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srtp: Add SRTP plugin based on libsrtp

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In large part the work of Gabriel Millaire
This commit is contained in:
Olivier Crête 2013-03-12 19:47:51 -04:00
parent 71768daceb
commit 4eb960d6c3
10 changed files with 2663 additions and 0 deletions
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12
configure.ac
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@ -1393,6 +1393,16 @@ AG_GST_CHECK_FEATURE(LIBMMS, [mms protocol library], libmms, [
])
AC_SUBST(LIBMMS_LIBS)
dnl *** libsrtp ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SRTP, true)
AG_GST_CHECK_FEATURE(SRTP, [srtp library], srtp, [
HAVE_SRTP="yes"
AG_GST_CHECK_LIBHEADER(SRTP, srtp, srtp_init, , srtp/srtp.h,
SRTP_LIBS="-lsrtp", HAVE_SRTP="no")
AC_SUBST(SRTP_LIBS)
])
dnl *** linsys ***
translit(dnm, m, l) AM_CONDITIONAL(USE_LINSYS, true)
AG_GST_CHECK_FEATURE(LINSYS, [Linear Systems SDI plugin], linsys, [
@ -2172,6 +2182,7 @@ AM_CONDITIONAL(USE_SNDFILE, false)
AM_CONDITIONAL(USE_SOUNDTOUCH, false)
AM_CONDITIONAL(USE_SPANDSP, false)
AM_CONDITIONAL(USE_SPC, false)
AM_CONDITIONAL(USE_SRTP, false)
AM_CONDITIONAL(USE_GME, false)
AM_CONDITIONAL(USE_GSETTINGS, false)
AM_CONDITIONAL(USE_XVID, false)
@ -2431,6 +2442,7 @@ ext/sndfile/Makefile
ext/soundtouch/Makefile
ext/spandsp/Makefile
ext/sndio/Makefile
ext/srtp/Makefile
ext/teletextdec/Makefile
ext/gme/Makefile
ext/gsettings/Makefile

8
ext/Makefile.am
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@ -318,6 +318,12 @@ else
SPC_DIR=
endif
if USE_SRTP
SRTP_DIR=srtp
else
SRTP_DIR=
endif
if USE_GME
GME_DIR=gme
else
@ -409,6 +415,7 @@ SUBDIRS=\
$(SPANDSP_DIR) \
$(GME_DIR) \
$(SPC_DIR) \
$(SRTP_DIR) \
$(TELETEXTDEC_DIR) \
$(TIMIDITY_DIR) \
$(XVID_DIR) \
@ -462,6 +469,7 @@ DIST_SUBDIRS = \
soundtouch \
spandsp \
spc \
srtp \
gme \
teletextdec \
timidity \

1
ext/srtp/.gitignore vendored Normal file
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@ -0,0 +1 @@
gstsrtp-enumtypes.[ch]

33
ext/srtp/Makefile.am Normal file
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@ -0,0 +1,33 @@
plugin_LTLIBRARIES = libgstsrtp.la
libgstsrtp_la_SOURCES = \
gstsrtp.c \
gstsrtp.h \
gstsrtpdec.c \
gstsrtpdec.h \
gstsrtpenc.c \
gstsrtpenc.h \
gstsrtp-enumtypes.c \
gstsrtp-enumtypes.h
libgstsrtp_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(SRTP_CFLAGS)
libgstsrtp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
-lgstrtp-@GST_API_VERSION@ \
$(GST_LIBS) \
$(SRTP_LIBS)
libgstsrtp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstsrtp_la_LIBTOOLFLAGS = --tag=disable-static
srtp_headers = \
gstsrtpdec.h \
gstsrtpenc.h
glib_enum_headers=$(srtp_headers)
glib_enum_define=GST_SRTP
glib_gen_prefix=gst_srtp
glib_gen_basename=gstsrtp
BUILT_SOURCES = gstsrtp-enumtypes.c gstsrtp-enumtypes.h
CLEANFILES = $(BUILT_SOURCES)
include $(top_srcdir)/common/gst-glib-gen.mak

233
ext/srtp/gstsrtp.c Normal file
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@ -0,0 +1,233 @@
/*
* GStreamer - GStreamer SRTP encoder and decoder
*
* Copyright 2009-2013 Collabora Ltd.
