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.

7 files changed, 1687 insertions, 0 deletions

diff --git a/ext/apexsink/LGPL-3.0.txt b/ext/apexsink/LGPL-3.0.txt
new file mode 100644
index 00000000..fc8a5de7
--- /dev/null
+++ b/ext/apexsink/LGPL-3.0.txt
@@ -0,0 +1,165 @@
+		   GNU LESSER GENERAL PUBLIC LICENSE
+                       Version 3, 29 June 2007
+
+ Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+
+  This version of the GNU Lesser General Public License incorporates
+the terms and conditions of version 3 of the GNU General Public
+License, supplemented by the additional permissions listed below.
+
+  0. Additional Definitions. 
+
+  As used herein, "this License" refers to version 3 of the GNU Lesser
+General Public License, and the "GNU GPL" refers to version 3 of the GNU
+General Public License.
+
+  "The Library" refers to a covered work governed by this License,
+other than an Application or a Combined Work as defined below.
+
+  An "Application" is any work that makes use of an interface provided
+by the Library, but which is not otherwise based on the Library.
+Defining a subclass of a class defined by the Library is deemed a mode
+of using an interface provided by the Library.
+
+  A "Combined Work" is a work produced by combining or linking an
+Application with the Library.  The particular version of the Library
+with which the Combined Work was made is also called the "Linked
+Version".
+
+  The "Minimal Corresponding Source" for a Combined Work means the
+Corresponding Source for the Combined Work, excluding any source code
+for portions of the Combined Work that, considered in isolation, are
+based on the Application, and not on the Linked Version.
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+  The "Corresponding Application Code" for a Combined Work means the
+object code and/or source code for the Application, including any data
+and utility programs needed for reproducing the Combined Work from the
+Application, but excluding the System Libraries of the Combined Work.
+
+  1. Exception to Section 3 of the GNU GPL.
+
+  You may convey a covered work under sections 3 and 4 of this License
+without being bound by section 3 of the GNU GPL.
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+  2. Conveying Modified Versions.
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+  If you modify a copy of the Library, and, in your modifications, a
+facility refers to a function or data to be supplied by an Application
+that uses the facility (other than as an argument passed when the
+facility is invoked), then you may convey a copy of the modified
+version:
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+   a) under this License, provided that you make a good faith effort to
+   ensure that, in the event an Application does not supply the
+   function or data, the facility still operates, and performs
+   whatever part of its purpose remains meaningful, or
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+   b) under the GNU GPL, with none of the additional permissions of
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+  3. Object Code Incorporating Material from Library Header Files.
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+  The object code form of an Application may incorporate material from
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+   a) Give prominent notice with each copy of the object code that the
+   Library is used in it and that the Library and its use are
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+   is a work based on the Library, and explaining where to find the
+   accompanying uncombined form of the same work.
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+  6. Revised Versions of the GNU Lesser General Public License.
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+  The Free Software Foundation may publish revised and/or new versions
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+versions will be similar in spirit to the present version, but may
+differ in detail to address new problems or concerns.
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diff --git a/ext/apexsink/Makefile.am b/ext/apexsink/Makefile.am
new file mode 100644
index 00000000..2b8ca57d
--- /dev/null
+++ b/ext/apexsink/Makefile.am
@@ -0,0 +1,8 @@
+plugin_LTLIBRARIES = libgstapexsink.la
+
+libgstapexsink_la_SOURCES = gstapexplugin.c gstapexraop.c gstapexsink.c
+libgstapexsink_la_CFLAGS = $(APEXSINK_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+libgstapexsink_la_LIBADD = $(APEXSINK_LIBS) $(GST_PLUGINS_BASE_LIBS) $(GST_LIBS) -lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR)
+libgstapexsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstapexraop.h gstapexsink.h
diff --git a/ext/apexsink/gstapexplugin.c b/ext/apexsink/gstapexplugin.c
new file mode 100644
index 00000000..014d5a2d
--- /dev/null
+++ b/ext/apexsink/gstapexplugin.c
@@ -0,0 +1,44 @@
+/* GStreamer AirPort Express Plugin
+ *
+ * Copyright (C) 2008 Jérémie Bernard [GRemi] <gremimail@gmail.com>
+ *
+ * gstapexpugin.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 <gst/gst.h>
+#include <gstapexsink.h>
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, GST_APEX_SINK_NAME, GST_RANK_PRIMARY,
+      GST_TYPE_APEX_SINK);
+}
+
+/* plugin export resolution */
+GST_PLUGIN_DEFINE
+    (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "apex",
+    "Apple AirPort Express Plugin",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/ext/apexsink/gstapexraop.c b/ext/apexsink/gstapexraop.c
new file mode 100644
index 00000000..164a4ccf
--- /dev/null
+++ b/ext/apexsink/gstapexraop.c
@@ -0,0 +1,690 @@
+/* GStreamer - 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.
+ * This interface accepts RAW PCM data and set it as AES encrypted ALAC while performing emission.
