sys/dshowdecwrapper/: Major rewrite of dshowdecwrapper. Converts code to

Original commit message from CVS:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowdecwrapper/gstdshowaudiodec.c:
* sys/dshowdecwrapper/gstdshowaudiodec.cpp:
* sys/dshowdecwrapper/gstdshowaudiodec.h:
* sys/dshowdecwrapper/gstdshowdecwrapper.c:
* sys/dshowdecwrapper/gstdshowdecwrapper.cpp:
* sys/dshowdecwrapper/gstdshowdecwrapper.h:
* sys/dshowdecwrapper/gstdshowfakesrc.cpp:
* sys/dshowdecwrapper/gstdshowfakesrc.h:
* sys/dshowdecwrapper/gstdshowutil.cpp:
* sys/dshowdecwrapper/gstdshowutil.h:
* sys/dshowdecwrapper/gstdshowvideodec.c:
* sys/dshowdecwrapper/gstdshowvideodec.cpp:
* sys/dshowdecwrapper/gstdshowvideodec.h:
Major rewrite of dshowdecwrapper. Converts code to
C++, moves to direct use of DirectShow base classes,
make a lot of code clearer, simplify, etc.
Fix decode of MP3 on Vista by working around an apparent
bug in the decoder.

1 files changed, 1074 insertions, 0 deletions

diff --git a/sys/dshowdecwrapper/gstdshowaudiodec.cpp b/sys/dshowdecwrapper/gstdshowaudiodec.cpp
new file mode 100644
index 00000000..dd2d98d7
--- /dev/null
+++ b/sys/dshowdecwrapper/gstdshowaudiodec.cpp
@@ -0,0 +1,1074 @@
+/*
+ * 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 AudioCodecEntry *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);
+
+/* 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, GstCaps *caps);
+
+/* All the GUIDs we want are generated from the FOURCC like this */
+#define GUID_MEDIASUBTYPE_FROM_FOURCC(fourcc) \
+    { fourcc , 0x0000, 0x0010, \
+    { 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71 }}
+
+static const AudioCodecEntry audio_dec_codecs[] = {
+  {"dshowadec_wma1",
+   "Windows Media Audio 7",
+   WAVE_FORMAT_MSAUDIO1,
+   "audio/x-wma, wmaversion = (int) 1"},
+  {"dshowadec_wma2",
+   "Windows Media Audio 8",
+   WAVE_FORMAT_WMAUDIO2,
+   "audio/x-wma, wmaversion = (int) 2"},
+  {"dshowadec_wma3",
+   "Windows Media Audio 9 Professional",
+   WAVE_FORMAT_WMAUDIO3,
+   "audio/x-wma, wmaversion = (int) 3"},
+  {"dshowadec_wma4",
+   "Windows Media Audio 9 Lossless",
+   WAVE_FORMAT_WMAUDIO_LOSSLESS,
+   "audio/x-wma, wmaversion = (int) 4"},
+  {"dshowadec_wms",
+   "Windows Media Audio Voice v9",
+   WAVE_FORMAT_WMAVOICE9,
+   "audio/x-wms"},
+  {"dshowadec_mp3",
+   "MPEG Layer 3 Audio",
+   WAVE_FORMAT_MPEGLAYER3,
+   "audio/mpeg, "
+       "mpegversion = (int) 1, "
+       "layer = (int)3, "
+       "rate = (int) [ 8000, 48000 ], "
+       "channels = (int) [ 1, 2 ], "
+       "parsed= (boolean) true"},
+  {"dshowadec_mpeg_1_2",
+   "MPEG Layer 1,2 Audio",
+   WAVE_FORMAT_MPEG,
+   "audio/mpeg, "
+       "mpegversion = (int) 1, "
+       "layer = (int) [ 1, 2 ], "
+       "rate = (int) [ 8000, 48000 ], "
+       "channels = (int) [ 1, 2 ], "
+       "parsed= (boolean) true"},
+};
+
+HRESULT AudioFakeSink::DoRenderSample(IMediaSample *pMediaSample)
+{
+  GstBuffer *out_buf = NULL;
+  gboolean in_seg = FALSE;
+  GstClockTime buf_start, buf_stop;
+  gint64 clip_start = 0, clip_stop = 0;
+  guint start_offset = 0, stop_offset;
+  GstClockTime duration;
+
+  if(pMediaSample)
+  {
+    BYTE *pBuffer = NULL;
+    LONGLONG lStart = 0, lStop = 0;
+    long size = pMediaSample->GetActualDataLength();