* @author: Gabriel Millaire <gabriel.millaire@collabora.co.uk>
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* 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
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include "gstsrtp.h"
#include <glib.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "gstsrtpenc.h"
#include "gstsrtpdec.h"
static void free_reporter_data (gpointer data);
GPrivate current_callback = G_PRIVATE_INIT (free_reporter_data);
struct GstSrtpEventReporterData
{
gboolean soft_limit_reached;
};
static void
free_reporter_data (gpointer data)
{
g_slice_free (struct GstSrtpEventReporterData, data);
}
static void
srtp_event_reporter (srtp_event_data_t * data)
{
struct GstSrtpEventReporterData *dat = g_private_get (&current_callback);
if (!dat)
return;
switch (data->event) {
case event_key_soft_limit:
dat->soft_limit_reached = TRUE;
break;
default:
break;
}
}
void
gst_srtp_init_event_reporter (void)
{
struct GstSrtpEventReporterData *dat = g_private_get (&current_callback);
if (!dat) {
dat = g_slice_new (struct GstSrtpEventReporterData);
g_private_set (&current_callback, dat);
}
dat->soft_limit_reached = FALSE;
srtp_install_event_handler (srtp_event_reporter);
}
const gchar *
enum_nick_from_value (GType enum_gtype, gint value)
{
GEnumClass *enum_class = g_type_class_ref (enum_gtype);
GEnumValue *enum_value;
const gchar *nick;
if (!enum_gtype)
return NULL;
enum_value = g_enum_get_value (enum_class, value);
nick = enum_value->value_nick;
g_type_class_unref (enum_class);
return nick;
}
gint
enum_value_from_nick (GType enum_gtype, const gchar * nick)
{
GEnumClass *enum_class = g_type_class_ref (enum_gtype);
GEnumValue *enum_value;
gint value;
if (!enum_gtype)
return 0;
enum_value = g_enum_get_value_by_nick (enum_class, nick);
value = enum_value->value;
g_type_class_unref (enum_class);
return value;
}
gboolean
gst_srtp_get_soft_limit_reached (void)
{
struct GstSrtpEventReporterData *dat = g_private_get (&current_callback);
if (dat)
return dat->soft_limit_reached;
return FALSE;
}
/* Get SSRC from RTCP buffer
*/
gboolean
rtcp_buffer_get_ssrc (GstBuffer * buf, guint32 * ssrc)
{
gboolean ret = FALSE;
GstRTCPBuffer rtcpbuf = GST_RTCP_BUFFER_INIT;
GstRTCPPacket packet;
/* Get SSRC from RR or SR packet (RTCP) */
if (!gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcpbuf))
return FALSE;
if (gst_rtcp_buffer_get_first_packet (&rtcpbuf, &packet)) {
do {
switch (gst_rtcp_packet_get_type (&packet)) {
case GST_RTCP_TYPE_RR:
*ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
ret = TRUE;
break;
case GST_RTCP_TYPE_SR:
gst_rtcp_packet_sr_get_sender_info (&packet, ssrc, NULL, NULL, NULL,
NULL);
ret = TRUE;
break;
default:
break;
}
} while (gst_rtcp_packet_move_to_next (&packet) && ret == FALSE);
}
gst_rtcp_buffer_unmap (&rtcpbuf);
return ret;
}
void
set_crypto_policy_cipher_auth (guint cipher, guint auth,
crypto_policy_t * policy)
{
switch (cipher) {
case GST_SRTP_CIPHER_AES_128_ICM:
policy->cipher_type = AES_ICM;
policy->cipher_key_len = 30;
break;
case GST_SRTP_CIPHER_NULL:
policy->cipher_type = NULL_CIPHER;
policy->cipher_key_len = 0;
break;
default:
g_assert_not_reached ();
}
switch (auth) {
case GST_SRTP_AUTH_HMAC_SHA1_80:
policy->auth_type = HMAC_SHA1;
policy->auth_key_len = 20;
policy->auth_tag_len = 10;
break;
case GST_SRTP_AUTH_HMAC_SHA1_32:
policy->auth_type = HMAC_SHA1;
policy->auth_key_len = 20;
policy->auth_tag_len = 4;
break;
case GST_SRTP_AUTH_NULL:
policy->auth_type = NULL_AUTH;
policy->auth_key_len = 0;
policy->auth_tag_len = 0;
break;
}
if (cipher == GST_SRTP_CIPHER_NULL && auth == GST_SRTP_AUTH_NULL)
policy->sec_serv = sec_serv_none;
else if (cipher == GST_SRTP_CIPHER_NULL)
policy->sec_serv = sec_serv_auth;
else if (auth == GST_SRTP_AUTH_NULL)
policy->sec_serv = sec_serv_conf;
else
policy->sec_serv = sec_serv_conf_and_auth;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
srtp_init ();
if (!gst_srtp_enc_plugin_init (plugin))
return FALSE;
if (!gst_srtp_dec_plugin_init (plugin))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
srtp,
"GStreamer SRTP",
plugin_init, VERSION, "LGPL", "GStreamer", "http://gstreamer.net/")

64
ext/srtp/gstsrtp.h Normal file
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@ -0,0 +1,64 @@
/*
* GStreamer - GStreamer SRTP encoder
*
* Copyright 2011-2013 Collabora Ltd.
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_SRTP_H__
#define __GST_SRTP_H__
#include <gst/gst.h>
#include <srtp/srtp.h>
void gst_srtp_init_event_reporter (void);
gboolean gst_srtp_get_soft_limit_reached (void);
gboolean rtcp_buffer_get_ssrc (GstBuffer * buf, guint32 * ssrc);
const gchar *enum_nick_from_value (GType enum_gtype, gint value);
gint enum_value_from_nick (GType enum_gtype, const gchar *nick);
void set_crypto_policy_cipher_auth (guint cipher, guint auth,
crypto_policy_t * policy);
#endif /* __GST_SRTP_H__ */

965
ext/srtp/gstsrtpdec.c Normal file
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@ -0,0 +1,965 @@
/*
* GStreamer - GStreamer SRTP decoder
*
* Copyright 2009-2011 Collabora Ltd.
