Log Message
[GStreamer] webrtc/disable-encryption.html is a crashing flaky https://bugs.webkit.org/show_bug.cgi?id=211166
Patch by Philippe Normand <pnorm...@igalia.com> on 2020-05-14 Reviewed by Xabier Rodriguez-Calvar. Source/WebCore: Make sure the audio and video mediastream source elements are correctly removed and disposed from their parent bin when resetting the track sources. Before this change there was a possibility of disposing the elements while they were still in PLAYING state. * platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp: (WebCore::_WebKitMediaStreamSrc::SourceData::reset): (WebCore::webkitMediaStreamSrcDispose): (WebCore::webkitMediaStreamSrcRemoveTrackByType): LayoutTests: * platform/gtk/TestExpectations: Marking webrtc/disable-encryption.html as no longer flaky.
Modified Paths
Diff
Modified: trunk/LayoutTests/ChangeLog (261684 => 261685)
--- trunk/LayoutTests/ChangeLog 2020-05-14 10:30:33 UTC (rev 261684)
+++ trunk/LayoutTests/ChangeLog 2020-05-14 12:07:18 UTC (rev 261685)
@@ -1,3 +1,12 @@
+2020-05-14 Philippe Normand <pnorm...@igalia.com>
+
+ [GStreamer] webrtc/disable-encryption.html is a crashing flaky
+ https://bugs.webkit.org/show_bug.cgi?id=211166
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ * platform/gtk/TestExpectations: Marking webrtc/disable-encryption.html as no longer flaky.
+
2020-05-13 Myles C. Maxfield <mmaxfi...@apple.com>
fast/text/multiple-codeunit-vertical-upright.html is failing
Modified: trunk/LayoutTests/platform/gtk/TestExpectations (261684 => 261685)
--- trunk/LayoutTests/platform/gtk/TestExpectations 2020-05-14 10:30:33 UTC (rev 261684)
+++ trunk/LayoutTests/platform/gtk/TestExpectations 2020-05-14 12:07:18 UTC (rev 261685)
@@ -2901,8 +2901,6 @@
webkit.org/b/210541 imported/w3c/web-platform-tests/html/cross-origin-embedder-policy/require-corp-load-from-cache-storage.https.html [ Failure Pass ]
-webkit.org/b/211166 webrtc/disable-encryption.html [ Crash Pass ]
-
webkit.org/b/211614 imported/w3c/web-platform-tests/wasm/jsapi/constructor/instantiate.any.worker.html [ Failure Pass ]
webkit.org/b/211764 http/wpt/webrtc/generateCertificate.html [ Failure Pass ]
Modified: trunk/Source/WebCore/ChangeLog (261684 => 261685)
--- trunk/Source/WebCore/ChangeLog 2020-05-14 10:30:33 UTC (rev 261684)
+++ trunk/Source/WebCore/ChangeLog 2020-05-14 12:07:18 UTC (rev 261685)
@@ -1,3 +1,19 @@
+2020-05-14 Philippe Normand <pnorm...@igalia.com>
+
+ [GStreamer] webrtc/disable-encryption.html is a crashing flaky
+ https://bugs.webkit.org/show_bug.cgi?id=211166
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ Make sure the audio and video mediastream source elements are correctly removed and disposed
+ from their parent bin when resetting the track sources. Before this change there was a
+ possibility of disposing the elements while they were still in PLAYING state.
