ext\neon\gstneonhttpsrc.c: Include unistd.h only if _HAVE_UNISTD_H is defined

Original commit message from CVS:
* ext\neon\gstneonhttpsrc.c:
Include unistd.h only if _HAVE_UNISTD_H is defined
* gst\mpegvideoparse\mpegvideoparse.c:
Use G_GUINT64_CONSTANT GLIB macro for constant
* sys\dshowsrcwrapper\gstdshowaudiosrc.c:
* sys\dshowsrcwrapper\gstdshowvideosrc.c:
* sys\dshowdecwrapper\gstdshowaudiodec.c:
* sys\dshowdecwrapper\gstdshowaudiodec.h:
* sys\dshowdecwrapper\gstdshowdecwrapper.c:
* sys\dshowdecwrapper\gstdshowdecwrapper.h:
* sys\dshowdecwrapper\gstdshowvideodec.c
* sys\dshowdecwrapper\gstdshowvideodec.h:
Add a DirectShow decoder wrapper.
* win32\MANIFEST:
Add new win32 files to MANIFEST
* win32\vs6\gst_plugins_bad.dsw:
* win32\vs6\libgstdshow.dsp:
* win32\vs6\libgstdshowdecwrapper.dsp:
* win32\vs6\libgstflv.dsp:
Add new projects to bad workspace

1 files changed, 1168 insertions, 0 deletions

diff --git a/sys/dshowdecwrapper/gstdshowaudiodec.c b/sys/dshowdecwrapper/gstdshowaudiodec.c
new file mode 100644
index 00000000..c231ac4b
--- /dev/null
+++ b/sys/dshowdecwrapper/gstdshowaudiodec.c
@@ -0,0 +1,1168 @@
+/*
+ * GStreamer DirectShow codecs wrapper
+ * Copyright <2006, 2007, 2008> Fluendo <gstreamer@fluendo.com>
+ * Copyright <2006, 2007, 2008> Pioneers of the Inevitable <songbird@songbirdnest.com>
+ * Copyright <2007,2008> Sebastien Moutte <sebastien@moutte.net>
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstdshowaudiodec.h"
+#include <mmreg.h>
+
+GST_DEBUG_CATEGORY_STATIC (dshowaudiodec_debug);
+#define GST_CAT_DEFAULT dshowaudiodec_debug
+
+GST_BOILERPLATE (GstDshowAudioDec, gst_dshowaudiodec, GstElement,
+    GST_TYPE_ELEMENT);
+static const CodecEntry *tmp;
+
+static void gst_dshowaudiodec_dispose (GObject * object);
+static GstStateChangeReturn gst_dshowaudiodec_change_state
+    (GstElement * element, GstStateChange transition);
+
+/* sink pad overwrites */
+static gboolean gst_dshowaudiodec_sink_setcaps (GstPad * pad, GstCaps * caps);
+static GstFlowReturn gst_dshowaudiodec_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean gst_dshowaudiodec_sink_event (GstPad * pad, GstEvent * event);
+
+/* callback used by directshow to push buffers */
+static gboolean gst_dshowaudiodec_push_buffer (byte * buffer, long size,
+    byte * src_object, UINT64 start, UINT64 stop);
+
+/* utils */
+static gboolean gst_dshowaudiodec_create_graph_and_filters (GstDshowAudioDec *
+    adec);
+static gboolean gst_dshowaudiodec_destroy_graph_and_filters (GstDshowAudioDec *
+    adec);
+static gboolean gst_dshowaudiodec_flush (GstDshowAudioDec * adec);
+static gboolean gst_dshowaudiodec_get_filter_settings (GstDshowAudioDec * adec);
+static gboolean gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec);
+
+/* gobal variable */
+const long bitrates[2][3][16] = {
+  /* version 0 */
+  {
+        /* one list per layer 1-3 */
+        {0, 32000, 48000, 56000, 64000, 80000, 96000, 112000, 128000, 144000,
+            160000, 176000, 192000, 224000, 256000, 0},
+        {0, 8000, 16000, 24000, 32000, 40000, 48000, 56000, 64000, 80000, 96000,
+            112000, 128000, 144000, 160000, 0},
+        {0, 8000, 16000, 24000, 32000, 40000, 48000, 56000, 64000, 80000, 96000,
+            112000, 128000, 144000, 160000, 0},
+      },
+  /* version 1 */
+  {
+        /* one list per layer 1-3 */
+        {0, 32000, 64000, 96000, 128000, 160000, 192000, 224000, 256000,
+            288000, 320000, 352000, 384000, 416000, 448000, 0},
+        {0, 32000, 48000, 56000, 64000, 80000, 96000, 112000, 128000,
+            160000, 192000, 224000, 256000, 320000, 384000, 0},
+        {0, 32000, 40000, 48000, 56000, 64000, 80000, 96000, 112000,
+            128000, 160000, 192000, 224000, 256000, 320000, 0},
+      }
+};
+
+#define GUID_MEDIATYPE_AUDIO    {0x73647561, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_PCM   {0x00000001, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_WMAV1 {0x00000160, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_WMAV2 {0x00000161, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_WMAV3 {0x00000162, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_WMAV4 {0x00000163, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_WMS   {0x0000000a, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_MP3   {0x00000055, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+#define GUID_MEDIASUBTYPE_MPEG1AudioPayload {0x00000050, 0x0000, 0x0010, { 0x80, 0x00, 0x00, 0xAA, 0x00, 0x38, 0x9b, 0x71 }}
+
+static const CodecEntry audio_dec_codecs[] = {
+  {"dshowadec_wma1",
+        "Windows Media Audio 7",
+        "DMO",
+        0x00000160,
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV1,