* @author: Gabriel Millaire <gabriel.millaire@collabora.co.uk>
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-srtpdec
*
* gstrtpdec acts as a decoder that removes security from SRTP and SRTCP
* packets (encryption and authentication) and out RTP and RTCP. It
* receives packet of type 'application/x-srtp' or 'application/x-srtcp'
* on its sink pad, and outs packets of type 'application/x-rtp' or
* 'application/x-rtcp' on its sink pad.
*
* For each packet received, it checks if the internal SSRC is in the list
* of streams already in use. If this is not the case, it sends a signal to
* the user to get the needed parameters to create a new stream : master
* key, encryption and authentication mecanisms for both RTP and RTCP. If
* the user can't provide those parameters, the buffer is dropped and a
* warning is emitted.
*
* This element uses libsrtp library. The encryption and authentication
* mecanisms available are :
*
* Encryption
* - AES_128_ICM (default, maximum security)
* - STRONGHOLD_CIPHER (same as AES_128_ICM)
* - NULL
*
* Authentication
* - HMAC_SHA1 (default, maximum protection)
* - STRONGHOLD_AUTH (same as HMAC_SHA1)
* - NULL
*
* Note that for SRTP protection, authentication is mandatory (non-null)
* if encryption is used (non-null).
*
* Each packet received is first analysed (checked for valid SSRC) then
* its buffer is unprotected with libsrtp, then pushed on the source pad.
* If protection failed or the stream could not be created, the buffer
* is dropped and a warning is emitted.
*
* When the maximum usage of the master key is reached, a soft-limit
* signal is sent to the user, and new parameters (master key) are needed
* in return. If the hard limit is reached, a flag is set and every
* subsequent packet is dropped, until a new key is set and the stream
* has been updated.
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch-0.10 udpsrc port=33333 caps="application/x-rtp,mkey=(string)bafbafbaf,..." ! rtpspeexdepay ! speexdec ! alsasink
* ]| Receive SPEEX SRTP or SRTCP packets through UDP using caps to specify
* master key and protection. It outs RTP or SRTP packets.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/gst.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include "gstsrtp.h"
#include "gstsrtp-enumtypes.h"
#include "gstsrtpdec.h"
GST_DEBUG_CATEGORY_STATIC (gst_srtp_dec_debug);
#define GST_CAT_DEFAULT gst_srtp_dec_debug
/* Filter signals and args */
enum
{
SIGNAL_REQUEST_KEY = 1,
SIGNAL_CLEAR_KEYS,
SIGNAL_SOFT_LIMIT,
SIGNAL_HARD_LIMIT,
LAST_SIGNAL
};
enum
{
PROP_0
};
/* the capabilities of the inputs and outputs.
*
* describe the real formats here.
*/
static GstStaticPadTemplate rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtp_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-srtp")
);
static GstStaticPadTemplate rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtp_src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-srtcp")
);
static GstStaticPadTemplate rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtcp")
);
static guint gst_srtp_dec_signals[LAST_SIGNAL] = { 0 };
G_DEFINE_TYPE (GstSrtpDec, gst_srtp_dec, GST_TYPE_ELEMENT);
static void gst_srtp_dec_clear_streams (GstSrtpDec * filter);
static gboolean gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_srtp_dec_sink_query_rtp (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_srtp_dec_sink_query_rtcp (GstPad * pad,
GstObject * parent, GstQuery * query);
static GstIterator *gst_srtp_dec_iterate_internal_links_rtp (GstPad * pad,
GstObject * parent);
static GstIterator *gst_srtp_dec_iterate_internal_links_rtcp (GstPad * pad,
GstObject * parent);
static GstFlowReturn gst_srtp_dec_chain_rtp (GstPad * pad,
GstObject * parent, GstBuffer * buf);
static GstFlowReturn gst_srtp_dec_chain_rtcp (GstPad * pad,
GstObject * parent, GstBuffer * buf);
static GstStateChangeReturn gst_srtp_dec_change_state (GstElement * element,
GstStateChange transition);
static GstSrtpDecSsrcStream *request_key_with_signal (GstSrtpDec * filter,
guint32 ssrc, gint signal);
struct _GstSrtpDecSsrcStream
{
guint32 ssrc;
GstCaps *caps;
GstBuffer *key;
guint rtp_cipher;
guint rtp_auth;
guint rtcp_cipher;
guint rtcp_auth;
};
/* initialize the srtpdec's class */
static void
gst_srtp_dec_class_init (GstSrtpDecClass * klass)
{
GstElementClass *gstelement_class;
gstelement_class = (GstElementClass *) klass;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtp_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtcp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtcp_sink_template));
gst_element_class_set_static_metadata (gstelement_class, "SRTP decoder",
"Filter/Network/SRTP",
"A SRTP and SRTCP decoder",
"Gabriel Millaire <millaire.gabriel@collabora.com>");
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_srtp_dec_change_state);
klass->clear_streams = GST_DEBUG_FUNCPTR (gst_srtp_dec_clear_streams);
/**
* GstSrtpDec::request-key:
* @gstsrtpdec: the element on which the signal is emitted
* @ssrc: The unique SSRC of the stream
*
* Signal emited to get the parameters relevant to stream
* with @ssrc. User should provide the key and the RTP and
* RTCP encryption ciphers and authentication, and return
* them wrapped in a GstCaps.