+ *
+ * Copyright (C) 2008 Jérémie Bernard [GRemi] <gremimail@gmail.com>
+ *
+ * gstapexraop.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 "gstapexraop.h"
+
+/* private constants */
+#define GST_APEX_RAOP_VOLUME_MIN 	-144
+#define GST_APEX_RAOP_VOLUME_MAX  	 0
+
+#define GST_APEX_RAOP_HDR_DEFAULT_LENGTH 1024
+#define GST_APEX_RAOP_SDP_DEFAULT_LENGTH 2048
+
+const static gchar GST_APEX_RAOP_RSA_PUBLIC_MOD[] =
+    "59dE8qLieItsH1WgjrcFRKj6eUWqi+bGLOX1HL3U3GhC/j0Qg90u3sG/1CUtwC"
+    "5vOYvfDmFI6oSFXi5ELabWJmT2dKHzBJKa3k9ok+8t9ucRqMd6DZHJ2YCCLlDR"
+    "KSKv6kDqnw4UwPdpOMXziC/AMj3Z/lUVX1G7WSHCAWKf1zNS1eLvqr+boEjXuB"
+    "OitnZ/bDzPHrTOZz0Dew0uowxf/+sG+NCK3eQJVxqcaJ/vEHKIVd2M+5qL71yJ"
+    "Q+87X6oV3eaYvt3zWZYD6z5vYTcrtij2VZ9Zmni/UAaHqn9JdsBWLUEpVviYnh"
+    "imNVvYFZeCXg/IdTQ+x4IRdiXNv5hEew==";
+
+const static gchar GST_APEX_RAOP_RSA_PUBLIC_EXP[] = "AQAB";
+
+const static gchar GST_APEX_RAOP_USER_AGENT[] =
+    "iTunes/4.6 (Macintosh; U; PPC Mac OS X 10.3)";
+
+const static guchar GST_APEX_RAOP_FRAME_HEADER[] = {
+  0x24, 0x00, 0x00, 0x00,
+  0xF0, 0xFF, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00,
+  0x00, 0x00, 0x00, 0x00
+};
+
+const static int GST_APEX_RAOP_FRAME_HEADER_SIZE = 16;
+
+const static int GST_APEX_RAOP_ALAC_HEADER_SIZE = 3;
+
+/* string extra utility */
+static gint
+g_strdel (gchar * str, gchar rc)
+{
+  int i = 0, j = 0, len, num = 0;
+  len = strlen (str);
+  while (i < len) {
+    if (str[i] == rc) {
+      for (j = i; j < len; j++)
+        str[j] = str[j + 1];
+      len--;
+      num++;
+    } else {
+      i++;
+    }
+  }
+  return num;
+}
+
+/* socket utilities */
+static int
+gst_apexraop_send (int desc, void *data, size_t len)
+{
+  int total = 0, bytesleft = len, n = 0;
+
+  while (total < len) {
+    n = send (desc, ((const char *) data) + total, bytesleft, 0);
+    if (n == -1)
+      break;
+    total += n;
+    bytesleft -= n;
+  }
+
+  return n == -1 ? -1 : total;
+}
+
+static int
+gst_apexraop_recv (int desc, void *data, size_t len)
+{
+  bzero (data, len);
+  return recv (desc, data, len, 0);
+}
+
+/* public opaque handle resolution */
+typedef struct
+{
+  guchar aes_ky[AES_BLOCK_SIZE];        /* AES random key */
+  guchar aes_iv[AES_BLOCK_SIZE];        /* AES random initial vector */
+
+  guchar url_abspath[16];       /* header url random absolute path addon, ANNOUNCE id */
+  gint cseq;                    /* header rtsp inc cseq */
+  guchar cid[24];               /* header client instance id */
+  gchar *session;               /* header raop negotiated session id, once SETUP performed */
+  gchar *ua;                    /* header user agent */
+
+  GstApExJackType jack_type;    /* APEX connected jack type, once ANNOUNCE performed */
+  GstApExJackStatus jack_status;        /* APEX connected jack status, once ANNOUNCE performed */
+
+  gchar *host;                  /* APEX target ip */
+  guint ctrl_port;              /* APEX target control port */
+  guint data_port;              /* APEX negotiated data port, once SETUP performed */
+
+  int ctrl_sd;                  /* control socket */
+  struct sockaddr_in ctrl_sd_in;
+
+  int data_sd;                  /* data socket */
+  struct sockaddr_in data_sd_in;
+}
+_GstApExRAOP;
+
+/* raop apex struct allocation */
+GstApExRAOP *
+gst_apexraop_new (const gchar * host, const guint16 port)
+{
+  _GstApExRAOP *apexraop;
+
+  apexraop = (_GstApExRAOP *) g_malloc0 (sizeof (_GstApExRAOP));
+
+  apexraop->host = g_strdup (host);
+  apexraop->ctrl_port = port;
+  apexraop->ua = g_strdup (GST_APEX_RAOP_USER_AGENT);
