Changeset 140272 in webkit

Timestamp:

Jan 20, 2013 3:36:20 AM (11 years ago)

Author:

Christophe Dumez

Message:

[gstreamer] assertion in g_object_unref from _WebKitWebAudioSourcePrivate destructor
​https://bugs.webkit.org/show_bug.cgi?id=107374

Reviewed by Philippe Normand.

WebKitWebAudioSrc object gets unrefed one time too many in
_WebKitWebAudioSourcePrivate destructor, causing an assertion
in g_object_unref. This patch passes NULL instead of
g_object_unref to gst_task_new() so that the WebKitWebAudioSrc
object does not get unrefed when the GstTask is destroyed.

No new tests, already covered by webaudio tests.

• platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:

(webkit_web_audio_src_init):

Location:

trunk/Source/WebCore

Files:

• ChangeLog (modified) (1 diff)
• platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (modified) (1 diff)

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2013-01-20  Christophe Dumez  <christophe.dumez@intel.com>
+
+        [gstreamer]  assertion in g_object_unref from _WebKitWebAudioSourcePrivate destructor
+        https://bugs.webkit.org/show_bug.cgi?id=107374
+
+        Reviewed by Philippe Normand.
+
+        WebKitWebAudioSrc object gets unrefed one time too many in
+        _WebKitWebAudioSourcePrivate destructor, causing an assertion
+        in g_object_unref. This patch passes NULL instead of
+        g_object_unref to gst_task_new() so that the WebKitWebAudioSrc
+        object does not get unrefed when the GstTask is destroyed.
+
+        No new tests, already covered by webaudio tests.
+
+        * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+        (webkit_web_audio_src_init):
+
 2013-01-20  Kentaro Hara  <haraken@chromium.org>
 

trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp

@@ -208 +208 @@
 #ifdef GST_API_VERSION_1
     g_rec_mutex_init(&priv->mutex);
-    priv->task = gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, reinterpret_cast<GDestroyNotify>(g_object_unref));
+    priv->task = gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, 0);
 #else
     g_static_rec_mutex_init(&priv->mutex);