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extra/gst-plugins-bad to 1.12.4-2

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This commit is contained in:
Kevin Mihelich 2018-01-14 16:27:26 +00:00
parent 53caf853b3
commit 1ff24ddc31
2 changed files with 841 additions and 3 deletions
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12
extra/gst-plugins-bad/PKGBUILD
View file

@ -8,7 +8,7 @@
pkgname=gst-plugins-bad
pkgver=1.12.4
pkgrel=1
pkgrel=2
pkgdesc="GStreamer Multimedia Framework Bad Plugins"
url="https://gstreamer.freedesktop.org/"
arch=(x86_64)
@ -23,9 +23,11 @@ makedepends=(python gobject-introspection gtk-doc git autoconf-archive vulkan-he
fluidsynth lilv opencv openexr)
_commit=cbdbd8d4f6893e6042dbf7b8258e23a8d2aaf081 # tags/1.12.4^0
source=("git+https://anongit.freedesktop.org/git/gstreamer/gst-plugins-bad#commit=$_commit"
"gst-common::git+https://anongit.freedesktop.org/git/gstreamer/common")
"gst-common::git+https://anongit.freedesktop.org/git/gstreamer/common"
srtp-Support-libsrtp2.patch)
sha256sums=('SKIP'
'SKIP')
'SKIP'
'12cb72b5972dcfbffa28176c5db9d701ebdbfb9dd8f26acf5500cc541abc87e7')
pkgver() {
cd $pkgname
@ -39,6 +41,10 @@ prepare() {
git config --local submodule.common.url "$srcdir/gst-common"
git submodule update
# libsrtp2
git cherry-pick -n 029e01743f 17121ebc57
patch -Np1 -i ../srtp-Support-libsrtp2.patch
sed -i 's/cmu_us_kal/&16/g' configure.ac ext/flite/gstflitetestsrc.c
NOCONFIGURE=1 ./autogen.sh

