1 files changed, 218 insertions, 203 deletions

diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c
index aeed48f13..1e77bf42f 100644
--- a/gst/mpegaudioparse/gstmpegaudioparse.c
+++ b/gst/mpegaudioparse/gstmpegaudioparse.c
@@ -32,35 +32,30 @@ static GstElementDetails mp3parse_details = {
   "Erik Walthinsen <omega@cse.ogi.edu>"
 };
 
-static GstStaticPadTemplate mp3_src_template =
-GST_STATIC_PAD_TEMPLATE (
-    "src",
+static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/mpeg, "
-      "mpegversion = (int) 1, "
-      "layer = (int) [ 1, 3 ], "
-      "rate = (int) [ 8000, 48000], "
-      "channels = (int) [ 1, 2 ]")
-);
-
-static GstStaticPadTemplate mp3_sink_template =
-GST_STATIC_PAD_TEMPLATE (
-    "sink",
+	"mpegversion = (int) 1, "
+	"layer = (int) [ 1, 3 ], "
+	"rate = (int) [ 8000, 48000], " "channels = (int) [ 1, 2 ]")
+    );
+
+static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/mpeg, "
-      "mpegversion = (int) 1"
-    )
-);
+    GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
+    );
 
 /* GstMPEGAudioParse signals and args */
-enum {
+enum
+{
   /* FILL ME */
   LAST_SIGNAL
 };
 
-enum {
+enum
+{
   ARG_0,
   ARG_SKIP,
   ARG_BIT_RATE,
@@ -68,63 +63,64 @@ enum {
 };
 
 
-static void	gst_mp3parse_class_init		(GstMPEGAudioParseClass *klass);
-static void	gst_mp3parse_base_init		(GstMPEGAudioParseClass *klass);
-static void	gst_mp3parse_init		(GstMPEGAudioParse *mp3parse);
+static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
+static void gst_mp3parse_base_init (GstMPEGAudioParseClass * klass);
+static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse);
 
-static void	gst_mp3parse_chain		(GstPad *pad,GstData *_data);
-static long	bpf_from_header			(GstMPEGAudioParse *parse, unsigned long header);
-static int	head_check			(unsigned long head);
+static void gst_mp3parse_chain (GstPad * pad, GstData * _data);
+static long bpf_from_header (GstMPEGAudioParse * parse, unsigned long header);
+static int head_check (unsigned long head);
 
-static void	gst_mp3parse_set_property		(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
-static void	gst_mp3parse_get_property		(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
-static GstElementStateReturn
-		gst_mp3parse_change_state	(GstElement *element);
+static void gst_mp3parse_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_mp3parse_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static GstElementStateReturn gst_mp3parse_change_state (GstElement * element);
 
 static GstElementClass *parent_class = NULL;
+
 /*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
 
 GType
-gst_mp3parse_get_type(void) {
+gst_mp3parse_get_type (void)
+{
   static GType mp3parse_type = 0;
 
   if (!mp3parse_type) {
     static const GTypeInfo mp3parse_info = {
-      sizeof(GstMPEGAudioParseClass),
-      (GBaseInitFunc)gst_mp3parse_base_init,
+      sizeof (GstMPEGAudioParseClass),
+      (GBaseInitFunc) gst_mp3parse_base_init,
       NULL,
-      (GClassInitFunc)gst_mp3parse_class_init,
+      (GClassInitFunc) gst_mp3parse_class_init,
       NULL,
       NULL,
-      sizeof(GstMPEGAudioParse),
+      sizeof (GstMPEGAudioParse),
       0,
-      (GInstanceInitFunc)gst_mp3parse_init,
+      (GInstanceInitFunc) gst_mp3parse_init,
     };
     mp3parse_type = g_type_register_static (GST_TYPE_ELEMENT,
-					    "GstMPEGAudioParse",
-					    &mp3parse_info, 0);
+	"GstMPEGAudioParse", &mp3parse_info, 0);
   }
   return mp3parse_type;
 }
 
 static guint mp3types_bitrates[2][3][16] =
-{ { {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448, },
-    {0, 32, 48, 56,  64,  80,  96, 112, 128, 160, 192, 224, 256, 320, 384, },
-    {0, 32, 40, 48,  56,  64,  80,  96, 112, 128, 160, 192, 224, 256, 320, } },
-  { {0, 32, 48, 56,  64,  80,  96, 112, 128, 144, 160, 176, 192, 224, 256, },
-    {0,  8, 16, 24,  32,  40,  48,  56,  64,  80,  96, 112, 128, 144, 160, },
-    {0,  8, 16, 24,  32,  40,  48,  56,  64,  80,  96, 112, 128, 144, 160, } },
+    { {{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
+    {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
+    {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}},
+{{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
+    {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
+    {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}},
 };
 
