sys/: New plugin for audio capture and playback using Windows Audio Session

Original commit message from CVS:
* sys/Makefile.am:
* sys/wasapi/Makefile.am:
* sys/wasapi/gstwasapi.c:
* sys/wasapi/gstwasapisink.c:
* sys/wasapi/gstwasapisink.h:
* sys/wasapi/gstwasapisrc.c:
* sys/wasapi/gstwasapisrc.h:
* sys/wasapi/gstwasapiutil.c:
* sys/wasapi/gstwasapiutil.h:
New plugin for audio capture and playback using Windows Audio Session
API (WASAPI) available with Vista and newer (#520901).
Comes with hardcoded caps and obviously needs lots of love. Haven't
had time to work on this code since it was written, was initially just
a quick experiment to play around with this new API.

8 files changed, 1203 insertions, 0 deletions

diff --git a/sys/wasapi/Makefile.am b/sys/wasapi/Makefile.am
new file mode 100644
index 00000000..90581783
--- /dev/null
+++ b/sys/wasapi/Makefile.am
@@ -0,0 +1,5 @@
+EXTRA_DIST = \
+	gstwasapi.c \
+	gstwasapisrc.c gstwasapisrc.h \
+	gstwasapisink.c gstwasapisink.h \
+	gstwasapiutil.c gstwasapiutil.h
diff --git a/sys/wasapi/gstwasapi.c b/sys/wasapi/gstwasapi.c
new file mode 100644
index 00000000..9fef21fe
--- /dev/null
+++ b/sys/wasapi/gstwasapi.c
@@ -0,0 +1,45 @@
+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * 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 "gstwasapisrc.h"
+#include "gstwasapisink.h"
+
+#ifdef HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  gboolean ret;
+
+  ret = gst_element_register (plugin, "wasapisrc",
+      GST_RANK_NONE, GST_TYPE_WASAPI_SRC);
+  if (!ret)
+    return ret;
+
+  return gst_element_register (plugin, "wasapisink",
+      GST_RANK_NONE, GST_TYPE_WASAPI_SINK);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "wasapi",
+    "Windows audio session API plugin",
+    plugin_init, VERSION, "LGPL", "GStreamer", "http://gstreamer.net/")
diff --git a/sys/wasapi/gstwasapisink.c b/sys/wasapi/gstwasapisink.c
new file mode 100644
index 00000000..f26e56d6
--- /dev/null
+++ b/sys/wasapi/gstwasapisink.c
@@ -0,0 +1,267 @@
+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * 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.
+ */
+
+/**
+ * SECTION:element-wasapisink
+ *
+ * Provides audio playback using the Windows Audio Session API available with
+ * Vista and newer.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch-0.10 -v audiotestsrc samplesperbuffer=160 ! wasapisink
+ * ]| Generate 20 ms buffers and render to the default audio device.
+ * </refsect2>
+ */
+
+#include "gstwasapisink.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
+#define GST_CAT_DEFAULT gst_wasapi_sink_debug
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "rate = (int) 8000, "
+        "channels = (int) 1, "
+        "signed = (boolean) TRUE, "
+        "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
+
+static void gst_wasapi_sink_dispose (GObject * object);
+static void gst_wasapi_sink_finalize (GObject * object);
+
+static void gst_wasapi_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
+    GstClockTime * start, GstClockTime * end);
+static gboolean gst_wasapi_sink_start (GstBaseSink * sink);
+static gboolean gst_wasapi_sink_stop (GstBaseSink * sink);
+static GstFlowReturn gst_wasapi_sink_render (GstBaseSink * sink,
+    GstBuffer * buffer);
+
+GST_BOILERPLATE (GstWasapiSink, gst_wasapi_sink, GstBaseSink,
+    GST_TYPE_BASE_SINK);
+
+static void
+gst_wasapi_sink_base_init (gpointer gclass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
+  static GstElementDetails element_details = {
+    "WasapiSrc",
+    "Sink/Audio",
+    "Stream audio to an audio capture device through WASAPI",
+    "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"
+  };
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+  gst_element_class_set_details (element_class, &element_details);
+}
+
+static void
+gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
+
+  gobject_class->dispose = gst_wasapi_sink_dispose;
+  gobject_class->finalize = gst_wasapi_sink_finalize;
+
+  gstbasesink_class->get_times = gst_wasapi_sink_get_times;
+  gstbasesink_class->start = gst_wasapi_sink_start;
+  gstbasesink_class->stop = gst_wasapi_sink_stop;
+  gstbasesink_class->render = gst_wasapi_sink_render;
+
+  GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
+      0, "Windows audio session API sink");
+}
+
