mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-15 04:46:32 +00:00
299 lines
7.1 KiB
C
299 lines
7.1 KiB
C
/*
|
|
* 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.
|
|
*/
|
|
|
|
|
|
#define GLIB_DISABLE_DEPRECATION_WARNINGS
|
|
|
|
#include "gstsrtp.h"
|
|
|
|
#include <gst/rtp/gstrtcpbuffer.h>
|
|
|
|
#include "gstsrtpenc.h"
|
|
#include "gstsrtpdec.h"
|
|
|
|
#ifndef HAVE_SRTP2
|
|
srtp_err_status_t
|
|
srtp_set_stream_roc (srtp_t session, guint32 ssrc, guint32 roc)
|
|
{
|
|
srtp_stream_t stream;
|
|
|
|
stream = srtp_get_stream (session, htonl (ssrc));
|
|
if (stream == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
rdbx_set_roc (&stream->rtp_rdbx, roc);
|
|
return srtp_err_status_ok;
|
|
}
|
|
|
|
srtp_err_status_t
|
|
srtp_get_stream_roc (srtp_t session, guint32 ssrc, guint32 * roc)
|
|
{
|
|
srtp_stream_t stream;
|
|
|
|
stream = srtp_get_stream (session, htonl (ssrc));
|
|
if (stream == NULL) {
|
|
return srtp_err_status_bad_param;
|
|
}
|
|
|
|
*roc = stream->rtp_rdbx.index >> 16;
|
|
return srtp_err_status_ok;
|
|
}
|
|
#endif
|
|
|
|
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_free (data);
|
|
}
|
|
|
|
|
|
static void
|
|
srtp_event_reporter (srtp_event_data_t * data)
|
|
{
|
|
struct GstSrtpEventReporterData *dat = g_private_get (¤t_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 (¤t_callback);
|
|
|
|
if (!dat) {
|
|
dat = g_new (struct GstSrtpEventReporterData, 1);
|
|
g_private_set (¤t_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 (¤t_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)) {
|
|
GstRTCPType type;
|
|
do {
|
|
type = gst_rtcp_packet_get_type (&packet);
|
|
switch (type) {
|
|
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;
|
|
case GST_RTCP_TYPE_RTPFB:
|
|
case GST_RTCP_TYPE_PSFB:
|
|
*ssrc = gst_rtcp_packet_fb_get_sender_ssrc (&packet);
|
|
ret = TRUE;
|
|
break;
|
|
case GST_RTCP_TYPE_APP:
|
|
*ssrc = gst_rtcp_packet_app_get_ssrc (&packet);
|
|
ret = TRUE;
|
|
break;
|
|
case GST_RTCP_TYPE_BYE:
|
|
*ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, 0);
|
|
ret = TRUE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
} while ((ret == FALSE) && (type != GST_RTCP_TYPE_INVALID) &&
|
|
gst_rtcp_packet_move_to_next (&packet));
|
|
}
|
|
|
|
gst_rtcp_buffer_unmap (&rtcpbuf);
|
|
|
|
return ret;
|
|
}
|
|
|
|
void
|
|
set_crypto_policy_cipher_auth (GstSrtpCipherType cipher,
|
|
GstSrtpAuthType auth, srtp_crypto_policy_t * policy)
|
|
{
|
|
switch (cipher) {
|
|
case GST_SRTP_CIPHER_AES_128_ICM:
|
|
policy->cipher_type = SRTP_AES_ICM_128;
|
|
break;
|
|
case GST_SRTP_CIPHER_AES_256_ICM:
|
|
policy->cipher_type = SRTP_AES_ICM_256;
|
|
break;
|
|
case GST_SRTP_CIPHER_AES_128_GCM:
|
|
policy->cipher_type = SRTP_AES_GCM_128;
|
|
break;
|
|
case GST_SRTP_CIPHER_AES_256_GCM:
|
|
policy->cipher_type = SRTP_AES_GCM_256;
|
|
break;
|
|
case GST_SRTP_CIPHER_NULL:
|
|
policy->cipher_type = SRTP_NULL_CIPHER;
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
policy->cipher_key_len = cipher_key_size (cipher);
|
|
|
|
switch (auth) {
|
|
case GST_SRTP_AUTH_HMAC_SHA1_80:
|
|
policy->auth_type = SRTP_HMAC_SHA1;
|
|
policy->auth_key_len = 20;
|
|
policy->auth_tag_len = 10;
|
|
break;
|
|
case GST_SRTP_AUTH_HMAC_SHA1_32:
|
|
policy->auth_type = SRTP_HMAC_SHA1;
|
|
policy->auth_key_len = 20;
|
|
policy->auth_tag_len = 4;
|
|
break;
|
|
case GST_SRTP_AUTH_NULL:
|
|
policy->auth_type = SRTP_NULL_AUTH;
|
|
policy->auth_key_len = 0;
|
|
if (cipher == GST_SRTP_CIPHER_AES_128_GCM
|
|
|| cipher == GST_SRTP_CIPHER_AES_256_GCM) {
|
|
policy->auth_tag_len = 16;
|
|
} else {
|
|
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;
|
|
}
|
|
|
|
guint
|
|
cipher_key_size (GstSrtpCipherType cipher)
|
|
{
|
|
guint size = 0;
|
|
|
|
switch (cipher) {
|
|
case GST_SRTP_CIPHER_AES_128_ICM:
|
|
size = SRTP_AES_ICM_128_KEY_LEN_WSALT;
|
|
break;
|
|
case GST_SRTP_CIPHER_AES_256_ICM:
|
|
size = SRTP_AES_ICM_256_KEY_LEN_WSALT;
|
|
break;
|
|
case GST_SRTP_CIPHER_AES_128_GCM:
|
|
size = SRTP_AES_GCM_128_KEY_LEN_WSALT;
|
|
break;
|
|
case GST_SRTP_CIPHER_AES_256_GCM:
|
|
size = SRTP_AES_GCM_256_KEY_LEN_WSALT;
|
|
break;
|
|
case GST_SRTP_CIPHER_NULL:
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
return size;
|
|
}
|