+
+ * platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
+ (WebCore::_WebKitMediaStreamSrc::SourceData::reset):
+ (WebCore::webkitMediaStreamSrcDispose):
+ (WebCore::webkitMediaStreamSrcRemoveTrackByType):
+
2020-05-14 Adrian Perez de Castro <ape...@igalia.com>
Non-unified build fixed, mid May 2020 edition
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp (261684 => 261685)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2020-05-14 10:30:33 UTC (rev 261684)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2020-05-14 12:07:18 UTC (rev 261685)
@@ -191,6 +191,16 @@
{
m_firstBufferPts = GST_CLOCK_TIME_NONE;
m_isVideo = isVideo;
+
+ if (!m_src)
+ return;
+
+ gst_element_set_locked_state(m_src.get(), true);
+ gst_element_set_state(m_src.get(), GST_STATE_NULL);
+ auto parent = adoptGRef(gst_object_get_parent(GST_OBJECT_CAST(m_src.get())));
+ if (parent)
+ gst_bin_remove(GST_BIN_CAST(parent.get()), m_src.get());
+ gst_element_set_locked_state(m_src.get(), false);
m_src = nullptr;
}
@@ -199,24 +209,25 @@
return m_src.get();
}
- static void needDataCb(GstElement*, guint, WebKitMediaStreamSrc::SourceData *self)
+ static void needDataCallback(WebKitMediaStreamSrc::SourceData* self, unsigned)
{
self->setEnoughData(false);
}
- static void enoughDataCb(GstElement*, WebKitMediaStreamSrc::SourceData *self)
+ static void enoughDataCallback(WebKitMediaStreamSrc::SourceData* self)
{
self->setEnoughData(true);
}
- void setSrc(GstElement *src)
+ void setSrc(GRefPtr<GstElement>&& src)
{
- m_src = adoptGRef(GST_ELEMENT(g_object_ref_sink(src)));
- if (GST_IS_APP_SRC(src)) {
- g_object_set(src, "is-live", true, "format", GST_FORMAT_TIME, "emit-signals", TRUE, "min-percent", 100, nullptr);
- g_signal_connect(src, "enough-data", G_CALLBACK(enoughDataCb), this);
- g_signal_connect(src, "need-data", G_CALLBACK(needDataCb), this);
- }
+ m_src = WTFMove(src);
+ if (!GST_IS_APP_SRC(m_src.get()))
+ return;
+
+ g_object_set(m_src.get(), "is-live", true, "format", GST_FORMAT_TIME, "emit-signals", true, "min-percent", 100, nullptr);
+ g_signal_connect_swapped(m_src.get(), "enough-data", G_CALLBACK(enoughDataCallback), this);
+ g_signal_connect_swapped(m_src.get(), "need-data", G_CALLBACK(needDataCallback), this);
}
bool isUsed()
@@ -369,15 +380,8 @@
{
WebKitMediaStreamSrc* self = WEBKIT_MEDIA_STREAM_SRC(object);
- if (self->audioSrc.isUsed()) {
- gst_bin_remove(GST_BIN(self), self->audioSrc.src());
- self->audioSrc.reset(false);
- }
-
- if (self->videoSrc.isUsed()) {
- gst_bin_remove(GST_BIN(self), self->videoSrc.src());
- self->videoSrc.reset(true);
- }
+ self->audioSrc.reset(false);
+ self->videoSrc.reset(true);
}
static void webkitMediaStreamSrcFinalize(GObject* object)
@@ -584,7 +588,8 @@
MediaStreamTrackPrivate* track, WebKitMediaStreamSrc::SourceData* data,
GstStaticPadTemplate* pad_template, bool onlyTrack)
{
- data->setSrc(gst_element_factory_make("appsrc", nullptr));
+ GRefPtr<GstElement> src = "" nullptr);
+ data->setSrc(WTFMove(src));
if (track->isCaptureTrack())
g_object_set(data->src(), "do-timestamp", true, nullptr);
@@ -624,19 +629,11 @@
static void webkitMediaStreamSrcRemoveTrackByType(WebKitMediaStreamSrc* self, RealtimeMediaSource::Type trackType)
{
- if (trackType == RealtimeMediaSource::Type::Audio) {
- if (self->audioSrc.isUsed()) {
- gst_element_set_state(self->audioSrc.src(), GST_STATE_NULL);
- gst_bin_remove(GST_BIN(self), self->audioSrc.src());
- self->audioSrc.reset(false);
- }
- } else if (trackType == RealtimeMediaSource::Type::Video) {
- if (self->videoSrc.isUsed()) {
- gst_element_set_state(self->videoSrc.src(), GST_STATE_NULL);
- gst_bin_remove(GST_BIN(self), self->videoSrc.src());
- self->videoSrc.reset(true);
- }
- } else
+ if (trackType == RealtimeMediaSource::Type::Audio)
+ self->audioSrc.reset(false);
+ else if (trackType == RealtimeMediaSource::Type::Video)
+ self->videoSrc.reset(true);
+ else
GST_INFO("Unsupported track type: %d", static_cast<int>(trackType));
}
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