directsoundsrc: LF and indent

1 files changed, 635 insertions, 544 deletions

diff --git a/sys/directsound/gstdirectsoundsrc.c b/sys/directsound/gstdirectsoundsrc.c
index 429d6b85..fce9d5c8 100644..100755
--- a/sys/directsound/gstdirectsoundsrc.c
+++ b/sys/directsound/gstdirectsoundsrc.c
@@ -1,587 +1,678 @@
-/*

- * GStreamer

- * Copyright 2005 Thomas Vander Stichele <thomas@apestaart.org>

- * Copyright 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>

- * Copyright 2005 Sébastien Moutte <sebastien@moutte.net>

- * Copyright 2006 Joni Valtanen <joni.valtanen@movial.fi>

- * 

- * Permission is hereby granted, free of charge, to any person obtaining a

- * copy of this software and associated documentation files (the "Software"),

- * to deal in the Software without restriction, including without limitation

- * the rights to use, copy, modify, merge, publish, distribute, sublicense,

- * and/or sell copies of the Software, and to permit persons to whom the

- * Software is furnished to do so, subject to the following conditions:

- *

- * The above copyright notice and this permission notice shall be included in

- * all copies or substantial portions of the Software.

- *

- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR

- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,

- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE

- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER

- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING

- * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER

- * DEALINGS IN THE SOFTWARE.

- *

- * Alternatively, the contents of this file may be used under the

- * GNU Lesser General Public License Version 2.1 (the "LGPL"), in

- * which case the following provisions apply instead of the ones

- * mentioned above:

- *

- * This library is free software; you can redistribute it and/or

- * modify it under the terms of the GNU Library General Public

- * License as published by the Free Software Foundation; either

- * version 2 of the License, or (at your option) any later version.

- *

- * This library is distributed in the hope that it will be useful,

- * but WITHOUT ANY WARRANTY; without even the implied warranty of

- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

- * Library General Public License for more details.

- *

- * You should have received a copy of the GNU Library General Public

- * License along with this library; if not, write to the

- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,

- * Boston, MA 02111-1307, USA.

- */  

-    

-/*

-  TODO: add device selection and check rate etc.

-*/ 

-    

-#ifdef HAVE_CONFIG_H

-#  include "config.h"

-#endif /* 
 */
-    

-#include <gst/gst.h>

-#include <gst/audio/gstbaseaudiosrc.h>

-    

-#include "gstdirectsoundsrc.h"

-    

-#include <windows.h>

-#include <dsound.h>

-    
GST_DEBUG_CATEGORY_STATIC (directsoundsrc_debug);
-

-#define GST_CAT_DEFAULT directsoundsrc_debug

-
static GstElementDetails gst_directsound_src_details =
-    
GST_ELEMENT_DETAILS ("Audio Source (DIRECTCSOUND)", 
"Source/Audio",
-    
"Capture from a soundcard via DIRECTSOUND",
-    
"Joni Valtanen <joni.valtanen@movial.fi>");
-


-/* defaults here */ 

-#define DEFAULT_DEVICE 0

-    

-/* properties */ 

-    enum
-{ 
PROP_0, 
PROP_DEVICE 

+/*
+ * GStreamer
+ * Copyright 2005 Thomas Vander Stichele <thomas@apestaart.org>
+ * Copyright 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
+ * Copyright 2005 Sébastien Moutte <sebastien@moutte.net>
+ * Copyright 2006 Joni Valtanen <joni.valtanen@movial.fi>
+ * 
+ * Permission is hereby granted, free of charge, to any person obtaining a
+ * copy of this software and associated documentation files (the "Software"),
+ * to deal in the Software without restriction, including without limitation
+ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
+ * and/or sell copies of the Software, and to permit persons to whom the
+ * Software is furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
+ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+ * DEALINGS IN THE SOFTWARE.
+ *
+ * Alternatively, the contents of this file may be used under the
+ * GNU Lesser General Public License Version 2.1 (the "LGPL"), in
+ * which case the following provisions apply instead of the ones
+ * mentioned above:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/*
+  TODO: add device selection and check rate etc.
+*/
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <gst/gst.h>
+#include <gst/audio/gstbaseaudiosrc.h>
+
+#include "gstdirectsoundsrc.h"
+
+#include <windows.h>
+#include <dsound.h>
+
+GST_DEBUG_CATEGORY_STATIC (directsoundsrc_debug);
+#define GST_CAT_DEFAULT directsoundsrc_debug
+
+static GstElementDetails gst_directsound_src_details =
+GST_ELEMENT_DETAILS ("Direct Sound Audio Src",
+    "Source/Audio",
+    "Capture from a soundcard via DIRECTSOUND",
+    "Joni Valtanen <joni.valtanen@movial.fi>");
+
+/* defaults here */
+#define DEFAULT_DEVICE 0
+
+/* properties */
+enum
+{
+  PROP_0,
+  PROP_DEVICE
 };
-


static HRESULT (WINAPI * pDSoundCaptureCreate) (LPGUID,
-    
LPDIRECTSOUNDCAPTURE *, 
LPUNKNOWN);
-

static void gst_directsound_src_finalise (GObject * object);
-

static void gst_directsound_src_set_property (GObject * object,
-    
guint prop_id, 
const GValue * value, 
GParamSpec * pspec);
-

