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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/270184">270184</a></dd>
<dt>Author</dt> <dd>philn@webkit.org</dd>
<dt>Date</dt> <dd>2020-11-27 01:08:39 -0800 (Fri, 27 Nov 2020)</dd>
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<h3>Log Message</h3>
<pre>[GStreamer] AudioSourceProvider can potentially invoke an already-freed client
https://bugs.webkit.org/show_bug.cgi?id=217952

Reviewed by Xabier Rodriguez-Calvar.

* platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp:
(WebCore::AudioSourceProviderGStreamer::deinterleavePadsConfigured): Check the provider has
a client before setting up the audio format.</pre>

<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreplatformaudiogstreamerAudioSourceProviderGStreamercpp">trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp</a></li>
</ul>

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<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (270183 => 270184)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog   2020-11-27 08:51:45 UTC (rev 270183)
+++ trunk/Source/WebCore/ChangeLog      2020-11-27 09:08:39 UTC (rev 270184)
</span><span class="lines">@@ -1,3 +1,14 @@
</span><ins>+2020-11-27  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer] AudioSourceProvider can potentially invoke an already-freed client
+        https://bugs.webkit.org/show_bug.cgi?id=217952
+
+        Reviewed by Xabier Rodriguez-Calvar.
+
+        * platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp:
+        (WebCore::AudioSourceProviderGStreamer::deinterleavePadsConfigured): Check the provider has
+        a client before setting up the audio format.
+
</ins><span class="cx"> 2020-11-26  Lauro Moura  <lmoura@igalia.com>
</span><span class="cx"> 
</span><span class="cx">         [GTK4] Build fix. Add cast when taking const data ownership
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiogstreamerAudioSourceProviderGStreamercpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp (270183 => 270184)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp   2020-11-27 08:51:45 UTC (rev 270183)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp      2020-11-27 09:08:39 UTC (rev 270184)
</span><span class="lines">@@ -108,7 +108,7 @@
</span><span class="cx"> {
</span><span class="cx">     m_notifier->invalidate();
</span><span class="cx"> 
</span><del>-    GRefPtr<GstElement> deinterleave = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "deinterleave"));
</del><ins>+    auto deinterleave = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "deinterleave"));
</ins><span class="cx">     if (deinterleave && m_client) {
</span><span class="cx">         g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadAddedHandlerId);
</span><span class="cx">         g_signal_handler_disconnect(deinterleave.get(), m_deinterleaveNoMorePadsHandlerId);
</span><span class="lines">@@ -115,10 +115,7 @@
</span><span class="cx">         g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadRemovedHandlerId);
</span><span class="cx">     }
</span><span class="cx"> 
</span><del>-#if ENABLE(MEDIA_STREAM)
-    if (m_pipeline)
-        gst_element_set_state(m_pipeline.get(), GST_STATE_NULL);
-#endif
</del><ins>+    setClient(nullptr);
</ins><span class="cx"> }
</span><span class="cx"> 
</span><span class="cx"> void AudioSourceProviderGStreamer::configureAudioBin(GstElement* audioBin, GstElement* audioSink)
</span><span class="lines">@@ -189,22 +186,26 @@
</span><span class="cx"> 
</span><span class="cx"> void AudioSourceProviderGStreamer::setClient(AudioSourceProviderClient* client)
</span><span class="cx"> {
</span><del>-    if (m_client)
</del><ins>+    if (m_client == client)
</ins><span class="cx">         return;
</span><span class="cx"> 
</span><del>-    ASSERT(client);
</del><span class="cx">     m_client = client;
</span><span class="cx"> 
</span><span class="cx"> #if ENABLE(MEDIA_STREAM)
</span><span class="cx">     if (m_pipeline)
</span><del>-        gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
</del><ins>+        gst_element_set_state(m_pipeline.get(), m_client ? GST_STATE_PLAYING : GST_STATE_NULL);
</ins><span class="cx"> #endif
</span><span class="cx"> 
</span><ins>+    // FIXME: This early return should ideally be replaced by a removal of the m_audioSinkBin from
+    // its parent pipeline. https://bugs.webkit.org/show_bug.cgi?id=219245
+    if (!m_client)
+        return;
+
</ins><span class="cx">     // The volume element is used to mute audio playback towards the
</span><span class="cx">     // autoaudiosink. This is needed to avoid double playback of audio
</span><span class="cx">     // from our audio sink and from the WebAudio AudioDestination node
</span><span class="cx">     // supposedly configured already by application side.
</span><del>-    GRefPtr<GstElement> volumeElement = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "volume"));
</del><ins>+    auto volumeElement = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "volume"));
</ins><span class="cx"> 
</span><span class="cx">     if (volumeElement)
</span><span class="cx">         g_object_set(volumeElement.get(), "mute", TRUE, nullptr);
</span><span class="lines">@@ -320,9 +321,9 @@
</span><span class="cx"> void AudioSourceProviderGStreamer::deinterleavePadsConfigured()
</span><span class="cx"> {
</span><span class="cx">     GST_DEBUG("Deinterleave configured, notifying client");
</span><del>-    m_notifier->notify(MainThreadNotification::DeinterleavePadsConfigured, [this] {
-        ASSERT(m_client);
-        m_client->setFormat(m_deinterleaveSourcePads, gSampleBitRate);
</del><ins>+    m_notifier->notify(MainThreadNotification::DeinterleavePadsConfigured, [numberOfChannels = m_deinterleaveSourcePads, sampleRate = gSampleBitRate, client = m_client] {
+        if (client)
+            client->setFormat(numberOfChannels, sampleRate);
</ins><span class="cx">     });
</span><span class="cx"> }
</span><span class="cx"> 
</span></span></pre>
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