+
+    pMediaSample->GetPointer(&pBuffer);
+    pMediaSample->GetTime(&lStart, &lStop);
+    
+    if (!GST_CLOCK_TIME_IS_VALID (mDec->timestamp)) {
+      // Convert REFERENCE_TIME to GST_CLOCK_TIME
+      mDec->timestamp = (GstClockTime)lStart * 100;
+    }
+    duration = (lStop - lStart) * 100;
+
+    buf_start = mDec->timestamp;
+    buf_stop = mDec->timestamp + duration;
+
+    /* save stop position to start next buffer with it */
+    mDec->timestamp = buf_stop;
+
+    /* check if this buffer is in our current segment */
+    in_seg = gst_segment_clip (mDec->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_DEBUG_OBJECT (mDec,
+          "buffer is out of segment, start %" GST_TIME_FORMAT " stop %"
+          GST_TIME_FORMAT, GST_TIME_ARGS (buf_start), GST_TIME_ARGS (buf_stop));
+      goto done;
+    }
+
+    /* buffer is entirely or partially in-segment, so allocate a
+     * GstBuffer for output, and clip if required */
+
+    /* allocate a new buffer for raw audio */
+    mDec->last_ret = gst_pad_alloc_buffer (mDec->srcpad, 
+        GST_BUFFER_OFFSET_NONE,
+        size,
+        GST_PAD_CAPS (mDec->srcpad), &out_buf);
+    if (!out_buf) {
+      GST_WARNING_OBJECT (mDec, "cannot allocate a new GstBuffer");
+      goto done;
+    }
+
+    /* set buffer properties */
+    GST_BUFFER_TIMESTAMP (out_buf) = buf_start;
+    GST_BUFFER_DURATION (out_buf) = duration;
+    memcpy (GST_BUFFER_DATA (out_buf), pBuffer,
+        MIN ((unsigned int)size, GST_BUFFER_SIZE (out_buf)));
+
+    /* we have to remove some heading samples */
+    if (clip_start > buf_start) {
+      start_offset = (guint)gst_util_uint64_scale_int (clip_start - buf_start,
+          mDec->rate, GST_SECOND) * mDec->depth / 8 * mDec->channels;
+    }
+    else
+      start_offset = 0;
+    /* we have to remove some trailing samples */
+    if (clip_stop < buf_stop) {
+      stop_offset = (guint)gst_util_uint64_scale_int (buf_stop - clip_stop,
+          mDec->rate, GST_SECOND) * mDec->depth / 8 * mDec->channels;
+    }
+    else
+      stop_offset = size;
+
+    /* 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 (mDec->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 */
+    mDec->timestamp = clip_stop;
+
+    GST_DEBUG_OBJECT (mDec,
+        "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)));
+
+    mDec->last_ret = gst_pad_push (mDec->srcpad, out_buf);
+  }
+
+done:
+  return S_OK;
+}
+    
+HRESULT AudioFakeSink::CheckMediaType(const CMediaType *pmt)
+{
+  if(pmt != NULL)
+  {
+    /* The Vista MP3 decoder (and possibly others?) outputs an 
+     * AM_MEDIA_TYPE with the wrong cbFormat. So, rather than using
+     * CMediaType.operator==, we implement a sufficient check ourselves.
+     * I think this is a bug in the MP3 decoder.
+     */
+    if (IsEqualGUID (pmt->majortype, m_MediaType.majortype) &&
+        IsEqualGUID (pmt->subtype, m_MediaType.subtype) &&
+        IsEqualGUID (pmt->formattype, m_MediaType.formattype))
+    {
+      /* Types are the same at the top-level. Now, we need to compare
+       * the format blocks.
+       * We special case WAVEFORMATEX to not check that 
+       * pmt->cbFormat == m_MediaType.cbFormat, though the actual format
+       * blocks must still be the same.