+static void
+gst_wasapi_sink_init (GstWasapiSink * self, GstWasapiSinkClass * gclass)
+{
+  self->rate = 8000;
+  self->buffer_time = 20 * GST_MSECOND;
+  self->period_time = 20 * GST_MSECOND;
+  self->latency = GST_CLOCK_TIME_NONE;
+
+  self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
+
+  CoInitialize (NULL);
+}
+
+static void
+gst_wasapi_sink_dispose (GObject * object)
+{
+  GstWasapiSink *self = GST_WASAPI_SINK (object);
+
+  if (self->event_handle != NULL) {
+    CloseHandle (self->event_handle);
+    self->event_handle = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_wasapi_sink_finalize (GObject * object)
+{
+  GstWasapiSink *self = GST_WASAPI_SINK (object);
+
+  CoUninitialize ();
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_wasapi_sink_get_times (GstBaseSink * sink,
+    GstBuffer * buffer, GstClockTime * start, GstClockTime * end)
+{
+  GstWasapiSink *self = GST_WASAPI_SINK (sink);
+
+  if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
+    *start = GST_BUFFER_TIMESTAMP (buffer);
+
+    if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
+      *end = *start + GST_BUFFER_DURATION (buffer);
+    } else {
+      *end = *start + self->buffer_time;
+    }
+
+    *start += self->latency;
+    *end += self->latency;
+  }
+}
+
+static gboolean
+gst_wasapi_sink_start (GstBaseSink * sink)
+{
+  GstWasapiSink *self = GST_WASAPI_SINK (sink);
+  gboolean res = FALSE;
+  IAudioClient *client = NULL;
+  HRESULT hr;
+  IAudioRenderClient *render_client = NULL;
+
+  if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
+          FALSE, self->rate, self->buffer_time, self->period_time,
+          AUDCLNT_STREAMFLAGS_EVENTCALLBACK, &client, &self->latency))
+    goto beach;
+
+  hr = IAudioClient_SetEventHandle (client, self->event_handle);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
+    goto beach;
+  }
+
+  hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
+      &render_client);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioClient::GetService "
+        "(IID_IAudioRenderClient) failed");
+    goto beach;
+  }
+
+  hr = IAudioClient_Start (client);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
+    goto beach;
+  }
+
+  self->client = client;
+  self->render_client = render_client;
+
+  res = TRUE;
+
+beach:
+  if (!res) {
+    if (render_client != NULL)
+      IUnknown_Release (render_client);
+
+    if (client != NULL)
+      IUnknown_Release (client);
+  }
+
+  return res;
+}
+
+static gboolean
+gst_wasapi_sink_stop (GstBaseSink * sink)
+{
+  GstWasapiSink *self = GST_WASAPI_SINK (sink);
+
+  if (self->client != NULL) {
+    IAudioClient_Stop (self->client);
+  }
+
+  if (self->render_client != NULL) {
+    IUnknown_Release (self->render_client);
+    self->render_client = NULL;
+  }
+
+  if (self->client != NULL) {
+    IUnknown_Release (self->client);
+    self->client = NULL;
+  }
+
+  return TRUE;
+}
+
+static GstFlowReturn
+gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
+{
+  GstWasapiSink *self = GST_WASAPI_SINK (sink);
+  GstFlowReturn ret = GST_FLOW_OK;
+  HRESULT hr;
+  gint16 *src = (gint16 *) GST_BUFFER_DATA (buffer);
+  gint16 *dst = NULL;
+  guint nsamples = GST_BUFFER_SIZE (buffer) / sizeof (gint16);
+  guint i;
+
+  WaitForSingleObject (self->event_handle, INFINITE);
+
+  hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples,
+      (BYTE **) & dst);
+  if (hr != S_OK) {
+    GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
+        ("IAudioRenderClient::GetBuffer () failed: %s",
+            gst_wasapi_util_hresult_to_string (hr)));
+    ret = GST_FLOW_ERROR;
+    goto beach;
+  }
+
+  for (i = 0; i < nsamples; i++) {
+    dst[0] = *src;
+    dst[1] = *src;
+
+    src++;
+    dst += 2;
+  }
+
+  hr = IAudioRenderClient_ReleaseBuffer (self->render_client, nsamples, 0);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioRenderClient::ReleaseBuffer () failed: %s",
+        gst_wasapi_util_hresult_to_string (hr));
+    ret = GST_FLOW_ERROR;
+    goto beach;
+  }
+
+beach:
+  return ret;
+}
diff --git a/sys/wasapi/gstwasapisink.h b/sys/wasapi/gstwasapisink.h
new file mode 100644
index 00000000..16cc22f1
--- /dev/null
+++ b/sys/wasapi/gstwasapisink.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * 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.