*/
gst_srtp_dec_signals[SIGNAL_REQUEST_KEY] =
g_signal_new ("request-key", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstSrtpDec::clear-keys:
* @gstsrtpdec: the element on which the signal is emitted
*
* Clear the internal list of streams
*/
gst_srtp_dec_signals[SIGNAL_CLEAR_KEYS] =
g_signal_new ("clear-keys", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (GstSrtpDecClass, clear_streams), NULL, NULL, NULL,
G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstSrtpDec::soft-limit:
* @gstsrtpdec: the element on which the signal is emitted
* @ssrc: The unique SSRC of the stream
*
* Signal emited when the stream with @ssrc has reached the
* soft limit of utilisation of it's master encryption key.
* User should provide a new key and new RTP and RTCP encryption
* ciphers and authentication, and return them wrapped in a
* GstCaps.
*/
gst_srtp_dec_signals[SIGNAL_SOFT_LIMIT] =
g_signal_new ("soft-limit", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstSrtpDec::hard-limit:
* @gstsrtpdec: the element on which the signal is emitted
* @ssrc: The unique SSRC of the stream
*
* Signal emited when the stream with @ssrc has reached the
* hard limit of utilisation of it's master encryption key.
* User should provide a new key and new RTP and RTCP encryption
* ciphers and authentication, and return them wrapped in a
* GstCaps. If user could not provide those parameters or signal
* is not answered, the buffers of this stream will be dropped.
*/
gst_srtp_dec_signals[SIGNAL_HARD_LIMIT] =
g_signal_new ("hard-limit", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
}
/* initialize the new element
* instantiate pads and add them to element
* set pad calback functions
* initialize instance structure
*/
static void
gst_srtp_dec_init (GstSrtpDec * filter)
{
filter->rtp_sinkpad =
gst_pad_new_from_static_template (&rtp_sink_template, "rtp_sink");
gst_pad_set_event_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_event_rtp));
gst_pad_set_query_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_query_rtp));
gst_pad_set_iterate_internal_links_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtp));
gst_pad_set_chain_function (filter->rtp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtp));
filter->rtp_srcpad =
gst_pad_new_from_static_template (&rtp_src_template, "rtp_src");
gst_pad_set_iterate_internal_links_function (filter->rtp_srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtp));
gst_pad_set_element_private (filter->rtp_sinkpad, filter->rtp_srcpad);
gst_pad_set_element_private (filter->rtp_srcpad, filter->rtp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtp_srcpad);
filter->rtcp_sinkpad =
gst_pad_new_from_static_template (&rtcp_sink_template, "rtcp_sink");
gst_pad_set_event_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_event_rtcp));
gst_pad_set_query_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_sink_query_rtcp));
gst_pad_set_iterate_internal_links_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtcp));
gst_pad_set_chain_function (filter->rtcp_sinkpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_chain_rtcp));
filter->rtcp_srcpad =
gst_pad_new_from_static_template (&rtcp_src_template, "rtcp_src");
gst_pad_set_iterate_internal_links_function (filter->rtcp_srcpad,
GST_DEBUG_FUNCPTR (gst_srtp_dec_iterate_internal_links_rtcp));
gst_pad_set_element_private (filter->rtcp_sinkpad, filter->rtcp_srcpad);
gst_pad_set_element_private (filter->rtcp_srcpad, filter->rtcp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtcp_sinkpad);
gst_element_add_pad (GST_ELEMENT (filter), filter->rtcp_srcpad);
filter->first_session = TRUE;
}
static void
remove_stream_by_ssrc (GstSrtpDec * filter, guint32 ssrc)
{
GstSrtpDecSsrcStream *stream = NULL;
stream = g_hash_table_lookup (filter->streams, GUINT_TO_POINTER (ssrc));
if (stream) {
srtp_remove_stream (filter->session, ssrc);
g_hash_table_remove (filter->streams, GUINT_TO_POINTER (ssrc));
}
}
static GstSrtpDecSsrcStream *
find_stream_by_ssrc (GstSrtpDec * filter, guint32 ssrc)
{
return g_hash_table_lookup (filter->streams, GUINT_TO_POINTER (ssrc));
}
/* get info from buffer caps
*/
static GstSrtpDecSsrcStream *
get_stream_from_caps (GstSrtpDec * filter, GstCaps * caps, guint32 ssrc)
