ENABLE_WEB_AUDIO)
endif ()
endif ()
+if (ENABLE_MEDIA_SOURCE AND PC_GSTREAMER_VERSION VERSION_LESS "1.14")
+ message(FATAL_ERROR "GStreamer 1.14 is needed for ENABLE_MEDIA_SOURCE.")
+endif ()
+
if (ENABLE_MEDIA_STREAM AND ENABLE_WEB_RTC)
+ if (PC_GSTREAMER_VERSION VERSION_LESS "1.12")
+ message(FATAL_ERROR "GStreamer 1.12 is needed for ENABLE_WEB_RTC.")
+ endif ()
SET_AND_EXPOSE_TO_BUILD(USE_LIBWEBRTC TRUE)
SET_AND_EXPOSE_TO_BUILD(WEBRTC_WEBKIT_BUILD TRUE)
else ()
diff -urp webkitgtk-2.32.0.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp webkitgtk-2.32.0/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2021-04-12 13:55:46.970577023 -0500
@@ -78,6 +78,7 @@ struct _WebKitWebAudioSrcPrivate {
GRefPtr<GstBufferPool> pool;
+ bool enableGapBufferSupport;
bool hasRenderedAudibleFrame { false };
Lock dispatchToRenderThreadLock;
@@ -92,6 +93,11 @@ 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()
@@ -364,7 +370,7 @@ 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)
@@ -424,7 +430,9 @@ 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-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2021-04-12 13:55:46.974577043 -0500
@@ -79,7 +79,7 @@ public:
const String& systemId() const { return m_systemId; }
String payloadContainerType() const
{
-#if GST_CHECK_VERSION(1, 16, 0)
+#if GST_CHECK_VERSION(1, 15, 0)
if (m_systemId == GST_PROTECTION_UNSPECIFIED_SYSTEM_ID)
return "webm"_s;
#endif
diff -urp webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2021-04-12 13:55:46.974577043 -0500
@@ -125,7 +125,7 @@ static GstCaps* transformCaps(GstBaseTra
// GST_PROTECTION_UNSPECIFIED_SYSTEM_ID was added in the GStreamer
// developement git master which will ship as version 1.16.0.
gst_structure_set_name(outgoingStructure.get(),
-#if GST_CHECK_VERSION(1, 16, 0)
+#if GST_CHECK_VERSION(1, 15, 0)
!g_strcmp0(klass->protectionSystemId(self), GST_PROTECTION_UNSPECIFIED_SYSTEM_ID) ? "application/x-webm-enc" :
#endif
"application/x-cenc");
diff -urp webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2021-03-25 08:33:45.000000000 -0500
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2021-04-12 13:55:46.970577023 -0500
@@ -152,7 +152,11 @@ Optional<GRefPtr<GstContext>> requestGLC
if (!g_strcmp0(contextType, "gst.gl.app_context")) {
GstContext* appContext = gst_context_new("gst.gl.app_context", TRUE);
GstStructure* structure = gst_context_writable_structure(appContext);
+#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 adoptGRef(appContext);
}
@@ -173,11 +177,15 @@ 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-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2021-04-12 13:55:46.970577023 -0500
@@ -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-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2021-04-12 14:10:55.707374702 -0500
@@ -153,6 +153,14 @@ static void busMessageCallback(GstBus*,
player->handleMessage(message);
}
+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;
@@ -841,15 +849,20 @@ bool MediaPlayerPrivateGStreamer::hasSin
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()
@@ -969,6 +982,7 @@ void MediaPlayerPrivateGStreamer::setPla
cleanURLString = cleanURLString.substring(0, url.pathEnd());
m_url = 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);
}
@@ -1936,6 +1950,7 @@ 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(URL(), uri);
+ convertToInternalProtocol(url);
m_origins.add(SecurityOrigin::create(url));
if (url != m_url) {
@@ -1974,6 +1989,11 @@ 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(), uri)));
+ }
} else if (gst_structure_has_name(structure, "GstCacheDownloadComplete")) {
GST_INFO_OBJECT(pipeline(), "Stream is fully downloaded, stopping monitoring downloading progress.");
m_fillTimer.stop();
diff -urp webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2021-04-12 13:55:46.974577043 -0500
@@ -51,6 +51,16 @@ 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
diff -urp webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2021-04-12 13:55:46.974577043 -0500
@@ -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-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2021-04-12 13:55:46.974577043 -0500
@@ -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-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2021-03-25 10:14:07.000000000 -0500
+++ webkitgtk-2.32.0/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2021-04-12 13:55:46.974577043 -0500
@@ -465,8 +465,12 @@ 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())) {
@@ -859,9 +863,15 @@ 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;
}
@@ -869,6 +879,10 @@ const gchar* const* webKitWebSrcGetProto
static URL convertPlaybinURI(const char* uriString)
{
URL url(URL(), uriString);
+ if (!webkitGstCheckVersion(1, 12, 0)) {
+ ASSERT(url.protocol().substring(0, 7) == "webkit+");
+ url.setProtocol(url.protocol().substring(7).toString());
+ }
return url;
}
diff -urp webkitgtk-2.32.0.orig/Source/WebCore/platform/GStreamer.cmake webkitgtk-2.32.0/Source/WebCore/platform/GStreamer.cmake
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/GStreamer.cmake 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/GStreamer.cmake 2021-04-12 13:55:46.970577023 -0500
@@ -142,13 +142,17 @@ if (ENABLE_VIDEO)
endif ()
if (ENABLE_MEDIA_STREAM OR ENABLE_WEB_RTC)
- list(APPEND WebCore_SYSTEM_INCLUDE_DIRECTORIES
- ${GSTREAMER_CODECPARSERS_INCLUDE_DIRS}
- )
- if (NOT USE_GSTREAMER_FULL)
- list(APPEND WebCore_LIBRARIES
- ${GSTREAMER_CODECPARSERS_LIBRARIES}
+ if (PC_GSTREAMER_VERSION VERSION_LESS "1.10")
+ message(FATAL_ERROR "GStreamer 1.10 is needed for ENABLE_MEDIA_STREAM or ENABLE_WEB_RTC")
+ else ()
+ list(APPEND WebCore_SYSTEM_INCLUDE_DIRECTORIES
+ ${GSTREAMER_CODECPARSERS_INCLUDE_DIRS}
)
+ if (NOT USE_GSTREAMER_FULL)
+ list(APPEND WebCore_LIBRARIES
+ ${GSTREAMER_CODECPARSERS_LIBRARIES}
+ )
+ endif ()
endif ()
endif ()
endif ()
diff -urp webkitgtk-2.32.0.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp webkitgtk-2.32.0/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
--- webkitgtk-2.32.0.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2021-02-26 03:57:13.000000000 -0600
+++ webkitgtk-2.32.0/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2021-04-12 13:55:46.974577043 -0500
@@ -401,7 +401,9 @@ static void webkitMediaStreamSrcDispose(
static GstStateChangeReturn webkitMediaStreamSrcChangeState(GstElement* element, GstStateChange transition)
{
+#if GST_CHECK_VERSION(1, 14, 0)
GST_DEBUG_OBJECT(element, "%s", gst_state_change_get_name(transition));
+#endif
if (transition == GST_STATE_CHANGE_PAUSED_TO_READY)
stopObservingTracks(WEBKIT_MEDIA_STREAM_SRC(element));
