port some more elements to 0.11

diff --git a/gst/alpha/gstalpha.c b/gst/alpha/gstalpha.c
index 70c9a32..084f83a 100644 (file)
--- a/gst/alpha/gstalpha.c
+++ b/gst/alpha/gstalpha.c
@@ -193,7 +193,8 @@ static void gst_alpha_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 static void gst_alpha_finalize (GObject * object);
 
-GST_BOILERPLATE (GstAlpha, gst_alpha, GstVideoFilter, GST_TYPE_VIDEO_FILTER);
+#define gst_alpha_parent_class parent_class
+G_DEFINE_TYPE (GstAlpha, gst_alpha, GST_TYPE_VIDEO_FILTER);
 
 #define GST_TYPE_ALPHA_METHOD (gst_alpha_method_get_type())
 static GType
@@ -215,32 +216,15 @@ gst_alpha_method_get_type (void)
 }
 
 static void
-gst_alpha_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class, "Alpha filter",
-      "Filter/Effect/Video",
-      "Adds an alpha channel to video - uniform or via chroma-keying",
-      "Wim Taymans <wim@fluendo.com>\n"
-      "Edward Hervey <edward.hervey@collabora.co.uk>\n"
-      "Jan Schmidt <thaytan@noraisin.net>");
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_alpha_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_alpha_src_template));
-
-  GST_DEBUG_CATEGORY_INIT (gst_alpha_debug, "alpha", 0,
-      "alpha - Element for adding alpha channel to streams");
-}
-
-static void
 gst_alpha_class_init (GstAlphaClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstBaseTransformClass *btrans_class = (GstBaseTransformClass *) klass;
 
+  GST_DEBUG_CATEGORY_INIT (gst_alpha_debug, "alpha", 0,
+      "alpha - Element for adding alpha channel to streams");
+
   gobject_class->set_property = gst_alpha_set_property;
   gobject_class->get_property = gst_alpha_get_property;
   gobject_class->finalize = gst_alpha_finalize;
@@ -290,6 +274,18 @@ gst_alpha_class_init (GstAlphaClass * klass)
           DEFAULT_PREFER_PASSTHROUGH,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class, "Alpha filter",
+      "Filter/Effect/Video",
+      "Adds an alpha channel to video - uniform or via chroma-keying",
+      "Wim Taymans <wim.taymans@gmail.com>\n"
+      "Edward Hervey <edward.hervey@collabora.co.uk>\n"
+      "Jan Schmidt <thaytan@noraisin.net>");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_alpha_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_alpha_src_template));
+
   btrans_class->start = GST_DEBUG_FUNCPTR (gst_alpha_start);
   btrans_class->transform = GST_DEBUG_FUNCPTR (gst_alpha_transform);
   btrans_class->before_transform =
@@ -300,7 +296,7 @@ gst_alpha_class_init (GstAlphaClass * klass)
 }
 
 static void
-gst_alpha_init (GstAlpha * alpha, GstAlphaClass * klass)
+gst_alpha_init (GstAlpha * alpha)
 {
   alpha->alpha = DEFAULT_ALPHA;
   alpha->method = DEFAULT_METHOD;

diff --git a/gst/alpha/gstalphacolor.c b/gst/alpha/gstalphacolor.c
index f4e25eb..7a26b19 100644 (file)
--- a/gst/alpha/gstalphacolor.c
+++ b/gst/alpha/gstalphacolor.c
@@ -62,8 +62,7 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
         GST_VIDEO_CAPS_YUV ("AYUV"))
     );
 
-GST_BOILERPLATE (GstAlphaColor, gst_alpha_color, GstVideoFilter,
-    GST_TYPE_VIDEO_FILTER);
+G_DEFINE_TYPE (GstAlphaColor, gst_alpha_color, GST_TYPE_VIDEO_FILTER);
 
 static GstCaps *gst_alpha_color_transform_caps (GstBaseTransform * btrans,
     GstPadDirection direction, GstCaps * caps);
@@ -73,26 +72,24 @@ static GstFlowReturn gst_alpha_color_transform_ip (GstBaseTransform * btrans,
     GstBuffer * inbuf);
 
 static void
-gst_alpha_color_base_init (gpointer g_class)
+gst_alpha_color_class_init (GstAlphaColorClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
+  GstBaseTransformClass *gstbasetransform_class =
+      (GstBaseTransformClass *) klass;
+
+  GST_DEBUG_CATEGORY_INIT (alpha_color_debug, "alphacolor", 0,
+      "ARGB<->AYUV colorspace conversion preserving the alpha channels");
 
-  gst_element_class_set_details_simple (element_class, "Alpha color filter",
+  gst_element_class_set_details_simple (gstelement_class, "Alpha color filter",
       "Filter/Converter/Video",
       "ARGB from/to AYUV colorspace conversion preserving the alpha channel",
       "Wim Taymans <wim@fluendo.com>");
 
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&sink_template));
-  gst_element_class_add_pad_template (element_class,
+  gst_element_class_add_pad_template (gstelement_class,
       gst_static_pad_template_get (&src_template));
-}
-
-static void
-gst_alpha_color_class_init (GstAlphaColorClass * klass)
-{
-  GstBaseTransformClass *gstbasetransform_class =
-      (GstBaseTransformClass *) klass;
 
   gstbasetransform_class->transform_caps =
       GST_DEBUG_FUNCPTR (gst_alpha_color_transform_caps);
@@ -101,12 +98,10 @@ gst_alpha_color_class_init (GstAlphaColorClass * klass)
   gstbasetransform_class->transform_ip =
       GST_DEBUG_FUNCPTR (gst_alpha_color_transform_ip);
 
-  GST_DEBUG_CATEGORY_INIT (alpha_color_debug, "alphacolor", 0,
-      "ARGB<->AYUV colorspace conversion preserving the alpha channels");
 }
 
 static void
-gst_alpha_color_init (GstAlphaColor * alpha, GstAlphaColorClass * g_class)
+gst_alpha_color_init (GstAlphaColor * alpha)
 {
   GstBaseTransform *btrans = GST_BASE_TRANSFORM (alpha);
 

diff --git a/gst/apetag/gstapedemux.c b/gst/apetag/gstapedemux.c
index d243c89..163a487 100644 (file)
--- a/gst/apetag/gstapedemux.c
+++ b/gst/apetag/gstapedemux.c
@@ -72,31 +72,27 @@ static GstTagDemuxResult gst_ape_demux_parse_tag (GstTagDemux * demux,
     GstBuffer * buffer, gboolean start_tag, guint * tag_size,
     GstTagList ** tags);
 
-GST_BOILERPLATE (GstApeDemux, gst_ape_demux, GstTagDemux, GST_TYPE_TAG_DEMUX);
+G_DEFINE_TYPE (GstApeDemux, gst_ape_demux, GST_TYPE_TAG_DEMUX);
 
 static void
-gst_ape_demux_base_init (gpointer klass)
+gst_ape_demux_class_init (GstApeDemuxClass * klass)
 {
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementClass *element_class;
+  GstTagDemuxClass *tagdemux_class;
 
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_factory));
+  GST_DEBUG_CATEGORY_INIT (apedemux_debug, "apedemux", 0,
+      "GStreamer APE tag demuxer");
+
+  tagdemux_class = GST_TAG_DEMUX_CLASS (klass);
+  element_class = GST_ELEMENT_CLASS (klass);
 
   gst_element_class_set_details_simple (element_class, "APE tag demuxer",
       "Codec/Demuxer/Metadata",
       "Read and output APE tags while demuxing the contents",
       "Tim-Philipp Müller <tim centricular net>");
 