static void gst_directsound_src_get_property (GObject * object,
-    
guint prop_id, 
GValue * value, 
GParamSpec * pspec);
-

static gboolean gst_directsound_src_open (GstAudioSrc * asrc);
-
static gboolean gst_directsound_src_close (GstAudioSrc * asrc);
-
static gboolean gst_directsound_src_prepare (GstAudioSrc * asrc,
-    
GstRingBufferSpec * spec);
-
static gboolean gst_directsound_src_unprepare (GstAudioSrc * asrc);
-
static void gst_directsound_src_reset (GstAudioSrc * asrc);
-
static GstCaps *gst_directsound_src_getcaps (GstBaseSrc * bsrc);
-

static guint gst_directsound_src_read (GstAudioSrc * asrc, 
gpointer data,
-    
guint length);
-

static void gst_directsound_src_dispose (GObject * object);
-

static void gst_directsound_src_do_init (GType type);
-

static guint gst_directsound_src_delay (GstAudioSrc * asrc);
-

static GstStaticPadTemplate directsound_src_src_factory =
-    
GST_STATIC_PAD_TEMPLATE (
"src", 
GST_PAD_SRC, 
GST_PAD_ALWAYS,
-    
GST_STATIC_CAPS ("audio/x-raw-int, " 

-        "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " 

-        "signed = (boolean) { TRUE, FALSE }, " 
 "width = (int) 16, " 

-        "depth = (int) 16, " 
 "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, 2 ]; " 
 "audio/x-raw-int, " 

-        "signed = (boolean) { TRUE, FALSE }, " 
 "width = (int) 8, " 

-        "depth = (int) 8, " 
 "rate = (int) [ 1, MAX ], "
-        "channels = (int) [ 1, 2 ]"));
-

static void 

-gst_directsound_src_do_init (GType type) 

+
+
+static HRESULT (WINAPI * pDSoundCaptureCreate) (LPGUID,
+    LPDIRECTSOUNDCAPTURE *, LPUNKNOWN);
+
+static void gst_directsound_src_finalise (GObject * object);
+
+static void gst_directsound_src_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+
+static void gst_directsound_src_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_directsound_src_open (GstAudioSrc * asrc);
+static gboolean gst_directsound_src_close (GstAudioSrc * asrc);
+static gboolean gst_directsound_src_prepare (GstAudioSrc * asrc,
+    GstRingBufferSpec * spec);
+static gboolean gst_directsound_src_unprepare (GstAudioSrc * asrc);
+static void gst_directsound_src_reset (GstAudioSrc * asrc);
+static GstCaps *gst_directsound_src_getcaps (GstBaseSrc * bsrc);
+
+static guint gst_directsound_src_read (GstAudioSrc * asrc,
+    gpointer data, guint length);
+
+static void gst_directsound_src_dispose (GObject * object);
+
+static void gst_directsound_src_do_init (GType type);
+
+static guint gst_directsound_src_delay (GstAudioSrc * asrc);
+
+static GstStaticPadTemplate directsound_src_src_factory =
+    GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
+        "signed = (boolean) { TRUE, FALSE }, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
+        "audio/x-raw-int, "
+        "signed = (boolean) { TRUE, FALSE }, "
+        "width = (int) 8, "
+        "depth = (int) 8, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
+
+static void
+gst_directsound_src_do_init (GType type)
 {
-  
GST_DEBUG_CATEGORY_INIT (directsoundsrc_debug, "directsoundsrc", 0,
-      
"DirectSound Src");
-


} 

GST_BOILERPLATE_FULL (GstDirectSoundSrc, gst_directsound_src,
-    GstAudioSrc, 
GST_TYPE_AUDIO_SRC, gst_directsound_src_do_init);
-

static void 

-gst_directsound_src_dispose (GObject * object) 

+  GST_DEBUG_CATEGORY_INIT (directsoundsrc_debug, "directsoundsrc", 0,
+      "DirectSound Src");
+
+
+}
+
+GST_BOILERPLATE_FULL (GstDirectSoundSrc, gst_directsound_src, GstAudioSrc,
+    GST_TYPE_AUDIO_SRC, gst_directsound_src_do_init);
+
+static void
+gst_directsound_src_dispose (GObject * object)
 {
-  
G_OBJECT_CLASS (parent_class)->dispose (object);
-
} 

static void 

+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
 
-gst_directsound_src_finalise (GObject * object) 

+static void
+gst_directsound_src_finalise (GObject * object)
 {
-  
GstDirectSoundSrc * dsoundsrc = GST_DIRECTSOUND_SRC (object);
-  

g_mutex_free (dsoundsrc->dsound_lock);
-
} 

static void 

+  GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (object);
+
+  g_mutex_free (dsoundsrc->dsound_lock);
+}
 
-gst_directsound_src_base_init (gpointer g_class) 

+static void
+gst_directsound_src_base_init (gpointer g_class)
 {
-  
GstElementClass * element_class = GST_ELEMENT_CLASS (g_class);
-  

GST_DEBUG ("initializing directsoundsrc base\n");
-  

gst_element_class_set_details (element_class, &gst_directsound_src_details);
-  

gst_element_class_add_pad_template (element_class,
-      
gst_static_pad_template_get (&directsound_src_src_factory));
-
} 



