File webkit2gtk3-old-gstreamer.patch of Package webkit2gtk3.23800
diff -urp webkitgtk-2.36.0.orig/Source/cmake/GStreamerChecks.cmake webkitgtk-2.36.0.gstreamer/Source/cmake/GStreamerChecks.cmake
--- webkitgtk-2.36.0.orig/Source/cmake/GStreamerChecks.cmake 2022-02-23 02:59:06.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/cmake/GStreamerChecks.cmake 2022-03-22 13:18:04.341943346 -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)
else ()
SET_AND_EXPOSE_TO_BUILD(USE_LIBWEBRTC FALSE)
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2022-02-23 02:59:01.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2022-03-22 13:18:04.341943346 -0500
@@ -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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2022-03-22 13:18:04.341943346 -0500
@@ -78,7 +78,7 @@ 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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2022-03-22 13:18:04.341943346 -0500
@@ -148,7 +148,7 @@ 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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2022-03-22 13:18:04.341943346 -0500
@@ -160,7 +160,11 @@ std::optional<GRefPtr<GstContext>> reque
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);
}
@@ -181,11 +185,15 @@ 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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2022-03-22 13:18:04.341943346 -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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp 2022-03-21 14:19:36.303412337 -0500
@@ -39,6 +39,7 @@
#include <wtf/Scope.h>
#include <wtf/glib/GUniquePtr.h>
#include <wtf/glib/RunLoopSourcePriority.h>
+#include <wtf/PrintStream.h>
#if USE(GSTREAMER_FULL)
#include <gst/gstinitstaticplugins.h>
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h 2022-03-16 08:48:02.000000000 -0500
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h 2022-03-21 15:12:02.672022277 -0500
@@ -83,14 +83,6 @@ inline GstClockTime toGstClockTime(const
return static_cast<GstClockTime>(toGstUnsigned64Time(mediaTime));
}
-inline MediaTime fromGstClockTime(GstClockTime time)
-{
- if (!GST_CLOCK_TIME_IS_VALID(time))
- return MediaTime::invalidTime();
-
- return MediaTime(GST_TIME_AS_USECONDS(time), G_USEC_PER_SEC);
-}
-
class GstMappedBuffer {
WTF_MAKE_NONCOPYABLE(GstMappedBuffer);
public:
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2022-03-22 13:18:04.345943367 -0500
@@ -51,7 +51,6 @@
#include "InbandTextTrackPrivateGStreamer.h"
#include "TextCombinerGStreamer.h"
#include "TextSinkGStreamer.h"
-#include "VideoFrameMetadataGStreamer.h"
#include "VideoTrackPrivateGStreamer.h"
#if ENABLE(MEDIA_STREAM)
@@ -134,6 +133,14 @@ 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;
@@ -816,15 +823,20 @@ 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()
@@ -922,6 +934,7 @@ 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);
}
@@ -1863,6 +1876,7 @@ 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) {
@@ -1901,6 +1915,11 @@ 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();
@@ -2721,26 +2740,31 @@ void MediaPlayerPrivateGStreamer::create
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));
- auto elementClass = makeString(gst_element_get_metadata(element, GST_ELEMENT_METADATA_KLASS));
- auto classifiers = elementClass.split('/');
-
- // Collect processing time metrics for video decoders and converters.
- if ((classifiers.contains("Converter"_s) || classifiers.contains("Decoder"_s)) && classifiers.contains("Video"_s) && !classifiers.contains("Parser"))
- webkitGstTraceProcessingTimeForElement(element);
-
- if (classifiers.contains("Decoder"_s) && classifiers.contains("Video"_s)) {
- player->configureVideoDecoder(element);
- return;
- }
if (g_str_has_prefix(elementName.get(), "downloadbuffer")) {
player->configureDownloadBuffer(element);
return;
}
- // This will set the multiqueue size to the default value.