+  apexraop->jack_type = GST_APEX_JACK_TYPE_UNDEFINED;
+  apexraop->jack_status = GST_APEX_JACK_STATUS_DISCONNECTED;
+
+  return (GstApExRAOP *) apexraop;
+}
+
+/* raop apex struct freeing */
+void
+gst_apexraop_free (GstApExRAOP * con)
+{
+  _GstApExRAOP *conn;
+  conn = (_GstApExRAOP *) con;
+
+  g_free (conn->host);
+  g_free (conn->session);
+  g_free (conn->ua);
+  g_free (conn);
+}
+
+/* host affectation */
+void
+gst_apexraop_set_host (GstApExRAOP * con, const gchar * host)
+{
+  _GstApExRAOP *conn;
+  conn = (_GstApExRAOP *) con;
+
+  g_free (conn->host);
+  conn->host = g_strdup (host);
+}
+
+/* host reader */
+gchar *
+gst_apexraop_get_host (GstApExRAOP * con)
+{
+  _GstApExRAOP *conn;
+  conn = (_GstApExRAOP *) con;
+
+  return g_strdup (conn->host);
+}
+
+/* control port affectation */
+void
+gst_apexraop_set_port (GstApExRAOP * con, const guint16 port)
+{
+  _GstApExRAOP *conn;
+  conn = (_GstApExRAOP *) con;
+
+  conn->ctrl_port = port;
+}
+
+/* control port reader */
+guint16
+gst_apexraop_get_port (GstApExRAOP * con)
+{
+  _GstApExRAOP *conn;
+  conn = (_GstApExRAOP *) con;
+
+  return conn->ctrl_port;
+}
+
+/* user agent affectation */
+void
+gst_apexraop_set_useragent (GstApExRAOP * con, const gchar * useragent)
+{
+  _GstApExRAOP *conn;
+  conn = (_GstApExRAOP *) con;
+
+  g_free (conn->ua);
+  conn->ua = g_strdup (useragent);
+}
+
+/* user agent reader */
+gchar *
+gst_apexraop_get_useragent (GstApExRAOP * con)
+{
+  _GstApExRAOP *conn;
+  conn = (_GstApExRAOP *) con;
+
+  return g_strdup (conn->ua);
+}
+
+/* raop apex connection sequence */
+GstRTSPStatusCode
+gst_apexraop_connect (GstApExRAOP * con)
+{
+  gchar *ac, *ky, *iv, *s, inaddr[INET_ADDRSTRLEN],
+      creq[GST_APEX_RAOP_SDP_DEFAULT_LENGTH],
+      hreq[GST_APEX_RAOP_HDR_DEFAULT_LENGTH], *req;
+  RSA *rsa;
+  guchar *mod, *exp, buf[4 + 8 + 16], rsakey[512];
+  gsize size;
+  struct sockaddr_in ioaddr;
+  socklen_t iolen;
+  GstRTSPStatusCode res;
+  _GstApExRAOP *conn;
+
+  conn = (_GstApExRAOP *) con;
+
+  if ((conn->ctrl_sd = socket (AF_INET, SOCK_STREAM, 0)) < 0)
+    return GST_RTSP_STS_DESTINATION_UNREACHABLE;
+
+  conn->ctrl_sd_in.sin_family = AF_INET;
+  conn->ctrl_sd_in.sin_port = htons (conn->ctrl_port);
+
+  if (!inet_aton (conn->host, &conn->ctrl_sd_in.sin_addr)) {
+    struct hostent *hp = (struct hostent *) gethostbyname (conn->host);
+    if (hp == NULL)
+      return GST_RTSP_STS_DESTINATION_UNREACHABLE;
+    memcpy (&conn->ctrl_sd_in.sin_addr, hp->h_addr, hp->h_length);
+  }
+
+  if (connect (conn->ctrl_sd, (struct sockaddr *) &conn->ctrl_sd_in,
+          sizeof (conn->ctrl_sd_in)) < 0)
+    return GST_RTSP_STS_DESTINATION_UNREACHABLE;
+
+  RAND_bytes (buf, sizeof (buf));
+  sprintf ((gchar *) conn->url_abspath, "%lu", *((gulong *) buf));
+  ac = g_base64_encode (buf + 12, 16);
+  g_strdel (ac, '=');
+  sprintf ((char *) conn->cid, "%08lx%08lx", *((gulong *) (buf + 4)),
+      *((gulong *) (buf + 8)));
+
+  RAND_bytes (conn->aes_ky, AES_BLOCK_SIZE);
+  RAND_bytes (conn->aes_iv, AES_BLOCK_SIZE);
+
+  rsa = RSA_new ();
+  mod = g_base64_decode (GST_APEX_RAOP_RSA_PUBLIC_MOD, &size);
+  rsa->n = BN_bin2bn (mod, size, NULL);
+  exp = g_base64_decode (GST_APEX_RAOP_RSA_PUBLIC_EXP, &size);
+  rsa->e = BN_bin2bn (exp, size, NULL);
+  size =
+      RSA_public_encrypt (AES_BLOCK_SIZE, conn->aes_ky, rsakey, rsa,
+      RSA_PKCS1_OAEP_PADDING);
+
+  ky = g_base64_encode (rsakey, size);
+  iv = g_base64_encode (conn->aes_iv, AES_BLOCK_SIZE);
+  g_strdel (ky, '=');
+  g_strdel (iv, '=');