-  GST_DEBUG_CATEGORY_INIT (apedemux_debug, "apedemux", 0,
-      "GStreamer APE tag demuxer");
-}
-
-static void
-gst_ape_demux_class_init (GstApeDemuxClass * klass)
-{
-  GstTagDemuxClass *tagdemux_class;
-
-  tagdemux_class = GST_TAG_DEMUX_CLASS (klass);
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_factory));
 
   tagdemux_class->identify_tag = GST_DEBUG_FUNCPTR (gst_ape_demux_identify_tag);
   tagdemux_class->parse_tag = GST_DEBUG_FUNCPTR (gst_ape_demux_parse_tag);
@@ -110,7 +106,7 @@ gst_ape_demux_class_init (GstApeDemuxClass * klass)
 }
 
 static void
-gst_ape_demux_init (GstApeDemux * apedemux, GstApeDemuxClass * gclass)
+gst_ape_demux_init (GstApeDemux * apedemux)
 {
   /* nothing to do here */
 }

diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c
index 5316c96..02db5fb 100644 (file)
--- a/gst/audiofx/audioamplify.c
+++ b/gst/audiofx/audioamplify.c
@@ -125,11 +125,7 @@ gst_audio_amplify_clipping_method_get_type (void)
     " rate=(int)[1,MAX],"                                             \
     " channels=(int)[1,MAX]"
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0, "audioamplify element");
-
-GST_BOILERPLATE_FULL (GstAudioAmplify, gst_audio_amplify, GstAudioFilter,
-    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+G_DEFINE_TYPE (GstAudioAmplify, gst_audio_amplify, GST_TYPE_AUDIO_FILTER);
 
 static gboolean gst_audio_amplify_set_process_function (GstAudioAmplify *
     filter, gint clipping, gint format, gint width);
@@ -279,28 +275,18 @@ MAKE_FLOAT_FUNCS (gdouble)
 /* GObject vmethod implementations */
 
 static void
-gst_audio_amplify_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstCaps *caps;
-
-  gst_element_class_set_details_simple (element_class, "Audio amplifier",
-      "Filter/Effect/Audio",
-      "Amplifies an audio stream by a given factor",
-      "Sebastian Dröge <slomo@circular-chaos.org>");
-
-  caps = gst_caps_from_string (ALLOWED_CAPS);
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
-      caps);
-  gst_caps_unref (caps);
-}
-
-static void
 gst_audio_amplify_class_init (GstAudioAmplifyClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstCaps *caps;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_amplify_debug, "audioamplify", 0,
+      "audioamplify element");
 
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
   gobject_class->set_property = gst_audio_amplify_set_property;
   gobject_class->get_property = gst_audio_amplify_get_property;
 
@@ -324,6 +310,16 @@ gst_audio_amplify_class_init (GstAudioAmplifyClass * klass)
           GST_TYPE_AUDIO_AMPLIFY_CLIPPING_METHOD, METHOD_CLIP,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class, "Audio amplifier",
+      "Filter/Effect/Audio",
+      "Amplifies an audio stream by a given factor",
+      "Sebastian Dröge <slomo@circular-chaos.org>");
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+
   GST_AUDIO_FILTER_CLASS (klass)->setup =
       GST_DEBUG_FUNCPTR (gst_audio_amplify_setup);
   GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
@@ -331,7 +327,7 @@ gst_audio_amplify_class_init (GstAudioAmplifyClass * klass)
 }
 
 static void
-gst_audio_amplify_init (GstAudioAmplify * filter, GstAudioAmplifyClass * klass)
+gst_audio_amplify_init (GstAudioAmplify * filter)
 {
   filter->amplification = 1.0;
   gst_audio_amplify_set_process_function (filter, METHOD_CLIP,
@@ -480,6 +476,8 @@ gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   GstAudioAmplify *filter = GST_AUDIO_AMPLIFY (base);
   guint num_samples;
   GstClockTime timestamp, stream_time;
+  guint8 *data;
+  gsize size;
 
   timestamp = GST_BUFFER_TIMESTAMP (buf);
   stream_time =
@@ -491,14 +489,16 @@ gst_audio_amplify_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   if (GST_CLOCK_TIME_IS_VALID (stream_time))
     gst_object_sync_values (G_OBJECT (filter), stream_time);
 
-  num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
-
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
     return GST_FLOW_OK;
 
-  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
+  num_samples = size / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  filter->process (filter, data, num_samples);
+
+  gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }

diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c
index 89a950c..95debde 100644 (file)
--- a/gst/audiofx/audiochebband.c
+++ b/gst/audiofx/audiochebband.c
@@ -96,11 +96,9 @@ enum
   PROP_POLES
 };
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0, "audiochebband element");
-
-GST_BOILERPLATE_FULL (GstAudioChebBand, gst_audio_cheb_band,
-    GstAudioFXBaseIIRFilter, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
+#define gst_audio_cheb_band_parent_class parent_class
+G_DEFINE_TYPE (GstAudioChebBand, gst_audio_cheb_band,
+    GST_TYPE_AUDIO_FX_BASE_IIR_FILTER);
 
 static void gst_audio_cheb_band_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec);
@@ -140,22 +138,15 @@ gst_audio_cheb_band_mode_get_type (void)
 /* GObject vmethod implementations */
 
 static void
-gst_audio_cheb_band_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_set_details_simple (element_class,
-      "Band pass & band reject filter", "Filter/Effect/Audio",
-      "Chebyshev band pass and band reject filter",
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
-
-static void
 gst_audio_cheb_band_class_init (GstAudioChebBandClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
+  GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0,
+      "audiochebband element");
+
   gobject_class->set_property = gst_audio_cheb_band_set_property;
   gobject_class->get_property = gst_audio_cheb_band_get_property;
   gobject_class->finalize = gst_audio_cheb_band_finalize;
@@ -193,12 +184,16 @@ gst_audio_cheb_band_class_init (GstAudioChebBandClass * klass)
           4, 32, 4,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class,
+      "Band pass & band reject filter", "Filter/Effect/Audio",
+      "Chebyshev band pass and band reject filter",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
   filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_setup);
 }
 
 static void
-gst_audio_cheb_band_init (GstAudioChebBand * filter,
-    GstAudioChebBandClass * klass)
+gst_audio_cheb_band_init (GstAudioChebBand * filter)
 {
   filter->lower_frequency = filter->upper_frequency = 0.0;
   filter->mode = MODE_BAND_PASS;

diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c
index 8d51fbb..bf50672 100644 (file)
--- a/gst/audiofx/audiocheblimit.c
+++ b/gst/audiofx/audiocheblimit.c
@@ -91,12 +91,9 @@ enum
   PROP_POLES
 };
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element");
-
-GST_BOILERPLATE_FULL (GstAudioChebLimit,
-    gst_audio_cheb_limit, GstAudioFXBaseIIRFilter,
-    GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
+#define gst_audio_cheb_limit_parent_class parent_class
+G_DEFINE_TYPE (GstAudioChebLimit,
+    gst_audio_cheb_limit, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER);
 
 static void gst_audio_cheb_limit_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec);
@@ -136,23 +133,15 @@ gst_audio_cheb_limit_mode_get_type (void)
 /* GObject vmethod implementations */
 
 static void
-gst_audio_cheb_limit_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_set_details_simple (element_class,
-      "Low pass & high pass filter",
-      "Filter/Effect/Audio",
-      "Chebyshev low pass and high pass filter",
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
-
-static void
 gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
+  GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0,
+      "audiocheblimit element");
+
   gobject_class->set_property = gst_audio_cheb_limit_set_property;
   gobject_class->get_property = gst_audio_cheb_limit_get_property;
   gobject_class->finalize = gst_audio_cheb_limit_finalize;
@@ -186,12 +175,17 @@ gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
           2, 32, 4,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class,
+      "Low pass & high pass filter",
+      "Filter/Effect/Audio",
+      "Chebyshev low pass and high pass filter",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
   filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup);
 }
 
 static void
-gst_audio_cheb_limit_init (GstAudioChebLimit * filter,
-    GstAudioChebLimitClass * klass)
+gst_audio_cheb_limit_init (GstAudioChebLimit * filter)
 {
   filter->cutoff = 0.0;
   filter->mode = MODE_LOW_PASS;

