Memory leak fixes

Original commit message from CVS:
Memory leak fixes
Allow level to take mono or stereo audio

ChangeLog

@@ -1,3 +1,19 @@
+2004-10-26 Iain <iaingnome@gmail.com>
+
+	* ext/ogg/gstoggdemux.c (gst_ogg_pad_new): Free the tag list.
+	* gst/audioscale/gstaudioscale.c (gst_audioscale_link): Free the copy
+	of the caps.
+	* gst/interleave/interleave.c (interleave_class_init): Hook up release
+	pad.
+	(interleave_release_pad): Remove the pad.
+	* gst/level/gstlevel.c: Allow the level to take 1 or 2 channels.
+	* sys/sunaudio/gstsunaudio.c (gst_sunaudio_setparams): Pay attention to
+	the set device.
+	* sys/xvimage/xvimagesink.c (gst_xvimagesink_get_xv_support): Free the
+	attrs
+	(gst_xvimagesink_xcontext_clear): Free the xcontext.
+	(gst_xvimagesink_finalize): Free the par.
+
 2004-10-26  Ronald S. Bultje  <rbultje@ronald.bitfreak.net>
 
 	* gst/avi/gstavimux.c: (gst_avimux_audsinkconnect),

ext/ogg/gstoggdemux.c

@@ -559,10 +559,12 @@ gst_ogg_demux_src_event (GstPad * pad, GstEvent * event)
 
       GST_OGG_SET_STATE (ogg, GST_OGG_STATE_PLAY);
       FOR_PAD_IN_CURRENT_CHAIN (ogg, pad,
-          pad->flags |= GST_OGG_PAD_NEEDS_DISCONT;);
+          pad->flags |= GST_OGG_PAD_NEEDS_DISCONT;
+          );
       if (GST_EVENT_SEEK_FLAGS (event) & GST_SEEK_FLAG_FLUSH) {
         FOR_PAD_IN_CURRENT_CHAIN (ogg, pad,
-            pad->flags |= GST_OGG_PAD_NEEDS_FLUSH;);
+            pad->flags |= GST_OGG_PAD_NEEDS_FLUSH;
+            );
       }
       GST_DEBUG_OBJECT (ogg,
           "initiating seeking to format %d, offset %" G_GUINT64_FORMAT, format,
@@ -636,7 +638,8 @@ gst_ogg_demux_handle_event (GstPad * pad, GstEvent * event)
       gst_event_unref (event);
       GST_FLAG_UNSET (ogg, GST_OGG_FLAG_WAIT_FOR_DISCONT);
       FOR_PAD_IN_CURRENT_CHAIN (ogg, pad,
-          pad->flags |= GST_OGG_PAD_NEEDS_DISCONT;);
+          pad->flags |= GST_OGG_PAD_NEEDS_DISCONT;
+          );
       break;
     default:
       gst_pad_event_default (pad, event);
@@ -922,7 +925,8 @@ _find_chain_get_unknown_part (GstOggDemux * ogg, gint64 * start, gint64 * end)
   *end = G_MAXINT64;
 
   g_assert (ogg->current_chain >= 0);
-  FOR_PAD_IN_CURRENT_CHAIN (ogg, pad, *start = MAX (*start, pad->end_offset););
+  FOR_PAD_IN_CURRENT_CHAIN (ogg, pad, *start = MAX (*start, pad->end_offset);
+      );
 
   if (ogg->setup_state == SETUP_FIND_LAST_CHAIN) {
     *end = gst_file_pad_get_length (ogg->sinkpad);
@@ -1051,7 +1055,8 @@ _find_streams_check (GstOggDemux * ogg)
   } else {
     endpos = G_MAXINT64;
     FOR_PAD_IN_CHAIN (ogg, pad, ogg->chains->len - 1,
-        endpos = MIN (endpos, pad->start_offset););
+        endpos = MIN (endpos, pad->start_offset);
+        );
   }
   if (!ogg->seek_skipped || gst_ogg_demux_position (ogg) >= endpos) {
     /* have we found the endposition for all streams yet? */
@@ -1240,6 +1245,8 @@ gst_ogg_pad_new (GstOggDemux * ogg, int serial)
   }
   gst_tag_list_add (list, GST_TAG_MERGE_REPLACE, GST_TAG_SERIAL, serial, NULL);
   gst_element_found_tags (GST_ELEMENT (ogg), list);
+  gst_tag_list_free (list);
+
   GST_LOG_OBJECT (ogg, "created new ogg src %p for stream with serial %d", ret,
       serial);
   ret->start_offset = ret->end_offset = -1;

gst/audioscale/gstaudioscale.c

@@ -84,17 +84,17 @@ enum
      {
        static GType audioscale_method_type = 0;
        static GEnumValue audioscale_methods[] =
-       {
+{
-         {
+  {
-         GST_RESAMPLE_NEAREST, "0", "Nearest"}
+  GST_RESAMPLE_NEAREST, "0", "Nearest"}
-         ,
+  ,
-         {
+  {
-         GST_RESAMPLE_BILINEAR, "1", "Bilinear"}
+  GST_RESAMPLE_BILINEAR, "1", "Bilinear"}
-         , {
+  , {
-         GST_RESAMPLE_SINC, "2", "Sinc"}
+  GST_RESAMPLE_SINC, "2", "Sinc"}
-         , {
+  , {
-         0, NULL, NULL}
+  0, NULL, NULL}
-       ,};
+,};
 