+
+  iolen = sizeof (struct sockaddr);
+  getsockname (conn->ctrl_sd, (struct sockaddr *) &ioaddr, &iolen);
+  inet_ntop (AF_INET, &(ioaddr.sin_addr), inaddr, INET_ADDRSTRLEN);
+
+  sprintf (creq,
+      "v=0\r\n"
+      "o=iTunes %s 0 IN IP4 %s\r\n"
+      "s=iTunes\r\n"
+      "c=IN IP4 %s\r\n"
+      "t=0 0\r\n"
+      "m=audio 0 RTP/AVP 96\r\n"
+      "a=rtpmap:96 AppleLossless\r\n"
+      "a=fmtp:96 %d 0 %d 40 10 14 %d 255 0 0 %d\r\n"
+      "a=rsaaeskey:%s\r\n"
+      "a=aesiv:%s\r\n",
+      conn->url_abspath,
+      inaddr,
+      conn->host,
+      GST_APEX_RAOP_SAMPLES_PER_FRAME,
+      GST_APEX_RAOP_BYTES_PER_CHANNEL * 8,
+      GST_APEX_RAOP_CHANNELS, GST_APEX_RAOP_BITRATE, ky, iv);
+
+  sprintf (hreq,
+      "ANNOUNCE rtsp://%s/%s RTSP/1.0\r\n"
+      "CSeq: %d\r\n"
+      "Client-Instance: %s\r\n"
+      "User-Agent: %s\r\n"
+      "Content-Type: application/sdp\r\n"
+      "Content-Length: %d\r\n"
+      "Apple-Challenge: %s\r\n",
+      conn->host,
+      conn->url_abspath, ++conn->cseq, conn->cid, conn->ua, strlen (creq), ac);
+
+  RSA_free (rsa);
+  g_free (ky);
+  g_free (iv);
+  g_free (ac);
+  g_free (mod);
+  g_free (exp);
+
+  req = g_strconcat (hreq, "\r\n", creq, NULL);
+
+  if (gst_apexraop_send (conn->ctrl_sd, req, strlen (req)) <= 0) {
+    g_free (req);
+    return GST_RTSP_STS_GONE;
+  }
+
+  g_free (req);
+
+  if (gst_apexraop_recv (conn->ctrl_sd, hreq,
+          GST_APEX_RAOP_HDR_DEFAULT_LENGTH) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  sscanf (hreq, "%*s %d", (int *) &res);
+
+  if (res != GST_RTSP_STS_OK)
+    return res;
+
+  s = g_strrstr (hreq, "Audio-Jack-Status");
+
+  if (s != NULL) {
+    gchar status[128];
+    sscanf (s, "%*s %s", status);
+
+    if (strcmp (status, "connected;") == 0)
+      conn->jack_status = GST_APEX_JACK_STATUS_CONNECTED;
+    else if (strcmp (status, "disconnected;") == 0)
+      conn->jack_status = GST_APEX_JACK_STATUS_DISCONNECTED;
+    else
+      conn->jack_status = GST_APEX_JACK_STATUS_UNDEFINED;
+
+    s = g_strrstr (s, "type=");
+
+    if (s != NULL) {
+      strtok (s, "=");
+      s = strtok (NULL, "\n");
+
+      if (strcmp (s, "analog"))
+        conn->jack_type = GST_APEX_JACK_TYPE_ANALOG;
+      else if (strcmp (s, "digital"))
+        conn->jack_type = GST_APEX_JACK_TYPE_DIGITAL;
+      else
+        conn->jack_type = GST_APEX_JACK_TYPE_UNDEFINED;
+    }
+  }
+
+  sprintf (hreq,
+      "SETUP rtsp://%s/%s RTSP/1.0\r\n"
+      "CSeq: %d\r\n"
+      "Client-Instance: %s\r\n"
+      "User-Agent: %s\r\n"
+      "Transport: RTP/AVP/TCP;unicast;interleaved=0-1;mode=record\r\n"
+      "\r\n", conn->host, conn->url_abspath, ++conn->cseq, conn->cid, conn->ua);
+
+  if (gst_apexraop_send (conn->ctrl_sd, hreq, strlen (hreq)) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  if (gst_apexraop_recv (conn->ctrl_sd, hreq,
+          GST_APEX_RAOP_HDR_DEFAULT_LENGTH) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  sscanf (hreq, "%*s %d", (int *) &res);
+
+  if (res != GST_RTSP_STS_OK)
+    return res;
+
+  s = g_strrstr (hreq, "Session");
+
+  if (s != NULL) {
+    gchar session[128];
+    sscanf (s, "%*s %s", session);
+    conn->session = g_strdup (session);
+  } else
+    return GST_RTSP_STS_PRECONDITION_FAILED;
+
+  s = g_strrstr (hreq, "server_port");
+  if (s != NULL) {
+    sscanf (s, "server_port=%d", &conn->data_port);
+  } else
+    return GST_RTSP_STS_PRECONDITION_FAILED;
+
+  sprintf (hreq,
+      "RECORD rtsp://%s/%s RTSP/1.0\r\n"
+      "CSeq: %d\r\n"
+      "Client-Instance: %s\r\n"