diff --git a/gst/audiofx/audiodynamic.c b/gst/audiofx/audiodynamic.c
index 33871d2..52b031a 100644 (file)
--- a/gst/audiofx/audiodynamic.c
+++ b/gst/audiofx/audiodynamic.c
@@ -83,11 +83,7 @@ enum
     " rate=(int)[1,MAX],"                                             \
     " channels=(int)[1,MAX]"
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0, "audiodynamic element");
-
-GST_BOILERPLATE_FULL (GstAudioDynamic, gst_audio_dynamic, GstAudioFilter,
-    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+G_DEFINE_TYPE (GstAudioDynamic, gst_audio_dynamic, GST_TYPE_AUDIO_FILTER);
 
 static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -213,27 +209,18 @@ gst_audio_dynamic_set_process_function (GstAudioDynamic * filter)
 /* GObject vmethod implementations */
 
 static void
-gst_audio_dynamic_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstCaps *caps;
-
-  gst_element_class_set_details_simple (element_class,
-      "Dynamic range controller", "Filter/Effect/Audio",
-      "Compressor and Expander", "Sebastian Dröge <slomo@circular-chaos.org>");
-
-  caps = gst_caps_from_string (ALLOWED_CAPS);
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
-      caps);
-  gst_caps_unref (caps);
-}
-
-static void
 gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstCaps *caps;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0,
+      "audiodynamic element");
 
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
   gobject_class->set_property = gst_audio_dynamic_set_property;
   gobject_class->get_property = gst_audio_dynamic_get_property;
 
@@ -263,6 +250,15 @@ gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
           1.0,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class,
+      "Dynamic range controller", "Filter/Effect/Audio",
+      "Compressor and Expander", "Sebastian Dröge <slomo@circular-chaos.org>");
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+
   GST_AUDIO_FILTER_CLASS (klass)->setup =
       GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);
   GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
@@ -270,7 +266,7 @@ gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
 }
 
 static void
-gst_audio_dynamic_init (GstAudioDynamic * filter, GstAudioDynamicClass * klass)
+gst_audio_dynamic_init (GstAudioDynamic * filter)
 {
   filter->ratio = 1.0;
   filter->threshold = 0.0;
@@ -696,6 +692,8 @@ gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
   guint num_samples;
   GstClockTime timestamp, stream_time;
+  guint8 *data;
+  gsize size;
 
   timestamp = GST_BUFFER_TIMESTAMP (buf);
   stream_time =
@@ -707,14 +705,16 @@ gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   if (GST_CLOCK_TIME_IS_VALID (stream_time))
     gst_object_sync_values (G_OBJECT (filter), stream_time);
 
-  num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
-
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
     return GST_FLOW_OK;
 
-  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
+  num_samples = size / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  filter->process (filter, data, num_samples);
+
+  gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }

diff --git a/gst/audiofx/audioecho.c b/gst/audiofx/audioecho.c
index cf4dab1..557e3ff 100644 (file)
--- a/gst/audiofx/audioecho.c
+++ b/gst/audiofx/audioecho.c
@@ -73,11 +73,8 @@ enum
     " rate=(int)[1,MAX],"                                             \
     " channels=(int)[1,MAX]"
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0, "audioecho element");
-
-GST_BOILERPLATE_FULL (GstAudioEcho, gst_audio_echo, GstAudioFilter,
-    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+#define gst_audio_echo_parent_class parent_class
+G_DEFINE_TYPE (GstAudioEcho, gst_audio_echo, GST_TYPE_AUDIO_FILTER);
 
 static void gst_audio_echo_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -99,28 +96,16 @@ static void gst_audio_echo_transform_double (GstAudioEcho * self,
 /* GObject vmethod implementations */
 
 static void
-gst_audio_echo_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstCaps *caps;
-
-  gst_element_class_set_details_simple (element_class, "Audio echo",
-      "Filter/Effect/Audio",
-      "Adds an echo or reverb effect to an audio stream",
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-
-  caps = gst_caps_from_string (ALLOWED_CAPS);
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
-      caps);
-  gst_caps_unref (caps);
-}
-
-static void
 gst_audio_echo_class_init (GstAudioEchoClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass;
   GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass;
+  GstCaps *caps;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0,
+      "audioecho element");
 
   gobject_class->set_property = gst_audio_echo_set_property;
   gobject_class->get_property = gst_audio_echo_get_property;
@@ -151,6 +136,16 @@ gst_audio_echo_class_init (GstAudioEchoClass * klass)
           0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
           | GST_PARAM_CONTROLLABLE));
 
+  gst_element_class_set_details_simple (gstelement_class, "Audio echo",
+      "Filter/Effect/Audio",
+      "Adds an echo or reverb effect to an audio stream",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+
   audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup);
   basetransform_class->transform_ip =
       GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip);
@@ -158,7 +153,7 @@ gst_audio_echo_class_init (GstAudioEchoClass * klass)
 }
 
 static void
-gst_audio_echo_init (GstAudioEcho * self, GstAudioEchoClass * klass)
+gst_audio_echo_init (GstAudioEcho * self)
 {
   self->delay = 1;
   self->max_delay = 1;
@@ -359,6 +354,8 @@ gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   GstAudioEcho *self = GST_AUDIO_ECHO (base);
   guint num_samples;
   GstClockTime timestamp, stream_time;
+  guint8 *data;
+  gsize size;
 
   timestamp = GST_BUFFER_TIMESTAMP (buf);
   stream_time =
@@ -370,9 +367,6 @@ gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   if (GST_CLOCK_TIME_IS_VALID (stream_time))
     gst_object_sync_values (G_OBJECT (self), stream_time);
 
-  num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8);
-
   if (self->buffer == NULL) {
     guint width, rate, channels;
 
@@ -395,7 +389,12 @@ gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf)
     }
   }
 
-  self->process (self, GST_BUFFER_DATA (buf), num_samples);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
+  num_samples = size / (GST_AUDIO_FILTER (self)->format.width / 8);
+
+  self->process (self, data, num_samples);
+
+  gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }

diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c
index b6102ae..3b3dac1 100644 (file)
--- a/gst/audiofx/audiofirfilter.c
+++ b/gst/audiofx/audiofirfilter.c
@@ -75,12 +75,9 @@ enum
 
 static guint gst_audio_fir_filter_signals[LAST_SIGNAL] = { 0, };
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_fir_filter_debug, "audiofirfilter", 0, \
-      "Generic audio FIR filter plugin");
-
-GST_BOILERPLATE_FULL (GstAudioFIRFilter, gst_audio_fir_filter, GstAudioFilter,
-    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
+#define gst_audio_fir_filter_parent_class parent_class
+G_DEFINE_TYPE (GstAudioFIRFilter, gst_audio_fir_filter,
+    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER);
 
 static void gst_audio_fir_filter_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -91,24 +88,17 @@ static void gst_audio_fir_filter_finalize (GObject * object);
 static gboolean gst_audio_fir_filter_setup (GstAudioFilter * base,
     GstRingBufferSpec * format);
 
