1 files changed, 859 insertions, 0 deletions

diff --git a/sys/dshowsrcwrapper/gstdshowaudiosrc.cpp b/sys/dshowsrcwrapper/gstdshowaudiosrc.cpp
new file mode 100755
index 000000000..8961c96d1
--- /dev/null
+++ b/sys/dshowsrcwrapper/gstdshowaudiosrc.cpp
@@ -0,0 +1,859 @@
+/* GStreamer
+ * Copyright (C)  2007 Sebastien Moutte <sebastien@moutte.net>
+ *
+ * gstdshowaudiosrc.c: 
+ *
+ * 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 "gstdshowaudiosrc.h"
+
+static const GstElementDetails gst_dshowaudiosrc_details =
+GST_ELEMENT_DETAILS ("Directshow audio capture source",
+    "Source/Audio",
+    "Receive data from a directshow audio capture graph",
+    "Sebastien Moutte <sebastien@moutte.net>");
+
+GST_DEBUG_CATEGORY_STATIC (dshowaudiosrc_debug);
+#define GST_CAT_DEFAULT dshowaudiosrc_debug
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "endianness = (int) { " G_STRINGIFY (G_BYTE_ORDER) " }, "
+        "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_dshowaudiosrc_init_interfaces (GType type);
+
+GST_BOILERPLATE_FULL (GstDshowAudioSrc, gst_dshowaudiosrc, GstAudioSrc,
+    GST_TYPE_AUDIO_SRC, gst_dshowaudiosrc_init_interfaces);
+
+enum
+{
+  PROP_0,
+  PROP_DEVICE,
+  PROP_DEVICE_NAME
+};
+
+static void gst_dshowaudiosrc_probe_interface_init (GstPropertyProbeInterface *
+    iface);
+static const GList *gst_dshowaudiosrc_probe_get_properties (GstPropertyProbe *
+    probe);
+static GValueArray *gst_dshowaudiosrc_probe_get_values (GstPropertyProbe *
+    probe, guint prop_id, const GParamSpec * pspec);
+static GValueArray *gst_dshowaudiosrc_get_device_name_values (GstDshowAudioSrc *
+    src);
+
+
+static void gst_dshowaudiosrc_dispose (GObject * gobject);
+static void gst_dshowaudiosrc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_dshowaudiosrc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static GstCaps *gst_dshowaudiosrc_get_caps (GstBaseSrc * src);
+static GstStateChangeReturn gst_dshowaudiosrc_change_state (GstElement *
+    element, GstStateChange transition);
+
+static gboolean gst_dshowaudiosrc_open (GstAudioSrc * asrc);
+static gboolean gst_dshowaudiosrc_prepare (GstAudioSrc * asrc,
+    GstRingBufferSpec * spec);
+static gboolean gst_dshowaudiosrc_unprepare (GstAudioSrc * asrc);
+static gboolean gst_dshowaudiosrc_close (GstAudioSrc * asrc);
+static guint gst_dshowaudiosrc_read (GstAudioSrc * asrc, gpointer data,
+    guint length);
+static guint gst_dshowaudiosrc_delay (GstAudioSrc * asrc);
+static void gst_dshowaudiosrc_reset (GstAudioSrc * asrc);
+
+/* utils */
+static GstCaps *gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc *
+    src, IPin * pin, IAMStreamConfig * streamcaps);
+static gboolean gst_dshowaudiosrc_push_buffer (byte * buffer, long size,
+    byte * src_object, UINT64 start, UINT64 stop);
+
+static void
+gst_dshowaudiosrc_init_interfaces (GType type)
+{
+  static const GInterfaceInfo dshowaudiosrc_info = {
+    (GInterfaceInitFunc) gst_dshowaudiosrc_probe_interface_init,
+    NULL,
+    NULL,
+  };
+
+  g_type_add_interface_static (type,
+      GST_TYPE_PROPERTY_PROBE, &dshowaudiosrc_info);
+}
+
+static void
+gst_dshowaudiosrc_probe_interface_init (GstPropertyProbeInterface * iface)
+{
+  iface->get_properties = gst_dshowaudiosrc_probe_get_properties;
+/*  iface->needs_probe    = gst_dshowaudiosrc_probe_needs_probe;
+  iface->probe_property = gst_dshowaudiosrc_probe_probe_property;*/
