REGRESSION(r274527): [GStreamer] media/webaudio-background-playback.h…

…tml now failing

https://bugs.webkit.org/show_bug.cgi?id=223840

Patch by Philippe Normand <pnormand@igalia.com> on 2021-03-28
Reviewed by Chris Dumez.

r274527 actually exposed a bug that was present since r271197. The AudioDestination pipeline
was not stopping properly because webkitGstSetElementStateSynchronously() was returning too
early.

This patch also includes a few improvements in the webaudiosrc element, most notably
regarding its preroll state. The element now stops emitting buffers downstream as soon as
the AudioDestination is scheduled to stop.

* platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
(webKitWebAudioSrcRenderAndPushFrames): Return early as soon as the destination is scheduled to stop.
(webKitWebAudioSrcChangeState): Synchronize preroll state with element state.
* platform/graphics/gstreamer/GStreamerCommon.cpp:
(WebCore::webkitGstSetElementStateSynchronously): targetState might be lower than current
state, e.g, when stopping a running pipeline, so we can't return early for those cases,
otherwise the pipeline won't stop.

Canonical link: https://commits.webkit.org/235852@main
git-svn-id: https://svn.webkit.org/repository/webkit/trunk@275149 268f45cc-cd09-0410-ab3c-d52691b4dbfc

Source/WebCore/ChangeLog

@@ -1,3 +1,26 @@
+2021-03-28  Philippe Normand  <pnormand@igalia.com>
+
+        REGRESSION(r274527): [GStreamer] media/webaudio-background-playback.html now failing
+        https://bugs.webkit.org/show_bug.cgi?id=223840
+
+        Reviewed by Chris Dumez.
+
+        r274527 actually exposed a bug that was present since r271197. The AudioDestination pipeline
+        was not stopping properly because webkitGstSetElementStateSynchronously() was returning too
+        early.
+
+        This patch also includes a few improvements in the webaudiosrc element, most notably
+        regarding its preroll state. The element now stops emitting buffers downstream as soon as
+        the AudioDestination is scheduled to stop.
+
+        * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+        (webKitWebAudioSrcRenderAndPushFrames): Return early as soon as the destination is scheduled to stop.
+        (webKitWebAudioSrcChangeState): Synchronize preroll state with element state.
+        * platform/graphics/gstreamer/GStreamerCommon.cpp:
+        (WebCore::webkitGstSetElementStateSynchronously): targetState might be lower than current
+        state, e.g, when stopping a running pipeline, so we can't return early for those cases,
+        otherwise the pipeline won't stop.
+
 2021-03-28  Chris Dumez  <cdumez@apple.com>
 
         Unreviewed, add an exception for a heap allocation on the WebAudio thread.

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

@@ -33,6 +33,7 @@
 #include <gst/pbutils/missing-plugins.h>
 #include <wtf/Condition.h>
 #include <wtf/Lock.h>
+#include <wtf/Scope.h>
 #include <wtf/glib/GUniquePtr.h>
 #include <wtf/glib/WTFGType.h>
 
@@ -337,6 +338,16 @@ static void webKitWebAudioSrcRenderAndPushFrames(GRefPtr<GstElement>&& element,
     auto* src = WEBKIT_WEB_AUDIO_SRC(element.get());
     auto* priv = src->priv;
 
+    auto notifyDispatchOnExit = makeScopeExit([priv] {
+        LockHolder lock(priv->dispatchLock);
+        priv->dispatchDone = true;
+        priv->dispatchCondition.notifyOne();
+    });
+
+    GST_TRACE_OBJECT(element.get(), "Playing: %d", priv->destination->isPlaying());
+    if (priv->hasRenderedAudibleFrame && !priv->destination->isPlaying())
+        return;
+
     ASSERT(channelBufferList.size() == priv->sources.size());
 
     GstClockTime timestamp = gst_util_uint64_scale(priv->numberOfSamples, GST_SECOND, priv->sampleRate);
@@ -381,12 +392,6 @@ static void webKitWebAudioSrcRenderAndPushFrames(GRefPtr<GstElement>&& element,
             failed = true;
         }
     }
-
-    {
-        LockHolder lock(priv->dispatchLock);
-        priv->dispatchDone = true;
-        priv->dispatchCondition.notifyOne();
-    }
 }
 
 static void webKitWebAudioSrcRenderIteration(WebKitWebAudioSrc* src)
@@ -473,6 +478,7 @@ static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, Gs
 
         gst_buffer_pool_set_active(priv->pool.get(), FALSE);
         priv->pool = nullptr;
+        priv->hasRenderedAudibleFrame = false;
         break;
     default:
         break;

Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.cpp

@@ -468,7 +468,7 @@ bool webkitGstSetElementStateSynchronously(GstElement* pipeline, GstState target
 
     GstState currentState;
     auto result = gst_element_get_state(pipeline, &currentState, nullptr, 10);
-    if (result == GST_STATE_CHANGE_SUCCESS && currentState >= targetState) {
+    if (result == GST_STATE_CHANGE_SUCCESS && currentState == targetState) {
         GST_DEBUG_OBJECT(pipeline, "Target state already reached");
         return true;
     }