-static guint mp3types_freqs[3][3] =
-{ {44100, 48000, 32000},
-  {22050, 24000, 16000}, 
-  {11025, 12000,  8000}};
+static guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
+{22050, 24000, 16000},
+{11025, 12000, 8000}
+};
 
 static inline guint
-mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
-				   guint *put_channels, guint *put_bitrate,
-				   guint *put_samplerate)
+mp3_type_frame_length_from_header (guint32 header, guint * put_layer,
+    guint * put_channels, guint * put_bitrate, guint * put_samplerate)
 {
   guint length;
   gulong mode, samplerate, bitrate, layer, channels, padding;
@@ -163,7 +159,7 @@ mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
 
   GST_DEBUG ("Calculated mp3 frame length of %u bytes", length);
   GST_DEBUG ("samplerate = %lu, bitrate = %lu, layer = %lu, channels = %lu",
-	     samplerate, bitrate, layer, channels);
+      samplerate, bitrate, layer, channels);
 
   if (put_layer)
     *put_layer = layer;
@@ -204,8 +200,7 @@ mp3_type_frame_length_from_header (guint32 header, guint *put_layer,
 #define GST_MP3_TYPEFIND_MIN_DATA (1440 * (GST_MP3_TYPEFIND_MIN_HEADERS + 1) - 1 + 3)
 
 static GstCaps *
-mp3_caps_create (guint layer, guint channels,
-		 guint bitrate, guint samplerate)
+mp3_caps_create (guint layer, guint channels, guint bitrate, guint samplerate)
 {
   GstCaps *new;
 
@@ -216,15 +211,14 @@ mp3_caps_create (guint layer, guint channels,
 
   new = gst_caps_new_simple ("audio/mpeg",
       "mpegversion", G_TYPE_INT, 1,
-      "layer",       G_TYPE_INT, layer,
-      "rate",        G_TYPE_INT, samplerate,
-      "channels",    G_TYPE_INT, channels, NULL);
+      "layer", G_TYPE_INT, layer,
+      "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
 
   return new;
 }
 
 static void
-gst_mp3parse_base_init (GstMPEGAudioParseClass *klass)
+gst_mp3parse_base_init (GstMPEGAudioParseClass * klass)
 {
   GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
 
@@ -236,22 +230,18 @@ gst_mp3parse_base_init (GstMPEGAudioParseClass *klass)
 }
 
 static void
-gst_mp3parse_class_init (GstMPEGAudioParseClass *klass)
+gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
 
-  gobject_class = (GObjectClass*)klass;
-  gstelement_class = (GstElementClass*)klass;
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
 
-  g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_SKIP,
-    g_param_spec_int("skip","skip","skip",
-                     G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
-  g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_BIT_RATE,
-    g_param_spec_int("bitrate","Bitrate","Bit Rate",
-                     G_MININT,G_MAXINT,0,G_PARAM_READABLE)); /* CHECKME */
+  g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, g_param_spec_int ("skip", "skip", "skip", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));	/* CHECKME */
+  g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE, g_param_spec_int ("bitrate", "Bitrate", "Bit Rate", G_MININT, G_MAXINT, 0, G_PARAM_READABLE));	/* CHECKME */
 
-  parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
+  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
 
   gobject_class->set_property = gst_mp3parse_set_property;
   gobject_class->get_property = gst_mp3parse_get_property;
@@ -260,18 +250,20 @@ gst_mp3parse_class_init (GstMPEGAudioParseClass *klass)
 }
 
