audio: Use G_DEFINE_TYPE instead of GST_BOILERPLATE

diff --git a/gst-libs/gst/audio/gstaudiosink.c b/gst-libs/gst/audio/gstaudiosink.c
index adea6be..88c6f7b 100644 (file)
--- a/gst-libs/gst/audio/gstaudiosink.c
+++ b/gst-libs/gst/audio/gstaudiosink.c
@@ -574,21 +574,16 @@ enum
   ARG_0,
 };
 
-#define _do_init(bla) \
+#define _do_init \
     GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element");
-
-GST_BOILERPLATE_FULL (GstAudioSink, gst_audio_sink, GstBaseAudioSink,
+#define gst_audio_sink_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstAudioSink, gst_audio_sink,
     GST_TYPE_BASE_AUDIO_SINK, _do_init);
 
 static GstRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink *
     sink);
 
 static void
-gst_audio_sink_base_init (gpointer g_class)
-{
-}
-
-static void
 gst_audio_sink_class_init (GstAudioSinkClass * klass)
 {
   GstBaseAudioSinkClass *gstbaseaudiosink_class;
@@ -602,7 +597,7 @@ gst_audio_sink_class_init (GstAudioSinkClass * klass)
 }
 
 static void
-gst_audio_sink_init (GstAudioSink * audiosink, GstAudioSinkClass * g_class)
+gst_audio_sink_init (GstAudioSink * audiosink)
 {
 }
 

diff --git a/gst-libs/gst/audio/gstaudiosrc.c b/gst-libs/gst/audio/gstaudiosrc.c
index b8f9e0b..57864d1 100644 (file)
--- a/gst-libs/gst/audio/gstaudiosrc.c
+++ b/gst-libs/gst/audio/gstaudiosrc.c
@@ -488,20 +488,15 @@ enum
   ARG_0,
 };
 
-#define _do_init(bla) \
+#define _do_init \
     GST_DEBUG_CATEGORY_INIT (gst_audio_src_debug, "audiosrc", 0, "audiosrc element");
-
-GST_BOILERPLATE_FULL (GstAudioSrc, gst_audio_src, GstBaseAudioSrc,
+#define gst_audio_src_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstAudioSrc, gst_audio_src,
     GST_TYPE_BASE_AUDIO_SRC, _do_init);
 
 static GstRingBuffer *gst_audio_src_create_ringbuffer (GstBaseAudioSrc * src);
 
 static void
-gst_audio_src_base_init (gpointer g_class)
-{
-}
-
-static void
 gst_audio_src_class_init (GstAudioSrcClass * klass)
 {
   GstBaseAudioSrcClass *gstbaseaudiosrc_class;
@@ -515,7 +510,7 @@ gst_audio_src_class_init (GstAudioSrcClass * klass)
 }
 
 static void
-gst_audio_src_init (GstAudioSrc * audiosrc, GstAudioSrcClass * g_class)
+gst_audio_src_init (GstAudioSrc * audiosrc)
 {
 }
 

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index da1bff0..69868fe 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -57,7 +57,6 @@ struct _GstBaseAudioSinkPrivate
 
   GstClockTime eos_time;
 
-  gboolean do_time_offset;
   /* number of microseconds we alow timestamps or clock slaving to drift
    * before resyncing */
   guint64 drift_tolerance;
@@ -119,10 +118,10 @@ gst_base_audio_sink_slave_method_get_type (void)
 }
 
 
-#define _do_init(bla) \
+#define _do_init \
     GST_DEBUG_CATEGORY_INIT (gst_base_audio_sink_debug, "baseaudiosink", 0, "baseaudiosink element");
-
-GST_BOILERPLATE_FULL (GstBaseAudioSink, gst_base_audio_sink, GstBaseSink,
+#define gst_base_audio_sink_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSink, gst_base_audio_sink,
     GST_TYPE_BASE_SINK, _do_init);
 
 static void gst_base_audio_sink_dispose (GObject * object);
@@ -167,11 +166,6 @@ static gboolean gst_base_audio_sink_query_pad (GstPad * pad, GstQuery * query);
 /* static guint gst_base_audio_sink_signals[LAST_SIGNAL] = { 0 }; */
 