+        "audio/x-wma, wmaversion = (int) 1",
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM,
+        "audio/x-raw-int, "
+        "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
+        "signed = (boolean) true, endianness = (int) "
+        G_STRINGIFY (G_LITTLE_ENDIAN)
+      },
+  {"dshowadec_wma2",
+        "Windows Media Audio 8",
+        "DMO",
+        0x00000161,
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV2,
+        "audio/x-wma, wmaversion = (int) 2",
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM,
+        "audio/x-raw-int, "
+        "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
+        "signed = (boolean) true, endianness = (int) "
+        G_STRINGIFY (G_LITTLE_ENDIAN)
+      },
+  {"dshowadec_wma3",
+        "Windows Media Audio 9 Professional",
+        "DMO",
+        0x00000162,
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV3,
+        "audio/x-wma, wmaversion = (int) 3",
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM,
+        "audio/x-raw-int, "
+        "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
+        "signed = (boolean) true, endianness = (int) "
+        G_STRINGIFY (G_LITTLE_ENDIAN)
+      },
+  {"dshowadec_wma4",
+        "Windows Media Audio 9 Lossless",
+        "DMO",
+        0x00000163,
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMAV4,
+        "audio/x-wma, wmaversion = (int) 4",
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM,
+        "audio/x-raw-int, "
+        "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
+        "signed = (boolean) true, endianness = (int) "
+        G_STRINGIFY (G_LITTLE_ENDIAN)
+      },
+  {"dshowadec_wms",
+        "Windows Media Audio Voice v9",
+        "DMO",
+        0x0000000a,
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_WMS,
+        "audio/x-wms",
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM,
+        "audio/x-raw-int, "
+        "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
+        "signed = (boolean) true, endianness = (int) "
+        G_STRINGIFY (G_LITTLE_ENDIAN)
+      },
+  {"dshowadec_mpeg1",
+        "MPEG-1 Layer 1,2,3 Audio",
+        "MPEG Layer-3 Decoder",
+        0x00000055,
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_MP3,
+        "audio/mpeg, "
+        "mpegversion = (int) 1, "
+        "layer = (int) { 1 , 2, 3 }, "
+        "rate = (int) [ 8000, 48000 ], "
+        "channels = (int) [ 1, 2 ], " "parsed= (boolean) true",
+        GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM,
+        "audio/x-raw-int, "
+        "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
+        "signed = (boolean) true, endianness = (int) "
+        G_STRINGIFY (G_LITTLE_ENDIAN)
+      }
+};
+
+/* Private map used when dshowadec_mpeg is loaded with layer=1 or 2.
+ * The problem is that gstreamer don't care about caps like layer when connecting pads.
+ * So I've only one element handling mpeg audio in the public codecs map and 
+ * when it's loaded for mp3, I'm releasing mpeg audio decoder and replace it by 
+ * the one described in this private map.
+*/
+static const CodecEntry audio_mpeg_1_2[] = { "dshowadec_mpeg_1_2",
+  "MPEG-1 Layer 1,2 Audio",
+  "MPEG Audio Decoder",
+  0x00000050,
+  GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_MPEG1AudioPayload,
+  "audio/mpeg, "
+      "mpegversion = (int) 1, "
+      "layer = (int) [ 1, 2 ], "
+      "rate = (int) [ 8000, 48000 ], "
+      "channels = (int) [ 1, 2 ], " "parsed= (boolean) true",
+  GUID_MEDIATYPE_AUDIO, GUID_MEDIASUBTYPE_PCM,
+  "audio/x-raw-int, "
+      "width = (int) { 1, 8, 16 }, depth = (int) { 1, 8, 16 }, "
+      "signed = (boolean) true, endianness = (int) "
+      G_STRINGIFY (G_LITTLE_ENDIAN)
+};
+
+static void
+gst_dshowaudiodec_base_init (GstDshowAudioDecClass * klass)
+{
+  GstPadTemplate *src, *sink;
+  GstCaps *srccaps, *sinkcaps;
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementDetails details;
+
+  klass->entry = tmp;
+  details.longname = g_strdup_printf ("DirectShow %s Decoder Wrapper",
+      tmp->element_longname);
+  details.klass = g_strdup ("Codec/Decoder/Audio");
+  details.description = g_strdup_printf ("DirectShow %s Decoder Wrapper",
+      tmp->element_longname);
+  details.author = "Sebastien Moutte <sebastien@moutte.net>";
+  gst_element_class_set_details (element_class, &details);
+  g_free (details.longname);
+  g_free (details.klass);
+  g_free (details.description);
+
+  sinkcaps = gst_caps_from_string (tmp->sinkcaps);
+  gst_caps_set_simple (sinkcaps,
+      "block_align", GST_TYPE_INT_RANGE, 0, G_MAXINT,
+      "bitrate", GST_TYPE_INT_RANGE, 0, G_MAXINT, NULL);
+