 static void
-gst_mp3parse_init (GstMPEGAudioParse *mp3parse)
+gst_mp3parse_init (GstMPEGAudioParse * mp3parse)
 {
-  mp3parse->sinkpad = gst_pad_new_from_template(
-      gst_static_pad_template_get (&mp3_sink_template), "sink");
-  gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->sinkpad);
-
-  gst_pad_set_chain_function(mp3parse->sinkpad,gst_mp3parse_chain);
-  gst_element_set_loop_function (GST_ELEMENT(mp3parse),NULL);
-
-  mp3parse->srcpad = gst_pad_new_from_template(
-      gst_static_pad_template_get (&mp3_src_template), "src");
-  gst_element_add_pad(GST_ELEMENT(mp3parse),mp3parse->srcpad);
+  mp3parse->sinkpad =
+      gst_pad_new_from_template (gst_static_pad_template_get
+      (&mp3_sink_template), "sink");
+  gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
+
+  gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
+  gst_element_set_loop_function (GST_ELEMENT (mp3parse), NULL);
+
+  mp3parse->srcpad =
+      gst_pad_new_from_template (gst_static_pad_template_get
+      (&mp3_src_template), "src");
+  gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
   gst_pad_use_explicit_caps (mp3parse->srcpad);
   /*gst_pad_set_type_id(mp3parse->srcpad, mp3frametype); */
 
@@ -283,71 +275,71 @@ gst_mp3parse_init (GstMPEGAudioParse *mp3parse)
 }
 
 static void
-gst_mp3parse_chain (GstPad *pad, GstData *_data)
+gst_mp3parse_chain (GstPad * pad, GstData * _data)
 {
   GstBuffer *buf = GST_BUFFER (_data);
   GstMPEGAudioParse *mp3parse;
   guchar *data;
-  glong size,offset = 0;
+  glong size, offset = 0;
   guint32 header;
   int bpf;
   GstBuffer *outbuf;
   guint64 last_ts;
 
-  g_return_if_fail(pad != NULL);
-  g_return_if_fail(GST_IS_PAD(pad));
-  g_return_if_fail(buf != NULL);
+  g_return_if_fail (pad != NULL);
+  g_return_if_fail (GST_IS_PAD (pad));
+  g_return_if_fail (buf != NULL);
 /*  g_return_if_fail(GST_IS_BUFFER(buf)); */
 
   mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
 
-  GST_DEBUG ("mp3parse: received buffer of %d bytes",GST_BUFFER_SIZE(buf));
+  GST_DEBUG ("mp3parse: received buffer of %d bytes", GST_BUFFER_SIZE (buf));
 
-  last_ts = GST_BUFFER_TIMESTAMP(buf);
+  last_ts = GST_BUFFER_TIMESTAMP (buf);
 
   /* FIXME, do flush */
   /*
-    if (mp3parse->partialbuf) {
-      gst_buffer_unref(mp3parse->partialbuf);
-      mp3parse->partialbuf = NULL;
-    }
-    mp3parse->in_flush = TRUE;
-    */
+     if (mp3parse->partialbuf) {
+     gst_buffer_unref(mp3parse->partialbuf);
+     mp3parse->partialbuf = NULL;
+     }
+     mp3parse->in_flush = TRUE;
+   */
 
   /* if we have something left from the previous frame */
   if (mp3parse->partialbuf) {
     GstBuffer *newbuf;
 
-    newbuf = gst_buffer_merge(mp3parse->partialbuf, buf);
+    newbuf = gst_buffer_merge (mp3parse->partialbuf, buf);
     /* and the one we received.. */
-    gst_buffer_unref(buf);
-    gst_buffer_unref(mp3parse->partialbuf);
+    gst_buffer_unref (buf);
+    gst_buffer_unref (mp3parse->partialbuf);
     mp3parse->partialbuf = newbuf;
-  }
-  else {
+  } else {
     mp3parse->partialbuf = buf;
   }
 
-  size = GST_BUFFER_SIZE(mp3parse->partialbuf);
-  data = GST_BUFFER_DATA(mp3parse->partialbuf);
+  size = GST_BUFFER_SIZE (mp3parse->partialbuf);
+  data = GST_BUFFER_DATA (mp3parse->partialbuf);
 
   /* while we still have bytes left -4 for the header */
-  while (offset < size-4) {
+  while (offset < size - 4) {
     int skipped = 0;
 
-    GST_DEBUG ("mp3parse: offset %ld, size %ld ",offset, size);
+    GST_DEBUG ("mp3parse: offset %ld, size %ld ", offset, size);
 
     /* search for a possible start byte */
-    for (;((data[offset] != 0xff) && (offset < size));offset++) skipped++;
+    for (; ((data[offset] != 0xff) && (offset < size)); offset++)
+      skipped++;
     if (skipped && !mp3parse->in_flush) {
-      GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes",offset,skipped);
+      GST_DEBUG ("mp3parse: **** now at %ld skipped %d bytes", offset, skipped);
     }
     /* construct the header word */
-    header = GUINT32_FROM_BE(*((guint32 *)(data+offset)));
+    header = GUINT32_FROM_BE (*((guint32 *) (data + offset)));
     /* if it's a valid header, go ahead and send off the frame */
-    if (head_check(header)) {
+    if (head_check (header)) {
       /* calculate the bpf of the frame */
-      bpf = bpf_from_header(mp3parse, header);
+      bpf = bpf_from_header (mp3parse, header);
 