-
-/* initialize the plugin's class */ 

-static void 

-gst_directsound_src_class_init (GstDirectSoundSrcClass * klass) 

+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  GST_DEBUG ("initializing directsoundsrc base\n");
+
+  gst_element_class_set_details (element_class, &gst_directsound_src_details);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&directsound_src_src_factory));
+}
+
+
+/* initialize the plugin's class */
+static void
+gst_directsound_src_class_init (GstDirectSoundSrcClass * klass)
 {
-  
GObjectClass * gobject_class;
-  
GstElementClass * gstelement_class;
-  
GstBaseSrcClass * gstbasesrc_class;
-  
GstBaseAudioSrcClass * gstbaseaudiosrc_class;
-  
GstAudioSrcClass * gstaudiosrc_class;
-  

gobject_class = (GObjectClass *) klass;
-  
gstelement_class = (GstElementClass *) klass;
-  
gstbasesrc_class = (GstBaseSrcClass *) klass;
-  
gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
-  
gstaudiosrc_class = (GstAudioSrcClass *) klass;
-  

GST_DEBUG ("initializing directsoundsrc class\n");
-  

gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_src_finalise);
-  
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_directsound_src_dispose);
-  
gobject_class->get_property =
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSrcClass *gstbasesrc_class;
+  GstBaseAudioSrcClass *gstbaseaudiosrc_class;
+  GstAudioSrcClass *gstaudiosrc_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasesrc_class = (GstBaseSrcClass *) klass;
+  gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
+  gstaudiosrc_class = (GstAudioSrcClass *) klass;
+
+  GST_DEBUG ("initializing directsoundsrc class\n");
+
+  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_src_finalise);
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_directsound_src_dispose);
+  gobject_class->get_property =
       GST_DEBUG_FUNCPTR (gst_directsound_src_get_property);
-  
gobject_class->set_property =
+  gobject_class->set_property =
       GST_DEBUG_FUNCPTR (gst_directsound_src_set_property);
-  

gstbasesrc_class->get_caps =
-      GST_DEBUG_FUNCPTR (gst_directsound_src_getcaps);
-  

gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_directsound_src_open);
-  
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_directsound_src_close);
-  
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_directsound_src_read);
-  
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_directsound_src_prepare);
-  
gstaudiosrc_class->unprepare =
+
+  gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_directsound_src_getcaps);
+
+  gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_directsound_src_open);
+  gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_directsound_src_close);
+  gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_directsound_src_read);
+  gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_directsound_src_prepare);
+  gstaudiosrc_class->unprepare =
       GST_DEBUG_FUNCPTR (gst_directsound_src_unprepare);
-  
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_src_delay);
-  
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_src_reset);
-


} 

static GstCaps *

+  gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_src_delay);
+  gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_src_reset);
+
+
+}
 
-gst_directsound_src_getcaps (GstBaseSrc * bsrc) 

+static GstCaps *
+gst_directsound_src_getcaps (GstBaseSrc * bsrc)
 {
-  
GstDirectSoundSrc * dsoundsrc;
-  
GstCaps * caps = NULL;
-  
GST_DEBUG ("get caps\n");
-  

dsoundsrc = GST_DIRECTSOUND_SRC (bsrc);
-  


caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD 

+  GstDirectSoundSrc *dsoundsrc;
+  GstCaps *caps = NULL;
+  GST_DEBUG ("get caps\n");
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (bsrc);
+
+
+  caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD
           (bsrc)));
-  
return caps;
-

}
+  return caps;
+
+}
 
-

static void 

+static void
 gst_directsound_src_set_property (GObject * object, guint prop_id,
-    
const GValue * value, GParamSpec * pspec) 

+    const GValue * value, GParamSpec * pspec)
 {
-  

-      //  GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);

-      GST_DEBUG ("set property\n");
-  

switch (prop_id)
-    
 {
-      

-#if 0

-          /* FIXME */ 

+  //  GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
+  GST_DEBUG ("set property\n");
+
+  switch (prop_id) {
+#if 0
+      /* FIXME */
     case PROP_DEVICE:
-      
src->device = g_value_get_uint (value);
-      
break;
-      

-#endif /* 
 */
+      src->device = g_value_get_uint (value);
+      break;
+#endif
     default:
-      
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      
break;
-    
}
-
}
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
 
-

static void 

+static void
 gst_directsound_src_get_property (GObject * object, guint prop_id,
-    
GValue * value, GParamSpec * pspec) 

+    GValue * value, GParamSpec * pspec)
 {
-  

-#if 0

-      GstDirectSoundSrc * src = GST_DIRECTSOUND_SRC (object);
-  

-#endif /* 
 */
-      
GST_DEBUG ("get property\n");
-  

switch (prop_id) {
-      

-#if 0

-          /* FIXME */ 

+#if 0
+  GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
+#endif
+
+  GST_DEBUG ("get property\n");
+
+  switch (prop_id) {
+#if 0
+      /* FIXME */
     case PROP_DEVICE:
-      
g_value_set_uint (value, src->device);
-      
break;
-      

-#endif /* 
 */
+      g_value_set_uint (value, src->device);
+      break;
+#endif
     default:
-      
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      
break;
-  
}
-
}
-
-



