- up to 2.16.1

diff --git a/build.patch b/build.patch
new file mode 100644 (file)
index 0000000..625626d
--- /dev/null
+++ b/build.patch
@@ -0,0 +1,10 @@
+--- mediastreamer-2.16.1/configure.ac~ 2017-06-23 10:52:57.000000000 +0200
++++ mediastreamer-2.16.1/configure.ac  2018-09-26 12:03:38.918658108 +0200
+@@ -1311,7 +1311,6 @@ AC_SUBST(MS_PUBLIC_CFLAGS)
+ 
+ AC_CONFIG_FILES(
+       Makefile
+-      po/Makefile.in
+       m4/Makefile
+       include/Makefile
+       include/mediastreamer2/Makefile

diff --git a/ffmpeg4.patch b/ffmpeg4.patch
deleted file mode 100644 (file)
index 6b7558c..0000000
--- a/ffmpeg4.patch
+++ /dev/null
@@ -1,20 +0,0 @@
---- mediastreamer-2.12.1/src/videofilters/nowebcam.c~  2018-04-28 19:50:35.000000000 +0200
-+++ mediastreamer-2.12.1/src/videofilters/nowebcam.c   2018-04-28 19:53:20.814288813 +0200
-@@ -32,7 +32,7 @@
- #ifndef NO_FFMPEG
- #include "ffmpeg-priv.h"
- #else
--#define FF_INPUT_BUFFER_PADDING_SIZE 32
-+#define AV_INPUT_BUFFER_PADDING_SIZE 32
- #endif
- 
- #if TARGET_OS_IPHONE
-@@ -248,7 +248,7 @@
-                       ms_error("Cannot load %s",jpgpath);
-                       return NULL;
-               }
--              jpgbuf=(uint8_t*)ms_malloc0(statbuf.st_size + FF_INPUT_BUFFER_PADDING_SIZE);
-+              jpgbuf=(uint8_t*)ms_malloc0(statbuf.st_size + AV_INPUT_BUFFER_PADDING_SIZE);
-               if (jpgbuf==NULL)
-               {
-                       close(fd);

diff --git a/libsrtp2.patch b/libsrtp2.patch
new file mode 100644 (file)
index 0000000..91cbae0
--- /dev/null
+++ b/libsrtp2.patch
@@ -0,0 +1,241 @@
+diff -u -r mediastreamer2-2.16.1/configure.ac mediastreamer2-2.16.1-libsrtp2/configure.ac
+--- mediastreamer2-2.16.1/configure.ac 2017-07-21 15:00:47.000000000 +0200
++++ mediastreamer2-2.16.1-libsrtp2/configure.ac        2018-01-13 00:06:22.194089121 +0100
+@@ -1041,7 +1041,7 @@
+ 
+ dnl check for libsrtp support (secure rtp)
+ AC_ARG_WITH(srtp,
+-      AC_HELP_STRING([--with-srtp], [Set prefix where libsrtp can be found or "none" to disable (ex:/usr or /usr/local) [[default=/usr]]]),
++      AC_HELP_STRING([--with-srtp], [Set prefix where libsrtp2 can be found or "none" to disable (ex:/usr or /usr/local) [[default=/usr]]]),
+       [srtp_prefix=$withval],
+       [if test "$prefix" != "NONE"; then
+               srtp_prefix=$prefix
+@@ -1051,22 +1051,22 @@
+ 
+ if test "${srtp_prefix}" != "none" ; then
+       if test "${srtp_prefix}" != "/usr" ; then
+-              SRTP_CFLAGS="-I${srtp_prefix}/include -I${srtp_prefix}/include/srtp"
++              SRTP_CFLAGS="-I${srtp_prefix}/include -I${srtp_prefix}/include/srtp2"
+               SRTP_LIBS="-L${srtp_prefix}/lib"
+       fi
+-      SRTP_LIBS="$SRTP_LIBS -lsrtp"
++      SRTP_LIBS="$SRTP_LIBS -lsrtp2"
+ 
+       dnl check srtp headers
+       CPPFLAGS_save=$CPPFLAGS
+       CPPFLAGS="$CPPFLAGS $SRTP_CFLAGS"
+-      AC_CHECK_HEADER([srtp/srtp.h],have_srtp_headers=yes)
++      AC_CHECK_HEADER([srtp2/srtp.h],have_srtp_headers=yes)
+       CPPFLAGS=$CPPFLAGS_save
+ 
+       dnl check for srtp lib
+       LDFLAGS_save=$LDFLAGS
