| /* |
| * 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 (¤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_slice_new (struct GstSrtpEventReporterData); |
| 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_APP: |
| case GST_RTCP_TYPE_RTPFB: |
| case GST_RTCP_TYPE_PSFB: |
| *ssrc = gst_rtcp_packet_fb_get_sender_ssrc (&packet); |
| 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, 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_AES_256_ICM: |
| policy->cipher_type = AES_ICM; |
| policy->cipher_key_len = 46; |
| 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; |
| } |
| |
| guint |
| cipher_key_size (GstSrtpCipherType cipher) |
| { |
| guint size = 0; |
| |
| switch (cipher) { |
| case GST_SRTP_CIPHER_AES_128_ICM: |
| size = 30; |
| break; |
| case GST_SRTP_CIPHER_AES_256_ICM: |
| size = 46; |
| break; |
| case GST_SRTP_CIPHER_NULL: |
| size = 0; |
| break; |
| default: |
| g_assert_not_reached (); |
| } |
| |
| return size; |
| } |
| |
| 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/") |