Index: jack/Makefile.am
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-bad/ext/jack/Makefile.am,v
retrieving revision 1.8
diff -p -u -w -r1.8 Makefile.am
--- jack/Makefile.am	8 Mar 2007 15:24:52 -0000	1.8
+++ jack/Makefile.am	4 Aug 2008 21:38:38 -0000
@@ -1,11 +1,11 @@
 
 plugin_LTLIBRARIES = libgstjack.la
 
-libgstjack_la_SOURCES = gstjack.c gstjackaudiosink.c gstjackaudioclient.c
+libgstjack_la_SOURCES = gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c
 libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS)
 libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(JACK_LIBS)
 libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 
-noinst_HEADERS = gstjackaudiosink.h gstjackaudioclient.h
+noinst_HEADERS = gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h
 
 EXTRA_DIST = README
Index: jack/gstjack.c
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-bad/ext/jack/gstjack.c,v
retrieving revision 1.46
diff -p -u -w -r1.46 gstjack.c
--- jack/gstjack.c	13 Mar 2008 14:25:19 -0000	1.46
+++ jack/gstjack.c	4 Aug 2008 21:38:38 -0000
@@ -21,11 +21,34 @@
 #include "config.h"
 #endif
 
+#include "gstjackaudiosrc.h"
 #include "gstjackaudiosink.h"
 
+GType
+gst_jack_connect_get_type()
+{
+  static GType jack_connect_type = 0;
+  static const GEnumValue jack_connect[] = {
+    {GST_JACK_CONNECT_NONE,
+      "Don't automatically connect ports to physical ports", "none"},
+    {GST_JACK_CONNECT_AUTO,
+      "Automatically connect ports to physical ports", "auto"},
+    {0, NULL, NULL},
+  };
+
+  if (!jack_connect_type) {
+    jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect);
+  }
+  return jack_connect_type;
+}
+
+
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
+  if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY,
+        GST_TYPE_JACK_AUDIO_SRC))
+    return FALSE;
   if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY,
           GST_TYPE_JACK_AUDIO_SINK))
     return FALSE;
@@ -33,8 +56,18 @@ plugin_init (GstPlugin * plugin)
   return TRUE;
 }
 
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+/* gstreamer looks for this structure to register jackaudiosrcs
+ *
+ * exchange the string 'Template jackaudiosrc' with your jackaudiosrc description
+ */
+GST_PLUGIN_DEFINE (
+    GST_VERSION_MAJOR,
     GST_VERSION_MINOR,
     "jack",
     "Jack elements",
-    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
+    plugin_init,
+    VERSION,
+    "LGPL",
+    "GStreamer",
+    "http://gstreamer.net/"
+    )
Index: jack/gstjack.h
===================================================================
RCS file: jack/gstjack.h
diff -N jack/gstjack.h
--- /dev/null	1 Jan 1970 00:00:00 -0000
+++ jack/gstjack.h	4 Aug 2008 21:38:38 -0000
@@ -0,0 +1,48 @@
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjacksink.h:
+ *
+ * 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_JACK_H_
+#define _GST_JACK_H_
+
+
+/**
+ * GstJackConnect:
+ * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports.
+ *     In this mode, the element will accept any number of input channels and will
+ *     create (but not connect) an output port for each channel.
+ * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each
+ *     output port to a random physical jack input pin. The sink will
+ *     expose the number of physical channels on its pad caps.
+ *
+ * Specify how the output ports will be connected.
+ */
+
+typedef enum {
+  GST_JACK_CONNECT_NONE,
+  GST_JACK_CONNECT_AUTO
+} GstJackConnect;
+
+typedef jack_default_audio_sample_t sample_t;
+
+#define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
+GType gst_jack_connect_get_type();
+
+#endif  // _GST_JACK_H_
Index: jack/gstjackaudiosink.c
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-bad/ext/jack/gstjackaudiosink.c,v
retrieving revision 1.10
diff -p -u -w -r1.10 gstjackaudiosink.c
--- jack/gstjackaudiosink.c	13 Jun 2008 11:59:19 -0000	1.10
+++ jack/gstjackaudiosink.c	4 Aug 2008 21:38:38 -0000
@@ -59,62 +59,11 @@
 #include <string.h>
 
 #include "gstjackaudiosink.h"
+#include "gstjackringbuffer.h"
 
 GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
 #define GST_CAT_DEFAULT gst_jack_audio_sink_debug
 