+  srccaps = gst_caps_from_string (tmp->srccaps);
+
+  sink = gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, sinkcaps);
+  src = gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS, srccaps);
+
+  /* register */
+  gst_element_class_add_pad_template (element_class, src);
+  gst_element_class_add_pad_template (element_class, sink);
+}
+
+static void
+gst_dshowaudiodec_class_init (GstDshowAudioDecClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_dshowaudiodec_dispose);
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_dshowaudiodec_change_state);
+
+  if (!parent_class)
+    parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+  if (!dshowaudiodec_debug) {
+    GST_DEBUG_CATEGORY_INIT (dshowaudiodec_debug, "dshowaudiodec", 0,
+        "Directshow filter audio decoder");
+  }
+}
+
+static void
+gst_dshowaudiodec_init (GstDshowAudioDec * adec,
+    GstDshowAudioDecClass * adec_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_GET_CLASS (adec);
+
+  /* setup pads */
+  adec->sinkpad =
+      gst_pad_new_from_template (gst_element_class_get_pad_template
+      (element_class, "sink"), "sink");
+
+  gst_pad_set_setcaps_function (adec->sinkpad, gst_dshowaudiodec_sink_setcaps);
+  gst_pad_set_event_function (adec->sinkpad, gst_dshowaudiodec_sink_event);
+  gst_pad_set_chain_function (adec->sinkpad, gst_dshowaudiodec_chain);
+  gst_element_add_pad (GST_ELEMENT (adec), adec->sinkpad);
+
+  adec->srcpad =
+      gst_pad_new_from_template (gst_element_class_get_pad_template
+      (element_class, "src"), "src");
+  gst_element_add_pad (GST_ELEMENT (adec), adec->srcpad);
+
+  adec->srcfilter = NULL;
+  adec->gstdshowsrcfilter = NULL;
+  adec->decfilter = NULL;
+  adec->sinkfilter = NULL;
+  adec->filtergraph = NULL;
+  adec->mediafilter = NULL;
+  adec->timestamp = GST_CLOCK_TIME_NONE;
+  adec->segment = gst_segment_new ();
+  adec->setup = FALSE;
+  adec->depth = 0;
+  adec->bitrate = 0;
+  adec->block_align = 0;
+  adec->channels = 0;
+  adec->rate = 0;
+  adec->layer = 0;
+  adec->codec_data = NULL;
+
+  CoInitializeEx (NULL, COINIT_MULTITHREADED);
+}
+
+static void
+gst_dshowaudiodec_dispose (GObject * object)
+{
+  GstDshowAudioDec *adec = (GstDshowAudioDec *) (object);
+
+  if (adec->segment) {
+    gst_segment_free (adec->segment);
+    adec->segment = NULL;
+  }
+
+  if (adec->codec_data) {
+    gst_buffer_unref (adec->codec_data);
+    adec->codec_data = NULL;
+  }
+
+  CoUninitialize ();
+}
+
+
+static GstStateChangeReturn
+gst_dshowaudiodec_change_state (GstElement * element, GstStateChange transition)
+{
+  GstDshowAudioDec *adec = (GstDshowAudioDec *) (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (!gst_dshowaudiodec_create_graph_and_filters (adec))
+        return GST_STATE_CHANGE_FAILURE;
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      break;
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      adec->depth = 0;
+      adec->bitrate = 0;
+      adec->block_align = 0;
+      adec->channels = 0;
+      adec->rate = 0;
+      adec->layer = 0;
+      if (adec->codec_data) {
+        gst_buffer_unref (adec->codec_data);
+        adec->codec_data = NULL;
+      }
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      if (!gst_dshowaudiodec_destroy_graph_and_filters (adec))
+        return GST_STATE_CHANGE_FAILURE;
+      break;
+    default:
+      break;
+  }
+
+  return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+}
+
+static gboolean
+gst_dshowaudiodec_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+  gboolean ret = FALSE;
+  GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad);
+  GstStructure *s = gst_caps_get_structure (caps, 0);
+  const GValue *v = NULL;
+
+  adec->timestamp = GST_CLOCK_TIME_NONE;
+
+  /* read data, only rate and channels are needed */
+  if (!gst_structure_get_int (s, "rate", &adec->rate) ||
+      !gst_structure_get_int (s, "channels", &adec->channels)) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("error getting audio specs from caps"), (NULL));
+    goto end;
+  }
+
+  gst_structure_get_int (s, "depth", &adec->depth);
+  gst_structure_get_int (s, "bitrate", &adec->bitrate);
+  gst_structure_get_int (s, "block_align", &adec->block_align);
+  gst_structure_get_int (s, "layer", &adec->layer);
+
+  if (adec->codec_data) {
+    gst_buffer_unref (adec->codec_data);
+    adec->codec_data = NULL;
+  }
+
+  if ((v = gst_structure_get_value (s, "codec_data")))
+    adec->codec_data = gst_buffer_ref (gst_value_get_buffer (v));
+
+  if (adec->layer != 1 && adec->layer != 2) {
+    /* setup dshow graph for all formats except for 
+     * MPEG-1 layer 1 and 2 for which we need negociate
+     * in _chain function.