+  iface->get_values = gst_dshowaudiosrc_probe_get_values;
+}
+
+static void
+gst_dshowaudiosrc_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+
+  gst_element_class_set_details (element_class, &gst_dshowaudiosrc_details);
+}
+
+static void
+gst_dshowaudiosrc_class_init (GstDshowAudioSrcClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSrcClass *gstbasesrc_class;
+  GstAudioSrcClass *gstaudiosrc_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasesrc_class = (GstBaseSrcClass *) klass;
+  gstaudiosrc_class = (GstAudioSrcClass *) klass;
+
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_dispose);
+  gobject_class->set_property =
+      GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_set_property);
+  gobject_class->get_property =
+      GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_get_property);
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_change_state);
+
+  gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_get_caps);
+
+  gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_open);
+  gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_prepare);
+  gstaudiosrc_class->unprepare =
+      GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_unprepare);
+  gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_close);
+  gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_read);
+  gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_delay);
+  gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_reset);
+
+  g_object_class_install_property
+      (gobject_class, PROP_DEVICE,
+      g_param_spec_string ("device", "Device",
+          "Directshow device reference (classID/name)", NULL,
+          static_cast<GParamFlags>(G_PARAM_READWRITE)));
+
+  g_object_class_install_property
+      (gobject_class, PROP_DEVICE_NAME,
+      g_param_spec_string ("device-name", "Device name",
+          "Human-readable name of the sound device", NULL, 
+          static_cast<GParamFlags>(G_PARAM_READWRITE)));
+
+  GST_DEBUG_CATEGORY_INIT (dshowaudiosrc_debug, "dshowaudiosrc", 0,
+      "Directshow audio source");
+}
+
+static void
+gst_dshowaudiosrc_init (GstDshowAudioSrc * src, GstDshowAudioSrcClass * klass)
+{
+  src->device = NULL;
+  src->device_name = NULL;
+  src->audio_cap_filter = NULL;
+  src->dshow_fakesink = NULL;
+  src->media_filter = NULL;
+  src->filter_graph = NULL;
+  src->caps = NULL;
+  src->pins_mediatypes = NULL;
+
+  src->gbarray = g_byte_array_new ();
+  src->gbarray_lock = g_mutex_new ();
+
+  src->is_running = FALSE;
+
+  CoInitializeEx (NULL, COINIT_MULTITHREADED);
+}
+
+static void
+gst_dshowaudiosrc_dispose (GObject * gobject)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (gobject);
+
+  if (src->device) {
+    g_free (src->device);
+    src->device = NULL;
+  }
+
+  if (src->device_name) {
+    g_free (src->device_name);
+    src->device_name = NULL;
+  }
+
+  if (src->caps) {
+    gst_caps_unref (src->caps);
+    src->caps = NULL;
+  }
+
+  if (src->pins_mediatypes) {
+    gst_dshow_free_pins_mediatypes (src->pins_mediatypes);
+    src->pins_mediatypes = NULL;
+  }
+
+  if (src->gbarray) {
+    g_byte_array_free (src->gbarray, TRUE);
+    src->gbarray = NULL;
+  }
+
+  if (src->gbarray_lock) {
+    g_mutex_free (src->gbarray_lock);
+    src->gbarray_lock = NULL;
+  }
+
+  /* clean dshow */
+  if (src->audio_cap_filter)