+ */
+
+#ifndef __GST_WASAPI_SINK_H__
+#define __GST_WASAPI_SINK_H__
+
+#include "gstwasapiutil.h"
+
+#include <gst/base/gstbasesink.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_WASAPI_SINK \
+  (gst_wasapi_sink_get_type ())
+#define GST_WASAPI_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_WASAPI_SINK, GstWasapiSink))
+#define GST_WASAPI_SINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_WASAPI_SINK, GstWasapiSinkClass))
+#define GST_IS_WASAPI_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_WASAPI_SINK))
+#define GST_IS_WASAPI_SINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_WASAPI_SINK))
+
+typedef struct _GstWasapiSink      GstWasapiSink;
+typedef struct _GstWasapiSinkClass GstWasapiSinkClass;
+
+struct _GstWasapiSink
+{
+  GstBaseSink base_sink;
+
+  guint rate;
+  GstClockTime buffer_time;
+  GstClockTime period_time;
+  GstClockTime latency;
+
+  IAudioClient * client;
+  IAudioRenderClient * render_client;
+  HANDLE event_handle;
+};
+
+struct _GstWasapiSinkClass
+{
+  GstBaseSinkClass parent_class;
+};
+
+GType gst_wasapi_sink_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_WASAPI_SINK_H__ */
+
diff --git a/sys/wasapi/gstwasapisrc.c b/sys/wasapi/gstwasapisrc.c
new file mode 100644
index 00000000..0a0edd2b
--- /dev/null
+++ b/sys/wasapi/gstwasapisrc.c
@@ -0,0 +1,443 @@
+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * 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.
+ */
+
+/**
+ * SECTION:element-wasapisrc
+ *
+ * Provides audio capture from the Windows Audio Session API available with
+ * Vista and newer.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch-0.10 -v wasapisrc ! fakesink
+ * ]| Capture from the default audio device and render to fakesink.
