gst-indent

Original commit message from CVS:
gst-indent

7 files changed, 312 insertions, 317 deletions

diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c
index d467af49..10a00142 100644
--- a/gst-libs/gst/audio/audio.c
+++ b/gst-libs/gst/audio/audio.c
@@ -26,7 +26,7 @@
 #include <gst/gststructure.h>
 
 int
-gst_audio_frame_byte_size (GstPad* pad)
+gst_audio_frame_byte_size (GstPad * pad)
 {
 /* calculate byte size of an audio frame
  * this should be moved closer to the gstreamer core
@@ -45,20 +45,20 @@ gst_audio_frame_byte_size (GstPad* pad)
 
   if (caps == NULL) {
     /* ERROR: could not get caps of pad */
-    g_warning ("gstaudio: could not get caps of pad %s:%s\n", 
-	       GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
+    g_warning ("gstaudio: could not get caps of pad %s:%s\n",
+	GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
     return 0;
   }
 
   structure = gst_caps_get_structure (caps, 0);
 
-  gst_structure_get_int (structure, "width",    &width);
+  gst_structure_get_int (structure, "width", &width);
   gst_structure_get_int (structure, "channels", &channels);
-  return (width / 8) * channels; 
+  return (width / 8) * channels;
 }
 
 long
-gst_audio_frame_length (GstPad* pad, GstBuffer* buf)
+gst_audio_frame_length (GstPad * pad, GstBuffer * buf)
 /* calculate length of buffer in frames
  * this should be moved closer to the gstreamer core
  * and be implemented for every mime type IMO
@@ -72,13 +72,13 @@ gst_audio_frame_length (GstPad* pad, GstBuffer* buf)
     /* error */
     return 0;
   /* FIXME: this function assumes the buffer size to be a whole multiple
-   *  	    of the frame byte size
+   *        of the frame byte size
    */
   return GST_BUFFER_SIZE (buf) / frame_byte_size;
 }
 
 long
-gst_audio_frame_rate (GstPad *pad)
+gst_audio_frame_rate (GstPad * pad)
 /*
  * calculate frame rate (based on caps of pad)
  * returns 0 if failed, rate if success
@@ -93,19 +93,18 @@ gst_audio_frame_rate (GstPad *pad)
 
   if (caps == NULL) {
     /* ERROR: could not get caps of pad */
-    g_warning ("gstaudio: could not get caps of pad %s:%s\n", 
-	       GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
+    g_warning ("gstaudio: could not get caps of pad %s:%s\n",
+	GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
     return 0;
-  }
-  else {
+  } else {
     structure = gst_caps_get_structure (caps, 0);
-    gst_structure_get_int  (structure, "rate", &rate);
+    gst_structure_get_int (structure, "rate", &rate);
     return rate;
   }
 }
 
-double 
-gst_audio_length (GstPad* pad, GstBuffer* buf)
+double
+gst_audio_length (GstPad * pad, GstBuffer * buf)
 {
 /* calculate length in seconds
  * of audio buffer buf
@@ -125,20 +124,17 @@ gst_audio_length (GstPad* pad, GstBuffer* buf)
   g_assert (GST_IS_BUFFER (buf));
   /* get caps of pad */
   caps = GST_PAD_CAPS (pad);
-  if (caps == NULL)
-  {
+  if (caps == NULL) {
     /* ERROR: could not get caps of pad */
-    g_warning ("gstaudio: could not get caps of pad %s:%s\n", 
-	       GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
+    g_warning ("gstaudio: could not get caps of pad %s:%s\n",
+	GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
     length = 0.0;
-  }
-  else
-  {
+  } else {
     structure = gst_caps_get_structure (caps, 0);
     bytes = GST_BUFFER_SIZE (buf);
-    gst_structure_get_int  (structure, "width",    &width);
-    gst_structure_get_int  (structure, "channels", &channels);
-    gst_structure_get_int  (structure, "rate",     &rate);
+    gst_structure_get_int (structure, "width", &width);
+    gst_structure_get_int (structure, "channels", &channels);
+    gst_structure_get_int (structure, "rate", &rate);
 
     g_assert (bytes != 0);
     g_assert (width != 0);
@@ -150,8 +146,8 @@ gst_audio_length (GstPad* pad, GstBuffer* buf)
   return length;
 }
 
-long 
-gst_audio_highest_sample_value (GstPad* pad)
+long
+gst_audio_highest_sample_value (GstPad * pad)
 /* calculate highest possible sample value
  * based on capabilities of pad
  */
@@ -160,25 +156,25 @@ gst_audio_highest_sample_value (GstPad* pad)
   gint width = 0;
   const GstCaps *caps = NULL;
   GstStructure *structure;
-  
+
   caps = GST_PAD_CAPS (pad);
-  if (caps == NULL)
-  {
-    g_warning ("gstaudio: could not get caps of pad %s:%s\n", 
-	       GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
+  if (caps == NULL) {
+    g_warning ("gstaudio: could not get caps of pad %s:%s\n",
+	GST_ELEMENT_NAME (gst_pad_get_parent (pad)), GST_PAD_NAME (pad));
   }
-  
+
   structure = gst_caps_get_structure (caps, 0);
-  gst_structure_get_int  (structure, "width", &width);
-  gst_structure_get_boolean  (structure, "signed", &is_signed);
-  
-  if (is_signed) --width;
+  gst_structure_get_int (structure, "width", &width);
+  gst_structure_get_boolean (structure, "signed", &is_signed);
+
+  if (is_signed)
+    --width;
   /* example : 16 bit, signed : samples between -32768 and 32767 */
   return ((long) (1 << width));
 }
 
-gboolean 
-gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
+gboolean
+gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf)
 /* check if the buffer size is a whole multiple of the frame size */
 {
   if (GST_BUFFER_SIZE (buf) % gst_audio_frame_byte_size (pad) == 0)
@@ -199,8 +195,8 @@ gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
  * number of list values, and each of the values, terminating with NULL
  */
 static void
-_gst_audio_structure_set_list (GstStructure *structure, const gchar *fieldname,
-                               GType type, int number, ...)
+_gst_audio_structure_set_list (GstStructure * structure,
+    const gchar * fieldname, GType type, int number, ...)
 {
   va_list varargs;
   GValue value = { 0 };
@@ -214,27 +210,27 @@ _gst_audio_structure_set_list (GstStructure *structure, const gchar *fieldname,
 
   va_start (varargs, number);
 
-  for (j = 0; j < number; ++j)
-  {
+  for (j = 0; j < number; ++j) {
     int i;
     gboolean b;
 
