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srtp: Support libsrtp2

Browse source 
For libsrtp 1, add defines that translate the new namespaced identifiers
to the old unnamespaced ones. Also move the code for setting and getting
a stream's ROC into two compat functions that match libsrtp2's API.

It seems that libsrtp2 properly supports changing the ROC without having
to touch the sequence numbers afterwards, given that srtp_set_stream_roc
sets a pending_roc field, so the entire roc_changed dance should not be
needed anymore. The compat functions for libsrtp 1 just contain our
preexisting hacks, however, so it's still needed there.

libsrtp2 has no means of discovering the streams in the session, so to
create the stats structure we need to iterate over our own set of SSRCs.
For this we also need to re-add the previously removed ssrcs_set to the
encoder.

https://bugzilla.gnome.org/show_bug.cgi?id=776901
This commit is contained in:
Jan Alexander Steffens (heftig) 2018-01-26 16:25:33 +01:00 committed by Sebastian Dröge
parent 49d30c901d
commit e9aa117200
11 changed files with 251 additions and 111 deletions
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12
configure.ac
View file

@ -1524,9 +1524,15 @@ AG_GST_CHECK_FEATURE(SRT, [srt library], srt, [
dnl *** libsrtp ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SRTP, true)
AG_GST_CHECK_FEATURE(SRTP, [srtp library], srtp, [
PKG_CHECK_MODULES(SRTP, libsrtp, HAVE_SRTP="yes",
AG_GST_CHECK_LIBHEADER(SRTP, srtp, srtp_init, , srtp/srtp.h, SRTP_LIBS="-lsrtp")
)
HAVE_SRTP="no"
AG_GST_PKG_CHECK_MODULES(SRTP, libsrtp2 >= 2.1.0)
if test x"$HAVE_SRTP" = x"yes"; then
AC_DEFINE([HAVE_SRTP2], 1, [Define if libsrtp2 is used])
else
PKG_CHECK_MODULES(SRTP, libsrtp, HAVE_SRTP="yes",
AG_GST_CHECK_LIBHEADER(SRTP, srtp, srtp_init, , srtp/srtp.h, SRTP_LIBS="-lsrtp")
)
fi
AC_SUBST(SRTP_LIBS)
AC_SUBST(SRTP_CFLAGS)
])

10
ext/srtp/Makefile.am
View file

@ -1,12 +1,14 @@
plugin_LTLIBRARIES = libgstsrtp.la
libgstsrtp_la_SOURCES = \
gstsrtp.c \
gstsrtp.h \
gstsrtp.c \
gstsrtp.h \
gstsrtpdec.c \
gstsrtpdec.h \
gstsrtpenc.c \
gstsrtpenc.h
gstsrtpenc.h \
gstsrtpenums.h \
$(NULL)
libgstsrtp_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(SRTP_CFLAGS)
libgstsrtp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
@ -15,7 +17,7 @@ libgstsrtp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) \
$(SRTP_LIBS)
libgstsrtp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
glib_enum_headers=gstsrtp.h
glib_enum_headers=gstsrtpenums.h
glib_enum_define=GST_SRTP
glib_gen_prefix=gst_srtp
glib_gen_basename=gstsrtp

58
ext/srtp/gstsrtp.c
View file

@ -21,21 +21,46 @@
* 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"
#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);
@ -185,38 +210,37 @@ rtcp_buffer_get_ssrc (GstBuffer * buf, guint32 * ssrc)
void
set_crypto_policy_cipher_auth (GstSrtpCipherType cipher,
GstSrtpAuthType auth, crypto_policy_t * policy)
GstSrtpAuthType auth, srtp_crypto_policy_t * policy)
{
switch (cipher) {
case GST_SRTP_CIPHER_AES_128_ICM:
policy->cipher_type = AES_ICM;
policy->cipher_key_len = 30;
policy->cipher_type = SRTP_AES_ICM_128;
break;
case GST_SRTP_CIPHER_AES_256_ICM:
policy->cipher_type = AES_ICM;
policy->cipher_key_len = 46;
policy->cipher_type = SRTP_AES_ICM_256;
break;
case GST_SRTP_CIPHER_NULL:
policy->cipher_type = NULL_CIPHER;
policy->cipher_key_len = 0;
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 = HMAC_SHA1;
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 = HMAC_SHA1;
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 = NULL_AUTH;
policy->auth_type = SRTP_NULL_AUTH;
policy->auth_key_len = 0;
policy->auth_tag_len = 0;
break;
@ -239,10 +263,10 @@ cipher_key_size (GstSrtpCipherType cipher)
switch (cipher) {
case GST_SRTP_CIPHER_AES_128_ICM:
size = 30;
size = SRTP_AES_ICM_128_KEY_LEN_WSALT;
break;
case GST_SRTP_CIPHER_AES_256_ICM:
size = 46;
size = SRTP_AES_ICM_256_KEY_LEN_WSALT;
break;
case GST_SRTP_CIPHER_NULL:
size = 0;