-/* Element class */
-static void
-gst_audio_fir_filter_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class,
-      "Audio FIR filter", "Filter/Effect/Audio",
-      "Generic audio FIR filter with custom filter kernel",
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
 
 static void
 gst_audio_fir_filter_class_init (GstAudioFIRFilterClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
+  GST_DEBUG_CATEGORY_INIT (gst_audio_fir_filter_debug, "audiofirfilter", 0,
+      "Generic audio FIR filter plugin");
+
   gobject_class->set_property = gst_audio_fir_filter_set_property;
   gobject_class->get_property = gst_audio_fir_filter_get_property;
   gobject_class->finalize = gst_audio_fir_filter_finalize;
@@ -140,6 +130,11 @@ gst_audio_fir_filter_class_init (GstAudioFIRFilterClass * klass)
       g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass),
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioFIRFilterClass, rate_changed),
       NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
+
+  gst_element_class_set_details_simple (gstelement_class,
+      "Audio FIR filter", "Filter/Effect/Audio",
+      "Generic audio FIR filter with custom filter kernel",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
 }
 
 static void
@@ -167,8 +162,7 @@ gst_audio_fir_filter_update_kernel (GstAudioFIRFilter * self, GValueArray * va)
 }
 
 static void
-gst_audio_fir_filter_init (GstAudioFIRFilter * self,
-    GstAudioFIRFilterClass * g_class)
+gst_audio_fir_filter_init (GstAudioFIRFilter * self)
 {
   GValue v = { 0, };
   GValueArray *va;

diff --git a/gst/audiofx/audiofxbasefirfilter.c b/gst/audiofx/audiofxbasefirfilter.c
index ac1b148..d74c0a6 100644 (file)
--- a/gst/audiofx/audiofxbasefirfilter.c
+++ b/gst/audiofx/audiofxbasefirfilter.c
@@ -49,10 +49,6 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
     " rate = (int) [ 1, MAX ], "                                      \
     " channels = (int) [ 1, MAX ]"
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_fir_filter_debug, "audiofxbasefirfilter", 0, \
-      "FIR filter base class");
-
 /* Switch from time-domain to FFT convolution for kernels >= this */
 #define FFT_THRESHOLD 32
 
@@ -66,8 +62,9 @@ enum
 #define DEFAULT_LOW_LATENCY FALSE
 #define DEFAULT_DRAIN_ON_CHANGES TRUE
 
-GST_BOILERPLATE_FULL (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
-    GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+#define gst_audio_fx_base_fir_filter_parent_class parent_class
+G_DEFINE_TYPE (GstAudioFXBaseFIRFilter, gst_audio_fx_base_fir_filter,
+    GST_TYPE_AUDIO_FILTER);
 
 static GstFlowReturn gst_audio_fx_base_fir_filter_transform (GstBaseTransform *
     base, GstBuffer * inbuf, GstBuffer * outbuf);
@@ -76,8 +73,8 @@ static gboolean gst_audio_fx_base_fir_filter_stop (GstBaseTransform * base);
 static gboolean gst_audio_fx_base_fir_filter_event (GstBaseTransform * base,
     GstEvent * event);
 static gboolean gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform *
-    base, GstPadDirection direction, GstCaps * caps, guint size,
-    GstCaps * othercaps, guint * othersize);
+    base, GstPadDirection direction, GstCaps * caps, gsize size,
+    GstCaps * othercaps, gsize * othersize);
 static gboolean gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
     GstRingBufferSpec * format);
 
@@ -541,22 +538,15 @@ gst_audio_fx_base_fir_filter_get_property (GObject * object, guint prop_id,
 }
 
 static void
-gst_audio_fx_base_fir_filter_base_init (gpointer g_class)
-{
-  GstCaps *caps;
-
-  caps = gst_caps_from_string (ALLOWED_CAPS);
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
-      caps);
-  gst_caps_unref (caps);
-}
-
-static void
 gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
   GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
+  GstCaps *caps;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_fir_filter_debug,
+      "audiofxbasefirfilter", 0, "FIR filter base class");
 
   gobject_class->dispose = gst_audio_fx_base_fir_filter_dispose;
   gobject_class->set_property = gst_audio_fx_base_fir_filter_set_property;
@@ -593,6 +583,11 @@ gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
           DEFAULT_DRAIN_ON_CHANGES,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+
   trans_class->transform =
       GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_transform);
   trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_fx_base_fir_filter_start);
@@ -604,8 +599,7 @@ gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
 }
 
 static void
-gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self,
-    GstAudioFXBaseFIRFilterClass * g_class)
+gst_audio_fx_base_fir_filter_init (GstAudioFXBaseFIRFilter * self)
 {
   self->kernel = NULL;
   self->buffer = NULL;
@@ -634,7 +628,8 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
   gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
   gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
   gint outsize, outsamples;
-  guint8 *in, *out;
+  guint8 *in, *out, *data;
+  gsize size;
 
   if (channels == 0 || rate == 0 || self->nsamples_in == 0) {
     self->buffer_fill = 0;
@@ -684,15 +679,16 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
 
     /* Convolve the residue with zeros to get the actual remaining data */
     in = g_new0 (guint8, outsize);
-    self->nsamples_out +=
-        self->process (self, in, GST_BUFFER_DATA (outbuf), outsamples);
+    data = gst_buffer_map (outbuf, &size, NULL, GST_MAP_READWRITE);
+    self->nsamples_out += self->process (self, in, data, outsamples);
+    gst_buffer_unmap (outbuf, data, size);
+
     g_free (in);
   } else {
     guint gensamples = 0;
-    guint8 *data;
 
     outbuf = gst_buffer_new_and_alloc (outsize);
-    data = GST_BUFFER_DATA (outbuf);
+    data = gst_buffer_map (outbuf, &size, NULL, GST_MAP_READWRITE);
 
     while (gensamples < outsamples) {
       guint step_insamples = self->block_length - self->buffer_fill;
@@ -710,6 +706,8 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
       g_free (out);
     }
     self->nsamples_out += gensamples;
+
+    gst_buffer_unmap (outbuf, data, size);
   }
 
   /* Set timestamp, offset, etc from the values we
@@ -734,7 +732,8 @@ gst_audio_fx_base_fir_filter_push_residue (GstAudioFXBaseFIRFilter * self)
   GST_DEBUG_OBJECT (self, "Pushing residue buffer of size %d with timestamp: %"
       GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
       G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
-      GST_BUFFER_SIZE (outbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+      gst_buffer_get_size (outbuf),
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
       GST_BUFFER_OFFSET_END (outbuf), outsamples);
 
@@ -778,8 +777,8 @@ gst_audio_fx_base_fir_filter_setup (GstAudioFilter * base,
 
 static gboolean
 gst_audio_fx_base_fir_filter_transform_size (GstBaseTransform * base,
-    GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
-    guint * othersize)
+    GstPadDirection direction, GstCaps * caps, gsize size, GstCaps * othercaps,
+    gsize * othersize)
 {
   GstAudioFXBaseFIRFilter *self = GST_AUDIO_FX_BASE_FIR_FILTER (base);
   guint blocklen;
@@ -817,8 +816,10 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
   gint channels = GST_AUDIO_FILTER_CAST (self)->format.channels;
   gint rate = GST_AUDIO_FILTER_CAST (self)->format.rate;
   gint width = GST_AUDIO_FILTER_CAST (self)->format.width / 8;
-  guint input_samples = (GST_BUFFER_SIZE (inbuf) / width) / channels;
-  guint output_samples = (GST_BUFFER_SIZE (outbuf) / width) / channels;
+  guint8 *indata, *outdata;
+  gsize insize, outsize;
+  guint input_samples;
+  guint output_samples;
   guint generated_samples;
   guint64 output_offset;
   gint64 diff = 0;
@@ -871,11 +872,18 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
     self->start_off = GST_BUFFER_OFFSET (inbuf);
   }
 
+  indata = gst_buffer_map (inbuf, &insize, NULL, GST_MAP_READ);
+  outdata = gst_buffer_map (outbuf, &outsize, NULL, GST_MAP_WRITE);
+
+  input_samples = (insize / width) / channels;
+  output_samples = (outsize / width) / channels;
+
   self->nsamples_in += input_samples;
 