+    src->audio_cap_filter->Release();
+
+  CoUninitialize ();
+
+  G_OBJECT_CLASS (parent_class)->dispose (gobject);
+}
+
+
+static const GList *
+gst_dshowaudiosrc_probe_get_properties (GstPropertyProbe * probe)
+{
+  GObjectClass *klass = G_OBJECT_GET_CLASS (probe);
+  static GList *props = NULL;
+
+  if (!props) {
+    GParamSpec *pspec;
+
+    pspec = g_object_class_find_property (klass, "device-name");
+    props = g_list_append (props, pspec);
+  }
+
+  return props;
+}
+
+static GValueArray *
+gst_dshowaudiosrc_get_device_name_values (GstDshowAudioSrc * src)
+{
+  GValueArray *array = g_value_array_new (0);
+  GValue value = { 0 };
+  ICreateDevEnum *devices_enum = NULL;
+  IEnumMoniker *moniker_enum = NULL;
+  IMoniker *moniker = NULL;
+  HRESULT hres = S_FALSE;
+  ULONG fetched;
+
+  g_value_init (&value, G_TYPE_STRING);
+
+  hres = CoCreateInstance (CLSID_SystemDeviceEnum, NULL, CLSCTX_INPROC_SERVER,
+      IID_ICreateDevEnum, (LPVOID *) &devices_enum);
+  if (hres != S_OK) {
+    GST_CAT_ERROR (dshowaudiosrc_debug,
+        "Can't create an instance of the system device enumerator (error=%d)",
+        hres);
+    array = NULL;
+    goto clean;
+  }
+
+  hres = devices_enum->CreateClassEnumerator(CLSID_AudioInputDeviceCategory, 
+    &moniker_enum, 0);
+  if (hres != S_OK || !moniker_enum) {
+    GST_CAT_ERROR (dshowaudiosrc_debug,
+        "Can't get enumeration of audio devices (error=%d)", hres);
+    array = NULL;
+    goto clean;
+  }
+
+  moniker_enum->Reset();
+
+  while (hres = moniker_enum->Next(1, &moniker, &fetched),
+      hres == S_OK) {
+    IPropertyBag *property_bag = NULL;
+
+    hres = moniker->BindToStorage(NULL, NULL, IID_IPropertyBag, 
+      (LPVOID *) &property_bag);
+    if (SUCCEEDED (hres) && property_bag) {
+      VARIANT varFriendlyName;
+
+      VariantInit (&varFriendlyName);
+      hres = property_bag->Read(L"FriendlyName", &varFriendlyName, NULL);
+      if (hres == S_OK && varFriendlyName.bstrVal) {
+        gchar *friendly_name =
+            g_utf16_to_utf8 ((const gunichar2 *) varFriendlyName.bstrVal,
+            wcslen (varFriendlyName.bstrVal), NULL, NULL, NULL);
+
+        g_value_set_string (&value, friendly_name);
+        g_value_array_append (array, &value);
+        g_value_unset (&value);
+        g_free (friendly_name);
+        SysFreeString (varFriendlyName.bstrVal);
+      }
+      property_bag->Release();
+    }
+    moniker->Release();
+  }
+
+clean:
+  if (moniker_enum)
+    moniker_enum->Release();
+
+  if (devices_enum)
+    devices_enum->Release();
+
+  return array;
+}
+
+static GValueArray *
+gst_dshowaudiosrc_probe_get_values (GstPropertyProbe * probe,
+    guint prop_id, const GParamSpec * pspec)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (probe);
+  GValueArray *array = NULL;
+
+  switch (prop_id) {
+    case PROP_DEVICE_NAME:
+      array = gst_dshowaudiosrc_get_device_name_values (src);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (probe, prop_id, pspec);
+      break;
+  }
+
+  return array;
+}
+
+static void
+gst_dshowaudiosrc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (object);
+
+  switch (prop_id) {
+    case PROP_DEVICE:
+    {
+      if (src->device) {
+        g_free (src->device);
+        src->device = NULL;
+      }
+      if (g_value_get_string (value)) {
+        src->device = g_strdup (g_value_get_string (value));