-typedef jack_default_audio_sample_t sample_t;
-
-#define GST_TYPE_JACK_RING_BUFFER        \
-        (gst_jack_ring_buffer_get_type())
-#define GST_JACK_RING_BUFFER(obj)        \
-        (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer))
-#define GST_JACK_RING_BUFFER_CLASS(klass) \
-        (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
-#define GST_JACK_RING_BUFFER_GET_CLASS(obj) \
-        (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER, GstJackRingBufferClass))
-#define GST_JACK_RING_BUFFER_CAST(obj)        \
-        ((GstJackRingBuffer *)obj)
-#define GST_IS_JACK_RING_BUFFER(obj)     \
-        (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER))
-#define GST_IS_JACK_RING_BUFFER_CLASS(klass)\
-        (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER))
-
-typedef struct _GstJackRingBuffer GstJackRingBuffer;
-typedef struct _GstJackRingBufferClass GstJackRingBufferClass;
-
-struct _GstJackRingBuffer
-{
-  GstRingBuffer object;
-
-  gint sample_rate;
-  gint buffer_size;
-  gint channels;
-};
-
-struct _GstJackRingBufferClass
-{
-  GstRingBufferClass parent_class;
-};
-
-static void gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass);
-static void gst_jack_ring_buffer_init (GstJackRingBuffer * ringbuffer,
-    GstJackRingBufferClass * klass);
-static void gst_jack_ring_buffer_dispose (GObject * object);
-static void gst_jack_ring_buffer_finalize (GObject * object);
-
-static GstRingBufferClass *ring_parent_class = NULL;
-
-static gboolean gst_jack_ring_buffer_open_device (GstRingBuffer * buf);
-static gboolean gst_jack_ring_buffer_close_device (GstRingBuffer * buf);
-static gboolean gst_jack_ring_buffer_acquire (GstRingBuffer * buf,
-    GstRingBufferSpec * spec);
-static gboolean gst_jack_ring_buffer_release (GstRingBuffer * buf);
-static gboolean gst_jack_ring_buffer_start (GstRingBuffer * buf);
-static gboolean gst_jack_ring_buffer_pause (GstRingBuffer * buf);
-static gboolean gst_jack_ring_buffer_stop (GstRingBuffer * buf);
-static guint gst_jack_ring_buffer_delay (GstRingBuffer * buf);
-
 static gboolean
 gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels)
 {
@@ -173,7 +122,7 @@ gst_jack_audio_sink_free_channels (GstJa
 
 /* ringbuffer abstract base class */
 static GType
-gst_jack_ring_buffer_get_type (void)
+gst_jack_ring_buffer_get_type()
 {
   static GType ringbuffer_type = 0;
 
@@ -689,24 +638,11 @@ enum
   PROP_LAST
 };
 
+#if 0
 #define GST_TYPE_JACK_CONNECT (gst_jack_connect_get_type())
-static GType
-gst_jack_connect_get_type (void)
-{
-  static GType jack_connect_type = 0;
-  static const GEnumValue jack_connect[] = {
-    {GST_JACK_CONNECT_NONE,
-        "Don't automatically connect ports to physical ports", "none"},
-    {GST_JACK_CONNECT_AUTO,
-        "Automatically connect ports to physical ports", "auto"},
-    {0, NULL, NULL},
-  };
-
-  if (!jack_connect_type) {
-    jack_connect_type = g_enum_register_static ("GstJackConnect", jack_connect);
-  }
-  return jack_connect_type;
-}
+GType
+gst_jack_connect_get_type();
+#endif
 
 #define _do_init(bla) \
     GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element");
@@ -849,6 +785,7 @@ gst_jack_audio_sink_getcaps (GstBaseSink
     max = 0;
     if (ports != NULL) {
       for (; ports[max]; max++);
+
       free (ports);
     } else
       max = 0;
Index: jack/gstjackaudiosink.h
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-bad/ext/jack/gstjackaudiosink.h,v
retrieving revision 1.3
diff -p -u -w -r1.3 gstjackaudiosink.h
--- jack/gstjackaudiosink.h	8 Mar 2007 15:24:52 -0000	1.3
+++ jack/gstjackaudiosink.h	4 Aug 2008 21:38:38 -0000
@@ -27,6 +27,7 @@
 #include <gst/gst.h>
 #include <gst/audio/gstbaseaudiosink.h>
 
+#include "gstjack.h"
 #include "gstjackaudioclient.h"
 
 G_BEGIN_DECLS
@@ -42,22 +43,6 @@ typedef struct _GstJackAudioSink GstJack
 typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass;
 
 /**
- * GstJackConnect:
- * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports.
- *     In this mode, the element will accept any number of input channels and will
- *     create (but not connect) an output port for each channel.
- * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each
- *     output port to a random physical jack input pin. The sink will
- *     expose the number of physical channels on its pad caps.
- *
- * Specify how the output ports will be connected.
- */
-typedef enum {
-  GST_JACK_CONNECT_NONE,
-  GST_JACK_CONNECT_AUTO
-} GstJackConnect;
-
-/**
  * GstJackAudioSink:
  * 
  * Opaque #GstJackAudioSink.
@@ -85,7 +70,7 @@ struct _GstJackAudioSinkClass {
   GstBaseAudioSinkClass parent_class;
 };
 