+       LDFLAGS="$LDFLAGS $SRTP_LIBS"
+       LIBS_save=$LIBS
+-      AC_CHECK_LIB(srtp,[srtp_init, crypto_policy_set_aes_cm_256_hmac_sha1_80, crypto_policy_set_aes_cm_256_hmac_sha1_32], have_srtp_lib=yes)
++      AC_CHECK_LIB(srtp2,[srtp_init, srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80, srtp_crypto_policy_set_aes_cm_256_hmac_sha1_32], have_srtp_lib=yes)
+       LDFLAGS=$LDFLAGS_save
+       LIBS=$LIBS_save
+ 
+@@ -1075,9 +1075,9 @@
+               LIBS_save=$LIBS
+               LDFLAGS="$LDFLAGS $SRTP_LIBS"
+               AC_CHECK_LIB(
+-                      srtp,
++                      srtp2,
+                       sha1_update,[
+-                              AC_MSG_WARN([This libsrtp version exports symbols conflicting with polarssl, resulting in a bad execution path. libsrtp will be statically linked])
++                              AC_MSG_WARN([This libsrtp2 version exports symbols conflicting with polarssl, resulting in a bad execution path. libsrtp2 will be statically linked])
+                               srtp_polarssl_conflict=yes
+                       ],
+                       srtp_polarssl_conflict=no
+@@ -1086,13 +1086,13 @@
+               LIBS=$LIBS_save
+ 
+               if test "$srtp_polarssl_conflict" = "yes"; then
+-                      srtp_static_lib="${srtp_prefix}/lib/libsrtp.a"
++                      srtp_static_lib="${srtp_prefix}/lib/libsrtp2.a"
+                       if test -f $srtp_static_lib -a -r $srtp_static_lib; then
+-                              SRTP_LIBS="${srtp_prefix}/lib/libsrtp.a -Bsymbolic -Wl,--version-script=\$(top_srcdir)/src/libsrtp.map"
++                              SRTP_LIBS="${srtp_prefix}/lib/libsrtp2.a -Bsymbolic -Wl,--version-script=\$(top_srcdir)/src/libsrtp2.map"
+                               AC_DEFINE(HAVE_SRTP, 1, [Defined when srtp support is compiled])
+                               have_srtp=true
+                       else
+-                              AC_MSG_WARN([Could not access to $srtp_static_lib. Please use --with-srtp=PREFIX to specify the prefix where libsrtp.a has been installed])
++                              AC_MSG_WARN([Could not access to $srtp_static_lib. Please use --with-srtp=PREFIX to specify the prefix where libsrtp2.a has been installed])
+                               SRTP_CFLAGS=
+                               SRTP_LIBS=
+                               have_srtp=true
+diff -u -r mediastreamer2-2.16.1/src/crypto/ms_srtp.c mediastreamer2-2.16.1-libsrtp2/src/crypto/ms_srtp.c
+--- mediastreamer2-2.16.1/src/crypto/ms_srtp.c 2017-07-21 15:00:47.000000000 +0200
++++ mediastreamer2-2.16.1-libsrtp2/src/crypto/ms_srtp.c        2018-01-13 00:14:48.575306030 +0100
+@@ -38,7 +38,7 @@
+ // Windows phone doesn't use make install
+ #include <srtp.h>
+ #else
+-#include <srtp/srtp.h>
++#include <srtp2/srtp.h>
+ #endif
+ 
+ 
+@@ -99,7 +99,7 @@
+ /**** Sender functions ****/
+ static int _process_on_send(RtpSession* session,MSSrtpStreamContext *ctx, mblk_t *m){
+       int slen;
+-      err_status_t err;
++      srtp_err_status_t err;