- if (g_str_has_prefix(elementName.get(), "uridecodebin"))
+ 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);
@@ -2784,27 +2808,6 @@ void MediaPlayerPrivateGStreamer::create
}), this);
}
-void MediaPlayerPrivateGStreamer::configureVideoDecoder(GstElement* decoder)
-{
- GUniquePtr<char> name(gst_element_get_name(decoder));
- if (g_str_has_prefix(name.get(), "v4l2"))
- m_videoDecoderPlatform = GstVideoDecoderPlatform::Video4Linux;
- else if (g_str_has_prefix(name.get(), "imxvpudec"))
- m_videoDecoderPlatform = GstVideoDecoderPlatform::ImxVPU;
- else if (g_str_has_prefix(name.get(), "omx"))
- m_videoDecoderPlatform = GstVideoDecoderPlatform::OpenMAX;
- else if (g_str_has_prefix(name.get(), "avdec")) {
- // Set the decoder maximum number of threads to a low, fixed value, not depending on the
- // platform. This also helps with processing metrics gathering. When using the default value
- // the decoder introduces artificial processing latency reflecting the maximum number of threads.
- g_object_set(decoder, "max-threads", 2, nullptr);
- }
-
-#if USE(TEXTURE_MAPPER_GL)
- updateTextureMapperFlags();
-#endif
-}
-
bool MediaPlayerPrivateGStreamer::didPassCORSAccessCheck() const
{
if (WEBKIT_IS_WEB_SRC(m_source.get()))
@@ -2916,8 +2919,6 @@ 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()) {
@@ -3222,7 +3223,7 @@ 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
@@ -3793,31 +3794,6 @@ 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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2022-03-22 13:18:04.345943367 -0500
@@ -53,6 +53,16 @@ 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
@@ -466,8 +476,6 @@ private:
void configureDownloadBuffer(GstElement*);
static void downloadBufferFileCreatedCallback(MediaPlayerPrivateGStreamer*);
- void configureVideoDecoder(GstElement*);
-
void setPlaybinURL(const URL& urlString);
void updateTracks(const GRefPtr<GstStreamCollection>&);
@@ -563,9 +571,6 @@ 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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp 2022-03-16 08:48:02.000000000 -0500
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.cpp 2022-03-22 14:08:37.241955857 -0500
@@ -24,7 +24,6 @@
#include "GStreamerCommon.h"
#include "PixelBuffer.h"
-#include "VideoFrameMetadataGStreamer.h"
#include <JavaScriptCore/JSCInlines.h>
#include <JavaScriptCore/TypedArrayInlines.h>
#include <algorithm>
@@ -33,7 +32,7 @@
namespace WebCore {
-MediaSampleGStreamer::MediaSampleGStreamer(GRefPtr<GstSample>&& sample, const FloatSize& presentationSize, const AtomString& trackId, VideoRotation videoRotation, bool videoMirrored, std::optional<VideoSampleMetadata>&& metadata)
+MediaSampleGStreamer::MediaSampleGStreamer(GRefPtr<GstSample>&& sample, const FloatSize& presentationSize, const AtomString& trackId, VideoRotation videoRotation, bool videoMirrored)
: m_pts(MediaTime::zeroTime())
, m_dts(MediaTime::zeroTime())
, m_duration(MediaTime::zeroTime())
@@ -46,9 +45,6 @@ MediaSampleGStreamer::MediaSampleGStream
GstBuffer* buffer = gst_sample_get_buffer(sample.get());
RELEASE_ASSERT(buffer);
- if (metadata)
- buffer = webkitGstBufferSetVideoSampleMetadata(buffer, WTFMove(metadata));
-
m_sample = sample;
initializeFromBuffer();
}
@@ -79,7 +75,7 @@ Ref<MediaSampleGStreamer> MediaSampleGSt
return adoptRef(*gstreamerMediaSample);
}
-Ref<MediaSampleGStreamer> MediaSampleGStreamer::createImageSample(PixelBuffer&& pixelBuffer, const IntSize& destinationSize, double frameRate, VideoRotation videoRotation, bool videoMirrored, std::optional<VideoSampleMetadata>&& metadata)
+Ref<MediaSampleGStreamer> MediaSampleGStreamer::createImageSample(PixelBuffer&& pixelBuffer, const IntSize& destinationSize, double frameRate, VideoRotation videoRotation, bool videoMirrored)
{
ensureGStreamerInitialized();
