From: Hyunjun Date: Fri, 29 May 2015 06:12:19 +0000 (+0900) Subject: 1.4.5 migration and bug fixes X-Git-Tag: accepted/tizen/common/20150601.145630^0 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=933bc9394fa510d051feecc98b2c97153e8fb448;p=platform%2Fupstream%2Fgst-rtsp-server.git 1.4.5 migration and bug fixes Change-Id: I7a16b3cbcc1e63bb3e090c6050f496b0abf76ea0 --- diff --git a/ChangeLog b/ChangeLog index 9ec36fd..c8c6360 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,9 +1,86 @@ +=== release 1.4.5 === + +2014-12-18 Sebastian Dröge + + * configure.ac: + releasing 1.4.5 + +2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: unref srtp decoder when leaving bin + https://bugzilla.gnome.org/show_bug.cgi?id=739481 + +=== release 1.4.4 === + +2014-11-06 13:18:04 +0100 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.4 + +2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: mikey memory leaks + https://bugzilla.gnome.org/show_bug.cgi?id=739383 + +=== release 1.4.3 === + +2014-09-24 12:51:21 +0300 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.3 + +=== release 1.4.2 === + +2014-09-19 15:13:37 +0300 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.2 + +2014-09-08 09:26:23 +0200 Srimanta Panda + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp-media: Make sure that sequence numbers are monotonic after pause + The sequence number is not monotonic for RTP packets after pause. The + reason is basepayloader generates a randon sequence number when the + pipeline goes from ready to pause. With this fix generation of sequence + number will be monotonic when going from pause to play request. + https://bugzilla.gnome.org/show_bug.cgi?id=736017 + +2014-08-28 13:35:15 +0200 Göran Jönsson + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Protect saved clients watch with a mutex + Fixes a crash when close() is called while merging clients + in handle_tunnel(). In that case close() would destroy the + watch while it is still being used in handle_tunnel(). + https://bugzilla.gnome.org/show_bug.cgi?id=735570 + === release 1.4.1 === -2014-08-27 Sebastian Dröge +2014-08-27 15:05:07 +0300 Sebastian Dröge + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.4.1 + * gst-rtsp-server.doap: + Release 1.4.1 2014-08-05 16:12:19 +0200 Sebastian Dröge diff --git a/Makefile.am b/Makefile.am index 2272985..100b512 100644 --- a/Makefile.am +++ b/Makefile.am @@ -3,9 +3,8 @@ DISTCHECK_CONFIGURE_FLAGS=--enable-gtk-doc SUBDIRS = \ gst \ common \ - pkgconfig \ - docs \ examples \ + pkgconfig \ tests DIST_SUBDIRS = $(SUBDIRS) diff --git a/Makefile.in b/Makefile.in index 901f570..f72cc77 100644 --- a/Makefile.in +++ b/Makefile.in @@ -94,8 +94,8 @@ DIST_COMMON = $(top_srcdir)/common/release.mak \ ChangeLog $(srcdir)/Makefile.in $(srcdir)/Makefile.am \ $(top_srcdir)/configure $(am__configure_deps) \ $(srcdir)/config.h.in $(srcdir)/gst-rtsp.spec.in COPYING \ - COPYING.LIB TODO compile config.guess config.sub install-sh \ - missing ltmain.sh + COPYING.LIB TODO compile config.guess config.sub depcomp \ + install-sh missing ltmain.sh subdir = . ACLOCAL_M4 = $(top_srcdir)/aclocal.m4 am__aclocal_m4_deps = $(top_srcdir)/common/m4/as-ac-expand.m4 \ diff --git a/NEWS b/NEWS index 3062b31..cc265ee 100644 --- a/NEWS +++ b/NEWS @@ -1,2 +1,2 @@ -This is GStreamer RTSP Server 1.4.1 +This is GStreamer RTSP Server 1.4.5 diff --git a/RELEASE b/RELEASE index 0052e43..f5876ce 100644 --- a/RELEASE +++ b/RELEASE @@ -1,6 +1,5 @@ -Release notes for GStreamer RTSP Server Library 1.4.1 - +Release notes for GStreamer RTSP Server Library 1.4.5 The GStreamer team is pleased to announce a bugfix release of the stable 1.4 release series. The 1.4 release series is adding new features on top @@ -29,7 +28,7 @@ risky as a bugfix. Bugs fixed in this release - * 732644 : RTSP PLAY with specified range replies with wrong range + * 739481 : rtsp-stream: leaks srtp decoder when leaving rtpbin ==== Download ==== @@ -66,6 +65,5 @@ subscribe to the gstreamer-devel list. Contributors to this release - * Arun Raghavan - * Sebastian Dröge + * Aleix Conchillo Flaqué   \ No newline at end of file diff --git a/aclocal.m4 b/aclocal.m4 index a653c19..f429e2a 100644 --- a/aclocal.m4 +++ b/aclocal.m4 @@ -123,10 +123,9 @@ _AM_IF_OPTION([no-dependencies],, [_AM_DEPENDENCIES([CCAS])])dnl # configured tree to be moved without reconfiguration. AC_DEFUN([AM_AUX_DIR_EXPAND], -[dnl Rely on autoconf to set up CDPATH properly. -AC_PREREQ([2.50])dnl -# expand $ac_aux_dir to an absolute path -am_aux_dir=`cd $ac_aux_dir && pwd` +[AC_REQUIRE([AC_CONFIG_AUX_DIR_DEFAULT])dnl +# Expand $ac_aux_dir to an absolute path. +am_aux_dir=`cd "$ac_aux_dir" && pwd` ]) # AM_CONDITIONAL -*- Autoconf -*- diff --git a/config.sub b/config.sub index d654d03..bba4efb 100755 --- a/config.sub +++ b/config.sub @@ -2,7 +2,7 @@ # Configuration validation subroutine script. # Copyright 1992-2014 Free Software Foundation, Inc. -timestamp='2014-05-01' +timestamp='2014-09-11' # This file is free software; you can redistribute it and/or modify it # under the terms of the GNU General Public License as published by @@ -302,6 +302,7 @@ case $basic_machine in | pdp10 | pdp11 | pj | pjl \ | powerpc | powerpc64 | powerpc64le | powerpcle \ | pyramid \ + | riscv32 | riscv64 \ | rl78 | rx \ | score \ | sh | sh[1234] | sh[24]a | sh[24]aeb | sh[23]e | sh[34]eb | sheb | shbe | shle | sh[1234]le | sh3ele \ @@ -828,6 +829,10 @@ case $basic_machine in basic_machine=powerpc-unknown os=-morphos ;; + moxiebox) + basic_machine=moxie-unknown + os=-moxiebox + ;; msdos) basic_machine=i386-pc os=-msdos @@ -1373,7 +1378,7 @@ case $os in | -cygwin* | -msys* | -pe* | -psos* | -moss* | -proelf* | -rtems* \ | -mingw32* | -mingw64* | -linux-gnu* | -linux-android* \ | -linux-newlib* | -linux-musl* | -linux-uclibc* \ - | -uxpv* | -beos* | -mpeix* | -udk* \ + | -uxpv* | -beos* | -mpeix* | -udk* | -moxiebox* \ | -interix* | -uwin* | -mks* | -rhapsody* | -darwin* | -opened* \ | -openstep* | -oskit* | -conix* | -pw32* | -nonstopux* \ | -storm-chaos* | -tops10* | -tenex* | -tops20* | -its* \ diff --git a/configure b/configure index 2b2deac..e2dfa91 100755 --- a/configure +++ b/configure @@ -1,6 +1,6 @@ #! /bin/sh # Guess values for system-dependent variables and create Makefiles. -# Generated by GNU Autoconf 2.69 for GStreamer RTSP Server Library 1.4.1. +# Generated by GNU Autoconf 2.69 for GStreamer RTSP Server Library 1.4.5. # # Report bugs to . # @@ -591,8 +591,8 @@ MAKEFLAGS= # Identity of this package. PACKAGE_NAME='GStreamer RTSP Server Library' PACKAGE_TARNAME='gst-rtsp-server' -PACKAGE_VERSION='1.4.1' -PACKAGE_STRING='GStreamer RTSP Server Library 1.4.1' +PACKAGE_VERSION='1.4.5' +PACKAGE_STRING='GStreamer RTSP Server Library 1.4.5' PACKAGE_BUGREPORT='http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer' PACKAGE_URL='' @@ -1508,7 +1508,7 @@ if test "$ac_init_help" = "long"; then # Omit some internal or obsolete options to make the list less imposing. # This message is too long to be a string in the A/UX 3.1 sh. cat <<_ACEOF -\`configure' configures GStreamer RTSP Server Library 1.4.1 to adapt to many kinds of systems. +\`configure' configures GStreamer RTSP Server Library 1.4.5 to adapt to many kinds of systems. Usage: $0 [OPTION]... [VAR=VALUE]... @@ -1579,7 +1579,7 @@ fi if test -n "$ac_init_help"; then case $ac_init_help in - short | recursive ) echo "Configuration of GStreamer RTSP Server Library 1.4.1:";; + short | recursive ) echo "Configuration of GStreamer RTSP Server Library 1.4.5:";; esac cat <<\_ACEOF @@ -1766,7 +1766,7 @@ fi test -n "$ac_init_help" && exit $ac_status if $ac_init_version; then cat <<\_ACEOF -GStreamer RTSP Server Library configure 1.4.1 +GStreamer RTSP Server Library configure 1.4.5 generated by GNU Autoconf 2.69 Copyright (C) 2012 Free Software Foundation, Inc. @@ -2044,7 +2044,7 @@ cat >config.log <<_ACEOF This file contains any messages produced by compilers while running configure, to aid debugging if configure makes a mistake. -It was created by GStreamer RTSP Server Library $as_me 1.4.1, which was +It was created by GStreamer RTSP Server Library $as_me 1.4.5, which was generated by GNU Autoconf 2.69. Invocation command line was $ $0 $@ @@ -2707,8 +2707,8 @@ test "$program_suffix" != NONE && ac_script='s/[\\$]/&&/g;s/;s,x,x,$//' program_transform_name=`$as_echo "$program_transform_name" | sed "$ac_script"` -# expand $ac_aux_dir to an absolute path -am_aux_dir=`cd $ac_aux_dir && pwd` +# Expand $ac_aux_dir to an absolute path. +am_aux_dir=`cd "$ac_aux_dir" && pwd` if test x"${MISSING+set}" != xset; then case $am_aux_dir in @@ -3021,7 +3021,7 @@ fi # Define the identity of the package. PACKAGE='gst-rtsp-server' - VERSION='1.4.1' + VERSION='1.4.5' cat >>confdefs.h <<_ACEOF @@ -3232,9 +3232,9 @@ fi - PACKAGE_VERSION_MAJOR=$(echo 1.4.1 | cut -d'.' -f1) - PACKAGE_VERSION_MINOR=$(echo 1.4.1 | cut -d'.' -f2) - PACKAGE_VERSION_MICRO=$(echo 1.4.1 | cut -d'.' -f3) + PACKAGE_VERSION_MAJOR=$(echo 1.4.5 | cut -d'.' -f1) + PACKAGE_VERSION_MINOR=$(echo 1.4.5 | cut -d'.' -f2) + PACKAGE_VERSION_MICRO=$(echo 1.4.5 | cut -d'.' -f3) @@ -3245,7 +3245,7 @@ fi { $as_echo "$as_me:${as_lineno-$LINENO}: checking nano version" >&5 $as_echo_n "checking nano version... " >&6; } - NANO=$(echo 1.4.1 | cut -d'.' -f4) + NANO=$(echo 1.4.5 | cut -d'.' -f4) if test x"$NANO" = x || test "x$NANO" = "x0" ; then { $as_echo "$as_me:${as_lineno-$LINENO}: result: 0 (release)" >&5 @@ -7889,10 +7889,10 @@ fi done - GST_CURRENT=401 + GST_CURRENT=405 GST_REVISION=0 - GST_AGE=401 - GST_LIBVERSION=401:0:401 + GST_AGE=405 + GST_LIBVERSION=405:0:405 @@ -17643,7 +17643,7 @@ cat >>$CONFIG_STATUS <<\_ACEOF || ac_write_fail=1 # report actual input values of CONFIG_FILES etc. instead of their # values after options handling. ac_log=" -This file was extended by GStreamer RTSP Server Library $as_me 1.4.1, which was +This file was extended by GStreamer RTSP Server Library $as_me 1.4.5, which was generated by GNU Autoconf 2.69. Invocation command line was CONFIG_FILES = $CONFIG_FILES @@ -17709,7 +17709,7 @@ _ACEOF cat >>$CONFIG_STATUS <<_ACEOF || ac_write_fail=1 ac_cs_config="`$as_echo "$ac_configure_args" | sed 's/^ //; s/[\\""\`\$]/\\\\&/g'`" ac_cs_version="\\ -GStreamer RTSP Server Library config.status 1.4.1 +GStreamer RTSP Server Library config.status 1.4.5 configured by $0, generated by GNU Autoconf 2.69, with options \\"\$ac_cs_config\\" diff --git a/configure.ac b/configure.ac index 5c7802c..7a65c6f 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ AC_PREREQ(2.62) dnl initialize autoconf dnl when going to/from release please set the nano (fourth number) right ! dnl releases only do Wall, cvs and prerelease does Werror too -AC_INIT([GStreamer RTSP Server Library], [1.4.1], +AC_INIT([GStreamer RTSP Server Library], [1.4.5], [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer], [gst-rtsp-server]) AG_GST_INIT @@ -53,7 +53,7 @@ dnl 1.2.5 => 205 dnl 1.10.9 (who knows) => 1009 dnl dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 401, 0, 401) +AS_LIBTOOL(GST, 405, 0, 405) dnl *** required versions of GStreamer stuff *** GST_REQ=1.4.0 diff --git a/docs/libs/html/GstRTSPAuth.html b/docs/libs/html/GstRTSPAuth.html index 1b1fc1b..b51425a 100644 --- a/docs/libs/html/GstRTSPAuth.html +++ b/docs/libs/html/GstRTSPAuth.html @@ -50,7 +50,7 @@ -GTlsCertificate * +GTlsCertificate * gst_rtsp_auth_get_tls_certificate () @@ -66,7 +66,7 @@ -gchar * +gchar * gst_rtsp_auth_make_basic () @@ -90,7 +90,7 @@ -gboolean +gboolean gst_rtsp_auth_check () @@ -172,7 +172,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPAuth
@@ -213,9 +213,9 @@ gst_rtsp_auth_new (void

gst_rtsp_auth_get_tls_certificate ()

-
GTlsCertificate *
+
GTlsCertificate *
 gst_rtsp_auth_get_tls_certificate (GstRTSPAuth *auth);

-
Get the GTlsCertificate used for negotiating TLS auth
+
Get the GTlsCertificate used for negotiating TLS auth
 .

 

 
Parameters

@@ -234,8 +234,8 @@ gst_rtsp_auth_get_tls_certificate (
 

 
Returns

-
 the GTlsCertificate of auth
-. g_object_unref() after
+
 the GTlsCertificate of auth
+. g_object_unref() after
 usage. 

 
[transfer full]

 

@@ -245,7 +245,7 @@ usage. 

 
gst_rtsp_auth_set_tls_certificate ()

 
void
 gst_rtsp_auth_set_tls_certificate (GstRTSPAuth *auth,
-                                   GTlsCertificate *cert);

+                                   GTlsCertificate *cert);

Set the TLS certificate for the auth. Client connections will only be accepted when TLS is negotiated.

@@ -264,7 +264,7 @@ be accepted when TLS is negotiated.

cert

-

a GTlsCertificate.

+

a GTlsCertificate.

[transfer none][allow-none] @@ -274,9 +274,9 @@ be accepted when TLS is negotiated.

gst_rtsp_auth_make_basic ()

-
gchar *
-gst_rtsp_auth_make_basic (const gchar *user,
-                          const gchar *pass);
+
gchar *
+gst_rtsp_auth_make_basic (const gchar *user,
+                          const gchar *pass);

Construct a Basic authorisation token from user and pass .

@@ -307,7 +307,7 @@ gst_rtsp_auth_make_basic (const user :pass . -g_free() after usage.

+g_free() after usage.

[transfer full]

@@ -316,7 +316,7 @@ gst_rtsp_auth_make_basic (const

gst_rtsp_auth_add_basic ()

void
 gst_rtsp_auth_add_basic (GstRTSPAuth *auth,
-                         const gchar *basic,
+                         const gchar *basic,
                          GstRTSPToken *token);

Add a basic token for the default authentication algorithm that enables the client with privileges listed in token @@ -354,7 +354,7 @@ enables the client with privileges listed in token

gst_rtsp_auth_remove_basic ()

void
 gst_rtsp_auth_remove_basic (GstRTSPAuth *auth,
-                            const gchar *basic);
+ const gchar *basic);

Add a basic token for the default authentication algorithm that enables the client with privileges from authgroup .

@@ -384,8 +384,8 @@ enables the client with privileges from authgroup

gst_rtsp_auth_check ()

-
gboolean
-gst_rtsp_auth_check (const gchar *check);
+
gboolean
+gst_rtsp_auth_check (const gchar *check);

Check if check is allowed in the current context.

@@ -503,7 +503,7 @@ be used for unauthenticated users.