       /********************************************************************************
       * robust seek support
@@ -361,107 +353,116 @@ gst_mp3parse_chain (GstPad *pad, GstData *_data)
       *   from previous frames.  In this case, seeking may be more complicated because
       *   the frames are not independently coded.
       ********************************************************************************/
-      if ( mp3parse->in_flush ) {
-        guint32 header2;
+      if (mp3parse->in_flush) {
+	guint32 header2;
 
-        if ((size-offset)<(bpf+4)) { if (mp3parse->in_flush) break; } /* wait until we have the the entire current frame as well as the next frame header */
-
-        header2 = GUINT32_FROM_BE(*((guint32 *)(data+offset+bpf)));
-        GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d", (unsigned int)header, (unsigned int)header2, bpf );
+	if ((size - offset) < (bpf + 4)) {
+	  if (mp3parse->in_flush)
+	    break;
+	}
+	/* wait until we have the the entire current frame as well as the next frame header */
+	header2 = GUINT32_FROM_BE (*((guint32 *) (data + offset + bpf)));
+	GST_DEBUG ("mp3parse: header=%08X, header2=%08X, bpf=%d",
+	    (unsigned int) header, (unsigned int) header2, bpf);
 
 /* mask the bits which are allowed to differ between frames */
 #define HDRMASK ~((0xF << 12)  /* bitrate */ | \
 		  (0x1 <<  9)  /* padding */ | \
-		  (0x3 <<  4)) /*mode extension*/
-
-        if ( (header2&HDRMASK) != (header&HDRMASK) ) { /* require 2 matching headers in a row */
-           GST_DEBUG ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)", (unsigned int)header, (unsigned int)header2, bpf );
-           offset++; /* This frame is invalid.  Start looking for a valid frame at the next position in the stream */
-           continue;
-        }
+		  (0x3 <<  4))	/*mode extension */
+
+	if ((header2 & HDRMASK) != (header & HDRMASK)) {	/* require 2 matching headers in a row */
+	  GST_DEBUG
+	      ("mp3parse: next header doesn't match (header=%08X, header2=%08X, bpf=%d)",
+	      (unsigned int) header, (unsigned int) header2, bpf);
+	  offset++;		/* This frame is invalid.  Start looking for a valid frame at the next position in the stream */
+	  continue;
+	}
 