@@ -98,9 +94,6 @@ Ref<MediaSampleGStreamer> MediaSampleGSt
auto height = size.height();
gst_buffer_add_video_meta(buffer.get(), GST_VIDEO_FRAME_FLAG_NONE, GST_VIDEO_FORMAT_BGRA, width, height);
- if (metadata)
- webkitGstBufferSetVideoSampleMetadata(buffer.get(), *metadata);
-
int frameRateNumerator, frameRateDenominator;
gst_util_double_to_fraction(frameRate, &frameRateNumerator, &frameRateDenominator);
@@ -135,28 +128,33 @@ Ref<MediaSampleGStreamer> MediaSampleGSt
void MediaSampleGStreamer::initializeFromBuffer()
{
+ auto createMediaTime =
+ [](GstClockTime time) -> MediaTime {
+ return MediaTime(GST_TIME_AS_USECONDS(time), G_USEC_PER_SEC);
+ };
+
const GstClockTime minimumDuration = 1000; // 1 us
auto* buffer = gst_sample_get_buffer(m_sample.get());
RELEASE_ASSERT(buffer);
if (GST_BUFFER_PTS_IS_VALID(buffer))
- m_pts = fromGstClockTime(GST_BUFFER_PTS(buffer));
+ m_pts = createMediaTime(GST_BUFFER_PTS(buffer));
if (GST_BUFFER_DTS_IS_VALID(buffer) || GST_BUFFER_PTS_IS_VALID(buffer))
- m_dts = fromGstClockTime(GST_BUFFER_DTS_OR_PTS(buffer));
+ m_dts = createMediaTime(GST_BUFFER_DTS_OR_PTS(buffer));
if (GST_BUFFER_DURATION_IS_VALID(buffer)) {
// Sometimes (albeit rarely, so far seen only at the end of a track)
// frames have very small durations, so small that may be under the
// precision we are working with and be truncated to zero.
// SourceBuffer algorithms are not expecting frames with zero-duration,
// so let's use something very small instead in those fringe cases.
- m_duration = fromGstClockTime(std::max(GST_BUFFER_DURATION(buffer), minimumDuration));
+ m_duration = createMediaTime(std::max(GST_BUFFER_DURATION(buffer), minimumDuration));
} else {
// Unfortunately, sometimes samples don't provide a duration. This can never happen in MP4 because of the way
// the format is laid out, but it's pretty common in WebM.
// The good part is that durations don't matter for playback, just for buffered ranges and coded frame deletion.
// We want to pick something small enough to not cause unwanted frame deletion, but big enough to never be
// mistaken for a rounding artifact.
- m_duration = fromGstClockTime(16666667); // 1/60 seconds
+ m_duration = createMediaTime(16666667); // 1/60 seconds
}
m_size = gst_buffer_get_size(buffer);
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/MediaSampleGStreamer.h 2022-03-22 13:54:58.421632900 -0500
@@ -26,7 +26,6 @@
#include "FloatSize.h"
#include "GStreamerCommon.h"
#include "MediaSample.h"
-#include "VideoSampleMetadata.h"
#include <wtf/text/AtomString.h>
namespace WebCore {
@@ -35,9 +34,9 @@ class PixelBuffer;
class MediaSampleGStreamer : public MediaSample {
public:
- static Ref<MediaSampleGStreamer> create(GRefPtr<GstSample>&& sample, const FloatSize& presentationSize, const AtomString& trackId, VideoRotation videoRotation = VideoRotation::None, bool videoMirrored = false, std::optional<VideoSampleMetadata>&& metadata = std::nullopt)
+ static Ref<MediaSampleGStreamer> create(GRefPtr<GstSample>&& sample, const FloatSize& presentationSize, const AtomString& trackId, VideoRotation videoRotation = VideoRotation::None, bool videoMirrored = false)
{
- return adoptRef(*new MediaSampleGStreamer(WTFMove(sample), presentationSize, trackId, videoRotation, videoMirrored, WTFMove(metadata)));
+ return adoptRef(*new MediaSampleGStreamer(WTFMove(sample), presentationSize, trackId, videoRotation, videoMirrored));
}
static Ref<MediaSampleGStreamer> createWrappedSample(const GRefPtr<GstSample>& sample, VideoRotation videoRotation = VideoRotation::None)
@@ -46,7 +45,7 @@ public:
}
static Ref<MediaSampleGStreamer> createFakeSample(GstCaps*, MediaTime pts, MediaTime dts, MediaTime duration, const FloatSize& presentationSize, const AtomString& trackId);