+     */
+    ret = gst_dshowaudiodec_setup_graph (adec);
+  }
+
+  ret = TRUE;
+end:
+  gst_object_unref (adec);
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_dshowaudiodec_chain (GstPad * pad, GstBuffer * buffer)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad);
+  gboolean discount = FALSE;
+
+  if (!adec->setup) {
+    if (adec->layer != 0) {
+      if (adec->codec_data) {
+        gst_buffer_unref (adec->codec_data);
+        adec->codec_data = NULL;
+      }
+      /* extract the 3 bytes of MPEG-1 audio frame header */
+      adec->codec_data = gst_buffer_create_sub (buffer, 1, 3);
+    }
+
+    /* setup dshow graph */
+    if (!gst_dshowaudiodec_setup_graph (adec)) {
+      return GST_FLOW_ERROR;
+    }
+  }
+
+  if (!adec->gstdshowsrcfilter) {
+    /* we are not setup */
+    ret = GST_FLOW_WRONG_STATE;
+    goto beach;
+  }
+
+  GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec, "chain (size %d)=> pts %"
+      GST_TIME_FORMAT " stop %" GST_TIME_FORMAT,
+      GST_BUFFER_SIZE (buffer), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer) +
+          GST_BUFFER_DURATION (buffer)));
+
+  /* if the incoming buffer has discont flag set => flush decoder data */
+  if (buffer && GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+    GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec,
+        "this buffer has a DISCONT flag (%" GST_TIME_FORMAT "), flushing",
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
+    gst_dshowaudiodec_flush (adec);
+    discount = TRUE;
+  }
+
+  /* push the buffer to the directshow decoder */
+  IGstDshowInterface_gst_push_buffer (adec->gstdshowsrcfilter,
+      GST_BUFFER_DATA (buffer), GST_BUFFER_TIMESTAMP (buffer),
+      GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer),
+      GST_BUFFER_SIZE (buffer), discount);
+
+beach:
+  gst_buffer_unref (buffer);
+  gst_object_unref (adec);
+  return ret;
+}
+
+static gboolean
+gst_dshowaudiodec_push_buffer (byte * buffer, long size, byte * src_object,
+    UINT64 dshow_start, UINT64 dshow_stop)
+{
+  GstDshowAudioDec *adec = (GstDshowAudioDec *) src_object;
+  GstBuffer *out_buf = NULL;
+  gboolean in_seg = FALSE;
+  gint64 buf_start, buf_stop;
+  gint64 clip_start = 0, clip_stop = 0;
+  size_t start_offset = 0, stop_offset = size;
+
+  if (!GST_CLOCK_TIME_IS_VALID (adec->timestamp)) {
+    adec->timestamp = dshow_start;
+  }
+
+  buf_start = adec->timestamp;
+  buf_stop = adec->timestamp + (dshow_stop - dshow_start);
+
+  /* save stop position to start next buffer with it */
+  adec->timestamp = buf_stop;
+
+  /* check if this buffer is in our current segment */
+  in_seg = gst_segment_clip (adec->segment, GST_FORMAT_TIME,
+      buf_start, buf_stop, &clip_start, &clip_stop);
+
+  /* if the buffer is out of segment do not push it downstream */
+  if (!in_seg) {
+    GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec,
+        "buffer is out of segment, start %" GST_TIME_FORMAT " stop %"
+        GST_TIME_FORMAT, GST_TIME_ARGS (buf_start), GST_TIME_ARGS (buf_stop));
+    return FALSE;
+  }
+
+  /* buffer is in our segment allocate a new out buffer and clip it if needed */
+
+  /* allocate a new buffer for raw audio */
+  gst_pad_alloc_buffer (adec->srcpad, GST_BUFFER_OFFSET_NONE,
+      size, GST_PAD_CAPS (adec->srcpad), &out_buf);
+  if (!out_buf) {
+    GST_CAT_ERROR_OBJECT (dshowaudiodec_debug, adec,
+        "can't not allocate a new GstBuffer");
+    return FALSE;
+  }
+
+  /* set buffer properties */
+  GST_BUFFER_SIZE (out_buf) = size;
+  GST_BUFFER_TIMESTAMP (out_buf) = buf_start;
+  GST_BUFFER_DURATION (out_buf) = buf_stop - buf_start;
+  memcpy (GST_BUFFER_DATA (out_buf), buffer, size);
+
+  /* we have to remove some heading samples */
+  if (clip_start > buf_start) {
+    start_offset = (size_t) gst_util_uint64_scale_int (clip_start - buf_start,
+        adec->rate, GST_SECOND) * adec->depth / 8 * adec->channels;
+  }
+  /* we have to remove some trailing samples */
+  if (clip_stop < buf_stop) {
+    stop_offset = (size_t) gst_util_uint64_scale_int (buf_stop - clip_stop,
+        adec->rate, GST_SECOND) * adec->depth / 8 * adec->channels;
+  }
+
+  /* truncating */
+  if ((start_offset != 0) || (stop_offset != (size_t) size)) {
+    GstBuffer *subbuf = gst_buffer_create_sub (out_buf, start_offset,
+        stop_offset - start_offset);
+
+    if (subbuf) {
+      gst_buffer_set_caps (subbuf, GST_PAD_CAPS (adec->srcpad));
+      gst_buffer_unref (out_buf);
+      out_buf = subbuf;
+    }
+  }
+
+  GST_BUFFER_TIMESTAMP (out_buf) = clip_start;
+  GST_BUFFER_DURATION (out_buf) = clip_stop - clip_start;
+
+  /* replace the saved stop position by the clipped one */
+  adec->timestamp = clip_stop;
+
+  GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec,
+      "push_buffer (size %d)=> pts %" GST_TIME_FORMAT " stop %" GST_TIME_FORMAT
+      " duration %" GST_TIME_FORMAT, size,