+       */
+      if (pmt->formattype == FORMAT_WaveFormatEx) {
+        if (pmt->cbFormat >= sizeof (WAVEFORMATEX) &&
+            m_MediaType.cbFormat >= sizeof (WAVEFORMATEX))
+        {
+          WAVEFORMATEX *wf1 = (WAVEFORMATEX *)pmt->pbFormat;
+          WAVEFORMATEX *wf2 = (WAVEFORMATEX *)m_MediaType.pbFormat;
+          if (wf1->cbSize == wf2->cbSize &&
+              memcmp (wf1, wf2, sizeof(WAVEFORMATEX) + wf1->cbSize) == 0)
+            return S_OK;
+        }
+      }
+      else {
+        if (pmt->cbFormat == m_MediaType.cbFormat &&
+             pmt->cbFormat == 0 ||
+             (pmt->pbFormat != NULL && m_MediaType.pbFormat != NULL &&
+                 memcmp (pmt->pbFormat, m_MediaType.pbFormat, pmt->cbFormat) == 0))
+          return S_OK;
+      }
+    }
+  }
+
+  return S_FALSE;
+}
+
+static void
+gst_dshowaudiodec_base_init (gpointer klass)
+{
+  GstDshowAudioDecClass *audiodec_class = (GstDshowAudioDecClass *)klass;
+  GstPadTemplate *src, *sink;
+  GstCaps *srccaps, *sinkcaps;
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementDetails details;
+
+  audiodec_class->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);
+
+  srccaps = gst_caps_from_string (
+      "audio/x-raw-int,"
+      "width = (int)[1, 32],"
+      "depth = (int)[1, 32],"
+      "rate = (int)[1, MAX],"
+      "channels = (int)[1, MAX],"
+      "signed = (boolean)true,"
+      "endianness = (int)" G_STRINGIFY(G_LITTLE_ENDIAN));
+
+  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 = (GstElementClass *)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);
+  HRESULT hr;
+
+  /* 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->fakesrc = NULL;
+  adec->fakesink = NULL;
+
+  adec->decfilter = 0;
+  adec->filtergraph = 0;
+  adec->mediafilter = 0;
+
+  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;
+
+  adec->last_ret = GST_FLOW_OK;
+
+  hr = CoInitialize (0);
+  if (SUCCEEDED(hr)) {
+    adec->comInitialized = TRUE;
+  }
+}
+
+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;
+  }
+
+  if (adec->comInitialized) {
+    CoUninitialize ();
+    adec->comInitialized = FALSE;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+
+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));
+
+  ret = gst_dshowaudiodec_setup_graph (adec, caps);
+end:
+  gst_object_unref (adec);
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_dshowaudiodec_chain (GstPad * pad, GstBuffer * buffer)
+{
+  GstDshowAudioDec *adec = (GstDshowAudioDec *) gst_pad_get_parent (pad);
+  bool discont = FALSE;
+
+  if (!adec->setup) {
+    /* we are not set up */
+    GST_WARNING_OBJECT (adec, "Decoder not set up, failing");
+    adec->last_ret = GST_FLOW_WRONG_STATE;
+    goto beach;
+  }
+
+  if (GST_FLOW_IS_FATAL (adec->last_ret)) {
+    GST_DEBUG_OBJECT (adec, "last decoding iteration generated a fatal error "
+        "%s", gst_flow_get_name (adec->last_ret));
+    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);
+    discont = TRUE;
+  }
+
+  /* push the buffer to the directshow decoder */
+  adec->fakesrc->GetOutputPin()->PushBuffer (
+      GST_BUFFER_DATA (buffer), GST_BUFFER_TIMESTAMP (buffer),
+      GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer),
+      GST_BUFFER_SIZE (buffer), (bool)discont);
+
+beach:
+  gst_buffer_unref (buffer);
+  gst_object_unref (adec);
+  return adec->last_ret;
+}
+
+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;