+       bool_t is_rtp=ctx->is_rtp;
+       rtp_header_t *rtp_header=is_rtp?(rtp_header_t*)m->b_rptr:NULL;
+       rtcp_common_header_t *rtcp_header=!is_rtp?(rtcp_common_header_t*)m->b_rptr:NULL;
+@@ -109,7 +109,7 @@
+               ms_mutex_lock(&ctx->mutex);
+               if (!ctx->secured) {
+                       /*does not make sense to protect, because we don't have any key*/
+-                      err=err_status_ok;
++                      err=srtp_err_status_ok;
+                       slen = 0; /*droping packets*/
+               } else {
+                       /* defragment incoming message and enlarge the buffer for srtp to write its data */
+@@ -120,7 +120,7 @@
+       } else if (rtcp_header && (slen>RTP_FIXED_HEADER_SIZE && rtcp_header->version==2)) {
+               ms_mutex_lock(&ctx->mutex);
+               if (!ctx->secured) {
+-                      err=err_status_ok;
++                      err=srtp_err_status_ok;
+                       /*does not make sense to protect, because we don't have any key*/
+                       slen = 0; /*droping packets*/
+               } else {
+@@ -135,7 +135,7 @@
+       }
+ 
+       /* check return code from srtp_protect */
+-      if (err==err_status_ok){
++      if (err==srtp_err_status_ok){
+               return slen;
+       }
+       ortp_error("srtp_protect%s() failed (%d) for stream ctx [%p]", is_rtp?"":"_rtcp", err,ctx);
+@@ -151,7 +151,7 @@
+ }
+ static int _process_on_receive(RtpSession* session,MSSrtpStreamContext *ctx, mblk_t *m, int err){
+       int slen;
+-      err_status_t srtp_err;
++      srtp_err_status_t srtp_err;
+       bool_t is_rtp=ctx->is_rtp;
+ 
+       /* keep NON-RTP data unencrypted */
+@@ -167,7 +167,7 @@
+ 
+       slen=err;
+       srtp_err = is_rtp?srtp_unprotect(ctx->srtp,m->b_rptr,&slen):srtp_unprotect_rtcp(ctx->srtp,m->b_rptr,&slen);
+-      if (srtp_err==err_status_ok) {
++      if (srtp_err==srtp_err_status_ok) {
+               return slen;
+       } else {
+               ms_error("srtp_unprotect%s() failed (%d) on stream ctx [%p]", is_rtp?"":"_rtcp", srtp_err,ctx);
+@@ -201,7 +201,7 @@
+ 
+ 
+ static int ms_media_stream_session_fill_srtp_context(MSMediaStreamSessions *sessions, bool_t is_send, bool_t is_rtp) {
+-      err_status_t err=0;
++      srtp_err_status_t err=0;
+       RtpTransport *transport=NULL;
+       MSSrtpStreamContext* stream_ctx = get_stream_context(sessions,is_send,is_rtp);
+ 
+@@ -258,28 +258,28 @@
+ }
+ 
+ 
+-static int ms_set_srtp_crypto_policy(MSCryptoSuite suite, crypto_policy_t *policy) {
++static int ms_set_srtp_srtp_crypto_policy(MSCryptoSuite suite, srtp_crypto_policy_t *policy) {
+       switch(suite){
+               case MS_AES_128_SHA1_32:
+                       // srtp doc says: not adapted to rtcp...
+-                      crypto_policy_set_aes_cm_128_hmac_sha1_32(policy);
++                      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32(policy);
+                       break;
+               case MS_AES_128_NO_AUTH:
+                       // srtp doc says: not adapted to rtcp...