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf)),
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf) +
+          GST_BUFFER_DURATION (out_buf)),
+      GST_TIME_ARGS (GST_BUFFER_DURATION (out_buf)));
+
+  gst_pad_push (adec->srcpad, out_buf);
+
+  return TRUE;
+}
+
+static gboolean
+gst_dshowaudiodec_sink_event (GstPad * pad, GstEvent * event)
+{
+  gboolean ret = TRUE;
+  GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad);
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_FLUSH_STOP:{
+      gst_dshowaudiodec_flush (adec);
+      ret = gst_pad_event_default (pad, event);
+      break;
+    }
+    case GST_EVENT_NEWSEGMENT:
+    {
+      GstFormat format;
+      gdouble rate;
+      gint64 start, stop, time;
+      gboolean update;
+
+      gst_event_parse_new_segment (event, &update, &rate, &format, &start,
+          &stop, &time);
+
+      GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec,
+          "received new segment from %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT,
+          GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+
+      if (update) {
+        GST_CAT_DEBUG_OBJECT (dshowaudiodec_debug, adec,
+            "closing current segment flushing..");
+        gst_dshowaudiodec_flush (adec);
+      }
+
+      /* save the new segment in our local current segment */
+      gst_segment_set_newsegment (adec->segment, update, rate, format, start,
+          stop, time);
+
+      ret = gst_pad_event_default (pad, event);
+      break;
+    }
+    default:
+      ret = gst_pad_event_default (pad, event);
+      break;
+  }
+  return ret;
+}
+
+static gboolean
+gst_dshowaudiodec_flush (GstDshowAudioDec * adec)
+{
+  if (!adec->gstdshowsrcfilter)
+    return FALSE;
+
+  /* flush dshow decoder and reset timestamp */
+  IGstDshowInterface_gst_flush (adec->gstdshowsrcfilter);
+  adec->timestamp = GST_CLOCK_TIME_NONE;
+
+  return TRUE;
+}
+
+
+static gboolean
+gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec)
+{
+  gboolean ret = FALSE;
+  GstDshowAudioDecClass *klass =
+      (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
+  HRESULT hres;
+  gint size = 0;
+  GstCaps *out;
+  AM_MEDIA_TYPE output_mediatype, input_mediatype;
+  WAVEFORMATEX *input_format = NULL, output_format;
+  IPin *output_pin = NULL, *input_pin = NULL;
+  IGstDshowInterface *gstdshowinterface = NULL;
+  CodecEntry *codec_entry = klass->entry;
+
+  if (adec->layer != 0) {
+    if (adec->layer == 1 || adec->layer == 2) {
+      /* for MPEG-1 layer 1 or 2 we have to release the current 
+       * MP3 decoder and create an instance of MPEG Audio Decoder
+       */
+      IBaseFilter_Release (adec->decfilter);
+      adec->decfilter = NULL;
+      codec_entry = audio_mpeg_1_2;
+      gst_dshow_find_filter (codec_entry->input_majortype,
+          codec_entry->input_subtype,
+          codec_entry->output_majortype,
+          codec_entry->output_subtype,
+          codec_entry->prefered_filter_substring, &adec->decfilter);
+      IFilterGraph_AddFilter (adec->filtergraph, adec->decfilter, L"decoder");
+    } else {
+      /* mp3 don't need to negociate with MPEG1WAVEFORMAT */
+      adec->layer = 0;
+    }
+  }
+
+  /* set mediatype on fakesrc filter output pin */
+  memset (&input_mediatype, 0, sizeof (AM_MEDIA_TYPE));
+  input_mediatype.majortype = codec_entry->input_majortype;
+  input_mediatype.subtype = codec_entry->input_subtype;
+  input_mediatype.bFixedSizeSamples = TRUE;
+  input_mediatype.bTemporalCompression = FALSE;
+  if (adec->block_align)
+    input_mediatype.lSampleSize = adec->block_align;
+  else
+    input_mediatype.lSampleSize = 8192; /* need to evaluate it dynamically */
+  input_mediatype.formattype = FORMAT_WaveFormatEx;
+
+  if (adec->layer != 0) {
+    MPEG1WAVEFORMAT *mpeg1_format;
+    BYTE b1, b2, b3;
+    gint samples, version, layer;
+
+    size = sizeof (MPEG1WAVEFORMAT);
+    input_format = g_malloc0 (size);
+    input_format->cbSize = sizeof (MPEG1WAVEFORMAT) - sizeof (WAVEFORMATEX);
+    mpeg1_format = (MPEG1WAVEFORMAT *) input_format;
+
+    /* initialize header bytes */
+    b1 = *GST_BUFFER_DATA (adec->codec_data);
+    b2 = *(GST_BUFFER_DATA (adec->codec_data) + 1);
+    b3 = *(GST_BUFFER_DATA (adec->codec_data) + 2);
+
+    /* fill MPEG1WAVEFORMAT using header */
+    input_format->wFormatTag = WAVE_FORMAT_MPEG;
+    mpeg1_format->wfx.nChannels = 2;
+    switch (b3 >> 6) {
+      case 0x00:
+        mpeg1_format->fwHeadMode = ACM_MPEG_STEREO;
+        break;
+      case 0x01:
+        mpeg1_format->fwHeadMode = ACM_MPEG_JOINTSTEREO;
+        break;
+      case 0x02:
+        mpeg1_format->fwHeadMode = ACM_MPEG_DUALCHANNEL;
+        break;
+      case 0x03:
+        mpeg1_format->fwHeadMode = ACM_MPEG_SINGLECHANNEL;
+        mpeg1_format->wfx.nChannels = 1;
+        break;
+    }
+
+    mpeg1_format->fwHeadModeExt = (WORD) (1 << (b3 >> 4));
+    mpeg1_format->wHeadEmphasis = (WORD) ((b3 & 0x03) + 1);