-/* initialize the new element

- * instantiate pads and add them to element

- * set functions

- * initialize structure

- */ 

-static void 

+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+
+/* initialize the new element
+ * instantiate pads and add them to element
+ * set functions
+ * initialize structure
+ */
+static void
 gst_directsound_src_init (GstDirectSoundSrc * src,
-    
GstDirectSoundSrcClass * gclass) 

+    GstDirectSoundSrcClass * gclass)
 {
-  
GST_DEBUG ("initializing directsoundsrc\n");
-  
src->dsound_lock = g_mutex_new ();
-
} 



static gboolean 

+  GST_DEBUG ("initializing directsoundsrc\n");
+  src->dsound_lock = g_mutex_new ();
+}
+
+
 
-gst_directsound_src_open (GstAudioSrc * asrc) 

+static gboolean
+gst_directsound_src_open (GstAudioSrc * asrc)
 {
-  
GstDirectSoundSrc * dsoundsrc;
-  
HRESULT hRes;                /* Result for windows functions */
-  

GST_DEBUG ("initializing directsoundsrc\n");
-  

dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
-  


-      /* Open dsound.dll */ 

-      dsoundsrc->DSoundDLL = LoadLibrary ("dsound.dll");
-  
if (!dsoundsrc->DSoundDLL) {
-    
goto dsound_open;
-  
}
-  


-      /* Building the DLL Calls */ 

-      pDSoundCaptureCreate = 

+  GstDirectSoundSrc *dsoundsrc;
+  HRESULT hRes;                 /* Result for windows functions */
+
+  GST_DEBUG ("initializing directsoundsrc\n");
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
+
+  /* Open dsound.dll */
+  dsoundsrc->DSoundDLL = LoadLibrary ("dsound.dll");
+  if (!dsoundsrc->DSoundDLL) {
+    goto dsound_open;
+  }
+
+  /* Building the DLL Calls */
+  pDSoundCaptureCreate =
       (void *) GetProcAddress (dsoundsrc->DSoundDLL,
-      
TEXT ("DirectSoundCaptureCreate"));
-  


-      /* If everything is not ok */ 

-      if (!pDSoundCaptureCreate) {
-    
goto capture_function;
-  
}
-  


-      /* FIXME: add here device selection */ 

-      /* Create capture object */ 

-      hRes = pDSoundCaptureCreate (NULL, &dsoundsrc->pDSC, NULL);
-  
if (FAILED (hRes)) {
-    
goto capture_object;
-  
}
-  

return TRUE;
-

capture_function:
 {
-    
FreeLibrary (dsoundsrc->DSoundDLL);
-    
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ, 

+      TEXT ("DirectSoundCaptureCreate"));
+
+  /* If everything is not ok */
+  if (!pDSoundCaptureCreate) {
+    goto capture_function;
+  }
+
+  /* FIXME: add here device selection */
+  /* Create capture object */
+  hRes = pDSoundCaptureCreate (NULL, &dsoundsrc->pDSC, NULL);
+  if (FAILED (hRes)) {
+    goto capture_object;
+  }
+
+  return TRUE;
+
+capture_function:
+  {
+    FreeLibrary (dsoundsrc->DSoundDLL);
+    GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
         ("Unable to get capturecreate function"), (NULL));
-    
return FALSE;
-  
}
-
capture_object:
 {
-    
FreeLibrary (dsoundsrc->DSoundDLL);
-    
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ, 

+    return FALSE;
+  }
+capture_object:
+  {
+    FreeLibrary (dsoundsrc->DSoundDLL);
+    GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
         ("Unable to create capture object"), (NULL));
-    
return FALSE;
-  
}
-
dsound_open:
 {
-    
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ, 

+    return FALSE;
+  }
+dsound_open:
+  {
+    DWORD err = GetLastError ();
+    GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
         ("Unable to open dsound.dll"), (NULL));
-    
return FALSE;
-  
}
-
}
+    g_print ("0x%x\n", HRESULT_FROM_WIN32 (err));
+    return FALSE;
+  }
+}
 
-

static gboolean 

-gst_directsound_src_close (GstAudioSrc * asrc) 

+static gboolean
+gst_directsound_src_close (GstAudioSrc * asrc)
 {
-  
GstDirectSoundSrc * dsoundsrc;
-  
HRESULT hRes;                /* Result for windows functions */
-  

GST_DEBUG ("initializing directsoundsrc\n");
-  

dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
-  


-      /* Release capture handler  */ 

-      hRes = IDirectSoundCapture_Release (dsoundsrc->pDSC);
-  


-      /* Close library */ 

-      FreeLibrary (dsoundsrc->DSoundDLL);
-  

return TRUE;
-
}
-
-

static gboolean 

-gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec) 

+  GstDirectSoundSrc *dsoundsrc;
+  HRESULT hRes;                 /* Result for windows functions */
+
+  GST_DEBUG ("initializing directsoundsrc\n");
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
+
+  /* Release capture handler  */
+  hRes = IDirectSoundCapture_Release (dsoundsrc->pDSC);
+
+  /* Close library */
+  FreeLibrary (dsoundsrc->DSoundDLL);
+
+  return TRUE;
+}
+
+static gboolean
+gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
 {
-  