{
GstSrtpDecSsrcStream *stream;
GstStructure *s;
GstBuffer *buf;
const gchar *rtp_cipher, *rtp_auth, *rtcp_cipher, *rtcp_auth;
/* Create new stream structure and set default values */
stream = g_slice_new0 (GstSrtpDecSsrcStream);
stream->ssrc = ssrc;
stream->key = NULL;
/* Get info from caps */
s = gst_caps_get_structure (caps, 0);
if (!s)
goto error;
rtp_cipher = gst_structure_get_string (s, "srtp-cipher");
rtp_auth = gst_structure_get_string (s, "srtp-auth");
rtcp_cipher = gst_structure_get_string (s, "srtcp-cipher");
rtcp_auth = gst_structure_get_string (s, "srtcp-auth");
if (!rtp_cipher || !rtp_auth || !rtcp_cipher || !rtcp_auth)
goto error;
if (!gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL) || !buf) {
goto error;
} else {
GST_DEBUG ("Got key [%p]", buf);
stream->key = buf;
}
stream->rtp_cipher = enum_value_from_nick (GST_TYPE_SRTP_CIPHER_TYPE,
rtp_cipher);
stream->rtp_auth = enum_value_from_nick (GST_TYPE_SRTP_AUTH_TYPE, rtp_auth);
stream->rtcp_cipher = enum_value_from_nick (GST_TYPE_SRTP_CIPHER_TYPE,
rtcp_cipher);
stream->rtcp_auth = enum_value_from_nick (GST_TYPE_SRTP_AUTH_TYPE, rtcp_auth);
if (stream->rtcp_cipher != NULL_CIPHER && stream->rtcp_auth == NULL_AUTH) {
GST_WARNING_OBJECT (filter,
"Cannot have SRTP NULL authentication with a not-NULL encryption"
" cipher.");
goto error;
}
return stream;
error:
g_slice_free (GstSrtpDecSsrcStream, stream);
return NULL;
}
/* Get SRTP params by signal
*/
static GstCaps *
signal_get_srtp_params (GstSrtpDec * filter, guint32 ssrc, gint signal)
{
GstCaps *caps = NULL;
g_signal_emit (filter, gst_srtp_dec_signals[signal], 0, ssrc, &caps);
if (caps != NULL)
GST_DEBUG_OBJECT (filter, "Caps received");
return caps;
}
/* Create a stream in the session
*/
static err_status_t
init_session_stream (GstSrtpDec * filter, guint32 ssrc,
GstSrtpDecSsrcStream * stream)
{
err_status_t ret;
srtp_policy_t policy;
GstMapInfo map;
memset (&policy, 0, sizeof (srtp_policy_t));
if (!stream)
return err_status_bad_param;
GST_INFO_OBJECT (filter, "Setting RTP policy...");
set_crypto_policy_cipher_auth (stream->rtp_cipher, stream->rtp_auth,
&policy.rtp);
GST_INFO_OBJECT (filter, "Setting RTCP policy...");
set_crypto_policy_cipher_auth (stream->rtcp_cipher, stream->rtcp_auth,
&policy.rtcp);
gst_buffer_map (stream->key, &map, GST_MAP_READ);
policy.ssrc.value = ssrc;
policy.ssrc.type = ssrc_specific;
policy.key = (guchar *) map.data;
policy.next = NULL;
/* If it is the first stream, create the session
* If not, add the stream policy to the session
*/
if (filter->first_session)
ret = srtp_create (&filter->session, &policy);
else
ret = srtp_add_stream (filter->session, &policy);
gst_buffer_unmap (stream->key, &map);
if (ret == err_status_ok) {
filter->first_session = FALSE;
g_hash_table_insert (filter->streams, GUINT_TO_POINTER (stream->ssrc),
stream);
}
return ret;
}
/* Return a stream structure for a given buffer
*/
static GstSrtpDecSsrcStream *
validate_buffer (GstSrtpDec * filter, GstBuffer * buf, guint32 * ssrc,
gboolean is_rtcp)
{
GstSrtpDecSsrcStream *stream = NULL;
if (!is_rtcp) {
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
if (!gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuf)) {
GST_WARNING_OBJECT (filter, "Invalid SRTP packet");
return NULL;
}
*ssrc = gst_rtp_buffer_get_ssrc (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
} else if (!rtcp_buffer_get_ssrc (buf, ssrc)) {
GST_WARNING_OBJECT (filter, "No SSRC found in buffer");
return NULL;
}
stream = find_stream_by_ssrc (filter, *ssrc);
if (stream)
return stream;
return request_key_with_signal (filter, *ssrc, SIGNAL_REQUEST_KEY);
}
/* Create new stream from params in caps
*/
static GstSrtpDecSsrcStream *
update_session_stream_from_caps (GstSrtpDec * filter, guint32 ssrc,
GstCaps * caps)
{
GstSrtpDecSsrcStream *stream = NULL;
err_status_t err;
g_return_val_if_fail (GST_IS_SRTP_DEC (filter), NULL);
g_return_val_if_fail (GST_IS_CAPS (caps), NULL);
/* Remove existing stream, if any */
remove_stream_by_ssrc (filter, ssrc);
stream = get_stream_from_caps (filter, caps, ssrc);
if (stream) {
/* Create new session stream */
err = init_session_stream (filter, ssrc, stream);
if (err != err_status_ok) {
