mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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299 lines
7.1 KiB
C
299 lines
7.1 KiB
C
/*
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* GStreamer - GStreamer SRTP encoder and decoder
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*
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* Copyright 2009-2013 Collabora Ltd.
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* @author: Gabriel Millaire <gabriel.millaire@collabora.co.uk>
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* @author: Olivier Crete <olivier.crete@collabora.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include "gstsrtp.h"
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#include <gst/rtp/gstrtcpbuffer.h>
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#include "gstsrtpenc.h"
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#include "gstsrtpdec.h"
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#ifndef HAVE_SRTP2
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srtp_err_status_t
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srtp_set_stream_roc (srtp_t session, guint32 ssrc, guint32 roc)
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{
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srtp_stream_t stream;
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stream = srtp_get_stream (session, htonl (ssrc));
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if (stream == NULL) {
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return srtp_err_status_bad_param;
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}
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rdbx_set_roc (&stream->rtp_rdbx, roc);
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return srtp_err_status_ok;
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}
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srtp_err_status_t
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srtp_get_stream_roc (srtp_t session, guint32 ssrc, guint32 * roc)
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{
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srtp_stream_t stream;
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stream = srtp_get_stream (session, htonl (ssrc));
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if (stream == NULL) {
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return srtp_err_status_bad_param;
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}
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*roc = stream->rtp_rdbx.index >> 16;
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return srtp_err_status_ok;
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}
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#endif
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static void free_reporter_data (gpointer data);
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GPrivate current_callback = G_PRIVATE_INIT (free_reporter_data);
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struct GstSrtpEventReporterData
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{
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gboolean soft_limit_reached;
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};
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static void
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free_reporter_data (gpointer data)
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{
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g_free (data);
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}
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static void
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srtp_event_reporter (srtp_event_data_t * data)
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{
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struct GstSrtpEventReporterData *dat = g_private_get (¤t_callback);
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if (!dat)
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return;
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switch (data->event) {
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case event_key_soft_limit:
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dat->soft_limit_reached = TRUE;
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break;
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default:
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break;
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}
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}
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void
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gst_srtp_init_event_reporter (void)
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{
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struct GstSrtpEventReporterData *dat = g_private_get (¤t_callback);
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if (!dat) {
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dat = g_new (struct GstSrtpEventReporterData, 1);
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g_private_set (¤t_callback, dat);
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}
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dat->soft_limit_reached = FALSE;
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srtp_install_event_handler (srtp_event_reporter);
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}
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const gchar *
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enum_nick_from_value (GType enum_gtype, gint value)
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{
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GEnumClass *enum_class = g_type_class_ref (enum_gtype);
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GEnumValue *enum_value;
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const gchar *nick;
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if (!enum_gtype)
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return NULL;
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enum_value = g_enum_get_value (enum_class, value);
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if (!enum_value)
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return NULL;
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nick = enum_value->value_nick;
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g_type_class_unref (enum_class);
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return nick;
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}
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gint
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enum_value_from_nick (GType enum_gtype, const gchar * nick)
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{
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GEnumClass *enum_class = g_type_class_ref (enum_gtype);
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GEnumValue *enum_value;
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gint value;
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if (!enum_gtype)
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return -1;
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enum_value = g_enum_get_value_by_nick (enum_class, nick);
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if (!enum_value)
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return -1;
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value = enum_value->value;
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g_type_class_unref (enum_class);
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return value;
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}
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gboolean
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gst_srtp_get_soft_limit_reached (void)
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{
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struct GstSrtpEventReporterData *dat = g_private_get (¤t_callback);
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if (dat)
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return dat->soft_limit_reached;
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return FALSE;
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}
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/* Get SSRC from RTCP buffer
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*/
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gboolean
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rtcp_buffer_get_ssrc (GstBuffer * buf, guint32 * ssrc)
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{
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gboolean ret = FALSE;
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GstRTCPBuffer rtcpbuf = GST_RTCP_BUFFER_INIT;
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GstRTCPPacket packet;
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/* Get SSRC from RR or SR packet (RTCP) */