-

GObjectClass parent_class;

+

GObjectClass parent_class;

    @@ -564,7 +564,7 @@ A response should be sent on error.

Check if the client can specify TTL, destination and port pair in multicast. No response is sent when the check returns -FALSE.

+FALSE.

@@ -581,7 +581,7 @@ the media factory and retrieve the role with the same name.

GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS

#define GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS   "transport.client-settings"
-

G_TYPE_BOOLEAN, TRUE if the client can specify TTL, destination and +

G_TYPE_BOOLEAN, TRUE if the client can specify TTL, destination and port pair in multicast.

@@ -589,7 +589,7 @@ the media factory and retrieve the role with the same name.

GST_RTSP_PERM_MEDIA_FACTORY_ACCESS

#define GST_RTSP_PERM_MEDIA_FACTORY_ACCESS      "media.factory.access"
-

G_TYPE_BOOLEAN, TRUE if the media can be accessed, FALSE will +

G_TYPE_BOOLEAN, TRUE if the media can be accessed, FALSE will return a 404 Not Found error when trying to access the media.

@@ -597,7 +597,7 @@ return a 404 Not Found error when trying to access the media.

GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT

#define GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT   "media.factory.construct"
-

G_TYPE_BOOLEAN, TRUE if the media can be constructed, FALSE will +

G_TYPE_BOOLEAN, TRUE if the media can be constructed, FALSE will return a 404 Not Found error when trying to access the media.

diff --git a/docs/libs/html/GstRTSPClient.html b/docs/libs/html/GstRTSPClient.html index e02e8a7..d806b44 100644 --- a/docs/libs/html/GstRTSPClient.html +++ b/docs/libs/html/GstRTSPClient.html @@ -132,7 +132,7 @@ -gboolean +gboolean gst_rtsp_client_set_connection () @@ -140,7 +140,7 @@ -guint +guint gst_rtsp_client_attach () @@ -148,7 +148,7 @@ -gboolean +gboolean (*GstRTSPClientSendFunc) () @@ -188,7 +188,7 @@ -GList * +GList * gst_rtsp_client_session_filter () @@ -207,7 +207,7 @@ -gboolean +gboolean drop-backlog Read / Write @@ -319,7 +319,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPClient
@@ -331,7 +331,7 @@ connection is open.

accepted and it inherits the GstRTSPMountPoints, GstRTSPSessionPool, GstRTSPAuth and GstRTSPThreadPool from the server.

The client connection should be configured with the GstRTSPConnection using -gst_rtsp_client_set_connection() before it can be attached to a GMainContext +gst_rtsp_client_set_connection() before it can be attached to a GMainContext using gst_rtsp_client_attach(). From then on the client will handle requests on the connection.

Use gst_rtsp_client_session_filter() to iterate or modify all the @@ -524,7 +524,7 @@ gst_rtsp_client_get_auth (

Returns

the GstRTSPAuth of client -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -586,7 +586,7 @@ gst_rtsp_client_get_thread_pool (

Returns

the GstRTSPThreadPool of client -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -656,7 +656,7 @@ The connection object returned remains valid until the client is freed.

gst_rtsp_client_set_connection ()

-
gboolean
+
gboolean
 gst_rtsp_client_set_connection (GstRTSPClient *client,
                                 GstRTSPConnection *conn);

 
Set the GstRTSPConnection of client
@@ -687,22 +687,22 @@ gst_rtsp_client_set_connection (
 

 
Returns

-
 TRUE on success.

+
 TRUE on success.

gst_rtsp_client_attach ()

-
guint
+
guint
 gst_rtsp_client_attach (GstRTSPClient *client,
-                        GMainContext *context);

+                        GMainContext *context);

Attaches client to context . When the mainloop for context is run, the client will be dispatched. When context - is NULL, the default context will be + is NULL, the default context will be used).

This function should be called when the client properties and urls are fully configured and the client is ready to start.

@@ -722,7 +722,7 @@ configured and the client is ready to start.

context

-

a GMainContext.

+

a GMainContext.

[allow-none] @@ -737,16 +737,16 @@ configured and the client is ready to start.

GstRTSPClientSendFunc ()

-
gboolean
+
gboolean
 (*GstRTSPClientSendFunc) (GstRTSPClient *client,
                           GstRTSPMessage *message,
-                          gboolean close,
-                          gpointer user_data);

+                          gboolean close,
+                          gpointer user_data);

This callback is called when client wants to send message . When close is -TRUE, the connection should be closed when the message has been sent.

+TRUE, the connection should be closed when the message has been sent.

Parameters

@@ -781,7 +781,7 @@ configured and the client is ready to start.

Returns

-

TRUE on success.

+

TRUE on success.

@@ -791,8 +791,8 @@ configured and the client is ready to start.

void
 gst_rtsp_client_set_send_func (GstRTSPClient *client,
                                GstRTSPClientSendFunc func,
-                               gpointer user_data,
-                               GDestroyNotify notify);
+ gpointer user_data, + GDestroyNotify notify);

Set func as the callback that will be called when a new message needs to be sent to the client. user_data @@ -883,7 +883,7 @@ gst_rtsp_client_send_message (GstRTSPMessage *message);

Send a message message to the remote end. message must be a -GST_RTSP_MESSAGE_REQUEST or a GST_RTSP_MESSAGE_RESPONSE.

+GST_RTSP_MESSAGE_REQUEST or a GST_RTSP_MESSAGE_RESPONSE.

Parameters

@@ -901,7 +901,7 @@ gst_rtsp_client_send_message (

session

 +the message to or NULL.

@@ -919,7 +919,7 @@ the message to or GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client, GstRTSPSession *sess, - gpointer user_data); + gpointer user_data);

This function will be called by the gst_rtsp_client_session_filter(). An implementation should return a value of GstRTSPFilterResult.

When this function returns GST_RTSP_FILTER_REMOVE, sess @@ -931,7 +931,7 @@ from client client .

A value of GST_RTSP_FILTER_REF will add sess - to the result GList of + to the result GList of gst_rtsp_client_session_filter().

Parameters

@@ -970,10 +970,10 @@ from client

gst_rtsp_client_session_filter ()

-
GList *
+
GList *
 gst_rtsp_client_session_filter (GstRTSPClient *client,
                                 GstRTSPClientSessionFilterFunc func,
-                                gpointer user_data);

+                                gpointer user_data);

Call func for each session managed by client . The result value of func @@ -994,10 +994,10 @@ locked so no further actions on client

If func returns GST_RTSP_FILTER_REF, the session will remain in client but -will also be added with an additional ref to the result GList of this +will also be added with an additional ref to the result GList of this function..

When func - is NULL, GST_RTSP_FILTER_REF will be assumed for each session.

+ is NULL, GST_RTSP_FILTER_REF will be assumed for each session.

Parameters

a GstRTSPSession to send -the message to or NULL.

[allow-none][transfer none]
@@ -1028,10 +1028,10 @@ function..

Returns

-

a GList with all +

a GList with all sessions for which func returned GST_RTSP_FILTER_REF. After usage, each -element in the GList should be unreffed before the list is freed.

+element in the GList should be unreffed before the list is freed.

[element-type GstRTSPSession][transfer full]

@@ -1088,7 +1088,7 @@ element in the

GObjectClass parent_class;

+ @@ -1197,7 +1197,7 @@ be sent for a tunneled connection. The response can be modified. Since 1.4

Property Details

The “drop-backlog” property

-
  “drop-backlog”             gboolean
+
  “drop-backlog”             gboolean

Drop data when the backlog queue is full.

Flags: Read / Write

Default value: TRUE

@@ -1223,7 +1223,7 @@ be sent for a tunneled connection. The response can be modified. Since 1.4

The “closed” signal

void
 user_function (GstRTSPClient *gstrtspclient,
-               gpointer       user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1232,7 +1232,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1241,7 +1241,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1250,7 +1250,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1259,7 +1259,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPSession *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1268,7 +1268,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1277,7 +1277,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1286,7 +1286,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1295,8 +1295,8 @@ user_function (< 
void
 user_function (GstRTSPClient  *client,
                GstRTSPContext *session,
-               gpointer        message,
-               gpointer        user_data)
+ gpointer message, + gpointer user_data)

Parameters

GObjectClass parent_class;

    
@@ -1337,7 +1337,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1346,7 +1346,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1355,7 +1355,7 @@ user_function (< 
void
 user_function (GstRTSPClient  *gstrtspclient,
                GstRTSPContext *arg1,
-               gpointer        user_data)
+ gpointer user_data)

Flags: Run Last

diff --git a/docs/libs/html/GstRTSPMedia.html b/docs/libs/html/GstRTSPMedia.html index 83c5721..7005873 100644 --- a/docs/libs/html/GstRTSPMedia.html +++ b/docs/libs/html/GstRTSPMedia.html @@ -92,7 +92,7 @@ - + @@ -394,7 +394,7 @@ - + @@ -409,12 +409,12 @@ - + - + @@ -424,7 +424,7 @@ - + @@ -502,7 +502,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPMedia
@@ -698,11 +698,11 @@ gst_rtsp_media_get_permissions (

gst_rtsp_media_set_shared ()

void
 gst_rtsp_media_set_shared (GstRTSPMedia *media,
-                           gboolean shared);
+ gboolean shared);

Set or unset if the pipeline for media can be shared will multiple clients. When shared - is TRUE, client requests for this media will share the media + is TRUE, client requests for this media will share the media pipeline.

Parameters

@@ -730,7 +730,7 @@ pipeline.

gst_rtsp_media_is_shared ()

-
gboolean
+
gboolean
 gst_rtsp_media_is_shared (GstRTSPMedia *media);

 
Check if the pipeline for media
  can be shared between multiple clients.

@@ -751,7 +751,7 @@ gst_rtsp_media_is_shared (
 
Returns

-
 TRUE if the media can be shared between clients.

+
 TRUE if the media can be shared between clients.
@@ -760,7 +760,7 @@ gst_rtsp_media_is_shared (

gst_rtsp_media_set_reusable ()

void
 gst_rtsp_media_set_reusable (GstRTSPMedia *media,
-                             gboolean reusable);
+ gboolean reusable);

Set or unset if the pipeline for media can be reused after the pipeline has been unprepared.

@@ -790,7 +790,7 @@ been unprepared.

gst_rtsp_media_is_reusable ()

-
gboolean
+
gboolean
 gst_rtsp_media_is_reusable (GstRTSPMedia *media);

 
Check if the pipeline for media
  can be reused after an unprepare.

@@ -811,7 +811,7 @@ gst_rtsp_media_is_reusable (
 
Returns

-
 TRUE if the media can be reused

+
 TRUE if the media can be reused
@@ -938,7 +938,7 @@ gst_rtsp_media_get_protocols (

gst_rtsp_media_set_eos_shutdown ()

void
 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media,
-                                 gboolean eos_shutdown);
+ gboolean eos_shutdown);

Set or unset if an EOS event will be sent to the pipeline for media before it is unprepared.

@@ -968,7 +968,7 @@ it is unprepared.

gst_rtsp_media_is_eos_shutdown ()

-
gboolean
+
gboolean
 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);

 
Check if the pipeline for media
  will send an EOS down the pipeline before
@@ -990,7 +990,7 @@ unpreparing.

Returns

-

TRUE if the media will send EOS before unpreparing.

+

TRUE if the media will send EOS before unpreparing.

@@ -1051,7 +1051,7 @@ gst_rtsp_media_get_address_pool (

Returns

the GstRTSPAddressPool of media -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -1061,7 +1061,7 @@ usage.

gst_rtsp_media_set_buffer_size ()

void
 gst_rtsp_media_set_buffer_size (GstRTSPMedia *media,
-                                guint size);
+ guint size);

Set the kernel UDP buffer size.

Parameters

@@ -1089,7 +1089,7 @@ gst_rtsp_media_set_buffer_size (

gst_rtsp_media_get_buffer_size ()

-
guint
+
guint
 gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);

 
Get the kernel UDP buffer size.

 

@@ -1116,7 +1116,7 @@ gst_rtsp_media_get_buffer_size (
 

 
gst_rtsp_media_setup_sdp ()

-
gboolean
+
gboolean
 gst_rtsp_media_setup_sdp (GstRTSPMedia *media,
                           GstSDPMessage *sdp,
                           GstSDPInfo *info);

@@ -1161,7 +1161,7 @@ information in the SDP.

 

 

 
gst_rtsp_media_prepare ()

-
gboolean
+
gboolean
 gst_rtsp_media_prepare (GstRTSPMedia *media,
                         GstRTSPThread *thread);

 
Prepare media
@@ -1188,7 +1188,7 @@ such as the duration.
+bus handler or NULL.

@@ -1196,14 +1196,14 @@ bus handler or

Returns

-

TRUE on success.

+

TRUE on success.

gst_rtsp_media_unprepare ()

-
gboolean
+
gboolean
 gst_rtsp_media_unprepare (GstRTSPMedia *media);

 
Unprepare media
 . After this call, the media should be prepared again before
@@ -1226,7 +1226,7 @@ must be created.

Returns

-

TRUE on success.

+

TRUE on success.

@@ -1325,7 +1325,7 @@ gst_rtsp_media_get_suspend_mode (

gst_rtsp_media_suspend ()

-
gboolean
+
gboolean
 gst_rtsp_media_suspend (GstRTSPMedia *media);

 
Suspend media
 . The state of the pipeline managed by media
@@ -1352,14 +1352,14 @@ with 
 
Returns

-
 TRUE on success.

+
 TRUE on success.

gst_rtsp_media_unsuspend ()

-
gboolean
+
gboolean
 gst_rtsp_media_unsuspend (GstRTSPMedia *media);

 
Unsuspend media
  if it was in a suspended state. This method does nothing
@@ -1381,7 +1381,7 @@ when the media was not in the suspended state.

Returns

-

TRUE on success.

+

TRUE on success.

@@ -1462,7 +1462,7 @@ exists.

gst_rtsp_media_n_streams ()

-
guint
+
guint
 gst_rtsp_media_n_streams (GstRTSPMedia *media);

 
Get the number of streams in this media.

 

@@ -1491,7 +1491,7 @@ gst_rtsp_media_n_streams (
gst_rtsp_media_get_stream ()

 
GstRTSPStream *
 gst_rtsp_media_get_stream (GstRTSPMedia *media,
-                           guint idx);

+                           guint idx);

Retrieve the stream with index idx from media .

@@ -1521,7 +1521,7 @@ gst_rtsp_media_get_stream (

Returns

the GstRTSPStream at index idx -or NULL when a stream with that index did not exist.

+or NULL when a stream with that index did not exist.

[nullable][transfer none]

@@ -1530,7 +1530,7 @@ or

gst_rtsp_media_find_stream ()

GstRTSPStream *
 gst_rtsp_media_find_stream (GstRTSPMedia *media,
-                            const gchar *control);
+ const gchar *control);

Find a stream in media with control as the control uri.

@@ -1560,7 +1560,7 @@ gst_rtsp_media_find_stream (

Returns

the GstRTSPStream with control uri control -or NULL when a stream with that control did +or NULL when a stream with that control did not exist.

[nullable][transfer none]

@@ -1568,7 +1568,7 @@ not exist.

gst_rtsp_media_seek ()

-
gboolean
+
gboolean
 gst_rtsp_media_seek (GstRTSPMedia *media,
                      GstRTSPTimeRange *range);

 
Seek the pipeline of media
@@ -1600,16 +1600,16 @@ gst_rtsp_media_seek (
 
Returns

-
 TRUE on success.

+
 TRUE on success.

gst_rtsp_media_get_range_string ()

-
gchar *
+
gchar *
 gst_rtsp_media_get_range_string (GstRTSPMedia *media,
-                                 gboolean play,
+                                 gboolean play,
                                  GstRTSPRangeUnit unit);

 
Get the current range as a string. media
  must be prepared with
@@ -1643,17 +1643,17 @@ gst_rtsp_media_get_range_string (
 

 
Returns

-
 The range as a string, g_free() after usage. 

+
 The range as a string, g_free() after usage. 

 
[transfer full]

gst_rtsp_media_set_state ()

-
gboolean
+
gboolean
 gst_rtsp_media_set_state (GstRTSPMedia *media,
                           GstState state,
-                          GPtrArray *transports);

+                          GPtrArray *transports);

Set the state of media to state and for the transports in transports @@ -1681,7 +1681,7 @@ gst_rtsp_media_set_state (

transports

-
 + @@ -1689,7 +1689,7 @@ gst_rtsp_media_set_state (

Returns

-

TRUE on success.

+

TRUE on success.

@@ -1792,7 +1792,7 @@ gst_rtsp_media_get_base_time (

gst_rtsp_media_use_time_provider ()

void
 gst_rtsp_media_use_time_provider (GstRTSPMedia *media,
-                                  gboolean time_provider);
+ gboolean time_provider);

Set media to provide a GstNetTimeProvider.

@@ -1821,7 +1821,7 @@ gst_rtsp_media_use_time_provider (

gst_rtsp_media_is_time_provider ()

-
gboolean
+
gboolean
 gst_rtsp_media_is_time_provider (GstRTSPMedia *media);

 
Check if media
  can provide a GstNetTimeProvider for its pipeline clock.