+-                      crypto_policy_set_aes_cm_128_null_auth(policy);
++                      srtp_crypto_policy_set_aes_cm_128_null_auth(policy);
+                       break;
+               case MS_NO_CIPHER_SHA1_80:
+-                      crypto_policy_set_null_cipher_hmac_sha1_80(policy);
++                      srtp_crypto_policy_set_null_cipher_hmac_sha1_80(policy);
+                       break;
+               case MS_AES_128_SHA1_80: /*default mode*/
+-                      crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
++                      srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80(policy);
+                       break;
+               case MS_AES_256_SHA1_80: // For backward compatibility
+               case MS_AES_CM_256_SHA1_80:
+-                  crypto_policy_set_aes_cm_256_hmac_sha1_80(policy);
+-            break;
++                      srtp_crypto_policy_set_aes_cm_256_hmac_sha1_80(policy);
++                      break;
+               case MS_AES_256_SHA1_32:
+-                      crypto_policy_set_aes_cm_256_hmac_sha1_32(policy);
++                      srtp_crypto_policy_set_aes_cm_256_hmac_sha1_32(policy);
+                       break;
+               case MS_CRYPTO_SUITE_INVALID:
+                       return -1;
+@@ -291,13 +291,13 @@
+ static int ms_add_srtp_stream(srtp_t srtp, MSCryptoSuite suite, uint32_t ssrc, const char* key, size_t key_length, bool_t is_send, bool_t is_rtp)
+ {
+       srtp_policy_t policy;
+-      err_status_t err;
+-      ssrc_t ssrc_conf;
++      srtp_err_status_t err;
++      srtp_ssrc_t ssrc_conf;
+ 
+       memset(&policy,0,sizeof(policy));
+ 
+       if (is_rtp) {
+-              if (ms_set_srtp_crypto_policy(suite, &policy.rtp) != 0) {
++              if (ms_set_srtp_srtp_crypto_policy(suite, &policy.rtp) != 0) {
+                       return -1;
+               }
+               /* check if key length match given policy */
+@@ -306,7 +306,7 @@
+                       return -1;
+               }
+       }else {
+-              if (ms_set_srtp_crypto_policy(suite, &policy.rtcp) != 0) {
++              if (ms_set_srtp_srtp_crypto_policy(suite, &policy.rtcp) != 0) {
+                       return -1;
+               }
+               if ((int)key_length != policy.rtcp.cipher_key_len) {
+@@ -326,7 +326,7 @@
+       policy.next = NULL;
+ 
+       err = srtp_add_stream(srtp, &policy);
+-      if (err != err_status_ok) {
++      if (err != srtp_err_status_ok) {
+               ms_error("Failed to add stream to srtp session (%d)", err);
+               return -1;
+       }
+@@ -344,7 +344,7 @@
+ int ms_srtp_init(void)
+ {
+ 
+-      err_status_t st=0;
++      srtp_err_status_t st=0;
+       ms_message("srtp init");
+       if (!srtp_init_done) {
+               st=srtp_init();
+@@ -352,7 +352,6 @@
+                       srtp_init_done++;
+               }else{
+                       ms_fatal("Couldn't initialize SRTP library: %d.", st);
+-                      err_reporting_init("mediastreamer2");
+               }
+       }else srtp_init_done++;
+       return (int)st;
+@@ -508,7 +507,7 @@
+ #else /* HAVE_SRTP */
+ 
+ typedef void* srtp_t;
+-typedef int err_status_t;
++typedef int srtp_err_status_t;
+ 
+ bool_t ms_srtp_supported(void){
+       return FALSE;

diff --git a/mediastreamer-ffmpeg.patch b/mediastreamer-ffmpeg.patch