-  generated_samples =
-      self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
-      input_samples);
+  generated_samples = self->process (self, indata, outdata, input_samples);
+
+  gst_buffer_unmap (inbuf, indata, insize);
+  gst_buffer_unmap (outbuf, outdata, outsize);
 
   g_assert (generated_samples <= output_samples);
   self->nsamples_out += generated_samples;
@@ -891,9 +899,9 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
     gint64 tmp = diff;
     diff = generated_samples - diff;
     generated_samples = tmp;
-    GST_BUFFER_DATA (outbuf) += diff * width * channels;
   }
-  GST_BUFFER_SIZE (outbuf) = generated_samples * width * channels;
+  gst_buffer_resize (outbuf, diff * width * channels,
+      generated_samples * width * channels);
 
   output_offset = self->nsamples_out - self->latency - generated_samples;
   GST_BUFFER_TIMESTAMP (outbuf) =
@@ -913,7 +921,8 @@ gst_audio_fx_base_fir_filter_transform (GstBaseTransform * base,
   GST_DEBUG_OBJECT (self, "Pushing buffer of size %d with timestamp: %"
       GST_TIME_FORMAT ", duration: %" GST_TIME_FORMAT ", offset: %"
       G_GUINT64_FORMAT ", offset_end: %" G_GUINT64_FORMAT ", nsamples_out: %d",
-      GST_BUFFER_SIZE (outbuf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+      gst_buffer_get_size (outbuf),
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
       GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), GST_BUFFER_OFFSET (outbuf),
       GST_BUFFER_OFFSET_END (outbuf), generated_samples);
 

diff --git a/gst/audiofx/audiofxbaseiirfilter.c b/gst/audiofx/audiofxbaseiirfilter.c
index 0b0fc34..939d102 100644 (file)
--- a/gst/audiofx/audiofxbaseiirfilter.c
+++ b/gst/audiofx/audiofxbaseiirfilter.c
@@ -42,12 +42,9 @@ GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
     " rate = (int) [ 1, MAX ],"                                       \
     " channels = (int) [ 1, MAX ]"
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug, "audiofxbaseiirfilter", 0, "Audio IIR Filter Base Class");
-
-GST_BOILERPLATE_FULL (GstAudioFXBaseIIRFilter,
-    gst_audio_fx_base_iir_filter, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
-    DEBUG_INIT);
+#define gst_audio_fx_base_iir_filter_parent_class parent_class
+G_DEFINE_TYPE (GstAudioFXBaseIIRFilter,
+    gst_audio_fx_base_iir_filter, GST_TYPE_AUDIO_FILTER);
 
 static gboolean gst_audio_fx_base_iir_filter_setup (GstAudioFilter * filter,
     GstRingBufferSpec * format);
@@ -64,17 +61,6 @@ static void process_32 (GstAudioFXBaseIIRFilter * filter,
 /* GObject vmethod implementations */
 
 static void
-gst_audio_fx_base_iir_filter_base_init (gpointer klass)
-{
-  GstCaps *caps;
-
-  caps = gst_caps_from_string (ALLOWED_CAPS);
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
-      caps);
-  gst_caps_unref (caps);
-}
-
-static void
 gst_audio_fx_base_iir_filter_dispose (GObject * object)
 {
   GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (object);
@@ -112,9 +98,18 @@ gst_audio_fx_base_iir_filter_class_init (GstAudioFXBaseIIRFilterClass * klass)
   GObjectClass *gobject_class = (GObjectClass *) klass;
   GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
+  GstCaps *caps;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_fx_base_iir_filter_debug,
+      "audiofxbaseiirfilter", 0, "Audio IIR Filter Base Class");
 
   gobject_class->dispose = gst_audio_fx_base_iir_filter_dispose;
 
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+
   filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_fx_base_iir_filter_setup);
 
   trans_class->transform_ip =
@@ -123,8 +118,7 @@ gst_audio_fx_base_iir_filter_class_init (GstAudioFXBaseIIRFilterClass * klass)
 }
 
 static void
-gst_audio_fx_base_iir_filter_init (GstAudioFXBaseIIRFilter * filter,
-    GstAudioFXBaseIIRFilterClass * klass)
+gst_audio_fx_base_iir_filter_init (GstAudioFXBaseIIRFilter * filter)
 {
   gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
 
@@ -357,6 +351,8 @@ gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base,
   GstAudioFXBaseIIRFilter *filter = GST_AUDIO_FX_BASE_IIR_FILTER (base);
   guint num_samples;
   GstClockTime timestamp, stream_time;
+  guint8 *data;
+  gsize size;
 
   timestamp = GST_BUFFER_TIMESTAMP (buf);
   stream_time =
@@ -368,15 +364,17 @@ gst_audio_fx_base_iir_filter_transform_ip (GstBaseTransform * base,
   if (GST_CLOCK_TIME_IS_VALID (stream_time))
     gst_object_sync_values (G_OBJECT (filter), stream_time);
 
-  num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
-
   if (gst_base_transform_is_passthrough (base))
     return GST_FLOW_OK;
 
   g_return_val_if_fail (filter->a != NULL, GST_FLOW_ERROR);
 
-  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
+  num_samples = size / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  filter->process (filter, data, num_samples);
+
+  gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }

diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c
index bac16b2..776bdb1 100644 (file)
--- a/gst/audiofx/audioiirfilter.c
+++ b/gst/audiofx/audioiirfilter.c
@@ -71,12 +71,9 @@ enum
 
 static guint gst_audio_iir_filter_signals[LAST_SIGNAL] = { 0, };
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_iir_filter_debug, "audioiirfilter", 0, \
-      "Generic audio IIR filter plugin");
-
-GST_BOILERPLATE_FULL (GstAudioIIRFilter, gst_audio_iir_filter, GstAudioFilter,
-    GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
+#define gst_audio_iir_filter_parent_class parent_class
+G_DEFINE_TYPE (GstAudioIIRFilter, gst_audio_iir_filter,
+    GST_TYPE_AUDIO_FX_BASE_IIR_FILTER);
 
 static void gst_audio_iir_filter_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -87,24 +84,16 @@ static void gst_audio_iir_filter_finalize (GObject * object);
 static gboolean gst_audio_iir_filter_setup (GstAudioFilter * base,
     GstRingBufferSpec * format);
 
-/* Element class */
-static void
-gst_audio_iir_filter_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class,
-      "Audio IIR filter", "Filter/Effect/Audio",
-      "Generic audio IIR filter with custom filter kernel",
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
-
 static void
 gst_audio_iir_filter_class_init (GstAudioIIRFilterClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
+  GST_DEBUG_CATEGORY_INIT (gst_audio_iir_filter_debug, "audioiirfilter", 0,
+      "Generic audio IIR filter plugin");
+
   gobject_class->set_property = gst_audio_iir_filter_set_property;
   gobject_class->get_property = gst_audio_iir_filter_get_property;
   gobject_class->finalize = gst_audio_iir_filter_finalize;
@@ -139,6 +128,11 @@ gst_audio_iir_filter_class_init (GstAudioIIRFilterClass * klass)
       g_signal_new ("rate-changed", G_TYPE_FROM_CLASS (klass),
       G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstAudioIIRFilterClass, rate_changed),
       NULL, NULL, gst_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
+
+  gst_element_class_set_details_simple (gstelement_class,
+      "Audio IIR filter", "Filter/Effect/Audio",
+      "Generic audio IIR filter with custom filter kernel",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
 }
 
 static void
@@ -184,8 +178,7 @@ gst_audio_iir_filter_update_coefficients (GstAudioIIRFilter * self,
 }
 
 static void
-gst_audio_iir_filter_init (GstAudioIIRFilter * self,
-    GstAudioIIRFilterClass * g_class)
+gst_audio_iir_filter_init (GstAudioIIRFilter * self)
 {
   GValue v = { 0, };
   GValueArray *a, *b;

diff --git a/gst/audiofx/audioinvert.c b/gst/audiofx/audioinvert.c
index c0a7da4..804b31b 100644 (file)
--- a/gst/audiofx/audioinvert.c
+++ b/gst/audiofx/audioinvert.c
@@ -78,11 +78,7 @@ enum
     " rate=(int)[1,MAX],"                                             \
     " channels=(int)[1,MAX]"
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0, "audioinvert element");
-
-GST_BOILERPLATE_FULL (GstAudioInvert, gst_audio_invert, GstAudioFilter,
-    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+G_DEFINE_TYPE (GstAudioInvert, gst_audio_invert, GST_TYPE_AUDIO_FILTER);
 
 static void gst_audio_invert_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -102,28 +98,18 @@ static void gst_audio_invert_transform_float (GstAudioInvert * filter,
 /* GObject vmethod implementations */
 
 static void
-gst_audio_invert_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstCaps *caps;
-
-  gst_element_class_set_details_simple (element_class, "Audio inversion",
-      "Filter/Effect/Audio",
-      "Swaps upper and lower half of audio samples",
-      "Sebastian Dröge <slomo@circular-chaos.org>");
-
-  caps = gst_caps_from_string (ALLOWED_CAPS);
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
-      caps);
-  gst_caps_unref (caps);
-}
-
-static void
 gst_audio_invert_class_init (GstAudioInvertClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstCaps *caps;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0,
+      "audioinvert element");
 