@@ -1843,7 +1843,7 @@ gst_rtsp_media_is_time_provider (
 

 
Returns

-
 TRUE if media
+
 TRUE if media
 can provide a GstNetTimeProvider.

 

 

@@ -1853,7 +1853,7 @@ can provide a 
gst_rtsp_media_get_time_provider ()

 
GstNetTimeProvider *
 gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
-                                  const gchar *address,
+                                  const gchar *address,
                                   guint16 port);

 
Get the GstNetTimeProvider for the clock used by media
 . The time provider
@@ -1876,7 +1876,7 @@ will listen on address
- + @@ -1946,7 +1946,7 @@ will listen on address - + @@ -2150,7 +2150,7 @@ after creating the SDP and when the client performs a PAUSED request.

Property Details

The “buffer-size” property

-
  “buffer-size”              guint
+
  “buffer-size”              guint

The kernel UDP buffer size to use.

Flags: Read / Write

Default value: 524288

@@ -2165,7 +2165,7 @@ after creating the SDP and when the client performs a PAUSED request.

The “eos-shutdown” property

-
  “eos-shutdown”             gboolean
+
  “eos-shutdown”             gboolean

Send an EOS event to the pipeline before unpreparing.

Flags: Read / Write

Default value: FALSE

@@ -2189,7 +2189,7 @@ after creating the SDP and when the client performs a PAUSED request.

The “reusable” property

-
  “reusable”                 gboolean
+
  “reusable”                 gboolean

If this media pipeline can be reused after an unprepare.

Flags: Read / Write

Default value: FALSE

@@ -2197,7 +2197,7 @@ after creating the SDP and when the client performs a PAUSED request.

The “shared” property

-
  “shared”                   gboolean
+
  “shared”                   gboolean

If this media pipeline can be shared.

Flags: Read / Write

Default value: FALSE

@@ -2213,7 +2213,7 @@ after creating the SDP and when the client performs a PAUSED request.

The “time-provider” property

-
  “time-provider”            gboolean
+
  “time-provider”            gboolean

Use a NetTimeProvider for clients.

Flags: Read / Write

Default value: FALSE

@@ -2225,8 +2225,8 @@ after creating the SDP and when the client performs a PAUSED request.

The “new-state” signal

void
 user_function (GstRTSPMedia *gstrtspmedia,
-               gint          arg1,
-               gpointer      user_data)
+ gint arg1, + gpointer user_data)

Flags: Run Last

@@ -2235,7 +2235,7 @@ user_function (void user_function (GstRTSPMedia *gstrtspmedia, GstRTSPStream *arg1, - gpointer user_data) + gpointer user_data)

Flags: Run Last

@@ -2243,7 +2243,7 @@ user_function (

The “prepared” signal

void
 user_function (GstRTSPMedia *gstrtspmedia,
-               gpointer      user_data)
+ gpointer user_data)

Flags: Run Last

@@ -2252,7 +2252,7 @@ user_function (void user_function (GstRTSPMedia *gstrtspmedia, GstRTSPStream *arg1, - gpointer user_data) + gpointer user_data)

Flags: Run Last

@@ -2260,8 +2260,8 @@ user_function (

The “target-state” signal

void
 user_function (GstRTSPMedia *gstrtspmedia,
-               gint          arg1,
-               gpointer      user_data)
+ gint arg1, + gpointer user_data)

Flags: Run Last

@@ -2269,7 +2269,7 @@ user_function (

The “unprepared” signal

void
 user_function (GstRTSPMedia *gstrtspmedia,
-               gpointer      user_data)
+ gpointer user_data)

Flags: Run Last

diff --git a/docs/libs/html/GstRTSPMediaFactory.html b/docs/libs/html/GstRTSPMediaFactory.html index 8ea1709..703b800 100644 --- a/docs/libs/html/GstRTSPMediaFactory.html +++ b/docs/libs/html/GstRTSPMediaFactory.html @@ -52,7 +52,7 @@ - + - + +gchar * @@ -257,7 +257,7 @@ - + @@ -312,7 +312,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPMediaFactory
@@ -345,7 +345,7 @@ gst_rtsp_media_factory_new (void<

gst_rtsp_media_factory_get_launch ()

-
gchar *
+
gchar *
 gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory *factory);

 
Get the gst_parse_launch() pipeline description that will be used in the
 default prepare vmethod.

@@ -366,7 +366,7 @@ default prepare vmethod.

Returns

-

the configured launch description. g_free() after +

the configured launch description. g_free() after usage.

[transfer full]

@@ -376,7 +376,7 @@ usage.

gst_rtsp_media_factory_set_launch ()

void
 gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory *factory,
-                                   const gchar *launch);
+ const gchar *launch);

The gst_parse_launch() line to use for constructing the pipeline in the default prepare vmethod.

The pipeline description should return a GstBin as the toplevel element @@ -474,8 +474,8 @@ gst_rtsp_media_factory_set_permissions

gst_rtsp_media_factory_add_role ()

void
 gst_rtsp_media_factory_add_role (GstRTSPMediaFactory *factory,
-                                 const gchar *role,
-                                 const gchar *fieldname,
+                                 const gchar *role,
+                                 const gchar *fieldname,
                                  ...);

A convenience method to add role with fieldname @@ -522,7 +522,7 @@ will be created and the role will be added to it.

gst_rtsp_media_factory_set_shared ()

void
 gst_rtsp_media_factory_set_shared (GstRTSPMediaFactory *factory,
-                                   gboolean shared);
+ gboolean shared);

Configure if media created from this factory can be shared between clients.

Parameters

@@ -550,7 +550,7 @@ gst_rtsp_media_factory_set_shared (

gst_rtsp_media_factory_is_shared ()

-
gboolean
+
gboolean
 gst_rtsp_media_factory_is_shared (GstRTSPMediaFactory *factory);

 
Get if media created from this factory can be shared between clients.

 

@@ -570,14 +570,14 @@ gst_rtsp_media_factory_is_shared (
 

 
Returns

-
 TRUE if the media will be shared between clients.

+
 TRUE if the media will be shared between clients.

 

 

 

 

 

 
gst_rtsp_media_factory_is_eos_shutdown ()

-
gboolean
+
gboolean
 gst_rtsp_media_factory_is_eos_shutdown
                                (GstRTSPMediaFactory *factory);

 
Get if media created from this factory will have an EOS event sent to the
@@ -599,7 +599,7 @@ pipeline before shutdown.

 

 

 
Returns

-
 TRUE if the media will receive EOS before shutdown.

+
 TRUE if the media will receive EOS before shutdown.
@@ -609,7 +609,7 @@ pipeline before shutdown.

void
 gst_rtsp_media_factory_set_eos_shutdown
                                (GstRTSPMediaFactory *factory,
-                                gboolean eos_shutdown);
+ gboolean eos_shutdown);

Configure if media created from this factory will have an EOS sent to the pipeline before shutdown.

@@ -779,7 +779,7 @@ gst_rtsp_media_factory_get_address_pool

Returns

the GstRTSPAddressPool of factory -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -820,7 +820,7 @@ gst_rtsp_media_factory_set_address_pool

gst_rtsp_media_factory_get_buffer_size ()

-
guint
+
guint
 gst_rtsp_media_factory_get_buffer_size
                                (GstRTSPMediaFactory *factory);

 
Get the kernel UDP buffer size.

@@ -851,7 +851,7 @@ gst_rtsp_media_factory_get_buffer_size
 
void
 gst_rtsp_media_factory_set_buffer_size
                                (GstRTSPMediaFactory *factory,
-                                guint size);

+                                guint size);

Set the kernel UDP buffer size.

Parameters

@@ -1053,7 +1053,7 @@ can contain multiple streams like audio and video.

 - + @@ -1115,7 +1115,7 @@ implementation will configure the 'shared' property of the media.

Property Details

The “buffer-size” property

-
  “buffer-size”              guint
+
  “buffer-size”              guint

The kernel UDP buffer size to use.

Flags: Read / Write

Default value: 524288

@@ -1123,7 +1123,7 @@ implementation will configure the 'shared' property of the media.

The “eos-shutdown” property

-
  “eos-shutdown”             gboolean
+
  “eos-shutdown”             gboolean

Send EOS down the pipeline before shutting down.

Flags: Read / Write

Default value: FALSE

@@ -1131,7 +1131,7 @@ implementation will configure the 'shared' property of the media.

The “launch” property

-
  “launch”                   gchar *
+
  “launch”                   gchar *

A launch description of the pipeline.

Flags: Read / Write

Default value: NULL

@@ -1155,7 +1155,7 @@ implementation will configure the 'shared' property of the media.

The “shared” property

-
  “shared”                   gboolean
+
  “shared”                   gboolean

If media from this factory is shared.

Flags: Read / Write

Default value: FALSE

@@ -1176,7 +1176,7 @@ implementation will configure the 'shared' property of the media.

void
 user_function (GstRTSPMediaFactory *gstrtspmediafactory,
                GstRTSPMedia        *arg1,
-               gpointer             user_data)
+ gpointer user_data)

Flags: Run Last

@@ -1185,7 +1185,7 @@ user_function (void user_function (GstRTSPMediaFactory *gstrtspmediafactory, GstRTSPMedia *arg1, - gpointer user_data) + gpointer user_data)

Flags: Run Last

diff --git a/docs/libs/html/GstRTSPMountPoints.html b/docs/libs/html/GstRTSPMountPoints.html index 838066e..a685b7b 100644 --- a/docs/libs/html/GstRTSPMountPoints.html +++ b/docs/libs/html/GstRTSPMountPoints.html @@ -74,7 +74,7 @@

Returns

the GstRTSPMediaFactory for path . -g_object_unref() after usage.

+g_object_unref() after usage.

[transfer full]

gst_rtsp_mount_points_make_path ()

-
gchar *
+
gchar *
 gst_rtsp_mount_points_make_path (GstRTSPMountPoints *mounts,
                                  const GstRTSPUrl *url);

 
Make a path string from url
@@ -294,7 +294,7 @@ gst_rtsp_mount_points_make_path (
 
Returns

 
 a path string for url
-, g_free() after usage. 

+, g_free() after usage. 

 
[transfer full]
@@ -327,7 +327,7 @@ gst_rtsp_mount_points_make_path ( - + diff --git a/docs/libs/html/GstRTSPServer.html b/docs/libs/html/GstRTSPServer.html index 435205c..e9547e9 100644 --- a/docs/libs/html/GstRTSPServer.html +++ b/docs/libs/html/GstRTSPServer.html @@ -52,7 +52,7 @@ +gchar * - + - + @@ -262,7 +262,7 @@ +gchar * @@ -311,7 +311,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPServer
@@ -327,8 +327,8 @@ network (0.0.0.0) and port 8554.

The server will require an SSL connection when a TLS certificate has been set in the auth object with gst_rtsp_auth_set_tls_certificate().

To start the server, use gst_rtsp_server_attach() to attach it to a -GMainContext. For more control, gst_rtsp_server_create_source() and -gst_rtsp_server_create_socket() can be used to get a GSource and GSocket +GMainContext. For more control, gst_rtsp_server_create_source() and +gst_rtsp_server_create_socket() can be used to get a GSource and GSocket respectively.

gst_rtsp_server_transfer_connection() can be used to transfer an existing socket to the RTSP server, for example from an HTTP server.

@@ -357,7 +357,7 @@ gst_rtsp_server_new (void<

gst_rtsp_server_get_address ()

-
gchar *
+
gchar *
 gst_rtsp_server_get_address (GstRTSPServer *server);

 
Get the address on which the server will accept connections.

 

@@ -377,7 +377,7 @@ gst_rtsp_server_get_address (
 
Returns

-
 the server address. g_free() after usage. 

+
 the server address. g_free() after usage. 

 
[transfer full]
@@ -386,7 +386,7 @@ gst_rtsp_server_get_address (

gst_rtsp_server_set_address ()

void
 gst_rtsp_server_set_address (GstRTSPServer *server,
-                             const gchar *address);
+ const gchar *address);

Configure server to accept connections on the given address.

This function must be called before the server is bound.

@@ -416,7 +416,7 @@ gst_rtsp_server_set_address (

gst_rtsp_server_get_service ()

-
gchar *
+
gchar *
 gst_rtsp_server_get_service (GstRTSPServer *server);

 
Get the service on which the server will accept connections.

 

@@ -436,7 +436,7 @@ gst_rtsp_server_get_service (
 
Returns

-
 the service. use g_free() after usage. 

+
 the service. use g_free() after usage. 

 
[transfer full]

 

 
@@ -445,7 +445,7 @@ gst_rtsp_server_get_service (
gst_rtsp_server_set_service ()

 
void
 gst_rtsp_server_set_service (GstRTSPServer *server,
-                             const gchar *service);

+                             const gchar *service);

Configure server to accept connections on the given service. service @@ -482,7 +482,7 @@ port. The actual used port can be retrieved with

gst_rtsp_server_get_backlog ()

-
gint
+
gint
 gst_rtsp_server_get_backlog (GstRTSPServer *server);

 
The maximum amount of queued requests for the server.

 

@@ -511,7 +511,7 @@ gst_rtsp_server_get_backlog (
gst_rtsp_server_set_backlog ()

 
void
 gst_rtsp_server_set_backlog (GstRTSPServer *server,
-                             gint backlog);

+                             gint backlog);

configure the maximum amount of requests that may be queued for the server.

This function must be called before the server is bound.

@@ -590,7 +590,7 @@ gst_rtsp_server_get_mount_points (

Returns

the GstRTSPMountPoints of server -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -651,7 +651,7 @@ gst_rtsp_server_get_session_pool (

Returns

-

the GstRTSPSessionPool used for sessions. g_object_unref() after +

the GstRTSPSessionPool used for sessions. g_object_unref() after usage.

[transfer full]

@@ -713,7 +713,7 @@ gst_rtsp_server_get_thread_pool (

Returns

the GstRTSPThreadPool of server -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -775,7 +775,7 @@ gst_rtsp_server_get_auth (

Returns

the GstRTSPAuth of server -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -815,12 +815,12 @@ gst_rtsp_server_set_auth (

gst_rtsp_server_transfer_connection ()

-
gboolean
+
gboolean
 gst_rtsp_server_transfer_connection (GstRTSPServer *server,
-                                     GSocket *socket,
-                                     const gchar *ip,
-                                     gint port,
-                                     const gchar *initial_buffer);

+                                     GSocket *socket,
+                                     const gchar *ip,
+                                     gint port,
+                                     const gchar *initial_buffer);

Take an existing network socket and use it for an RTSP connection. This is used when transferring a socket from an HTTP server which should be used as an RTSP over HTTP tunnel. The initial_buffer @@ -872,11 +872,11 @@ that the HTTP server read from the socket while parsing the HTTP header.

gst_rtsp_server_io_func ()

-
gboolean
-gst_rtsp_server_io_func (GSocket *socket,
-                         GIOCondition condition,
+
gboolean
+gst_rtsp_server_io_func (GSocket *socket,
+                         GIOCondition condition,
                          GstRTSPServer *server);

-
A default GSocketSourceFunc that creates a new GstRTSPClient to accept and handle a
+
A default GSocketSourceFunc that creates a new GstRTSPClient to accept and handle a
 new connection on socket
  or server
 .

@@ -891,7 +891,7 @@ new connection on socket
- + @@ -917,11 +917,11 @@ new connection on socket

gst_rtsp_server_create_socket ()

-
GSocket *
+
GSocket *
 gst_rtsp_server_create_socket (GstRTSPServer *server,
-                               GCancellable *cancellable,
-                               GError **error);

-
Create a GSocket for server
+                               GCancellable *cancellable,
+                               GError **error);
+

Create a GSocket for server . The socket will listen on the configured service.

@@ -940,12 +940,12 @@ configured service.

 - + - + @@ -953,8 +953,8 @@ configured service.

Returns

-

the GSocket for server -or NULL when an error +

the GSocket for server +or NULL when an error occurred.

[transfer full]

@@ -962,18 +962,18 @@ occurred.

gst_rtsp_server_create_source ()

-
GSource *
+
GSource *
 gst_rtsp_server_create_source (GstRTSPServer *server,
-                               GCancellable *cancellable,
-                               GError **error);

-
Create a GSource for server
+                               GCancellable *cancellable,
+                               GError **error);
+

Create a GSource for server . The new source will have a default -GSocketSourceFunc of gst_rtsp_server_io_func().

+GSocketSourceFunc of gst_rtsp_server_io_func().

cancellable - if not NULL can be used to cancel the source, which will cause + if not NULL can be used to cancel the source, which will cause the source to trigger, reporting the current condition (which is likely 0 unless cancellation happened at the same time as a condition change). You can -check for this in the callback using g_cancellable_is_cancelled().

+check for this in the callback using g_cancellable_is_cancelled().