-GType gst_jack_audio_sink_get_type (void);
+GType gst_jack_audio_sink_get_type ();
 
 G_END_DECLS
 
Index: jack/gstjackaudiosrc.c
===================================================================
RCS file: jack/gstjackaudiosrc.c
diff -N jack/gstjackaudiosrc.c
--- /dev/null	1 Jan 1970 00:00:00 -0000
+++ jack/gstjackaudiosrc.c	4 Aug 2008 21:38:39 -0000
@@ -0,0 +1,841 @@
+/* GStreamer
+ * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
+ * 
+ * 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.
+ */
+
+/**
+ * SECTION:element-jackaudiosrc
+ * @see_also: #GstBaseAudioSrc, #GstRingBuffer
+ *
+ * A Src that inputs data from Jack ports.
+ * 
+ * It will create N Jack ports named in_&lt;name&gt;_&lt;num&gt; where 
+ * &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
+ * Each port corresponds to a gstreamer channel.
+ * 
+ * The samplerate as exposed on the caps is always the same as the samplerate of
+ * the jack server.
+ * 
+ * When the #GstJackAudioSrc:connect property is set to auto, this element
+ * will try to connect each input port to a random physical jack output pin. 
+ * 
+ * When the #GstJackAudioSrc:connect property is set to none, the element will
+ * accept any number of output channels and will create (but not connect) an
+ * input port for each channel.
+ * 
+ * The element will generate an error when the Jack server is shut down when it
+ * was PAUSED or PLAYING. This element does not support dynamic rate and buffer
+ * size changes at runtime.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch jackaudiosrc connect=0 ! jackaudiosink connect=0
+ * ]| Get audio input into gstreamer from jack.
+ * </refsect2>
+ *
+ * Last reviewed on 2008-07-22 (0.10.4)
+ */
+
+#include <gst/gst.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "gstjackaudiosrc.h"
+#include "gstjackringbuffer.h"
+
+GST_DEBUG_CATEGORY_STATIC(gst_jackaudiosrc_debug);
+#define GST_CAT_DEFAULT gst_jackaudiosrc_debug
+
+static gboolean 
+gst_jack_audio_src_allocate_channels(GstJackAudioSrc * src, gint channels)
+{
+  jack_client_t *client;
+
+  client = gst_jack_audio_client_get_client(src->client);
+
+  /* remove ports we don't need */
+  while (src->port_count > channels) 
+    jack_port_unregister(client, src->ports[--src->port_count]);
+
+  /* alloc enough input ports */
+  src->ports = g_realloc(src->ports, sizeof(jack_port_t *) * channels);
+
+  /* create an input port for each channel */
+  while (src->port_count < channels) 
+  {
+    gchar *name;
+
+    /* port names start from 1 and are local to the element */
+    name =
+      g_strdup_printf("in_%s_%d", GST_ELEMENT_NAME(src),
+          src->port_count + 1);
+    src->ports[src->port_count] =
+      jack_port_register(client, name, JACK_DEFAULT_AUDIO_TYPE,
+          JackPortIsInput, 0);
+    if (src->ports[src->port_count] == NULL)
+      return FALSE;
+
+    src->port_count++;
+
+    g_free(name);
+  }
+  return TRUE;
+}
+
+static void 
+gst_jack_audio_src_free_channels(GstJackAudioSrc * src)
+{
+  gint res, i = 0;
+  jack_client_t *client;
+
+  client = gst_jack_audio_client_get_client(src->client);
+
+  /* get rid of all ports */
+  while (src->port_count) 
+  {
+    GST_LOG_OBJECT(src, "unregister port %d", i);
+    if ((res = jack_port_unregister(client, src->ports[i++]))) 
+      GST_DEBUG_OBJECT(src, "unregister of port failed (%d)", res);
+
+    src->port_count--;
+  }
+  g_free(src->ports);
+  src->ports = NULL;
+}
+
+/* ringbuffer abstract base class */
+static GType 
+gst_jack_ring_buffer_get_type()
+{
+  static GType ringbuffer_type = 0;
+
+  if (!ringbuffer_type) {
+    static const GTypeInfo ringbuffer_info = { sizeof(GstJackRingBufferClass),
+      NULL,
+      NULL,
+      (GClassInitFunc) gst_jack_ring_buffer_class_init,
+      NULL,
+      NULL,
+      sizeof(GstJackRingBuffer),
+      0,
+      (GInstanceInitFunc) gst_jack_ring_buffer_init,
+      NULL
+    };
+
+    ringbuffer_type =
+      g_type_register_static(GST_TYPE_RING_BUFFER,
+          "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
+  }
+  return ringbuffer_type;
+}
+
+static void 
+gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstObjectClass *gstobject_class;
+  GstRingBufferClass *gstringbuffer_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstobject_class = (GstObjectClass *) klass;
+  gstringbuffer_class = (GstRingBufferClass *) klass;
+