48
ext/srtp/gstsrtp.h
View file

@ -45,23 +45,43 @@
#ifndef __GST_SRTP_H__
#define __GST_SRTP_H__
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "gstsrtpenums.h"
#include "gstsrtp-enumtypes.h"
#include <gst/gst.h>
#include <srtp/srtp.h>
#ifdef HAVE_SRTP2
# include <srtp2/srtp.h>
# include <srtp2/crypto_types.h>
#else
# include <srtp/srtp.h>
# include <srtp/srtp_priv.h>
typedef enum
{
GST_SRTP_CIPHER_NULL,
GST_SRTP_CIPHER_AES_128_ICM,
GST_SRTP_CIPHER_AES_256_ICM
} GstSrtpCipherType;
# define srtp_crypto_policy_t crypto_policy_t
# define SRTP_AES_ICM_128 AES_ICM
# define SRTP_AES_ICM_256 AES_ICM
# define SRTP_NULL_CIPHER NULL_CIPHER
# define SRTP_AES_ICM_128_KEY_LEN_WSALT 30
# define SRTP_AES_ICM_256_KEY_LEN_WSALT 46
# define SRTP_HMAC_SHA1 HMAC_SHA1
# define SRTP_NULL_AUTH NULL_AUTH
# define srtp_err_status_t err_status_t
# define srtp_err_status_ok err_status_ok
# define srtp_err_status_bad_param err_status_bad_param
# define srtp_err_status_key_expired err_status_key_expired
# define srtp_err_status_auth_fail err_status_auth_fail
# define srtp_err_status_cipher_fail err_status_cipher_fail
# define srtp_err_status_fail err_status_fail
typedef enum
{
GST_SRTP_AUTH_NULL,
GST_SRTP_AUTH_HMAC_SHA1_32,
GST_SRTP_AUTH_HMAC_SHA1_80
} GstSrtpAuthType;
srtp_err_status_t srtp_set_stream_roc (srtp_t session, guint32 ssrc,
guint32 roc);
srtp_err_status_t srtp_get_stream_roc (srtp_t session, guint32 ssrc,
guint32 * roc);
#endif
void gst_srtp_init_event_reporter (void);
gboolean gst_srtp_get_soft_limit_reached (void);
@ -72,7 +92,7 @@ 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 (GstSrtpCipherType cipher,
GstSrtpAuthType auth, crypto_policy_t * policy);
GstSrtpAuthType auth, srtp_crypto_policy_t * policy);
guint cipher_key_size (GstSrtpCipherType cipher);

