[arch-commits] Commit in gst-plugins-bad/trunk (PKGBUILD srtp-Support-libsrtp2.patch)
Jan Steffens
heftig at archlinux.org
Thu Feb 1 19:17:57 UTC 2018
Date: Thursday, February 1, 2018 @ 19:17:56
Author: heftig
Revision: 315741
1.12.4-3
Modified:
gst-plugins-bad/trunk/PKGBUILD
Deleted:
gst-plugins-bad/trunk/srtp-Support-libsrtp2.patch
-----------------------------+
PKGBUILD | 15
srtp-Support-libsrtp2.patch | 832 ------------------------------------------
2 files changed, 6 insertions(+), 841 deletions(-)
Modified: PKGBUILD
===================================================================
--- PKGBUILD 2018-02-01 19:17:42 UTC (rev 315740)
+++ PKGBUILD 2018-02-01 19:17:56 UTC (rev 315741)
@@ -4,7 +4,7 @@
pkgname=gst-plugins-bad
pkgver=1.12.4
-pkgrel=2
+pkgrel=3
pkgdesc="GStreamer Multimedia Framework Bad Plugins"
url="https://gstreamer.freedesktop.org/"
arch=(x86_64)
@@ -13,17 +13,15 @@
libmodplug libgme wayland libofa openjpeg2 libwebp libsrtp gnutls glu sbc rtmpdump
libgudev graphene schroedinger libexif libdvdread libvdpau libmpeg2 wildmidi ladspa
openal libusb vulkan-icd-loader libfdk-aac faac soundtouch spandsp neon
- webrtc-audio-processing libdc1394 libmpcdec zvbi)
+ webrtc-audio-processing libdc1394 libmpcdec zvbi openexr libbs2b)
makedepends=(python valgrind gobject-introspection gtk-doc git autoconf-archive vulkan-headers
gtk3 clutter librsvg libtiger qt5-declarative qt5-x11extras qt5-wayland zbar
- fluidsynth lilv opencv openexr)
+ fluidsynth lilv opencv)
_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"
- srtp-Support-libsrtp2.patch)
+ "gst-common::git+https://anongit.freedesktop.org/git/gstreamer/common")
sha256sums=('SKIP'
- 'SKIP'
- '12cb72b5972dcfbffa28176c5db9d701ebdbfb9dd8f26acf5500cc541abc87e7')
+ 'SKIP')
pkgver() {
cd $pkgname
@@ -38,8 +36,7 @@
git submodule update
# libsrtp2
- git cherry-pick -n 029e01743f 17121ebc57
- patch -Np1 -i ../srtp-Support-libsrtp2.patch
+ git cherry-pick -n 029e01743f 17121ebc57 e9aa117200
sed -i 's/cmu_us_kal/&16/g' configure.ac ext/flite/gstflitetestsrc.c
Deleted: srtp-Support-libsrtp2.patch
===================================================================
--- srtp-Support-libsrtp2.patch 2018-02-01 19:17:42 UTC (rev 315740)
+++ srtp-Support-libsrtp2.patch 2018-02-01 19:17:56 UTC (rev 315741)
@@ -1,832 +0,0 @@
-From 25713711519864ae2f666802d05b136772c3b84b Mon Sep 17 00:00:00 2001
-Message-Id: <25713711519864ae2f666802d05b136772c3b84b.1515749240.git.jsteffens at make.tv>
-From: "Jan Alexander Steffens (heftig)" <jsteffens at 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
\ No newline at end of file
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