+  ring_parent_class = g_type_class_peek_parent(klass);
+
+  gobject_class->dispose = GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_dispose);
+  gobject_class->finalize = GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_finalize);
+
+  gstringbuffer_class->open_device =
+    GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_open_device);
+  gstringbuffer_class->close_device =
+    GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_close_device);
+  gstringbuffer_class->acquire =
+    GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_acquire);
+  gstringbuffer_class->release =
+    GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_release);
+  gstringbuffer_class->start = GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_start);
+  gstringbuffer_class->pause = GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_pause);
+  gstringbuffer_class->resume = GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_start);
+  gstringbuffer_class->stop = GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_stop);
+
+  gstringbuffer_class->delay = GST_DEBUG_FUNCPTR(gst_jack_ring_buffer_delay);
+}
+
+/* this is the callback of jack. This should be RT-safe.
+ * Writes samples from the jack input port's buffer to the gst ring buffer.
+ */
+static int 
+jack_process_cb(jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSrc *src;
+  GstRingBuffer *buf;
+  GstJackRingBuffer *abuf;
+  gint len, givenLen;
+  guint8 *writeptr, *dataStart;
+  gint writeseg;
+  gint channels, i, j;
+  sample_t **buffers, *data;
+
+  buf = GST_RING_BUFFER_CAST(arg);
+  abuf = GST_JACK_RING_BUFFER_CAST(arg);
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  channels = buf->spec.channels;
+  len = sizeof(sample_t) * nframes * channels;
+
+  /* alloc pointers to samples */
+  buffers = g_alloca(sizeof(sample_t *) * channels);
+  data = g_alloca(len);
+
+  /* get input buffers */
+  for (i = 0; i < channels; i++) 
+    buffers[i] = (sample_t *)jack_port_get_buffer(src->ports[i], nframes);
+
+  //writeptr = data; 
+  dataStart = (guint8*) data;
+
+  /* the samples in the jack input buffers have to be interleaved into the 
+   * ringbuffer 
+   */
+
+  for (i = 0; i < nframes; ++i) 
+    for (j = 0; j < channels; ++j)
+      *data++ = buffers[j][i]; 
+
+  if (gst_ring_buffer_prepare_read(buf, &writeseg, &writeptr, &givenLen))
+  {
+    memcpy(writeptr, (char *) dataStart, givenLen);
+
+    GST_DEBUG("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr, len / channels, channels);
+
+    /* clear written samples in the ringbuffer */
+    // gst_ring_buffer_clear(buf, 0);
+
+    /* we wrote one segment */
+    gst_ring_buffer_advance(buf, 1);
+  }
+  return 0;
+}
+
+/* we error out */
+static int 
+jack_sample_rate_cb(jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST(arg);
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(arg));
+
+  if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR(src, RESOURCE, SETTINGS,
+        (NULL), ("Jack changed the sample rate, which is not supported"));
+    return 1;
+  }
+}
+
+/* we error out */
+static int 
+jack_buffer_size_cb(jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST(arg);
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(arg));
+
+  if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR(src, RESOURCE, SETTINGS,
+        (NULL), ("Jack changed the buffer size, which is not supported"));
+    return 1;
+  }
+}
+
+static void 
+jack_shutdown_cb(void *arg)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(arg));
+
+  GST_DEBUG_OBJECT(src, "shutdown");
+
+  GST_ELEMENT_ERROR(src, RESOURCE, NOT_FOUND,
+      (NULL), ("Jack server shutdown"));
+}
+
+static void 
+gst_jack_ring_buffer_init(GstJackRingBuffer * buf, GstJackRingBufferClass * g_class)
+{
+  buf->channels = -1;
+  buf->buffer_size = -1;
+  buf->sample_rate = -1;
+}
+
+static void 
+gst_jack_ring_buffer_dispose(GObject * object)
+{
+  G_OBJECT_CLASS(ring_parent_class)->dispose(object);
+}
+
+static void 
+gst_jack_ring_buffer_finalize(GObject * object)
+{
+  GstJackRingBuffer *ringbuffer;
+  ringbuffer = GST_JACK_RING_BUFFER_CAST(object);
+  G_OBJECT_CLASS(ring_parent_class)->finalize(object);
+}
+
+/* the _open_device method should make a connection with the server
+*/
+static gboolean 
+gst_jack_ring_buffer_open_device(GstRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+  jack_status_t status = 0;