74
ext/srtp/gstsrtpdec.c
View file

@ -108,20 +108,10 @@
*
*/
#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"
#include <srtp/srtp_priv.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_srtp_dec_debug);
#define GST_CAT_DEFAULT gst_srtp_dec_debug
@ -409,7 +399,10 @@ gst_srtp_dec_init (GstSrtpDec * filter)
gst_element_add_pad (GST_ELEMENT (filter), filter->rtcp_srcpad);
filter->first_session = TRUE;
#ifndef HAVE_SRTP2
filter->roc_changed = FALSE;
#endif
}
static GstStructure *
@ -425,19 +418,26 @@ gst_srtp_dec_create_stats (GstSrtpDec * filter)
g_value_init (&v, GST_TYPE_STRUCTURE);
if (filter->session) {
srtp_stream_t stream = filter->session->stream_list;
while (stream) {
GHashTableIter iter;
gpointer key;
g_hash_table_iter_init (&iter, filter->streams);
while (g_hash_table_iter_next (&iter, &key, NULL)) {
GstStructure *ss;
guint32 ssrc = GUINT32_FROM_BE (stream->ssrc);
guint32 roc = stream->rtp_rdbx.index >> 16;
guint32 ssrc = GPOINTER_TO_UINT (key);
srtp_err_status_t status;
guint32 roc;
status = srtp_get_stream_roc (filter->session, ssrc, &roc);
if (status != srtp_err_status_ok) {
continue;
}
ss = gst_structure_new ("application/x-srtp-stream",
"ssrc", G_TYPE_UINT, ssrc, "roc", G_TYPE_UINT, roc, NULL);
g_value_take_boxed (&v, ss);
gst_value_array_append_value (&va, &v);
stream = stream->next;
}
}
@ -556,7 +556,8 @@ get_stream_from_caps (GstSrtpDec * filter, GstCaps * caps, guint32 ssrc)
goto error;
}
if (stream->rtcp_cipher != NULL_CIPHER && stream->rtcp_auth == NULL_AUTH) {
if (stream->rtcp_cipher != SRTP_NULL_CIPHER &&
stream->rtcp_auth == SRTP_NULL_AUTH) {
GST_WARNING_OBJECT (filter,
"Cannot have SRTP NULL authentication with a not-NULL encryption"
" cipher.");
@ -594,11 +595,11 @@ signal_get_srtp_params (GstSrtpDec * filter, guint32 ssrc, gint signal)
/* Create a stream in the session
*/
static err_status_t
static srtp_err_status_t
init_session_stream (GstSrtpDec * filter, guint32 ssrc,
GstSrtpDecSsrcStream * stream)
{
err_status_t ret;
srtp_err_status_t ret;
srtp_policy_t policy;
GstMapInfo map;
guchar tmp[1];
@ -606,7 +607,7 @@ init_session_stream (GstSrtpDec * filter, guint32 ssrc,
memset (&policy, 0, sizeof (srtp_policy_t));
if (!stream)
return err_status_bad_param;
return srtp_err_status_bad_param;
GST_INFO_OBJECT (filter, "Setting RTP policy...");
set_crypto_policy_cipher_auth (stream->rtp_cipher, stream->rtp_auth,
@ -638,17 +639,20 @@ init_session_stream (GstSrtpDec * filter, guint32 ssrc,
if (stream->key)
gst_buffer_unmap (stream->key, &map);
if (ret == err_status_ok) {
srtp_stream_t srtp_stream;
if (ret == srtp_err_status_ok) {
srtp_err_status_t status;
srtp_stream = srtp_get_stream (filter->session, htonl (ssrc));
if (srtp_stream) {
status = srtp_set_stream_roc (filter->session, ssrc, stream->roc);
#ifdef HAVE_SRTP2
(void) status; /* Ignore unused variable */
#else
if (status == srtp_err_status_ok) {
/* Here, we just set the ROC, but we also need to set the initial
* RTP sequence number later, otherwise libsrtp will not be able
* to get the right packet index. */
rdbx_set_roc (&srtp_stream->rtp_rdbx, stream->roc);
filter->roc_changed = TRUE;
}
#endif
filter->first_session = FALSE;
g_hash_table_insert (filter->streams, GUINT_TO_POINTER (stream->ssrc),
@ -713,7 +717,7 @@ update_session_stream_from_caps (GstSrtpDec * filter, guint32 ssrc,
{
GstSrtpDecSsrcStream *stream = NULL;
GstSrtpDecSsrcStream *old_stream = NULL;
err_status_t err;
srtp_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);
@ -751,7 +755,7 @@ update_session_stream_from_caps (GstSrtpDec * filter, guint32 ssrc,
/* Create new session stream */
err = init_session_stream (filter, ssrc, stream);
if (err != err_status_ok) {
if (err != srtp_err_status_ok) {
if (stream->key)
gst_buffer_unref (stream->key);
g_slice_free (GstSrtpDecSsrcStream, stream);
@ -1139,7 +1143,7 @@ gst_srtp_dec_decode_buffer (GstSrtpDec * filter, GstPad * pad, GstBuffer * buf,
gboolean is_rtcp, guint32 ssrc)
{
GstMapInfo map;
err_status_t err;
srtp_err_status_t err;
gint size;
GST_LOG_OBJECT (pad, "Received %s buffer of size %" G_GSIZE_FORMAT
@ -1159,6 +1163,7 @@ unprotect:
if (is_rtcp)
err = srtp_unprotect_rtcp (filter->session, map.data, &size);
else {
#ifndef HAVE_SRTP2
/* If ROC has changed, we know we need to set the initial RTP
* sequence number too. */
if (filter->roc_changed) {
@ -1183,18 +1188,19 @@ unprotect:
filter->roc_changed = FALSE;
}
#endif
err = srtp_unprotect (filter->session, map.data, &size);
}
GST_OBJECT_UNLOCK (filter);
if (err != err_status_ok) {
if (err != srtp_err_status_ok) {
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:
case srtp_err_status_key_expired:
GST_OBJECT_LOCK (filter);
/* Update stream */
@ -1212,10 +1218,10 @@ unprotect:
"dropping");
}
break;
case err_status_auth_fail:
case srtp_err_status_auth_fail:
GST_WARNING_OBJECT (filter, "Error authentication packet, dropping");
break;
case err_status_cipher_fail:
case srtp_err_status_cipher_fail:
GST_WARNING_OBJECT (filter, "Error while decrypting packet, dropping");
break;
default:

