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openSUSE:Step:15-SP4
webkit2gtk3.20650
webkit2gtk3-old-gstreamer.patch
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File webkit2gtk3-old-gstreamer.patch of Package webkit2gtk3.20650
diff -urp webkitgtk-2.32.3.orig/Source/cmake/GStreamerChecks.cmake webkitgtk-2.32.3/Source/cmake/GStreamerChecks.cmake --- webkitgtk-2.32.3.orig/Source/cmake/GStreamerChecks.cmake 2021-02-26 03:57:17.000000000 -0600 +++ webkitgtk-2.32.3/Source/cmake/GStreamerChecks.cmake 2021-07-26 13:05:20.805378339 -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-2.32.3.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp webkitgtk-2.32.3/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2021-05-10 02:59:16.000000000 -0500 +++ webkitgtk-2.32.3/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2021-07-26 13:05:20.805378339 -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() @@ -375,7 +381,7 @@ 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) @@ -429,7 +435,9 @@ 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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2021-02-26 03:57:13.000000000 -0600 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2021-07-26 13:05:20.809378360 -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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2021-05-05 00:33:24.000000000 -0500 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2021-07-26 13:05:20.809378360 -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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2021-05-10 02:59:16.000000000 -0500 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2021-07-26 13:05:20.809378360 -0500 @@ -168,7 +168,11 @@ 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); } @@ -189,11 +193,15 @@ 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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2021-02-26 03:57:13.000000000 -0600 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2021-07-26 13:05:20.809378360 -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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2021-07-23 02:47:17.000000000 -0500 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2021-07-26 13:05:20.817378403 -0500 @@ -134,6 +134,14 @@ 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; @@ -823,15 +831,20 @@ 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() @@ -951,6 +964,7 @@ 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); } @@ -1911,6 +1925,7 @@ 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) { @@ -1949,6 +1964,11 @@ 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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2021-05-05 00:33:24.000000000 -0500 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2021-07-26 13:05:20.817378403 -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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2021-02-26 03:57:13.000000000 -0600 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2021-07-26 13:05:20.821378424 -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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2021-02-26 03:57:13.000000000 -0600 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2021-07-26 13:05:20.821378424 -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-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2021-03-25 10:14:07.000000000 -0500 +++ webkitgtk-2.32.3/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2021-07-26 13:05:20.821378424 -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-2.32.3.orig/Source/WebCore/platform/GStreamer.cmake webkitgtk-2.32.3/Source/WebCore/platform/GStreamer.cmake --- webkitgtk-2.32.3.orig/Source/WebCore/platform/GStreamer.cmake 2021-02-26 03:57:13.000000000 -0600 +++ webkitgtk-2.32.3/Source/WebCore/platform/GStreamer.cmake 2021-07-26 13:05:20.821378424 -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-2.32.3.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp webkitgtk-2.32.3/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp --- webkitgtk-2.32.3.orig/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2021-05-05 00:33:24.000000000 -0500 +++ webkitgtk-2.32.3/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2021-07-26 13:05:20.825378446 -0500 @@ -399,7 +399,9 @@ 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));
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