+  }
+
+  gst_object_unref (adec);
+
+  return ret;
+}
+
+static gboolean
+gst_dshowaudiodec_flush (GstDshowAudioDec * adec)
+{
+  if (!adec->fakesrc)
+    return FALSE;
+
+  /* flush dshow decoder and reset timestamp */
+  adec->fakesrc->GetOutputPin()->Flush();
+  adec->timestamp = GST_CLOCK_TIME_NONE;
+
+  return TRUE;
+}
+
+static AM_MEDIA_TYPE *
+dshowaudiodec_set_input_format (GstDshowAudioDec *adec, GstCaps *caps)
+{
+  AM_MEDIA_TYPE *mediatype;
+  WAVEFORMATEX *format;
+  GstDshowAudioDecClass *klass =
+      (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
+  const AudioCodecEntry *codec_entry = klass->entry;
+  int size;
+
+  mediatype = (AM_MEDIA_TYPE *)g_malloc0 (sizeof(AM_MEDIA_TYPE));
+  mediatype->majortype = MEDIATYPE_Audio;
+  GUID subtype = GUID_MEDIASUBTYPE_FROM_FOURCC (0x00000000);
+  subtype.Data1 = codec_entry->format;
+  mediatype->subtype = subtype;
+  mediatype->bFixedSizeSamples = TRUE;
+  mediatype->bTemporalCompression = FALSE;
+  if (adec->block_align)
+    mediatype->lSampleSize = adec->block_align;
+  else
+    mediatype->lSampleSize = 8192; /* need to evaluate it dynamically */
+  mediatype->formattype = FORMAT_WaveFormatEx;
+
+  /* We need this special behaviour for layers 1 and 2 (layer 3 uses a different
+   * decoder which doesn't need this */
+  if (adec->layer == 1 || adec->layer == 2) {
+    MPEG1WAVEFORMAT *mpeg1_format;
+    int version, samples;
+    GstStructure *structure = gst_caps_get_structure (caps, 0);
+
+    size = sizeof (MPEG1WAVEFORMAT);
+    format = (WAVEFORMATEX *)g_malloc0 (size);
+    format->cbSize = sizeof (MPEG1WAVEFORMAT) - sizeof (WAVEFORMATEX);
+    format->wFormatTag = WAVE_FORMAT_MPEG;
+
+    mpeg1_format = (MPEG1WAVEFORMAT *) format;
+
+    mpeg1_format->wfx.nChannels = adec->channels;
+    if (adec->channels == 2)
+      mpeg1_format->fwHeadMode = ACM_MPEG_STEREO;
+    else
+      mpeg1_format->fwHeadMode = ACM_MPEG_SINGLECHANNEL;
+    
+    mpeg1_format->fwHeadModeExt = 0;
+    mpeg1_format->wHeadEmphasis = 0;
+    mpeg1_format->fwHeadFlags = 0;
+
+    switch (adec->layer) {
+      case 1:
+        mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER3;
+        break;
+      case 2:
+        mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER2;
+        break;
+      case 3:
+        mpeg1_format->fwHeadLayer = ACM_MPEG_LAYER1;
+        break;
+    };
+
+    gst_structure_get_int (structure, "mpegaudioversion", &version);
+    if (adec->layer == 1) {
+      samples = 384;
+    } else {
+      if (version == 1) {
+        samples = 576;
+      } else {
+        samples = 1152;
+      }
+    }
+    mpeg1_format->wfx.nBlockAlign = (WORD) samples;
+    mpeg1_format->wfx.nSamplesPerSec = adec->rate;
+    mpeg1_format->dwHeadBitrate = 128000; /* This doesn't seem to matter */
+    mpeg1_format->wfx.nAvgBytesPerSec = mpeg1_format->dwHeadBitrate / 8;
+  } 
+  else 
+  {
+    size = sizeof (WAVEFORMATEX) +
+        (adec->codec_data ? GST_BUFFER_SIZE (adec->codec_data) : 0);
+    format = (WAVEFORMATEX *)g_malloc0 (size);
+    if (adec->codec_data) {     /* Codec data is appended after our header */
+      memcpy (((guchar *) format) + sizeof (WAVEFORMATEX),
+          GST_BUFFER_DATA (adec->codec_data),
+          GST_BUFFER_SIZE (adec->codec_data));
+      format->cbSize = GST_BUFFER_SIZE (adec->codec_data);
+    }
+
+    format->wFormatTag = codec_entry->format;
