Add apexsink for audio output to Apple AirPort Express Wireless devices. Fixes bug #542510.

Original commit message from CVS:
Patch by: Jérémie Bernard <gremimail at gmail dot com>
* configure.ac:
* ext/apexsink/LGPL-3.0.txt:
* ext/apexsink/Makefile.am:
* ext/apexsink/gstapexplugin.c: (plugin_init):
* ext/apexsink/gstapexraop.c: (g_strdel), (gst_apexraop_send),
(gst_apexraop_recv), (gst_apexraop_new), (gst_apexraop_free),
(gst_apexraop_set_host), (gst_apexraop_get_host),
(gst_apexraop_set_port), (gst_apexraop_get_port),
(gst_apexraop_set_useragent), (gst_apexraop_get_useragent),
(gst_apexraop_connect), (gst_apexraop_get_jacktype),
(gst_apexraop_get_jackstatus), (gst_apexraop_close),
(gst_apexraop_set_volume), (gst_apexraop_write_bits),
(gst_apexraop_write), (gst_apexraop_flush):
* ext/apexsink/gstapexraop.h:
* ext/apexsink/gstapexsink.c: (gst_apexsink_jackstatus_get_type),
(gst_apexsink_jacktype_get_type), (gst_apexsink_interfaces_init),
(gst_apexsink_implements_interface_init),
(gst_apexsink_mixer_interface_init),
(gst_apexsink_interface_supported),
(gst_apexsink_mixer_list_tracks), (gst_apexsink_mixer_set_volume),
(gst_apexsink_mixer_get_volume), (gst_apexsink_base_init),
(gst_apexsink_class_init), (gst_apexsink_init),
(gst_apexsink_set_property), (gst_apexsink_get_property),
(gst_apexsink_finalise), (gst_apexsink_open),
(gst_apexsink_prepare), (gst_apexsink_write),
(gst_apexsink_unprepare), (gst_apexsink_delay),
(gst_apexsink_reset), (gst_apexsink_close):
* ext/apexsink/gstapexsink.h:
Add apexsink for audio output to Apple AirPort Express Wireless
devices. Fixes bug #542510.

1 files changed, 571 insertions, 0 deletions

diff --git a/ext/apexsink/gstapexsink.c b/ext/apexsink/gstapexsink.c
new file mode 100644
index 00000000..5957093d
--- /dev/null
+++ b/ext/apexsink/gstapexsink.c
@@ -0,0 +1,571 @@
+/* GStreamer - AirPort Express Audio Sink -
+ *
+ * Remote Audio Access Protocol (RAOP) as used in Apple iTunes to stream music to the Airport Express (ApEx) -
+ * RAOP is based on the Real Time Streaming Protocol (RTSP) but with an extra challenge-response RSA based authentication step.
+ *
+ * RAW PCM input only as defined by the following GST_STATIC_PAD_TEMPLATE
+ *
+ * Copyright (C) 2008 Jérémie Bernard [GRemi] <gremimail@gmail.com>
+ *
+ * gstapexsink.c 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstapexsink.h"
+
+GST_DEBUG_CATEGORY_STATIC (apexsink_debug);
+#define GST_CAT_DEFAULT apexsink_debug
+
+static GstStaticPadTemplate gst_apexsink_sink_factory = GST_STATIC_PAD_TEMPLATE
+    ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS
+    (GST_APEX_RAOP_INPUT_TYPE ","
+        "width = (int) " GST_APEX_RAOP_INPUT_WIDTH ","
+        "depth = (int) " GST_APEX_RAOP_INPUT_DEPTH ","
+        "endianness = (int) " GST_APEX_RAOP_INPUT_ENDIAN ","
+        "channels = (int) " GST_APEX_RAOP_INPUT_CHANNELS ","
+        "rate = (int) " GST_APEX_RAOP_INPUT_BIT_RATE ","
+        "signed = (boolean) " GST_APEX_RAOP_INPUT_SIGNED)
+    );
+
+
+enum
+{
+  APEX_PROP_HOST = 1,
+  APEX_PROP_PORT,
+  APEX_PROP_VOLUME,
+  APEX_PROP_JACK_TYPE,
+  APEX_PROP_JACK_STATUS,
+};
+
+#define DEFAULT_APEX_HOST		""