832
extra/gst-plugins-bad/srtp-Support-libsrtp2.patch Normal file
View file

@ -0,0 +1,832 @@
From 25713711519864ae2f666802d05b136772c3b84b Mon Sep 17 00:00:00 2001
Message-Id: <25713711519864ae2f666802d05b136772c3b84b.1515749240.git.jsteffens@make.tv>
From: "Jan Alexander Steffens (heftig)" <jsteffens@make.tv>
Date: Thu, 11 Jan 2018 16:06:24 +0100
Subject: [PATCH] srtp: Support libsrtp2
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
---
configure.ac | 12 ++++++---
ext/srtp/gstsrtp.c | 58 +++++++++++++++++++++++++++++-------------
ext/srtp/gstsrtp.h | 35 ++++++++++++++++++++++++--
ext/srtp/gstsrtpdec.c | 70 ++++++++++++++++++++++++++++-----------------------
ext/srtp/gstsrtpdec.h | 5 ++--
ext/srtp/gstsrtpenc.c | 65 ++++++++++++++++++++++++++++-------------------
ext/srtp/gstsrtpenc.h | 5 ++--
ext/srtp/meson.build | 15 ++++++++---
8 files changed, 178 insertions(+), 87 deletions(-)
diff --git a/configure.ac b/configure.ac
index fc15bfd8a..7a8714360 100644
--- a/configure.ac
+++ b/configure.ac
@@ -2470,9 +2470,15 @@ AC_SUBST(LIBMMS_LIBS)
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)
+ 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)
])
diff --git a/ext/srtp/gstsrtp.c b/ext/srtp/gstsrtp.c
index 1f2d3015c..0c1e235ab 100644
--- a/ext/srtp/gstsrtp.c
+++ b/ext/srtp/gstsrtp.c
@@ -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);
@@ -185,64 +210,63 @@ 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;
}
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;
+ 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;
diff --git a/ext/srtp/gstsrtp.h b/ext/srtp/gstsrtp.h
index 6bc4fb943..8ea1cf9dc 100644
--- a/ext/srtp/gstsrtp.h
+++ b/ext/srtp/gstsrtp.h
@@ -45,34 +45,65 @@
#ifndef __GST_SRTP_H__
#define __GST_SRTP_H__
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
#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>
+
+# 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
+
+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
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;
void gst_srtp_init_event_reporter (void);
gboolean gst_srtp_get_soft_limit_reached (void);
gboolean rtcp_buffer_get_ssrc (GstBuffer * buf, guint32 * ssrc);
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);
diff --git a/ext/srtp/gstsrtpdec.c b/ext/srtp/gstsrtpdec.c
index 4ad989fb2..ae47704f0 100644
--- a/ext/srtp/gstsrtpdec.c
+++ b/ext/srtp/gstsrtpdec.c
@@ -108,21 +108,13 @@
*
*/
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
+#include "gstsrtpdec.h"
-#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>
-
GST_DEBUG_CATEGORY_STATIC (gst_srtp_dec_debug);
#define GST_CAT_DEFAULT gst_srtp_dec_debug
@@ -409,35 +401,45 @@ 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 *
gst_srtp_dec_create_stats (GstSrtpDec * filter)
{
GstStructure *s;
GValue va = G_VALUE_INIT;
GValue v = G_VALUE_INIT;
s = gst_structure_new_empty ("application/x-srtp-decoder-stats");
g_value_init (&va, GST_TYPE_ARRAY);
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 +558,8 @@ 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.");
@@ -594,19 +597,19 @@ 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];
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 +641,20 @@ 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 +719,7 @@ 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 +757,7 @@ 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);
@@ -1139,83 +1145,85 @@ 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
" with SSRC = %u", is_rtcp ? "RTCP" : "RTP", gst_buffer_get_size (buf),
ssrc);
/* Change buffer to remove protection */
buf = gst_buffer_make_writable (buf);
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
size = map.size;
unprotect:
gst_srtp_init_event_reporter ();
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) {
srtp_stream_t stream;
stream = srtp_get_stream (filter->session, htonl (ssrc));
if (stream) {
guint16 seqnum = 0;
GstRTPBuffer rtpbuf = GST_RTP_BUFFER_INIT;
gst_rtp_buffer_map (buf,
GST_MAP_READ | GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING, &rtpbuf);
seqnum = gst_rtp_buffer_get_seq (&rtpbuf);
gst_rtp_buffer_unmap (&rtpbuf);
/* We finally add the RTP sequence number to the current
* rollover counter. */
stream->rtp_rdbx.index &= ~0xFFFF;
stream->rtp_rdbx.index |= seqnum;
}
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 */
if (find_stream_by_ssrc (filter, ssrc)) {
GST_OBJECT_UNLOCK (filter);
if (request_key_with_signal (filter, ssrc, SIGNAL_HARD_LIMIT)) {
GST_OBJECT_LOCK (filter);
goto unprotect;
} else {
GST_WARNING_OBJECT (filter, "Hard limit reached, no new key, "
"dropping");
}
} else {
GST_WARNING_OBJECT (filter, "Could not find matching stream, "
"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:
diff --git a/ext/srtp/gstsrtpdec.h b/ext/srtp/gstsrtpdec.h
index 644092a50..f26435a43 100644
--- a/ext/srtp/gstsrtpdec.h
+++ b/ext/srtp/gstsrtpdec.h
@@ -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
diff --git a/ext/srtp/gstsrtpenc.c b/ext/srtp/gstsrtpenc.c
index 893ece8be..c5884da40 100644
--- a/ext/srtp/gstsrtpenc.c
+++ b/ext/srtp/gstsrtpenc.c
@@ -102,22 +102,15 @@
* 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 <stdio.h>
-#include "gstsrtpenc.h"
-
-#include "gstsrtp.h"
#include "gstsrtp-enumtypes.h"
-#include <srtp/srtp_priv.h>
-
GST_DEBUG_CATEGORY_STATIC (gst_srtp_enc_debug);
#define GST_CAT_DEFAULT gst_srtp_enc_debug
@@ -355,57 +348,58 @@ 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
max_cipher_key_size (GstSrtpEnc * filter)
{
guint rtp_size, rtcp_size;
rtp_size = cipher_key_size (filter->rtp_cipher);
rtcp_size = cipher_key_size (filter->rtcp_cipher);
return (rtp_size > rtcp_size) ? rtp_size : rtcp_size;
}
/* Create stream
*
* 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];
memset (&policy, 0, sizeof (srtp_policy_t));
if (HAS_CRYPTO (filter)) {
guint expected;
gsize keysize;
if (filter->key == NULL) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
("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);
keysize = gst_buffer_get_size (filter->key);
if (expected != keysize) {
GST_OBJECT_UNLOCK (filter);
GST_ELEMENT_ERROR (filter, LIBRARY, SETTINGS,
("Master key size is wrong"),
("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 +444,8 @@ 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;
@@ -604,35 +600,46 @@ 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);
}
static GstStructure *
gst_srtp_enc_create_stats (GstSrtpEnc * filter)
{
GstStructure *s;
GValue va = G_VALUE_INIT;
GValue v = G_VALUE_INIT;
s = gst_structure_new_empty ("application/x-srtp-encoder-stats");
g_value_init (&va, GST_TYPE_ARRAY);
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 +808,12 @@ 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 +1003,9 @@ 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"),
@@ -1025,39 +1038,39 @@ 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);
size_max = size + SRTP_MAX_TRAILER_LEN + 10;
bufout = gst_buffer_new_allocate (NULL, size_max, NULL);
gst_buffer_map (bufout, &mapout, GST_MAP_READWRITE);
gst_buffer_extract (buf, 0, mapout.data, size);
GST_OBJECT_LOCK (filter);
gst_srtp_init_event_reporter ();
if (is_rtcp)
err = srtp_protect_rtcp (filter->session, mapout.data, &size);
else
err = srtp_protect (filter->session, mapout.data, &size);
GST_OBJECT_UNLOCK (filter);
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);
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 +1291,8 @@ 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;
diff --git a/ext/srtp/gstsrtpenc.h b/ext/srtp/gstsrtpenc.h
index 842c56f54..5c64e0b3e 100644
--- a/ext/srtp/gstsrtpenc.h
+++ b/ext/srtp/gstsrtpenc.h
@@ -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
diff --git a/ext/srtp/meson.build b/ext/srtp/meson.build
index 09f511118..3fb508703 100644
--- a/ext/srtp/meson.build
+++ b/ext/srtp/meson.build
@@ -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', 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',
command : [mkenums, glib_mkenums, '@OUTPUT@', '@INPUT@'])
gstsrtp_c = custom_target('gstsrtpenum_c',
output : 'gstsrtp-enumtypes.c',
input : 'gstsrtp.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],
--
2.15.1