+  const gchar *name;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  GST_DEBUG_OBJECT(src, "open");
+
+  name = g_get_application_name();
+  if (!name)
+    name = "GStreamer";
+
+  src->client = gst_jack_audio_client_new(name, src->server,
+      GST_JACK_CLIENT_SOURCE,
+      jack_shutdown_cb,
+      jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
+  if (src->client == NULL)
+    goto could_not_open;
+
+  GST_DEBUG_OBJECT(src, "opened");
+
+  return TRUE;
+
+  /* ERRORS */
+could_not_open:
+  {
+    if (status & JackServerFailed) {
+      GST_ELEMENT_ERROR(src, RESOURCE, NOT_FOUND,
+          (NULL), ("Cannot connect to the Jack server (status %d)", status));
+    } else {
+      GST_ELEMENT_ERROR(src, RESOURCE, OPEN_WRITE,
+          (NULL), ("Jack client open error (status %d)", status));
+    }
+    return FALSE;
+  }
+}
+
+/* close the connection with the server
+*/
+static gboolean 
+gst_jack_ring_buffer_close_device(GstRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  GST_DEBUG_OBJECT(src, "close");
+
+  gst_jack_audio_src_free_channels(src);
+  gst_jack_audio_client_free(src->client);
+  src->client = NULL;
+
+  return TRUE;
+}
+
+
+/* allocate a buffer and setup resources to process the audio samples of
+ * the format as specified in @spec.
+ *
+ * We allocate N jack ports, one for each channel. If we are asked to
+ * automatically make a connection with physical ports, we connect as many
+ * ports as there are physical ports, leaving leftover ports unconnected.
+ *
+ * It is assumed that samplerate and number of channels are acceptable since our
+ * getcaps method will always provide correct values. If unacceptable caps are
+ * received for some reason, we fail here.
+ */
+static gboolean 
+gst_jack_ring_buffer_acquire(GstRingBuffer * buf, GstRingBufferSpec * spec)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+  const char **ports;
+  gint sample_rate, buffer_size;
+  gint i, channels, res;
+  jack_client_t *client;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+  abuf = GST_JACK_RING_BUFFER_CAST(buf);
+
+  GST_DEBUG_OBJECT(src, "acquire");
+
+  client = gst_jack_audio_client_get_client(src->client);
+
+  /* sample rate must be that of the server */
+  sample_rate = jack_get_sample_rate(client);
+  if (sample_rate != spec->rate)
+    goto wrong_samplerate;
+
+  channels = spec->channels;
+
+  if (!gst_jack_audio_src_allocate_channels(src, channels))
+    goto out_of_ports;
+
+  buffer_size = jack_get_buffer_size(client);
+
+  /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
+   * for all channels  */
+  spec->segsize = buffer_size * sizeof(gfloat) * channels;
+  spec->latency_time = gst_util_uint64_scale(spec->segsize,
+      (GST_SECOND / GST_USECOND), spec->rate * spec->bytes_per_sample);
+  /* segtotal based on buffer-time latency */
+  spec->segtotal = spec->buffer_time / spec->latency_time;
+
+  GST_DEBUG_OBJECT(src, "segsize %d, segtotal %d", spec->segsize,
+      spec->segtotal);
+
+  /* allocate the ringbuffer memory now */
+  buf->data = gst_buffer_new_and_alloc(spec->segtotal * spec->segsize);
+  memset(GST_BUFFER_DATA(buf->data), 0, GST_BUFFER_SIZE(buf->data));
+
+  if ((res = gst_jack_audio_client_set_active(src->client, TRUE)))
+    goto could_not_activate;
+
+  /* if we need to automatically connect the ports, do so now. We must do this
+   * after activating the client. */
+  if (src->connect == GST_JACK_CONNECT_AUTO) {
+    /* find all the physical output ports. A physical output port is a port
+     * associated with a hardware device. Someone needs connect to a physical
+     * port in order to capture something. */
+    ports = jack_get_ports(client, NULL, NULL, JackPortIsPhysical | JackPortIsOutput);
+    if (ports == NULL) {
+      /* no ports? fine then we don't do anything except for posting a warning
+       * message. */
+      GST_ELEMENT_WARNING(src, RESOURCE, NOT_FOUND, (NULL),
+          ("No physical output ports found, leaving ports unconnected"));
+      goto done;
+    }
+
+    for (i = 0; i < channels; i++) 
+    {
+      /* stop when all output ports are exhausted */
+      if (ports[i] == NULL) {
+        /* post a warning that we could not connect all ports */
+        GST_ELEMENT_WARNING(src, RESOURCE, NOT_FOUND, (NULL),
+            ("No more physical ports, leaving some ports unconnected"));
