Modified: trunk/Source/WebCore/ChangeLog (138544 => 138545)
--- trunk/Source/WebCore/ChangeLog 2012-12-28 20:32:06 UTC (rev 138544)
+++ trunk/Source/WebCore/ChangeLog 2012-12-28 21:04:18 UTC (rev 138545)
@@ -1,3 +1,22 @@
+2012-12-28 Philippe Normand <[email protected]>
+
+ [GStreamer] GstMessage handler in AudioDestination
+ https://bugs.webkit.org/show_bug.cgi?id=105292
+
+ Reviewed by Martin Robinson.
+
+ Handle GStreamer warnings and errors coming from the playback
+ pipeline. For now we only display warnings on the console.
+
+ * platform/audio/gstreamer/AudioDestinationGStreamer.cpp:
+ (WebCore::messageCallback):
+ (WebCore):
+ (WebCore::AudioDestinationGStreamer::AudioDestinationGStreamer):
+ (WebCore::AudioDestinationGStreamer::finishBuildingPipelineAfterWavParserPadReady):
+ (WebCore::AudioDestinationGStreamer::handleMessage):
+ * platform/audio/gstreamer/AudioDestinationGStreamer.h:
+ (AudioDestinationGStreamer):
+
2012-12-28 Martin Robinson <[email protected]>
[GTK][WK2] Add support for IME Composition
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp (138544 => 138545)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2012-12-28 20:32:06 UTC (rev 138544)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2012-12-28 21:04:18 UTC (rev 138545)
@@ -36,6 +36,11 @@
// needs to handle this number of frames per cycle as well.
const unsigned framesToPull = 128;
+gboolean messageCallback(GstBus* bus, GstMessage* message, AudioDestinationGStreamer* destination)
+{
+ return destination->handleMessage(message);
+}
+
PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, float sampleRate)
{
return adoptPtr(new AudioDestinationGStreamer(callback, sampleRate));
@@ -58,6 +63,9 @@
, m_isPlaying(false)
{
m_pipeline = gst_pipeline_new("play");
+ GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
+ 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,
@@ -79,6 +87,8 @@
AudioDestinationGStreamer::~AudioDestinationGStreamer()
{
+ GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
+ g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
gst_element_set_state(m_pipeline, GST_STATE_NULL);
gst_object_unref(m_pipeline);
}
@@ -119,6 +129,28 @@
gst_element_sync_state_with_parent(audioSink.leakRef());
}
+gboolean AudioDestinationGStreamer::handleMessage(GstMessage* message)
+{
+ GOwnPtr<GError> error;
+ GOwnPtr<gchar> debug;
+
+ switch (GST_MESSAGE_TYPE(message)) {
+ case GST_MESSAGE_WARNING:
+ gst_message_parse_warning(message, &error.outPtr(), &debug.outPtr());
+ g_warning("Warning: %d, %s. Debug output: %s", error->code, error->message, debug.get());
+ break;
+ case GST_MESSAGE_ERROR:
+ gst_message_parse_error(message, &error.outPtr(), &debug.outPtr());
+ g_warning("Error: %d, %s. Debug output: %s", error->code, error->message, debug.get());
+ gst_element_set_state(m_pipeline, GST_STATE_NULL);
+ m_isPlaying = false;
+ break;
+ default:
+ break;
+ }
+ return TRUE;
+}
+
void AudioDestinationGStreamer::start()
{
ASSERT(m_wavParserAvailable);
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.h (138544 => 138545)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.h 2012-12-28 20:32:06 UTC (rev 138544)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.h 2012-12-28 21:04:18 UTC (rev 138545)
@@ -24,6 +24,7 @@
typedef struct _GstElement GstElement;
typedef struct _GstPad GstPad;
+typedef struct _GstMessage GstMessage;
namespace WebCore {
@@ -40,6 +41,7 @@
AudioIOCallback& callback() const { return m_callback; }
void finishBuildingPipelineAfterWavParserPadReady(GstPad*);
+ gboolean handleMessage(GstMessage*);
private:
AudioIOCallback& m_callback;
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp (138544 => 138545)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp 2012-12-28 20:32:06 UTC (rev 138544)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp 2012-12-28 21:04:18 UTC (rev 138545)
@@ -148,7 +148,7 @@
AudioFileReader::~AudioFileReader()
{
if (m_pipeline) {
- GRefPtr<GstBus> bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+ GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
gst_element_set_state(m_pipeline, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(m_pipeline));
@@ -331,7 +331,7 @@
// A deinterleave element is added once a src pad becomes available in decodebin.
m_pipeline = gst_pipeline_new(0);
- GRefPtr<GstBus> bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
+ GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
gst_bus_add_signal_watch(bus.get());
g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