deleted file mode 100644 (file)
index 2f3359d..0000000
--- a/mediastreamer-ffmpeg.patch
+++ /dev/null
@@ -1,11 +0,0 @@
---- mediastreamer-2.12.0/configure.ac.orig     2015-11-02 14:44:12.252105408 +0100
-+++ mediastreamer-2.12.0/configure.ac  2015-11-10 19:53:44.207627872 +0100
-@@ -82,7 +82,7 @@
- esac
- 
- if test "$strictness" = "yes" ; then
--      STRICT_OPTIONS="$STRICT_OPTIONS -Werror"
-+      STRICT_OPTIONS="$STRICT_OPTIONS -Werror -Wno-error=deprecated-declarations"
-       CFLAGS="$CFLAGS -fno-strict-aliasing"
- fi
- AC_SUBST(STRICT_OPTIONS)

diff --git a/mediastreamer-imagedir.patch b/mediastreamer-imagedir.patch
deleted file mode 100644 (file)
index d1bc200..0000000
--- a/mediastreamer-imagedir.patch
+++ /dev/null
@@ -1,24 +0,0 @@
-diff -dur -x '*~' mediastreamer-2.9.0.orig/src/Makefile.am mediastreamer-2.9.0/src/Makefile.am
---- mediastreamer-2.9.0.orig/src/Makefile.am   2013-05-30 10:09:06.000000000 +0200
-+++ mediastreamer-2.9.0/src/Makefile.am        2013-06-17 10:35:37.000000000 +0200
-@@ -363,7 +363,7 @@
- 
- AM_OBJCFLAGS=$(AM_CFLAGS)
- 
--imgdir=$(datadir)/images/
-+imgdir=$(datadir)/pixmaps/mediastreamer
- 
- img_DATA=voip/nowebcamCIF.jpg
- 
-diff -dur -x '*~' mediastreamer-2.9.0.orig/src/videofilters/nowebcam.c mediastreamer-2.9.0/src/videofilters/nowebcam.c
---- mediastreamer-2.9.0.orig/src/videofilters/nowebcam.c       2013-02-20 14:38:36.000000000 +0100
-+++ mediastreamer-2.9.0/src/videofilters/nowebcam.c    2013-06-17 10:37:20.000000000 +0200
-@@ -1783,7 +1783,7 @@
- 
- static char *def_image=NULL;
- 
--static const char *def_image_path=PACKAGE_DATA_DIR "/images/" NOWEBCAM_JPG ".jpg";
-+static const char *def_image_path="/usr/share/pixmaps/mediastreamer/" NOWEBCAM_JPG ".jpg";
- 
- 
- mblk_t *ms_load_nowebcam(MSVideoSize *reqsize, int idx){

diff --git a/mediastreamer-werror.patch b/mediastreamer-werror.patch
deleted file mode 100644 (file)
index f7f7fbf..0000000
--- a/mediastreamer-werror.patch
+++ /dev/null
@@ -1,11 +0,0 @@
---- mediastreamer-2.12.1/src/videofilters/msv4l2.c.orig        2015-10-19 13:42:10.603994267 +0200
-+++ mediastreamer-2.12.1/src/videofilters/msv4l2.c     2016-03-31 18:44:22.923935372 +0200
-@@ -343,7 +343,7 @@
- 
- static int set_camera_feature(V4l2State *s, unsigned int ctl_id, int value, const char *feature_name){
-       struct v4l2_ext_control ctl={0};
--      struct v4l2_ext_controls ctls={0};
-+      struct v4l2_ext_controls ctls={{0}};
-       struct v4l2_queryctrl queryctrl={0};
-       
-       queryctrl.id = ctl_id;

diff --git a/mediastreamer.spec b/mediastreamer.spec
index 7702d4b35bfc61118c93ad75fff3b145212bbfd1..8ae4089b03777f7ab1aa52dbaad12c6cf88df614 100644 (file)
--- a/mediastreamer.spec
+++ b/mediastreamer.spec
@@ -19,16 +19,14 @@
 Summary:       Audio/Video real-time streaming
 Summary(pl.UTF-8):     Przesyłanie strumieni audio/video w czasie rzeczywistym 
 Name:          mediastreamer
-Version:       2.12.1
-Release:       4
+Version:       2.16.1
+Release:       1
 License:       GPL v2+
 Group:         Libraries
-Source0:       http://download-mirror.savannah.gnu.org/releases/linphone/mediastreamer/%{name}-%{version}.tar.gz
-# Source0-md5: 1ca115e94a718638fbecf8352c169861
-Patch0:                %{name}-imagedir.patch
-Patch1:                %{name}-ffmpeg.patch
-Patch2:                %{name}-werror.patch
-Patch3:                ffmpeg4.patch