+      }
+      break;
+    }
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_dshowaudiosrc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+
+}
+
+static GstCaps *
+gst_dshowaudiosrc_get_caps (GstBaseSrc * basesrc)
+{
+  HRESULT hres = S_OK;
+  IBindCtx *lpbc = NULL;
+  IMoniker *audiom = NULL;
+  DWORD dwEaten;
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (basesrc);
+  gunichar2 *unidevice = NULL;
+
+  if (src->device) {
+    g_free (src->device);
+    src->device = NULL;
+  }
+
+  src->device =
+      gst_dshow_getdevice_from_devicename (&CLSID_AudioInputDeviceCategory,
+      &src->device_name);
+  if (!src->device) {
+    GST_CAT_ERROR (dshowaudiosrc_debug, "No audio device found.");
+    return NULL;
+  }
+  unidevice =
+      g_utf8_to_utf16 (src->device, strlen (src->device), NULL, NULL, NULL);
+
+  if (!src->audio_cap_filter) {
+    hres = CreateBindCtx (0, &lpbc);
+    if (SUCCEEDED (hres)) {
+      hres = MkParseDisplayName (lpbc, (LPCOLESTR) unidevice, &dwEaten, &audiom);
+      if (SUCCEEDED (hres)) {
+        hres = audiom->BindToObject(lpbc, NULL, IID_IBaseFilter, 
+          (LPVOID *) &src->audio_cap_filter);
+        audiom->Release();
+      }
+      lpbc->Release();
+    }
+  }
+
+  if (src->audio_cap_filter && !src->caps) {
+    /* get the capture pins supported types */
+    IPin *capture_pin = NULL;
+    IEnumPins *enumpins = NULL;
+    HRESULT hres;
+
+    hres = src->audio_cap_filter->EnumPins(&enumpins);
+    if (SUCCEEDED (hres)) {
+      while (enumpins->Next(1, &capture_pin, NULL) == S_OK) {
+        IKsPropertySet *pKs = NULL;
+
+        hres = capture_pin->QueryInterface(IID_IKsPropertySet, (LPVOID *) &pKs);
+        if (SUCCEEDED (hres) && pKs) {
+          DWORD cbReturned;
+          GUID pin_category;
+          RPC_STATUS rpcstatus;
+
+          hres =
+              pKs->Get(AMPROPSETID_Pin,
+              AMPROPERTY_PIN_CATEGORY, NULL, 0, &pin_category, sizeof (GUID),
+              &cbReturned);
+
+          /* we only want capture pins */
+          if (UuidCompare (&pin_category, (UUID *) &PIN_CATEGORY_CAPTURE,
+                  &rpcstatus) == 0) {
+            IAMStreamConfig *streamcaps = NULL;
+
+            if (SUCCEEDED (capture_pin->QueryInterface(IID_IAMStreamConfig, (LPVOID *) &streamcaps))) {
+              src->caps =
+                  gst_dshowaudiosrc_getcaps_from_streamcaps (src, capture_pin,
+                  streamcaps);
+              streamcaps->Release();
+            }
+          }
+          pKs->Release();
+        }
+        capture_pin->Release();
+      }
+      enumpins->Release();
+    }
+  }
+
+  if (unidevice) {
+    g_free (unidevice);
+  }
+
+  if (src->caps) {
+    return gst_caps_ref (src->caps);
+  }
+
+  return NULL;
+}
+
+static GstStateChangeReturn
+gst_dshowaudiosrc_change_state (GstElement * element, GstStateChange transition)
+{
+  HRESULT hres = S_FALSE;
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      if (src->media_filter)
+        hres = src->media_filter->Run(0);
+      if (hres != S_OK) {
+        GST_CAT_ERROR (dshowaudiosrc_debug,
+            "Can't RUN the directshow capture graph (error=%d)", hres);
+        src->is_running = FALSE;
+        return GST_STATE_CHANGE_FAILURE;
+      } else {
+        src->is_running = TRUE;
+      }
+      break;