GstDirectSoundSrc * dsoundsrc;
-  
WAVEFORMATEX wfx;            /* Wave format structure */
-  
HRESULT hRes;                /* Result for windows functions */
-  
DSCBUFFERDESC descSecondary; /* Capturebuffer decsiption */
-  

int fmt = 0;                /* audio format */
-  

dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
-  

GST_DEBUG ("initializing directsoundsrc\n");
-  


-      /* Define buffer */ 

-      memset (&wfx, 0, sizeof (WAVEFORMATEX));
-  
wfx.wFormatTag = WAVE_FORMAT_PCM;    /* should be WAVE_FORMAT_PCM */
-  
wfx.nChannels = spec->channels;
-  
wfx.nSamplesPerSec = spec->rate;     /* 8000|11025|22050|44100 */
-  
wfx.wBitsPerSample = spec->width;    // 8|16;

-  
wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8);
-  
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
-  
wfx.cbSize = 0;              /* This size is allways for PCM-format */
-  


-      /* 1 or 2 Channels etc...

-         FIXME: Never really tested. Is this ok?

-       */ 

-      if (spec->width == 16 && spec->channels == 1) {
-    
spec->format = GST_S16_LE;
-  
} else if (spec->width == 16 && spec->channels == 2) {
-    
spec->format = GST_U16_LE;
-  
} else if (spec->width == 8 && spec->channels == 1) {
-    
spec->format = GST_S8;
-  
} else if (spec->width == 8 && spec->channels == 2) {
-    
spec->format = GST_U8;
-  
}
-  


-      /* Set the buffer size to two seconds. 

-         This should never reached. 

-       */ 

-      dsoundsrc->buffer_size = wfx.nAvgBytesPerSec * 2;
-  


-      //notifysize * 16; //spec->width; /*original 16*/

-      GST_DEBUG ("Buffer size: %d", dsoundsrc->buffer_size);
-  


-      /* Init secondary buffer desciption */ 

-      memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
-  
descSecondary.dwSize = sizeof (DSCBUFFERDESC);
-  
descSecondary.dwFlags = 0;
-  
descSecondary.dwReserved = 0;
-  


-      /* This is not primary buffer so have to set size  */ 

-      descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
-  
descSecondary.lpwfxFormat = &wfx;
-  


-      /* Create buffer */ 

-      hRes =
-      IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC, 
&descSecondary,
-      
&dsoundsrc->pDSBSecondary, 
NULL);
-  
if (hRes != DS_OK) {
-    
goto capture_buffer;
-  
}
-  

spec->channels = wfx.nChannels;
-  
spec->rate = wfx.nSamplesPerSec;
-  
spec->bytes_per_sample = (spec->width / 8) * spec->channels;
-  
dsoundsrc->bytes_per_sample = spec->bytes_per_sample;
-  

GST_DEBUG ("latency time: %llu - buffer time: %llu", 
spec->latency_time,
-      spec->buffer_time);
-  


-      /* Buffer-time should be allways more than 2*latency */ 

-      if (spec->buffer_time < spec->latency_time * 2) {
-    
spec->buffer_time = spec->latency_time * 2;
-    
GST_WARNING ("buffer-time was less than latency");
-  
}
-  


-      /* Save the times */ 

-      dsoundsrc->buffer_time = spec->buffer_time;
-  
dsoundsrc->latency_time = spec->latency_time;
-  

dsoundsrc->latency_size =
-      (gint) wfx.nAvgBytesPerSec * 
dsoundsrc->latency_time / 1000000.0;
-  


spec->segsize =
-      (guint) (((double) spec->buffer_time / 1000000.0) * 
wfx.nAvgBytesPerSec);
-  


-      /* just in case */ 

-      if (spec->segsize < 1)
-    
spec->segsize = 1;
-  

spec->segtotal = spec->width * (wfx.nAvgBytesPerSec / spec->segsize);
-  

GST_DEBUG ("bytes/sec: %d, buffer size: %d, segsize: %d, segtotal: %d",
-      
wfx.nAvgBytesPerSec, 
dsoundsrc->buffer_size, 
spec->segsize,
-      
spec->segtotal);
-  

spec->silence_sample[0] = 0;
-  
spec->silence_sample[1] = 0;
-  
spec->silence_sample[2] = 0;
-  
spec->silence_sample[3] = 0;
-  

if (spec->width != 16 && spec->width != 8)
-    
goto dodgy_width;
-  


-      /* Not readed anything yet */ 

-      dsoundsrc->current_circular_offset = 0;
-  

GST_DEBUG ("GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, \

-GstRingBufferSpec->bytes_per_sample: %d\n\

-WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, \

-WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n", 
spec->channels, spec->rate, spec->bytes_per_sample, 
wfx.nSamplesPerSec, wfx.wBitsPerSample, 
wfx.nBlockAlign, wfx.nAvgBytesPerSec);
-  