5
ext/srtp/gstsrtpdec.h
View file

@ -47,8 +47,7 @@
#ifndef __GST_SRTPDEC_H__
#define __GST_SRTPDEC_H__
#include <gst/gst.h>
#include <srtp/srtp.h>
#include "gstsrtp.h"
G_BEGIN_DECLS
@ -84,7 +83,9 @@ struct _GstSrtpDec
gboolean rtp_has_segment;
gboolean rtcp_has_segment;
#ifndef HAVE_SRTP2
gboolean roc_changed;
#endif
};
struct _GstSrtpDecClass

67
ext/srtp/gstsrtpenc.c
View file

@ -102,21 +102,12 @@
* while to other clients.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "gstsrtpenc.h"
#include <gst/gst.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <string.h>
#include "gstsrtpenc.h"
#include "gstsrtp.h"
#include "gstsrtp-enumtypes.h"
#include <srtp/srtp_priv.h>
#include <stdio.h>
GST_DEBUG_CATEGORY_STATIC (gst_srtp_enc_debug);
#define GST_CAT_DEFAULT gst_srtp_enc_debug
@ -355,6 +346,7 @@ gst_srtp_enc_init (GstSrtpEnc * filter)
filter->rtcp_auth = DEFAULT_RTCP_AUTH;
filter->replay_window_size = DEFAULT_REPLAY_WINDOW_SIZE;
filter->allow_repeat_tx = DEFAULT_ALLOW_REPEAT_TX;
filter->ssrcs_set = g_hash_table_new (g_direct_hash, g_direct_equal);
}
static guint
@ -372,10 +364,10 @@ max_cipher_key_size (GstSrtpEnc * filter)
*
* Should be called with the filter locked
*/
static err_status_t
static srtp_err_status_t
gst_srtp_enc_create_session (GstSrtpEnc * filter)
{
err_status_t ret;
srtp_err_status_t ret;
srtp_policy_t policy;
GstMapInfo map;
guchar tmp[1];
@ -392,7 +384,7 @@ gst_srtp_enc_create_session (GstSrtpEnc * filter)
("Cipher is not NULL, key must be set"),
("Cipher is not NULL, key must be set"));
GST_OBJECT_LOCK (filter);
return err_status_fail;
return srtp_err_status_fail;
}
expected = max_cipher_key_size (filter);
@ -405,7 +397,7 @@ gst_srtp_enc_create_session (GstSrtpEnc * filter)
("Expected master key of %d bytes, but received %" G_GSIZE_FORMAT
" bytes", expected, keysize));
GST_OBJECT_LOCK (filter);
return err_status_fail;
return srtp_err_status_fail;
}
}
@ -450,6 +442,8 @@ gst_srtp_enc_reset_no_lock (GstSrtpEnc * filter)
if (!filter->first_session) {
srtp_dealloc (filter->session);
filter->session = NULL;
g_hash_table_remove_all (filter->ssrcs_set);
}
filter->first_session = TRUE;
@ -604,6 +598,10 @@ gst_srtp_enc_dispose (GObject * object)
gst_buffer_unref (filter->key);
filter->key = NULL;
if (filter->ssrcs_set)
g_hash_table_unref (filter->ssrcs_set);
filter->ssrcs_set = NULL;
G_OBJECT_CLASS (gst_srtp_enc_parent_class)->dispose (object);
}
@ -620,19 +618,26 @@ gst_srtp_enc_create_stats (GstSrtpEnc * filter)
g_value_init (&v, GST_TYPE_STRUCTURE);
if (filter->session) {
srtp_stream_t stream = filter->session->stream_list;
while (stream) {
GHashTableIter iter;
gpointer key;
g_hash_table_iter_init (&iter, filter->ssrcs_set);
