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

Browse source 
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.
This commit is contained in:
Jérémie Bernard 2008-08-28 17:01:30 +00:00 committed by Sebastian Dröge
parent 53025584c3
commit a72dc6992e
9 changed files with 1847 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

35
ChangeLog
View file

@ -1,3 +1,38 @@
2008-08-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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.
2008-08-28 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),

125
configure.ac
View file

@ -346,6 +346,129 @@ AG_GST_CHECK_FEATURE(AMRWB, [amrwb library], amrwb, [
AC_SUBST(AMRWB_LIBS))
])
dnl *** apexsink ***
translit(dnm, m, l) AM_CONDITIONAL(USE_APEXSINK, true)
AG_GST_CHECK_FEATURE(APEXSINK, [AirPort Express Wireless sink], apexsink, [
HAVE_APEXSINK="yes"
# The big search for OpenSSL
# copied from openssh's configure.ac
AC_ARG_WITH(ssl-dir,
[ --with-ssl-dir=PATH Specify path to OpenSSL installation ],
[
if test "x$withval" != "xno" ; then
tryssldir=$withval
fi
]
)
AC_SEARCH_LIBS(socket, [socket])
AC_SEARCH_LIBS(gethostbyname, [nsl])
saved_LIBS="$LIBS"
saved_LDFLAGS="$LDFLAGS"
saved_CPPFLAGS="$CPPFLAGS"
if test "x$prefix" != "xNONE" ; then
tryssldir="$tryssldir $prefix"
fi
AC_CACHE_CHECK([for OpenSSL directory], ac_cv_openssldir, [
for ssldir in $tryssldir "" /usr/local/openssl /usr/lib/openssl /usr/local/ssl /usr/lib/ssl /usr/local /usr/athena /usr/pkg /opt /opt/openssl ; do
CPPFLAGS="$saved_CPPFLAGS"
LDFLAGS="$saved_LDFLAGS"
LIBS="$saved_LIBS -lssl -lcrypto"
# Skip directories if they don't exist
if test ! -z "$ssldir" -a ! -d "$ssldir" ; then
continue;
fi
if test ! -z "$ssldir" -a "x$ssldir" != "x/usr"; then
# Try to use $ssldir/lib if it exists, otherwise
# $ssldir
if test -d "$ssldir/lib" ; then
LDFLAGS="-L$ssldir/lib $saved_LDFLAGS"
if test ! -z "$need_dash_r" ; then
LDFLAGS="-R$ssldir/lib $LDFLAGS"
fi
else
LDFLAGS="-L$ssldir $saved_LDFLAGS"
if test ! -z "$need_dash_r" ; then
LDFLAGS="-R$ssldir $LDFLAGS"
fi
fi
# Try to use $ssldir/include if it exists, otherwise
# $ssldir
if test -d "$ssldir/include" ; then
CPPFLAGS="-I$ssldir/include $saved_CPPFLAGS"
else
CPPFLAGS="-I$ssldir $saved_CPPFLAGS"
fi
fi
# Basic test to check for compatible version and correct linking
# *does not* test for RSA - that comes later.
AC_TRY_RUN(
[
#include <string.h>
#include <openssl/rand.h>
int main(void)
{
char a[2048];
memset(a, 0, sizeof(a));
RAND_add(a, sizeof(a), sizeof(a));
return(RAND_status() <= 0);
}
],
[
found_crypto=1
break;
], []
)
if test ! -z "$found_crypto" ; then
break;
fi
done
if test -z "$found_crypto" ; then
HAVE_APEXSINK="no"
fi
if test -z "$ssldir" ; then
ssldir="(system)"
fi
ac_cv_openssldir=$ssldir
])
if (test ! -z "$ac_cv_openssldir" && test "x$ac_cv_openssldir" != "x(system)") ;
then
dnl Need to recover ssldir - test above runs in subshell
ssldir=$ac_cv_openssldir
if test ! -z "$ssldir" -a "x$ssldir" != "x/usr"; then
# Try to use $ssldir/lib if it exists, otherwise
# $ssldir
if test -d "$ssldir/lib" ; then
APEXSINK_LIBS="-L$ssldir/lib $saved_LDFLAGS"
if test ! -z "$need_dash_r" ; then
APEXSINK_LIBS="-R$ssldir/lib $LDFLAGS"
fi
else
APEXSINK_LDFLAGS="-L$ssldir $saved_LDFLAGS"
if test ! -z "$need_dash_r" ; then
APEXSINK_LIBS="-R$ssldir $LDFLAGS"
fi
fi
# Try to use $ssldir/include if it exists, otherwise
# $ssldir
if test -d "$ssldir/include" ; then
APEXSINK_CFLAGS="-I$ssldir/include $saved_CPPFLAGS"
else
APEXSINK_CFLAGS="-I$ssldir $saved_CPPFLAGS"
fi
fi
fi
APEXSINK_LIBS="$APEXSINK_LIBS $LIBS"
LIBS="$saved_LIBS"
])
dnl *** BZ2 ***
translit(dnm, m, l) AM_CONDITIONAL(USE_BZ2, true)
AG_GST_CHECK_FEATURE(BZ2, [bz2 library], bz2, [
@ -1146,6 +1269,7 @@ AM_CONDITIONAL(USE_QUICKTIME, false)
AM_CONDITIONAL(USE_VCD, false)
AM_CONDITIONAL(USE_ALSA, false)
AM_CONDITIONAL(USE_AMRWB, false)
AM_CONDITIONAL(USE_APEXSINK, false)
AM_CONDITIONAL(USE_BZ2, false)
AM_CONDITIONAL(USE_CDAUDIO, false)
AM_CONDITIONAL(USE_CELT, false)
@ -1308,6 +1432,7 @@ examples/directfb/Makefile
examples/switch/Makefile
ext/amrwb/Makefile
ext/alsaspdif/Makefile
ext/apexsink/Makefile
ext/bz2/Makefile
ext/cdaudio/Makefile
ext/celt/Makefile

165
ext/apexsink/LGPL-3.0.txt Normal file
View file

@ -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.
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
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1. Exception to Section 3 of the GNU GPL.
You may convey a covered work under sections 3 and 4 of this License
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2. Conveying Modified Versions.
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3. Object Code Incorporating Material from Library Header Files.
The object code form of an Application may incorporate material from
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6. Revised Versions of the GNU Lesser General Public License.
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Each version is given a distinguishing version number. If the
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8
ext/apexsink/Makefile.am Normal file
View file

@ -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

44
ext/apexsink/gstapexplugin.c Normal file
View file

@ -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)

690
ext/apexsink/gstapexraop.c Normal file
View file

@ -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;
}

124
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/* 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

571
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/* 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;
}

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/* 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