- static Ref<MediaSampleGStreamer> createImageSample(PixelBuffer&&, const IntSize& destinationSize = { }, double frameRate = 1, VideoRotation videoRotation = VideoRotation::None, bool videoMirrored = false, std::optional<VideoSampleMetadata>&& metadata = std::nullopt);
+ static Ref<MediaSampleGStreamer> createImageSample(PixelBuffer&&, const IntSize& destinationSize = { }, double frameRate = 1, VideoRotation videoRotation = VideoRotation::None, bool videoMirrored = false);
void extendToTheBeginning();
MediaTime presentationTime() const override { return m_pts; }
@@ -70,7 +69,7 @@ public:
bool videoMirrored() const override { return m_videoMirrored; }
protected:
- MediaSampleGStreamer(GRefPtr<GstSample>&&, const FloatSize& presentationSize, const AtomString& trackId, VideoRotation = VideoRotation::None, bool videoMirrored = false, std::optional<VideoSampleMetadata>&& = std::nullopt);
+ MediaSampleGStreamer(GRefPtr<GstSample>&&, const FloatSize& presentationSize, const AtomString& trackId, VideoRotation = VideoRotation::None, bool videoMirrored = false);
MediaSampleGStreamer(const GRefPtr<GstSample>&, VideoRotation = VideoRotation::None);
virtual ~MediaSampleGStreamer() = default;
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2022-03-22 13:18:04.345943367 -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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2022-03-22 13:18:04.345943367 -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-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2022-03-22 13:18:04.345943367 -0500
@@ -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())) {
@@ -868,9 +872,15 @@ 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;
}
@@ -878,6 +888,10 @@ 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-2.36.0.orig/Source/WebCore/platform/GStreamer.cmake webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/GStreamer.cmake
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/GStreamer.cmake 2022-02-23 02:59:01.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/GStreamer.cmake 2022-03-22 13:18:04.345943367 -0500
@@ -24,7 +24,6 @@ if (ENABLE_VIDEO OR ENABLE_WEB_AUDIO)
platform/graphics/gstreamer/TextCombinerPadGStreamer.cpp
platform/graphics/gstreamer/TextSinkGStreamer.cpp
platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp
- platform/graphics/gstreamer/VideoFrameMetadataGStreamer.cpp
platform/graphics/gstreamer/VideoSinkGStreamer.cpp
platform/graphics/gstreamer/VideoTrackPrivateGStreamer.cpp
platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp
@@ -146,13 +145,17 @@ 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-2.36.0.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp 2022-03-21 15:12:02.676022299 -0500
@@ -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, 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 {
- VideoSampleMetadata metadata;
- metadata.captureTime = MonotonicTime::now().secondsSinceEpoch();
- auto* buffer = GST_PAD_PROBE_INFO_BUFFER(info);
- auto* modifiedBuffer = webkitGstBufferSetVideoSampleMetadata(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-2.36.0.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2022-03-21 15:12:02.676022299 -0500
@@ -31,7 +31,6 @@
#include "GStreamerCommon.h"
#include "MediaSampleGStreamer.h"
#include "MediaStreamPrivate.h"
-#include "VideoFrameMetadataGStreamer.h"
#include "VideoTrackPrivateMediaStream.h"
#include <gst/app/gstappsrc.h>
@@ -62,10 +61,8 @@ 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());
@@ -256,19 +253,19 @@ public:
if (!m_parent)
return;
- auto sampleSize = sample.presentationSize();
- IntSize captureSize(sampleSize.width(), sampleSize.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* mediaSample = static_cast<MediaSampleGStreamer*>(&sample);
auto videoRotation = sample.videoRotation();
bool videoMirrored = sample.videoMirrored();