       }
 
       /* if we don't have the whole frame... */
       if ((size - offset) < bpf) {
-        GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ",(size-offset), bpf);
+	GST_DEBUG ("mp3parse: partial buffer needed %ld < %d ", (size - offset),
+	    bpf);
 	break;
       } else {
-        guint bitrate, layer, rate, channels;
-
-        if (!mp3_type_frame_length_from_header (header, &layer,
-						&channels,
-						&bitrate, &rate)) {
-	  g_error("Header failed internal error");
-        }
-        if (channels != mp3parse->channels ||
-            rate     != mp3parse->rate     ||
-            layer    != mp3parse->layer    ||
-            bitrate  != mp3parse->bit_rate) {
-          GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
-
-          gst_pad_set_explicit_caps(mp3parse->srcpad, caps);
-
-          mp3parse->channels = channels;
-          mp3parse->layer    = layer;
-          mp3parse->rate     = rate;
-          mp3parse->bit_rate = bitrate;
-        }
-
-        outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,bpf);
-
-        offset += bpf;
+	guint bitrate, layer, rate, channels;
+
+	if (!mp3_type_frame_length_from_header (header, &layer,
+		&channels, &bitrate, &rate)) {
+	  g_error ("Header failed internal error");
+	}
+	if (channels != mp3parse->channels ||
+	    rate != mp3parse->rate ||
+	    layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
+	  GstCaps *caps = mp3_caps_create (layer, channels, bitrate, rate);
+
+	  gst_pad_set_explicit_caps (mp3parse->srcpad, caps);
+
+	  mp3parse->channels = channels;
+	  mp3parse->layer = layer;
+	  mp3parse->rate = rate;
+	  mp3parse->bit_rate = bitrate;
+	}
+
+	outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, bpf);
+
+	offset += bpf;
 	if (mp3parse->skip == 0) {
-          GST_DEBUG ("mp3parse: pushing buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
+	  GST_DEBUG ("mp3parse: pushing buffer of %d bytes",
+	      GST_BUFFER_SIZE (outbuf));
 	  if (mp3parse->in_flush) {
 	    /* FIXME do some sort of flush event */
 	    mp3parse->in_flush = FALSE;
 	  }
-	  GST_BUFFER_TIMESTAMP(outbuf) = last_ts;
-	  GST_BUFFER_DURATION(outbuf) = 8 * (GST_SECOND/1000) * GST_BUFFER_SIZE(outbuf) / mp3parse->bit_rate;
-
-          if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
-            gst_pad_push(mp3parse->srcpad,GST_DATA (outbuf));
-          } else {
-            GST_DEBUG ("No capsnego yet, delaying buffer push");
-            gst_buffer_unref (outbuf);
-          }
-	}
-	else {
-          GST_DEBUG ("mp3parse: skipping buffer of %d bytes",GST_BUFFER_SIZE(outbuf));
-          gst_buffer_unref(outbuf);
+	  GST_BUFFER_TIMESTAMP (outbuf) = last_ts;
+	  GST_BUFFER_DURATION (outbuf) =
+	      8 * (GST_SECOND / 1000) * GST_BUFFER_SIZE (outbuf) /
+	      mp3parse->bit_rate;
+
+	  if (GST_PAD_CAPS (mp3parse->srcpad) != NULL) {
+	    gst_pad_push (mp3parse->srcpad, GST_DATA (outbuf));
+	  } else {
+	    GST_DEBUG ("No capsnego yet, delaying buffer push");
+	    gst_buffer_unref (outbuf);
+	  }
+	} else {
+	  GST_DEBUG ("mp3parse: skipping buffer of %d bytes",
+	      GST_BUFFER_SIZE (outbuf));
+	  gst_buffer_unref (outbuf);
 	  mp3parse->skip--;
 	}
       }
     } else {
       offset++;
-      if (!mp3parse->in_flush) GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
+      if (!mp3parse->in_flush)
+	GST_DEBUG ("mp3parse: *** wrong header, skipping byte (FIXME?)");
     }
   }
   /* if we have processed this block and there are still */
   /* bytes left not in a partial block, copy them over. */
-  if (size-offset > 0) {
+  if (size - offset > 0) {
     glong remainder = (size - offset);
-    GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",remainder);
 
-    outbuf = gst_buffer_create_sub(mp3parse->partialbuf,offset,remainder);
-    gst_buffer_unref(mp3parse->partialbuf);
+    GST_DEBUG ("mp3parse: partial buffer needed %ld for trailing bytes",
+	remainder);
+
+    outbuf = gst_buffer_create_sub (mp3parse->partialbuf, offset, remainder);
+    gst_buffer_unref (mp3parse->partialbuf);
     mp3parse->partialbuf = outbuf;
-  }
-  else {
-    gst_buffer_unref(mp3parse->partialbuf);
+  } else {
+    gst_buffer_unref (mp3parse->partialbuf);
     mp3parse->partialbuf = NULL;
   }
 }
 
 static long
-bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
+bpf_from_header (GstMPEGAudioParse * parse, unsigned long header)
 {
   guint bitrate, layer, rate, channels, length;
 
   if (!(length = mp3_type_frame_length_from_header (header, &layer,
-						    &channels,
-						    &bitrate, &rate))) {
+	      &channels, &bitrate, &rate))) {
     return 0;
   }
 