+#define DEFAULT_APEX_PORT		5000
+#define DEFAULT_APEX_VOLUME		75
+#define DEFAULT_APEX_JACK_TYPE		GST_APEX_JACK_TYPE_UNDEFINED
+#define DEFAULT_APEX_JACK_STATUS	GST_APEX_JACK_STATUS_UNDEFINED
+
+/* genum apex jack resolution */
+GType
+gst_apexsink_jackstatus_get_type (void)
+{
+  static GType jackstatus_type = 0;
+  static GEnumValue jackstatus[] = {
+    {GST_APEX_JACK_STATUS_UNDEFINED, "GST_APEX_JACK_STATUS_UNDEFINED",
+        "Jack status undefined"},
+    {GST_APEX_JACK_STATUS_DISCONNECTED, "GST_APEX_JACK_STATUS_DISCONNECTED",
+        "Jack disconnected"},
+    {GST_APEX_JACK_STATUS_CONNECTED, "GST_APEX_JACK_STATUS_CONNECTED",
+        "Jack connected"},
+    {0, NULL, NULL},
+  };
+
+  if (!jackstatus_type) {
+    jackstatus_type = g_enum_register_static ("GstApExJackStatus", jackstatus);
+  }
+
+  return jackstatus_type;
+}
+
+GType
+gst_apexsink_jacktype_get_type (void)
+{
+  static GType jacktype_type = 0;
+  static GEnumValue jacktype[] = {
+    {GST_APEX_JACK_TYPE_UNDEFINED, "GST_APEX_JACK_TYPE_UNDEFINED",
+        "Undefined jack type"},
+    {GST_APEX_JACK_TYPE_ANALOG, "GST_APEX_JACK_TYPE_ANALOG", "Analog jack"},
+    {GST_APEX_JACK_TYPE_DIGITAL, "GST_APEX_JACK_TYPE_DIGITAL", "Digital jack"},
+    {0, NULL, NULL},
+  };
+
+  if (!jacktype_type) {
+    jacktype_type = g_enum_register_static ("GstApExJackType", jacktype);
+  }
+
+  return jacktype_type;
+}
+
+
+static void gst_apexsink_base_init (gpointer g_class);
+static void gst_apexsink_class_init (GstApExSinkClass * klass);
+static void gst_apexsink_init (GstApExSink * apexsink,
+    GstApExSinkClass * g_class);
+
+static void gst_apexsink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_apexsink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_apexsink_finalise (GObject * object);
+
+static gboolean gst_apexsink_open (GstAudioSink * asink);
+static gboolean gst_apexsink_prepare (GstAudioSink * asink,
+    GstRingBufferSpec * spec);
+static guint gst_apexsink_write (GstAudioSink * asink, gpointer data,
+    guint length);
+static gboolean gst_apexsink_unprepare (GstAudioSink * asink);
+static guint gst_apexsink_delay (GstAudioSink * asink);
+static void gst_apexsink_reset (GstAudioSink * asink);
+static gboolean gst_apexsink_close (GstAudioSink * asink);
+
+/* mixer interface standard api */
+static void gst_apexsink_interfaces_init (GType type);
+static void gst_apexsink_implements_interface_init (GstImplementsInterfaceClass
+    * iface);
+static void gst_apexsink_mixer_interface_init (GstMixerClass * iface);
+
+static gboolean gst_apexsink_interface_supported (GstImplementsInterface *
+    iface, GType iface_type);
+static const GList *gst_apexsink_mixer_list_tracks (GstMixer * mixer);
+static void gst_apexsink_mixer_set_volume (GstMixer * mixer,
+    GstMixerTrack * track, gint * volumes);
+static void gst_apexsink_mixer_get_volume (GstMixer * mixer,
+    GstMixerTrack * track, gint * volumes);
+
+GST_BOILERPLATE_FULL (GstApExSink, gst_apexsink, GstAudioSink,
+    GST_TYPE_AUDIO_SINK, gst_apexsink_interfaces_init);
+
+/* apex sink interface(s) stuff */
+static void
+gst_apexsink_interfaces_init (GType type)
