Added fully functional jackaudiosink.

Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/jack/Makefile.am:
* ext/jack/gstjack.c: (plugin_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type),
(gst_jack_ring_buffer_class_init), (jack_process_cb),
(jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose),
(gst_jack_ring_buffer_finalize),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause),
(gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay),
(gst_jack_connect_get_type), (gst_jack_audio_sink_base_init),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_set_property),
(gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps),
(gst_jack_audio_sink_create_ringbuffer):
* ext/jack/gstjackaudiosink.h:
Added fully functional jackaudiosink.

1 files changed, 846 insertions, 0 deletions

diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c
new file mode 100644
index 00000000..5a0cadf8
--- /dev/null
+++ b/ext/jack/gstjackaudiosink.c
@@ -0,0 +1,846 @@
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjackaudiosink.c: jack audio sink implementation
+ *
+ * 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:gstjacksink
+ * @short_description: JACK audio sink
+ * @see_also: #GstBaseAudioSink, #GstRingBuffer
+ *
+ * A Sink that outputs data to Jack ports. 
+ *
+ * It will create N Jack ports named out_<num> where <num> 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 ::connect property is set to auto, this element will try to connect
+ * each output port to a random physical jack input pin. In this mode, the sink
+ * will expose the number of physical channels on its pad caps. 
+ *
+ * When the ::connect property is set to none, the element will accept any
+ * number of input channels and will create (but not connect) an output 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.
+ *
+ * Last reviewed on 2006-11-30 (0.10.4)
+ */
+#include <string.h>
+
+#include "gstjackaudiosink.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;
+
+  jack_port_t **outport;
+};
+
+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);
+
+/* ringbuffer abstract base class */
+static GType
+gst_jack_ring_buffer_get_type (void)
+{
+  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,
+        "GstJackAudioSinkRingBuffer", &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 RT-safe.
+ */
+static int
+jack_process_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSink *sink;
+  GstRingBuffer *buf;
+  GstJackRingBuffer *abuf;
+  gint readseg, len;
+  guint8 *readptr;
+  gint i, j, flen, channels;
+  sample_t **buffers, *data;
+
+  buf = GST_RING_BUFFER_CAST (arg);
+  abuf = GST_JACK_RING_BUFFER_CAST (arg);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  channels = buf->spec.channels;
+
+  /* alloc pointers to samples */
+  buffers = g_alloca (sizeof (sample_t *) * channels);
+
+  /* get target buffers */
+  for (i = 0; i < channels; i++) {
+    buffers[i] = (sample_t *) jack_port_get_buffer (abuf->outport[i], nframes);
+  }
+
+  if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
+    flen = len / channels;
+
+    if (nframes * sizeof (sample_t) != flen)
+      goto wrong_size;
+
+    /* copy samples */
+    GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, readptr,
+        flen, channels);
+    data = (sample_t *) readptr;
+
+    /* copy and interleave into target buffers */
+    for (i = 0; i < nframes; i++) {
+      for (j = 0; j < channels; j++) {
+        buffers[j][i] = *data++;
+      }
+    }
+
+    /* clear written samples */
+    gst_ring_buffer_clear (buf, readseg);
+
+    /* we wrote one segment */
+    gst_ring_buffer_advance (buf, 1);
+  } else {
+    /* write silence to all buffers */
+    for (i = 0; i < channels; i++) {
+      memset (buffers[i], 0, nframes * sizeof (sample_t));
+    }
+  }
+  return 0;
+
+  /* ERRORS */
+wrong_size:
+  {
+    GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
+        nframes * sizeof (sample_t), flen);
+    return 1;
+  }
+}
+
+/* we error out */
+static int
+jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (arg);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
+
+  if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR (sink, 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)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (arg);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
+
+  if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
+        (NULL), ("Jack changed the buffer size, which is not supported"));
+    return 1;
+  }
+}
+
+static void
+jack_shutdown_cb (void *arg)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
+
+  GST_DEBUG_OBJECT (sink, "shutdown");
+
+  GST_ELEMENT_ERROR (sink, 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)
+{
+  GstJackAudioSink *sink;
+  jack_options_t options;
+  jack_status_t status = 0;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "open");
+
+  /* never start a server */
+  options = JackNoStartServer;
+  /* if we have a servername, use it */
+  if (sink->server != NULL)
+    options |= JackServerName;
+  /* open the client */
+  sink->client = jack_client_open ("GStreamer", options, &status, sink->server);