+      "User-Agent: %s\r\n"
+      "Session: %s\r\n"
+      "Range: npt=0-\r\n"
+      "RTP-Info: seq=0;rtptime=0\r\n"
+      "\r\n",
+      conn->host,
+      conn->url_abspath, ++conn->cseq, conn->cid, conn->ua, conn->session);
+
+  if (gst_apexraop_send (conn->ctrl_sd, hreq, strlen (hreq)) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  if (gst_apexraop_recv (conn->ctrl_sd, hreq,
+          GST_APEX_RAOP_HDR_DEFAULT_LENGTH) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  sscanf (hreq, "%*s %d", (int *) &res);
+
+  if (res != GST_RTSP_STS_OK)
+    return res;
+
+  if ((conn->data_sd = socket (AF_INET, SOCK_STREAM, 0)) < 0)
+    return GST_RTSP_STS_DESTINATION_UNREACHABLE;
+
+  conn->data_sd_in.sin_family = AF_INET;
+  conn->data_sd_in.sin_port = htons (conn->data_port);
+
+  memcpy (&conn->data_sd_in.sin_addr, &conn->ctrl_sd_in.sin_addr,
+      sizeof (conn->ctrl_sd_in.sin_addr));
+
+  if (connect (conn->data_sd, (struct sockaddr *) &conn->data_sd_in,
+          sizeof (conn->data_sd_in)) < 0)
+    return GST_RTSP_STS_DESTINATION_UNREACHABLE;
+
+  return res;
+}
+
+/* raop apex jack type access */
+GstApExJackType
+gst_apexraop_get_jacktype (GstApExRAOP * con)
+{
+  _GstApExRAOP *conn;
+
+  conn = (_GstApExRAOP *) con;
+
+  if (!conn)
+    return GST_APEX_JACK_TYPE_UNDEFINED;
+
+  return conn->jack_type;
+}
+
+/* raop apex jack status access */
+GstApExJackStatus
+gst_apexraop_get_jackstatus (GstApExRAOP * con)
+{
+  _GstApExRAOP *conn;
+
+  conn = (_GstApExRAOP *) con;
+
+  if (!conn)
+    return GST_APEX_JACK_STATUS_UNDEFINED;
+
+  return conn->jack_status;
+}
+
+/* raop apex sockets close */
+void
+gst_apexraop_close (GstApExRAOP * con)
+{
+  gchar hreq[GST_APEX_RAOP_HDR_DEFAULT_LENGTH];
+  _GstApExRAOP *conn;
+
+  conn = (_GstApExRAOP *) con;
+
+  sprintf (hreq,
+      "TEARDOWN rtsp://%s/%s RTSP/1.0\r\n"
+      "CSeq: %d\r\n"
+      "Client-Instance: %s\r\n"
+      "User-Agent: %s\r\n"
+      "Session: %s\r\n"
+      "\r\n",
+      conn->host,
+      conn->url_abspath, ++conn->cseq, conn->cid, conn->ua, conn->session);
+
+  gst_apexraop_send (conn->ctrl_sd, hreq, strlen (hreq));
+  gst_apexraop_recv (conn->ctrl_sd, hreq, GST_APEX_RAOP_HDR_DEFAULT_LENGTH);
+
+  if (conn->ctrl_sd != 0)
+    close (conn->ctrl_sd);
+  if (conn->data_sd != 0)
+    close (conn->data_sd);
+}
+
+/* raop apex volume set */
+GstRTSPStatusCode
+gst_apexraop_set_volume (GstApExRAOP * con, const guint volume)
+{
+  gint v;
+  gchar creq[GST_APEX_RAOP_SDP_DEFAULT_LENGTH],
+      hreq[GST_APEX_RAOP_HDR_DEFAULT_LENGTH], *req, vol[128];
+  GstRTSPStatusCode res;
+  _GstApExRAOP *conn;
+
+  conn = (_GstApExRAOP *) con;
+
+  v = GST_APEX_RAOP_VOLUME_MIN + (GST_APEX_RAOP_VOLUME_MAX -
+      GST_APEX_RAOP_VOLUME_MIN) * volume / 100.;
+  sprintf (vol, "volume: %d.000000\r\n", v);
+
+  sprintf (creq, "%s\r\n", vol);
+
+  sprintf (hreq,
+      "SET_PARAMETER rtsp://%s/%s RTSP/1.0\r\n"
+      "CSeq: %d\r\n"
+      "Client-Instance: %s\r\n"
+      "User-Agent: %s\r\n"
+      "Session: %s\r\n"
+      "Content-Type: text/parameters\r\n"
+      "Content-Length: %d\r\n",
+      conn->host,
+      conn->url_abspath,
+      ++conn->cseq, conn->cid, conn->ua, conn->session, strlen (creq)
+      );
+
+  req = g_strconcat (hreq, "\r\n", creq, NULL);
+
+  if (gst_apexraop_send (conn->ctrl_sd, req, strlen (req)) <= 0) {
+    g_free (req);
+    return GST_RTSP_STS_GONE;
+  }
+
+  g_free (req);
+
+  if (gst_apexraop_recv (conn->ctrl_sd, hreq,
+          GST_APEX_RAOP_HDR_DEFAULT_LENGTH) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  sscanf (hreq, "%*s %d", (int *) &res);