while (g_hash_table_iter_next (&iter, &key, NULL)) {
GstStructure *ss;
guint32 ssrc = GUINT32_FROM_BE (stream->ssrc);
guint32 roc = stream->rtp_rdbx.index >> 16;
guint32 ssrc = GPOINTER_TO_UINT (key);
srtp_err_status_t status;
guint32 roc;
status = srtp_get_stream_roc (filter->session, ssrc, &roc);
if (status != srtp_err_status_ok) {
continue;
}
ss = gst_structure_new ("application/x-srtp-stream",
"ssrc", G_TYPE_UINT, ssrc, "roc", G_TYPE_UINT, roc, NULL);
g_value_take_boxed (&v, ss);
gst_value_array_append_value (&va, &v);
stream = stream->next;
}
}
@ -801,6 +806,12 @@ gst_srtp_enc_sink_setcaps (GstPad * pad, GstSrtpEnc * filter,
GST_OBJECT_LOCK (filter);
if (gst_structure_has_field_typed (ps, "ssrc", G_TYPE_UINT)) {
guint ssrc;
gst_structure_get_uint (ps, "ssrc", &ssrc);
g_hash_table_add (filter->ssrcs_set, GUINT_TO_POINTER (ssrc));
}
if (HAS_CRYPTO (filter))
gst_structure_set (ps, "srtp-key", GST_TYPE_BUFFER, filter->key, NULL);
@ -990,9 +1001,9 @@ gst_srtp_enc_check_set_caps (GstSrtpEnc * filter, GstPad * pad,
}
if (filter->first_session) {
err_status_t status = gst_srtp_enc_create_session (filter);
srtp_err_status_t status = gst_srtp_enc_create_session (filter);
if (status != err_status_ok) {
if (status != srtp_err_status_ok) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, INIT,
("Could not initialize SRTP encoder"),
@ -1025,7 +1036,7 @@ gst_srtp_enc_process_buffer (GstSrtpEnc * filter, GstPad * pad,
gint size_max, size;
GstBuffer *bufout = NULL;
GstMapInfo mapout;
err_status_t err;
srtp_err_status_t err;
/* Create a bigger buffer to add protection */
size = gst_buffer_get_size (buf);
@ -1049,7 +1060,7 @@ gst_srtp_enc_process_buffer (GstSrtpEnc * filter, GstPad * pad,
gst_buffer_unmap (bufout, &mapout);
if (err == err_status_ok) {
if (err == srtp_err_status_ok) {
/* Buffer protected */
gst_buffer_set_size (bufout, size);
gst_buffer_copy_into (bufout, buf, GST_BUFFER_COPY_METADATA, 0, -1);
@ -1057,7 +1068,7 @@ gst_srtp_enc_process_buffer (GstSrtpEnc * filter, GstPad * pad,
GST_LOG_OBJECT (pad, "Encoding %s buffer of size %d",
is_rtcp ? "RTCP" : "RTP", size);
} else if (err == err_status_key_expired) {
} else if (err == srtp_err_status_key_expired) {
GST_ELEMENT_ERROR (GST_ELEMENT_CAST (filter), STREAM, ENCODE,
("Key usage limit has been reached"),
@ -1278,8 +1289,8 @@ gst_srtp_enc_change_state (GstElement * element, GstStateChange transition)
}
}
}
if ((filter->rtcp_cipher != NULL_CIPHER)
&& (filter->rtcp_auth == NULL_AUTH)) {
if ((filter->rtcp_cipher != SRTP_NULL_CIPHER)
&& (filter->rtcp_auth == SRTP_NULL_AUTH)) {
GST_ERROR_OBJECT (filter,
"RTCP authentication can't be NULL if encryption is not NULL.");
return GST_STATE_CHANGE_FAILURE;