gst_buffer_unref (stream->key);
g_slice_free (GstSrtpDecSsrcStream, stream);
stream = NULL;
}
}
return stream;
}
static void
clear_stream (GstSrtpDecSsrcStream * stream)
{
gst_buffer_unref (stream->key);
g_slice_free (GstSrtpDecSsrcStream, stream);
}
/* Clear the policy list
*/
static void
gst_srtp_dec_clear_streams (GstSrtpDec * filter)
{
guint nb = 0;
GST_OBJECT_LOCK (filter);
if (!filter->first_session)
srtp_dealloc (filter->session);
nb = g_hash_table_size (filter->streams);
g_hash_table_destroy (filter->streams);
filter->streams = NULL;
filter->first_session = TRUE;
GST_OBJECT_UNLOCK (filter);
GST_DEBUG_OBJECT (filter, "Cleared %d streams", nb);
}
/* Send a signal
*/
static GstSrtpDecSsrcStream *
request_key_with_signal (GstSrtpDec * filter, guint32 ssrc, gint signal)
{
GstCaps *caps;
GstSrtpDecSsrcStream *stream = NULL;
caps = signal_get_srtp_params (filter, ssrc, signal);
if (caps) {
GstSrtpDecSsrcStream *stream =
update_session_stream_from_caps (filter, ssrc, caps);
if (stream)
GST_DEBUG_OBJECT (filter, "New stream set with SSRC %d", ssrc);
else
GST_WARNING_OBJECT (filter, "Could not set stream with SSRC %d", ssrc);
gst_caps_unref (caps);
}
return stream;
}
static gboolean
gst_srtp_dec_sink_setcaps (GstPad * pad, GstObject * parent,
GstCaps * caps, gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad;
GstStructure *ps;
gboolean ret = FALSE;
g_return_val_if_fail (gst_caps_is_fixed (caps), FALSE);
ps = gst_caps_get_structure (caps, 0);
if (gst_structure_has_field_typed (ps, "ssrc", G_TYPE_UINT) &&
gst_structure_has_field_typed (ps, "srtp-key", GST_TYPE_BUFFER) &&
gst_structure_has_field_typed (ps, "srtp-cipher", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtp-auth", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtcp-cipher", G_TYPE_STRING) &&
gst_structure_has_field_typed (ps, "srtcp-auth", G_TYPE_STRING)) {
guint ssrc;
gst_structure_get_uint (ps, "ssrc", &ssrc);
if (!update_session_stream_from_caps (filter, ssrc, caps)) {
GST_WARNING_OBJECT (pad, "Could not create session from pad caps: %"
GST_PTR_FORMAT, caps);
return FALSE;
}
}
caps = gst_caps_copy (caps);
ps = gst_caps_get_structure (caps, 0);
gst_structure_remove_fields (ps, "srtp-key", "srtp-cipher", "srtp-auth",
"srtcp-cipher", "srtcp-auth", NULL);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-rtcp");
else
gst_structure_set_name (ps, "application/x-rtp");
otherpad = gst_pad_get_element_private (pad);
ret = gst_pad_set_caps (otherpad, caps);
gst_caps_unref (caps);
return ret;
}
static gboolean
gst_srtp_dec_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event, gboolean is_rtcp)
{
GstCaps *caps;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
gst_event_parse_caps (event, &caps);
return gst_srtp_dec_sink_setcaps (pad, parent, caps, FALSE);
default:
return gst_pad_event_default (pad, parent, event);
}
}
static gboolean
gst_srtp_dec_sink_event_rtp (GstPad * pad, GstObject * parent, GstEvent * event)
{
return gst_srtp_dec_sink_event (pad, parent, event, FALSE);
}
static gboolean
gst_srtp_dec_sink_event_rtcp (GstPad * pad, GstObject * parent,
GstEvent * event)
{
return gst_srtp_dec_sink_event (pad, parent, event, TRUE);
}
static gboolean
gst_srtp_dec_sink_query (GstPad * pad, GstObject * parent, GstQuery * query,
gboolean is_rtcp)
{
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter = NULL;
GstCaps *other_filter = NULL;
GstCaps *template_caps;
GstPad *otherpad;
GstCaps *other_caps;
GstCaps *ret;
int i;
gst_query_parse_caps (query, &filter);
otherpad = (GstPad *) gst_pad_get_element_private (pad);
if (filter) {
other_filter = gst_caps_copy (filter);
for (i = 0; i < gst_caps_get_size (other_filter); i++) {
GstStructure *ps = gst_caps_get_structure (other_filter, i);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-rtcp");
else
gst_structure_set_name (ps, "application/x-rtp");
gst_structure_remove_fields (ps, "srtp-key", "srtp-cipher",
"srtp-auth", "srtcp-cipher", "srtcp-auth", NULL);
}
}
other_caps = gst_pad_peer_query_caps (otherpad, other_filter);
if (other_filter)
gst_caps_unref (other_filter);
if (!other_caps) {
goto return_template;
}
template_caps = gst_pad_get_pad_template_caps (otherpad);
