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-rw-r--r--sys/dshowsrcwrapper/gstdshowaudiosrc.c879
1 files changed, 879 insertions, 0 deletions
diff --git a/sys/dshowsrcwrapper/gstdshowaudiosrc.c b/sys/dshowsrcwrapper/gstdshowaudiosrc.c
new file mode 100644
index 00000000..87c12789
--- /dev/null
+++ b/sys/dshowsrcwrapper/gstdshowaudiosrc.c
@@ -0,0 +1,879 @@
+/* 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.
+ */
+
+#include "gstdshowaudiosrc.h"
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+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, 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, 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) {
+ IBaseFilter_Release (src->audio_cap_filter);
+ }
+
+ CoUninitialize ();
+}
+
+
+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, (void **) &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 =
+ ICreateDevEnum_CreateClassEnumerator (devices_enum,
+ &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;
+ }
+
+ IEnumMoniker_Reset (moniker_enum);
+
+ while (hres = IEnumMoniker_Next (moniker_enum, 1, &moniker, &fetched),
+ hres == S_OK) {
+ IPropertyBag *property_bag = NULL;
+
+ hres =
+ IMoniker_BindToStorage (moniker, NULL, NULL, &IID_IPropertyBag,
+ (void **) &property_bag);
+ if (SUCCEEDED (hres) && property_bag) {
+ VARIANT varFriendlyName;
+
+ VariantInit (&varFriendlyName);
+ hres =
+ IPropertyBag_Read (property_bag, 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);
+ }
+ IPropertyBag_Release (property_bag);
+ }
+ IMoniker_Release (moniker);
+ }
+
+clean:
+ if (moniker_enum) {
+ IEnumMoniker_Release (moniker_enum);
+ }
+
+ if (devices_enum) {
+ ICreateDevEnum_Release (devices_enum);
+ }
+
+ 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, unidevice, &dwEaten, &audiom);
+ if (SUCCEEDED (hres)) {
+ hres =
+ IMoniker_BindToObject (audiom, lpbc, NULL, &IID_IBaseFilter,
+ &src->audio_cap_filter);
+ IMoniker_Release (audiom);
+ }
+ IBindCtx_Release (lpbc);
+ }
+ }
+
+ if (src->audio_cap_filter && !src->caps) {
+ /* get the capture pins supported types */
+ IPin *capture_pin = NULL;
+ IEnumPins *enumpins = NULL;
+ HRESULT hres;
+
+ hres = IBaseFilter_EnumPins (src->audio_cap_filter, &enumpins);
+ if (SUCCEEDED (hres)) {
+ while (IEnumPins_Next (enumpins, 1, &capture_pin, NULL) == S_OK) {
+ IKsPropertySet *pKs = NULL;
+
+ hres =
+ IPin_QueryInterface (capture_pin, &IID_IKsPropertySet,
+ (void **) &pKs);
+ if (SUCCEEDED (hres) && pKs) {
+ DWORD cbReturned;
+ GUID pin_category;
+ RPC_STATUS rpcstatus;
+
+ hres =
+ IKsPropertySet_Get (pKs, &AMPROPSETID_Pin,
+ AMPROPERTY_PIN_CATEGORY, NULL, 0, &pin_category, sizeof (GUID),
+ &cbReturned);
+
+ /* we only want capture pins */
+ if (UuidCompare (&pin_category, &PIN_CATEGORY_CAPTURE,
+ &rpcstatus) == 0) {
+ IAMStreamConfig *streamcaps = NULL;
+
+ if (SUCCEEDED (IPin_QueryInterface (capture_pin,
+ &IID_IAMStreamConfig, (void **) &streamcaps))) {
+ src->caps =
+ gst_dshowaudiosrc_getcaps_from_streamcaps (src, capture_pin,
+ streamcaps);
+ IAMStreamConfig_Release (streamcaps);
+ }
+ }
+ IKsPropertySet_Release (pKs);
+ }
+ IPin_Release (capture_pin);
+ }
+ IEnumPins_Release (enumpins);
+ }
+ }
+
+ 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 = IMediaFilter_Run (src->media_filter, 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 = IMediaFilter_Stop (src->media_filter);
+ 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 = IFilterGraph_QueryInterface (src->filter_graph, &IID_IMediaFilter,
+ (void **) &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 =
+ IFilterGraph_AddFilter (src->filter_graph, 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 =
+ IFilterGraph_AddFilter (src->filter_graph, 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) {
+ IBaseFilter_Release (src->dshow_fakesink);
+ src->dshow_fakesink = NULL;
+ }
+
+ if (src->media_filter) {
+ IMediaFilter_Release (src->media_filter);
+ src->media_filter = NULL;
+ }
+ if (src->filter_graph) {
+ IFilterGraph_Release (src->filter_graph);
+ 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 =
+ IBaseFilter_QueryInterface (src->dshow_fakesink,
+ &IID_IGstDshowInterface, (void **) &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;
+ }
+
+ IGstDshowInterface_gst_set_media_type (srcinterface,
+ pin_mediatype->mediatype);
+ IGstDshowInterface_gst_set_buffer_callback (srcinterface,
+ gst_dshowaudiosrc_push_buffer, (byte *) src);
+
+ if (srcinterface) {
+ IGstDshowInterface_Release (srcinterface);
+ }
+
+ 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 =
+ IFilterGraph_ConnectDirect (src->filter_graph,
+ pin_mediatype->capture_pin, input_pin, NULL);
+ IPin_Release (input_pin);
+
+ 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) {
+ IGstDshowInterface_Release (srcinterface);
+ }
+
+ 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 = IFilterGraph_Disconnect (src->filter_graph, output_pin);
+ IPin_Release (output_pin);
+ }
+
+ gst_dshow_get_pin_from_filter (src->dshow_fakesink, PINDIR_INPUT, &input_pin);
+ if (input_pin) {
+ hres = IFilterGraph_Disconnect (src->filter_graph, input_pin);
+ IPin_Release (input_pin);
+ }
+
+ 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 */
+ IFilterGraph_RemoveFilter (src->filter_graph, src->audio_cap_filter);
+ IFilterGraph_RemoveFilter (src->filter_graph, src->dshow_fakesink);
+
+ /*release our gstreamer dshow sink */
+ IBaseFilter_Release (src->dshow_fakesink);
+ src->dshow_fakesink = NULL;
+
+ /*release media filter interface */
+ IMediaFilter_Release (src->media_filter);
+ src->media_filter = NULL;
+
+ /*release the filter graph manager */
+ IFilterGraph_Release (src->filter_graph);
+ 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;
+
+ IAMStreamConfig_GetNumberOfCapabilities (streamcaps, &icount, &isize);
+
+ if (isize != sizeof (ascc))
+ return NULL;
+
+ for (; i < icount; i++) {
+ GstCapturePinMediaType *pin_mediatype = g_new0 (GstCapturePinMediaType, 1);
+
+ IPin_AddRef (pin);
+ pin_mediatype->capture_pin = pin;
+
+ hres =
+ IAMStreamConfig_GetStreamCaps (streamcaps, 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, &MEDIASUBTYPE_PCM,
+ &rpcstatus) == 0 && rpcstatus == RPC_S_OK)
+ && (UuidCompare (&pin_mediatype->mediatype->formattype,
+ &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;
+}