+{
+  static const GInterfaceInfo implements_interface_info =
+      { (GInterfaceInitFunc) gst_apexsink_implements_interface_init, NULL,
+    NULL
+  };
+  static const GInterfaceInfo mixer_interface_info =
+      { (GInterfaceInitFunc) gst_apexsink_mixer_interface_init, NULL, NULL };
+
+  g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
+      &implements_interface_info);
+  g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_interface_info);
+}
+
+static void
+gst_apexsink_implements_interface_init (GstImplementsInterfaceClass * iface)
+{
+  iface->supported = gst_apexsink_interface_supported;
+}
+
+static void
+gst_apexsink_mixer_interface_init (GstMixerClass * iface)
+{
+  GST_MIXER_TYPE (iface) = GST_MIXER_SOFTWARE;
+
+  iface->list_tracks = gst_apexsink_mixer_list_tracks;
+  iface->set_volume = gst_apexsink_mixer_set_volume;
+  iface->get_volume = gst_apexsink_mixer_get_volume;
+}
+
+static gboolean
+gst_apexsink_interface_supported (GstImplementsInterface * iface,
+    GType iface_type)
+{
+  g_return_val_if_fail (iface_type == GST_TYPE_MIXER, FALSE);
+
+  return TRUE;
+}
+
+static const GList *
+gst_apexsink_mixer_list_tracks (GstMixer * mixer)
+{
+  GstApExSink *apexsink = GST_APEX_SINK (mixer);
+
+  return apexsink->tracks;
+}
+
+static void
+gst_apexsink_mixer_set_volume (GstMixer * mixer, GstMixerTrack * track,
+    gint * volumes)
+{
+  GstApExSink *apexsink = GST_APEX_SINK (mixer);
+
+  apexsink->volume = volumes[0];
+
+  if (apexsink->gst_apexraop != NULL)
+    gst_apexraop_set_volume (apexsink->gst_apexraop, apexsink->volume);
+}
+
+static void
+gst_apexsink_mixer_get_volume (GstMixer * mixer, GstMixerTrack * track,
+    gint * volumes)
+{
+  GstApExSink *apexsink = GST_APEX_SINK (mixer);
+
+  volumes[0] = apexsink->volume;
+}
+
+/* sink base init */
+static void
+gst_apexsink_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_set_details_simple (element_class,
+      "Apple AirPort Express Audio Sink", "Sink/Audio/Wireless",
+      "Output stream to an AirPort Express",
+      "Jérémie Bernard [GRemi] <gremimail@gmail.com>");
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_apexsink_sink_factory));
+}
+
+/* sink class init */
+static void
+gst_apexsink_class_init (GstApExSinkClass * klass)
+{
+  GST_DEBUG_CATEGORY_INIT (apexsink_debug, GST_APEX_SINK_NAME, 0,
+      "AirPort Express sink");
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  ((GObjectClass *) klass)->get_property =
+      GST_DEBUG_FUNCPTR (gst_apexsink_get_property);
+  ((GObjectClass *) klass)->set_property =
+      GST_DEBUG_FUNCPTR (gst_apexsink_set_property);
+  ((GObjectClass *) klass)->finalize =
+      GST_DEBUG_FUNCPTR (gst_apexsink_finalise);
+
+  ((GstAudioSinkClass *) klass)->open = GST_DEBUG_FUNCPTR (gst_apexsink_open);
+  ((GstAudioSinkClass *) klass)->prepare =
+      GST_DEBUG_FUNCPTR (gst_apexsink_prepare);
+  ((GstAudioSinkClass *) klass)->write = GST_DEBUG_FUNCPTR (gst_apexsink_write);
+  ((GstAudioSinkClass *) klass)->unprepare =
+      GST_DEBUG_FUNCPTR (gst_apexsink_unprepare);
+  ((GstAudioSinkClass *) klass)->delay = GST_DEBUG_FUNCPTR (gst_apexsink_delay);