     GValue list_value = { 0 };
 
-    switch (type)
-    {
+    switch (type) {
       case G_TYPE_INT:
-        i = va_arg (varargs, int);
-        g_value_init (&list_value, G_TYPE_INT);
-        g_value_set_int (&list_value, i);
-        break;
+	i = va_arg (varargs, int);
+
+	g_value_init (&list_value, G_TYPE_INT);
+	g_value_set_int (&list_value, i);
+	break;
       case G_TYPE_BOOLEAN:
-        b = va_arg (varargs, gboolean);
-        g_value_init (&list_value, G_TYPE_BOOLEAN);
-        g_value_set_boolean (&list_value, b);
-        break;
-       default:
-        g_warning ("_gst_audio_structure_set_list: LIST of given type not implemented.");
+	b = va_arg (varargs, gboolean);
+	g_value_init (&list_value, G_TYPE_BOOLEAN);
+	g_value_set_boolean (&list_value, b);
+	break;
+      default:
+	g_warning
+	    ("_gst_audio_structure_set_list: LIST of given type not implemented.");
     }
     g_array_append_val (array, list_value);
 
@@ -244,38 +240,38 @@ _gst_audio_structure_set_list (GstStructure *structure, const gchar *fieldname,
 }
 
 void
-gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag)
+gst_audio_structure_set_int (GstStructure * structure, GstAudioFieldFlag flag)
 {
   if (flag & GST_AUDIO_FIELD_RATE)
-    gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+    gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT,
+	NULL);
   if (flag & GST_AUDIO_FIELD_CHANNELS)
-    gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+    gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT,
+	NULL);
   if (flag & GST_AUDIO_FIELD_ENDIANNESS)
-    _gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2, G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
+    _gst_audio_structure_set_list (structure, "endianness", G_TYPE_INT, 2,
+	G_LITTLE_ENDIAN, G_BIG_ENDIAN, NULL);
   if (flag & GST_AUDIO_FIELD_WIDTH)
-    _gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32, NULL);
+    _gst_audio_structure_set_list (structure, "width", G_TYPE_INT, 3, 8, 16, 32,
+	NULL);
   if (flag & GST_AUDIO_FIELD_DEPTH)
     gst_structure_set (structure, "depth", GST_TYPE_INT_RANGE, 1, 32, NULL);
   if (flag & GST_AUDIO_FIELD_SIGNED)
-    _gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE, FALSE, NULL);
+    _gst_audio_structure_set_list (structure, "signed", G_TYPE_BOOLEAN, 2, TRUE,
+	FALSE, NULL);
   if (flag & GST_AUDIO_FIELD_BUFFER_FRAMES)
-    gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
+    gst_structure_set (structure, "buffer-frames", GST_TYPE_INT_RANGE, 1,
+	G_MAXINT, NULL);
 }
 
 static gboolean
-plugin_init (GstPlugin *plugin)
+plugin_init (GstPlugin * plugin)
 {
   return TRUE;
 }
 
-GST_PLUGIN_DEFINE (
-  GST_VERSION_MAJOR,
-  GST_VERSION_MINOR,
-  "gstaudio",
-  "Support services for audio plugins",
-  plugin_init,
-  VERSION,
-  GST_LICENSE,
-  GST_PACKAGE,
-  GST_ORIGIN
-);
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "gstaudio",
+    "Support services for audio plugins",
+    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE, GST_ORIGIN);
diff --git a/gst-libs/gst/audio/audio.h b/gst-libs/gst/audio/audio.h
index 4a3428dd..5f21e018 100644
--- a/gst-libs/gst/audio/audio.h
+++ b/gst-libs/gst/audio/audio.h
@@ -26,7 +26,6 @@
 #define __GST_AUDIO_AUDIO_H__
 
 G_BEGIN_DECLS
-
 /* For people that are looking at this source: the purpose of these defines is
  * to make GstCaps a bit easier, in that you don't have to know all of the
  * properties that need to be defined. you can just use these macros. currently
@@ -50,9 +49,7 @@ G_BEGIN_DECLS
  *
  * Andy Wingo, 18 August 2001
  * Thomas, 6 September 2002 */
-
 #define GST_AUDIO_DEF_RATE 44100
-
 #define GST_AUDIO_INT_PAD_TEMPLATE_CAPS \
   "audio/x-raw-int, " \
   "rate = (int) [ 1, MAX ], " \
@@ -60,9 +57,7 @@ G_BEGIN_DECLS
   "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
   "width = (int) { 8, 16, 32 }, " \
   "depth = (int) [ 1, 32 ], " \
-  "signed = (boolean) { true, false }" 
-
-
+  "signed = (boolean) { true, false }"
 /* "standard" int audio is native order, 16 bit stereo. */
 #define GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS \
   "audio/x-raw-int, " \
@@ -71,8 +66,7 @@ G_BEGIN_DECLS
   "endianness = (int) BYTE_ORDER, " \
   "width = (int) 16, " \
   "depth = (int) 16, " \
-  "signed = (boolean) true" 
-
+  "signed = (boolean) true"
 #define GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS \
   "audio/x-raw-float, " \
   "rate = (int) [ 1, MAX ], " \
@@ -80,7 +74,6 @@ G_BEGIN_DECLS
   "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " \
   "width = (int) { 32, 64 }, " \
   "buffer-frames = (int) [ 1, MAX]"
-
 /* "standard" float audio is native order, 32 bit mono. */
 #define GST_AUDIO_FLOAT_STANDARD_PAD_TEMPLATE_CAPS \
   "audio/x-raw-float, " \
@@ -88,43 +81,42 @@ G_BEGIN_DECLS
   "channels = (int) 1, " \
   "endianness = (int) BYTE_ORDER, " \
   "buffer-frames = (int) [ 1, MAX]"
-
 /*
  * this library defines and implements some helper functions for audio
  * handling
  */
-
 /* get byte size of audio frame (based on caps of pad */
-int      gst_audio_frame_byte_size      (GstPad* pad);
+int gst_audio_frame_byte_size (GstPad * pad);
 
 /* get length in frames of buffer */
-long     gst_audio_frame_length         (GstPad* pad, GstBuffer* buf);
+long gst_audio_frame_length (GstPad * pad, GstBuffer * buf);
 
 /* get frame rate based on caps */
-long     gst_audio_frame_rate           (GstPad *pad);
+long gst_audio_frame_rate (GstPad * pad);
 
 /* calculate length in seconds of audio buffer buf based on caps of pad */
-double   gst_audio_length               (GstPad* pad, GstBuffer* buf);
+double gst_audio_length (GstPad * pad, GstBuffer * buf);
 