+
+  return res;
+}
+
+/* raop apex raw data alac encapsulation, encryption and emission, http://wiki.multimedia.cx/index.php?title=Apple_Lossless_Audio_Coding */
+static void inline
+gst_apexraop_write_bits (guchar * buffer, int data, int numbits,
+    int *bit_offset, int *byte_offset)
+{
+  const static guchar masks[] =
+      { 0x01, 0x03, 0x07, 0x0F, 0x1F, 0x3F, 0x7F, 0xFF };
+
+  if (((*bit_offset) != 0) && (((*bit_offset) + numbits) > 8)) {
+    gint numwritebits;
+    guchar bitstowrite;
+
+    numwritebits = 8 - (*bit_offset);
+    bitstowrite =
+        (guchar) ((data >> (numbits - numwritebits)) << (8 - (*bit_offset) -
+            numwritebits));
+    buffer[(*byte_offset)] |= bitstowrite;
+    numbits -= numwritebits;
+    (*bit_offset) = 0;
+    (*byte_offset)++;
+  }
+
+  while (numbits >= 8) {
+    guchar bitstowrite;
+
+    bitstowrite = (guchar) ((data >> (numbits - 8)) & 0xFF);
+    buffer[(*byte_offset)] |= bitstowrite;
+    numbits -= 8;
+    (*bit_offset) = 0;
+    (*byte_offset)++;
+  }
+
+  if (numbits > 0) {
+    guchar bitstowrite;
+    bitstowrite =
+        (guchar) ((data & masks[numbits]) << (8 - (*bit_offset) - numbits));
+    buffer[(*byte_offset)] |= bitstowrite;
+    (*bit_offset) += numbits;
+    if ((*bit_offset) == 8) {
+      (*byte_offset)++;
+      (*bit_offset) = 0;
+    }
+  }
+}
+
+guint
+gst_apexraop_write (GstApExRAOP * con, gpointer rawdata, guint length)
+{
+  guchar *buffer, *frame_data;
+  gushort len;
+  gint bit_offset, byte_offset, i, out_len, res;
+  EVP_CIPHER_CTX aes_ctx;
+  _GstApExRAOP *conn;
+
+  conn = (_GstApExRAOP *) con;
+
+  buffer =
+      (guchar *) g_malloc0 (GST_APEX_RAOP_FRAME_HEADER_SIZE +
+      GST_APEX_RAOP_ALAC_HEADER_SIZE + length);
+
+  memcpy (buffer, GST_APEX_RAOP_FRAME_HEADER, GST_APEX_RAOP_FRAME_HEADER_SIZE);
+
+  len =
+      length + GST_APEX_RAOP_FRAME_HEADER_SIZE +
+      GST_APEX_RAOP_ALAC_HEADER_SIZE - 4;
+  buffer[2] = len >> 8;
+  buffer[3] = len & 0xff;
+
+  bit_offset = 0;
+  byte_offset = 0;
+  frame_data = buffer + GST_APEX_RAOP_FRAME_HEADER_SIZE;
+
+  gst_apexraop_write_bits (frame_data, 1, 3, &bit_offset, &byte_offset);        /* channels, 0 mono, 1 stereo */
+  gst_apexraop_write_bits (frame_data, 0, 4, &bit_offset, &byte_offset);        /* unknown */
+  gst_apexraop_write_bits (frame_data, 0, 8, &bit_offset, &byte_offset);        /* unknown (12 bits) */
+  gst_apexraop_write_bits (frame_data, 0, 4, &bit_offset, &byte_offset);
+  gst_apexraop_write_bits (frame_data, 0, 1, &bit_offset, &byte_offset);        /* has size flag */
+  gst_apexraop_write_bits (frame_data, 0, 2, &bit_offset, &byte_offset);        /* unknown */
+  gst_apexraop_write_bits (frame_data, 1, 1, &bit_offset, &byte_offset);        /* no compression flag */
+
+  for (i = 0; i < length; i += 2) {
+    gst_apexraop_write_bits (frame_data, ((guchar *) rawdata)[i + 1], 8,
+        &bit_offset, &byte_offset);
+    gst_apexraop_write_bits (frame_data, ((guchar *) rawdata)[i], 8,
+        &bit_offset, &byte_offset);
+  }
+
+  EVP_CIPHER_CTX_init (&aes_ctx);
+  EVP_CipherInit_ex (&aes_ctx, EVP_aes_128_cbc (), NULL, conn->aes_ky,
+      conn->aes_iv, AES_ENCRYPT);
+  EVP_CipherUpdate (&aes_ctx, frame_data, &out_len, frame_data, /*( */
+      GST_APEX_RAOP_ALAC_HEADER_SIZE +
+      length /*) / AES_BLOCK_SIZE * AES_BLOCK_SIZE */ );
+  EVP_CIPHER_CTX_cleanup (&aes_ctx);
+
+  res =
+      gst_apexraop_send (conn->data_sd, buffer,
+      GST_APEX_RAOP_FRAME_HEADER_SIZE + GST_APEX_RAOP_ALAC_HEADER_SIZE +