+  if (sink->client == NULL)
+    goto could_not_open;
+
+  /* set our callbacks  */
+  jack_set_process_callback (sink->client, jack_process_cb, buf);
+  /* these callbacks cause us to error */
+  jack_set_buffer_size_callback (sink->client, jack_buffer_size_cb, buf);
+  jack_set_sample_rate_callback (sink->client, jack_sample_rate_cb, buf);
+  jack_on_shutdown (sink->client, jack_shutdown_cb, buf);
+
+  GST_DEBUG_OBJECT (sink, "opened");
+
+  return TRUE;
+
+  /* ERRORS */
+could_not_open:
+  {
+    if (status & JackServerFailed) {
+      GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
+          (NULL), ("Cannot connect to the Jack server (status %d)", status));
+    } else {
+      GST_ELEMENT_ERROR (sink, 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)
+{
+  GstJackAudioSink *sink;
+  gint res;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "close");
+
+  if ((res = jack_client_close (sink->client))) {
+    /* just a warning, we assume the client is gone. */
+    GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE,
+        (NULL), ("Jack client close error (%d)", res));
+  }
+  sink->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 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)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+  const char **ports;
+  gint sample_rate, buffer_size;
+  gint i, channels, res;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+  abuf = GST_JACK_RING_BUFFER_CAST (buf);
+
+  GST_DEBUG_OBJECT (sink, "acquire");
+
+  /* sample rate must be that of the server */
+  sample_rate = jack_get_sample_rate (sink->client);
+  if (sample_rate != spec->rate)
+    goto wrong_samplerate;
+
+  channels = spec->channels;
+
+  /* alloc enough output ports */
+  abuf->outport = g_new (jack_port_t *, channels);
+
+  /* create an output port for each channel */
+  for (i = 0; i < channels; i++) {
+    gchar *name;
+
+    /* port names start from 1 */
+    name = g_strdup_printf ("out_%d", i + 1);
+    abuf->outport[i] = jack_port_register (sink->client, name,
+        JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0);
+    if (abuf->outport[i] == NULL)
+      goto out_of_ports;
+
+    g_free (name);
+  }
+
+  buffer_size = jack_get_buffer_size (sink->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 (sink, "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 = jack_activate (sink->client)))
+    goto could_not_activate;
+
+  /* if we need to automatically connect the ports, do so now. We must do this
+   * after activating the client. */
+  if (sink->connect == GST_JACK_CONNECT_AUTO) {
+    /* find all the physical input ports. A physical input port is a port
+     * associated with a hardware device. Someone needs connect to a physical
+     * port in order to hear something. */
+    ports = jack_get_ports (sink->client, NULL, NULL,
+        JackPortIsPhysical | JackPortIsInput);
+    if (ports == NULL) {
+      /* no ports? fine then we don't do anything except for posting a warning
+       * message. */
+      GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
+          ("No physical input ports found, leaving ports unconnected"));
+      goto done;
+    }
+
+    for (i = 0; i < channels; i++) {
+      /* stop when all input ports are exhausted */
+      if (ports[i] == NULL) {
+        /* post a warning that we could not connect all ports */
+        GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
+            ("No more physical ports, leaving some ports unconnected"));
+        break;
+      }
+      /* connect the port to a physical port */
+      if ((res = jack_connect (sink->client, jack_port_name (abuf->outport[i]),
+                  ports[i])))
+        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 (sink, 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 (sink, RESOURCE, SETTINGS, (NULL),
+        ("Cannot allocate more Jack ports"));
+    return FALSE;
+  }
+could_not_activate:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+        ("Could not activate client (%d)", res));
+    return FALSE;
+  }
+cannot_connect:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+        ("Could not connect output ports to physical ports (%d)", res));
+    free (ports);
+    return FALSE;
+  }
+}
+
+/* function is called with LOCK */
+static gboolean
+gst_jack_ring_buffer_release (GstRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+  gint i, res;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (buf);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "release");
+
+  if ((res = jack_deactivate (sink->client))) {
+    /* we only warn, this means the server is probably shut down and the client
+     * is gone anyway. */
+    GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL),
+        ("Could not deactivate Jack client (%d)", res));
+  }
+
+  /* remove all ports */
+  for (i = 0; i < abuf->channels; i++) {
+    GST_LOG_OBJECT (sink, "unregister port %d", i);
+    if ((res = jack_port_unregister (sink->client, abuf->outport[i]))) {
+      GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
+    }
+    abuf->outport[i] = NULL;
+  }