 /* calculate highest possible sample value based on capabilities of pad */
-long     gst_audio_highest_sample_value (GstPad* pad);
+long gst_audio_highest_sample_value (GstPad * pad);
 
 /* check if the buffer size is a whole multiple of the frame size */
-gboolean gst_audio_is_buffer_framed     (GstPad* pad, GstBuffer* buf);
+gboolean gst_audio_is_buffer_framed (GstPad * pad, GstBuffer * buf);
 
 /* functions useful for _getcaps functions */
-typedef enum {
-  GST_AUDIO_FIELD_RATE          = (1 << 0),
-  GST_AUDIO_FIELD_CHANNELS      = (1 << 1),
-  GST_AUDIO_FIELD_ENDIANNESS    = (1 << 2),
-  GST_AUDIO_FIELD_WIDTH         = (1 << 3),
-  GST_AUDIO_FIELD_DEPTH         = (1 << 4),
-  GST_AUDIO_FIELD_SIGNED        = (1 << 5),
+typedef enum
+{
+  GST_AUDIO_FIELD_RATE = (1 << 0),
+  GST_AUDIO_FIELD_CHANNELS = (1 << 1),
+  GST_AUDIO_FIELD_ENDIANNESS = (1 << 2),
+  GST_AUDIO_FIELD_WIDTH = (1 << 3),
+  GST_AUDIO_FIELD_DEPTH = (1 << 4),
+  GST_AUDIO_FIELD_SIGNED = (1 << 5),
   GST_AUDIO_FIELD_BUFFER_FRAMES = (1 << 6)
 } GstAudioFieldFlag;
 
-void gst_audio_structure_set_int (GstStructure *structure, GstAudioFieldFlag flag);
+void gst_audio_structure_set_int (GstStructure * structure,
+    GstAudioFieldFlag flag);
 
 G_END_DECLS
-
 #endif /* __GST_AUDIO_AUDIO_H__ */
diff --git a/gst-libs/gst/audio/audioclock.c b/gst-libs/gst/audio/audioclock.c
index 77c7e576..d6d1da45 100644
--- a/gst-libs/gst/audio/audioclock.c
+++ b/gst-libs/gst/audio/audioclock.c
@@ -26,23 +26,24 @@
 
 #include "audioclock.h"
 
-static void 		gst_audio_clock_class_init 	(GstAudioClockClass *klass);
-static void 		gst_audio_clock_init 		(GstAudioClock *clock);
+static void gst_audio_clock_class_init (GstAudioClockClass * klass);
+static void gst_audio_clock_init (GstAudioClock * clock);
 
-static GstClockTime	gst_audio_clock_get_internal_time (GstClock *clock);
-static GstClockReturn	gst_audio_clock_id_wait_async	(GstClock *clock,
-                                                         GstClockEntry *entry);
-static void		gst_audio_clock_id_unschedule	(GstClock *clock,
-                                                         GstClockEntry *entry);
+static GstClockTime gst_audio_clock_get_internal_time (GstClock * clock);
+static GstClockReturn gst_audio_clock_id_wait_async (GstClock * clock,
+    GstClockEntry * entry);
+static void gst_audio_clock_id_unschedule (GstClock * clock,
+    GstClockEntry * entry);
 
 static GstSystemClockClass *parent_class = NULL;
+
 /* static guint gst_audio_clock_signals[LAST_SIGNAL] = { 0 }; */
-  
+
 GType
 gst_audio_clock_get_type (void)
-{ 
+{
   static GType clock_type = 0;
-	    
+
   if (!clock_type) {
     static const GTypeInfo clock_info = {
       sizeof (GstAudioClockClass),
@@ -57,32 +58,32 @@ gst_audio_clock_get_type (void)
       NULL
     };
     clock_type = g_type_register_static (GST_TYPE_SYSTEM_CLOCK, "GstAudioClock",
-                                         &clock_info, 0);
+	&clock_info, 0);
   }
   return clock_type;
 }
 
 
 static void
-gst_audio_clock_class_init (GstAudioClockClass *klass)
+gst_audio_clock_class_init (GstAudioClockClass * klass)
 {
   GObjectClass *gobject_class;
   GstObjectClass *gstobject_class;
   GstClockClass *gstclock_class;
 
-  gobject_class = (GObjectClass*) klass;
-  gstobject_class = (GstObjectClass*) klass;
-  gstclock_class = (GstClockClass*) klass;
+  gobject_class = (GObjectClass *) klass;
+  gstobject_class = (GstObjectClass *) klass;
+  gstclock_class = (GstClockClass *) klass;
 
   parent_class = g_type_class_ref (GST_TYPE_SYSTEM_CLOCK);
 
-  gstclock_class->get_internal_time	= gst_audio_clock_get_internal_time;
-  gstclock_class->wait_async		= gst_audio_clock_id_wait_async;
-  gstclock_class->unschedule		= gst_audio_clock_id_unschedule;
+  gstclock_class->get_internal_time = gst_audio_clock_get_internal_time;
+  gstclock_class->wait_async = gst_audio_clock_id_wait_async;
+  gstclock_class->unschedule = gst_audio_clock_id_unschedule;
 }
 
 static void
-gst_audio_clock_init (GstAudioClock *clock)
+gst_audio_clock_init (GstAudioClock * clock)
 {
   gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
 
@@ -90,20 +91,22 @@ gst_audio_clock_init (GstAudioClock *clock)
   clock->prev2 = 0;
 }
 
-GstClock*
-gst_audio_clock_new (gchar *name, GstAudioClockGetTimeFunc func, gpointer user_data)
+GstClock *
+gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
+    gpointer user_data)
 {
-  GstAudioClock *aclock = GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
+  GstAudioClock *aclock =
+      GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
 
   aclock->func = func;
   aclock->user_data = user_data;
   aclock->adjust = 0;
 
-  return (GstClock*)aclock;
+  return (GstClock *) aclock;
 }
 
 void
-gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active)
+gst_audio_clock_set_active (GstAudioClock * aclock, gboolean active)
 {
   GstClockTime time;
   GstClock *clock;
@@ -117,8 +120,9 @@ gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active)
     aclock->adjust = time - aclock->func (clock, aclock->user_data);
   } else {
     GTimeVal timeval;
+
     g_get_current_time (&timeval);
-	
+
     aclock->adjust = GST_TIMEVAL_TO_TIME (timeval) - time;
   }
 
