blob: 427db7cf5ccc48ce8ab72ceb925874ee1cd3a315 [file] [log] [blame]
/*
* 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);
if (!enum_value)
return NULL;
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 -1;
enum_value = g_enum_get_value_by_nick (enum_class, nick);
if (!enum_value)
return -1;
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 (GstSrtpCipherType cipher,
GstSrtpAuthType auth, crypto_policy_t * policy)
{
switch (cipher) {
case GST_SRTP_CIPHER_AES_128_ICM:
policy->cipher_type = AES_128_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/")