Parameters

-gboolean +gboolean gst_rtsp_media_is_shared () @@ -108,7 +108,7 @@
-gboolean +gboolean gst_rtsp_media_is_reusable () @@ -156,7 +156,7 @@
-gboolean +gboolean gst_rtsp_media_is_eos_shutdown () @@ -188,7 +188,7 @@
-guint +guint gst_rtsp_media_get_buffer_size () @@ -196,7 +196,7 @@
-gboolean +gboolean gst_rtsp_media_setup_sdp () @@ -204,7 +204,7 @@
-gboolean +gboolean gst_rtsp_media_prepare () @@ -212,7 +212,7 @@
-gboolean +gboolean gst_rtsp_media_unprepare () @@ -244,7 +244,7 @@
-gboolean +gboolean gst_rtsp_media_suspend () @@ -252,7 +252,7 @@
-gboolean +gboolean gst_rtsp_media_unsuspend () @@ -276,7 +276,7 @@
-guint +guint gst_rtsp_media_n_streams () @@ -300,7 +300,7 @@
-gboolean +gboolean gst_rtsp_media_seek () @@ -308,7 +308,7 @@
-gchar * +gchar * gst_rtsp_media_get_range_string () @@ -316,7 +316,7 @@
-gboolean +gboolean gst_rtsp_media_set_state () @@ -356,7 +356,7 @@
-gboolean +gboolean gst_rtsp_media_is_time_provider () @@ -383,7 +383,7 @@
guintguint buffer-size Read / Write
Read / Write / Construct Only
gbooleangboolean eos-shutdown Read / Write
Read / Write
gbooleangboolean reusable Read / Write
gbooleangboolean shared Read / Write
Read / Write
gbooleangboolean time-provider Read / Write

thread

a GstRTSPThread to run the -bus handler or NULL.

[transfer full][allow-none]

a GPtrArray of GstRTSPStreamTransport pointers.

a GPtrArray of GstRTSPStreamTransport pointers.

[transfer none][element-type GstRtspServer.RTSPStreamTransport]

address

an address or NULL.

an address or NULL.

[allow-none]

GObjectClass parent_class;

GObjectClass parent_class;

    
-gchar * +gchar * gst_rtsp_media_factory_get_launch () @@ -100,7 +100,7 @@
-gboolean +gboolean gst_rtsp_media_factory_is_shared () @@ -108,7 +108,7 @@
-gboolean +gboolean gst_rtsp_media_factory_is_eos_shutdown () @@ -172,7 +172,7 @@
-guint +guint gst_rtsp_media_factory_get_buffer_size () @@ -231,18 +231,18 @@
guintguint buffer-size Read / Write
gbooleangboolean eos-shutdown Read / Write
-gchar *launch Read / Write
Read / Write
gbooleangboolean shared Read / Write

GObjectClass parent_class;

GObjectClass parent_class;

    
-gchar * +gchar * gst_rtsp_mount_points_make_path () @@ -104,7 +104,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPMountPoints
@@ -138,7 +138,7 @@ gst_rtsp_mount_points_new (void

gst_rtsp_mount_points_add_factory ()

void
 gst_rtsp_mount_points_add_factory (GstRTSPMountPoints *mounts,
-                                   const gchar *path,
+                                   const gchar *path,
                                    GstRTSPMediaFactory *factory);

Attach factory to the mount point path @@ -183,7 +183,7 @@ used after calling this function.

gst_rtsp_mount_points_remove_factory ()

void
 gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints *mounts,
-                                      const gchar *path);
+ const gchar *path);

Remove the GstRTSPMediaFactory associated with path in mounts .

@@ -215,13 +215,13 @@ gst_rtsp_mount_points_remove_factory (

gst_rtsp_mount_points_match ()

GstRTSPMediaFactory *
 gst_rtsp_mount_points_match (GstRTSPMountPoints *mounts,
-                             const gchar *path,
-                             gint *matched);
+ const gchar *path, + gint *matched);

Find the factory in mounts that has the longest match with path .

If matched - is NULL, path + is NULL, path will match the factory exactly otherwise the amount of characters that matched is returned in matched .

@@ -257,14 +257,14 @@ matched.

GObjectClass parent_class;

GObjectClass parent_class;

    
-gchar * +gchar * gst_rtsp_server_get_address () @@ -68,7 +68,7 @@
-gchar * +gchar * gst_rtsp_server_get_service () @@ -84,7 +84,7 @@
-gint +gint gst_rtsp_server_get_backlog () @@ -172,7 +172,7 @@
-gboolean +gboolean gst_rtsp_server_transfer_connection () @@ -180,7 +180,7 @@
-gboolean +gboolean gst_rtsp_server_io_func () @@ -188,7 +188,7 @@
-GSocket * +GSocket * gst_rtsp_server_create_socket () @@ -196,7 +196,7 @@
-GSource * +GSource * gst_rtsp_server_create_source () @@ -204,7 +204,7 @@
-guint +guint gst_rtsp_server_attach () @@ -220,7 +220,7 @@
-GList * +GList * gst_rtsp_server_client_filter () @@ -240,17 +240,17 @@
-gchar *address Read / Write
gintgint backlog Read / Write
gintgint bound-port Read
-gchar *service Read / Write

socket

a GSocket

a GSocket

  

cancellable

a GCancellable.

a GCancellable.

[allow-none]

error

a GError.

a GError.

[out]
@@ -990,12 +990,12 @@ check for this in the callback using

cancellable

- + - + @@ -1003,24 +1003,24 @@ check for this in the callback using

Returns

-

the GSource for server -or NULL when an error -occurred. Free with g_source_unref().

+

the GSource for server +or NULL when an error +occurred. Free with g_source_unref().

[transfer full]

gst_rtsp_server_attach ()

-
guint
+
guint
 gst_rtsp_server_attach (GstRTSPServer *server,
-                        GMainContext *context);

+                        GMainContext *context);

Attaches server to context . When the mainloop for context is run, the server will be dispatched. When context - is NULL, the default context will be + is NULL, the default context will be used).

This function should be called when the server properties and urls are fully configured and the server is ready to start.

@@ -1040,7 +1040,7 @@ configured and the server is ready to start.

 - + @@ -1058,7 +1058,7 @@ configured and the server is ready to start.

GstRTSPFilterResult
 (*GstRTSPServerClientFilterFunc) (GstRTSPServer *server,
                                   GstRTSPClient *client,
-                                  gpointer user_data);
+ gpointer user_data);

This function will be called by the gst_rtsp_server_client_filter(). An implementation should return a value of GstRTSPFilterResult.

When this function returns GST_RTSP_FILTER_REMOVE, client @@ -1070,7 +1070,7 @@ from server server .

A value of GST_RTSP_FILTER_REF will add client - to the result GList of + to the result GList of gst_rtsp_server_client_filter().

Parameters

@@ -1109,10 +1109,10 @@ from server

gst_rtsp_server_client_filter ()

-
GList *
+
GList *
 gst_rtsp_server_client_filter (GstRTSPServer *server,
                                GstRTSPServerClientFilterFunc func,
-                               gpointer user_data);

+                               gpointer user_data);

Call func for each client managed by server . The result value of func @@ -1133,10 +1133,10 @@ locked so no further actions on server

If func returns GST_RTSP_FILTER_REF, the client will remain in server but -will also be added with an additional ref to the result GList of this +will also be added with an additional ref to the result GList of this function..

When func - is NULL, GST_RTSP_FILTER_REF will be assumed for each client.

+ is NULL, GST_RTSP_FILTER_REF will be assumed for each client.

Parameters

a GCancellable or NULL.

a GCancellable or NULL.

[allow-none]

error

a GError.

a GError.

[out]

context

a GMainContext.

a GMainContext.

[allow-none]
@@ -1167,10 +1167,10 @@ function..

Returns

-

a GList with all +

a GList with all clients for which func returned GST_RTSP_FILTER_REF. After usage, each -element in the GList should be unreffed before the list is freed.

+element in the GList should be unreffed before the list is freed.

[element-type GstRTSPClient][transfer full]

@@ -1206,7 +1206,7 @@ it.

 - + @@ -1233,7 +1233,7 @@ mount-points, auth, session-pool and thread-pool on the client.

Property Details

The “address” property

-
  “address”                  gchar *
+
  “address”                  gchar *

The address the server uses to listen on.

Flags: Read / Write

Default value: "0.0.0.0"

@@ -1241,7 +1241,7 @@ mount-points, auth, session-pool and thread-pool on the client.

The “backlog” property

-
  “backlog”                  gint
+
  “backlog”                  gint

The maximum length to which the queue of pending connections may grow.

Flags: Read / Write

Allowed values: >= 0

@@ -1250,7 +1250,7 @@ mount-points, auth, session-pool and thread-pool on the client.

The “bound-port” property

-
  “bound-port”               gint
+
  “bound-port”               gint

The port number the server is listening on.

Flags: Read

Allowed values: [-1,65535]

@@ -1266,7 +1266,7 @@ mount-points, auth, session-pool and thread-pool on the client.

The “service” property

-
  “service”                  gchar *
+
  “service”                  gchar *

The service or port number the server uses to listen on.

Flags: Read / Write

Default value: "8554"

@@ -1286,7 +1286,7 @@ mount-points, auth, session-pool and thread-pool on the client.

void
 user_function (GstRTSPServer *gstrtspserver,
                GstRTSPClient *arg1,
-               gpointer       user_data)
+ gpointer user_data)

Flags: Run Last

diff --git a/docs/libs/html/GstRTSPSession.html b/docs/libs/html/GstRTSPSession.html index e61e4a9..38cc654 100644 --- a/docs/libs/html/GstRTSPSession.html +++ b/docs/libs/html/GstRTSPSession.html @@ -50,7 +50,7 @@
- +gchar * - + - + @@ -216,7 +216,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPSession
@@ -244,7 +244,7 @@ with a url. Use

gst_rtsp_session_new ()

GstRTSPSession *
-gst_rtsp_session_new (const gchar *sessionid);
+gst_rtsp_session_new (const gchar *sessionid);

Create a new GstRTSPSession instance with sessionid .

@@ -271,7 +271,7 @@ gst_rtsp_session_new (const

gst_rtsp_session_get_sessionid ()

-
const gchar *
+
const gchar *
 gst_rtsp_session_get_sessionid (GstRTSPSession *session);

 
Get the sessionid of session
 .

@@ -302,7 +302,7 @@ is alive. 

 

 

 
gst_rtsp_session_get_header ()

-
gchar *
+
gchar *
 gst_rtsp_session_get_header (GstRTSPSession *session);

 
Get the string that can be placed in the Session header field.

 

@@ -323,7 +323,7 @@ gst_rtsp_session_get_header (
 
Returns

 
 the Session header of session
-. g_free() after usage. 

+. g_free() after usage. 

 
[transfer full]

 

 

@@ -332,7 +332,7 @@ gst_rtsp_session_get_header (
gst_rtsp_session_set_timeout ()

 
void
 gst_rtsp_session_set_timeout (GstRTSPSession *session,
-                              guint timeout);

+                              guint timeout);

Configure session for a timeout of timeout seconds. The session will be @@ -364,7 +364,7 @@ cleaned up when there is no activity for timeout

gst_rtsp_session_get_timeout ()

-
guint
+
guint
 gst_rtsp_session_get_timeout (GstRTSPSession *session);

 
Get the timeout value of session
 .

@@ -462,9 +462,9 @@ amount of time as 
 
gst_rtsp_session_next_timeout ()

-
gint
+
gint
 gst_rtsp_session_next_timeout (GstRTSPSession *session,
-                               GTimeVal *now);

+                               GTimeVal *now);

 
Get the amount of milliseconds till the session will expire.

 

 
Parameters

@@ -497,9 +497,9 @@ gst_rtsp_session_next_timeout (
 

 
gst_rtsp_session_is_expired ()

-
gboolean
+
gboolean
 gst_rtsp_session_is_expired (GstRTSPSession *session,
-                             GTimeVal *now);

+                             GTimeVal *now);

 
Check if session
  timeout out.

 

@@ -526,7 +526,7 @@ gst_rtsp_session_is_expired (
 
Returns

-
 TRUE if session
+
 TRUE if session
 timed out

 

 

@@ -536,7 +536,7 @@ timed out

 
gst_rtsp_session_manage_media ()

 
GstRTSPSessionMedia *
 gst_rtsp_session_manage_media (GstRTSPSession *sess,
-                               const gchar *path,
+                               const gchar *path,
                                GstRTSPMedia *media);

 
Manage the media object obj
  in sess
@@ -582,7 +582,7 @@ object. 

 

 

 
gst_rtsp_session_release_media ()

-
gboolean
+
gboolean
 gst_rtsp_session_release_media (GstRTSPSession *sess,
                                 GstRTSPSessionMedia *media);

 
Release the managed media
@@ -612,7 +612,7 @@ gst_rtsp_session_release_media (
 

 
Returns

-
 TRUE if there are more media session left in sess
+
 TRUE if there are more media session left in sess
 .

 

 

@@ -622,8 +622,8 @@ gst_rtsp_session_release_media (
gst_rtsp_session_get_media ()

 
GstRTSPSessionMedia *
 gst_rtsp_session_get_media (GstRTSPSession *sess,
-                            const gchar *path,
-                            gint *matched);

+                            const gchar *path,
+                            gint *matched);

 
Get the session media for path
 . matched
  will contain the number of matched
@@ -670,7 +670,7 @@ in sess
 
GstRTSPFilterResult
 (*GstRTSPSessionFilterFunc) (GstRTSPSession *sess,
                              GstRTSPSessionMedia *media,
-                             gpointer user_data);

+                             gpointer user_data);

This function will be called by the gst_rtsp_session_filter(). An implementation should return a value of GstRTSPFilterResult.

When this function returns GST_RTSP_FILTER_REMOVE, media @@ -682,7 +682,7 @@ from sess sess .

A value of GST_RTSP_FILTER_REF will add media - to the result GList of + to the result GList of gst_rtsp_session_filter().

Parameters

@@ -721,10 +721,10 @@ from sess

gst_rtsp_session_filter ()

-
GList *
+
GList *
 gst_rtsp_session_filter (GstRTSPSession *sess,
                          GstRTSPSessionFilterFunc func,
-                         gpointer user_data);

+                         gpointer user_data);

Call func for each media in sess . The result value of func @@ -745,10 +745,10 @@ locked so no further actions on sess

If func returns GST_RTSP_FILTER_REF, the media will remain in sess but -will also be added with an additional ref to the result GList of this +will also be added with an additional ref to the result GList of this function..

When func - is NULL, GST_RTSP_FILTER_REF will be assumed for all media.

+ is NULL, GST_RTSP_FILTER_REF will be assumed for all media.

Parameters

GObjectClass parent_class;

GObjectClass parent_class;

    
const gchar * +const gchar * gst_rtsp_session_get_sessionid () @@ -58,7 +58,7 @@
-gchar * +gchar * gst_rtsp_session_get_header () @@ -74,7 +74,7 @@
-guint +guint gst_rtsp_session_get_timeout () @@ -106,7 +106,7 @@
-gint +gint gst_rtsp_session_next_timeout () @@ -114,7 +114,7 @@
-gboolean +gboolean gst_rtsp_session_is_expired () @@ -130,7 +130,7 @@
-gboolean +gboolean gst_rtsp_session_release_media () @@ -154,7 +154,7 @@
-GList * +GList * gst_rtsp_session_filter () @@ -174,17 +174,17 @@
-gchar *sessionid Read / Write / Construct Only
guintguint timeout Read / Write
gbooleangboolean timeout-always-visible Read / Write
@@ -782,7 +782,7 @@ function..

a GList with all media for which func returned GST_RTSP_FILTER_REF. After usage, each -element in the GList should be unreffed before the list is freed.

+element in the GList should be unreffed before the list is freed.

[element-type GstRTSPSessionMedia][transfer full]

@@ -847,7 +847,7 @@ identified with the url of a media.

Property Details

The “sessionid” property

-
  “sessionid”                gchar *
+
  “sessionid”                gchar *

the session id.

Flags: Read / Write / Construct Only

Default value: NULL

@@ -855,7 +855,7 @@ identified with the url of a media.

The “timeout” property

-
  “timeout”                  guint
+
  “timeout”                  guint

the timeout of the session (0 = never).

Flags: Read / Write

Default value: 60

@@ -863,7 +863,7 @@ identified with the url of a media.

The “timeout-always-visible” property

-
  “timeout-always-visible”   gboolean
+
  “timeout-always-visible”   gboolean

timeout always visible in header.

Flags: Read / Write

Default value: FALSE

diff --git a/docs/libs/html/GstRTSPSessionPool.html b/docs/libs/html/GstRTSPSessionPool.html index a598a0c..718aef7 100644 --- a/docs/libs/html/GstRTSPSessionPool.html +++ b/docs/libs/html/GstRTSPSessionPool.html @@ -52,7 +52,7 @@
- + @@ -192,7 +192,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPSessionPool
@@ -220,7 +220,7 @@ gst_rtsp_session_pool_new (voidCreate a new GstRTSPSessionPool instance.

Returns

-

A new GstRTSPSessionPool. g_object_unref() after +

A new GstRTSPSessionPool. g_object_unref() after usage.

[transfer full]

@@ -228,7 +228,7 @@ usage.

gst_rtsp_session_pool_get_max_sessions ()

-
guint
+
guint
 gst_rtsp_session_pool_get_max_sessions
                                (GstRTSPSessionPool *pool);

 
Get the maximum allowed number of sessions in pool
@@ -261,7 +261,7 @@ amount of sessions.

 
void
 gst_rtsp_session_pool_set_max_sessions
                                (GstRTSPSessionPool *pool,
-                                guint max);

+                                guint max);

Configure the maximum allowed number of sessions in pool to max . @@ -292,7 +292,7 @@ A value of 0 means an unlimited amount of sessions.

gst_rtsp_session_pool_get_n_sessions ()

-
guint
+
guint
 gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool *pool);

 
Get the amount of active sessions in pool
 .