+    format->nChannels = adec->channels;
+    format->nSamplesPerSec = adec->rate;
+    format->nAvgBytesPerSec = adec->bitrate / 8;
+    format->nBlockAlign = adec->block_align;
+    format->wBitsPerSample = adec->depth;
+  }
+
+  mediatype->cbFormat = size;
+  mediatype->pbFormat = (BYTE *) format;
+
+  return mediatype;
+}
+
+static AM_MEDIA_TYPE *
+dshowaudiodec_set_output_format (GstDshowAudioDec *adec)
+{
+  AM_MEDIA_TYPE *mediatype;
+  WAVEFORMATEX *format;
+  GstDshowAudioDecClass *klass =
+      (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
+  const AudioCodecEntry *codec_entry = klass->entry;
+
+  if (!gst_dshowaudiodec_get_filter_settings (adec)) {
+    return NULL;
+  }
+
+  format = (WAVEFORMATEX *)g_malloc0(sizeof (WAVEFORMATEX));
+  format->wFormatTag = WAVE_FORMAT_PCM;
+  format->wBitsPerSample = adec->depth;
+  format->nChannels = adec->channels;
+  format->nBlockAlign = adec->channels * (adec->depth / 8);
+  format->nSamplesPerSec = adec->rate;
+  format->nAvgBytesPerSec = format->nBlockAlign * adec->rate;
+
+  mediatype = (AM_MEDIA_TYPE *)g_malloc0(sizeof (AM_MEDIA_TYPE));
+  mediatype->majortype = MEDIATYPE_Audio;
+  GUID subtype = GUID_MEDIASUBTYPE_FROM_FOURCC (WAVE_FORMAT_PCM);
+  mediatype->subtype = subtype;
+  mediatype->bFixedSizeSamples = TRUE;
+  mediatype->bTemporalCompression = FALSE;
+  mediatype->lSampleSize = format->nBlockAlign;
+  mediatype->formattype = FORMAT_WaveFormatEx;
+  mediatype->cbFormat = sizeof (WAVEFORMATEX);
+  mediatype->pbFormat = (BYTE *)format;
+  
+  return mediatype;
+}
+
+static void
+dshowadec_free_mediatype (AM_MEDIA_TYPE *mediatype)
+{
+  if (mediatype->pbFormat)
+    g_free (mediatype->pbFormat);
+  g_free (mediatype);
+}
+
+static gboolean
+gst_dshowaudiodec_setup_graph (GstDshowAudioDec * adec, GstCaps *caps)
+{
+  gboolean ret = FALSE;
+  GstDshowAudioDecClass *klass =
+      (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
+  HRESULT hres;
+  GstCaps *outcaps;
+  AM_MEDIA_TYPE *output_mediatype = NULL;
+  AM_MEDIA_TYPE *input_mediatype = NULL;
+  CComPtr<IPin> output_pin;
+  CComPtr<IPin> input_pin;
+  const AudioCodecEntry *codec_entry = klass->entry;
+  CComQIPtr<IBaseFilter> srcfilter;
+  CComQIPtr<IBaseFilter> sinkfilter;
+
+  input_mediatype = dshowaudiodec_set_input_format (adec, caps);
+
+  adec->fakesrc->GetOutputPin()->SetMediaType (input_mediatype);
+
+  srcfilter = adec->fakesrc;
+
+  /* connect our fake source to decoder */
+  output_pin = gst_dshow_get_pin_from_filter (srcfilter, PINDIR_OUTPUT);
+  if (!output_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get output pin from our directshow fakesrc filter"), (NULL));
+    goto end;
+  }
+  input_pin = gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_INPUT);
+  if (!input_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get input pin from decoder filter"), (NULL));
+    goto end;
+  }
+
+  hres = adec->filtergraph->ConnectDirect (output_pin, input_pin,
+      NULL);
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't connect fakesrc with decoder (error=%x)", hres), (NULL));
+    goto end;
+  }
+
+  output_mediatype = dshowaudiodec_set_output_format (adec);
+  if (!output_mediatype) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get audio output format from decoder"), (NULL));
+    goto end;
+  }
+
+  adec->fakesink->SetMediaType(output_mediatype);
+