+ * </refsect2>
+ */
+
+#include "gstwasapisrc.h"
+#include <gst/audio/gstaudioclock.h>
+
+GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
+#define GST_CAT_DEFAULT gst_wasapi_src_debug
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw-int, "
+        "width = (int) 16, "
+        "depth = (int) 16, "
+        "rate = (int) 8000, "
+        "channels = (int) 1, "
+        "signed = (boolean) TRUE, "
+        "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
+
+static void gst_wasapi_src_dispose (GObject * object);
+static void gst_wasapi_src_finalize (GObject * object);
+
+static GstClock *gst_wasapi_src_provide_clock (GstElement * element);
+
+static gboolean gst_wasapi_src_start (GstBaseSrc * src);
+static gboolean gst_wasapi_src_stop (GstBaseSrc * src);
+static gboolean gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query);
+
+static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
+
+static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
+    gpointer user_data);
+
+GST_BOILERPLATE (GstWasapiSrc, gst_wasapi_src, GstPushSrc, GST_TYPE_PUSH_SRC);
+
+static void
+gst_wasapi_src_base_init (gpointer gclass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
+  static GstElementDetails element_details = {
+    "WasapiSrc",
+    "Source/Audio",
+    "Stream audio from an audio capture device through WASAPI",
+    "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"
+  };
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+  gst_element_class_set_details (element_class, &element_details);
+}
+
+static void
+gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+  GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
+  GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
+
+  gobject_class->dispose = gst_wasapi_src_dispose;
+  gobject_class->finalize = gst_wasapi_src_finalize;
+
+  gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
+
+  gstbasesrc_class->start = gst_wasapi_src_start;
+  gstbasesrc_class->stop = gst_wasapi_src_stop;
+  gstbasesrc_class->query = gst_wasapi_src_query;
+
+  gstpushsrc_class->create = gst_wasapi_src_create;
+
+  GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
+      0, "Windows audio session API source");
+}
+
+static void
+gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
+{
+  GstBaseSrc *basesrc = GST_BASE_SRC (self);
+
+  gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
+  gst_base_src_set_live (basesrc, TRUE);
+
+  self->rate = 8000;
+  self->buffer_time = 20 * GST_MSECOND;
+  self->period_time = 20 * GST_MSECOND;
+  self->latency = GST_CLOCK_TIME_NONE;
+  self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
+
+  self->start_time = GST_CLOCK_TIME_NONE;
+  self->next_time = GST_CLOCK_TIME_NONE;
+
+  self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
+      gst_wasapi_src_get_time, self);
+
+  CoInitialize (NULL);
+}
+
+static void
+gst_wasapi_src_dispose (GObject * object)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (object);
+
+  if (self->clock != NULL) {
+    gst_object_unref (self->clock);
+    self->clock = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_wasapi_src_finalize (GObject * object)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (object);
+
+  CoUninitialize ();
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static GstClock *
+gst_wasapi_src_provide_clock (GstElement * element)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (element);
+  GstClock *clock;
+
+  GST_OBJECT_LOCK (self);
+
+  if (self->client_clock == NULL)
+    goto wrong_state;
+
+  clock = GST_CLOCK (gst_object_ref (self->clock));
+
+  GST_OBJECT_UNLOCK (self);
+  return clock;
+
+  /* ERRORS */
+wrong_state:
+  {
+    GST_OBJECT_UNLOCK (self);
+    GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
+    return NULL;
+  }
+}
+
+static gboolean
+gst_wasapi_src_start (GstBaseSrc * src)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (src);
+  gboolean res = FALSE;
+  IAudioClient *client = NULL;
+  IAudioClock *client_clock = NULL;
+  guint64 client_clock_freq = 0;
+  IAudioCaptureClient *capture_client = NULL;
+  HRESULT hr;
+
+  if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
+          TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
+          &self->latency))
+    goto beach;
+
+  hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
+        "failed");
+    goto beach;
+  }
+
+  hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed");
+    goto beach;
+  }
+
+  hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
+      &capture_client);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioClient::GetService "
+        "(IID_IAudioCaptureClient) failed");
+    goto beach;
+  }
+
+  hr = IAudioClient_Start (client);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
+    goto beach;
+  }
+
+  self->client = client;
+  self->client_clock = client_clock;
+  self->client_clock_freq = client_clock_freq;
+  self->capture_client = capture_client;
+
+  res = TRUE;
+
+beach:
+  if (!res) {
+    if (capture_client != NULL)
+      IUnknown_Release (capture_client);
+
+    if (client_clock != NULL)
+      IUnknown_Release (client_clock);
+
+    if (client != NULL)
+      IUnknown_Release (client);
+  }
+
+  return res;