if (m_videoRotation != videoRotation || m_videoMirrored != videoMirrored) {
@@ -281,12 +278,6 @@ 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 = mediaSample->platformSample().sample.gstSample;
- if (!m_configuredSize.isEmpty() && m_lastKnownSize != m_configuredSize) {
- m_lastKnownSize = m_configuredSize;
- updateBlackFrame(gst_sample_get_caps(gstSample));
- }
-
if (m_track.enabled()) {
GST_TRACE_OBJECT(m_src.get(), "Pushing video frame from enabled track");
pushSample(gstSample);
@@ -331,12 +322,6 @@ private:
void pushBlackFrame()
{
GST_TRACE_OBJECT(m_src.get(), "Pushing black video frame");
- VideoSampleMetadata metadata;
- metadata.captureTime = MonotonicTime::now().secondsSinceEpoch();
- auto* buffer = webkitGstBufferSetVideoSampleMetadata(gst_sample_get_buffer(m_blackFrame.get()), metadata);
- // 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());
}
@@ -350,7 +335,6 @@ private:
bool m_isObserving { false };
RefPtr<AudioTrackPrivateMediaStream> m_audioTrack;
RefPtr<VideoTrackPrivateMediaStream> m_videoTrack;
- IntSize m_configuredSize;
IntSize m_lastKnownSize;
GRefPtr<GstSample> m_blackFrame;
MediaSample::VideoRotation m_videoRotation { MediaSample::VideoRotation::None };
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeVideoSourceGStreamer.cpp 2022-03-21 15:12:02.676022299 -0500
@@ -162,9 +162,7 @@ void MockRealtimeVideoSourceGStreamer::u
if (!pixelBuffer)
return;
- std::optional<VideoSampleMetadata> metadata;
- metadata->captureTime = MonotonicTime::now().secondsSinceEpoch();
- auto sample = MediaSampleGStreamer::createImageSample(WTFMove(*pixelBuffer), size(), frameRate(), sampleRotation(), false, WTFMove(metadata));
+ auto sample = MediaSampleGStreamer::createImageSample(WTFMove(*pixelBuffer), size(), frameRate(), sampleRotation());
sample->offsetTimestampsBy(MediaTime::createWithDouble((elapsedTime() + 100_ms).seconds()));
dispatchMediaSampleToObservers(sample.get(), { });
}
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.cpp 2022-03-21 16:13:53.111610814 -0500
@@ -63,7 +63,7 @@ void RealtimeIncomingVideoSourceLibWebRT
videoSampleAvailable(MediaSampleGStreamer::createWrappedSample(framebuffer->getSample(), static_cast<MediaSample::VideoRotation>(frame.rotation())), { });
} else {
auto gstSample = convertLibWebRTCVideoFrameToGStreamerSample(frame);
- auto metadata = std::make_optional(metadataFromVideoFrame(frame));
+ auto sample = MediaSampleGStreamer::create(WTFMove(gstSample), { }, { });
videoSampleAvailable(MediaSampleGStreamer::create(WTFMove(gstSample), { }, { }, static_cast<MediaSample::VideoRotation>(frame.rotation()), false, WTFMove(metadata)), { });
}
}
diff -urp webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h
--- webkitgtk-2.36.0.orig/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h 2022-02-23 02:59:02.000000000 -0600
+++ webkitgtk-2.36.0.gstreamer/Source/WebCore/platform/mediastream/libwebrtc/gstreamer/RealtimeIncomingVideoSourceLibWebRTC.h 2022-03-21 15:12:02.676022299 -0500
@@ -45,6 +45,8 @@ private:
// rtc::VideoSinkInterface
void OnFrame(const webrtc::VideoFrame&) final;
+ void setCapsFromSettings();
+ GRefPtr<GstCaps> m_caps;
};
} // namespace WebCore
diff -urp webkitgtk-2.36.0.orig/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml webkitgtk-2.36.0.gstreamer/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml
--- webkitgtk-2.36.0.orig/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml 2022-03-16 08:48:02.000000000 -0500
+++ webkitgtk-2.36.0.gstreamer/Source/WTF/Scripts/Preferences/WebPreferencesExperimental.yaml 2022-03-21 15:12:02.672022277 -0500
@@ -1201,11 +1201,9 @@ 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
# FIXME: This is on by default in WebKit2. Perhaps we should consider turning it on for WebKitLegacy as well.