ret = gst_caps_intersect_full (other_caps, template_caps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (other_caps);
gst_caps_unref (template_caps);
ret = gst_caps_make_writable (ret);
for (i = 0; i < gst_caps_get_size (ret); i++) {
GstStructure *ps = gst_caps_get_structure (ret, i);
if (is_rtcp)
gst_structure_set_name (ps, "application/x-srtcp");
else
gst_structure_set_name (ps, "application/x-srtp");
}
gst_query_set_caps_result (query, ret);
return TRUE;
return_template:
ret = gst_pad_get_pad_template_caps (pad);
gst_query_set_caps_result (query, ret);
gst_caps_unref (ret);
return TRUE;
}
default:
return gst_pad_query_default (pad, parent, query);
}
}
static gboolean
gst_srtp_dec_sink_query_rtp (GstPad * pad, GstObject * parent, GstQuery * query)
{
return gst_srtp_dec_sink_query (pad, parent, query, FALSE);
}
static gboolean
gst_srtp_dec_sink_query_rtcp (GstPad * pad, GstObject * parent,
GstQuery * query)
{
return gst_srtp_dec_sink_query (pad, parent, query, TRUE);
}
static GstIterator *
gst_srtp_dec_iterate_internal_links (GstPad * pad, GstObject * parent,
gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad = NULL;
GstIterator *it = NULL;
otherpad = (GstPad *) gst_pad_get_element_private (pad);
if (otherpad) {
GValue val = { 0 };
g_value_init (&val, GST_TYPE_PAD);
g_value_set_object (&val, otherpad);
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
g_value_unset (&val);
} else {
GST_ELEMENT_ERROR (GST_ELEMENT_CAST (filter), CORE, PAD, (NULL),
("Unable to get linked pad"));
}
return it;
}
static GstIterator *
gst_srtp_dec_iterate_internal_links_rtp (GstPad * pad, GstObject * parent)
{
return gst_srtp_dec_iterate_internal_links (pad, parent, FALSE);
}
static GstIterator *
gst_srtp_dec_iterate_internal_links_rtcp (GstPad * pad, GstObject * parent)
{
return gst_srtp_dec_iterate_internal_links (pad, parent, TRUE);
}
static GstFlowReturn
gst_srtp_dec_chain (GstPad * pad, GstObject * parent, GstBuffer * buf,
gboolean is_rtcp)
{
GstSrtpDec *filter = GST_SRTP_DEC (parent);
GstPad *otherpad;
err_status_t err = err_status_ok;
GstSrtpDecSsrcStream *stream = NULL;
GstFlowReturn ret = GST_FLOW_OK;
gint size;
guint32 ssrc = 0;
GstMapInfo map;
GST_OBJECT_LOCK (filter);
/* Check if this stream exists, if not create a new stream */
if (!(stream = validate_buffer (filter, buf, &ssrc, is_rtcp))) {
GST_OBJECT_UNLOCK (filter);
goto drop_buffer;
}
GST_LOG_OBJECT (pad, "Received %s buffer of size %d with SSRC = %u",
is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf), ssrc);
/* Change buffer to remove protection */
buf = gst_buffer_make_writable (buf);
unprotect:
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
size = map.size;
gst_srtp_init_event_reporter ();
if (is_rtcp)
err = srtp_unprotect_rtcp (filter->session, map.data, &size);
else
err = srtp_unprotect (filter->session, map.data, &size);
gst_buffer_unmap (buf, &map);
GST_OBJECT_UNLOCK (filter);
if (err == err_status_ok) {
gst_buffer_set_size (buf, size);
otherpad = (GstPad *) gst_pad_get_element_private (pad);
/* If all is well, we may have reached soft limit */
if (gst_srtp_get_soft_limit_reached ())
request_key_with_signal (filter, ssrc, SIGNAL_SOFT_LIMIT);
/* Push buffer to source pad */
ret = gst_pad_push (otherpad, buf);
} else { /* srtp_unprotect failed */
GST_WARNING_OBJECT (pad,
"Unable to unprotect buffer (unprotect failed code %d)", err);
/* Signal user depending on type of error */
switch (err) {
case err_status_key_expired:
GST_OBJECT_LOCK (filter);
/* Update stream */
if ((stream = find_stream_by_ssrc (filter, ssrc))) {
GST_OBJECT_UNLOCK (filter);
if (request_key_with_signal (filter, ssrc, SIGNAL_HARD_LIMIT)) {
GST_OBJECT_LOCK (filter);
goto unprotect;
}
goto drop_buffer;
}
break;
case err_status_auth_fail:
case err_status_cipher_fail:
GST_ELEMENT_ERROR (filter, STREAM, DECRYPT,
("Error while decryption stream"), (NULL));
ret = GST_FLOW_ERROR;
goto drop_buffer;
default:
goto drop_buffer;
}
}
return ret;
/* Drop buffer, except if gst_pad_push returned OK or an error */
drop_buffer:
GST_WARNING_OBJECT (pad, "Dropping buffer");
gst_buffer_unref (buf);
return ret;
}
static GstFlowReturn
gst_srtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_dec_chain (pad, parent, buf, FALSE);
}
static GstFlowReturn