+  outcaps = gst_caps_new_simple ("audio/x-raw-int",
+      "width", G_TYPE_INT, adec->depth,
+      "depth", G_TYPE_INT, adec->depth,
+      "rate", G_TYPE_INT, adec->rate,
+      "channels", G_TYPE_INT, adec->channels, 
+      "signed", G_TYPE_BOOLEAN, TRUE,
+      "endianness", G_TYPE_INT, G_LITTLE_ENDIAN,
+      NULL);
+
+  if (!gst_pad_set_caps (adec->srcpad, outcaps)) {
+    gst_caps_unref (outcaps);
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Failed to negotiate output"), (NULL));
+    goto end;
+  }
+  gst_caps_unref (outcaps);
+
+  /* connect the decoder to our fake sink */
+  output_pin = gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT);
+  if (!output_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get output pin from our decoder filter"), (NULL));
+    goto end;
+  }
+
+  sinkfilter = adec->fakesink;
+  input_pin = gst_dshow_get_pin_from_filter (sinkfilter, PINDIR_INPUT);
+  if (!input_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't get input pin from our directshow fakesink filter"), (NULL));
+    goto end;
+  }
+
+  hres = adec->filtergraph->ConnectDirect(output_pin, input_pin, NULL);
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't connect decoder with fakesink (error=%x)", hres), (NULL));
+    goto end;
+  }
+
+  hres = adec->mediafilter->Run (-1);
+  if (hres != S_OK) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("Can't run the directshow graph (error=%x)", hres), (NULL));
+    goto end;
+  }
+
+  ret = TRUE;
+  adec->setup = TRUE;
+end:
+  if (input_mediatype)
+    dshowadec_free_mediatype (input_mediatype);
+  if (output_mediatype)
+    dshowadec_free_mediatype (output_mediatype);
+
+  return ret;
+}
+
+static gboolean
+gst_dshowaudiodec_get_filter_settings (GstDshowAudioDec * adec)
+{
+  CComPtr<IPin> output_pin;
+  CComPtr<IEnumMediaTypes> enum_mediatypes;
+  HRESULT hres;
+  ULONG fetched;
+  BOOL ret = FALSE;
+
+  if (adec->decfilter == 0)
+    return FALSE;
+
+  output_pin = gst_dshow_get_pin_from_filter (adec->decfilter, PINDIR_OUTPUT);
+  if (!output_pin) {
+    GST_ELEMENT_ERROR (adec, CORE, NEGOTIATION,
+        ("failed getting ouput pin from the decoder"), (NULL));
+    return FALSE;
+  }
+
+  hres = output_pin->EnumMediaTypes (&enum_mediatypes);
+  if (hres == S_OK && enum_mediatypes) {
+    AM_MEDIA_TYPE *mediatype = NULL;
+
+    enum_mediatypes->Reset();
+    while (!ret && enum_mediatypes->Next(1, &mediatype, &fetched) == S_OK) 
+    {
+      if (IsEqualGUID (mediatype->subtype, MEDIASUBTYPE_PCM) &&
+          IsEqualGUID (mediatype->formattype, FORMAT_WaveFormatEx))
+      {
+        WAVEFORMATEX *audio_info = (WAVEFORMATEX *) mediatype->pbFormat;
+
+        adec->channels = audio_info->nChannels;
+        adec->depth = audio_info->wBitsPerSample;
+        adec->rate = audio_info->nSamplesPerSec;
+        ret = TRUE;
+      }
+      DeleteMediaType (mediatype);
+    }
+  }
+
+  return ret;
+}
+
+static gboolean
+gst_dshowaudiodec_create_graph_and_filters (GstDshowAudioDec * adec)
+{
+  HRESULT hres;
+  GstDshowAudioDecClass *klass =
+      (GstDshowAudioDecClass *) G_OBJECT_GET_CLASS (adec);
+  CComQIPtr<IBaseFilter> srcfilter;
+  CComQIPtr<IBaseFilter> sinkfilter;
+  GUID insubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (klass->entry->format);
+  GUID outsubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (WAVE_FORMAT_PCM);
+
+  /* create the filter graph manager object */
