inter: Port to 1.0 API

Also remove a lot of empty, non-implemented methods

1 files changed, 13 insertions, 134 deletions

diff --git a/gst/inter/gstinteraudiosink.c b/gst/inter/gstinteraudiosink.c
index 8cc3888cc..b67c6fc1d 100644
--- a/gst/inter/gstinteraudiosink.c
+++ b/gst/inter/gstinteraudiosink.c
@@ -57,30 +57,14 @@ static void gst_inter_audio_sink_set_property (GObject * object,
     guint property_id, const GValue * value, GParamSpec * pspec);
 static void gst_inter_audio_sink_get_property (GObject * object,
     guint property_id, GValue * value, GParamSpec * pspec);
-static void gst_inter_audio_sink_dispose (GObject * object);
 static void gst_inter_audio_sink_finalize (GObject * object);
 
-static GstCaps *gst_inter_audio_sink_get_caps (GstBaseSink * sink);
-static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
-    GstCaps * caps);
-static GstFlowReturn gst_inter_audio_sink_buffer_alloc (GstBaseSink * sink,
-    guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf);
 static void gst_inter_audio_sink_get_times (GstBaseSink * sink,
     GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
 static gboolean gst_inter_audio_sink_start (GstBaseSink * sink);
 static gboolean gst_inter_audio_sink_stop (GstBaseSink * sink);
-static gboolean gst_inter_audio_sink_unlock (GstBaseSink * sink);
-static gboolean gst_inter_audio_sink_event (GstBaseSink * sink,
-    GstEvent * event);
-static GstFlowReturn gst_inter_audio_sink_preroll (GstBaseSink * sink,
-    GstBuffer * buffer);
 static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
     GstBuffer * buffer);
-static GstStateChangeReturn gst_inter_audio_sink_async_play (GstBaseSink *
-    sink);
-static gboolean gst_inter_audio_sink_activate_pull (GstBaseSink * sink,
-    gboolean active);
-static gboolean gst_inter_audio_sink_unlock_stop (GstBaseSink * sink);
 
 enum
 {
@@ -94,28 +78,25 @@ static GstStaticPadTemplate gst_inter_audio_sink_sink_template =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) BYTE_ORDER, "
-        "signed = (boolean) true, "
-        "width = (int) 16, "
-        "depth = (int) 16, " "rate = (int) 48000, " "channels = (int) 2")
+    GST_STATIC_CAPS ("audio/x-raw, format = (string) " GST_AUDIO_NE (S16) ", "
+        "rate = (int) 48000, channels = (int) 2")
     );
 
 
 /* class initialization */
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_inter_audio_sink_debug_category, "interaudiosink", 0, \
-      "debug category for interaudiosink element");
 
-GST_BOILERPLATE_FULL (GstInterAudioSink, gst_inter_audio_sink, GstBaseSink,
-    GST_TYPE_BASE_SINK, DEBUG_INIT);
+G_DEFINE_TYPE (GstInterAudioSink, gst_inter_audio_sink, GST_TYPE_BASE_SINK);
 
 static void
-gst_inter_audio_sink_base_init (gpointer g_class)
+gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass);
 
+  GST_DEBUG_CATEGORY_INIT (gst_inter_audio_sink_debug_category,
+      "interaudiosink", 0, "debug category for interaudiosink element");
   gst_element_class_add_pad_template (element_class,
       gst_static_pad_template_get (&gst_inter_audio_sink_sink_template));
 
@@ -124,41 +105,15 @@ gst_inter_audio_sink_base_init (gpointer g_class)
       "Sink/Audio",
       "Virtual audio sink for internal process communication",
       "David Schleef <ds@schleef.org>");
-}
-
-static void
-gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
-{
-  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
-  GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass);
 