+  ((GstAudioSinkClass *) klass)->reset = GST_DEBUG_FUNCPTR (gst_apexsink_reset);
+  ((GstAudioSinkClass *) klass)->close = GST_DEBUG_FUNCPTR (gst_apexsink_close);
+
+  g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_HOST,
+      g_param_spec_string ("host", "Host", "AirPort Express target host",
+          DEFAULT_APEX_HOST, G_PARAM_READWRITE));
+  g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_PORT,
+      g_param_spec_uint ("port", "Port", "AirPort Express target port", 0,
+          32000, DEFAULT_APEX_PORT, G_PARAM_READWRITE));
+  g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_VOLUME,
+      g_param_spec_uint ("volume", "Volume", "AirPort Express target volume", 0,
+          100, DEFAULT_APEX_VOLUME, G_PARAM_READWRITE));
+  g_object_class_install_property ((GObjectClass *) klass, APEX_PROP_JACK_TYPE,
+      g_param_spec_enum ("jack_type", "Jack Type",
+          "AirPort Express connected jack type", GST_APEX_SINK_JACKTYPE_TYPE,
+          DEFAULT_APEX_JACK_TYPE, G_PARAM_READABLE));
+  g_object_class_install_property ((GObjectClass *) klass,
+      APEX_PROP_JACK_STATUS, g_param_spec_enum ("jack_status", "Jack Status",
+          "AirPort Express jack connection status",
+          GST_APEX_SINK_JACKSTATUS_TYPE, DEFAULT_APEX_JACK_STATUS,
+          G_PARAM_READABLE));
+}
+
+/* sink plugin instance init */
+static void
+gst_apexsink_init (GstApExSink * apexsink, GstApExSinkClass * g_class)
+{
+  GstMixerTrack *track = NULL;
+
+  track = g_object_new (GST_TYPE_MIXER_TRACK, NULL);
+  track->label = g_strdup ("Airport Express");
+  track->num_channels = GST_APEX_RAOP_CHANNELS;
+  track->min_volume = 0;
+  track->max_volume = 100;
+  track->flags = GST_MIXER_TRACK_OUTPUT;
+
+  apexsink->host = g_strdup (DEFAULT_APEX_HOST);
+  apexsink->port = DEFAULT_APEX_PORT;
+  apexsink->volume = DEFAULT_APEX_VOLUME;
+  apexsink->gst_apexraop = NULL;
+  apexsink->tracks = g_list_append (apexsink->tracks, track);
+
+  GST_INFO_OBJECT (apexsink,
+      "ApEx sink default initialization, target=\"%s\", port=\"%d\", volume=\"%d\%\"",
+      apexsink->host, apexsink->port, apexsink->volume);
+}
+
+/* apex sink set property */
+static void
+gst_apexsink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstApExSink *sink = GST_APEX_SINK (object);
+
+  switch (prop_id) {
+    case APEX_PROP_HOST:
+    {
+      if (sink->gst_apexraop == NULL) {
+        g_free (sink->host);
+        sink->host = g_value_dup_string (value);
+
+        GST_INFO_OBJECT (sink, "ApEx sink target set to \"%s\"", sink->host);
+      } else
+        G_OBJECT_WARN_INVALID_PSPEC (object, "host", prop_id, pspec);
+    }
+      break;
+    case APEX_PROP_PORT:
+    {
+      if (sink->gst_apexraop == NULL) {
+        sink->port = g_value_get_uint (value);
+
+        GST_INFO_OBJECT (sink, "ApEx port set to \"%d\"", sink->port);
+      } else
+        G_OBJECT_WARN_INVALID_PSPEC (object, "port", prop_id, pspec);
+    }
+      break;
+    case APEX_PROP_VOLUME:
+    {
+      sink->volume = g_value_get_uint (value);
+
+      if (sink->gst_apexraop != NULL)
+        gst_apexraop_set_volume (sink->gst_apexraop, sink->volume);