@@ -126,22 +130,22 @@ gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active)
 }
 
 static GstClockTime
-gst_audio_clock_get_internal_time (GstClock *clock)
+gst_audio_clock_get_internal_time (GstClock * clock)
 {
   GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
-  
+
   if (aclock->active) {
     return aclock->func (clock, aclock->user_data) + aclock->adjust;
   } else {
     GTimeVal timeval;
-    
+
     g_get_current_time (&timeval);
     return GST_TIMEVAL_TO_TIME (timeval);
   }
 }
 
 void
-gst_audio_clock_update_time (GstAudioClock *aclock, GstClockTime time)
+gst_audio_clock_update_time (GstAudioClock * aclock, GstClockTime time)
 {
   /* I don't know of a purpose in updating these; perhaps they can be removed */
   aclock->prev2 = aclock->prev1;
@@ -150,43 +154,41 @@ gst_audio_clock_update_time (GstAudioClock *aclock, GstClockTime time)
   /* FIXME: the wait_async subsystem should be made threadsafe, but I don't want
    * to lock and unlock a mutex on every iteration... */
   while (aclock->async_entries) {
-    GstClockEntry *entry = (GstClockEntry*)aclock->async_entries->data;
-    
+    GstClockEntry *entry = (GstClockEntry *) aclock->async_entries->data;
+
     if (entry->time > time)
       break;
 
-    entry->func ((GstClock*)aclock, time, entry, entry->user_data);
+    entry->func ((GstClock *) aclock, time, entry, entry->user_data);
 
     aclock->async_entries = g_slist_delete_link (aclock->async_entries,
-                                                 aclock->async_entries);
+	aclock->async_entries);
     /* do I need to free the entry? */
   }
 }
 
 static gint
-compare_clock_entries (GstClockEntry *entry1, GstClockEntry *entry2)
+compare_clock_entries (GstClockEntry * entry1, GstClockEntry * entry2)
 {
   return entry1->time - entry2->time;
 }
 
 static GstClockReturn
-gst_audio_clock_id_wait_async (GstClock *clock, GstClockEntry *entry)
+gst_audio_clock_id_wait_async (GstClock * clock, GstClockEntry * entry)
 {
-  GstAudioClock *aclock = (GstAudioClock*)clock;
-  
+  GstAudioClock *aclock = (GstAudioClock *) clock;
+
   aclock->async_entries = g_slist_insert_sorted (aclock->async_entries,
-                                                 entry,
-                                                 (GCompareFunc)compare_clock_entries);
+      entry, (GCompareFunc) compare_clock_entries);
 
   /* is this the proper return val? */
   return GST_CLOCK_EARLY;
 }
 
 static void
-gst_audio_clock_id_unschedule (GstClock *clock, GstClockEntry *entry)
+gst_audio_clock_id_unschedule (GstClock * clock, GstClockEntry * entry)
 {
-  GstAudioClock *aclock = (GstAudioClock*)clock;
-  
-  aclock->async_entries = g_slist_remove (aclock->async_entries,
-                                          entry);
+  GstAudioClock *aclock = (GstAudioClock *) clock;
+
+  aclock->async_entries = g_slist_remove (aclock->async_entries, entry);
 }
diff --git a/gst-libs/gst/audio/audioclock.h b/gst-libs/gst/audio/audioclock.h
index 17439242..abb07541 100644
--- a/gst-libs/gst/audio/audioclock.h
+++ b/gst-libs/gst/audio/audioclock.h
@@ -27,7 +27,6 @@
 #include <gst/gstsystemclock.h>
 
 G_BEGIN_DECLS
-
 #define GST_TYPE_AUDIO_CLOCK \
   (gst_audio_clock_get_type())
 #define GST_AUDIO_CLOCK(obj) \
@@ -38,14 +37,15 @@ G_BEGIN_DECLS
   (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CLOCK))
 #define GST_IS_AUDIO_CLOCK_CLASS(obj) \
   (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CLOCK))
-
 typedef struct _GstAudioClock GstAudioClock;
 typedef struct _GstAudioClockClass GstAudioClockClass;
 
-typedef GstClockTime (*GstAudioClockGetTimeFunc) (GstClock *clock, gpointer user_data);
+typedef GstClockTime (*GstAudioClockGetTimeFunc) (GstClock * clock,
+    gpointer user_data);
 
 
-struct _GstAudioClock {
+struct _GstAudioClock
+{
   GstSystemClock clock;
 
   GstClockTime prev1, prev2;
@@ -63,19 +63,19 @@ struct _GstAudioClock {
   gpointer _gst_reserved[GST_PADDING];
 };
 
-struct _GstAudioClockClass {
+struct _GstAudioClockClass
+{
   GstSystemClockClass parent_class;
 
   gpointer _gst_reserved[GST_PADDING];
 };
 
-GType           gst_audio_clock_get_type 	(void);
-GstClock*	gst_audio_clock_new		(gchar *name, GstAudioClockGetTimeFunc func,
-                                                 gpointer user_data);
-void		gst_audio_clock_set_active 	(GstAudioClock *aclock, gboolean active);
+GType gst_audio_clock_get_type (void);
+GstClock *gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
+    gpointer user_data);
+void gst_audio_clock_set_active (GstAudioClock * aclock, gboolean active);
 
-void		gst_audio_clock_update_time	(GstAudioClock *aclock, GstClockTime time);
+void gst_audio_clock_update_time (GstAudioClock * aclock, GstClockTime time);
 
 G_END_DECLS
-
 #endif /* __GST_AUDIO_CLOCK_H__ */
diff --git a/gst-libs/gst/audio/gstaudiofilter.c b/gst-libs/gst/audio/gstaudiofilter.c
index 35ed4875..ad491bb6 100644
--- a/gst-libs/gst/audio/gstaudiofilter.c
+++ b/gst-libs/gst/audio/gstaudiofilter.c
@@ -29,26 +29,30 @@
 
 
 /* GstAudiofilter signals and args */
-enum {
+enum
+{
   /* FILL ME */
   LAST_SIGNAL
 };
 
-enum {
+enum
+{
   ARG_0,
   ARG_METHOD,
   /* FILL ME */
 };
 
-static void     gst_audiofilter_base_init       (gpointer g_class);
-static void	gst_audiofilter_class_init	(gpointer g_class, gpointer class_data);
-static void	gst_audiofilter_init		(GTypeInstance *instance, gpointer g_class);
+static void gst_audiofilter_base_init (gpointer g_class);
+static void gst_audiofilter_class_init (gpointer g_class, gpointer class_data);
+static void gst_audiofilter_init (GTypeInstance * instance, gpointer g_class);
 
-static void	gst_audiofilter_set_property		(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
-static void	gst_audiofilter_get_property		(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
+static void gst_audiofilter_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_audiofilter_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
 