@@ -351,7 +351,7 @@ gst_rtsp_session_pool_create (
gst_rtsp_session_pool_find ()

 
GstRTSPSession *
 gst_rtsp_session_pool_find (GstRTSPSessionPool *pool,
-                            const gchar *sessionid);

+                            const gchar *sessionid);

Find the session with sessionid in pool . The access time of the session @@ -381,14 +381,14 @@ will be updated with

Returns

the GstRTSPSession with sessionid -or NULL when the session did not exist. g_object_unref() after usage.

+or NULL when the session did not exist. g_object_unref() after usage.

[transfer full][nullable]

gst_rtsp_session_pool_remove ()

-
gboolean
+
gboolean
 gst_rtsp_session_pool_remove (GstRTSPSessionPool *pool,
                               GstRTSPSession *sess);

 
Remove sess
@@ -419,14 +419,14 @@ gst_rtsp_session_pool_remove (
 
Returns

-
 TRUE if the session was found and removed.

+
 TRUE if the session was found and removed.

gst_rtsp_session_pool_cleanup ()

-
guint
+
guint
 gst_rtsp_session_pool_cleanup (GstRTSPSessionPool *pool);

 
Inspect all the sessions in pool
  and remove the sessions that are inactive
@@ -455,9 +455,9 @@ for more than their timeout.

 

 

 
GstRTSPSessionPoolFunc ()

-
gboolean
+
gboolean
 (*GstRTSPSessionPoolFunc) (GstRTSPSessionPool *pool,
-                           gpointer user_data);

+                           gpointer user_data);

 
The function that will be called from the GSource watch on the session pool.

 
The function will be called when the pool must be cleaned up because one or
 more sessions timed out.

@@ -485,16 +485,16 @@ more sessions timed out.

 

 

 
Returns

-
 FALSE if the source should be removed.

+
 FALSE if the source should be removed.

gst_rtsp_session_pool_create_watch ()

-
GSource *
+
GSource *
 gst_rtsp_session_pool_create_watch (GstRTSPSessionPool *pool);

-
Create a GSource that will be dispatched when the session should be cleaned
+
Create a GSource that will be dispatched when the session should be cleaned
 up.

 

 
Parameters

@@ -513,7 +513,7 @@ up.

 

 

 
Returns

-
 a GSource. 

+
 a GSource. 

 
[transfer full]
@@ -523,7 +523,7 @@ up.

GstRTSPFilterResult
 (*GstRTSPSessionPoolFilterFunc) (GstRTSPSessionPool *pool,
                                  GstRTSPSession *session,
-                                 gpointer user_data);
+ gpointer user_data);

This function will be called by the gst_rtsp_session_pool_filter(). An implementation should return a value of GstRTSPFilterResult.

When this function returns GST_RTSP_FILTER_REMOVE, session @@ -535,7 +535,7 @@ from pool pool .

A value of GST_RTSP_FILTER_REF will add session - to the result GList of + to the result GList of gst_rtsp_session_pool_filter().

Parameters

@@ -574,10 +574,10 @@ from pool

gst_rtsp_session_pool_filter ()

-
GList *
+
GList *
 gst_rtsp_session_pool_filter (GstRTSPSessionPool *pool,
                               GstRTSPSessionPoolFilterFunc func,
-                              gpointer user_data);

+                              gpointer user_data);

Call func for each session in pool . The result value of func @@ -601,7 +601,7 @@ expired state with gst_rtsp_session_set_expired() will also be added with an additional ref to the result GList of this function..

When func - is NULL, GST_RTSP_FILTER_REF will be assumed for all sessions.

+ is NULL, GST_RTSP_FILTER_REF will be assumed for all sessions.

Parameters

-guint +guint gst_rtsp_session_pool_get_max_sessions () @@ -68,7 +68,7 @@
-guint +guint gst_rtsp_session_pool_get_n_sessions () @@ -92,7 +92,7 @@
-gboolean +gboolean gst_rtsp_session_pool_remove () @@ -100,7 +100,7 @@
-guint +guint gst_rtsp_session_pool_cleanup () @@ -108,7 +108,7 @@
-gboolean +gboolean (*GstRTSPSessionPoolFunc) () @@ -116,7 +116,7 @@
-GSource * +GSource * gst_rtsp_session_pool_create_watch () @@ -132,7 +132,7 @@
-GList * +GList * gst_rtsp_session_pool_filter () @@ -150,7 +150,7 @@
guintguint max-sessions Read / Write
@@ -672,7 +672,7 @@ attached to a - + @@ -701,7 +701,7 @@ custom session ids and should not check if the session exists.

Property Details

The “max-sessions” property

-
  “max-sessions”             guint
+
  “max-sessions”             guint

the maximum amount of sessions (0 = unlimited).

Flags: Read / Write

Default value: 0

@@ -714,7 +714,7 @@ custom session ids and should not check if the session exists.

void
 user_function (GstRTSPSessionPool *gstrtspsessionpool,
                GstRTSPSession     *arg1,
-               gpointer            user_data)
+ gpointer user_data)

Flags: Run Last

diff --git a/docs/libs/html/GstRTSPStream.html b/docs/libs/html/GstRTSPStream.html index fcbe74c..fee8a7d 100644 --- a/docs/libs/html/GstRTSPStream.html +++ b/docs/libs/html/GstRTSPStream.html @@ -50,7 +50,7 @@

GObjectClass parent_class;

GObjectClass parent_class;

    
-guint +guint gst_rtsp_stream_get_index () @@ -66,7 +66,7 @@
-gchar * +gchar * gst_rtsp_stream_get_control () @@ -82,7 +82,7 @@
-gboolean +gboolean gst_rtsp_stream_has_control () @@ -90,7 +90,7 @@
-guint +guint gst_rtsp_stream_get_mtu () @@ -106,7 +106,7 @@
-gint +gint gst_rtsp_stream_get_dscp_qos () @@ -154,7 +154,7 @@
-gboolean +gboolean gst_rtsp_stream_is_transport_supported () @@ -186,7 +186,7 @@
-gboolean +gboolean gst_rtsp_stream_join_bin () @@ -194,7 +194,7 @@
-gboolean +gboolean gst_rtsp_stream_leave_bin () @@ -218,7 +218,7 @@
-GObject * +GObject * gst_rtsp_stream_get_rtpsession () @@ -234,7 +234,7 @@
-gboolean +gboolean gst_rtsp_stream_get_rtpinfo () @@ -250,7 +250,7 @@
-guint +guint gst_rtsp_stream_get_pt () @@ -274,7 +274,7 @@
-gboolean +gboolean gst_rtsp_stream_add_transport () @@ -282,7 +282,7 @@
-gboolean +gboolean gst_rtsp_stream_remove_transport () @@ -290,7 +290,7 @@
-GSocket * +GSocket * gst_rtsp_stream_get_rtp_socket () @@ -298,7 +298,7 @@
-GSocket * +GSocket * gst_rtsp_stream_get_rtcp_socket () @@ -306,7 +306,7 @@
-gboolean +gboolean gst_rtsp_stream_set_blocked () @@ -314,7 +314,7 @@
-gboolean +gboolean gst_rtsp_stream_is_blocking () @@ -322,7 +322,7 @@
-gboolean +gboolean gst_rtsp_stream_update_crypto () @@ -338,7 +338,7 @@
-GList * +GList * gst_rtsp_stream_transport_filter () @@ -368,7 +368,7 @@

Object Hierarchy

-
    GObject
+
    GObject
     ╰── GstRTSPStream
@@ -396,7 +396,7 @@ the destination again.

gst_rtsp_stream_new ()

GstRTSPStream *
-gst_rtsp_stream_new (guint idx,
+gst_rtsp_stream_new (guint idx,
                      GstElement *payloader,
                      GstPad *srcpad);

Create a new media stream with index idx @@ -440,7 +440,7 @@ gst_rtsp_stream_new (

gst_rtsp_stream_get_index ()

-
guint
+
guint
 gst_rtsp_stream_get_index (GstRTSPStream *stream);

 
Get the stream index.

 
Return: the stream index.

@@ -491,7 +491,7 @@ gst_rtsp_stream_get_srcpad (
 
gst_rtsp_stream_get_control ()

-
gchar *
+
gchar *
 gst_rtsp_stream_get_control (GstRTSPStream *stream);

 
Get the control string to identify this stream.

 

@@ -511,7 +511,7 @@ gst_rtsp_stream_get_control (
 
Returns

-
 the control string. g_free() after usage. 

+
 the control string. g_free() after usage. 

 
[transfer full]
@@ -520,7 +520,7 @@ gst_rtsp_stream_get_control (

gst_rtsp_stream_set_control ()

void
 gst_rtsp_stream_set_control (GstRTSPStream *stream,
-                             const gchar *control);
+ const gchar *control);

Set the control string in stream .

@@ -549,9 +549,9 @@ gst_rtsp_stream_set_control (

gst_rtsp_stream_has_control ()

-
gboolean
+
gboolean
 gst_rtsp_stream_has_control (GstRTSPStream *stream,
-                             const gchar *control);

+                             const gchar *control);

Check if stream has the control string control .

@@ -579,7 +579,7 @@ gst_rtsp_stream_has_control (

Returns

-

TRUE is stream +

TRUE is stream has control as the control string

@@ -588,7 +588,7 @@ as the control string

gst_rtsp_stream_get_mtu ()

-
guint
+
guint
 gst_rtsp_stream_get_mtu (GstRTSPStream *stream);

 
Get the configured MTU in the payloader of stream
 .

@@ -618,7 +618,7 @@ gst_rtsp_stream_get_mtu (
gst_rtsp_stream_set_mtu ()

 
void
 gst_rtsp_stream_set_mtu (GstRTSPStream *stream,
-                         guint mtu);

+                         guint mtu);

Configure the mtu in the payloader of stream to mtu .

@@ -648,7 +648,7 @@ gst_rtsp_stream_set_mtu (

gst_rtsp_stream_get_dscp_qos ()

-
gint
+
gint
 gst_rtsp_stream_get_dscp_qos (GstRTSPStream *stream);

 
Get the configured DSCP QoS in of the outgoing sockets.

 

@@ -677,7 +677,7 @@ gst_rtsp_stream_get_dscp_qos (
gst_rtsp_stream_set_dscp_qos ()

 
void
 gst_rtsp_stream_set_dscp_qos (GstRTSPStream *stream,
-                              gint dscp_qos);

+                              gint dscp_qos);

Configure the dscp qos of the outgoing sockets to dscp_qos .

@@ -824,7 +824,7 @@ gst_rtsp_stream_set_protocols (

gst_rtsp_stream_is_transport_supported ()

-
gboolean
+
gboolean
 gst_rtsp_stream_is_transport_supported
                                (GstRTSPStream *stream,
                                 GstRTSPTransport *transport);

@@ -854,7 +854,7 @@ gst_rtsp_stream_is_transport_supported

Returns

-

TRUE if transport +

TRUE if transport can be handled by stream .

@@ -885,7 +885,7 @@ gst_rtsp_stream_get_address_pool (

Returns

the GstRTSPAddressPool of stream -. g_object_unref() after +. g_object_unref() after usage.

[transfer full]

@@ -927,10 +927,10 @@ gst_rtsp_stream_set_address_pool (

gst_rtsp_stream_reserve_address ()

GstRTSPAddress *
 gst_rtsp_stream_reserve_address (GstRTSPStream *stream,
-                                 const gchar *address,
-                                 guint port,
-                                 guint n_ports,
-                                 guint ttl);
+ const gchar *address, + guint port, + guint n_ports, + guint ttl);

Reserve address and port as the address and port of stream @@ -975,7 +975,7 @@ gst_rtsp_stream_reserve_address (

Returns

the GstRTSPAddress of stream -or NULL when +or NULL when the address could be reserved. gst_rtsp_address_free() after usage.

[nullable]

@@ -983,7 +983,7 @@ the address could be reserved.

gst_rtsp_stream_join_bin ()

-
gboolean
+
gboolean
 gst_rtsp_stream_join_bin (GstRTSPStream *stream,
                           GstBin *bin,
                           GstElement *rtpbin,
@@ -1033,21 +1033,21 @@ added to bin

Returns

-

TRUE on success.

+

TRUE on success.

gst_rtsp_stream_leave_bin ()

-
gboolean
+
gboolean
 gst_rtsp_stream_leave_bin (GstRTSPStream *stream,
                            GstBin *bin,
                            GstElement *rtpbin);

 
Remove the elements of stream
  from bin
 .

-
Return: TRUE on success.

+
Return: TRUE on success.

 

 
Parameters

 

@@ -1082,7 +1082,7 @@ gst_rtsp_stream_leave_bin (void
 gst_rtsp_stream_get_server_port (GstRTSPStream *stream,
                                  GstRTSPRange *server_port,
-                                 GSocketFamily family);
+                                 GSocketFamily family);
 
Fill server_port
  with the port pair used by the server. This function can
 only be called when stream
@@ -1120,7 +1120,7 @@ only be called when stream
 
gst_rtsp_stream_get_multicast_address ()

 
GstRTSPAddress *
 gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
-                                       GSocketFamily family);

+                                       GSocketFamily family);
 
Get the multicast address of stream
  for family
 .

@@ -1140,7 +1140,7 @@ gst_rtsp_stream_get_multicast_address (
family

-
+
@@ -1149,7 +1149,7 @@ gst_rtsp_stream_get_multicast_address (
 
Returns

 
 the GstRTSPAddress of stream
-or NULL when no address could be allocated. gst_rtsp_address_free()
+or NULL when no address could be allocated. gst_rtsp_address_free()
 after usage. 

 
[transfer full][nullable]

 
@@ -1157,7 +1157,7 @@ after usage. 

 

 

 
gst_rtsp_stream_get_rtpsession ()

-
GObject *
+
GObject *
 gst_rtsp_stream_get_rtpsession (GstRTSPStream *stream);

 
Get the RTP session of this stream.

 

@@ -1186,7 +1186,7 @@ gst_rtsp_stream_get_rtpsession (
gst_rtsp_stream_get_ssrc ()

 
void
 gst_rtsp_stream_get_ssrc (GstRTSPStream *stream,
-                          guint *ssrc);

+                          guint *ssrc);

 
Get the SSRC used by the RTP session of this stream. This function can only
 be called when stream
  has been joined.

@@ -1216,11 +1216,11 @@ be called when stream
 

 

 
gst_rtsp_stream_get_rtpinfo ()

-
gboolean
+
gboolean
 gst_rtsp_stream_get_rtpinfo (GstRTSPStream *stream,
-                             guint *rtptime,
-                             guint *seq,
-                             guint *clock_rate,
+                             guint *rtptime,
+                             guint *seq,
+                             guint *clock_rate,
                              GstClockTime *running_time);

 
Retrieve the current rtptime, seq and running-time. This is used to
 construct a RTPInfo reply header.

@@ -1263,7 +1263,7 @@ construct a RTPInfo reply header.

 

 

 
Returns

-
 TRUE when rtptime, seq and running-time could be determined.

+
 TRUE when rtptime, seq and running-time could be determined.

 

 

 

@@ -1300,7 +1300,7 @@ after usage. 

 

 

 
gst_rtsp_stream_get_pt ()

-
guint
+
guint
 gst_rtsp_stream_get_pt (GstRTSPStream *stream);

 
Get the stream payload type.

 
Return: the stream payload type.

@@ -1399,7 +1399,7 @@ message has been received from a client using the TCP transport.

 

 

 
gst_rtsp_stream_add_transport ()

-
gboolean
+
gboolean
 gst_rtsp_stream_add_transport (GstRTSPStream *stream,
                                GstRTSPStreamTransport *trans);

 
Add the transport in trans
@@ -1436,7 +1436,7 @@ then also be send to the values configured in trans<
 

 

 
Returns

-
 TRUE if trans
+
 TRUE if trans
 was added

 

 

@@ -1444,7 +1444,7 @@ was added

 

 

 
gst_rtsp_stream_remove_transport ()

-
gboolean
+
gboolean
 gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
                                   GstRTSPStreamTransport *trans);

 
Remove the transport in trans
@@ -1481,7 +1481,7 @@ not be sent to the values configured in trans
 

 

 
Returns

-
 TRUE if trans
+
 TRUE if trans
 was removed

 

 

@@ -1489,9 +1489,9 @@ was removed

 

 

 
gst_rtsp_stream_get_rtp_socket ()

-
GSocket *
+
GSocket *
 gst_rtsp_stream_get_rtp_socket (GstRTSPStream *stream,
-                                GSocketFamily family);

+                                GSocketFamily family);

 
Get the RTP socket from stream
  for a family
 .

@@ -1521,7 +1521,7 @@ gst_rtsp_stream_get_rtp_socket (
 

 
Returns

-
 the RTP socket or NULL if no
+
 the RTP socket or NULL if no
 socket could be allocated for family
 . Unref after usage. 

 
[transfer full][nullable]

@@ -1530,9 +1530,9 @@ socket could be allocated for family
 

 

 
gst_rtsp_stream_get_rtcp_socket ()

-
GSocket *
+
GSocket *
 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream *stream,
-                                 GSocketFamily family);

+                                 GSocketFamily family);

 
Get the RTCP socket from stream
  for a family
 .