+
+      GST_INFO_OBJECT (sink, "ApEx volume set to \"%d\%\"", sink->volume);
+    }
+      break;
+    default:
+    {
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+    }
+      break;
+  }
+}
+
+/* apex sink get property */
+static void
+gst_apexsink_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstApExSink *sink = GST_APEX_SINK (object);
+
+  switch (prop_id) {
+    case APEX_PROP_HOST:
+    {
+      g_value_set_string (value, sink->host);
+    }
+      break;
+    case APEX_PROP_PORT:
+    {
+      g_value_set_uint (value, sink->port);
+    }
+      break;
+    case APEX_PROP_VOLUME:
+    {
+      g_value_set_uint (value, sink->volume);
+    }
+      break;
+    case APEX_PROP_JACK_TYPE:
+    {
+      g_value_set_enum (value, gst_apexraop_get_jacktype (sink->gst_apexraop));
+    }
+      break;
+    case APEX_PROP_JACK_STATUS:
+    {
+      g_value_set_enum (value,
+          gst_apexraop_get_jackstatus (sink->gst_apexraop));
+    }
+      break;
+    default:
+    {
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+    }
+      break;
+  }
+}
+
+/* apex sink finalize */
+static void
+gst_apexsink_finalise (GObject * object)
+{
+  GstApExSink *sink = GST_APEX_SINK (object);
+
+  if (sink->tracks) {
+    g_list_foreach (sink->tracks, (GFunc) g_object_unref, NULL);
+    g_list_free (sink->tracks);
+    sink->tracks = NULL;
+  }
+
+  g_free (sink->host);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* sink open : open the device */
+static gboolean
+gst_apexsink_open (GstAudioSink * asink)
+{
+  int res;
+  GstApExSink *apexsink = (GstApExSink *) asink;
+
+  apexsink->gst_apexraop = gst_apexraop_new (apexsink->host, apexsink->port);
+
+  if ((res = gst_apexraop_connect (apexsink->gst_apexraop)) != GST_RTSP_STS_OK) {
+    GST_ERROR_OBJECT (apexsink,
+        "%s : network or RAOP failure, connection refused or timeout, RTSP code=%d",
+        apexsink->host, res);
+    return FALSE;
+  }
+
+  GST_INFO_OBJECT (apexsink,
+      "OPEN : ApEx sink successfully connected to \"%s:%d\", ANNOUNCE, SETUP and RECORD requests performed",
+      apexsink->host, apexsink->port);
+
+  switch (gst_apexraop_get_jackstatus (apexsink->gst_apexraop)) {
+    case GST_APEX_JACK_STATUS_CONNECTED:
+    {
+      GST_INFO_OBJECT (apexsink, "OPEN : ApEx jack is connected");
+    }
+      break;
+    case GST_APEX_JACK_STATUS_DISCONNECTED:
+    {
+      GST_WARNING_OBJECT (apexsink, "OPEN : ApEx jack is disconnected !");
+    }
+      break;
+    default:
+    {
+      GST_WARNING_OBJECT (apexsink, "OPEN : ApEx jack status is undefined !");
+    }
+      break;
+  }
+
+  switch (gst_apexraop_get_jacktype (apexsink->gst_apexraop)) {
+    case GST_APEX_JACK_TYPE_ANALOG:
+    {
+      GST_INFO_OBJECT (apexsink, "OPEN : ApEx jack type is analog");
+    }
+      break;
+    case GST_APEX_JACK_TYPE_DIGITAL:
+    {
+      GST_INFO_OBJECT (apexsink, "OPEN : ApEx jack type is digital");
+    }
+      break;
+    default:
+    {
+      GST_WARNING_OBJECT (apexsink, "OPEN : ApEx jack type is undefined !");
+    }
+      break;
+  }
+
+  if ((res =
+          gst_apexraop_set_volume (apexsink->gst_apexraop,
