Diff
Modified: trunk/ChangeLog (271398 => 271399)
--- trunk/ChangeLog 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/ChangeLog 2021-01-12 12:33:25 UTC (rev 271399)
@@ -1,3 +1,12 @@
+2021-01-12 Philippe Normand <[email protected]>
+
+ [GStreamer] Bump version requirement
+ https://bugs.webkit.org/show_bug.cgi?id=220356
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ * Source/cmake/GStreamerChecks.cmake: Bump required version to 1.14.
+
2021-01-12 Xabier Rodriguez Calvar <[email protected]>
[GStreamer] Switch from ENABLE_ to USE_ in native audio/video and text sink options
Modified: trunk/Source/WebCore/ChangeLog (271398 => 271399)
--- trunk/Source/WebCore/ChangeLog 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/ChangeLog 2021-01-12 12:33:25 UTC (rev 271399)
@@ -1,3 +1,46 @@
+2021-01-12 Philippe Normand <[email protected]>
+
+ [GStreamer] Bump version requirement
+ https://bugs.webkit.org/show_bug.cgi?id=220356
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ Remove compile-time and runtime GStreamer version checks that are
+ not needed anymore since we now require at least GStreamer 1.14 at
+ configure time.
+
+ * platform/GStreamer.cmake:
+ * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+ (_WebKitWebAudioSrcPrivate::_WebKitWebAudioSrcPrivate):
+ (webKitWebAudioSrcRenderAndPushFrames):
+ (webKitWebAudioSrcChangeState):
+ * platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp:
+ (requestGLContext):
+ (webKitGLVideoSinkChangeState):
+ * platform/graphics/gstreamer/GStreamerAudioMixer.cpp:
+ (WebCore::GStreamerAudioMixer::ensureState):
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
+ (WebCore::MediaPlayerPrivateGStreamer::wouldTaintOrigin const):
+ (WebCore::MediaPlayerPrivateGStreamer::setPlaybinURL):
+ (WebCore::MediaPlayerPrivateGStreamer::handleMessage):
+ (WebCore::MediaPlayerPrivateGStreamer::createGSTPlayBin):
+ (WebCore::convertToInternalProtocol): Deleted.
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h:
+ * platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp:
+ (PlatformDisplay::tryEnsureGstGLContext const):
+ * platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp:
+ (webKitAudioSinkChangeState):
+ * platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp:
+ (webKitWebSrcCreate):
+ (webKitWebSrcGetProtocols):
+ (convertPlaybinURI):
+ * platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h:
+ (WebCore::InitData::payloadContainerType const):
+ * platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp:
+ (transformCaps):
+ * platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
+ (webkitMediaStreamSrcChangeState):
+
2021-01-12 Xabier Rodriguez Calvar <[email protected]>
[GStreamer] Switch from ENABLE_ to USE_ in native audio/video and text sink options
Modified: trunk/Source/WebCore/platform/GStreamer.cmake (271398 => 271399)
--- trunk/Source/WebCore/platform/GStreamer.cmake 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/GStreamer.cmake 2021-01-12 12:33:25 UTC (rev 271399)
@@ -141,17 +141,13 @@
endif ()
if (ENABLE_MEDIA_STREAM OR ENABLE_WEB_RTC)
- 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}
+ list(APPEND WebCore_SYSTEM_INCLUDE_DIRECTORIES
+ ${GSTREAMER_CODECPARSERS_INCLUDE_DIRS}
+ )
+ if (NOT USE_GSTREAMER_FULL)
+ list(APPEND WebCore_LIBRARIES
+ ${GSTREAMER_CODECPARSERS_LIBRARIES}
)
- if (NOT USE_GSTREAMER_FULL)
- list(APPEND WebCore_LIBRARIES
- ${GSTREAMER_CODECPARSERS_LIBRARIES}
- )
- endif ()
endif ()
endif ()
endif ()
Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -78,7 +78,6 @@
GRefPtr<GstBufferPool> pool;
- bool enableGapBufferSupport;
bool hasRenderedAudibleFrame { false };
Lock dispatchToRenderThreadLock;
@@ -93,11 +92,6 @@
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()
@@ -370,7 +364,7 @@
GST_BUFFER_TIMESTAMP(buffer.get()) = outputTimestamp.position.nanoseconds();
GST_BUFFER_DURATION(buffer.get()) = duration;
- if (priv->enableGapBufferSupport && priv->bus->channel(i)->isSilent())
+ if (priv->bus->channel(i)->isSilent())
GST_BUFFER_FLAG_SET(buffer.get(), GST_BUFFER_FLAG_GAP);
if (failed)
@@ -430,9 +424,7 @@
auto* src = ""
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:
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/GLVideoSinkGStreamer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -133,11 +133,7 @@
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);
}
@@ -158,15 +154,11 @@
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;
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerAudioMixer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -57,9 +57,8 @@
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);
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -153,14 +153,6 @@
player->handleMessage(message);
}
-static void convertToInternalProtocol(URL& url)
-{
- if (webkitGstCheckVersion(1, 12, 0))
- return;
- if (url.protocolIsInHTTPFamily() || url.protocolIsBlob())
- url.setProtocol(makeString("webkit+", url.protocol()));
-}
-
static void initializeDebugCategory()
{