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      if (src->media_filter)
+        hres = src->media_filter->Stop();
+      if (hres != S_OK) {
+        GST_CAT_ERROR (dshowaudiosrc_debug,
+            "Can't STOP the directshow capture graph (error=%d)", hres);
+        return GST_STATE_CHANGE_FAILURE;
+      }
+      src->is_running = FALSE;
+
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+
+  return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+}
+
+static gboolean
+gst_dshowaudiosrc_open (GstAudioSrc * asrc)
+{
+  HRESULT hres = S_FALSE;
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
+
+  hres = CoCreateInstance (CLSID_FilterGraph, NULL, CLSCTX_INPROC,
+      IID_IFilterGraph, (LPVOID *) & src->filter_graph);
+  if (hres != S_OK || !src->filter_graph) {
+    GST_CAT_ERROR (dshowaudiosrc_debug,
+        "Can't create an instance of the directshow graph manager (error=%d)",
+        hres);
+    goto error;
+  }
+
+  hres = src->filter_graph->QueryInterface(IID_IMediaFilter, (LPVOID *) &src->media_filter);
+  if (hres != S_OK || !src->media_filter) {
+    GST_CAT_ERROR (dshowaudiosrc_debug,
+        "Can't get IMediacontrol interface from the graph manager (error=%d)",
+        hres);
+    goto error;
+  }
+
+  hres = CoCreateInstance (CLSID_DshowFakeSink, NULL, CLSCTX_INPROC,
+      IID_IBaseFilter, (LPVOID *) & src->dshow_fakesink);
+  if (hres != S_OK || !src->dshow_fakesink) {
+    GST_CAT_ERROR (dshowaudiosrc_debug,
+        "Can't create an instance of the directshow fakesink (error=%d)", hres);
+    goto error;
+  }
+
+  hres = src->filter_graph->AddFilter(src->audio_cap_filter, L"capture");
+  if (hres != S_OK) {
+    GST_CAT_ERROR (dshowaudiosrc_debug,
+        "Can't add the directshow capture filter to the graph (error=%d)",
+        hres);
+    goto error;
+  }
+
+  hres = src->filter_graph->AddFilter(src->dshow_fakesink, L"fakesink");
+  if (hres != S_OK) {
+    GST_CAT_ERROR (dshowaudiosrc_debug,
+        "Can't add our fakesink filter to the graph (error=%d)", hres);
+    goto error;
+  }
+
+  return TRUE;
+
+error:
+  if (src->dshow_fakesink) {
+    src->dshow_fakesink->Release();
+    src->dshow_fakesink = NULL;
+  }
+
+  if (src->media_filter) {
+    src->media_filter->Release();
+    src->media_filter = NULL;
+  }
+  if (src->filter_graph) {
+    src->filter_graph->Release();
+    src->filter_graph = NULL;
+  }
+
+  return FALSE;
+}
+
+static gboolean
+gst_dshowaudiosrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
+{
+  HRESULT hres;
+  IGstDshowInterface *srcinterface = NULL;
+  IPin *input_pin = NULL;
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
+
+  /* search the negociated caps in our caps list to get its index and the corresponding mediatype */
+  if (gst_caps_is_subset (spec->caps, src->caps)) {
+    guint i = 0;
+    gint res = -1;
+
+    for (; i < gst_caps_get_size (src->caps) && res == -1; i++) {
+      GstCaps *capstmp = gst_caps_copy_nth (src->caps, i);
+
+      if (gst_caps_is_subset (spec->caps, capstmp)) {
+        res = i;
+      }
+      gst_caps_unref (capstmp);
+    }
+
+    if (res != -1 && src->pins_mediatypes) {
+      /*get the corresponding media type and build the dshow graph */
+      GstCapturePinMediaType *pin_mediatype = NULL;
+      GList *type = g_list_nth (src->pins_mediatypes, res);
+
+      if (type) {
+        pin_mediatype = (GstCapturePinMediaType *) type->data;