+    mpeg1_format->fwHeadFlags = (WORD) (((b2 & 1) ? ACM_MPEG_PRIVATEBIT : 0) +
+        ((b3 & 8) ? ACM_MPEG_COPYRIGHT : 0) +
+        ((b3 & 4) ? ACM_MPEG_ORIGINALHOME : 0) +
+        ((b1 & 1) ? ACM_MPEG_PROTECTIONBIT : 0) + ACM_MPEG_ID_MPEG1);
+
+    layer = (b1 >> 1) & 3;
+    switch (layer) {
+      case 1:
+        mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER3;
+        layer = 3;
+        break;
+      case 2:
+        mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER2;
+        break;
+      case 3:
+        mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER1;
+        layer = 1;
+        break;
+    };
+
+    version = (b1 >> 3) & 1;
+    if (layer == 1) {
+      samples = 384;
+    } else {
+      if (version == 0) {
+        samples = 576;
+      } else {
+        samples = 1152;
+      }
+    }
+    mpeg1_format->wfx.nBlockAlign = (WORD) samples;
+    mpeg1_format->wfx.nSamplesPerSec = adec->rate;
+    mpeg1_format->dwHeadBitrate = bitrates[version][layer - 1][b2 >> 4];
+    mpeg1_format->wfx.nAvgBytesPerSec = mpeg1_format->dwHeadBitrate / 8;
+  } else {
+    size = sizeof (WAVEFORMATEX) +
+        (adec->codec_data ? GST_BUFFER_SIZE (adec->codec_data) : 0);
+    input_format = g_malloc0 (size);
+    if (adec->codec_data) {     /* Codec data is appended after our header */
+      memcpy (((guchar *) input_format) + sizeof (WAVEFORMATEX),
+          GST_BUFFER_DATA (adec->codec_data),
+          GST_BUFFER_SIZE (adec->codec_data));
+      input_format->cbSize = GST_BUFFER_SIZE (adec->codec_data);
+    }
+
+    input_format->wFormatTag = codec_entry->format;
+    input_format->nChannels = adec->channels;
+    input_format->nSamplesPerSec = adec->rate;
+    input_format->nAvgBytesPerSec = adec->bitrate / 8;
+    input_format->nBlockAlign = adec->block_align;
+    input_format->wBitsPerSample = adec->depth;
+  }
+
+  input_mediatype.cbFormat = size;
+  input_mediatype.pbFormat = (BYTE *) input_format;
+
+  hres = IBaseFilter_QueryInterface (adec->srcfilter, &IID_IGstDshowInterface,
+      (void **) &gstdshowinterface);
+  if (hres != S_OK || !gstdshowinterface) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get IGstDshowInterface interface from dshow fakesrc filter (error=%d)",
+            hres), (NULL));
+    goto end;
+  }
+
+  /* save a reference to IGstDshowInterface to use it processing functions */
+  if (!adec->gstdshowsrcfilter) {
+    adec->gstdshowsrcfilter = gstdshowinterface;
+    IBaseFilter_AddRef (adec->gstdshowsrcfilter);
+  }
+
+  IGstDshowInterface_gst_set_media_type (gstdshowinterface, &input_mediatype);
+  IGstDshowInterface_Release (gstdshowinterface);
+  gstdshowinterface = NULL;
+
+  /* connect our fake source to decoder */
+  gst_dshow_get_pin_from_filter (adec->srcfilter, PINDIR_OUTPUT, &output_pin);
+  if (!output_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get output pin from our directshow fakesrc filter"), (NULL));
+    goto end;
+  }
+  gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_INPUT, &input_pin);
+  if (!input_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get input pin from decoder filter"), (NULL));
+    goto end;
+  }
+
+  hres =
+      IFilterGraph_ConnectDirect (adec->filtergraph, output_pin, input_pin,
+      NULL);
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't connect fakesrc with decoder (error=%d)", hres), (NULL));
+    goto end;
+  }
+
+  IPin_Release (input_pin);
+  IPin_Release (output_pin);
+  input_pin = NULL;
+  output_pin = NULL;
+
+  if (!gst_dshowaudiodec_get_filter_settings (adec)) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get audio depth from decoder"), (NULL));
+    goto end;
+  }
+
+  /* set mediatype on fake sink input pin */
+  memset (&output_format, 0, sizeof (WAVEFORMATEX));
+  output_format.wFormatTag = WAVE_FORMAT_PCM;
+  output_format.wBitsPerSample = adec->depth;
+  output_format.nChannels = adec->channels;
+  output_format.nBlockAlign = adec->channels * (adec->depth / 8);
+  output_format.nSamplesPerSec = adec->rate;
+  output_format.nAvgBytesPerSec = output_format.nBlockAlign * adec->rate;
+
+  memset (&output_mediatype, 0, sizeof (AM_MEDIA_TYPE));
+  output_mediatype.majortype = codec_entry->output_majortype;
+  output_mediatype.subtype = codec_entry->output_subtype;
+  output_mediatype.bFixedSizeSamples = TRUE;
+  output_mediatype.bTemporalCompression = FALSE;
+  output_mediatype.lSampleSize = output_format.nBlockAlign;
+  output_mediatype.formattype = FORMAT_WaveFormatEx;
+  output_mediatype.cbFormat = sizeof (WAVEFORMATEX);
+  output_mediatype.pbFormat = (char *) &output_format;
+
+  hres = IBaseFilter_QueryInterface (adec->sinkfilter, &IID_IGstDshowInterface,
+      (void **) &gstdshowinterface);
+  if (hres != S_OK || !gstdshowinterface) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get IGstDshowInterface interface from dshow fakesink filter (error=%d)",
+            hres), (NULL));