+  g_free (abuf->outport);
+  abuf->outport = NULL;
+  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)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "start");
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_pause (GstRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "pause");
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_stop (GstRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "stop");
+
+  return TRUE;
+}
+
+static guint
+gst_jack_ring_buffer_delay (GstRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+  guint res = 0;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "delay %u", res);
+
+  return res;
+}
+
+/* elementfactory information */
+static const GstElementDetails gst_jack_audio_sink_details =
+GST_ELEMENT_DETAILS ("Audio Sink (Jack)",
+    "Sink/Audio",
+    "Output to Jack",
+    "Wim Taymans <wim@fluendo.com>");
+
+static GstStaticPadTemplate jackaudiosink_sink_factory =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    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 ]")
+    );
+
+/* AudioSink signals and args */
+enum
+{
+  /* FILL ME */
+  SIGNAL_LAST
+};
+
+#define DEFAULT_PROP_CONNECT 	GST_JACK_CONNECT_AUTO
+#define DEFAULT_PROP_SERVER 	NULL
+
+enum
+{
+  PROP_0,
+  PROP_CONNECT,
+  PROP_SERVER,
+  PROP_LAST
+};
+
+#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;
+}
+
+#define _do_init(bla) \
+    GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0, "jacksink element");
+
+GST_BOILERPLATE_FULL (GstJackAudioSink, gst_jack_audio_sink, GstBaseAudioSink,
+    GST_TYPE_BASE_AUDIO_SINK, _do_init);
+
+static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink);
+static GstRingBuffer *gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink *
+    sink);
+
+static void
+gst_jack_audio_sink_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_set_details (element_class, &gst_jack_audio_sink_details);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&jackaudiosink_sink_factory));
+}
+
+static void
+gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSinkClass *gstbasesink_class;
+  GstBaseAudioSinkClass *gstbaseaudiosink_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasesink_class = (GstBaseSinkClass *) klass;
+  gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass;
+
+  gobject_class->get_property =
+      GST_DEBUG_FUNCPTR (gst_jack_audio_sink_get_property);
+  gobject_class->set_property =
+      GST_DEBUG_FUNCPTR (gst_jack_audio_sink_set_property);
+
+  g_object_class_install_property (gobject_class, PROP_CONNECT,
+      g_param_spec_enum ("connect", "Connect",
+          "Specify how the output 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));
+
+  gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
+
+  gstbaseaudiosink_class->create_ringbuffer =
+      GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
+}
+
+static void
+gst_jack_audio_sink_init (GstJackAudioSink * sink,
+    GstJackAudioSinkClass * g_class)
+{
+  sink->connect = DEFAULT_PROP_CONNECT;
+  sink->server = g_strdup (DEFAULT_PROP_SERVER);
+}
+
+static void
+gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (object);
+
+  switch (prop_id) {
+    case PROP_CONNECT:
+      sink->connect = g_value_get_enum (value);
+      break;
+    case PROP_SERVER:
+      g_free (sink->server);
+      sink->server = g_value_dup_string (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (object);
+
+  switch (prop_id) {
+    case PROP_CONNECT:
+      g_value_set_enum (value, sink->connect);
+      break;
+    case PROP_SERVER:
+      g_value_set_string (value, sink->server);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_jack_audio_sink_getcaps (GstBaseSink * bsink)
+{
+  GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
+  const char **ports;
+  gint min, max;
+  gint rate;
+
+  if (sink->client == NULL)
+    goto no_client;
+
+  if (sink->connect == GST_JACK_CONNECT_AUTO) {
+    /* get a port count, this is the number of channels we can automatically
+     * connect. */
+    ports = jack_get_ports (sink->client, NULL, NULL,
+        JackPortIsPhysical | JackPortIsInput);
+    max = 0;
+    if (ports != NULL) {
+      for (; ports[max]; max++);
+      free (ports);
+    } else
+      max = 0;
+  } else {
+    /* we allow any number of pads, somoething else is going to connect the
+     * pads. */
+    max = G_MAXINT;
+  }
+  min = MIN (1, max);
+
+  rate = jack_get_sample_rate (sink->client);
+
+  GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
+
+  if (!sink->caps) {
+    sink->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 (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
+
+  return gst_caps_ref (sink->caps);
+
+  /* ERRORS */
+no_client:
+  {
+    GST_DEBUG_OBJECT (sink, "device not open, using template caps");
+    /* base class will get template caps for us when we return NULL */
+    return NULL;
+  }
+}
+
+static GstRingBuffer *
+gst_jack_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
+{
+  GstRingBuffer *buffer;
+
+  buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
+  GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
+
+  return buffer;
+}