@@ -1562,7 +1562,7 @@ gst_rtsp_stream_get_rtcp_socket (
 

 
Returns

-
 the RTCP socket or NULL if no
+
 the RTCP socket or NULL if no
 socket could be allocated for family
 . Unref after usage. 

 
[transfer full][nullable]

@@ -1571,9 +1571,9 @@ socket could be allocated for family
 

 

 
gst_rtsp_stream_set_blocked ()

-
gboolean
+
gboolean
 gst_rtsp_stream_set_blocked (GstRTSPStream *stream,
-                             gboolean blocked);

+                             gboolean blocked);

 
Blocks or unblocks the dataflow on stream
 .

 

@@ -1600,14 +1600,14 @@ gst_rtsp_stream_set_blocked (
 
Returns

-
 TRUE on success

+
 TRUE on success

 

 

 

 

 

 
gst_rtsp_stream_is_blocking ()

-
gboolean
+
gboolean
 gst_rtsp_stream_is_blocking (GstRTSPStream *stream);

 
Check if stream
  is blocking on a GstBuffer.

@@ -1628,7 +1628,7 @@ gst_rtsp_stream_is_blocking (
 
Returns

-
 TRUE if stream
+
 TRUE if stream
 is blocking

 

 

@@ -1636,9 +1636,9 @@ is blocking

 

 

 
gst_rtsp_stream_update_crypto ()

-
gboolean
+
gboolean
 gst_rtsp_stream_update_crypto (GstRTSPStream *stream,
-                               guint ssrc,
+                               guint ssrc,
                                GstCaps *crypto);

 
Update the new crypto information for ssrc
  in stream
@@ -1647,7 +1647,7 @@ for ssrc
  did not exist, it will be added. If information
 for ssrc
  existed, it will be replaced. If crypto
- is NULL, it will
+ is NULL, it will
 be removed from stream
 .

 

@@ -1679,7 +1679,7 @@ be removed from stream
 

 

 
Returns

-
 TRUE if crypto
+
 TRUE if crypto
 could be updated

 

 

@@ -1690,7 +1690,7 @@ could be updated

 
GstRTSPFilterResult
 (*GstRTSPStreamTransportFilterFunc) (GstRTSPStream *stream,
                                      GstRTSPStreamTransport *trans,
-                                     gpointer user_data);

+                                     gpointer user_data);

 
This function will be called by the gst_rtsp_stream_transport_filter(). An
 implementation should return a value of GstRTSPFilterResult.

 
When this function returns GST_RTSP_FILTER_REMOVE, trans
@@ -1702,7 +1702,7 @@ from stream
 stream
 .

 
A value of GST_RTSP_FILTER_REF will add trans
- to the result GList of
+ to the result GList of
 gst_rtsp_stream_transport_filter().

 

 
Parameters

@@ -1741,10 +1741,10 @@ from stream
 

 

 
gst_rtsp_stream_transport_filter ()

-
GList *
+
GList *
 gst_rtsp_stream_transport_filter (GstRTSPStream *stream,
                                   GstRTSPStreamTransportFilterFunc func,
-                                  gpointer user_data);

+                                  gpointer user_data);

 
Call func
  for each transport managed by stream
 . The result value of func
@@ -1765,10 +1765,10 @@ locked so no further actions on stream
 
If func
  returns GST_RTSP_FILTER_REF, the transport will remain in stream
  but
-will also be added with an additional ref to the result GList of this
+will also be added with an additional ref to the result GList of this
 function..

 
When func
- is NULL, GST_RTSP_FILTER_REF will be assumed for each transport.

+ is NULL, GST_RTSP_FILTER_REF will be assumed for each transport.

 

 
Parameters

 
the GSocketFamily
the GSocketFamily

  
 

 

@@ -1799,10 +1799,10 @@ function..

 
 

 
Returns

-
 a GList with all
+
 a GList with all
 transports for which func
 returned GST_RTSP_FILTER_REF. After usage, each
-element in the GList should be unreffed before the list is freed. 

+element in the GList should be unreffed before the list is freed. 

 
[element-type GstRTSPStreamTransport][transfer full]

 

 
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html b/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html
index 2ba729e..aec2ed1 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html
@@ -81,7 +81,7 @@
 
 
-
+
@@ -564,7 +564,7 @@ IPv6 addresses

 
-
+
@@ -653,7 +653,7 @@ IPv6 addresses

 
-
+
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPContext.html b/docs/libs/html/gst-rtsp-server-GstRTSPContext.html
index 05b6d41..00c9702 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPContext.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPContext.html
@@ -212,7 +212,7 @@ is on the top of the stack).

 
 
-
+
@@ -222,27 +222,27 @@ is on the top of the stack).

 
-
+
-
+
-
+
-
+
-
+
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html b/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html
index cf5778e..ba6ee82 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html
@@ -57,7 +57,7 @@
 
-
+
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html b/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html
index 4ce53af..f793fe9 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html
@@ -65,7 +65,7 @@
 
 
-
-
+
+
-
-
+
+
@@ -544,13 +544,13 @@ structures.

 
 
-
+
-
-
+
+
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPToken.html b/docs/libs/html/gst-rtsp-server-GstRTSPToken.html
index 8279c79..9bd5be3 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPToken.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPToken.html
@@ -95,7 +95,7 @@
 
 
-

 
-gboolean
+gboolean
 
 
 gst_rtsp_address_pool_add_range ()
@@ -89,7 +89,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_address_pool_has_unicast_addresses ()
@@ -282,10 +282,10 @@ gst_rtsp_address_pool_dump (
 
gst_rtsp_address_pool_add_range ()

-
gboolean
+
gboolean
 gst_rtsp_address_pool_add_range (GstRTSPAddressPool *pool,
-                                 const gchar *min_address,
-                                 const gchar *max_address,
+                                 const gchar *min_address,
+                                 const gchar *max_address,
                                  guint16 min_port,
                                  guint16 max_port,
                                  guint8 ttl);

@@ -354,14 +354,14 @@ to all available IPv4 or IPv6 addresses.

 
 

 
Returns

-
 TRUE if the addresses could be added.

+
 TRUE if the addresses could be added.

 

 

 
 

 

 
gst_rtsp_address_pool_has_unicast_addresses ()

-
gboolean
+
gboolean
 gst_rtsp_address_pool_has_unicast_addresses
                                (GstRTSPAddressPool *pool);

 
Used to know if the pool includes any unicast addresses.

@@ -382,7 +382,7 @@ gst_rtsp_address_pool_has_unicast_addresses
 

 

 
Returns

-
 TRUE if the pool includes any unicast addresses, FALSE otherwise

+
 TRUE if the pool includes any unicast addresses, FALSE otherwise

 

 

 
@@ -392,7 +392,7 @@ gst_rtsp_address_pool_has_unicast_addresses
 
GstRTSPAddress *
 gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool *pool,
                                        GstRTSPAddressFlags flags,
-                                       gint n_ports);

+                                       gint n_ports);

 
Take an address and ports from pool
 . flags
  can be used to control the
@@ -430,7 +430,7 @@ one can be found in port
 

 
Returns

 
 a GstRTSPAddress that should be freed with
-gst_rtsp_address_free after use or NULL when no address could be
+gst_rtsp_address_free after use or NULL when no address could be
 acquired. 

 
[nullable]

 

@@ -440,10 +440,10 @@ acquired. 

 
gst_rtsp_address_pool_reserve_address ()

 
GstRTSPAddressPoolResult
 gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool *pool,
-                                       const gchar *ip_address,
-                                       guint port,
-                                       guint n_ports,
-                                       guint ttl,
+                                       const gchar *ip_address,
+                                       guint port,
+                                       guint n_ports,
+                                       guint ttl,
                                        GstRTSPAddress **address);

 
Take a specific address and ports from pool
 . n_ports
@@ -554,7 +554,7 @@ IPv6 addresses

 		 
 

 
gchar *address;
gchar *address;

 
the address

  
 
 
 

 
gint n_ports;
gint n_ports;

 
number of ports

  
 

 
 
GObject parent;
GObject parent;

 
the parent GObject

  
 

 
GstRTSPAuth *auth;
the current auth object or NULL
the current auth object or NULL

  
 

 

 

 
GstRTSPSession *session;
the session, can be NULL
the session, can be NULL

  
 

 

 
GstRTSPSessionMedia *sessmedia;
the session media for the url can be NULL
the session media for the url can be NULL

  
 

 

 
GstRTSPMediaFactory *factory;
the media factory for the url, can be NULL
the media factory for the url, can be NULL

  
 

 

 
GstRTSPMedia *media;
the media for the url can be NULL
the media for the url can be NULL

  
 

 

 
GstRTSPStream *stream;
the stream for the url can be NULL
the stream for the url can be NULL

  
 

 

 

 
-gchar *
+gchar *
 
 
 gst_rtsp_media_factory_uri_get_uri ()
@@ -111,7 +111,7 @@ gst_rtsp_media_factory_uri_new (v
 
gst_rtsp_media_factory_uri_set_uri ()

 
void
 gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI *factory,
-                                    const gchar *uri);

+                                    const gchar *uri);
 
Set the URI of the resource that will be streamed by this factory.

 

 
Parameters

@@ -139,7 +139,7 @@ gst_rtsp_media_factory_uri_set_uri (
 

 
gst_rtsp_media_factory_uri_get_uri ()

-
gchar *
+
gchar *
 gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI *factory);

 
Get the URI that will provide media for this factory.

 

@@ -159,7 +159,7 @@ gst_rtsp_media_factory_uri_get_uri (
 

 
Returns

-
 the configured URI. g_free() after usage. 

+
 the configured URI. g_free() after usage. 

 
[transfer full]

 

 

diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPPermissions.html b/docs/libs/html/gst-rtsp-server-GstRTSPPermissions.html
index 37fb587..d6cfc0f 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPPermissions.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPPermissions.html
@@ -96,7 +96,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_permissions_is_allowed ()
@@ -129,7 +129,7 @@ object that performs the checks on the permissions and the current
 GstRTSPToken.

 
As a convenience function, gst_rtsp_permissions_is_allowed() can be used to
 check if the permissions contains a role that contains the boolean value
-TRUE for the the given key.

+TRUE for the the given key.

 
Last reviewed on 2013-07-15 (1.0.0)

 
 

@@ -200,8 +200,8 @@ gst_rtsp_permissions_unref (
gst_rtsp_permissions_add_role ()

 
void
 gst_rtsp_permissions_add_role (GstRTSPPermissions *permissions,
-                               const gchar *role,
-                               const gchar *fieldname,
+                               const gchar *role,
+                               const gchar *fieldname,
                                ...);

 
Add a new role
  to permissions
@@ -245,8 +245,8 @@ are the same layout as 
gst_rtsp_permissions_add_role_valist ()

 
void
 gst_rtsp_permissions_add_role_valist (GstRTSPPermissions *permissions,
-                                      const gchar *role,
-                                      const gchar *fieldname,
+                                      const gchar *role,
+                                      const gchar *fieldname,
                                       va_list var_args);

 
Add a new role
  to permissions
@@ -290,7 +290,7 @@ are set according to the varargs in a manner similar to 
gst_rtsp_permissions_remove_role ()

 
void
 gst_rtsp_permissions_remove_role (GstRTSPPermissions *permissions,
-                                  const gchar *role);

+                                  const gchar *role);
 
Remove all permissions for role
  in permissions
 .

@@ -322,7 +322,7 @@ gst_rtsp_permissions_remove_role (
gst_rtsp_permissions_get_role ()

 
const GstStructure *
 gst_rtsp_permissions_get_role (GstRTSPPermissions *permissions,
-                               const gchar *role);

+                               const gchar *role);
 
Get all permissions for role
  in permissions
 .

@@ -360,10 +360,10 @@ is valid. 

 

 

 
gst_rtsp_permissions_is_allowed ()

-
gboolean
+
gboolean
 gst_rtsp_permissions_is_allowed (GstRTSPPermissions *permissions,
-                                 const gchar *role,
-                                 const gchar *permission);

+                                 const gchar *role,
+                                 const gchar *permission);

 
Check if role
  in permissions
  is given permission for permission
@@ -397,7 +397,7 @@ gst_rtsp_permissions_is_allowed (
 

 
Returns

-
 TRUE if role
+
 TRUE if role
 is allowed permission
 .

 

diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPSdp.html b/docs/libs/html/gst-rtsp-server-GstRTSPSdp.html
index 00d6db7..88f0d70 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPSdp.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPSdp.html
@@ -40,7 +40,7 @@
 
 

 
-gboolean
+gboolean
 
 
 gst_rtsp_sdp_from_media ()
@@ -69,7 +69,7 @@
 
Functions

 

 
gst_rtsp_sdp_from_media ()

-
gboolean
+
gboolean
 gst_rtsp_sdp_from_media (GstSDPMessage *sdp,
                          GstSDPInfo *info,
                          GstRTSPMedia *media);

diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPSessionMedia.html b/docs/libs/html/gst-rtsp-server-GstRTSPSessionMedia.html
index da892d7..f36b686 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPSessionMedia.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPSessionMedia.html
@@ -49,7 +49,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_session_media_matches ()
@@ -73,7 +73,7 @@
 

 

 
-gchar *
+gchar *
 
 
 gst_rtsp_session_media_get_rtpinfo ()
@@ -81,7 +81,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_session_media_set_state ()
@@ -121,7 +121,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_session_media_alloc_channels ()
@@ -164,7 +164,7 @@ transports.

 

 
gst_rtsp_session_media_new ()

 
GstRTSPSessionMedia *
-gst_rtsp_session_media_new (const gchar *path,
+gst_rtsp_session_media_new (const gchar *path,
                             GstRTSPMedia *media);

 
Create a new GstRTSPSessionMedia that manages the streams
 in media
@@ -204,10 +204,10 @@ in media
 

 

 
gst_rtsp_session_media_matches ()

-
gboolean
+
gboolean
 gst_rtsp_session_media_matches (GstRTSPSessionMedia *media,
-                                const gchar *path,
-                                gint *matched);

+                                const gchar *path,
+                                gint *matched);

 
Check if the path of media
  matches path
 . It path
@@ -244,7 +244,7 @@ matched characters is returned in matched
 

 
Returns

-
 TRUE when path
+
 TRUE when path
 matches the path of media
 .

 

@@ -312,7 +312,7 @@ gst_rtsp_session_media_get_base_time (
 
gst_rtsp_session_media_get_rtpinfo ()

-
gchar *
+
gchar *
 gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia *media);

 
Retrieve the RTP-Info header string for all streams in media
 
@@ -335,14 +335,14 @@ with configured transports.

 

 
Returns

 
 The RTP-Info as a string or
-NULL when no RTP-Info could be generated, g_free() after usage. 

+NULL when no RTP-Info could be generated, g_free() after usage. 

 
[transfer full][nullable]

 

 

 

 

 
gst_rtsp_session_media_set_state ()

-
gboolean
+
gboolean
 gst_rtsp_session_media_set_state (GstRTSPSessionMedia *media,
                                   GstState state);

 
Tell the media object media
@@ -372,7 +372,7 @@ gst_rtsp_session_media_set_state (
 

 
Returns

-
 TRUE on success.

+
 TRUE on success.

 

 

 

@@ -442,7 +442,7 @@ gst_rtsp_session_media_set_rtsp_state (
gst_rtsp_session_media_get_transport ()

 
GstRTSPStreamTransport *
 gst_rtsp_session_media_get_transport (GstRTSPSessionMedia *media,
-                                      guint idx);

+                                      guint idx);
 
Get a previously created GstRTSPStreamTransport for the stream at idx
 .

 

@@ -523,7 +523,7 @@ gst_rtsp_session_media_set_transport (
 
gst_rtsp_session_media_alloc_channels ()

-
gboolean
+
gboolean
 gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia *media,
                                        GstRTSPRange *range);

 
Fill range
@@ -553,7 +553,7 @@ interleaved transport.

 

 

 
Returns

-
 TRUE on success.

+
 TRUE on success.

 

 

 

diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html b/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html
index e065a93..6ae49fe 100644
--- a/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html
@@ -87,7 +87,7 @@
 

 

 
-gchar *
+gchar *
 
 
 gst_rtsp_stream_transport_get_rtpinfo ()
@@ -95,7 +95,7 @@
 

 

 
-gboolean
+gboolean
 
 
 (*GstRTSPSendFunc) ()
@@ -135,7 +135,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_stream_transport_set_active ()
@@ -151,7 +151,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_stream_transport_is_timed_out ()
@@ -159,7 +159,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_stream_transport_send_rtcp ()
@@ -167,7 +167,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_stream_transport_send_rtp ()
@@ -408,7 +408,7 @@ gst_rtsp_stream_transport_set_url (
 

 
gst_rtsp_stream_transport_get_rtpinfo ()

-
gchar *
+
gchar *
 gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport *trans,
                                        GstClockTime start_time);

 
Get the RTP-Info string for trans
@@ -440,18 +440,18 @@ gst_rtsp_stream_transport_get_rtpinfo (
Returns

 
 the RTPInfo string for trans
 and start_time
-or NULL when the RTP-Info could not be
-determined. g_free() after usage. 