+        break;
+      }
+      GST_DEBUG_OBJECT(src, "try connecting to %s",
+          jack_port_name(src->ports[i]));
+      /* connect the physical port to a port */
+
+      res = jack_connect(client, ports[i], jack_port_name(src->ports[i]));
+      g_print("connecting to %s\n", jack_port_name(src->ports[i]));
+      if (res != 0 && res != EEXIST)
+        goto cannot_connect;
+    }
+    free(ports);
+  }
+done:
+
+  abuf->sample_rate = sample_rate;
+  abuf->buffer_size = buffer_size;
+  abuf->channels = spec->channels;
+
+  return TRUE;
+
+  /* ERRORS */
+wrong_samplerate:
+  {
+    GST_ELEMENT_ERROR(src, RESOURCE, SETTINGS, (NULL),
+        ("Wrong samplerate, server is running at %d and we received %d",
+         sample_rate, spec->rate));
+    return FALSE;
+  }
+out_of_ports:
+  {
+    GST_ELEMENT_ERROR(src, RESOURCE, SETTINGS, (NULL),
+        ("Cannot allocate more Jack ports"));
+    return FALSE;
+  }
+could_not_activate:
+  {
+    GST_ELEMENT_ERROR(src, RESOURCE, SETTINGS, (NULL),
+        ("Could not activate client (%d:%s)", res, g_strerror(res)));
+    return FALSE;
+  }
+cannot_connect:
+  {
+    GST_ELEMENT_ERROR(src, RESOURCE, SETTINGS, (NULL),
+        ("Could not connect input ports to physical ports (%d:%s)",
+         res, g_strerror(res)));
+    free(ports);
+    return FALSE;
+  }
+}
+
+/* function is called with LOCK */
+static gboolean 
+gst_jack_ring_buffer_release(GstRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+  gint res;
+
+  abuf = GST_JACK_RING_BUFFER_CAST(buf);
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  GST_DEBUG_OBJECT(src, "release");
+
+  if ((res = gst_jack_audio_client_set_active(src->client, FALSE))) {
+    /* we only warn, this means the server is probably shut down and the client
+     * is gone anyway. */
+    GST_ELEMENT_WARNING(src, RESOURCE, CLOSE, (NULL),
+        ("Could not deactivate Jack client (%d)", res));
+  }
+
+  abuf->channels = -1;
+  abuf->buffer_size = -1;
+  abuf->sample_rate = -1;
+
+  /* free the buffer */
+  gst_buffer_unref(buf->data);
+  buf->data = NULL;
+
+  return TRUE;
+}
+
+static gboolean 
+gst_jack_ring_buffer_start(GstRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  GST_DEBUG_OBJECT(src, "start");
+
+  return TRUE;
+}
+
+static gboolean 
+gst_jack_ring_buffer_pause(GstRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  GST_DEBUG_OBJECT(src, "pause");
+
+  return TRUE;
+}
+
+static gboolean 
+gst_jack_ring_buffer_stop(GstRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  GST_DEBUG_OBJECT(src, "stop");
+
+  return TRUE;
+}
+
+static guint 
+gst_jack_ring_buffer_delay(GstRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+  guint res = 0;
+
+  src = GST_JACK_AUDIO_SRC(GST_OBJECT_PARENT(buf));
+
+  GST_DEBUG_OBJECT(src, "delay %u", res);
+
+  return res;
+}
+
+/* Audiosrc signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+#define DEFAULT_PROP_CONNECT 	GST_JACK_CONNECT_AUTO
+#define DEFAULT_PROP_SERVER 	NULL
+
+enum
+{
+  PROP_0,
+  PROP_CONNECT,
+  PROP_SERVER,
+  PROP_LAST
+};
+
+
+/* the capabilities of the inputs and outputs.
+ *
+ * describe the real formats here.
+ */
+
+static GstStaticPadTemplate src_factory = 
+GST_STATIC_PAD_TEMPLATE("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS("audio/x-raw-float, "
+      "endianness = (int) { " G_STRINGIFY(G_BYTE_ORDER) " }, "
+      "width = (int) 32, "
+      "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
+    );
+
+#define _do_init(bla) \
+  GST_DEBUG_CATEGORY_INIT(gst_jackaudiosrc_debug, "jacksrc", 0, "jacksrc element");
+
+GST_BOILERPLATE_FULL(GstJackAudioSrc, gst_jackaudiosrc, GstBaseAudioSrc,
+    GST_TYPE_BASE_AUDIO_SRC, _do_init);
+
+static void gst_jackaudiosrc_set_property(GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_jackaudiosrc_get_property(GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_jackaudiosrc_getcaps(GstBaseSrc * bsrc);
+static GstRingBuffer *gst_jackaudiosrc_create_ringbuffer(GstBaseAudioSrc *src);
+
+/* GObject vmethod implementations */
+
+static void 
+gst_jackaudiosrc_base_init(gpointer gclass)
+{
+  static GstElementDetails gst_jackaudiosrc_details = {
+    "Audio Source (Jack)",
+    "Source/Audio",
+    "Input from Jack",
+    "Tristan Matthews <tristan@sat.qc.ca>"
+  };
+  GstElementClass *element_class = GST_ELEMENT_CLASS(gclass);
+
+  gst_element_class_add_pad_template(element_class,