@@ -471,40 +472,57 @@ bpf_from_header (GstMPEGAudioParse *parse, unsigned long header)
 static gboolean
 head_check (unsigned long head)
 {
-  GST_DEBUG ("checking mp3 header 0x%08lx",head);
+  GST_DEBUG ("checking mp3 header 0x%08lx", head);
   /* if it's not a valid sync */
   if ((head & 0xffe00000) != 0xffe00000) {
-    GST_DEBUG ("invalid sync");return FALSE; }
+    GST_DEBUG ("invalid sync");
+    return FALSE;
+  }
   /* if it's an invalid MPEG version */
   if (((head >> 19) & 3) == 0x1) {
-    GST_DEBUG ("invalid MPEG version");return FALSE; }
+    GST_DEBUG ("invalid MPEG version");
+    return FALSE;
+  }
   /* if it's an invalid layer */
   if (!((head >> 17) & 3)) {
-    GST_DEBUG ("invalid layer");return FALSE; }
+    GST_DEBUG ("invalid layer");
+    return FALSE;
+  }
   /* if it's an invalid bitrate */
   if (((head >> 12) & 0xf) == 0x0) {
-    GST_DEBUG ("invalid bitrate");return FALSE; }
+    GST_DEBUG ("invalid bitrate");
+    return FALSE;
+  }
   if (((head >> 12) & 0xf) == 0xf) {
-    GST_DEBUG ("invalid bitrate");return FALSE; }
+    GST_DEBUG ("invalid bitrate");
+    return FALSE;
+  }
   /* if it's an invalid samplerate */
   if (((head >> 10) & 0x3) == 0x3) {
-    GST_DEBUG ("invalid samplerate");return FALSE; }
-  if ((head & 0xffff0000) == 0xfffe0000) { 
-    GST_DEBUG ("invalid sync");return FALSE; }
+    GST_DEBUG ("invalid samplerate");
+    return FALSE;
+  }
+  if ((head & 0xffff0000) == 0xfffe0000) {
+    GST_DEBUG ("invalid sync");
+    return FALSE;
+  }
   if (head & 0x00000002) {
-        GST_DEBUG ("invalid emphasis");return FALSE; }
+    GST_DEBUG ("invalid emphasis");
+    return FALSE;
+  }
 
   return TRUE;
 }
 
 static void
-gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+gst_mp3parse_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
 {
   GstMPEGAudioParse *src;
 
   /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail(GST_IS_MP3PARSE(object));
-  src = GST_MP3PARSE(object);
+  g_return_if_fail (GST_IS_MP3PARSE (object));
+  src = GST_MP3PARSE (object);
 
   switch (prop_id) {
     case ARG_SKIP:
@@ -516,13 +534,14 @@ gst_mp3parse_set_property (GObject *object, guint prop_id, const GValue *value,
 }
 
 static void
-gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+gst_mp3parse_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
 {
   GstMPEGAudioParse *src;
 
   /* it's not null if we got it, but it might not be ours */
-  g_return_if_fail(GST_IS_MP3PARSE(object));
-  src = GST_MP3PARSE(object);
+  g_return_if_fail (GST_IS_MP3PARSE (object));
+  src = GST_MP3PARSE (object);
 
   switch (prop_id) {
     case ARG_SKIP:
@@ -537,43 +556,39 @@ gst_mp3parse_get_property (GObject *object, guint prop_id, GValue *value, GParam
   }
 }
 
-static GstElementStateReturn 
-gst_mp3parse_change_state (GstElement *element) 
+static GstElementStateReturn
+gst_mp3parse_change_state (GstElement * element)
 {
   GstMPEGAudioParse *src;
 
-  g_return_val_if_fail(GST_IS_MP3PARSE(element), GST_STATE_FAILURE);
-  src = GST_MP3PARSE(element);
+  g_return_val_if_fail (GST_IS_MP3PARSE (element), GST_STATE_FAILURE);
+  src = GST_MP3PARSE (element);
 
   switch (GST_STATE_TRANSITION (element)) {
     case GST_STATE_PAUSED_TO_READY:
-      src->channels = -1; src->rate = -1; src->layer = -1;
+      src->channels = -1;
+      src->rate = -1;
+      src->layer = -1;
       break;
     default:
       break;
   }
 
-  if (GST_ELEMENT_CLASS(parent_class)->change_state)
-    return GST_ELEMENT_CLASS(parent_class)->change_state(element);
+  if (GST_ELEMENT_CLASS (parent_class)->change_state)
+    return GST_ELEMENT_CLASS (parent_class)->change_state (element);
 
   return GST_STATE_SUCCESS;
 }
 
 static gboolean
-plugin_init (GstPlugin *plugin)
+plugin_init (GstPlugin * plugin)
 {
   return gst_element_register (plugin, "mp3parse",
-			       GST_RANK_NONE, GST_TYPE_MP3PARSE);
+      GST_RANK_NONE, GST_TYPE_MP3PARSE);
 }
 
-GST_PLUGIN_DEFINE (
-  GST_VERSION_MAJOR,
-  GST_VERSION_MINOR,
-  "mpegaudioparse",
-  "MPEG-1 layer 1/2/3 audio parser",
-  plugin_init,
-  VERSION,
-  "LGPL",
-  GST_PACKAGE,
-  GST_ORIGIN
-)
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "mpegaudioparse",
+    "MPEG-1 layer 1/2/3 audio parser",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)