Title: [138576] trunk/Source/WebCore
- Revision
- 138576
- Author
- [email protected]
- Date
- 2012-12-30 06:57:20 -0800 (Sun, 30 Dec 2012)
Log Message
Regression(r138545): Makes WebAudio tests crash
https://bugs.webkit.org/show_bug.cgi?id=105869
Reviewed by Philippe Normand.
Stop using smart pointers and adoptGRef() for the
GstBus object and call gst_object_unref() manually.
This fixes assertions hits when calling adoptGRef()
since the GstBus object has a floating reference.
Alternatively, we could keep using smart pointers
and stop calling adoptGRef() so that
gst_object_ref_sink() is called and the floating
flag is cleared but I think it is clearer to do
it manually here.
No new tests, already covered by existing tests.
* platform/audio/gstreamer/AudioDestinationGStreamer.cpp:
(WebCore::messageCallback):
(WebCore::AudioDestinationGStreamer::AudioDestinationGStreamer):
(WebCore::AudioDestinationGStreamer::~AudioDestinationGStreamer):
* platform/audio/gstreamer/AudioFileReaderGStreamer.cpp:
(WebCore::AudioFileReader::~AudioFileReader):
(WebCore::AudioFileReader::decodeAudioForBusCreation):
Modified Paths
Diff
Modified: trunk/Source/WebCore/ChangeLog (138575 => 138576)
--- trunk/Source/WebCore/ChangeLog 2012-12-30 14:52:13 UTC (rev 138575)
+++ trunk/Source/WebCore/ChangeLog 2012-12-30 14:57:20 UTC (rev 138576)
@@ -1,3 +1,31 @@
+2012-12-30 Christophe Dumez <[email protected]>
+
+ Regression(r138545): Makes WebAudio tests crash
+ https://bugs.webkit.org/show_bug.cgi?id=105869
+
+ Reviewed by Philippe Normand.
+
+ Stop using smart pointers and adoptGRef() for the
+ GstBus object and call gst_object_unref() manually.
+ This fixes assertions hits when calling adoptGRef()
+ since the GstBus object has a floating reference.
+
+ Alternatively, we could keep using smart pointers
+ and stop calling adoptGRef() so that
+ gst_object_ref_sink() is called and the floating
+ flag is cleared but I think it is clearer to do
+ it manually here.
+
+ No new tests, already covered by existing tests.
+
+ * platform/audio/gstreamer/AudioDestinationGStreamer.cpp:
+ (WebCore::messageCallback):
+ (WebCore::AudioDestinationGStreamer::AudioDestinationGStreamer):
+ (WebCore::AudioDestinationGStreamer::~AudioDestinationGStreamer):
+ * platform/audio/gstreamer/AudioFileReaderGStreamer.cpp:
+ (WebCore::AudioFileReader::~AudioFileReader):
+ (WebCore::AudioFileReader::decodeAudioForBusCreation):
+
2012-12-29 Dimitri Glazkov <[email protected]>
Rename StyleResolver's member variable that holds SelectorChecker to a proper name.
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp (138575 => 138576)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2012-12-30 14:52:13 UTC (rev 138575)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2012-12-30 14:57:20 UTC (rev 138576)
@@ -36,7 +36,7 @@
// needs to handle this number of frames per cycle as well.
const unsigned framesToPull = 128;
-gboolean messageCallback(GstBus* bus, GstMessage* message, AudioDestinationGStreamer* destination)
+gboolean messageCallback(GstBus*, GstMessage* message, AudioDestinationGStreamer* destination)
{
return destination->handleMessage(message);
}
@@ -63,9 +63,11 @@
, m_isPlaying(false)
{
m_pipeline = gst_pipeline_new("play");
- GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
- gst_bus_add_signal_watch(bus.get());
- g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
+ GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+ ASSERT(bus);
+ gst_bus_add_signal_watch(bus);
+ g_signal_connect(bus, "message", G_CALLBACK(messageCallback), this);
+ gst_object_unref(bus);
GstElement* webkitAudioSrc = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC,
"rate", sampleRate,
@@ -87,8 +89,10 @@
AudioDestinationGStreamer::~AudioDestinationGStreamer()
{
- GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
- g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
+ GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+ ASSERT(bus);
+ g_signal_handlers_disconnect_by_func(bus, reinterpret_cast<gpointer>(messageCallback), this);
+ gst_object_unref(bus);
gst_element_set_state(m_pipeline, GST_STATE_NULL);
gst_object_unref(m_pipeline);
}
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp (138575 => 138576)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp 2012-12-30 14:52:13 UTC (rev 138575)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp 2012-12-30 14:57:20 UTC (rev 138576)
@@ -148,8 +148,10 @@
AudioFileReader::~AudioFileReader()
{
if (m_pipeline) {
- GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
- g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
+ GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+ ASSERT(bus);
+ g_signal_handlers_disconnect_by_func(bus, reinterpret_cast<gpointer>(messageCallback), this);
+ gst_object_unref(bus);
gst_element_set_state(m_pipeline, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(m_pipeline));
}
@@ -331,9 +333,11 @@
// A deinterleave element is added once a src pad becomes available in decodebin.
m_pipeline = gst_pipeline_new(0);
- GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
- gst_bus_add_signal_watch(bus.get());
- g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
+ GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+ ASSERT(bus);
+ gst_bus_add_signal_watch(bus);
+ g_signal_connect(bus, "message", G_CALLBACK(messageCallback), this);
+ gst_object_unref(bus);
GstElement* source;
if (m_data) {
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