+      gst_static_pad_template_get(&src_factory));
+  gst_element_class_set_details(element_class, &gst_jackaudiosrc_details);
+}
+
+/* initialize the jackaudiosrc's class */
+static void 
+gst_jackaudiosrc_class_init(GstJackAudioSrcClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSrcClass *gstbasesrc_class;
+  GstBaseAudioSrcClass *gstbaseaudiosrc_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  gstbasesrc_class = (GstBaseSrcClass *) klass;
+  gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
+
+  gobject_class->set_property = GST_DEBUG_FUNCPTR(gst_jackaudiosrc_set_property);
+  gobject_class->get_property = GST_DEBUG_FUNCPTR(gst_jackaudiosrc_get_property);
+
+  g_object_class_install_property(gobject_class, PROP_CONNECT,
+      g_param_spec_enum("connect", "Connect",
+        "Specify how the input ports will be connected",
+        GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT, G_PARAM_READWRITE));
+
+  g_object_class_install_property(gobject_class, PROP_SERVER,
+      g_param_spec_string("server", "Server",
+        "The Jack server to connect to (NULL = default)",
+        DEFAULT_PROP_SERVER, G_PARAM_READWRITE));
+
+  gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR(gst_jackaudiosrc_getcaps);
+  gstbaseaudiosrc_class->create_ringbuffer =
+    GST_DEBUG_FUNCPTR(gst_jackaudiosrc_create_ringbuffer);
+
+  /* ref class from a thread-safe context to work around missing bit of
+   * thread-safety in GObject */
+  g_type_class_ref(GST_TYPE_JACK_RING_BUFFER);
+
+  gst_jack_audio_client_init();
+}
+
+/* initialize the new element
+ * instantiate pads and add them to element
+ * set pad calback functions
+ * initialize instance structure
+ */
+static void 
+gst_jackaudiosrc_init(GstJackAudioSrc * src, GstJackAudioSrcClass * gclass)
+{
+  //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
+  src->connect = DEFAULT_PROP_CONNECT;
+  src->server = g_strdup(DEFAULT_PROP_SERVER);
+  src->ports = NULL;
+  src->port_count = 0;
+}
+
+static void 
+gst_jackaudiosrc_set_property(GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSrc *src = GST_JACK_AUDIO_SRC(object);
+
+  switch (prop_id) 
+  {
+    case PROP_CONNECT:
+      src->connect = g_value_get_enum(value);
+      break;
+    case PROP_SERVER:
+      g_free(src->server);
+      src->server = g_value_dup_string(value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec);
+      break;
+  }
+}
+
+static void 
+gst_jackaudiosrc_get_property(GObject * object, guint prop_id, 
+    GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSrc *src = GST_JACK_AUDIO_SRC(object);
+
+  switch (prop_id) 
+  {
+    case PROP_CONNECT:
+      g_value_set_enum(value, src->connect);
+      break;
+    case PROP_SERVER:
+      g_value_set_string(value, src->server);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID(object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_jackaudiosrc_getcaps(GstBaseSrc * bsrc)
+{
+  GstJackAudioSrc *src = GST_JACK_AUDIO_SRC(bsrc);
+  const char **ports;
+  gint min, max;
+  gint rate;
+  jack_client_t *client;
+
+  if (src->client == NULL)
+    goto no_client;
+
+  client = gst_jack_audio_client_get_client(src->client);
+
+  if (src->connect == GST_JACK_CONNECT_AUTO) {
+    /* get a port count, this is the number of channels we can automatically
+     * connect. */
+    ports = jack_get_ports(client, NULL, NULL,
+        JackPortIsPhysical | JackPortIsOutput);
+    max = 0;
+    if (ports != NULL) {
+      for (; ports[max]; max++);
+
+      free(ports);
+    } else
+      max = 0;
+  } else {
+    /* we allow any number of pads, something else is going to connect the
+     * pads. */
+    max = G_MAXINT;
+  }
+  min = MIN(1, max);
+
+  rate = jack_get_sample_rate(client);
+
+  GST_DEBUG_OBJECT(src, "got %d-%d ports, samplerate: %d", min, max, rate);
+
+  if (!src->caps) {
+    src->caps = gst_caps_new_simple("audio/x-raw-float",
+        "endianness", G_TYPE_INT, G_BYTE_ORDER,
+        "width", G_TYPE_INT, 32,
+        "rate", G_TYPE_INT, rate,
+        "channels", GST_TYPE_INT_RANGE, min, max, NULL);
+  }
+  GST_INFO_OBJECT(src, "returning caps %" GST_PTR_FORMAT, src->caps);
+
+  return gst_caps_ref(src->caps);
+
+  /* ERRORS */
+no_client:
+  {
+    GST_DEBUG_OBJECT(src, "device not open, using template caps");
+    /* base class will get template caps for us when we return NULL */