+or NULL when the RTP-Info could not be
+determined. g_free() after usage. 

 
[transfer full][nullable]

 

 
 

 

 
GstRTSPSendFunc ()

-
gboolean
+
gboolean
 (*GstRTSPSendFunc) (GstBuffer *buffer,
                     guint8 channel,
-                    gpointer user_data);

+                    gpointer user_data);

 
Function registered with gst_rtsp_stream_transport_set_callbacks() and
 called when buffer
  must be sent on channel
@@ -485,7 +485,7 @@ called when buffer
 

 

 
Returns

-
 TRUE on success

+
 TRUE on success

 

 

 
@@ -497,8 +497,8 @@ gst_rtsp_stream_transport_set_callbacks
                                (GstRTSPStreamTransport *trans,
                                 GstRTSPSendFunc send_rtp,
                                 GstRTSPSendFunc send_rtcp,
-                                gpointer user_data,
-                                GDestroyNotify notify);
+                                gpointer user_data,
+                                GDestroyNotify notify);
 
Install callbacks that will be called when data for a stream should be sent
 to a client. This is usually used when sending RTP/RTCP over TCP.

 

@@ -543,7 +543,7 @@ to a client. This is usually used when sending RTP/RTCP over TCP.

 

 
GstRTSPKeepAliveFunc ()

 
void
-(*GstRTSPKeepAliveFunc) (gpointer user_data);

+(*GstRTSPKeepAliveFunc) (gpointer user_data);
 
Function registered with gst_rtsp_stream_transport_set_keepalive() and called
 when the stream is active.

 

@@ -569,8 +569,8 @@ when the stream is active.

 gst_rtsp_stream_transport_set_keepalive
                                (GstRTSPStreamTransport *trans,
                                 GstRTSPKeepAliveFunc keep_alive,
-                                gpointer user_data,
-                                GDestroyNotify notify);
+                                gpointer user_data,
+                                GDestroyNotify notify);
 
Install callbacks that will be called when RTCP packets are received from the
 receiver of trans
 .

@@ -633,9 +633,9 @@ gst_rtsp_stream_transport_keep_alive (
 
gst_rtsp_stream_transport_set_active ()

-
gboolean
+
gboolean
 gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport *trans,
-                                      gboolean active);

+                                      gboolean active);

 
Activate or deactivate datatransfer configured in trans
 .

 

@@ -663,7 +663,7 @@ gst_rtsp_stream_transport_set_active (
 
Returns

-
 TRUE when the state was changed.

+
 TRUE when the state was changed.

 

 

 

@@ -673,7 +673,7 @@ gst_rtsp_stream_transport_set_active (void
 gst_rtsp_stream_transport_set_timed_out
                                (GstRTSPStreamTransport *trans,
-                                gboolean timedout);
+                                gboolean timedout);
 
Set the timed out state of trans
  to timedout
 

@@ -703,7 +703,7 @@ gst_rtsp_stream_transport_set_timed_out
 

 

 
gst_rtsp_stream_transport_is_timed_out ()

-
gboolean
+
gboolean
 gst_rtsp_stream_transport_is_timed_out
                                (GstRTSPStreamTransport *trans);

 
Check if trans
@@ -725,7 +725,7 @@ gst_rtsp_stream_transport_is_timed_out
 

 

 
Returns

-
 TRUE if trans
+
 TRUE if trans
 timed out.

 

 

@@ -733,7 +733,7 @@ timed out.

 

 

 
gst_rtsp_stream_transport_send_rtcp ()

-
gboolean
+
gboolean
 gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
                                      GstBuffer *buffer);

 
Send buffer
@@ -763,14 +763,14 @@ gst_rtsp_stream_transport_send_rtcp (
 

 
Returns

-
 TRUE on success

+
 TRUE on success

 

 

 

 

 

 
gst_rtsp_stream_transport_send_rtp ()

-
gboolean
+
gboolean
 gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
                                     GstBuffer *buffer);

 
Send buffer
@@ -800,7 +800,7 @@ gst_rtsp_stream_transport_send_rtp (
 

 
Returns

-
 TRUE on success

+
 TRUE on success

 

 

 

@@ -823,7 +823,7 @@ gst_rtsp_stream_transport_send_rtp (
 
 
GObject parent;
GObject parent;

 
parent instance

  
 

 
-gboolean
+gboolean
 
 
 gst_rtsp_thread_reuse ()
@@ -89,7 +89,7 @@
 

 

 
-gint
+gint
 
 
 gst_rtsp_thread_pool_get_max_threads ()
@@ -251,7 +251,7 @@ gst_rtsp_thread_unref (
 
gst_rtsp_thread_reuse ()

-
gboolean
+
gboolean
 gst_rtsp_thread_reuse (GstRTSPThread *thread);

 
Reuse the mainloop of thread
 

@@ -272,7 +272,7 @@ gst_rtsp_thread_reuse (
 
Returns

-
 TRUE if the mainloop could be reused

+
 TRUE if the mainloop could be reused

 

 
 
@@ -316,7 +316,7 @@ gst_rtsp_thread_pool_new (void
 

 
gst_rtsp_thread_pool_get_max_threads ()

-
gint
+
gint
 gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool *pool);

 
Get the maximum number of threads used for client connections.
 See gst_rtsp_thread_pool_set_max_threads().

@@ -346,7 +346,7 @@ See 
gst_rtsp_thread_pool_set_max_threads ()

 
void
 gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool *pool,
-                                      gint max_threads);

+                                      gint max_threads);

 
Set the maximum threads used by the pool to handle client requests.
 A value of 0 will use the pool mainloop, a value of -1 will use an
 unlimited number of threads.

@@ -490,13 +490,13 @@ structures.

 
 
 

 
GMainContext *context;
a GMainContext
GMainContext *context;
a GMainContext

  
 

 
GMainLoop *loop;
a GMainLoop
GMainLoop *loop;
a GMainLoop

  
 

 
 
GObjectClass parent_class;
GObjectClass parent_class;

  
  
 

 
GThreadPool *pool;
a GThreadPool used internally
GThreadPool *pool;
a GThreadPool used internally

  
 

 

 
const gchar *
+const gchar *
 
 
 gst_rtsp_token_get_string ()
@@ -103,7 +103,7 @@
 

 

 
-gboolean
+gboolean
 
 
 gst_rtsp_token_is_allowed ()
@@ -155,7 +155,7 @@ gst_rtsp_token_new_empty (void
 
gst_rtsp_token_new ()

 
GstRTSPToken *
-gst_rtsp_token_new (const gchar *firstfield,
+gst_rtsp_token_new (const gchar *firstfield,
                     ...);

 
Create a new Authorization token with the given fieldnames and values.
 Arguments are given similar to gst_structure_new().

@@ -191,7 +191,7 @@ Arguments are given similar to 
 
gst_rtsp_token_new_valist ()

 
GstRTSPToken *
-gst_rtsp_token_new_valist (const gchar *firstfield,
+gst_rtsp_token_new_valist (const gchar *firstfield,
                            va_list var_args);

 
Create a new Authorization token with the given fieldnames and values.
 Arguments are given similar to gst_structure_new_valist().

@@ -330,7 +330,7 @@ gst_rtsp_token_writable_structure (token
-is writable and will never return NULL.

+is writable and will never return NULL.

 
MT safe. 

 
[transfer none]

 
@@ -338,9 +338,9 @@ is writable and will never return 
 
gst_rtsp_token_get_string ()

-
const gchar *
+
const gchar *
 gst_rtsp_token_get_string (GstRTSPToken *token,
-                           const gchar *field);

+                           const gchar *field);

 
Get the string value of field
  in token
 .

@@ -371,7 +371,7 @@ gst_rtsp_token_get_string (field
 in
 token
-or NULL when field
+or NULL when field
 is not defined in token
 . The string
 becomes invalid when you free token
@@ -382,12 +382,12 @@ becomes invalid when you free token
 

 

 
gst_rtsp_token_is_allowed ()

-
gboolean
+
gboolean
 gst_rtsp_token_is_allowed (GstRTSPToken *token,
-                           const gchar *field);

+                           const gchar *field);

 
Check if token
  has a boolean field
- and if it is set to TRUE.

+ and if it is set to TRUE.

 

 
Parameters

 

@@ -412,9 +412,9 @@ gst_rtsp_token_is_allowed (
 
Returns

-
 TRUE if token
+
 TRUE if token
 has a boolean field named field
-set to TRUE.

+set to TRUE.

 

 
 
diff --git a/docs/libs/html/index.html b/docs/libs/html/index.html
index ad0e2e2..d6b8af9 100644
--- a/docs/libs/html/index.html
+++ b/docs/libs/html/index.html
@@ -15,7 +15,7 @@
 

 

 

-      for GStreamer RTSP Server 1.4.1
+      for GStreamer RTSP Server 1.4.5
     

 

 

diff --git a/docs/libs/html/rtsp-server-hierarchy.html b/docs/libs/html/rtsp-server-hierarchy.html
index f4d68b7..4d752f2 100644
--- a/docs/libs/html/rtsp-server-hierarchy.html
+++ b/docs/libs/html/rtsp-server-hierarchy.html
@@ -23,7 +23,7 @@
 

 Object Hierarchy

 
-    GObject
+    GObject
     ├── GstRTSPAuth
     ├── GstRTSPMountPoints
     ├── GstRTSPMediaFactory
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index a549266..8dfa661 100644
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,46 @@ RTSP server library based on GStreamer
 
  
   
+   1.4.5
+   1.4
+   
+   2014-12-18
+   
+  
+ 
+
+ 
+  
+   1.4.4
+   1.4
+   
+   2014-11-06
+   
+  
+ 
+
+ 
+  
+   1.4.3
+   1.4
+   
+   2014-09-24
+   
+  
+ 
+
+ 
+  
+   1.4.2
+   1.4
+   
+   2014-09-19
+   
+  
+ 
+
+ 
+  
    1.4.1
    1.4
    
diff --git a/gst/rtsp-server/from_upstream.diff b/gst/rtsp-server/from_upstream.diff
deleted file mode 100644
index 99065f5..0000000
--- a/gst/rtsp-server/from_upstream.diff
+++ /dev/null
@@ -1,255 +0,0 @@
-diff -urN /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/Makefile.am ./Makefile.am
---- /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/Makefile.am	2014-01-13 01:50:54.000000000 +0900
-+++ ./Makefile.am	2015-01-15 16:49:47.091243189 +0900
-@@ -7,6 +7,7 @@
- 		rtsp-thread-pool.h \
- 		rtsp-media.h \
- 		rtsp-media-factory.h \
-+		rtsp-media-factory-wfd.h \
- 		rtsp-media-factory-uri.h \
- 		rtsp-mount-points.h \
- 		rtsp-permissions.h \
-@@ -16,8 +17,11 @@
- 		rtsp-session-media.h \
- 		rtsp-session-pool.h \
- 		rtsp-token.h \
-+		rtsp-client-wfd.h \
- 		rtsp-client.h \
--		rtsp-server.h
-+		rtsp-server-wfd.h \
-+		rtsp-server.h \
-+		gstwfdmessage.h
- 
- c_sources = \
- 	rtsp-auth.c \
-@@ -28,6 +32,7 @@
- 	rtsp-thread-pool.c \
- 	rtsp-media.c \
- 	rtsp-media-factory.c \
-+	rtsp-media-factory-wfd.c \
- 	rtsp-media-factory-uri.c \
- 	rtsp-mount-points.c \
- 	rtsp-permissions.c \
-@@ -37,7 +42,10 @@
- 	rtsp-session-media.c \
- 	rtsp-session-pool.c \
- 	rtsp-token.c \
-+	gstwfdmessage.c \
-+	rtsp-client-wfd.c \
- 	rtsp-client.c \
-+	rtsp-server-wfd.c \
- 	rtsp-server.c
- 
- noinst_HEADERS = 
-diff -urN /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-client.c ./rtsp-client.c
---- /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-client.c	2014-02-10 21:43:30.000000000 +0900
-+++ ./rtsp-client.c	2015-02-23 16:33:37.059801061 +0900
-@@ -137,6 +137,12 @@
-     GstRTSPContext * ctx);
- static gchar *default_make_path_from_uri (GstRTSPClient * client,
-     const GstRTSPUrl * uri);
-+static gboolean default_handle_options_request (GstRTSPClient * client,
-+    GstRTSPContext * ctx);
-+static gboolean default_handle_set_param_request (GstRTSPClient * client,
-+    GstRTSPContext * ctx);
-+static gboolean default_handle_get_param_request (GstRTSPClient * client,
-+    GstRTSPContext * ctx);
- 
- G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
- 
-@@ -159,6 +165,9 @@
-   klass->params_set = default_params_set;
-   klass->params_get = default_params_get;
-   klass->make_path_from_uri = default_make_path_from_uri;
-+  klass->handle_options_request = default_handle_options_request;
-+  klass->handle_set_param_request = default_handle_set_param_request;
-+  klass->handle_get_param_request = default_handle_get_param_request;
- 
-   g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
-       g_param_spec_object ("session-pool", "Session Pool",
-@@ -869,7 +878,7 @@
- }
- 
- static gboolean
--handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
-+default_handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
- {
-   GstRTSPResult res;
-   guint8 *data;
-@@ -906,7 +915,7 @@
- }
- 
- static gboolean
--handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
-+default_handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
- {
-   GstRTSPResult res;
-   guint8 *data;
-@@ -1462,6 +1471,8 @@
-   if (media == NULL)
-     goto media_not_found_no_reply;
- 
-+  /* FIXME-WFD : wfd url problem */
-+#if 0
-   if (path[matched] == '\0')
-     goto control_not_found;
- 
-@@ -1469,6 +1480,9 @@
-   path[matched] = '\0';
-   /* control is remainder */
-   control = &path[matched + 1];
-+#else
-+  control = g_strdup ("stream=0");
-+#endif
- 
-   /* find the stream now using the control part */
-   stream = gst_rtsp_media_find_stream (media, control);
-@@ -1534,6 +1548,10 @@
-   /* create and serialize the server transport */
-   st = make_server_transport (client, ctx, ct);
-   trans_str = gst_rtsp_transport_as_text (st);
-+
-+  /* FIXME-WFD : Temporarily force to set profile string */
-+  trans_str = g_strjoinv ("RTP/AVP/UDP", g_strsplit (trans_str, "RTP/AVP", -1));
-+
-   gst_rtsp_transport_free (st);
- 
-   /* construct the response now */
-@@ -1820,7 +1838,7 @@
- }
- 
- static gboolean
--handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
-+default_handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
- {
-   GstRTSPMethod options;
-   gchar *str;
-@@ -1906,6 +1924,9 @@
-   GstRTSPContext sctx = { NULL }, *ctx;
-   GstRTSPMessage response = { 0 };
-   gchar *sessid;
-+  GstRTSPClientClass *klass;
-+
-+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
- 
-   if (!(ctx = gst_rtsp_context_get_current ())) {
-     ctx = &sctx;
-@@ -1995,7 +2016,7 @@
-   /* now see what is asked and dispatch to a dedicated handler */
-   switch (method) {
-     case GST_RTSP_OPTIONS:
--      handle_options_request (client, ctx);
-+      klass->handle_options_request (client, ctx);
-       break;
-     case GST_RTSP_DESCRIBE:
-       handle_describe_request (client, ctx);
-@@ -2013,10 +2034,10 @@
-       handle_teardown_request (client, ctx);
-       break;
-     case GST_RTSP_SET_PARAMETER:
--      handle_set_param_request (client, ctx);
-+      klass->handle_set_param_request (client, ctx);
-       break;
-     case GST_RTSP_GET_PARAMETER:
--      handle_get_param_request (client, ctx);
-+      klass->handle_get_param_request (client, ctx);
-       break;
-     case GST_RTSP_ANNOUNCE:
-     case GST_RTSP_RECORD:
-diff -urN /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-client.h ./rtsp-client.h
---- /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-client.h	2014-01-13 01:50:54.000000000 +0900
-+++ ./rtsp-client.h	2015-02-17 15:57:28.047757756 +0900
-@@ -102,6 +102,10 @@
-   GstRTSPResult   (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
-   gchar *         (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
- 
-+  gboolean        (*handle_options_request) (GstRTSPClient * client, GstRTSPContext * ctx);
-+  gboolean        (*handle_set_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
-+  gboolean        (*handle_get_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
-+
-   /* signals */
-   void     (*closed)                  (GstRTSPClient *client);
-   void     (*new_session)             (GstRTSPClient *client, GstRTSPSession *session);
-@@ -177,7 +181,7 @@
-                                                                 GstRTSPSession *sess,
-                                                                 gpointer user_data);
- 
--GList *                gst_rtsp_client_session_filter    (GstRTSPClient *client,
-+GList *               gst_rtsp_client_session_filter    (GstRTSPClient *client,
-                                                           GstRTSPClientSessionFilterFunc func,
-                                                           gpointer user_data);
- 
-diff -urN /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-media.c ./rtsp-media.c
---- /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-media.c	2014-02-22 00:11:11.000000000 +0900
-+++ ./rtsp-media.c	2015-02-17 15:57:25.483757666 +0900
-@@ -241,10 +241,12 @@
-           "If this media pipeline can be reused after an unprepare",
-           DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- 
-+#ifdef GST_TYPE_RTSP_PROFILE
-   g_object_class_install_property (gobject_class, PROP_PROFILES,
-       g_param_spec_flags ("profiles", "Profiles",
-           "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
-           DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-+#endif
- 
-   g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
-       g_param_spec_flags ("protocols", "Protocols",
-diff -urN /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-stream.c ./rtsp-stream.c
---- /home/zzoon.ko/opensrc/tarball/gst-rtsp-server-1.2.3/gst/rtsp-server/rtsp-stream.c	2014-02-22 00:11:11.000000000 +0900
-+++ ./rtsp-stream.c	2015-02-17 15:57:25.527757667 +0900
-@@ -173,10 +173,12 @@
-           "The control string for this stream", DEFAULT_CONTROL,
-           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
- 
-+#ifdef GST_TYPE_RTSP_PROFILE
-   g_object_class_install_property (gobject_class, PROP_PROFILES,
-       g_param_spec_flags ("profiles", "Profiles",
-           "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
-           DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-+#endif
- 
-   g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
-       g_param_spec_flags ("protocols", "Protocols",
-@@ -1001,6 +1003,9 @@
-       inetaddr = g_inet_address_new_any (family);
-   }
- 
-+  /* FIXME-WFD : Force to set 19000 as port number */
-+  tmp_rtp = 19000;
-+
-   rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
-   if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
-     g_object_unref (rtp_sockaddr);
-@@ -1171,9 +1176,14 @@
-       G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
-       &priv->server_port_v4, &priv->server_addr_v4);
- 
-+  /* FIXME-WFD : force to disable ipv6 mode in WFD mode */
-+#if 0
-   priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
-       G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
-       &priv->server_port_v6, &priv->server_addr_v6);
-+#else
-+  priv->have_ipv6 = FALSE;
-+#endif
- 
-   return priv->have_ipv4 || priv->have_ipv6;
- }
-@@ -1552,6 +1562,8 @@
- 
-   /* get the session */
-   g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
-+  GST_DEBUG("zzoon unref internal session");
-+  g_object_unref(priv->session);
- 
-   g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
-       stream);
-@@ -1841,6 +1853,7 @@
-   gst_object_unref (priv->send_src[1]);
-   priv->send_src[1] = NULL;
- 
-+  GST_DEBUG("Unref rtp session");
-   g_object_unref (priv->session);
-   priv->session = NULL;
-   if (priv->caps)
diff --git a/gst/rtsp-server/rtsp-client-wfd.c b/gst/rtsp-server/rtsp-client-wfd.c
index 27607ef..267fcfa 100644
--- a/gst/rtsp-server/rtsp-client-wfd.c
+++ b/gst/rtsp-server/rtsp-client-wfd.c
@@ -46,7 +46,6 @@
 #include "rtsp-media-factory-wfd.h"
 #include "rtsp-sdp.h"
 #include "rtsp-params.h"
-#include "gstwfdmessage.h"
 