+    goto end;
+  }
+
+  IGstDshowInterface_gst_set_media_type (gstdshowinterface, &output_mediatype);
+  IGstDshowInterface_gst_set_buffer_callback (gstdshowinterface,
+      gst_dshowaudiodec_push_buffer, (byte *) adec);
+  IGstDshowInterface_Release (gstdshowinterface);
+  gstdshowinterface = NULL;
+
+  /* negotiate output */
+  out = gst_caps_from_string (codec_entry->srccaps);
+  gst_caps_set_simple (out,
+      "width", G_TYPE_INT, adec->depth,
+      "depth", G_TYPE_INT, adec->depth,
+      "rate", G_TYPE_INT, adec->rate,
+      "channels", G_TYPE_INT, adec->channels, NULL);
+  if (!gst_pad_set_caps (adec->srcpad, out)) {
+    gst_caps_unref (out);
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Failed to negotiate output"), (NULL));
+    goto end;
+  }
+  gst_caps_unref (out);
+
+  /* connect the decoder to our fake sink */
+  gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT, &output_pin);
+  if (!output_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get output pin from our decoder filter"), (NULL));
+    goto end;
+  }
+  gst_dshow_get_pin_from_filter (adec->sinkfilter, PINDIR_INPUT, &input_pin);
+  if (!input_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get input pin from our directshow fakesink filter"), (NULL));
+    goto end;
+  }
+
+  hres =
+      IFilterGraph_ConnectDirect (adec->filtergraph, output_pin, input_pin,
+      NULL);
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't connect decoder with fakesink (error=%d)", hres), (NULL));
+    goto end;
+  }
+
+  hres = IMediaFilter_Run (adec->mediafilter, -1);
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't run the directshow graph (error=%d)", hres), (NULL));
+    goto end;
+  }
+
+  ret = TRUE;
+  adec->setup = TRUE;
+end:
+  gst_object_unref (adec);
+  if (input_format)
+    g_free (input_format);
+  if (gstdshowinterface)
+    IGstDshowInterface_Release (gstdshowinterface);
+  if (input_pin)
+    IPin_Release (input_pin);
+  if (output_pin)
+    IPin_Release (output_pin);
+
+  return ret;
+}
+
+static gboolean
+gst_dshowaudiodec_get_filter_settings (GstDshowAudioDec * adec)
+{
+  IPin *output_pin = NULL;
+  IEnumMediaTypes *enum_mediatypes = NULL;
+  HRESULT hres;
+  ULONG fetched;
+  BOOL ret = FALSE;
+
+  if (!adec->decfilter)
+    return FALSE;
+
+  if (!gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT,
+          &output_pin)) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("failed getting ouput pin from the decoder"), (NULL));
+    return FALSE;
+  }
+
+  hres = IPin_EnumMediaTypes (output_pin, &enum_mediatypes);
+  if (hres == S_OK && enum_mediatypes) {
+    AM_MEDIA_TYPE *mediatype = NULL;
+
+    IEnumMediaTypes_Reset (enum_mediatypes);
+    while (hres =
+        IEnumMoniker_Next (enum_mediatypes, 1, &mediatype, &fetched),
+        hres == S_OK) {
+      RPC_STATUS rpcstatus;
+
+      if ((UuidCompare (&mediatype->subtype, &MEDIASUBTYPE_PCM, &rpcstatus) == 0
+              && rpcstatus == RPC_S_OK) &&
+          (UuidCompare (&mediatype->formattype, &FORMAT_WaveFormatEx,
+                  &rpcstatus) == 0 && rpcstatus == RPC_S_OK)) {
+        WAVEFORMATEX *audio_info = (WAVEFORMATEX *) mediatype->pbFormat;
+
+        adec->channels = audio_info->nChannels;
+        adec->depth = audio_info->wBitsPerSample;
+        adec->rate = audio_info->nSamplesPerSec;
+        ret = TRUE;
+      }
+      gst_dshow_free_mediatype (mediatype);
+      if (ret)
+        break;
+    }
+    IEnumMediaTypes_Release (enum_mediatypes);
+  }
+  if (output_pin) {
+    IPin_Release (output_pin);
+  }
+
+  return ret;
+}
+
+static gboolean
+gst_dshowaudiodec_create_graph_and_filters (GstDshowAudioDec * adec)
+{
+  BOOL ret = FALSE;
+  HRESULT hres = S_FALSE;
+  GstDshowAudioDecClass *klass =
+      (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
+
+  /* create the filter graph manager object */
+  hres = CoCreateInstance (&CLSID_FilterGraph, NULL, CLSCTX_INPROC,
+      &IID_IFilterGraph, (LPVOID *) & adec->filtergraph);
+  if (hres != S_OK || !adec->filtergraph) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't create an instance of the directshow graph manager (error=%d)",
+            hres), (NULL));
+    goto error;
+  }
+
+  hres = IFilterGraph_QueryInterface (adec->filtergraph, &IID_IMediaFilter,
+      (void **) &adec->mediafilter);
+  if (hres != S_OK || !adec->mediafilter) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't get IMediacontrol interface from the graph manager (error=%d)",
+            hres), (NULL));
+    goto error;
+  }
+
+  /* create fake src filter */
+  hres = CoCreateInstance (&CLSID_DshowFakeSrc, NULL, CLSCTX_INPROC,
+      &IID_IBaseFilter, (LPVOID *) & adec->srcfilter);
+  if (hres != S_OK || !adec->srcfilter) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't create an instance of the directshow fakesrc (error=%d)", hres),
+        (NULL));
+    goto error;
+  }
+