-static void	gst_audiofilter_chain		(GstPad *pad, GstData *_data);
-GstCaps * gst_audiofilter_class_get_capslist(GstAudiofilterClass *klass);
+static void gst_audiofilter_chain (GstPad * pad, GstData * _data);
+GstCaps *gst_audiofilter_class_get_capslist (GstAudiofilterClass * klass);
 
 static GstElementClass *parent_class = NULL;
 
@@ -59,23 +63,24 @@ gst_audiofilter_get_type (void)
 
   if (!audiofilter_type) {
     static const GTypeInfo audiofilter_info = {
-      sizeof(GstAudiofilterClass),
+      sizeof (GstAudiofilterClass),
       gst_audiofilter_base_init,
       NULL,
       gst_audiofilter_class_init,
       NULL,
       NULL,
-      sizeof(GstAudiofilter),
+      sizeof (GstAudiofilter),
       0,
       gst_audiofilter_init,
     };
-    audiofilter_type = g_type_register_static(GST_TYPE_ELEMENT,
+    audiofilter_type = g_type_register_static (GST_TYPE_ELEMENT,
 	"GstAudiofilter", &audiofilter_info, G_TYPE_FLAG_ABSTRACT);
   }
   return audiofilter_type;
 }
 
-static void gst_audiofilter_base_init (gpointer g_class)
+static void
+gst_audiofilter_base_init (gpointer g_class)
 {
   static GstElementDetails audiofilter_details = {
     "Audio filter base class",
@@ -89,24 +94,25 @@ static void gst_audiofilter_base_init (gpointer g_class)
   gst_element_class_set_details (element_class, &audiofilter_details);
 }
 
-static void gst_audiofilter_class_init (gpointer g_class, gpointer class_data)
+static void
+gst_audiofilter_class_init (gpointer g_class, gpointer class_data)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
   GstAudiofilterClass *klass;
 
-  klass = (GstAudiofilterClass *)g_class;
-  gobject_class = (GObjectClass*)klass;
-  gstelement_class = (GstElementClass*)klass;
+  klass = (GstAudiofilterClass *) g_class;
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
 
-  parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
+  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
 
   gobject_class->set_property = gst_audiofilter_set_property;
   gobject_class->get_property = gst_audiofilter_get_property;
 }
 
 static GstPadLinkReturn
-gst_audiofilter_link (GstPad *pad, const GstCaps *caps)
+gst_audiofilter_link (GstPad * pad, const GstCaps * caps)
 {
   GstAudiofilter *audiofilter;
   GstPadLinkReturn ret;
@@ -114,11 +120,10 @@ gst_audiofilter_link (GstPad *pad, const GstCaps *caps)
   GstStructure *structure;
   GstAudiofilterClass *audiofilter_class;
 
-  GST_DEBUG("gst_audiofilter_link");
+  GST_DEBUG ("gst_audiofilter_link");
   audiofilter = GST_AUDIOFILTER (gst_pad_get_parent (pad));
-  audiofilter_class = GST_AUDIOFILTER_CLASS (
-      G_OBJECT_GET_CLASS (audiofilter));
-  
+  audiofilter_class = GST_AUDIOFILTER_CLASS (G_OBJECT_GET_CLASS (audiofilter));
+
 
   if (pad == audiofilter->srcpad) {
     link_ret = gst_pad_try_set_caps (audiofilter->sinkpad, caps);
@@ -135,53 +140,55 @@ gst_audiofilter_link (GstPad *pad, const GstCaps *caps)
   if (strcmp (gst_structure_get_name (structure), "audio/x-raw-int") == 0) {
     ret = gst_structure_get_int (structure, "depth", &audiofilter->depth);
     ret &= gst_structure_get_int (structure, "width", &audiofilter->width);
-    ret &= gst_structure_get_int (structure, "channels", &audiofilter->channels);
+    ret &=
+	gst_structure_get_int (structure, "channels", &audiofilter->channels);
   } else if (strcmp (gst_structure_get_name (structure), "audio/x-raw-float")
       == 0) {
 
   } else {
-    g_assert_not_reached();
+    g_assert_not_reached ();
   }
   ret &= gst_structure_get_int (structure, "rate", &audiofilter->rate);
 
-  audiofilter->bytes_per_sample = (audiofilter->width/8) *
-    audiofilter->channels;
+  audiofilter->bytes_per_sample = (audiofilter->width / 8) *
+      audiofilter->channels;
 
-  if (audiofilter_class->setup) (audiofilter_class->setup) (audiofilter);
+  if (audiofilter_class->setup)
+    (audiofilter_class->setup) (audiofilter);
 
   return GST_PAD_LINK_OK;
 }
 
 static void
-gst_audiofilter_init (GTypeInstance *instance, gpointer g_class)
+gst_audiofilter_init (GTypeInstance * instance, gpointer g_class)
 {
   GstAudiofilter *audiofilter = GST_AUDIOFILTER (instance);
   GstPadTemplate *pad_template;
 
-  GST_DEBUG("gst_audiofilter_init");
-
-  pad_template = gst_element_class_get_pad_template(GST_ELEMENT_CLASS(g_class),
-      "sink");
-  g_return_if_fail(pad_template != NULL);
-  audiofilter->sinkpad = gst_pad_new_from_template(pad_template, "sink");
-  gst_element_add_pad(GST_ELEMENT(audiofilter),audiofilter->sinkpad);
-  gst_pad_set_chain_function(audiofilter->sinkpad,gst_audiofilter_chain);
-  gst_pad_set_link_function(audiofilter->sinkpad,gst_audiofilter_link);
-  gst_pad_set_getcaps_function(audiofilter->sinkpad,gst_pad_proxy_getcaps);
-
-  pad_template = gst_element_class_get_pad_template(GST_ELEMENT_CLASS(g_class),
-      "src");
-  g_return_if_fail(pad_template != NULL);
-  audiofilter->srcpad = gst_pad_new_from_template(pad_template, "src");
-  gst_element_add_pad(GST_ELEMENT(audiofilter),audiofilter->srcpad);
-  gst_pad_set_link_function(audiofilter->srcpad,gst_audiofilter_link);
-  gst_pad_set_getcaps_function(audiofilter->srcpad,gst_pad_proxy_getcaps);
+  GST_DEBUG ("gst_audiofilter_init");
+
+  pad_template =
+      gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
+  g_return_if_fail (pad_template != NULL);
+  audiofilter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
+  gst_element_add_pad (GST_ELEMENT (audiofilter), audiofilter->sinkpad);
+  gst_pad_set_chain_function (audiofilter->sinkpad, gst_audiofilter_chain);
+  gst_pad_set_link_function (audiofilter->sinkpad, gst_audiofilter_link);
+  gst_pad_set_getcaps_function (audiofilter->sinkpad, gst_pad_proxy_getcaps);
+
+  pad_template =
+      gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
+  g_return_if_fail (pad_template != NULL);
+  audiofilter->srcpad = gst_pad_new_from_template (pad_template, "src");
+  gst_element_add_pad (GST_ELEMENT (audiofilter), audiofilter->srcpad);
+  gst_pad_set_link_function (audiofilter->srcpad, gst_audiofilter_link);
+  gst_pad_set_getcaps_function (audiofilter->srcpad, gst_pad_proxy_getcaps);
 