+
+        hres = src->dshow_fakesink->QueryInterface(IID_IGstDshowInterface, (LPVOID *) &srcinterface);
+        if (hres != S_OK || !srcinterface) {
+          GST_CAT_ERROR (dshowaudiosrc_debug,
+              "Can't get IGstDshowInterface interface from our dshow fakesink filter (error=%d)",
+              hres);
+          goto error;
+        }
+
+        srcinterface->gst_set_media_type(pin_mediatype->mediatype);
+        srcinterface->gst_set_buffer_callback(
+          (push_buffer_func) gst_dshowaudiosrc_push_buffer, (byte *) src);
+
+        if (srcinterface)
+          srcinterface->Release();
+
+        gst_dshow_get_pin_from_filter (src->dshow_fakesink, PINDIR_INPUT,
+            &input_pin);
+        if (!input_pin) {
+          GST_CAT_ERROR (dshowaudiosrc_debug,
+              "Can't get input pin from our directshow fakesink filter");
+          goto error;
+        }
+
+        hres = src->filter_graph->ConnectDirect(pin_mediatype->capture_pin, 
+          input_pin, NULL);
+        input_pin->Release();
+
+        if (hres != S_OK) {
+          GST_CAT_ERROR (dshowaudiosrc_debug,
+              "Can't connect capture filter with fakesink filter (error=%d)",
+              hres);
+          goto error;
+        }
+
+        spec->segsize = spec->rate * spec->channels;
+        spec->segtotal = 1;
+      }
+    }
+  }
+
+  return TRUE;
+
+error:
+  if (srcinterface)
+    srcinterface->Release();
+
+  return FALSE;
+}
+
+static gboolean
+gst_dshowaudiosrc_unprepare (GstAudioSrc * asrc)
+{
+  IPin *input_pin = NULL, *output_pin = NULL;
+  HRESULT hres = S_FALSE;
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
+
+  /* disconnect filters */
+  gst_dshow_get_pin_from_filter (src->audio_cap_filter, PINDIR_OUTPUT,
+      &output_pin);
+  if (output_pin) {
+    hres = src->filter_graph->Disconnect(output_pin);
+    output_pin->Release();
+  }
+
+  gst_dshow_get_pin_from_filter (src->dshow_fakesink, PINDIR_INPUT, &input_pin);
+  if (input_pin) {
+    hres = src->filter_graph->Disconnect(input_pin);
+    input_pin->Release();
+  }
+
+  return TRUE;
+}
+
+static gboolean
+gst_dshowaudiosrc_close (GstAudioSrc * asrc)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
+
+  if (!src->filter_graph)
+    return TRUE;
+
+  /*remove filters from the graph */
+  src->filter_graph->RemoveFilter(src->audio_cap_filter);
+  src->filter_graph->RemoveFilter(src->dshow_fakesink);
+
+  /*release our gstreamer dshow sink */
+  src->dshow_fakesink->Release();
+  src->dshow_fakesink = NULL;
+
+  /*release media filter interface */
+  src->media_filter->Release();
+  src->media_filter = NULL;
+
+  /*release the filter graph manager */
+  src->filter_graph->Release();
+  src->filter_graph = NULL;
+
+  return TRUE;
+}
+
+static guint
+gst_dshowaudiosrc_read (GstAudioSrc * asrc, gpointer data, guint length)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
+  guint ret = 0;
+
+  if (!src->is_running)
+    return -1;
+
+  if (src->gbarray) {
+  test:
+    if (src->gbarray->len >= length) {
+      g_mutex_lock (src->gbarray_lock);
+      memcpy (data, src->gbarray->data + (src->gbarray->len - length), length);
+      g_byte_array_remove_range (src->gbarray, src->gbarray->len - length,
+          length);
+      ret = length;
+      g_mutex_unlock (src->gbarray_lock);
+    } else {