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
   gobject_class->set_property = gst_audio_invert_set_property;
   gobject_class->get_property = gst_audio_invert_get_property;
 
@@ -133,6 +119,16 @@ gst_audio_invert_class_init (GstAudioInvertClass * klass)
           0.0,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class, "Audio inversion",
+      "Filter/Effect/Audio",
+      "Swaps upper and lower half of audio samples",
+      "Sebastian Dröge <slomo@circular-chaos.org>");
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+
   GST_AUDIO_FILTER_CLASS (klass)->setup =
       GST_DEBUG_FUNCPTR (gst_audio_invert_setup);
   GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
@@ -140,7 +136,7 @@ gst_audio_invert_class_init (GstAudioInvertClass * klass)
 }
 
 static void
-gst_audio_invert_init (GstAudioInvert * filter, GstAudioInvertClass * klass)
+gst_audio_invert_init (GstAudioInvert * filter)
 {
   filter->degree = 0.0;
   gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
@@ -236,6 +232,8 @@ gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   GstAudioInvert *filter = GST_AUDIO_INVERT (base);
   guint num_samples;
   GstClockTime timestamp, stream_time;
+  guint8 *data;
+  gsize size;
 
   timestamp = GST_BUFFER_TIMESTAMP (buf);
   stream_time =
@@ -247,14 +245,16 @@ gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   if (GST_CLOCK_TIME_IS_VALID (stream_time))
     gst_object_sync_values (G_OBJECT (filter), stream_time);
 
-  num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
-
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
     return GST_FLOW_OK;
 
-  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
+  num_samples = size / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  filter->process (filter, data, num_samples);
+
+  gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }

diff --git a/gst/audiofx/audiokaraoke.c b/gst/audiofx/audiokaraoke.c
index c6fb93e..bc2a786 100644 (file)
--- a/gst/audiofx/audiokaraoke.c
+++ b/gst/audiofx/audiokaraoke.c
@@ -85,11 +85,7 @@ enum
     " rate=(int)[1,MAX],"                                             \
     " channels=(int)[1,MAX]"
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_karaoke_debug, "audiokaraoke", 0, "audiokaraoke element");
-
-GST_BOILERPLATE_FULL (GstAudioKaraoke, gst_audio_karaoke, GstAudioFilter,
-    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+G_DEFINE_TYPE (GstAudioKaraoke, gst_audio_karaoke, GST_TYPE_AUDIO_FILTER);
 
 static void gst_audio_karaoke_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -109,27 +105,18 @@ static void gst_audio_karaoke_transform_float (GstAudioKaraoke * filter,
 /* GObject vmethod implementations */
 
 static void
-gst_audio_karaoke_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-  GstCaps *caps;
-
-  gst_element_class_set_details_simple (element_class, "AudioKaraoke",
-      "Filter/Effect/Audio",
-      "Removes voice from sound", "Wim Taymans <wim.taymans@gmail.com>");
-
-  caps = gst_caps_from_string (ALLOWED_CAPS);
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
-      caps);
-  gst_caps_unref (caps);
-}
-
-static void
 gst_audio_karaoke_class_init (GstAudioKaraokeClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstCaps *caps;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_karaoke_debug, "audiokaraoke", 0,
+      "audiokaraoke element");
 
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
   gobject_class->set_property = gst_audio_karaoke_set_property;
   gobject_class->get_property = gst_audio_karaoke_get_property;
 
@@ -153,6 +140,15 @@ gst_audio_karaoke_class_init (GstAudioKaraokeClass * klass)
           "The Frequency width of the filter", 0.0, 100.0, DEFAULT_FILTER_WIDTH,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class, "AudioKaraoke",
+      "Filter/Effect/Audio",
+      "Removes voice from sound", "Wim Taymans <wim.taymans@gmail.com>");
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+
   GST_AUDIO_FILTER_CLASS (klass)->setup =
       GST_DEBUG_FUNCPTR (gst_audio_karaoke_setup);
   GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
@@ -160,7 +156,7 @@ gst_audio_karaoke_class_init (GstAudioKaraokeClass * klass)
 }
 
 static void
-gst_audio_karaoke_init (GstAudioKaraoke * filter, GstAudioKaraokeClass * klass)
+gst_audio_karaoke_init (GstAudioKaraoke * filter)
 {
   gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
   gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
@@ -339,6 +335,8 @@ gst_audio_karaoke_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   GstAudioKaraoke *filter = GST_AUDIO_KARAOKE (base);
   guint num_samples;
   GstClockTime timestamp, stream_time;
+  guint8 *data;
+  gsize size;
 
   timestamp = GST_BUFFER_TIMESTAMP (buf);
   stream_time =
@@ -350,14 +348,16 @@ gst_audio_karaoke_transform_ip (GstBaseTransform * base, GstBuffer * buf)
   if (GST_CLOCK_TIME_IS_VALID (stream_time))
     gst_object_sync_values (G_OBJECT (filter), stream_time);
 
-  num_samples =
-      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
-
   if (gst_base_transform_is_passthrough (base) ||
       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
     return GST_FLOW_OK;
 
-  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+  data = gst_buffer_map (buf, &size, NULL, GST_MAP_READWRITE);
+  num_samples = size / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  filter->process (filter, data, num_samples);
+
+  gst_buffer_unmap (buf, data, size);
 
   return GST_FLOW_OK;
 }

diff --git a/gst/audiofx/audiopanorama.c b/gst/audiofx/audiopanorama.c
index 768533c..7077db1 100644 (file)
--- a/gst/audiofx/audiopanorama.c
+++ b/gst/audiofx/audiopanorama.c
@@ -117,11 +117,7 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
         "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true")
     );
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_panorama_debug, "audiopanorama", 0, "audiopanorama element");
-
-GST_BOILERPLATE_FULL (GstAudioPanorama, gst_audio_panorama, GstBaseTransform,
-    GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
+G_DEFINE_TYPE (GstAudioPanorama, gst_audio_panorama, GST_TYPE_BASE_TRANSFORM);
 
 static void gst_audio_panorama_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -180,26 +176,17 @@ static GstAudioPanoramaProcessFunc panorama_process_functions[2][2][2] = {
 /* GObject vmethod implementations */
 
 static void
-gst_audio_panorama_base_init (gpointer klass)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&src_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&sink_template));
-  gst_element_class_set_details_simple (element_class, "Stereo positioning",
-      "Filter/Effect/Audio",
-      "Positions audio streams in the stereo panorama",
-      "Stefan Kost <ensonic@users.sf.net>");
-}
-
-static void
 gst_audio_panorama_class_init (GstAudioPanoramaClass * klass)
 {
   GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  GST_DEBUG_CATEGORY_INIT (gst_audio_panorama_debug, "audiopanorama", 0,
+      "audiopanorama element");
 
   gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
   gobject_class->set_property = gst_audio_panorama_set_property;
   gobject_class->get_property = gst_audio_panorama_get_property;
 