return TRUE;
-

wrong_format:
 {
-    
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ, 

+
+  GstDirectSoundSrc *dsoundsrc;
+  WAVEFORMATEX wfx;             /* Wave format structure */
+  HRESULT hRes;                 /* Result for windows functions */
+  DSCBUFFERDESC descSecondary;  /* Capturebuffer decsiption */
+
+  int fmt = 0;                  /* audio format */
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
+
+  GST_DEBUG ("initializing directsoundsrc\n");
+
+  /* Define buffer */
+  memset (&wfx, 0, sizeof (WAVEFORMATEX));
+  wfx.wFormatTag = WAVE_FORMAT_PCM;     /* should be WAVE_FORMAT_PCM */
+  wfx.nChannels = spec->channels;
+  wfx.nSamplesPerSec = spec->rate;      /* 8000|11025|22050|44100 */
+  wfx.wBitsPerSample = spec->width;     // 8|16;
+
+  wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8);
+  wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
+  wfx.cbSize = 0;               /* This size is allways for PCM-format */
+
+  /* 1 or 2 Channels etc...
+     FIXME: Never really tested. Is this ok?
+   */
+  if (spec->width == 16 && spec->channels == 1) {
+    spec->format = GST_S16_LE;
+  } else if (spec->width == 16 && spec->channels == 2) {
+    spec->format = GST_U16_LE;
+  } else if (spec->width == 8 && spec->channels == 1) {
+    spec->format = GST_S8;
+  } else if (spec->width == 8 && spec->channels == 2) {
+    spec->format = GST_U8;
+  }
+
+  /* Set the buffer size to two seconds. 
+     This should never reached. 
+   */
+  dsoundsrc->buffer_size = wfx.nAvgBytesPerSec * 2;
+
+  //notifysize * 16; //spec->width; /*original 16*/
+  GST_DEBUG ("Buffer size: %d", dsoundsrc->buffer_size);
+
+  /* Init secondary buffer desciption */
+  memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
+  descSecondary.dwSize = sizeof (DSCBUFFERDESC);
+  descSecondary.dwFlags = 0;
+  descSecondary.dwReserved = 0;
+
+  /* This is not primary buffer so have to set size  */
+  descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
+  descSecondary.lpwfxFormat = &wfx;
+
+  /* Create buffer */
+  hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
+      &descSecondary, &dsoundsrc->pDSBSecondary, NULL);
+  if (hRes != DS_OK) {
+    goto capture_buffer;
+  }
+
+  spec->channels = wfx.nChannels;
+  spec->rate = wfx.nSamplesPerSec;
+  spec->bytes_per_sample = (spec->width / 8) * spec->channels;
+  dsoundsrc->bytes_per_sample = spec->bytes_per_sample;
+
+  GST_DEBUG ("latency time: %llu - buffer time: %llu",
+      spec->latency_time, spec->buffer_time);
+
+  /* Buffer-time should be allways more than 2*latency */
+  if (spec->buffer_time < spec->latency_time * 2) {
+    spec->buffer_time = spec->latency_time * 2;
+    GST_WARNING ("buffer-time was less than latency");
+  }
+
+  /* Save the times */
+  dsoundsrc->buffer_time = spec->buffer_time;
+  dsoundsrc->latency_time = spec->latency_time;
+
+  dsoundsrc->latency_size = (gint) wfx.nAvgBytesPerSec *
+      dsoundsrc->latency_time / 1000000.0;
+
+
+  spec->segsize = (guint) (((double) spec->buffer_time / 1000000.0) *
+      wfx.nAvgBytesPerSec);
+
+  /* just in case */
+  if (spec->segsize < 1)
+    spec->segsize = 1;
+
+  spec->segtotal = spec->width * (wfx.nAvgBytesPerSec / spec->segsize);
+
+  GST_DEBUG ("bytes/sec: %d, buffer size: %d, segsize: %d, segtotal: %d",
+      wfx.nAvgBytesPerSec,
+      dsoundsrc->buffer_size, spec->segsize, spec->segtotal);
+
+  spec->silence_sample[0] = 0;
+  spec->silence_sample[1] = 0;
+  spec->silence_sample[2] = 0;
+  spec->silence_sample[3] = 0;
+
+  if (spec->width != 16 && spec->width != 8)
+    goto dodgy_width;
+
+  /* Not readed anything yet */
+  dsoundsrc->current_circular_offset = 0;
+
+  GST_DEBUG ("GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, \
+GstRingBufferSpec->bytes_per_sample: %d\n\
+WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, \
+WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n", spec->channels, spec->rate, spec->bytes_per_sample, wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nBlockAlign, wfx.nAvgBytesPerSec);
+
+  return TRUE;
+
+wrong_format:
+  {
+    GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
         ("Unable to get format %d", spec->format), (NULL));
-    
return FALSE;
-  
}
-
capture_buffer:
 {
-    
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ, 

+    return FALSE;
+  }
+capture_buffer:
+  {
+    GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
         ("Unable to create capturebuffer"), (NULL));
-    
return FALSE;
-  
}
-
dodgy_width:
 {
-    
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ, 

+    return FALSE;
+  }
+dodgy_width:
+  {
+    GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
         ("Unexpected width %d", spec->width), (NULL));
-    
return FALSE;
-  
}
-

}
+    return FALSE;
+  }
 
-

static gboolean 

-gst_directsound_src_unprepare (GstAudioSrc * asrc) 

+}
+
+static gboolean
+gst_directsound_src_unprepare (GstAudioSrc * asrc)
 {
-  
GstDirectSoundSrc * dsoundsrc;
-  

HRESULT hRes;               /* Result for windows functions */
-  


-      /* Resets */ 

-      GST_DEBUG ("unpreparing directsoundsrc");
-  

dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
-  


-      /* Stop capturing */ 

-      hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
-  


-      /* Release buffer  */ 

-      hRes = IDirectSoundCaptureBuffer_Release (dsoundsrc->pDSBSecondary);
-  

return TRUE;
-

}
-
-


-/* 

-return number of readed bytes */ 

-static guint 

-gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length) 