5
ext/srtp/gstsrtpenc.h
View file

@ -47,8 +47,7 @@
#ifndef __GST_SRTPENC_H__
#define __GST_SRTPENC_H__
#include <gst/gst.h>
#include <srtp/srtp.h>
#include "gstsrtp.h"
G_BEGIN_DECLS
@ -84,6 +83,8 @@ struct _GstSrtpEnc
guint replay_window_size;
gboolean allow_repeat_tx;
GHashTable *ssrcs_set;
};
struct _GstSrtpEncClass

62
ext/srtp/gstsrtpenums.h Normal file
View file

@ -0,0 +1,62 @@
/*
* 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_ENUMS_H__
#define __GST_SRTP_ENUMS_H__
typedef enum
{
GST_SRTP_CIPHER_NULL,
GST_SRTP_CIPHER_AES_128_ICM,
GST_SRTP_CIPHER_AES_256_ICM
} GstSrtpCipherType;
typedef enum
{
GST_SRTP_AUTH_NULL,
GST_SRTP_AUTH_HMAC_SHA1_32,
GST_SRTP_AUTH_HMAC_SHA1_80
} GstSrtpAuthType;
#endif /* __GST_SRTP_ENUMS_H__ */

19
ext/srtp/meson.build
View file

@ -4,27 +4,34 @@ srtp_sources = [
'gstsrtpenc.c',
]
srtp_dep = dependency('libsrtp', required : false)
if not srtp_dep.found() and cc.has_header_symbol('srtp/srtp.h', 'srtp_init')
srtp_dep = cc.find_library('srtp', required : false)
srtp_cargs = []
srtp_dep = dependency('libsrtp2', version : '>= 2.1.0', required : false)
if srtp_dep.found()
srtp_cargs += ['-DHAVE_SRTP2']
else
srtp_dep = dependency('libsrtp', required : false)
if not srtp_dep.found() and cc.has_header_symbol('srtp/srtp.h', 'srtp_init')
srtp_dep = cc.find_library('srtp', required : false)
endif
endif
if srtp_dep.found()
mkenums = find_program('srtp_mkenum.py')
gstsrtp_h = custom_target('gstsrtpenum_h',
output : 'gstsrtp-enumtypes.h',
input : 'gstsrtp.h',
input : 'gstsrtpenums.h',
command : [mkenums, glib_mkenums, '@OUTPUT@', '@INPUT@'])
gstsrtp_c = custom_target('gstsrtpenum_c',
output : 'gstsrtp-enumtypes.c',
input : 'gstsrtp.h',
input : 'gstsrtpenums.h',
depends : [gstsrtp_h],
command : [mkenums, glib_mkenums, '@OUTPUT@', '@INPUT@'])
gstsrtp = library('gstsrtp',
srtp_sources, gstsrtp_c, gstsrtp_h,
c_args : gst_plugins_bad_args,
c_args : gst_plugins_bad_args + srtp_cargs,
link_args : noseh_link_args,
include_directories : [configinc],
dependencies : [gstrtp_dep, gstvideo_dep, srtp_dep],

2
ext/srtp/srtp_mkenum.py
View file

@ -18,7 +18,7 @@ h_array = ['--fhead',
]
c_array = ['--fhead',
"#include \"gstsrtp-enumtypes.h\"\n\n#include \"gstsrtp.h\"",
"#include \"gstsrtp-enumtypes.h\"\n\n#include \"gstsrtpenums.h\"",
'--fprod',
"\n/* enumerations from \"@filename@\" */",
'--vhead',