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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/177341">177341</a></dd>
<dt>Author</dt> <dd>commit-queue@webkit.org</dd>
<dt>Date</dt> <dd>2014-12-16 00:17:58 -0800 (Tue, 16 Dec 2014)</dd>
</dl>
<h3>Log Message</h3>
<pre>[GStreamer] Fix deadlock when shutting down AudioDestination
https://bugs.webkit.org/show_bug.cgi?id=139496
Patch by Sebastian Dröge <sebastian@centricular.com> on 2014-12-16
Reviewed by Philippe Normand.
* platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
(webKitWebAudioSrcLoop):
(webKitWebAudioSrcChangeState):
Sometimes we would wait forever for the task to shut down. This
was happening because of a bug in GStreamer that caused joining
a paused task to deadlock.</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreplatformaudiogstreamerWebKitWebAudioSourceGStreamercpp">trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (177340 => 177341)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2014-12-16 08:16:46 UTC (rev 177340)
+++ trunk/Source/WebCore/ChangeLog 2014-12-16 08:17:58 UTC (rev 177341)
</span><span class="lines">@@ -1,3 +1,17 @@
</span><ins>+2014-12-16 Sebastian Dröge <sebastian@centricular.com>
+
+ [GStreamer] Fix deadlock when shutting down AudioDestination
+ https://bugs.webkit.org/show_bug.cgi?id=139496
+
+ Reviewed by Philippe Normand.
+
+ * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+ (webKitWebAudioSrcLoop):
+ (webKitWebAudioSrcChangeState):
+ Sometimes we would wait forever for the task to shut down. This
+ was happening because of a bug in GStreamer that caused joining
+ a paused task to deadlock.
+
</ins><span class="cx"> 2014-12-15 Dhi Aurrahman <diorahman@rockybars.com>
</span><span class="cx">
</span><span class="cx"> Extend :lang()'s selector checker to handle ranges with '*' properly and perform matching within the ASCII range
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiogstreamerWebKitWebAudioSourceGStreamercpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (177340 => 177341)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2014-12-16 08:16:46 UTC (rev 177340)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2014-12-16 08:17:58 UTC (rev 177341)
</span><span class="lines">@@ -314,7 +314,7 @@
</span><span class="cx"> ASSERT(priv->provider);
</span><span class="cx"> if (!priv->provider || !priv->bus) {
</span><span class="cx"> GST_ELEMENT_ERROR(src, CORE, FAILED, ("Internal WebAudioSrc error"), ("Can't start without provider or bus"));
</span><del>- gst_task_pause(src->priv->task.get());
</del><ins>+ gst_task_stop(src->priv->task.get());
</ins><span class="cx"> return;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -339,8 +339,11 @@
</span><span class="cx"> g_free(buffer);
</span><span class="cx"> channelBufferList = g_slist_delete_link(channelBufferList, channelBufferList);
</span><span class="cx"> }
</span><del>- GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to allocate buffer for flow: %s", gst_flow_get_name(ret)));
- gst_task_pause(src->priv->task.get());
</del><ins>+
+ // FLUSHING and EOS are not errors.
+ if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
+ GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to allocate buffer for flow: %s", gst_flow_get_name(ret)));
+ gst_task_stop(src->priv->task.get());
</ins><span class="cx"> return;
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -359,6 +362,7 @@
</span><span class="cx"> GSList* sourcesIt = priv->sources;
</span><span class="cx"> GSList* buffersIt = channelBufferList;
</span><span class="cx">
</span><ins>+ GstFlowReturn ret = GST_FLOW_OK;
</ins><span class="cx"> for (i = 0; sourcesIt && buffersIt; sourcesIt = g_slist_next(sourcesIt), buffersIt = g_slist_next(buffersIt), ++i) {
</span><span class="cx"> GstElement* appsrc = static_cast<GstElement*>(sourcesIt->data);
</span><span class="cx"> AudioSrcBuffer* buffer = static_cast<AudioSrcBuffer*>(buffersIt->data);
</span><span class="lines">@@ -368,11 +372,16 @@
</span><span class="cx"> gst_buffer_unmap(channelBuffer, &buffer->info);
</span><span class="cx"> g_free(buffer);
</span><span class="cx">
</span><del>- GstFlowReturn ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc), channelBuffer);
- if (ret != GST_FLOW_OK) {
- GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc), gst_flow_get_name(ret)));
- gst_task_pause(src->priv->task.get());
- }
</del><ins>+ if (ret == GST_FLOW_OK) {
+ ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc), channelBuffer);
+ if (ret != GST_FLOW_OK) {
+ // FLUSHING and EOS are not errors.
+ if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
+ GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc), gst_flow_get_name(ret)));
+ gst_task_stop(src->priv->task.get());
+ }
+ } else
+ gst_buffer_unref(channelBuffer);
</ins><span class="cx"> }
</span><span class="cx">
</span><span class="cx"> g_slist_free(channelBufferList);
</span><span class="lines">@@ -417,6 +426,9 @@
</span><span class="cx"> }
</span><span class="cx"> case GST_STATE_CHANGE_PAUSED_TO_READY:
</span><span class="cx"> GST_DEBUG_OBJECT(src, "PAUSED->READY");
</span><ins>+#if GST_CHECK_VERSION(1, 4, 0)
+ gst_buffer_pool_set_flushing(src->priv->pool, TRUE);
+#endif
</ins><span class="cx"> if (!gst_task_join(src->priv->task.get()))
</span><span class="cx"> returnValue = GST_STATE_CHANGE_FAILURE;
</span><span class="cx"> gst_buffer_pool_set_active(src->priv->pool, FALSE);
</span></span></pre>
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