+}
+
+static gboolean
+gst_wasapi_src_stop (GstBaseSrc * src)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (src);
+
+  if (self->client != NULL) {
+    IAudioClient_Stop (self->client);
+  }
+
+  if (self->capture_client != NULL) {
+    IUnknown_Release (self->capture_client);
+    self->capture_client = NULL;
+  }
+
+  if (self->client_clock != NULL) {
+    IUnknown_Release (self->client_clock);
+    self->client_clock = NULL;
+  }
+
+  if (self->client != NULL) {
+    IUnknown_Release (self->client);
+    self->client = NULL;
+  }
+
+  return TRUE;
+}
+
+static gboolean
+gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (src);
+  gboolean ret = FALSE;
+
+  GST_DEBUG_OBJECT (self, "query for %s",
+      gst_query_type_get_name (GST_QUERY_TYPE (query)));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_LATENCY:{
+      GstClockTime min_latency, max_latency;
+
+      min_latency = self->latency + self->period_time;
+      max_latency = min_latency;
+
+      GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
+          " max %" GST_TIME_FORMAT,
+          GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
+
+      gst_query_set_latency (query, TRUE, min_latency, max_latency);
+      ret = TRUE;
+      break;
+    }
+
+    default:
+      ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
+      break;
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (src);
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstClock *clock;
+  GstClockTime timestamp, duration = self->period_time;
+  HRESULT hr;
+  gint16 *samples = NULL;
+  guint32 nsamples_read = 0, nsamples;
+  DWORD flags = 0;
+  guint64 devpos;
+
+  GST_OBJECT_LOCK (self);
+  clock = GST_ELEMENT_CLOCK (self);
+  if (clock != NULL)
+    gst_object_ref (clock);
+  GST_OBJECT_UNLOCK (self);
+
+  if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
+    GstClockID id;
+
+    id = gst_clock_new_single_shot_id (clock, self->next_time);
+    gst_clock_id_wait (id, NULL);
+    gst_clock_id_unref (id);
+  }
+
+  do {
+    hr = IAudioCaptureClient_GetBuffer (self->capture_client,
+        (BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
+  }
+  while (hr == AUDCLNT_S_BUFFER_EMPTY);
+
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
+        gst_wasapi_util_hresult_to_string (hr));
+    ret = GST_FLOW_ERROR;
+    goto beach;
+  }
+
+  if (flags != 0) {
+    GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
+        devpos, flags);
+  }
+
+  /* FIXME: Why do we get 1024 sometimes and not a multiple of
+   *        samples_per_buffer? Shouldn't WASAPI provide a DISCONT
+   *        flag if we read too slow?
+   */
+  nsamples = nsamples_read;
+  g_assert (nsamples >= self->samples_per_buffer);
+  if (nsamples > self->samples_per_buffer) {
+    GST_WARNING_OBJECT (self,
+        "devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
+        devpos, nsamples, self->samples_per_buffer);
+
+    nsamples = self->samples_per_buffer;
+  }
+
+  if (clock == NULL || clock == self->clock) {
+    timestamp =
+        gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
+  } else {
+    GstClockTime base_time;
+
+    timestamp = gst_clock_get_time (clock);
+
+    base_time = GST_ELEMENT_CAST (self)->base_time;
+    if (timestamp > base_time)
+      timestamp -= base_time;
+    else
+      timestamp = 0;
+
+    if (timestamp > duration)
+      timestamp -= duration;
+    else
+      timestamp = 0;
+  }
+
+  ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
+      devpos,
+      nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
+
+  if (ret == GST_FLOW_OK) {
+    guint i;
+    gint16 *dst;
+
+    GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
+    GST_BUFFER_TIMESTAMP (*buf) = timestamp;
+    GST_BUFFER_DURATION (*buf) = duration;
+
+    dst = (gint16 *) GST_BUFFER_DATA (*buf);
+    for (i = 0; i < nsamples; i++) {
+      *dst = *samples;
+
+      samples += 2;
+      dst++;
+    }
+  }
+
+  hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
+        gst_wasapi_util_hresult_to_string (hr));
+    ret = GST_FLOW_ERROR;
+    goto beach;
+  }
+
+beach:
+  if (clock != NULL)
+    gst_object_unref (clock);
+
+  return ret;
+}
+
+static GstClockTime
+gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
+{
+  GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
+  HRESULT hr;
+  guint64 devpos;
+  GstClockTime result;
+
+  if (G_UNLIKELY (self->client_clock == NULL))
+    return GST_CLOCK_TIME_NONE;
+
+  hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
+  if (G_UNLIKELY (hr != S_OK))
+    return GST_CLOCK_TIME_NONE;
+
+  result = gst_util_uint64_scale_int (devpos, GST_SECOND,
+      self->client_clock_freq);
+
+  /*
+     GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
+     " frequency = %" G_GUINT64_FORMAT
+     " result = %" G_GUINT64_FORMAT " ms",
+     devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));
+   */
+
+  return result;
+}
diff --git a/sys/wasapi/gstwasapisrc.h b/sys/wasapi/gstwasapisrc.h
new file mode 100644
index 00000000..26d14fbe
--- /dev/null
+++ b/sys/wasapi/gstwasapisrc.h
@@ -0,0 +1,74 @@
+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * 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.