+      length);
+
+  g_free (buffer);
+
+  return (guint) ((res >=
+          (GST_APEX_RAOP_FRAME_HEADER_SIZE +
+              GST_APEX_RAOP_ALAC_HEADER_SIZE)) ? (res -
+          GST_APEX_RAOP_FRAME_HEADER_SIZE -
+          GST_APEX_RAOP_ALAC_HEADER_SIZE) : 0);
+}
+
+/* raop apex buffer flush */
+GstRTSPStatusCode
+gst_apexraop_flush (GstApExRAOP * con)
+{
+  gchar hreq[GST_APEX_RAOP_HDR_DEFAULT_LENGTH];
+  GstRTSPStatusCode res;
+  _GstApExRAOP *conn;
+
+  conn = (_GstApExRAOP *) con;
+
+  sprintf (hreq,
+      "FLUSH rtsp://%s/%s RTSP/1.0\r\n"
+      "CSeq: %d\r\n"
+      "Client-Instance: %s\r\n"
+      "User-Agent: %s\r\n"
+      "Session: %s\r\n"
+      "RTP-Info: seq=0;rtptime=0\r\n"
+      "\r\n",
+      conn->host,
+      conn->url_abspath, ++conn->cseq, conn->cid, conn->ua, conn->session);
+
+  if (gst_apexraop_send (conn->ctrl_sd, hreq, strlen (hreq)) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  if (gst_apexraop_recv (conn->ctrl_sd, hreq,
+          GST_APEX_RAOP_HDR_DEFAULT_LENGTH) <= 0)
+    return GST_RTSP_STS_GONE;
+
+  sscanf (hreq, "%*s %d", (int *) &res);
+
+  return res;
+}
diff --git a/ext/apexsink/gstapexraop.h b/ext/apexsink/gstapexraop.h
new file mode 100644
index 00000000..fe1ba41b
--- /dev/null
+++ b/ext/apexsink/gstapexraop.h
@@ -0,0 +1,124 @@
+/* GStreamer - 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.
+ * This interface accepts RAW PCM data and set it as AES encrypted ALAC while performing emission.
+ *
+ * Copyright (C) 2008 Jérémie Bernard [GRemi] <gremimail@gmail.com>
+ *
+ * gstapexraop.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_APEXRAOP_H__
+#define __GST_APEXRAOP_H__
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspdefs.h>
+
+#include <openssl/rand.h>
+#include <openssl/rsa.h>
+#include <openssl/aes.h>
+#include <openssl/evp.h>
+
+#include <string.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include <netinet/in.h>
+#include <netdb.h>
+
+#include <arpa/inet.h>
+
+G_BEGIN_DECLS
+
+/* raop fixed parameters */
+#define GST_APEX_RAOP_BITRATE				44100
+#define GST_APEX_RAOP_SAMPLES_PER_FRAME			4096
+#define GST_APEX_RAOP_BYTES_PER_CHANNEL			2
+#define GST_APEX_RAOP_CHANNELS				2
+#define GST_APEX_RAOP_BYTES_PER_SAMPLE			(GST_APEX_RAOP_CHANNELS * GST_APEX_RAOP_BYTES_PER_CHANNEL)
+
+/* gst associated caps fields specification */
+#define GST_APEX_RAOP_INPUT_TYPE			"audio/x-raw-int"
+#define GST_APEX_RAOP_INPUT_WIDTH			"16"
+#define GST_APEX_RAOP_INPUT_DEPTH	            	GST_APEX_RAOP_INPUT_WIDTH
+#define GST_APEX_RAOP_INPUT_ENDIAN			"LITTLE_ENDIAN"
+#define GST_APEX_RAOP_INPUT_CHANNELS            	"2"
+#define GST_APEX_RAOP_INPUT_BIT_RATE			"44100"
+#define GST_APEX_RAOP_INPUT_SIGNED			"TRUE"
+
+typedef enum
+{
+  GST_APEX_JACK_TYPE_UNDEFINED = 0,
+  GST_APEX_JACK_TYPE_ANALOG,
+  GST_APEX_JACK_TYPE_DIGITAL,
+}
+GstApExJackType;
+
+typedef enum
+{
+  GST_APEX_JACK_STATUS_UNDEFINED = 0,
+  GST_APEX_JACK_STATUS_DISCONNECTED,
+  GST_APEX_JACK_STATUS_CONNECTED,
+}
+GstApExJackStatus;
+
+/* raop context handle */
+typedef struct
+{
+} GstApExRAOP;
+
+/* host might be null and port might be 0 while instanciating */
+GstApExRAOP *gst_apexraop_new (const gchar * host, const guint16 port);
+void gst_apexraop_free (GstApExRAOP * conn);
+
+/* must not be connected yet while setting the host target */
+void gst_apexraop_set_host (GstApExRAOP * conn, const gchar * host);
+gchar *gst_apexraop_get_host (GstApExRAOP * conn);
+
+/* must not be connected yet while setting the port target */
+void gst_apexraop_set_port (GstApExRAOP * conn, const guint16 port);