 #define GST_RTSP_WFD_CLIENT_GET_PRIVATE(obj)  \
    (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClientPrivate))
@@ -2086,6 +2085,18 @@ gst_rtsp_wfd_client_set_video_supported_resolution (GstRTSPWFDClient * client,
   return res;
 }
 
+GstRTSPResult
+gst_rtsp_wfd_client_set_video_native_resolution (GstRTSPWFDClient * client,
+    guint64 native_reso)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  priv->video_native_resolution = native_reso;
+  GST_DEBUG ("Native Resolution : %"G_GUINT64_FORMAT, native_reso);
+
+  return res;
+}
+
 static gboolean
 wfd_ckeck_keep_alive_response (gpointer userdata)
 {
diff --git a/gst/rtsp-server/rtsp-client-wfd.h b/gst/rtsp-server/rtsp-client-wfd.h
index 4300181..5f429e3 100644
--- a/gst/rtsp-server/rtsp-client-wfd.h
+++ b/gst/rtsp-server/rtsp-client-wfd.h
@@ -34,6 +34,7 @@ typedef struct _GstRTSPWFDClientPrivate GstRTSPWFDClientPrivate;
 #include "rtsp-sdp.h"
 #include "rtsp-auth.h"
 #include "rtsp-client.h"
+#include "gstwfdmessage.h"
 
 #define GST_TYPE_RTSP_WFD_CLIENT              (gst_rtsp_wfd_client_get_type ())
 #define GST_IS_RTSP_WFD_CLIENT(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_WFD_CLIENT))
@@ -122,7 +123,8 @@ GstRTSPResult         gst_rtsp_wfd_client_trigger_request (
 
 GstRTSPResult         gst_rtsp_wfd_client_set_video_supported_resolution (
                           GstRTSPWFDClient * client, guint64 supported_reso);
-
+GstRTSPResult         gst_rtsp_wfd_client_set_video_native_resolution (
+		                      GstRTSPWFDClient * client, guint64 native_reso);
 /**
  * GstRTSPWFDClientSessionFilterFunc:
  * @client: a #GstRTSPWFDClient object
diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 45abd90..ac1ce96 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -59,6 +59,7 @@ struct _GstRTSPClientPrivate
 {
   GMutex lock;                  /* protects everything else */
   GMutex send_lock;
+  GMutex watch_lock;
   GstRTSPConnection *connection;
   GstRTSPWatch *watch;
   GMainContext *watch_context;
@@ -287,6 +288,7 @@ gst_rtsp_client_init (GstRTSPClient * client)
 
   g_mutex_init (&priv->lock);
   g_mutex_init (&priv->send_lock);
+  g_mutex_init (&priv->watch_lock);
   priv->close_seq = 0;
   priv->drop_backlog = DEFAULT_DROP_BACKLOG;
 }
@@ -415,6 +417,7 @@ gst_rtsp_client_finalize (GObject * obj)
   g_free (priv->server_ip);
   g_mutex_clear (&priv->lock);
   g_mutex_clear (&priv->send_lock);
+  g_mutex_clear (&priv->watch_lock);
 
   G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
 }
@@ -1703,6 +1706,7 @@ handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
     gst_caps_unref (caps);
   }
   gst_mikey_message_unref (msg);
+  gst_buffer_unref (key);
 
   return TRUE;
 
@@ -1790,7 +1794,9 @@ handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
         handle_mikey_data (client, ctx, data, size);
       }
     }
+    g_strfreev (split);
   }
+  g_strfreev (specs);
   return TRUE;
 }
 
@@ -3188,7 +3194,9 @@ closed (GstRTSPWatch * watch, gpointer user_data)
   }
 
   gst_rtsp_watch_set_flushing (watch, TRUE);
+  g_mutex_lock (&priv->watch_lock);
   gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+  g_mutex_unlock (&priv->watch_lock);
 
   return GST_RTSP_OK;
 }
@@ -3308,6 +3316,7 @@ handle_tunnel (GstRTSPClient * client)
 
     opriv = oclient->priv;
 
+    g_mutex_lock (&opriv->watch_lock);
     if (opriv->watch == NULL)
       goto tunnel_closed;
 
@@ -3317,6 +3326,7 @@ handle_tunnel (GstRTSPClient * client)
     gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
     gst_rtsp_watch_reset (priv->watch);
     gst_rtsp_watch_reset (opriv->watch);
+    g_mutex_unlock (&opriv->watch_lock);
     g_object_unref (oclient);
 
     /* the old client owns the tunnel now, the new one will be freed */
@@ -3336,6 +3346,7 @@ no_tunnelid:
 tunnel_closed:
   {
     GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
+    g_mutex_unlock (&opriv->watch_lock);
     g_object_unref (oclient);
     return FALSE;
   }
diff --git a/gst/rtsp-server/rtsp-media-factory-wfd.c b/gst/rtsp-server/rtsp-media-factory-wfd.c
index 137930a..c0f50f2 100644
--- a/gst/rtsp-server/rtsp-media-factory-wfd.c
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.c
@@ -717,6 +717,7 @@ _rtsp_media_factory_wfd_create_xvcapture_bin (GstRTSPMediaFactoryWFD * factory,
   GstElement *vcaps = NULL;
   gchar *vcodec = NULL;
   GstElement *venc = NULL;
+  GstElement *vparse = NULL;
   GstElement *vqueue = NULL;
   GstRTSPMediaFactoryWFDPrivate *priv = NULL;
 
@@ -763,14 +764,21 @@ _rtsp_media_factory_wfd_create_xvcapture_bin (GstRTSPMediaFactoryWFD * factory,
   g_object_set (venc, "idr-period", 120, NULL);
   g_object_set (venc, "skip-inbuf", priv->video_enc_skip_inbuf_value, NULL);
 
+  vparse = gst_element_factory_make ("h264parse", "videoparse");
+  if (NULL == vparse) {
+    GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+    goto create_error;
+  }
+  g_object_set (vparse, "config-interval", 1, NULL);
+
   vqueue = gst_element_factory_make ("queue", "video-queue");
   if (!vqueue) {
     GST_ERROR_OBJECT (factory, "failed to create video queue element");
     goto create_error;
   }
 
-  gst_bin_add_many (srcbin, videosrc, vcaps, venc, vqueue, NULL);
-  if (!gst_element_link_many (videosrc, vcaps, venc, vqueue, NULL)) {
+  gst_bin_add_many (srcbin, videosrc, vcaps, venc, vparse, vqueue, NULL);
+  if (!gst_element_link_many (videosrc, vcaps, venc, vparse, vqueue, NULL)) {
     GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
     goto create_error;
   }
diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
index 48868f5..6b81e25 100644
--- a/gst/rtsp-server/rtsp-media.c
+++ b/gst/rtsp-server/rtsp-media.c
@@ -2713,6 +2713,14 @@ no_setup_sdp:
   }
 }
 
+static void
+do_set_seqnum (GstRTSPStream * stream)
+{
+  guint16 seq_num;
+  seq_num = gst_rtsp_stream_get_current_seqnum (stream);
+  gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
+}
+
 /* call with state_lock */
 gboolean
 default_suspend (GstRTSPMedia * media)
@@ -2735,6 +2743,12 @@ default_suspend (GstRTSPMedia * media)
       ret = set_target_state (media, GST_STATE_NULL, TRUE);
       if (ret == GST_STATE_CHANGE_FAILURE)
         goto state_failed;
+      /* Because payloader needs to set the sequence number as
+       * monotonic, we need to preserve the sequence number
+       * after pause. (otherwise going from pause to play,  which
+       * is actually from NULL to PLAY will create a new sequence
+       * number. */
+      g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
       break;
     default:
       break;
diff --git a/gst/rtsp-server/rtsp-server-wfd.c b/gst/rtsp-server/rtsp-server-wfd.c
index 8e14f97..eb56e53 100644
--- a/gst/rtsp-server/rtsp-server-wfd.c
+++ b/gst/rtsp-server/rtsp-server-wfd.c
@@ -71,6 +71,7 @@ struct _GstRTSPWFDServerPrivate
 
   /* the clients that are connected */
   GList *clients;
+  guint64 native_resolution;
   guint64 supported_resolution;
 };
 
@@ -118,6 +119,7 @@ gst_rtsp_wfd_server_init (GstRTSPWFDServer * server)
   GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE (server);
 
   server->priv = priv;
+  server->priv->native_resolution = 0;
   server->priv->supported_resolution = 1;
   GST_INFO_OBJECT (server, "New server is initialized");
 }
@@ -227,6 +229,9 @@ create_client_wfd (GstRTSPServer * server)
   gst_rtsp_wfd_client_set_video_supported_resolution (client,
         priv->supported_resolution);
 
+  gst_rtsp_wfd_client_set_video_native_resolution (client,
+        priv->native_resolution);
+
   GST_RTSP_WFD_SERVER_UNLOCK (server);
 
   return GST_RTSP_CLIENT (client);
@@ -279,3 +284,18 @@ gst_rtsp_wfd_server_set_supported_reso(GstRTSPWFDServer *server, guint64 support
   GST_RTSP_WFD_SERVER_UNLOCK (server);
   return res;
 }
+GstRTSPResult
+gst_rtsp_wfd_server_set_video_native_reso (GstRTSPWFDServer *server, guint64 native_reso)
+{
+	  GstRTSPResult res = GST_RTSP_OK;
+	  GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+	  g_return_val_if_fail (GST_IS_RTSP_WFD_SERVER (server), GST_RTSP_ERROR);
+
+	  GST_RTSP_WFD_SERVER_LOCK (server);
+
+	  priv->native_resolution = native_reso;
+
+	  GST_RTSP_WFD_SERVER_UNLOCK (server);
+	  return res;
+}
diff --git a/gst/rtsp-server/rtsp-server-wfd.h b/gst/rtsp-server/rtsp-server-wfd.h
index 890a0fe..c1da877 100644
--- a/gst/rtsp-server/rtsp-server-wfd.h
+++ b/gst/rtsp-server/rtsp-server-wfd.h
@@ -78,7 +78,8 @@ GType                 gst_rtsp_wfd_server_get_type             (void);
 GstRTSPWFDServer *    gst_rtsp_wfd_server_new                  (void);
 GstRTSPResult         gst_rtsp_wfd_server_trigger_request      (GstRTSPServer *server, GstWFDTriggerType type);
 
-GstRTSPResult gst_rtsp_wfd_server_set_supported_reso(GstRTSPWFDServer *server, guint64 supported_reso);
+GstRTSPResult         gst_rtsp_wfd_server_set_supported_reso (GstRTSPWFDServer *server, guint64 supported_reso);
+GstRTSPResult         gst_rtsp_wfd_server_set_video_native_reso    (GstRTSPWFDServer *server, guint64 native_reso);
 
 #if 0
 void                  gst_rtsp_server_set_address          (GstRTSPServer *server, const gchar *address);
diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index b618f7c..862404b 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -1734,7 +1734,7 @@ gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
   priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
   g_free (name);
 
-  /* get the session, TODO-WFD: This session needs to be decrease ref count? */
+  /* get the session */
   g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
 
   g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
@@ -2035,6 +2035,8 @@ gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
 
   if (priv->srtpenc)
     gst_object_unref (priv->srtpenc);
+  if (priv->srtpdec)
+    gst_object_unref (priv->srtpdec);
 
   priv->is_joined = FALSE;
   g_mutex_unlock (&priv->lock);
@@ -2476,6 +2478,48 @@ gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
 }
 
 /**
+ * gst_rtsp_stream_set_seqnum:
+ * @stream: a #GstRTSPStream
+ * @seqnum: a new sequence number
+ *
+ * Configure the sequence number in the payloader of @stream to @seqnum.
+ */
+void
+gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_seqnum:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured sequence number in the payloader of @stream.
+ *
+ * Returns: the sequence number of the payloader.
+ */
+guint16
+gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  guint seqnum;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+  priv = stream->priv;
+
+  g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
+
+  return seqnum;
+}
+
+/**
  * gst_rtsp_stream_transport_filter:
  * @stream: a #GstRTSPStream
  * @func: (scope call) (allow-none): a callback
diff --git a/gst/rtsp-server/rtsp-stream.h b/gst/rtsp-server/rtsp-stream.h
index 1dbfdf3..135689e 100644
--- a/gst/rtsp-server/rtsp-stream.h
+++ b/gst/rtsp-server/rtsp-stream.h
@@ -153,6 +153,9 @@ gboolean          gst_rtsp_stream_query_position   (GstRTSPStream * stream,
 gboolean          gst_rtsp_stream_query_stop       (GstRTSPStream * stream,
                                                     gint64 * stop);
 
+void              gst_rtsp_stream_set_seqnum_offset          (GstRTSPStream *stream, guint16 seqnum);
+guint16           gst_rtsp_stream_get_current_seqnum          (GstRTSPStream *stream);
+
 /**
  * GstRTSPStreamTransportFilterFunc:
  * @stream: a #GstRTSPStream object
diff --git a/ltmain.sh b/ltmain.sh
index 3fd54df..bffda54 100644
--- a/ltmain.sh
+++ b/ltmain.sh
@@ -70,7 +70,7 @@
 #         compiler:		$LTCC
 #         compiler flags:		$LTCFLAGS
 #         linker:		$LD (gnu? $with_gnu_ld)
-#         $progname:	(GNU libtool) 2.4.2 Debian-2.4.2-1.10
+#         $progname:	(GNU libtool) 2.4.2 Debian-2.4.2-1.11
 #         automake:	$automake_version
 #         autoconf:	$autoconf_version
 #
@@ -80,7 +80,7 @@
 
 PROGRAM=libtool
 PACKAGE=libtool
-VERSION="2.4.2 Debian-2.4.2-1.10"
+VERSION="2.4.2 Debian-2.4.2-1.11"
 TIMESTAMP=""
 package_revision=1.3337
 
diff --git a/packaging/gst-rtsp-server.spec b/packaging/gst-rtsp-server.spec
old mode 100755
new mode 100644
index 2a1edc5..30832e8
--- a/packaging/gst-rtsp-server.spec
+++ b/packaging/gst-rtsp-server.spec
@@ -1,6 +1,6 @@
 Name:       gst-rtsp-server
 Summary:    Multimedia Framework Library
-Version:    1.4.1
+Version:    1.4.5
 Release:    0
 Group:      System/Libraries
 License:    LGPLv2+