+ */
+
+#ifndef __GST_WASAPI_SRC_H__
+#define __GST_WASAPI_SRC_H__
+
+#include "gstwasapiutil.h"
+
+#include <gst/base/gstpushsrc.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_WASAPI_SRC \
+  (gst_wasapi_src_get_type ())
+#define GST_WASAPI_SRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_WASAPI_SRC, GstWasapiSrc))
+#define GST_WASAPI_SRC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_WASAPI_SRC, GstWasapiSrcClass))
+#define GST_IS_WASAPI_SRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_WASAPI_SRC))
+#define GST_IS_WASAPI_SRC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_WASAPI_SRC))
+
+typedef struct _GstWasapiSrc      GstWasapiSrc;
+typedef struct _GstWasapiSrcClass GstWasapiSrcClass;
+
+struct _GstWasapiSrc
+{
+  GstPushSrc audio_src;
+
+  GstClock * clock;
+
+  guint rate;
+  GstClockTime buffer_time;
+  GstClockTime period_time;
+  GstClockTime latency;
+  guint samples_per_buffer;
+
+  IAudioClient * client;
+  IAudioClock * client_clock;
+  guint64 client_clock_freq;
+  IAudioCaptureClient * capture_client;
+
+  GstClockTime start_time;
+  GstClockTime next_time;
+};
+
+struct _GstWasapiSrcClass
+{
+  GstPushSrcClass parent_class;
+};
+
+GType gst_wasapi_src_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_WASAPI_SRC_H__ */
+
diff --git a/sys/wasapi/gstwasapiutil.c b/sys/wasapi/gstwasapiutil.c
new file mode 100644
index 00000000..79c92495
--- /dev/null
+++ b/sys/wasapi/gstwasapiutil.c
@@ -0,0 +1,261 @@
+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * 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 "gstwasapiutil.h"
+
+#include <mmdeviceapi.h>
+
+/* These seem to be missing in the Windows SDK... */
+const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c,
+  {0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e}
+};
+const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35,
+  {0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6}
+};
+const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,
+  {0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2}
+};
+const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b,
+  {0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7}
+};
+const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0,
+  {0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17}
+};
+const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483,
+  {0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}
+};
+
+const gchar *
+gst_wasapi_util_hresult_to_string (HRESULT hr)
+{
+  const gchar *s = "AUDCLNT_E_UNKNOWN";
+
+  switch (hr) {
+    case AUDCLNT_E_NOT_INITIALIZED:
+      s = "AUDCLNT_E_NOT_INITIALIZED";
+      break;
+    case AUDCLNT_E_ALREADY_INITIALIZED:
+      s = "AUDCLNT_E_ALREADY_INITIALIZED";
+      break;
+    case AUDCLNT_E_WRONG_ENDPOINT_TYPE:
+      s = "AUDCLNT_E_WRONG_ENDPOINT_TYPE";
+      break;
+    case AUDCLNT_E_DEVICE_INVALIDATED:
+      s = "AUDCLNT_E_DEVICE_INVALIDATED";
+      break;
+    case AUDCLNT_E_NOT_STOPPED:
+      s = "AUDCLNT_E_NOT_STOPPED";
+      break;
+    case AUDCLNT_E_BUFFER_TOO_LARGE:
+      s = "AUDCLNT_E_BUFFER_TOO_LARGE";
+      break;
+    case AUDCLNT_E_OUT_OF_ORDER:
+      s = "AUDCLNT_E_OUT_OF_ORDER";
+      break;
+    case AUDCLNT_E_UNSUPPORTED_FORMAT:
+      s = "AUDCLNT_E_UNSUPPORTED_FORMAT";
+      break;
+    case AUDCLNT_E_INVALID_SIZE:
+      s = "AUDCLNT_E_INVALID_SIZE";
+      break;
+    case AUDCLNT_E_DEVICE_IN_USE:
+      s = "AUDCLNT_E_DEVICE_IN_USE";