+  hres = adec->filtergraph.CoCreateInstance (
+      CLSID_FilterGraph, NULL, CLSCTX_INPROC);
+  if (FAILED (hres)) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't create an instance of the directshow graph manager (error=%d)",
+            hres), (NULL));
+    goto error;
+  }
+
+  hres = adec->filtergraph->QueryInterface (&adec->mediafilter);
+  if (FAILED (hres)) {
+    GST_WARNING_OBJECT (adec, "Can't QI filtergraph to mediafilter");
+    goto error;
+  }
+
+  /* create fake src filter */
+  adec->fakesrc = new FakeSrc();
+  /* Created with a refcount of zero, so increment that */
+  adec->fakesrc->AddRef();
+
+  /* create decoder filter */
+  if (!gst_dshow_find_filter (MEDIATYPE_Audio,
+          insubtype,
+          MEDIATYPE_Audio,
+          outsubtype,
+          NULL, &adec->decfilter)) {
+    GST_ELEMENT_ERROR (adec, STREAM, FAILED,
+        ("Can't create an instance of the decoder filter"), (NULL));
+    goto error;
+  }
+
+  /* create fake sink filter */
+  adec->fakesink = new AudioFakeSink(adec);
+  /* Created with a refcount of zero, so increment that */
+  adec->fakesink->AddRef();
+
+  /* add filters to the graph */
+  srcfilter = adec->fakesrc;
+  hres = adec->filtergraph->AddFilter (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 = adec->filtergraph->AddFilter(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;
+  }
+
+  sinkfilter = adec->fakesink;
+  hres = adec->filtergraph->AddFilter(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->fakesrc) {
+    adec->fakesrc->Release();
+    adec->fakesrc = NULL;
+  }
+  if (adec->fakesink) {
+    adec->fakesink->Release();
+    adec->fakesink = NULL;
+  }
+  adec->decfilter = 0;
+  adec->mediafilter = 0;
+  adec->filtergraph = 0;
+
+  return FALSE;
+}
+
+static gboolean
+gst_dshowaudiodec_destroy_graph_and_filters (GstDshowAudioDec * adec)
+{
+  if (adec->mediafilter) {
+    adec->mediafilter->Stop();
+  }
+
+  if (adec->fakesrc) {
+    if (adec->filtergraph) {
+      CComQIPtr<IBaseFilter> filter = adec->fakesrc;
+      adec->filtergraph->RemoveFilter(filter);
+    }
+    adec->fakesrc->Release();
+    adec->fakesrc = NULL;
+  }
+  if (adec->decfilter) {
+    if (adec->filtergraph)
+      adec->filtergraph->RemoveFilter(adec->decfilter);
+    adec->decfilter = 0;
+  }
+  if (adec->fakesink) {
+    if (adec->filtergraph) {
+      CComQIPtr<IBaseFilter> filter = adec->fakesink;
+      adec->filtergraph->RemoveFilter(filter);
+    }
+
+    adec->fakesink->Release();
+    adec->fakesink = NULL;
+  }
+  adec->mediafilter = 0;
+  adec->filtergraph = 0;
+
+  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;
+  HRESULT hr;
+
+  GST_DEBUG_CATEGORY_INIT (dshowaudiodec_debug, "dshowaudiodec", 0,
+      "Directshow filter audio decoder");
+
+  hr = CoInitialize(0);
+  for (i = 0; i < sizeof (audio_dec_codecs) / sizeof (AudioCodecEntry); i++) {
+    GType type;
+
+    GUID insubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (audio_dec_codecs[i].format);
+    GUID outsubtype = GUID_MEDIASUBTYPE_FROM_FOURCC (WAVE_FORMAT_PCM);
+    if (gst_dshow_find_filter (MEDIATYPE_Audio,
+            insubtype,
+            MEDIATYPE_Audio,
+            outsubtype,
+            NULL, 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, (GTypeFlags)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);
+    }
+  }
+
+  if (SUCCEEDED(hr))
+    CoUninitialize ();
+
+  return TRUE;
+}