[CherryPick] [gstreamer] GstBus signal watch should be removed on clean up

[Title] [gstreamer] GstBus signal watch should be removed on clean up
[Issues] NA
[Problem] Memory leak fix in WebAudio.
[Solution] Cherry picked.
[Cherry-Picker] Praveen R Jadhav <praveen.j@samsung.com>

https://bugs.webkit.org/show_bug.cgi?id=107544

Reviewed by Philippe Normand.

Our gstreamer backend code currently calls gst_bus_add_signal_watch()
on GstBus to add a signal watch. As per the gstreamer  documentation,
"To clean up, the caller is responsible for calling
gst_bus_remove_signal_watch() as many times as this function is
called". This is because gst_bus_add_signal_watch() causes the GstBus
object to be ref'd and gst_bus_remove_signal_watch() needs to be
called to properly unref it.

This patch makes sure that gst_bus_remove_signal_watch() is called
on the GstBus object when cleaning up. This patch also uses smart
pointers for GstBus objects for consistency.

No new tests, no behavior change for layout tests.

* platform/audio/gstreamer/AudioDestinationGStreamer.cpp:
(WebCore::AudioDestinationGStreamer::AudioDestinationGStreamer):
(WebCore::AudioDestinationGStreamer::~AudioDestinationGStreamer):
* platform/audio/gstreamer/AudioFileReaderGStreamer.cpp:
(WebCore::AudioFileReader::~AudioFileReader):
(WebCore::AudioFileReader::decodeAudioForBusCreation):
* platform/graphics/gstreamer/GStreamerGWorld.cpp:
(WebCore::GStreamerGWorld::GStreamerGWorld):
* platform/graphics/gstreamer/GStreamerVersioning.cpp:
(webkitGstPipelineGetBus):
* platform/graphics/gstreamer/GStreamerVersioning.h:
* platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
(WebCore::MediaPlayerPrivateGStreamer::~MediaPlayerPrivateGStreamer):

git-svn-id: http://svn.webkit.org/repository/webkit/trunk@140443 268f45cc-cd09-0410-ab3c-d52691b4dbfc

Conflicts:

	Source/WebCore/ChangeLog
	Source/WebCore/platform/graphics/gstreamer/GStreamerGWorld.cpp
	Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp

Change-Id: Ib1142fe97c9641e7c40cdba75f5e7a4ba2d28ff0

diff --git a/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp
index 9e3cde4..9df5401 100644 (file)
--- a/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp
+++ b/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp
@@ -125,22 +125,15 @@ AudioDestinationGStreamer::AudioDestinationGStreamer(AudioIOCallback& callback,
     , m_audioSessionManager(AudioSessionManagerGStreamerTizen::createAudioSessionManager())
 #endif
 {
-#if !ENABLE(TIZEN_WEB_AUDIO)
-    static bool gstInitialized = false;
-    if (!gstInitialized)
-        gstInitialized = gst_init_check(0, 0, 0);
-    ASSERT_WITH_MESSAGE(gstInitialized, "GStreamer initialization failed");
-#endif
 #if ENABLE(TIZEN_GSTREAMER_AUDIO)
     if (m_audioSessionManager)
         m_audioSessionManager->registerAudioSessionManager(MM_SESSION_TYPE_SHARE, AudioDestinationAudioSessionNotifyCallback, this);
 #endif
     m_pipeline = gst_pipeline_new("play");
-    GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+    GRefPtr<GstBus> bus = webkitGstPipelineGetBus(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);
+    gst_bus_add_signal_watch(bus.get());
+    g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
 
     GstElement* webkitAudioSrc = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC,
                                                                             "rate", sampleRate,
@@ -169,10 +162,11 @@ AudioDestinationGStreamer::AudioDestinationGStreamer(AudioIOCallback& callback,
 
 AudioDestinationGStreamer::~AudioDestinationGStreamer()
 {
-    GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+    GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
     ASSERT(bus);
-    g_signal_handlers_disconnect_by_func(bus, reinterpret_cast<gpointer>(messageCallback), this);
-    gst_object_unref(bus);
+    g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
+    gst_bus_remove_signal_watch(bus.get());
+
     gst_element_set_state(m_pipeline, GST_STATE_NULL);
 
 #if ENABLE(TIZEN_GSTREAMER_AUDIO)

diff --git a/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp
index ac692a7..5a9126b 100644 (file)
--- a/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp
+++ b/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp
@@ -175,10 +175,11 @@ AudioFileReader::AudioFileReader(const void* data, size_t dataSize)
 AudioFileReader::~AudioFileReader()
 {
     if (m_pipeline) {
-        GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+        GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
         ASSERT(bus);
-        g_signal_handlers_disconnect_by_func(bus, reinterpret_cast<gpointer>(messageCallback), this);
-        gst_object_unref(bus);
+        g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
+        gst_bus_remove_signal_watch(bus.get());
+
         gst_element_set_state(m_pipeline, GST_STATE_NULL);
         gst_object_unref(GST_OBJECT(m_pipeline));
     }
@@ -422,11 +423,10 @@ void AudioFileReader::decodeAudioForBusCreation()
     // A deinterleave element is added once a src pad becomes available in decodebin.
     m_pipeline = gst_pipeline_new(0);
 
-    GstBus* bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+    GRefPtr<GstBus> bus = webkitGstPipelineGetBus(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);
+    gst_bus_add_signal_watch(bus.get());
+    g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
 
     GstElement* source;
     if (m_data) {
@@ -451,15 +451,6 @@ PassOwnPtr<AudioBus> AudioFileReader::createBus(float sampleRate, bool mixToMono
 {
     m_sampleRate = sampleRate;
 
-#if !ENABLE(TIZEN_WEB_AUDIO)
-    static bool gstInitialized = false;
-    if (!gstInitialized)
-        gstInitialized = gst_init_check(0, 0, 0);
-
-    if (!gstInitialized)
-        return nullptr;
-#endif
-
     m_frontLeftBuffers = gst_buffer_list_new();
     m_frontRightBuffers = gst_buffer_list_new();