+  /* create decoder filter */
+  if (!gst_dshow_find_filter (klass->entry->input_majortype,
+          klass->entry->input_subtype,
+          klass->entry->output_majortype,
+          klass->entry->output_subtype,
+          klass->entry->prefered_filter_substring, &adec->decfilter)) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't create an instance of the decoder filter"), (NULL));
+    goto error;
+  }
+
+  /* create fake sink filter */
+  hres = CoCreateInstance (&CLSID_DshowFakeSink, NULL, CLSCTX_INPROC,
+      &IID_IBaseFilter, (LPVOID *) & adec->sinkfilter);
+  if (hres != S_OK || !adec->sinkfilter) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't create an instance of the directshow fakesink (error=%d)",
+            hres), (NULL));
+    goto error;
+  }
+
+  /* add filters to the graph */
+  hres = IFilterGraph_AddFilter (adec->filtergraph, adec->srcfilter, L"src");
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't add fakesrc filter to the graph (error=%d)", hres), (NULL));
+    goto error;
+  }
+
+  hres =
+      IFilterGraph_AddFilter (adec->filtergraph, adec->decfilter, L"decoder");
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't add decoder filter to the graph (error=%d)", hres), (NULL));
+    goto error;
+  }
+
+  hres = IFilterGraph_AddFilter (adec->filtergraph, adec->sinkfilter, L"sink");
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't add fakesink filter to the graph (error=%d)", hres), (NULL));
+    goto error;
+  }
+
+  return TRUE;
+
+error:
+  if (adec->srcfilter) {
+    IBaseFilter_Release (adec->srcfilter);
+    adec->srcfilter = NULL;
+  }
+  if (adec->decfilter) {
+    IBaseFilter_Release (adec->decfilter);
+    adec->decfilter = NULL;
+  }
+  if (adec->sinkfilter) {
+    IBaseFilter_Release (adec->sinkfilter);
+    adec->sinkfilter = NULL;
+  }
+  if (adec->mediafilter) {
+    IMediaFilter_Release (adec->mediafilter);
+    adec->mediafilter = NULL;
+  }
+  if (adec->filtergraph) {
+    IFilterGraph_Release (adec->filtergraph);
+    adec->filtergraph = NULL;
+  }
+
+  return FALSE;
+}
+
+static gboolean
+gst_dshowaudiodec_destroy_graph_and_filters (GstDshowAudioDec * adec)
+{
+  if (adec->mediafilter) {
+    IMediaFilter_Stop (adec->mediafilter);
+  }
+
+  if (adec->gstdshowsrcfilter) {
+    IGstDshowInterface_Release (adec->gstdshowsrcfilter);
+    adec->gstdshowsrcfilter = NULL;
+  }
+  if (adec->srcfilter) {
+    if (adec->filtergraph)
+      IFilterGraph_RemoveFilter (adec->filtergraph, adec->srcfilter);
+    IBaseFilter_Release (adec->srcfilter);
+    adec->srcfilter = NULL;
+  }
+  if (adec->decfilter) {
+    if (adec->filtergraph)
+      IFilterGraph_RemoveFilter (adec->filtergraph, adec->decfilter);
+    IBaseFilter_Release (adec->decfilter);
+    adec->decfilter = NULL;
+  }
+  if (adec->sinkfilter) {
+    if (adec->filtergraph)
+      IFilterGraph_RemoveFilter (adec->filtergraph, adec->sinkfilter);
+    IBaseFilter_Release (adec->sinkfilter);
+    adec->sinkfilter = NULL;
+  }
+  if (adec->mediafilter) {
+    IMediaFilter_Release (adec->mediafilter);
+    adec->mediafilter = NULL;
+  }
+  if (adec->filtergraph) {
+    IFilterGraph_Release (adec->filtergraph);
+    adec->filtergraph = NULL;
+  }
+
+  adec->setup = FALSE;
+
+  return TRUE;
+}
+
+gboolean
+dshow_adec_register (GstPlugin * plugin)
+{
+  GTypeInfo info = {
+    sizeof (GstDshowAudioDecClass),
+    (GBaseInitFunc) gst_dshowaudiodec_base_init,
+    NULL,
+    (GClassInitFunc) gst_dshowaudiodec_class_init,
+    NULL,
+    NULL,
+    sizeof (GstDshowAudioDec),
+    0,
+    (GInstanceInitFunc) gst_dshowaudiodec_init,
+  };
+  gint i;
+
+  GST_DEBUG_CATEGORY_INIT (dshowaudiodec_debug, "dshowaudiodec", 0,
+      "Directshow filter audio decoder");
+
+  CoInitializeEx (NULL, COINIT_MULTITHREADED);
+  for (i = 0; i < sizeof (audio_dec_codecs) / sizeof (CodecEntry); i++) {
+    GType type;
+
+    if (gst_dshow_find_filter (audio_dec_codecs[i].input_majortype,
+            audio_dec_codecs[i].input_subtype,
+            audio_dec_codecs[i].output_majortype,
+            audio_dec_codecs[i].output_subtype,
+            audio_dec_codecs[i].prefered_filter_substring, NULL)) {
+
+      GST_CAT_DEBUG (dshowaudiodec_debug, "Registering %s",
+          audio_dec_codecs[i].element_name);
+
+      tmp = &audio_dec_codecs[i];
+      type =
+          g_type_register_static (GST_TYPE_ELEMENT,
+          audio_dec_codecs[i].element_name, &info, 0);
+      if (!gst_element_register (plugin, audio_dec_codecs[i].element_name,
+              GST_RANK_PRIMARY, type)) {
+        return FALSE;
+      }
+      GST_CAT_DEBUG (dshowaudiodec_debug, "Registered %s",
+          audio_dec_codecs[i].element_name);
+    } else {
+      GST_CAT_DEBUG (dshowaudiodec_debug,
+          "Element %s not registered (the format is not supported by the system)",
+          audio_dec_codecs[i].element_name);
+    }
+  }
+
+  CoUninitialize ();
+  return TRUE;
+}