+      break;
+    case AUDCLNT_E_BUFFER_OPERATION_PENDING:
+      s = "AUDCLNT_E_BUFFER_OPERATION_PENDING";
+      break;
+    case AUDCLNT_E_THREAD_NOT_REGISTERED:
+      s = "AUDCLNT_E_THREAD_NOT_REGISTERED";
+      break;
+    case AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED:
+      s = "AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED";
+      break;
+    case AUDCLNT_E_ENDPOINT_CREATE_FAILED:
+      s = "AUDCLNT_E_ENDPOINT_CREATE_FAILED";
+      break;
+    case AUDCLNT_E_SERVICE_NOT_RUNNING:
+      s = "AUDCLNT_E_SERVICE_NOT_RUNNING";
+      break;
+    case AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED:
+      s = "AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED";
+      break;
+    case AUDCLNT_E_EXCLUSIVE_MODE_ONLY:
+      s = "AUDCLNT_E_EXCLUSIVE_MODE_ONLY";
+      break;
+    case AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL:
+      s = "AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL";
+      break;
+    case AUDCLNT_E_EVENTHANDLE_NOT_SET:
+      s = "AUDCLNT_E_EVENTHANDLE_NOT_SET";
+      break;
+    case AUDCLNT_E_INCORRECT_BUFFER_SIZE:
+      s = "AUDCLNT_E_INCORRECT_BUFFER_SIZE";
+      break;
+    case AUDCLNT_E_BUFFER_SIZE_ERROR:
+      s = "AUDCLNT_E_BUFFER_SIZE_ERROR";
+      break;
+    case AUDCLNT_E_CPUUSAGE_EXCEEDED:
+      s = "AUDCLNT_E_CPUUSAGE_EXCEEDED";
+      break;
+    case AUDCLNT_S_BUFFER_EMPTY:
+      s = "AUDCLNT_S_BUFFER_EMPTY";
+      break;
+    case AUDCLNT_S_THREAD_ALREADY_REGISTERED:
+      s = "AUDCLNT_S_THREAD_ALREADY_REGISTERED";
+      break;
+    case AUDCLNT_S_POSITION_STALLED:
+      s = "AUDCLNT_S_POSITION_STALLED";
+      break;
+  }
+
+  return s;
+}
+
+gboolean
+gst_wasapi_util_get_default_device_client (GstElement * element,
+    gboolean capture,
+    guint rate,
+    GstClockTime buffer_time,
+    GstClockTime period_time,
+    DWORD flags, IAudioClient ** ret_client, GstClockTime * ret_latency)
+{
+  gboolean res = FALSE;
+  HRESULT hr;
+  IMMDeviceEnumerator *enumerator = NULL;
+  IMMDevice *device = NULL;
+  IAudioClient *client = NULL;
+  REFERENCE_TIME latency_rt, def_period, min_period;
+  WAVEFORMATEXTENSIBLE format;
+
+  hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
+      &IID_IMMDeviceEnumerator, &enumerator);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (element, "CoCreateInstance (MMDeviceEnumerator) failed");
+    goto beach;
+  }
+
+  hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint (enumerator,
+      (capture) ? eCapture : eRender, eCommunications, &device);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (element,
+        "IMMDeviceEnumerator::GetDefaultAudioEndpoint () failed");
+    goto beach;
+  }
+
+  hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL,
+      &client);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (element, "IMMDevice::Activate (IID_IAudioClient) failed");
+    goto beach;
+  }
+
+  hr = IAudioClient_GetDevicePeriod (client, &def_period, &min_period);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (element, "IAudioClient::GetDevicePeriod () failed");
+    goto beach;
+  }
+
+  ZeroMemory (&format, sizeof (format));
+  format.Format.cbSize = sizeof (format) - sizeof (format.Format);
+  format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
+  format.Format.nChannels = 2;
+  format.Format.nSamplesPerSec = rate;
+  format.Format.wBitsPerSample = 16;
+  format.Format.nBlockAlign = format.Format.nChannels