@@ -224,6 +211,16 @@ gst_audio_panorama_class_init (GstAudioPanoramaClass * klass)
           GST_TYPE_AUDIO_PANORAMA_METHOD, METHOD_PSYCHOACOUSTIC,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class, "Stereo positioning",
+      "Filter/Effect/Audio",
+      "Positions audio streams in the stereo panorama",
+      "Stefan Kost <ensonic@users.sf.net>");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&src_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&sink_template));
+
   GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
       GST_DEBUG_FUNCPTR (gst_audio_panorama_get_unit_size);
   GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
@@ -235,8 +232,7 @@ gst_audio_panorama_class_init (GstAudioPanoramaClass * klass)
 }
 
 static void
-gst_audio_panorama_init (GstAudioPanorama * filter,
-    GstAudioPanoramaClass * klass)
+gst_audio_panorama_init (GstAudioPanorama * filter)
 {
 
   filter->panorama = 0;
@@ -655,8 +651,9 @@ gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
     GstBuffer * outbuf)
 {
   GstAudioPanorama *filter = GST_AUDIO_PANORAMA (base);
-  guint num_samples = GST_BUFFER_SIZE (outbuf) / (2 * filter->width);
   GstClockTime timestamp, stream_time;
+  guint8 *indata, *outdata;
+  gsize insize, outsize;
 
   timestamp = GST_BUFFER_TIMESTAMP (inbuf);
   stream_time =
@@ -668,14 +665,20 @@ gst_audio_panorama_transform (GstBaseTransform * base, GstBuffer * inbuf,
   if (GST_CLOCK_TIME_IS_VALID (stream_time))
     gst_object_sync_values (G_OBJECT (filter), stream_time);
 
+  indata = gst_buffer_map (inbuf, &insize, NULL, GST_MAP_READ);
+  outdata = gst_buffer_map (outbuf, &outsize, NULL, GST_MAP_WRITE);
+
   if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
     GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
-    memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
-    return GST_FLOW_OK;
+    memset (outdata, 0, outsize);
+  } else {
+    guint num_samples = outsize / (2 * filter->width);
+
+    filter->process (filter, indata, outdata, num_samples);
   }
 
-  filter->process (filter, GST_BUFFER_DATA (inbuf),
-      GST_BUFFER_DATA (outbuf), num_samples);
+  gst_buffer_unmap (inbuf, indata, insize);
+  gst_buffer_unmap (outbuf, outdata, outsize);
 
   return GST_FLOW_OK;
 }

diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c
index 07356e2..db7e1d5 100644 (file)
--- a/gst/audiofx/audiowsincband.c
+++ b/gst/audiofx/audiowsincband.c
@@ -138,12 +138,9 @@ gst_gst_audio_wsincband_window_get_type (void)
   return gtype;
 }
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_gst_audio_wsincband_debug, "audiowsincband", 0, \
-      "Band-pass and Band-reject Windowed sinc filter plugin");
-
-GST_BOILERPLATE_FULL (GstAudioWSincBand, gst_audio_wsincband, GstAudioFilter,
-    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
+#define gst_audio_wsincband_parent_class parent_class
+G_DEFINE_TYPE (GstAudioWSincBand, gst_audio_wsincband,
+    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER);
 
 static void gst_audio_wsincband_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -157,27 +154,16 @@ static gboolean gst_audio_wsincband_setup (GstAudioFilter * base,
 
 #define POW2(x)  (x)*(x)
 
-/* Element class */
-static void
-gst_audio_wsincband_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class,
-      "Band pass & band reject filter", "Filter/Effect/Audio",
-      "Band pass and band reject windowed sinc filter",
-      "Thomas Vander Stichele <thomas at apestaart dot org>, "
-      "Steven W. Smith, "
-      "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
-
 static void
 gst_audio_wsincband_class_init (GstAudioWSincBandClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
+  GST_DEBUG_CATEGORY_INIT (gst_gst_audio_wsincband_debug, "audiowsincband", 0,
+      "Band-pass and Band-reject Windowed sinc filter plugin");
+
   gobject_class->set_property = gst_audio_wsincband_set_property;
   gobject_class->get_property = gst_audio_wsincband_get_property;
   gobject_class->finalize = gst_audio_wsincband_finalize;
@@ -210,12 +196,19 @@ gst_audio_wsincband_class_init (GstAudioWSincBandClass * klass)
           WINDOW_HAMMING,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class,
+      "Band pass & band reject filter", "Filter/Effect/Audio",
+      "Band pass and band reject windowed sinc filter",
+      "Thomas Vander Stichele <thomas at apestaart dot org>, "
+      "Steven W. Smith, "
+      "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
   filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsincband_setup);
 }
 
 static void
-gst_audio_wsincband_init (GstAudioWSincBand * self,
-    GstAudioWSincBandClass * g_class)
+gst_audio_wsincband_init (GstAudioWSincBand * self)
 {
   self->kernel_length = 101;
   self->lower_frequency = 0.0;

diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c
index aa0cf4b..288cee3 100644 (file)
--- a/gst/audiofx/audiowsinclimit.c
+++ b/gst/audiofx/audiowsinclimit.c
@@ -137,12 +137,9 @@ gst_audio_wsinclimit_window_get_type (void)
   return gtype;
 }
 
-#define DEBUG_INIT(bla) \
-  GST_DEBUG_CATEGORY_INIT (gst_audio_wsinclimit_debug, "audiowsinclimit", 0, \
-      "Low-pass and High-pass Windowed sinc filter plugin");
-
-GST_BOILERPLATE_FULL (GstAudioWSincLimit, gst_audio_wsinclimit, GstAudioFilter,
-    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER, DEBUG_INIT);
+#define gst_audio_wsinclimit_parent_class parent_class
+G_DEFINE_TYPE (GstAudioWSincLimit, gst_audio_wsinclimit,
+    GST_TYPE_AUDIO_FX_BASE_FIR_FILTER);
 
 static void gst_audio_wsinclimit_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec);
@@ -156,28 +153,16 @@ static gboolean gst_audio_wsinclimit_setup (GstAudioFilter * base,
 
 #define POW2(x)  (x)*(x)
 
-/* Element class */
-
-static void
-gst_audio_wsinclimit_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class,
-      "Low pass & high pass filter", "Filter/Effect/Audio",
-      "Low pass and high pass windowed sinc filter",
-      "Thomas Vander Stichele <thomas at apestaart dot org>, "
-      "Steven W. Smith, "
-      "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
-      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
-}
-
 static void
 gst_audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
 
+  GST_DEBUG_CATEGORY_INIT (gst_audio_wsinclimit_debug, "audiowsinclimit", 0,
+      "Low-pass and High-pass Windowed sinc filter plugin");
+
   gobject_class->set_property = gst_audio_wsinclimit_set_property;
   gobject_class->get_property = gst_audio_wsinclimit_get_property;
   gobject_class->finalize = gst_audio_wsinclimit_finalize;
@@ -206,12 +191,19 @@ gst_audio_wsinclimit_class_init (GstAudioWSincLimitClass * klass)
           WINDOW_HAMMING,
           G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class,
+      "Low pass & high pass filter", "Filter/Effect/Audio",
+      "Low pass and high pass windowed sinc filter",
+      "Thomas Vander Stichele <thomas at apestaart dot org>, "
+      "Steven W. Smith, "
+      "Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+
   filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_wsinclimit_setup);
 }
 
 static void
-gst_audio_wsinclimit_init (GstAudioWSincLimit * self,
-    GstAudioWSincLimitClass * g_class)
+gst_audio_wsinclimit_init (GstAudioWSincLimit * self)
 {
   self->mode = MODE_LOW_PASS;
   self->window = WINDOW_HAMMING;

diff --git a/gst/videofilter/gstgamma.c b/gst/videofilter/gstgamma.c
index 167680e..eaa0dc8 100644 (file)
--- a/gst/videofilter/gstgamma.c
+++ b/gst/videofilter/gstgamma.c
@@ -130,27 +130,13 @@ static void gst_gamma_before_transform (GstBaseTransform * transform,
 
 static void gst_gamma_calculate_tables (GstGamma * gamma);
 