   audiofilter->inited = FALSE;
 }
 
 static void
-gst_audiofilter_chain (GstPad *pad, GstData *data)
+gst_audiofilter_chain (GstPad * pad, GstData * data)
 {
   GstBuffer *inbuf = GST_BUFFER (data);
   GstAudiofilter *audiofilter;
@@ -196,60 +203,60 @@ gst_audiofilter_chain (GstPad *pad, GstData *data)
 
   audiofilter = GST_AUDIOFILTER (gst_pad_get_parent (pad));
   //g_return_if_fail (audiofilter->inited);
-  audiofilter_class = GST_AUDIOFILTER_CLASS (
-      G_OBJECT_GET_CLASS (audiofilter));
+  audiofilter_class = GST_AUDIOFILTER_CLASS (G_OBJECT_GET_CLASS (audiofilter));
 
   GST_DEBUG ("gst_audiofilter_chain: got buffer of %d bytes in '%s'",
-      GST_BUFFER_SIZE(inbuf), GST_OBJECT_NAME (audiofilter));
- 
-  if(audiofilter->passthru){
-    gst_pad_push(audiofilter->srcpad, data);
+      GST_BUFFER_SIZE (inbuf), GST_OBJECT_NAME (audiofilter));
+
+  if (audiofilter->passthru) {
+    gst_pad_push (audiofilter->srcpad, data);
     return;
   }
 
   audiofilter->size = GST_BUFFER_SIZE (inbuf);
   audiofilter->n_samples = audiofilter->size / audiofilter->bytes_per_sample;
 
-  if (gst_data_is_writable(data)) {
+  if (gst_data_is_writable (data)) {
     if (audiofilter_class->filter_inplace) {
       (audiofilter_class->filter_inplace) (audiofilter, inbuf);
       outbuf = inbuf;
     } else {
-      outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE(inbuf));
-      GST_BUFFER_DURATION(outbuf) = GST_BUFFER_DURATION(inbuf);
-      GST_BUFFER_TIMESTAMP(outbuf) = GST_BUFFER_TIMESTAMP(inbuf);
+      outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE (inbuf));
+      GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (inbuf);
+      GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (inbuf);
 
       (audiofilter_class->filter) (audiofilter, outbuf, inbuf);
-      gst_buffer_unref(inbuf);
+      gst_buffer_unref (inbuf);
     }
   } else {
-    outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE(inbuf));
-    GST_BUFFER_DURATION(outbuf) = GST_BUFFER_DURATION(inbuf);
-    GST_BUFFER_TIMESTAMP(outbuf) = GST_BUFFER_TIMESTAMP(inbuf);
+    outbuf = gst_buffer_new_and_alloc (GST_BUFFER_SIZE (inbuf));
+    GST_BUFFER_DURATION (outbuf) = GST_BUFFER_DURATION (inbuf);
+    GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (inbuf);
     if (audiofilter_class->filter) {
       (audiofilter_class->filter) (audiofilter, outbuf, inbuf);
     } else {
-      memcpy(GST_BUFFER_DATA(outbuf), GST_BUFFER_DATA(inbuf),
-          GST_BUFFER_SIZE(inbuf));
+      memcpy (GST_BUFFER_DATA (outbuf), GST_BUFFER_DATA (inbuf),
+	  GST_BUFFER_SIZE (inbuf));
 
       (audiofilter_class->filter_inplace) (audiofilter, outbuf);
     }
-    gst_buffer_unref(inbuf);
+    gst_buffer_unref (inbuf);
   }
 
-  gst_pad_push(audiofilter->srcpad, GST_DATA (outbuf));
+  gst_pad_push (audiofilter->srcpad, GST_DATA (outbuf));
 }
 
 static void
-gst_audiofilter_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+gst_audiofilter_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
 {
   GstAudiofilter *src;
 
   /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail(GST_IS_AUDIOFILTER(object));
-  src = GST_AUDIOFILTER(object);
+  g_return_if_fail (GST_IS_AUDIOFILTER (object));
+  src = GST_AUDIOFILTER (object);
 
-  GST_DEBUG("gst_audiofilter_set_property");
+  GST_DEBUG ("gst_audiofilter_set_property");
   switch (prop_id) {
     default:
       break;
@@ -257,13 +264,14 @@ gst_audiofilter_set_property (GObject *object, guint prop_id, const GValue *valu
 }
 
 static void
-gst_audiofilter_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+gst_audiofilter_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
 {
   GstAudiofilter *src;
 
   /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail(GST_IS_AUDIOFILTER(object));
-  src = GST_AUDIOFILTER(object);
+  g_return_if_fail (GST_IS_AUDIOFILTER (object));
+  src = GST_AUDIOFILTER (object);
 
   switch (prop_id) {
     default:
@@ -272,37 +280,31 @@ gst_audiofilter_get_property (GObject *object, guint prop_id, GValue *value, GPa
   }
 }
 
-void gst_audiofilter_class_add_pad_templates (
-    GstAudiofilterClass *audiofilter_class, const GstCaps *caps)
+void
+gst_audiofilter_class_add_pad_templates (GstAudiofilterClass *
+    audiofilter_class, const GstCaps * caps)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (audiofilter_class);
 
-  audiofilter_class->caps = gst_caps_copy(caps);
+  audiofilter_class->caps = gst_caps_copy (caps);
 
   gst_element_class_add_pad_template (element_class,
-      gst_pad_template_new("src", GST_PAD_SRC, GST_PAD_ALWAYS,
-        gst_caps_copy(caps)));
+      gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+	  gst_caps_copy (caps)));
 
   gst_element_class_add_pad_template (element_class,
-      gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
-        gst_caps_copy(caps)));
+      gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+	  gst_caps_copy (caps)));
 }
 
 static gboolean
-plugin_init (GstPlugin *plugin)
+plugin_init (GstPlugin * plugin)
 {
   return TRUE;
 }
 