+      if (src->is_running) {
+        Sleep (100);
+        goto test;
+      }
+    }
+  }
+
+  return ret;
+}
+
+static guint
+gst_dshowaudiosrc_delay (GstAudioSrc * asrc)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
+  guint ret = 0;
+
+  if (src->gbarray) {
+    g_mutex_lock (src->gbarray_lock);
+    if (src->gbarray->len) {
+      ret = src->gbarray->len / 4;
+    }
+    g_mutex_unlock (src->gbarray_lock);
+  }
+
+  return ret;
+}
+
+static void
+gst_dshowaudiosrc_reset (GstAudioSrc * asrc)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
+
+  g_mutex_lock (src->gbarray_lock);
+  g_byte_array_remove_range (src->gbarray, 0, src->gbarray->len);
+  g_mutex_unlock (src->gbarray_lock);
+}
+
+static GstCaps *
+gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc * src, IPin * pin,
+    IAMStreamConfig * streamcaps)
+{
+  GstCaps *caps = NULL;
+  HRESULT hres = S_OK;
+  RPC_STATUS rpcstatus;
+  int icount = 0;
+  int isize = 0;
+  AUDIO_STREAM_CONFIG_CAPS ascc;
+  int i = 0;
+
+  if (!streamcaps)
+    return NULL;
+
+  streamcaps->GetNumberOfCapabilities(&icount, &isize);
+
+  if (isize != sizeof (ascc))
+    return NULL;
+
+  for (; i < icount; i++) {
+    GstCapturePinMediaType *pin_mediatype = g_new0 (GstCapturePinMediaType, 1);
+
+    pin->AddRef();
+    pin_mediatype->capture_pin = pin;
+
+    hres = streamcaps->GetStreamCaps(i, &pin_mediatype->mediatype,
+        (BYTE *) & ascc);
+    if (hres == S_OK && pin_mediatype->mediatype) {
+      GstCaps *mediacaps = NULL;
+
+      if (!caps)
+        caps = gst_caps_new_empty ();
+
+      if ((UuidCompare (&pin_mediatype->mediatype->subtype, (UUID *) &MEDIASUBTYPE_PCM,
+                  &rpcstatus) == 0 && rpcstatus == RPC_S_OK)
+          && (UuidCompare (&pin_mediatype->mediatype->formattype,
+                  (UUID *) &FORMAT_WaveFormatEx, &rpcstatus) == 0
+              && rpcstatus == RPC_S_OK)) {
+        WAVEFORMATEX *wavformat =
+            (WAVEFORMATEX *) pin_mediatype->mediatype->pbFormat;
+        mediacaps =
+            gst_caps_new_simple ("audio/x-raw-int", "width", G_TYPE_INT,
+            wavformat->wBitsPerSample, "depth", G_TYPE_INT,
+            wavformat->wBitsPerSample, "endianness", G_TYPE_INT, G_BYTE_ORDER,
+            "signed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT,
+            wavformat->nChannels, "rate", G_TYPE_INT, wavformat->nSamplesPerSec,
+            NULL);
+
+        if (mediacaps) {
+          src->pins_mediatypes =
+              g_list_append (src->pins_mediatypes, pin_mediatype);
+          gst_caps_append (caps, mediacaps);
+        } else {
+          gst_dshow_free_pin_mediatype (pin_mediatype);
+        }
+      } else {
+        gst_dshow_free_pin_mediatype (pin_mediatype);
+      }
+    } else {
+      gst_dshow_free_pin_mediatype (pin_mediatype);
+    }
+  }
+
+  if (caps && gst_caps_is_empty (caps)) {
+    gst_caps_unref (caps);
+    caps = NULL;
+  }
+
+  return caps;
+}
+
+static gboolean
+gst_dshowaudiosrc_push_buffer (byte * buffer, long size, byte * src_object,
+    UINT64 start, UINT64 stop)
+{
+  GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (src_object);
+
+  if (!buffer || size == 0 || !src) {
+    return FALSE;
+  }
+
+  g_mutex_lock (src->gbarray_lock);
+  g_byte_array_prepend (src->gbarray, (guint8 *) buffer, size);
+  g_mutex_unlock (src->gbarray_lock);
+
+  return TRUE;
+}