static std::once_flag onceFlag;
@@ -848,22 +840,15 @@
Optional<bool> MediaPlayerPrivateGStreamer::wouldTaintOrigin(const SecurityOrigin& origin) const
{
- if (webkitGstCheckVersion(1, 12, 0)) {
- GST_TRACE_OBJECT(pipeline(), "Checking %u origins", m_origins.size());
- for (auto& responseOrigin : m_origins) {
- if (!origin.canAccess(*responseOrigin)) {
- GST_DEBUG_OBJECT(pipeline(), "Found reachable response origin");
- return true;
- }
+ GST_TRACE_OBJECT(pipeline(), "Checking %u origins", m_origins.size());
+ for (auto& responseOrigin : m_origins) {
+ if (!origin.canAccess(*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;
+ GST_DEBUG_OBJECT(pipeline(), "No valid response origin found");
+ return false;
}
void MediaPlayerPrivateGStreamer::simulateAudioInterruption()
@@ -983,7 +968,6 @@
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);
}
@@ -1973,7 +1957,6 @@
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) {
@@ -2012,11 +1995,6 @@
} 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();
@@ -2816,11 +2794,7 @@
m_textAppSinkPad = adoptGRef(gst_element_get_static_pad(m_textAppSink.get(), "sink"));
ASSERT(m_textAppSinkPad);
- GRefPtr<GstCaps> textCaps;
- if (webkitGstCheckVersion(1, 14, 0))
- textCaps = adoptGRef(gst_caps_new_empty_simple("application/x-subtitle-vtt"));
- else
- textCaps = adoptGRef(gst_caps_new_empty_simple("text/vtt"));
+ auto textCaps = adoptGRef(gst_caps_new_empty_simple("application/x-subtitle-vtt"));
g_object_set(m_textAppSink.get(), "emit-signals", TRUE, "enable-last-sample", FALSE, "caps", textCaps.get(), nullptr);
g_signal_connect_swapped(m_textAppSink.get(), "new-sample", G_CALLBACK(newTextSampleCallback), this);
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2021-01-12 12:33:25 UTC (rev 271399)
@@ -51,16 +51,6 @@
#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
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/PlatformDisplayGStreamer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -98,21 +98,13 @@
if (!contextHandle)
return false;
- bool shouldAdoptRef = webkitGstCheckVersion(1, 14, 0);
-
- if (shouldAdoptRef)
- m_gstGLDisplay = adoptGRef(createGstGLDisplay(*this));
- else
- m_gstGLDisplay = createGstGLDisplay(*this);
+ m_gstGLDisplay = adoptGRef(createGstGLDisplay(*this));
if (!m_gstGLDisplay)
return false;
GstGLPlatform glPlatform = sharedContext->isEGLContext() ? GST_GL_PLATFORM_EGL : GST_GL_PLATFORM_GLX;
- 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);
+ m_gstGLContext = adoptGRef(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();
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/WebKitAudioSinkGStreamer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -256,9 +256,7 @@
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)
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -465,12 +465,8 @@
// 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. 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) {
+ // 2) A GstContext query made here.
+ if (!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())) {
@@ -863,15 +859,9 @@
const gchar* const* webKitWebSrcGetProtocols(GType)
{
static const char* protocols[4];
- 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[0] = "http";
+ protocols[1] = "https";
+ protocols[2] = "blob";
protocols[3] = nullptr;
return protocols;
}
@@ -879,10 +869,6 @@
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;
}
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/eme/GStreamerEMEUtilities.h 2021-01-12 12:33:25 UTC (rev 271399)
@@ -79,7 +79,7 @@
const String& systemId() const { return m_systemId; }
String payloadContainerType() const
{
-#if GST_CHECK_VERSION(1, 15, 0)
+#if GST_CHECK_VERSION(1, 16, 0)
if (m_systemId == GST_PROTECTION_UNSPECIFIED_SYSTEM_ID)
return "webm"_s;
#endif
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/eme/WebKitCommonEncryptionDecryptorGStreamer.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -125,7 +125,7 @@
// 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, 15, 0)
+#if GST_CHECK_VERSION(1, 16, 0)
!g_strcmp0(klass->protectionSystemId(self), GST_PROTECTION_UNSPECIFIED_SYSTEM_ID) ? "application/x-webm-enc" :
#endif
"application/x-cenc");
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp (271398 => 271399)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2021-01-12 12:33:25 UTC (rev 271399)
@@ -401,9 +401,7 @@
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));
Modified: trunk/Source/cmake/GStreamerChecks.cmake (271398 => 271399)
--- trunk/Source/cmake/GStreamerChecks.cmake 2021-01-12 12:10:08 UTC (rev 271398)
+++ trunk/Source/cmake/GStreamerChecks.cmake 2021-01-12 12:33:25 UTC (rev 271399)
@@ -28,7 +28,7 @@
list(APPEND GSTREAMER_COMPONENTS audio fft)
endif ()
- find_package(GStreamer 1.10.0 REQUIRED COMPONENTS ${GSTREAMER_COMPONENTS})
+ find_package(GStreamer 1.14.0 REQUIRED COMPONENTS ${GSTREAMER_COMPONENTS})
if (ENABLE_WEB_AUDIO)
if (NOT PC_GSTREAMER_AUDIO_FOUND OR NOT PC_GSTREAMER_FFT_FOUND)
@@ -52,14 +52,7 @@
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 ()