gst_srtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buf)
{
return gst_srtp_dec_chain (pad, parent, buf, TRUE);
}
static GstStateChangeReturn
gst_srtp_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstSrtpDec *filter;
filter = GST_SRTP_DEC (element);
GST_OBJECT_LOCK (filter);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!filter->first_session)
gst_srtp_dec_clear_streams (filter);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
filter->streams = g_hash_table_new_full (g_direct_hash, g_direct_equal,
NULL, (GDestroyNotify) clear_stream);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
default:
break;
}
GST_OBJECT_UNLOCK (filter);
res = GST_ELEMENT_CLASS (gst_srtp_dec_parent_class)->change_state (element,
transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_srtp_dec_clear_streams (filter);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
}
/* entry point to initialize the plug-in
* initialize the plug-in itself
* register the element factories and other features
*/
gboolean
gst_srtp_dec_plugin_init (GstPlugin * srtpdec)
{
GST_DEBUG_CATEGORY_INIT (gst_srtp_dec_debug, "srtpdec", 0, "SRTP dec");
return gst_element_register (srtpdec, "srtpdec", GST_RANK_NONE,
GST_TYPE_SRTP_DEC);
}

96
ext/srtp/gstsrtpdec.h Normal file
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@ -0,0 +1,96 @@
/*
* GStreamer - GStreamer SRTP decoder
*
* Copyright 2009 Collabora Ltd.
* @author: Gabriel Millaire <gabriel.millaire@collabora.co.uk>
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_SRTPDEC_H__
#define __GST_SRTPDEC_H__
#include <gst/gst.h>
#include <srtp/srtp.h>
G_BEGIN_DECLS
#define GST_TYPE_SRTP_DEC \
(gst_srtp_dec_get_type())
#define GST_SRTP_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SRTP_DEC,GstSrtpDec))
#define GST_SRTP_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SRTP_DEC,GstSrtpDecClass))
#define GST_IS_SRTP_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SRTP_DEC))
#define GST_IS_SRTP_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SRTP_DEC))
typedef struct _GstSrtpDec GstSrtpDec;
typedef struct _GstSrtpDecClass GstSrtpDecClass;
typedef struct _GstSrtpDecSsrcStream GstSrtpDecSsrcStream;
struct _GstSrtpDec
{
GstElement element;
GstPad *rtp_sinkpad, *rtp_srcpad;
GstPad *rtcp_sinkpad, *rtcp_srcpad;
gboolean ask_update;
srtp_t session;
gboolean first_session;
GHashTable *streams;
};
struct _GstSrtpDecClass
{
GstElementClass parent_class;
void (*clear_streams) (GstSrtpDec * filter);
};
GType gst_srtp_dec_get_type (void);
gboolean gst_srtp_dec_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_SRTPDEC_H__ */

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/*
* GStreamer - GStreamer SRTP encoder
*
* Copyright 2009 Collabora Ltd.
* @author: Gabriel Millaire <gabriel.millaire@collabora.co.uk>
* @author: Olivier Crete <olivier.crete@collabora.com>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_SRTPENC_H__
#define __GST_SRTPENC_H__
#include <gst/gst.h>
#include <srtp/srtp.h>
G_BEGIN_DECLS
#define GST_TYPE_SRTP_ENC \
(gst_srtp_enc_get_type())
#define GST_SRTP_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SRTP_ENC,GstSrtpEnc))
#define GST_SRTP_ENC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SRTP_ENC,GstSrtpEncClass))
#define GST_IS_SRTP_ENC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SRTP_ENC))
#define GST_IS_SRTP_ENC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SRTP_ENC))
typedef struct _GstSrtpEnc GstSrtpEnc;
typedef struct _GstSrtpEncClass GstSrtpEncClass;
typedef enum
{
GST_SRTP_CIPHER_NULL,
GST_SRTP_CIPHER_AES_128_ICM
} GstSrtpCipherType;
typedef enum
{
GST_SRTP_AUTH_NULL,
GST_SRTP_AUTH_HMAC_SHA1_32,
GST_SRTP_AUTH_HMAC_SHA1_80
} GstSrtpAuthType;
struct _GstSrtpEnc
{
GstElement element;
gboolean random_key;
GstBuffer *key;
guint rtp_cipher;
guint rtp_auth;
guint rtcp_cipher;
guint rtcp_auth;
srtp_t session;
gboolean first_session;
gboolean key_changed;
gboolean hard_limit_reached;
gboolean soft_limit_reached;
GHashTable *ssrcs_set;
GType key_type;
};
struct _GstSrtpEncClass
{
GstElementClass parent_class;
};
GType gst_srtp_enc_get_type (void);
gboolean gst_srtp_enc_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_SRTPENC_H__ */