-GST_PLUGIN_DEFINE (
-  GST_VERSION_MAJOR,
-  GST_VERSION_MINOR,
-  "gstaudiofilter",
-  "Audio filter parent class",
-  plugin_init,
-  VERSION,
-  "LGPL",
-  GST_PACKAGE,
-  GST_ORIGIN
-)
-
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "gstaudiofilter",
+    "Audio filter parent class",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
diff --git a/gst-libs/gst/audio/gstaudiofilter.h b/gst-libs/gst/audio/gstaudiofilter.h
index 9786e16c..534d4c6b 100644
--- a/gst-libs/gst/audio/gstaudiofilter.h
+++ b/gst-libs/gst/audio/gstaudiofilter.h
@@ -25,17 +25,15 @@
 #include <gst/gst.h>
 
 
-G_BEGIN_DECLS
-
-typedef struct _GstAudiofilter GstAudiofilter;
+G_BEGIN_DECLS typedef struct _GstAudiofilter GstAudiofilter;
 typedef struct _GstAudiofilterClass GstAudiofilterClass;
 
-typedef void (*GstAudiofilterFilterFunc)(GstAudiofilter *filter,
-    GstBuffer *outbuf, GstBuffer *inbuf);
-typedef void (*GstAudiofilterInplaceFilterFunc)(GstAudiofilter *filter,
-    GstBuffer *buffer);
+typedef void (*GstAudiofilterFilterFunc) (GstAudiofilter * filter,
+    GstBuffer * outbuf, GstBuffer * inbuf);
+typedef void (*GstAudiofilterInplaceFilterFunc) (GstAudiofilter * filter,
+    GstBuffer * buffer);
 
-typedef void (*GstAudiofilterSetupFunc) (GstAudiofilter *filter);
+typedef void (*GstAudiofilterSetupFunc) (GstAudiofilter * filter);
 
 
 #define GST_TYPE_AUDIOFILTER \
@@ -49,10 +47,11 @@ typedef void (*GstAudiofilterSetupFunc) (GstAudiofilter *filter);
 #define GST_IS_AUDIOFILTER_CLASS(obj) \
   (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOFILTER))
 
-struct _GstAudiofilter {
+struct _GstAudiofilter
+{
   GstElement element;
 
-  GstPad *sinkpad,*srcpad;
+  GstPad *sinkpad, *srcpad;
 
   /* audio state */
   gboolean inited;
@@ -68,7 +67,8 @@ struct _GstAudiofilter {
   int bytes_per_sample;
 };
 
-struct _GstAudiofilterClass {
+struct _GstAudiofilterClass
+{
   GstElementClass parent_class;
 
   GstCaps *caps;
@@ -77,11 +77,10 @@ struct _GstAudiofilterClass {
   GstAudiofilterFilterFunc filter;
 };
 
-GType gst_audiofilter_get_type(void);
+GType gst_audiofilter_get_type (void);
 
-void gst_audiofilter_class_add_pad_templates (GstAudiofilterClass *audiofilterclass, const GstCaps *caps);
+void gst_audiofilter_class_add_pad_templates (GstAudiofilterClass *
+    audiofilterclass, const GstCaps * caps);
 
 G_END_DECLS
-
 #endif /* __GST_AUDIOFILTER_H__ */
-
diff --git a/gst-libs/gst/audio/gstaudiofiltertemplate.c b/gst-libs/gst/audio/gstaudiofiltertemplate.c
index c7c0ce2b..994fdc59 100644
--- a/gst-libs/gst/audio/gstaudiofiltertemplate.c
+++ b/gst-libs/gst/audio/gstaudiofiltertemplate.c
@@ -48,37 +48,47 @@ typedef struct _GstAudiofilterTemplateClass GstAudiofilterTemplateClass;
 #define GST_IS_AUDIOFILTER_TEMPLATE_CLASS(obj) \
   (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIOFILTER_TEMPLATE))
 
-struct _GstAudiofilterTemplate {
+struct _GstAudiofilterTemplate
+{
   GstAudiofilter audiofilter;
 
 };
 
-struct _GstAudiofilterTemplateClass {
+struct _GstAudiofilterTemplateClass
+{
   GstAudiofilterClass parent_class;
 
 };
 
 
-enum {
+enum
+{
   /* FILL ME */
   LAST_SIGNAL
 };
 
-enum {
+enum
+{
   ARG_0,
   /* FILL ME */
 };
 
-static void     gst_audiofilter_template_base_init       (gpointer g_class);
-static void	gst_audiofilter_template_class_init	(gpointer g_class, gpointer class_data);
-static void gst_audiofilter_template_init (GTypeInstance *instance, gpointer g_class);
+static void gst_audiofilter_template_base_init (gpointer g_class);
+static void gst_audiofilter_template_class_init (gpointer g_class,
+    gpointer class_data);
+static void gst_audiofilter_template_init (GTypeInstance * instance,
+    gpointer g_class);
 
-static void	gst_audiofilter_template_set_property		(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
-static void	gst_audiofilter_template_get_property		(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
+static void gst_audiofilter_template_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audiofilter_template_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
 
-static void gst_audiofilter_template_setup (GstAudiofilter *audiofilter);
-static void gst_audiofilter_template_filter (GstAudiofilter *audiofilter, GstBuffer *outbuf, GstBuffer *inbuf);
-static void gst_audiofilter_template_filter_inplace (GstAudiofilter *audiofilter, GstBuffer *buf);
+static void gst_audiofilter_template_setup (GstAudiofilter * audiofilter);
+static void gst_audiofilter_template_filter (GstAudiofilter * audiofilter,
+    GstBuffer * outbuf, GstBuffer * inbuf);
+static void gst_audiofilter_template_filter_inplace (GstAudiofilter *
+    audiofilter, GstBuffer * buf);
 
 GType
 gst_audiofilter_template_get_type (void)
@@ -87,23 +97,24 @@ gst_audiofilter_template_get_type (void)
 
   if (!audiofilter_template_type) {
     static const GTypeInfo audiofilter_template_info = {
-      sizeof(GstAudiofilterTemplateClass),
+      sizeof (GstAudiofilterTemplateClass),
       gst_audiofilter_template_base_init,
       NULL,
       gst_audiofilter_template_class_init,
       NULL,
       gst_audiofilter_template_init,
-      sizeof(GstAudiofilterTemplate),
+      sizeof (GstAudiofilterTemplate),
       0,
       NULL,
     };
-    audiofilter_template_type = g_type_register_static(GST_TYPE_AUDIOFILTER,
+    audiofilter_template_type = g_type_register_static (GST_TYPE_AUDIOFILTER,
 	"GstAudiofilterTemplate", &audiofilter_template_info, 0);
   }
   return audiofilter_template_type;
 }
 