+              apexsink->volume)) != GST_RTSP_STS_OK) {
+    GST_WARNING_OBJECT (apexsink,
+        "%s : could not set initial volume to \"%d\%\", RTSP code=%d",
+        apexsink->host, apexsink->volume, res);
+  } else {
+    GST_INFO_OBJECT (apexsink,
+        "OPEN : ApEx sink successfully set volume to \"%d\%\"",
+        apexsink->volume);
+  }
+
+  return TRUE;
+}
+
+/* prepare sink : configure the device with the specified format */
+static gboolean
+gst_apexsink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
+{
+  GstApExSink *apexsink = (GstApExSink *) asink;
+
+  apexsink->latency_time = spec->latency_time;
+
+  spec->segsize =
+      GST_APEX_RAOP_SAMPLES_PER_FRAME * GST_APEX_RAOP_BYTES_PER_SAMPLE;
+  spec->segtotal = 1;
+
+  bzero (spec->silence_sample, sizeof (spec->silence_sample));
+
+  GST_INFO_OBJECT (apexsink,
+      "PREPARE : ApEx sink ready to stream at %dHz, %d bytes per sample, %d channels, %d bytes segments (%dkB/s)",
+      spec->rate, spec->bytes_per_sample, spec->channels, spec->segsize,
+      spec->rate * spec->bytes_per_sample / 1000);
+
+  return TRUE;
+}
+
+/* sink write : write samples to the device */
+static guint
+gst_apexsink_write (GstAudioSink * asink, gpointer data, guint length)
+{
+  GstApExSink *apexsink = (GstApExSink *) asink;
+
+  if (gst_apexraop_write (apexsink->gst_apexraop, data, length) != length) {
+    GST_INFO_OBJECT (apexsink,
+        "WRITE : %d bytes not fully sended, skipping frame samples...", length);
+  } else {
+    GST_INFO_OBJECT (apexsink, "WRITE : %d bytes sended", length);
+
+    usleep ((gulong) ((length * 1000000.) / (GST_APEX_RAOP_BITRATE *
+                GST_APEX_RAOP_BYTES_PER_SAMPLE) - apexsink->latency_time));
+  }
+
+  return length;
+}
+
+/* unprepare sink : undo operations done by prepare */
+static gboolean
+gst_apexsink_unprepare (GstAudioSink * asink)
+{
+  GstApExSink *apexsink = (GstApExSink *) asink;
+
+  GST_INFO_OBJECT (apexsink, "UNPREPARE");
+
+  return TRUE;
+}
+
+/* delay sink : get the estimated number of samples written but not played yet by the device */
+static guint
+gst_apexsink_delay (GstAudioSink * asink)
+{
+  GstApExSink *apexsink = (GstApExSink *) asink;
+
+  GST_INFO_OBJECT (apexsink, "DELAY");
+
+  return 0;
+}
+
+/* reset sink : unblock writes and flush the device */
+static void
+gst_apexsink_reset (GstAudioSink * asink)
+{
+  int res;
+  GstApExSink *apexsink = (GstApExSink *) asink;
+
+  GST_INFO_OBJECT (apexsink, "RESET : flushing buffer...");
+
+  if ((res = gst_apexraop_flush (apexsink->gst_apexraop)) == GST_RTSP_STS_OK) {
+    GST_INFO_OBJECT (apexsink, "RESET : ApEx buffer flush success");
+  } else {
+    GST_WARNING_OBJECT (apexsink,
+        "RESET : could not flush ApEx buffer, RTSP code=%d", res);
+  }
+}
+
+/* sink close : close the device */
+static gboolean
+gst_apexsink_close (GstAudioSink * asink)
+{
+  GstApExSink *apexsink = (GstApExSink *) asink;
+
+  gst_apexraop_close (apexsink->gst_apexraop);
+  gst_apexraop_free (apexsink->gst_apexraop);
+
+  GST_INFO_OBJECT (apexsink, "CLOSE : ApEx sink closed connection");
+
+  return TRUE;
+}