+guint16 gst_apexraop_get_port (GstApExRAOP * conn);
+
+/* optional affectation, default iTunes user agent internaly used */
+void gst_apexraop_set_useragent (GstApExRAOP * conn, const gchar * useragent);
+gchar *gst_apexraop_get_useragent (GstApExRAOP * conn);
+
+/* once allocation and configuration performed, manages the raop ANNOUNCE, SETUP and RECORD sequences, 
+ * open both ctrl and data channels */
+GstRTSPStatusCode gst_apexraop_connect (GstApExRAOP * conn);
+
+/* close the currently used session, manages raop TEARDOWN sequence and closes the used sockets */
+void gst_apexraop_close (GstApExRAOP * conn);
+
+/* once connected, set the apex target volume, manages SET_PARAMETER sequence */
+GstRTSPStatusCode gst_apexraop_set_volume (GstApExRAOP * conn,
+    const guint volume);
+
+/* write raw samples typed as defined by the fixed raop parameters, flush the apex buffer */
+guint gst_apexraop_write (GstApExRAOP * conn, gpointer rawdata, guint length);
+GstRTSPStatusCode gst_apexraop_flush (GstApExRAOP * conn);
+
+/* retrieve the connected apex jack type and status */
+GstApExJackType gst_apexraop_get_jacktype (GstApExRAOP * conn);
+GstApExJackStatus gst_apexraop_get_jackstatus (GstApExRAOP * conn);
+
+G_END_DECLS
+
+#endif
+
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;
+}
diff --git a/ext/apexsink/gstapexsink.h b/ext/apexsink/gstapexsink.h
new file mode 100644
index 00000000..11d2f1ef
--- /dev/null
+++ b/ext/apexsink/gstapexsink.h
@@ -0,0 +1,85 @@
+/* GStreamer - AirPort Express (ApEx) 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 regarding the expected gstapexraop input format.
+ *
+ * Copyright (C) 2008 Jérémie Bernard [GRemi] <gremimail@gmail.com>
+ *
+ * gstapexsink.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_APEXSINK_H__
+#define __GST_APEXSINK_H__
+
+#include "gstapexraop.h"
+
+#include <gst/audio/gstaudiosink.h>
+#include <gst/interfaces/mixer.h>
+
+G_BEGIN_DECLS
+
+/* standard gstreamer macros */
+#define GST_TYPE_APEX_SINK           	 	(gst_apexsink_get_type())
+#define GST_APEX_SINK(obj)            		(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_APEX_SINK,GstApExSink))
+#define GST_APEX_SINK_CLASS(klass)    		(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_APEX_SINK,GstApExSinkClass))
+#define GST_IS_APEX_SINK(obj)         		(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_APEX_SINK))
+#define GST_IS_APEX_SINK_CLASS(klass) 		(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_APEX_SINK))
+#define GST_APEX_SINK_CAST(obj)       		((GstApExSink*)(obj))
+#define GST_APEX_SINK_NAME			"apexsink"
+#define GST_APEX_SINK_JACKTYPE_TYPE		(gst_apexsink_jacktype_get_type())
+#define GST_APEX_SINK_JACKSTATUS_TYPE		(gst_apexsink_jackstatus_get_type())
+/* ApEx classes declaration */
+typedef struct _GstApExSink GstApExSink;
+typedef struct _GstApExSinkClass GstApExSinkClass;
+
+struct _GstApExSink
+{
+  /* base definition */
+  GstAudioSink sink;
+
+  /* public read/write sink properties */
+  gchar *host;
+  guint port;
+  guint volume;
+
+  /* private attributes : latency time local copy, tracks list of the mixer interface */
+  guint64 latency_time;
+  GList *tracks;
+
+  /* private apex client */
+  GstApExRAOP *gst_apexraop;
+};
+
+struct _GstApExSinkClass
+{
+  GstAudioSinkClass parent_class;
+};
+
+/* genum jack access */
+GType gst_apexsink_jackstatus_get_type (void);
+GType gst_apexsink_jacktype_get_type (void);
+
+/* audio sink standard api */
+GType gst_apexsink_get_type (void);
+
+G_END_DECLS
+
+#endif