 static void
-gst_base_audio_sink_base_init (gpointer g_class)
-{
-}
-
-static void
 gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
 {
   GObjectClass *gobject_class;
@@ -257,10 +251,8 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
 }
 
 static void
-gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
-    GstBaseAudioSinkClass * g_class)
+gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink)
 {
-  GstPluginFeature *feature;
   GstBaseSink *basesink;
 
   baseaudiosink->priv = GST_BASE_AUDIO_SINK_GET_PRIVATE (baseaudiosink);
@@ -283,25 +275,6 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
   /* install some custom pad_query functions */
   gst_pad_set_query_function (GST_BASE_SINK_PAD (baseaudiosink),
       GST_DEBUG_FUNCPTR (gst_base_audio_sink_query_pad));
-
-  baseaudiosink->priv->do_time_offset = TRUE;
-
-  /* check the factory, pulsesink < 0.10.17 does the timestamp offset itself so
-   * we should not do ourselves */
-  feature =
-      GST_PLUGIN_FEATURE_CAST (GST_ELEMENT_CLASS (g_class)->elementfactory);
-  GST_DEBUG ("created from factory %p", feature);
-
-  /* HACK for old pulsesink that did the time_offset themselves */
-  if (feature) {
-    if (strcmp (gst_plugin_feature_get_name (feature), "pulsesink") == 0) {
-      if (!gst_plugin_feature_check_version (feature, 0, 10, 17)) {
-        /* we're dealing with an old pulsesink, we need to disable time corection */
-        GST_DEBUG ("disable time offset");
-        baseaudiosink->priv->do_time_offset = FALSE;
-      }
-    }
-  }
 }
 
 static void
@@ -1585,20 +1558,18 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
       GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
 
   /* bring to position in the ringbuffer */
-  if (sink->priv->do_time_offset) {
-    time_offset =
-        GST_AUDIO_CLOCK_CAST (sink->provided_clock)->abidata.ABI.time_offset;
-    GST_DEBUG_OBJECT (sink,
-        "time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
-    if (render_start > time_offset)
-      render_start -= time_offset;
-    else
-      render_start = 0;
-    if (render_stop > time_offset)
-      render_stop -= time_offset;
-    else
-      render_stop = 0;
-  }
+  time_offset =
+      GST_AUDIO_CLOCK_CAST (sink->provided_clock)->abidata.ABI.time_offset;
+  GST_DEBUG_OBJECT (sink,
+      "time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
+  if (render_start > time_offset)
+    render_start -= time_offset;
+  else
+    render_start = 0;
+  if (render_stop > time_offset)
+    render_stop -= time_offset;
+  else
+    render_stop = 0;
 
   /* and bring the time to the rate corrected offset in the buffer */
   render_start = gst_util_uint64_scale_int (render_start,

diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.c b/gst-libs/gst/audio/gstbaseaudiosrc.c
index 8135e43..48ac2df 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosrc.c
+++ b/gst-libs/gst/audio/gstbaseaudiosrc.c
@@ -119,8 +119,9 @@ _do_init (GType type)
 #endif /* ENABLE_NLS */
 }
 
-GST_BOILERPLATE_FULL (GstBaseAudioSrc, gst_base_audio_src, GstPushSrc,
-    GST_TYPE_PUSH_SRC, _do_init);
+#define gst_base_audio_src_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstBaseAudioSrc, gst_base_audio_src, GST_TYPE_PUSH_SRC,
+    _do_init (g_define_type_id));
 
 static void gst_base_audio_src_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -149,11 +150,6 @@ static void gst_base_audio_src_fixate (GstBaseSrc * bsrc, GstCaps * caps);
 /* static guint gst_base_audio_src_signals[LAST_SIGNAL] = { 0 }; */
 
 static void
-gst_base_audio_src_base_init (gpointer g_class)
-{
-}
-
-static void
 gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
 {
   GObjectClass *gobject_class;
@@ -241,8 +237,7 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
 }
 
 static void
-gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
-    GstBaseAudioSrcClass * g_class)
+gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc)
 {
   baseaudiosrc->priv = GST_BASE_AUDIO_SRC_GET_PRIVATE (baseaudiosrc);