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SUSE:SLE-15-SP1:GA
webkit2gtk3.24549
webkit2gtk3-old-gstreamer.patch
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File webkit2gtk3-old-gstreamer.patch of Package webkit2gtk3.24549
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.
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