+      * (format.Format.wBitsPerSample / 8);
+  format.Format.nAvgBytesPerSec = format.Format.nSamplesPerSec
+      * format.Format.nBlockAlign;
+  format.Samples.wValidBitsPerSample = 16;
+  format.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
+  format.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+
+  hr = IAudioClient_Initialize (client, AUDCLNT_SHAREMODE_EXCLUSIVE,    /* or AUDCLNT_SHAREMODE_SHARED */
+      flags, buffer_time / 100, /* buffer duration in 100s of ns */
+      period_time / 100,        /* periodicity in 100s of ns */
+      (WAVEFORMATEX *) & format, NULL);
+  if (hr != S_OK) {
+    GST_ELEMENT_ERROR (element, RESOURCE, OPEN_READ, (NULL),
+        ("IAudioClient::Initialize () failed: %s",
+            gst_wasapi_util_hresult_to_string (hr)));
+    goto beach;
+  }
+
+  hr = IAudioClient_GetStreamLatency (client, &latency_rt);
+  if (hr != S_OK) {
+    GST_ERROR_OBJECT (element, "IAudioClient::GetStreamLatency () failed");
+    goto beach;
+  }
+
+  GST_INFO_OBJECT (element, "default period: %d (%d ms), "
+      "minimum period: %d (%d ms), "
+      "latency: %d (%d ms)",
+      (guint32) def_period, (guint32) def_period / 10000,
+      (guint32) min_period, (guint32) min_period / 10000,
+      (guint32) latency_rt, (guint32) latency_rt / 10000);
+
+  IUnknown_AddRef (client);
+  *ret_client = client;
+
+  *ret_latency = latency_rt * 100;
+
+  res = TRUE;
+
+beach:
+  if (client != NULL)
+    IUnknown_Release (client);
+
+  if (device != NULL)
+    IUnknown_Release (device);
+
+  if (enumerator != NULL)
+    IUnknown_Release (enumerator);
+
+  return res;
+}
+
+#if 0
+static WAVEFORMATEXTENSIBLE *
+gst_wasapi_src_probe_device_format (GstWasapiSrc * self, IMMDevice * device)
+{
+  HRESULT hr;
+  IPropertyStore *props = NULL;
+  PROPVARIANT format_prop;
+  WAVEFORMATEXTENSIBLE *format = NULL;
+
+  hr = IMMDevice_OpenPropertyStore (device, STGM_READ, &props);
+  if (hr != S_OK)
+    goto beach;
+
+  PropVariantInit (&format_prop);
+  hr = IPropertyStore_GetValue (props, &PKEY_AudioEngine_DeviceFormat,
+      &format_prop);
+  if (hr != S_OK)
+    goto beach;
+
+  format = (WAVEFORMATEXTENSIBLE *) format_prop.blob.pBlobData;
+
+  /* hmm: HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\MMDevices\Audio\Capture\{64adb8b7-9716-4c02-8929-96e53f5642da}\Properties */
+
+beach:
+  if (props != NULL)
+    IUnknown_Release (props);
+
+  return format;
+}
+#endif
diff --git a/sys/wasapi/gstwasapiutil.h b/sys/wasapi/gstwasapiutil.h
new file mode 100644
index 00000000..fc060abf
--- /dev/null
+++ b/sys/wasapi/gstwasapiutil.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * 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.
+ */
+
+#ifndef __GST_WASAPI_UTIL_H__
+#define __GST_WASAPI_UTIL_H__
+
+#include <gst/gst.h>
+
+#include <audioclient.h>
+
+const gchar *
+gst_wasapi_util_hresult_to_string (HRESULT hr);
+
+gboolean
+gst_wasapi_util_get_default_device_client (GstElement * element,
+                                           gboolean capture,
+                                           guint rate,
+                                           GstClockTime buffer_time,
+                                           GstClockTime period_time,
+                                           DWORD flags,
+                                           IAudioClient ** ret_client,
+                                           GstClockTime * ret_latency);
+
+#endif /* __GST_WASAPI_UTIL_H__ */
+