-static void gst_audiofilter_template_base_init (gpointer g_class)
+static void
+gst_audiofilter_template_base_init (gpointer g_class)
 {
   static GstElementDetails audiofilter_template_details = {
     "Audio filter template",
@@ -128,16 +139,16 @@ gst_audiofilter_template_class_init (gpointer g_class, gpointer class_data)
   GstAudiofilterTemplateClass *klass;
   GstAudiofilterClass *audiofilter_class;
 
-  klass = (GstAudiofilterTemplateClass *)g_class;
-  gobject_class = (GObjectClass*)klass;
-  gstelement_class = (GstElementClass*)klass;
-  audiofilter_class = (GstAudiofilterClass *)g_class;
+  klass = (GstAudiofilterTemplateClass *) g_class;
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  audiofilter_class = (GstAudiofilterClass *) g_class;
 
 #if 0
-  g_object_class_install_property(gobject_class, ARG_METHOD,
-      g_param_spec_enum("method","method","method",
-      GST_TYPE_AUDIOTEMPLATE_METHOD, GST_AUDIOTEMPLATE_METHOD_1,
-      G_PARAM_READWRITE));
+  g_object_class_install_property (gobject_class, ARG_METHOD,
+      g_param_spec_enum ("method", "method", "method",
+	  GST_TYPE_AUDIOTEMPLATE_METHOD, GST_AUDIOTEMPLATE_METHOD_1,
+	  G_PARAM_READWRITE));
 #endif
 
   gobject_class->set_property = gst_audiofilter_template_set_property;
@@ -146,11 +157,11 @@ gst_audiofilter_template_class_init (gpointer g_class, gpointer class_data)
   audiofilter_class->setup = gst_audiofilter_template_setup;
   audiofilter_class->filter = gst_audiofilter_template_filter;
   audiofilter_class->filter_inplace = gst_audiofilter_template_filter_inplace;
-audiofilter_class->filter = NULL;
+  audiofilter_class->filter = NULL;
 }
 
 static void
-gst_audiofilter_template_init (GTypeInstance *instance, gpointer g_class)
+gst_audiofilter_template_init (GTypeInstance * instance, gpointer g_class)
 {
   //GstAudiofilterTemplate *audiofilter_template = GST_AUDIOFILTER_TEMPLATE (instance);
   //GstAudiofilter *audiofilter = GST_AUDIOFILTER (instance);
@@ -162,15 +173,16 @@ gst_audiofilter_template_init (GTypeInstance *instance, gpointer g_class)
 }
 
 static void
-gst_audiofilter_template_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+gst_audiofilter_template_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
 {
   GstAudiofilterTemplate *src;
 
   /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail(GST_IS_AUDIOFILTER_TEMPLATE(object));
-  src = GST_AUDIOFILTER_TEMPLATE(object);
+  g_return_if_fail (GST_IS_AUDIOFILTER_TEMPLATE (object));
+  src = GST_AUDIOFILTER_TEMPLATE (object);
 
-  GST_DEBUG("gst_audiofilter_template_set_property");
+  GST_DEBUG ("gst_audiofilter_template_set_property");
   switch (prop_id) {
     default:
       break;
@@ -178,13 +190,14 @@ gst_audiofilter_template_set_property (GObject *object, guint prop_id, const GVa
 }
 
 static void
-gst_audiofilter_template_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+gst_audiofilter_template_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
 {
   GstAudiofilterTemplate *src;
 
   /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail(GST_IS_AUDIOFILTER_TEMPLATE(object));
-  src = GST_AUDIOFILTER_TEMPLATE(object);
+  g_return_if_fail (GST_IS_AUDIOFILTER_TEMPLATE (object));
+  src = GST_AUDIOFILTER_TEMPLATE (object);
 
   switch (prop_id) {
     default:
@@ -194,7 +207,7 @@ gst_audiofilter_template_get_property (GObject *object, guint prop_id, GValue *v
 }
 
 static gboolean
-plugin_init (GstPlugin *plugin)
+plugin_init (GstPlugin * plugin)
 {
   if (!gst_library_load ("gstaudiofilter"))
     return FALSE;
@@ -203,20 +216,13 @@ plugin_init (GstPlugin *plugin)
       GST_TYPE_AUDIOFILTER_TEMPLATE);
 }
 
-GST_PLUGIN_DEFINE (
-  GST_VERSION_MAJOR,
-  GST_VERSION_MINOR,
-  "gstaudiofilter_template",
-  "Audio filter template",
-  plugin_init,
-  VERSION,
-  "LGPL",
-  GST_PACKAGE,
-  GST_ORIGIN
-)
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "gstaudiofilter_template",
+    "Audio filter template",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)
 
-static void
-gst_audiofilter_template_setup (GstAudiofilter *audiofilter)
+     static void gst_audiofilter_template_setup (GstAudiofilter * audiofilter)
 {
   GstAudiofilterTemplate *audiofilter_template;
 
@@ -234,8 +240,8 @@ gst_audiofilter_template_setup (GstAudiofilter *audiofilter)
  * with a minimum of memory copies. */
 
 static void
-gst_audiofilter_template_filter (GstAudiofilter *audiofilter,
-    GstBuffer *outbuf, GstBuffer *inbuf)
+gst_audiofilter_template_filter (GstAudiofilter * audiofilter,
+    GstBuffer * outbuf, GstBuffer * inbuf)
 {
   GstAudiofilterTemplate *audiofilter_template;
 
@@ -245,13 +251,12 @@ gst_audiofilter_template_filter (GstAudiofilter *audiofilter,
   /* do something interesting here.  This simply copies the source
    * to the destination. */
 
-  memcpy (GST_BUFFER_DATA (outbuf), GST_BUFFER_DATA (inbuf),
-      audiofilter->size);
+  memcpy (GST_BUFFER_DATA (outbuf), GST_BUFFER_DATA (inbuf), audiofilter->size);
 }
 
 static void
-gst_audiofilter_template_filter_inplace (GstAudiofilter *audiofilter,
-    GstBuffer *buf)
+gst_audiofilter_template_filter_inplace (GstAudiofilter * audiofilter,
+    GstBuffer * buf)
 {
   GstAudiofilterTemplate *audiofilter_template;
 
@@ -262,4 +267,3 @@ gst_audiofilter_template_filter_inplace (GstAudiofilter *audiofilter,
    * to the destination. */
 
 }
-