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if (!gst_rtcp_buffer_map (buf, GST_MAP_READ, &rtcpbuf))
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return FALSE;
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if (gst_rtcp_buffer_get_first_packet (&rtcpbuf, &packet)) {
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GstRTCPType type;
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do {
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type = gst_rtcp_packet_get_type (&packet);
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switch (type) {
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case GST_RTCP_TYPE_RR:
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*ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
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ret = TRUE;
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break;
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case GST_RTCP_TYPE_SR:
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gst_rtcp_packet_sr_get_sender_info (&packet, ssrc, NULL, NULL, NULL,
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NULL);
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ret = TRUE;
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break;
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case GST_RTCP_TYPE_RTPFB:
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case GST_RTCP_TYPE_PSFB:
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*ssrc = gst_rtcp_packet_fb_get_sender_ssrc (&packet);
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ret = TRUE;
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break;
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case GST_RTCP_TYPE_APP:
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*ssrc = gst_rtcp_packet_app_get_ssrc (&packet);
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ret = TRUE;
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break;
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case GST_RTCP_TYPE_BYE:
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*ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, 0);
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ret = TRUE;
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break;
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default:
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break;
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}
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} while ((ret == FALSE) && (type != GST_RTCP_TYPE_INVALID) &&
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gst_rtcp_packet_move_to_next (&packet));
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}
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gst_rtcp_buffer_unmap (&rtcpbuf);
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return ret;
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}
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void
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set_crypto_policy_cipher_auth (GstSrtpCipherType cipher,
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GstSrtpAuthType auth, srtp_crypto_policy_t * policy)
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{
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switch (cipher) {
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case GST_SRTP_CIPHER_AES_128_ICM:
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policy->cipher_type = SRTP_AES_ICM_128;
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break;
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case GST_SRTP_CIPHER_AES_256_ICM:
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policy->cipher_type = SRTP_AES_ICM_256;
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break;
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case GST_SRTP_CIPHER_AES_128_GCM:
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policy->cipher_type = SRTP_AES_GCM_128;
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break;
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case GST_SRTP_CIPHER_AES_256_GCM:
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policy->cipher_type = SRTP_AES_GCM_256;
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break;
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case GST_SRTP_CIPHER_NULL:
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policy->cipher_type = SRTP_NULL_CIPHER;
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break;
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default:
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g_assert_not_reached ();
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}
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policy->cipher_key_len = cipher_key_size (cipher);
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switch (auth) {
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case GST_SRTP_AUTH_HMAC_SHA1_80:
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policy->auth_type = SRTP_HMAC_SHA1;
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policy->auth_key_len = 20;
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policy->auth_tag_len = 10;
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break;
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case GST_SRTP_AUTH_HMAC_SHA1_32:
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policy->auth_type = SRTP_HMAC_SHA1;
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policy->auth_key_len = 20;
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policy->auth_tag_len = 4;
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break;
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case GST_SRTP_AUTH_NULL:
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policy->auth_type = SRTP_NULL_AUTH;
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policy->auth_key_len = 0;
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if (cipher == GST_SRTP_CIPHER_AES_128_GCM
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|| cipher == GST_SRTP_CIPHER_AES_256_GCM) {
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policy->auth_tag_len = 16;
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} else {
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policy->auth_tag_len = 0;
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}
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break;
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}
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if (cipher == GST_SRTP_CIPHER_NULL && auth == GST_SRTP_AUTH_NULL)
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policy->sec_serv = sec_serv_none;
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else if (cipher == GST_SRTP_CIPHER_NULL)
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policy->sec_serv = sec_serv_auth;
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else if (auth == GST_SRTP_AUTH_NULL)
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policy->sec_serv = sec_serv_conf;
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else
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policy->sec_serv = sec_serv_conf_and_auth;
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}
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guint
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cipher_key_size (GstSrtpCipherType cipher)
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{
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guint size = 0;
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switch (cipher) {
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case GST_SRTP_CIPHER_AES_128_ICM:
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size = SRTP_AES_ICM_128_KEY_LEN_WSALT;
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break;
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case GST_SRTP_CIPHER_AES_256_ICM:
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size = SRTP_AES_ICM_256_KEY_LEN_WSALT;
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break;
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case GST_SRTP_CIPHER_AES_128_GCM:
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size = SRTP_AES_GCM_128_KEY_LEN_WSALT;
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break;
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case GST_SRTP_CIPHER_AES_256_GCM:
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size = SRTP_AES_GCM_256_KEY_LEN_WSALT;
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break;
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case GST_SRTP_CIPHER_NULL:
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break;
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default:
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g_assert_not_reached ();
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}
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return size;
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}
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