+Source0:       https://linphone.org/releases/sources//mediastreamer/%{name}-%{version}.tar.gz
+# Source0-md5: 15b8b129a922180855d04d58cdd08d43
+Patch0:                build.patch
+Patch1:                libsrtp2.patch
 URL:           http://www.linphone.org/technical-corner/mediastreamer2/overview
 %{?with_opengl:BuildRequires:  OpenGL-GLX-devel}
 BuildRequires: SDL-devel >= 1.2.0
@@ -37,7 +35,7 @@ BuildRequires:        SDL-devel >= 1.2.0
 BuildRequires: autoconf >= 2.53
 BuildRequires: automake >= 1:1.9
 %{?with_bcg729:BuildRequires:  bcg729-devel >= 1.0}
-%{?with_zrtp:BuildRequires:    bzrtp-devel >= 1.0.0}
+%{?with_zrtp:BuildRequires:    bzrtp-devel >= 1.0.6}
 BuildRequires: doxygen
 # libavcodec >= 51.0.0, libswscale >= 0.7.0
 BuildRequires: ffmpeg-devel
@@ -55,7 +53,7 @@ BuildRequires:        libv4l-devel
 BuildRequires: libvpx-devel >= 0.9.6
 %{?with_matroska:BuildRequires:        matroska-foundation-devel}
 BuildRequires: opus-devel >= 0.9.0
-BuildRequires: ortp-devel >= 0.24.0
+BuildRequires: ortp-devel >= 1.0.0
 BuildRequires: pkgconfig
 BuildRequires: polarssl-devel
 %{?with_portaudio:BuildRequires:       portaudio-devel}
@@ -64,18 +62,18 @@ BuildRequires:      sed >= 4.0
 BuildRequires: spandsp-devel >= 0.0.6
 BuildRequires: speex-devel >= 1:1.2-beta3
 BuildRequires: speexdsp-devel >= 1.2-beta3
-%{?with_srtp:BuildRequires:    srtp-devel}
+%{?with_srtp:BuildRequires:    libsrtp2-devel}
 BuildRequires: xorg-lib-libX11-devel
 BuildRequires: xorg-lib-libXv-devel
 BuildRequires: xxd
 %{?with_bcg729:Requires:       bcg729 >= 1.0}
-%{?with_zrtp:Requires: bzrtp >= 1.0.0}
+%{?with_zrtp:Requires: bzrtp >= 1.0.6}
 %{?with_opengl:Requires:       glew >= 1.5}
 Requires:      libtheora >= 1.0-0.alpha7
 Requires:      libupnp >= 1.6
 Requires:      libvpx >= 0.9.6
 Requires:      opus >= 0.9.0
-Requires:      ortp >= 0.24.0
+Requires:      ortp >= 1.0.0
 %{?with_pulseaudio:Requires:   pulseaudio-libs >= 0.9.21}
 Requires:      spandsp >= 0.0.6
 Requires:      speex >= 1:1.2-beta3
@@ -100,7 +98,7 @@ Requires:    %{name} = %{version}-%{release}
 %{?with_opengl:Requires:       OpenGL-devel}
 %{?with_alsa:Requires: alsa-lib-devel}
 %{?with_bcg729:Requires:       bcg729-devel >= 1.0}
-%{?with_zrtp:Requires: bzrtp-devel >= 1.0.0}
+%{?with_zrtp:Requires: bzrtp-devel >= 1.0.6}
 Requires:      ffmpeg-devel
 %{?with_opengl:Requires:       glew-devel >= 1.5}
 Requires:      libtheora-devel >= 1.0-0.alpha7
@@ -110,14 +108,14 @@ Requires: libv4l-devel
 Requires:      libvpx-devel >= 0.9.6
 %{?with_matroska:Requires:     matroska-foundation-devel}
 Requires:      opus-devel >= 0.9.0
-Requires:      ortp-devel >= 0.24.0
+Requires:      ortp-devel >= 1.0.0
 Requires:      polarssl-devel
 %{?with_portaudio:Requires:    portaudio-devel}
 %{?with_pulseaudio:Requires:   pulseaudio-devel >= 0.9.21}
 Requires:      spandsp-devel >= 0.0.6
 Requires:      speex-devel >= 1:1.2-beta3
 Requires:      speexdsp-devel >= 1.2-beta3
-%{?with_srtp:Requires: srtp-devel}
+%{?with_srtp:Requires: libsrtp2-devel}
 Requires:      xorg-lib-libX11-devel
 Requires:      xorg-lib-libXv-devel
 
@@ -144,11 +142,13 @@ Statyczne biblioteki mediastreamer.
 %setup -q
 %patch0 -p1
 %patch1 -p1
-%patch2 -p1
-%patch3 -p1
+
+[ ! -e gitversion.h ] && echo '#define MS2_GIT_VERSION "%{version}"' > src/gitversion.h
 
 %build
 %{__libtoolize}
+%{__gettextize}
+%{__intltoolize}
 %{__aclocal}
 %{__autoconf}
 %{__autoheader}