+  GstDirectSoundSrc *dsoundsrc;
+
+  HRESULT hRes;                 /* Result for windows functions */
+
+  /* Resets */
+  GST_DEBUG ("unpreparing directsoundsrc");
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
+
+  /* Stop capturing */
+  hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
+
+  /* Release buffer  */
+  hRes = IDirectSoundCaptureBuffer_Release (dsoundsrc->pDSBSecondary);
+
+  return TRUE;
+
+}
+
+/* 
+return number of readed bytes */
+static guint
+gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
 {
-  
GstDirectSoundSrc * dsoundsrc;
-  

HRESULT hRes;               /* Result for windows functions */
-  
DWORD dwCurrentCaptureCursor = 0;
-  
DWORD dwBufferSize = 0;
-  

LPVOID pLockedBuffer1 = NULL;
-  
LPVOID pLockedBuffer2 = NULL;
-  
DWORD dwSizeBuffer1 = 0;
-  
DWORD dwSizeBuffer2 = 0;
-  

DWORD dwStatus = 0;
-  

GST_DEBUG ("reading directsoundsrc\n");
-  

dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
-  

GST_DSOUND_LOCK (dsoundsrc);
-  


-      /* Get current buffer status */ 

-      hRes =
-      IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
-      
&dwStatus);
-  


-      /* Starting capturing if not allready */ 

-      if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
-    
hRes =
-        IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
-        
DSCBSTART_LOOPING);
-    

-        //    Sleep (dsoundsrc->latency_time/1000);

-        GST_DEBUG ("capture started");
-  
}
-  


-      //  calculate_buffersize:

-      while (length > dwBufferSize) {
-    
Sleep (dsoundsrc->latency_time / 1000);
-    

hRes =
-        
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
-        
&dwCurrentCaptureCursor, 
NULL);
-    


-        /* calculate the buffer */ 

-        if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
-      
dwBufferSize = dsoundsrc->buffer_size - 

-          (dsoundsrc->current_circular_offset - 
dwCurrentCaptureCursor);
-    
} else {
-      
dwBufferSize =
-          
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
-    
}
-  


}                          // while (...

-  

-      /* Lock the buffer */ 

-      hRes =
-      IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
-      
dsoundsrc->current_circular_offset, 
length, 
&pLockedBuffer1,
-      
&dwSizeBuffer1, 
&pLockedBuffer2, 
&dwSizeBuffer2, 
0L);
-  


-      /* Copy buffer data to another buffer */ 

-      if (hRes == DS_OK) {
-    
memcpy (data, pLockedBuffer1, dwSizeBuffer1);
-  
}
-  


-      /* ...and if something is in another buffer */ 

-      if (pLockedBuffer2 != NULL) {
-    
memcpy ((data + dwSizeBuffer1), pLockedBuffer2, dwSizeBuffer2);
-  
}
-  

dsoundsrc->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
-  
dsoundsrc->current_circular_offset %= dsoundsrc->buffer_size;
-  

IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary, 
pLockedBuffer1,
-      
dwSizeBuffer1, 
pLockedBuffer2, 
dwSizeBuffer2);
-  

GST_DSOUND_UNLOCK (dsoundsrc);
-  


-      /* return length (readed data size in bytes) */ 

-      return length;
-

}
-
-

static guint 

-gst_directsound_src_delay (GstAudioSrc * asrc) 

+  GstDirectSoundSrc *dsoundsrc;
+
+  HRESULT hRes;                 /* Result for windows functions */
+  DWORD dwCurrentCaptureCursor = 0;
+  DWORD dwBufferSize = 0;
+
+  LPVOID pLockedBuffer1 = NULL;
+  LPVOID pLockedBuffer2 = NULL;
+  DWORD dwSizeBuffer1 = 0;
+  DWORD dwSizeBuffer2 = 0;
+
+  DWORD dwStatus = 0;
+
+  GST_DEBUG ("reading directsoundsrc\n");
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
+
+  GST_DSOUND_LOCK (dsoundsrc);
+
+  /* Get current buffer status */
+  hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
+      &dwStatus);
+
+  /* Starting capturing if not allready */
+  if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
+    hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
+        DSCBSTART_LOOPING);
+    //    Sleep (dsoundsrc->latency_time/1000);
+    GST_DEBUG ("capture started");
+  }
+  //  calculate_buffersize:
+  while (length > dwBufferSize) {
+    Sleep (dsoundsrc->latency_time / 1000);
+
+    hRes =
+        IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
+        &dwCurrentCaptureCursor, NULL);
+
+    /* calculate the buffer */
+    if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
+      dwBufferSize = dsoundsrc->buffer_size -
+          (dsoundsrc->current_circular_offset - dwCurrentCaptureCursor);
+    } else {
+      dwBufferSize =
+          dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
+    }
+
+
+  }                             // while (...
+
+  /* Lock the buffer */
+  hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
+      dsoundsrc->current_circular_offset,
+      length,
+      &pLockedBuffer1, &dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
+
+  /* Copy buffer data to another buffer */
+  if (hRes == DS_OK) {
+    memcpy (data, pLockedBuffer1, dwSizeBuffer1);
+  }
+
+  /* ...and if something is in another buffer */
+  if (pLockedBuffer2 != NULL) {
+    memcpy (((guchar *) data + dwSizeBuffer1), pLockedBuffer2, dwSizeBuffer2);
+  }
+
+  dsoundsrc->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
+  dsoundsrc->current_circular_offset %= dsoundsrc->buffer_size;
+
+  IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
+      pLockedBuffer1, dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
+
+  GST_DSOUND_UNLOCK (dsoundsrc);
+
+  /* return length (readed data size in bytes) */
+  return length;
+
+}
+
+static guint
+gst_directsound_src_delay (GstAudioSrc * asrc)
 {
-  
GstDirectSoundSrc * dsoundsrc;
-  
HRESULT hRes;
-  
DWORD dwCurrentCaptureCursor;
-  
DWORD dwBytesInQueue = 0;
-  
gint nNbSamplesInQueue = 0;
-  