-GST_BOILERPLATE (GstGamma, gst_gamma, GstVideoFilter, GST_TYPE_VIDEO_FILTER);
-
-static void
-gst_gamma_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class, "Video gamma correction",
-      "Filter/Effect/Video",
-      "Adjusts gamma on a video stream", "Arwed v. Merkatz <v.merkatz@gmx.net");
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_gamma_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_gamma_src_template));
-}
+G_DEFINE_TYPE (GstGamma, gst_gamma, GST_TYPE_VIDEO_FILTER);
 
 static void
 gst_gamma_class_init (GstGammaClass * g_class)
 {
   GObjectClass *gobject_class = (GObjectClass *) g_class;
+  GstElementClass *gstelement_class = (GstElementClass *) g_class;
   GstBaseTransformClass *trans_class = (GstBaseTransformClass *) g_class;
 
   GST_DEBUG_CATEGORY_INIT (gamma_debug, "gamma", 0, "gamma");
@@ -163,6 +149,15 @@ gst_gamma_class_init (GstGammaClass * g_class)
           0.01, 10, DEFAULT_PROP_GAMMA,
           GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS | G_PARAM_READWRITE));
 
+  gst_element_class_set_details_simple (gstelement_class,
+      "Video gamma correction", "Filter/Effect/Video",
+      "Adjusts gamma on a video stream", "Arwed v. Merkatz <v.merkatz@gmx.net");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_gamma_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_gamma_src_template));
+
   trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_gamma_set_caps);
   trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_gamma_transform_ip);
   trans_class->before_transform =
@@ -170,7 +165,7 @@ gst_gamma_class_init (GstGammaClass * g_class)
 }
 
 static void
-gst_gamma_init (GstGamma * gamma, GstGammaClass * g_class)
+gst_gamma_init (GstGamma * gamma)
 {
   /* properties */
   gamma->gamma = DEFAULT_PROP_GAMMA;

diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c
index 7fc56a6..b59a97b 100644 (file)
--- a/gst/videofilter/gstvideobalance.c
+++ b/gst/videofilter/gstvideobalance.c
@@ -121,28 +121,13 @@ static void gst_video_balance_set_property (GObject * object, guint prop_id,
 static void gst_video_balance_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 
-static void
-_do_init (GType video_balance_type)
-{
-  static const GInterfaceInfo iface_info = {
-    (GInterfaceInitFunc) gst_video_balance_interface_init,
-    NULL,
-    NULL,
-  };
-  static const GInterfaceInfo colorbalance_info = {
-    (GInterfaceInitFunc) gst_video_balance_colorbalance_init,
-    NULL,
-    NULL,
-  };
-
-  g_type_add_interface_static (video_balance_type,
-      GST_TYPE_IMPLEMENTS_INTERFACE, &iface_info);
-  g_type_add_interface_static (video_balance_type, GST_TYPE_COLOR_BALANCE,
-      &colorbalance_info);
-}
-
-GST_BOILERPLATE_FULL (GstVideoBalance, gst_video_balance, GstVideoFilter,
-    GST_TYPE_VIDEO_FILTER, _do_init);
+#define gst_video_balance_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstVideoBalance, gst_video_balance,
+    GST_TYPE_VIDEO_FILTER,
+    G_IMPLEMENT_INTERFACE (GST_TYPE_IMPLEMENTS_INTERFACE,
+        gst_video_balance_interface_init);
+    G_IMPLEMENT_INTERFACE (GST_TYPE_COLOR_BALANCE,
+        gst_video_balance_colorbalance_init));
 
 /*
  * look-up tables (LUT).
@@ -526,22 +511,6 @@ not_negotiated:
 }
 
 static void
-gst_video_balance_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class, "Video balance",
-      "Filter/Effect/Video",
-      "Adjusts brightness, contrast, hue, saturation on a video stream",
-      "David Schleef <ds@schleef.org>");
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_video_balance_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_video_balance_src_template));
-}
-
-static void
 gst_video_balance_finalize (GObject * object)
 {
   GList *channels = NULL;
@@ -568,6 +537,7 @@ static void
 gst_video_balance_class_init (GstVideoBalanceClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
 
   GST_DEBUG_CATEGORY_INIT (videobalance_debug, "videobalance", 0,
@@ -593,6 +563,16 @@ gst_video_balance_class_init (GstVideoBalanceClass * klass)
           DEFAULT_PROP_SATURATION,
           GST_PARAM_CONTROLLABLE | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class, "Video balance",
+      "Filter/Effect/Video",
+      "Adjusts brightness, contrast, hue, saturation on a video stream",
+      "David Schleef <ds@schleef.org>");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_video_balance_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_video_balance_src_template));
+
   trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_video_balance_set_caps);
   trans_class->transform_ip =
       GST_DEBUG_FUNCPTR (gst_video_balance_transform_ip);
@@ -601,8 +581,7 @@ gst_video_balance_class_init (GstVideoBalanceClass * klass)
 }
 
 static void
-gst_video_balance_init (GstVideoBalance * videobalance,
-    GstVideoBalanceClass * klass)
+gst_video_balance_init (GstVideoBalance * videobalance)
 {
   const gchar *channels[4] = { "HUE", "SATURATION",
     "BRIGHTNESS", "CONTRAST"

diff --git a/gst/videofilter/gstvideoflip.c b/gst/videofilter/gstvideoflip.c
index 3facdd1..adeea45 100644 (file)
--- a/gst/videofilter/gstvideoflip.c
+++ b/gst/videofilter/gstvideoflip.c
@@ -130,8 +130,8 @@ gst_video_flip_method_get_type (void)
   return video_flip_method_type;
 }
 
-GST_BOILERPLATE (GstVideoFlip, gst_video_flip, GstVideoFilter,
-    GST_TYPE_VIDEO_FILTER);
+#define gst_video_flip_parent_class parent_class
+G_DEFINE_TYPE (GstVideoFlip, gst_video_flip, GST_TYPE_VIDEO_FILTER);
 
 static GstCaps *
 gst_video_flip_transform_caps (GstBaseTransform * trans,
@@ -1047,24 +1047,10 @@ gst_video_flip_get_property (GObject * object, guint prop_id, GValue * value,
 }
 
 static void
-gst_video_flip_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  gst_element_class_set_details_simple (element_class, "Video flipper",
-      "Filter/Effect/Video",
-      "Flips and rotates video", "David Schleef <ds@schleef.org>");
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_video_flip_sink_template));
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&gst_video_flip_src_template));
-}
-
-static void
 gst_video_flip_class_init (GstVideoFlipClass * klass)
 {
   GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
   GstBaseTransformClass *trans_class = (GstBaseTransformClass *) klass;
 
   GST_DEBUG_CATEGORY_INIT (video_flip_debug, "videoflip", 0, "videoflip");
@@ -1077,6 +1063,15 @@ gst_video_flip_class_init (GstVideoFlipClass * klass)
           GST_TYPE_VIDEO_FLIP_METHOD, PROP_METHOD_DEFAULT,
           GST_PARAM_CONTROLLABLE | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  gst_element_class_set_details_simple (gstelement_class, "Video flipper",
+      "Filter/Effect/Video",
+      "Flips and rotates video", "David Schleef <ds@schleef.org>");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_video_flip_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_video_flip_src_template));
+
   trans_class->transform_caps =
       GST_DEBUG_FUNCPTR (gst_video_flip_transform_caps);
   trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_video_flip_set_caps);
@@ -1088,7 +1083,7 @@ gst_video_flip_class_init (GstVideoFlipClass * klass)
 }
 
 static void
-gst_video_flip_init (GstVideoFlip * videoflip, GstVideoFlipClass * klass)
+gst_video_flip_init (GstVideoFlip * videoflip)
 {
   videoflip->method = PROP_METHOD_DEFAULT;
   gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (videoflip), TRUE);