        if (!audioscale_method_type) {
          audioscale_method_type = g_enum_register_static ("GstAudioscaleMethod",
@@ -127,8 +127,7 @@ GType audioscale_get_type (void)
 
   if (!audioscale_type)
   {
-    static const GTypeInfo audioscale_info =
+    static const GTypeInfo audioscale_info = {
-    {
     sizeof (AudioscaleClass),
           gst_audioscale_base_init,
           NULL,
@@ -188,8 +187,7 @@ static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
   int rate_min, rate_max;
 
   if (G_VALUE_TYPE (src) == G_TYPE_INT ||
-      G_VALUE_TYPE (src) == GST_TYPE_INT_RANGE)
+      G_VALUE_TYPE (src) == GST_TYPE_INT_RANGE) {
-  {
     if (G_VALUE_TYPE (src) == G_TYPE_INT) {
       rate_min = g_value_get_int (src);
       rate_max = rate_min;
@@ -213,8 +211,7 @@ static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
     g_value_init (dest, GST_TYPE_LIST);
     for (i = 0; i < gst_value_list_get_size (src); i++) {
       const GValue *s = gst_value_list_get_value (src, i);
-      GValue d =
+      GValue d = {
-      {
       0};
       int j;
 
@@ -222,8 +219,7 @@ static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
 
       for (j = 0; j < gst_value_list_get_size (dest); j++) {
         const GValue *s2 = gst_value_list_get_value (dest, j);
-        GValue d2 =
+        GValue d2 = {
-        {
         0};
 
         gst_value_union (&d2, &d, s2);
@@ -242,8 +238,7 @@ static void gst_audioscale_expand_value (GValue * dest, const GValue * src)
 
     if (gst_value_list_get_size (dest) == 1) {
       const GValue *s = gst_value_list_get_value (dest, 0);
-      GValue d =
+      GValue d = {
-      {
       0};
 
       gst_value_init_and_copy (&d, s);
@@ -268,8 +263,7 @@ static void gst_audioscale_expand_caps (GstCaps * caps)
   for (i = 0; i < gst_caps_get_size (caps); i++) {
     GstStructure *structure = gst_caps_get_structure (caps, i);
     const GValue *value;
-    GValue dest =
+    GValue dest = {
-    {
     0};
 
     value = gst_structure_get_value (structure, "rate");
@@ -312,8 +306,7 @@ static GstCaps *gst_audioscale_fixate (GstPad * pad, const GstCaps * caps)
 
     audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
     r = &(audioscale->gst_resample_template);
-  if (pad == audioscale->srcpad)
+  if (pad == audioscale->srcpad) {
-  {
     otherpad = audioscale->sinkpad;
     rate = r->i_rate;
   } else
@@ -351,8 +344,7 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
     audioscale = GST_AUDIOSCALE (gst_pad_get_parent (pad));
     r = &(audioscale->gst_resample_template);
 
-  if (pad == audioscale->srcpad)
+  if (pad == audioscale->srcpad) {
-  {
     otherpad = audioscale->sinkpad;
     rate = &r->o_rate;
     otherrate = &r->i_rate;
@@ -372,6 +364,7 @@ static GstPadLinkReturn gst_audioscale_link (GstPad * pad, const GstCaps * caps)
   copy = gst_caps_copy (caps);
   gst_audioscale_expand_caps (copy);
   link_ret = gst_pad_try_set_caps_nonfixed (otherpad, copy);
+  gst_caps_free (copy);
   if (GST_PAD_LINK_FAILED (link_ret))
     return link_ret;
 

sys/xvimage/xvimagesink.c

@@ -736,7 +736,7 @@ gst_xvimagesink_get_xv_support (GstXvImageSink * xvimagesink,
   /* Set XV_AUTOPAINT_COLORKEY */
   {
     int count;
-    const XvAttribute *const attr = XvQueryPortAttributes (xcontext->disp,
+    XvAttribute *const attr = XvQueryPortAttributes (xcontext->disp,
         xcontext->xv_port_id, &count);
     static const char autopaint[] = "XV_AUTOPAINT_COLORKEY";
 
@@ -747,6 +747,8 @@ gst_xvimagesink_get_xv_support (GstXvImageSink * xvimagesink,
         XvSetPortAttribute (xcontext->disp, xcontext->xv_port_id, atom, 1);
         break;
       }
+
+    XFree (attr);
   }
 
   /* We get all image formats supported by our port */
@@ -1090,6 +1092,7 @@ gst_xvimagesink_xcontext_clear (GstXvImageSink * xvimagesink)
 
   g_mutex_unlock (xvimagesink->x_lock);
 
+  g_free (xvimagesink->xcontext);
   xvimagesink->xcontext = NULL;
 }
 
@@ -1887,6 +1890,7 @@ gst_xvimagesink_finalize (GObject * object)
     xvimagesink->display_name = NULL;
   }
 
+  g_free (xvimagesink->par);
   g_mutex_free (xvimagesink->x_lock);
   g_mutex_free (xvimagesink->pool_lock);