GST_DEBUG ("Delay\n");
-  

dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
-  


-      /* evaluate the number of samples in queue in the circular buffer */ 

-      hRes =
-      
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
-      
&dwCurrentCaptureCursor, 
NULL);
-  

-      /* FIXME: Check is this calculated right */ 

-      if (hRes == S_OK) {
-    
if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
-      
dwBytesInQueue =
-          
dsoundsrc->buffer_size - (dsoundsrc->current_circular_offset -
-          
dwCurrentCaptureCursor);
-    
} else {
-      
dwBytesInQueue =
-          
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
-    
}
-    

nNbSamplesInQueue = dwBytesInQueue / dsoundsrc->bytes_per_sample;
-  
}
-  

return nNbSamplesInQueue;
-
}
-
-

static void 

-gst_directsound_src_reset (GstAudioSrc * asrc) 

+  GstDirectSoundSrc *dsoundsrc;
+  HRESULT hRes;
+  DWORD dwCurrentCaptureCursor;
+  DWORD dwBytesInQueue = 0;
+  gint nNbSamplesInQueue = 0;
+
+  GST_DEBUG ("Delay\n");
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
+
+  /* evaluate the number of samples in queue in the circular buffer */
+  hRes =
+      IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
+      &dwCurrentCaptureCursor, NULL);
+  /* FIXME: Check is this calculated right */
+  if (hRes == S_OK) {
+    if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
+      dwBytesInQueue =
+          dsoundsrc->buffer_size - (dsoundsrc->current_circular_offset -
+          dwCurrentCaptureCursor);
+    } else {
+      dwBytesInQueue =
+          dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
+    }
+
+    nNbSamplesInQueue = dwBytesInQueue / dsoundsrc->bytes_per_sample;
+  }
+
+  return nNbSamplesInQueue;
+}
+
+static void
+gst_directsound_src_reset (GstAudioSrc * asrc)
 {
-  
GstDirectSoundSrc * dsoundsrc;
-  
LPVOID pLockedBuffer = NULL;
-  
DWORD dwSizeBuffer = 0;
-  

GST_DEBUG ("reset directsoundsrc\n");
-  

dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
-  


-#if 0

-      IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
-  

-#endif /* 
 */
-      
GST_DSOUND_LOCK (dsoundsrc);
-  

if (dsoundsrc->pDSBSecondary) {
-    

-        /*stop capturing */ 

-        HRESULT hRes =
-        IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
-    


-        /*reset position */ 

-        /*    hRes = IDirectSoundCaptureBuffer_SetCurrentPosition (dsoundsrc->pDSBSecondary, 0); */ 

-        

-        /*reset the buffer */ 

-        hRes =
-        IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
-        
dsoundsrc->current_circular_offset, dsoundsrc->buffer_size,
-        
pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
-    

if (SUCCEEDED (hRes)) {
-      
memset (pLockedBuffer, 0, dwSizeBuffer);
-      

hRes =
-          
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
-          pLockedBuffer, 
dwSizeBuffer, NULL, 0);
-    
}
-    
dsoundsrc->current_circular_offset = 0;
-  

}
-  

GST_DSOUND_UNLOCK (dsoundsrc);
-
}
-
-



+  GstDirectSoundSrc *dsoundsrc;
+  LPVOID pLockedBuffer = NULL;
+  DWORD dwSizeBuffer = 0;
+
+  GST_DEBUG ("reset directsoundsrc\n");
+
+  dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
+
+#if 0
+  IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
+#endif
+
+  GST_DSOUND_LOCK (dsoundsrc);
+
+  if (dsoundsrc->pDSBSecondary) {
+    /*stop capturing */
+    HRESULT hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
+
+    /*reset position */
+    /*    hRes = IDirectSoundCaptureBuffer_SetCurrentPosition (dsoundsrc->pDSBSecondary, 0); */
+
+    /*reset the buffer */
+    hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
+        dsoundsrc->current_circular_offset, dsoundsrc->buffer_size,
+        pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
+
+    if (SUCCEEDED (hRes)) {
+      memset (pLockedBuffer, 0, dwSizeBuffer);
+
+      hRes =
+          IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
+          pLockedBuffer, dwSizeBuffer, NULL, 0);
+    }
+    dsoundsrc->current_circular_offset = 0;
+
+  }
+
+  GST_DSOUND_UNLOCK (dsoundsrc);
+}