   gobject_class->set_property = gst_inter_audio_sink_set_property;
   gobject_class->get_property = gst_inter_audio_sink_get_property;
-  gobject_class->dispose = gst_inter_audio_sink_dispose;
   gobject_class->finalize = gst_inter_audio_sink_finalize;
-  base_sink_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_caps);
-  base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
-  if (0)
-    base_sink_class->buffer_alloc =
-        GST_DEBUG_FUNCPTR (gst_inter_audio_sink_buffer_alloc);
   base_sink_class->get_times =
       GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_times);
   base_sink_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_start);
   base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
-  base_sink_class->unlock = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_unlock);
-  if (0)
-    base_sink_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_event);
-  //if (0)
-  base_sink_class->preroll = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_preroll);
   base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
-  if (0)
-    base_sink_class->async_play =
-        GST_DEBUG_FUNCPTR (gst_inter_audio_sink_async_play);
-  if (0)
-    base_sink_class->activate_pull =
-        GST_DEBUG_FUNCPTR (gst_inter_audio_sink_activate_pull);
-  base_sink_class->unlock_stop =
-      GST_DEBUG_FUNCPTR (gst_inter_audio_sink_unlock_stop);
 
   g_object_class_install_property (gobject_class, PROP_CHANNEL,
       g_param_spec_string ("channel", "Channel",
@@ -167,8 +122,7 @@ gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
 }
 
 static void
-gst_inter_audio_sink_init (GstInterAudioSink * interaudiosink,
-    GstInterAudioSinkClass * interaudiosink_class)
+gst_inter_audio_sink_init (GstInterAudioSink * interaudiosink)
 {
   interaudiosink->channel = g_strdup ("default");
 }
@@ -207,16 +161,6 @@ gst_inter_audio_sink_get_property (GObject * object, guint property_id,
 }
 
 void
-gst_inter_audio_sink_dispose (GObject * object)
-{
-  /* GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object); */
-
-  /* clean up as possible.  may be called multiple times */
-
-  G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-void
 gst_inter_audio_sink_finalize (GObject * object)
 {
   GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
@@ -224,33 +168,10 @@ gst_inter_audio_sink_finalize (GObject * object)
   /* clean up object here */
   g_free (interaudiosink->channel);
 
-  G_OBJECT_CLASS (parent_class)->finalize (object);
+  G_OBJECT_CLASS (gst_inter_audio_sink_parent_class)->finalize (object);
 }
 
 
-
-static GstCaps *
-gst_inter_audio_sink_get_caps (GstBaseSink * sink)
-{
-
-  return NULL;
-}
-
-static gboolean
-gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
-{
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_inter_audio_sink_buffer_alloc (GstBaseSink * sink, guint64 offset,
-    guint size, GstCaps * caps, GstBuffer ** buf)
-{
-
-  return GST_FLOW_ERROR;
-}
-
 static void
 gst_inter_audio_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
     GstClockTime * start, GstClockTime * end)
@@ -302,39 +223,18 @@ gst_inter_audio_sink_stop (GstBaseSink * sink)
   return TRUE;
 }
 
-static gboolean
-gst_inter_audio_sink_unlock (GstBaseSink * sink)
-{
-
-  return TRUE;
-}
-
-static gboolean
-gst_inter_audio_sink_event (GstBaseSink * sink, GstEvent * event)
-{
-
-  return TRUE;
-}
-
-static GstFlowReturn
-gst_inter_audio_sink_preroll (GstBaseSink * sink, GstBuffer * buffer)
-{
-
-  return GST_FLOW_OK;
-}
-
 static GstFlowReturn
 gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
 {
   GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
   int n;
 
-  GST_DEBUG ("render %d", GST_BUFFER_SIZE (buffer));
+  GST_DEBUG ("render %d", gst_buffer_get_size (buffer));
 
   g_mutex_lock (interaudiosink->surface->mutex);
   n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
 #define SIZE 1600
-  if (n > (1600 * 3)) {
+  if (n > (SIZE * 3)) {
     GST_WARNING ("flushing 800 samples");
     gst_adapter_flush (interaudiosink->surface->audio_adapter, (SIZE / 2) * 4);
     n -= (SIZE / 2);
@@ -345,24 +245,3 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
 
   return GST_FLOW_OK;
 }
-
-static GstStateChangeReturn
-gst_inter_audio_sink_async_play (GstBaseSink * sink)
-{
-
-  return GST_STATE_CHANGE_SUCCESS;
-}
-
-static gboolean
-gst_inter_audio_sink_activate_pull (GstBaseSink * sink, gboolean active)
-{
-
-  return TRUE;
-}
-
-static gboolean
-gst_inter_audio_sink_unlock_stop (GstBaseSink * sink)
-{
-
-  return TRUE;
-}