+    return NULL;
+  }
+}
+
+static GstRingBuffer *
+gst_jackaudiosrc_create_ringbuffer(GstBaseAudioSrc * src)
+{
+  GstRingBuffer *buffer;
+
+  buffer = g_object_new(GST_TYPE_JACK_RING_BUFFER, NULL);
+  GST_DEBUG_OBJECT(src, "created ringbuffer @%p", buffer);
+
+  return buffer;
+}
+
Index: jack/gstjackaudiosrc.h
===================================================================
RCS file: jack/gstjackaudiosrc.h
diff -N jack/gstjackaudiosrc.h
--- /dev/null	1 Jan 1970 00:00:00 -0000
+++ jack/gstjackaudiosrc.h	4 Aug 2008 21:38:39 -0000
@@ -0,0 +1,94 @@
+/* GStreamer
+ * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
+ *
+ * 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.
+ */
+
+#ifndef __GST_JACK_AUDIO_SRC_H__
+#define __GST_JACK_AUDIO_SRC_H__
+
+#include <jack/jack.h>
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosrc.h>
+
+#include "gstjackaudioclient.h"
+#include "gstjack.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_JACK_AUDIO_SRC             (gst_jackaudiosrc_get_type())
+#define GST_JACK_AUDIO_SRC(obj)             (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc))
+#define GST_JACK_AUDIO_SRC_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
+#define GST_IS_JACK_AUDIO_SRC(obj)          (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC))
+#define GST_IS_JACK_AUDIO_SRC_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC))
+
+typedef struct _GstJackAudioSrc GstJackAudioSrc;
+typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass;
+
+struct _GstJackAudioSrc
+{
+    GstBaseAudioSrc src;
+
+    /*< private >*/
+    /* cached caps */
+    GstCaps         *caps;
+
+    /* properties */
+    GstJackConnect connect;
+    gchar           *server;
+
+    /* our client */
+    GstJackAudioClient *client;
+
+    /* our ports */
+    jack_port_t    **ports;
+    int port_count;
+};
+
+struct _GstJackAudioSrcClass
+{
+    GstBaseAudioSrcClass parent_class;
+};
+
+GType gst_jackaudiosrc_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_JACK_AUDIO_SRC_H__ */
Index: jack/gstjackringbuffer.h
===================================================================
RCS file: jack/gstjackringbuffer.h
diff -N jack/gstjackringbuffer.h
--- /dev/null	1 Jan 1970 00:00:00 -0000
+++ jack/gstjackringbuffer.h	4 Aug 2008 21:38:39 -0000
@@ -0,0 +1,87 @@
+/*
+ * GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
+ *
+ * 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.
+ */
+
+
+#define GST_TYPE_JACK_RING_BUFFER               (gst_jack_ring_buffer_get_type())
+#define GST_JACK_RING_BUFFER(obj)               (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer))
+#define GST_JACK_RING_BUFFER_CLASS(klass)       (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
+#define GST_JACK_RING_BUFFER_GET_CLASS(obj)     (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
+#define GST_JACK_RING_BUFFER_CAST(obj)          ((GstJackRingBuffer *)obj)
+#define GST_IS_JACK_RING_BUFFER(obj)            (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER))
+#define GST_IS_JACK_RING_BUFFER_CLASS(klass)    (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER))
+
+typedef struct _GstJackRingBuffer GstJackRingBuffer;
+typedef struct _GstJackRingBufferClass GstJackRingBufferClass;
+
+struct _GstJackRingBuffer
+{
+  GstRingBuffer object;
+
+  gint sample_rate;
+  gint buffer_size;
+  gint channels;
+};
+
+struct _GstJackRingBufferClass
+{
+  GstRingBufferClass parent_class;
+};
+
+static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass);
+static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer,
+    GstJackRingBufferClass * klass);
+static void gst_jack_ring_buffer_dispose(GObject * object);
+static void gst_jack_ring_buffer_finalize(GObject * object);
+
+static GstRingBufferClass *ring_parent_class = NULL;
+
+static gboolean gst_jack_ring_buffer_open_device(GstRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_close_device(GstRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_acquire(GstRingBuffer * buf,GstRingBufferSpec * spec);
+static gboolean gst_jack_ring_buffer_release(GstRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_start(GstRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_pause(GstRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_stop(GstRingBuffer * buf);
+static guint gst_jack_ring_buffer_delay(GstRingBuffer * buf);
+
