Merge branch 'master' into 0.11

Conflicts:
	common
	configure.ac

.gitignore

@@ -10,6 +10,7 @@
 *~
 .deps
 .libs
+ABOUT-NLS
 INSTALL
 Makefile
 Makefile.in
@@ -21,6 +22,7 @@ config.guess
 config.h
 config.h.in
 config.log
+config.rpath
 config.status
 config.sub
 configure
@@ -34,3 +36,5 @@ bindings/python/rtspserver.c
 tags
 gst-rtsp.spec
 stamp-h.in
+
+/m4/*m4

Makefile.am

@@ -3,11 +3,11 @@ DISTCHECK_CONFIGURE_FLAGS=--enable-gtk-doc
 SUBDIRS = 		\
 	gst \
 	bindings	\
-	m4   		\
 	common \
 	pkgconfig \
 	docs \
-	examples
+	examples \
+	tests
 
 DIST_SUBDIRS = $(SUBDIRS)
 
@@ -16,13 +16,15 @@ EXTRA_DIST = \
 	AUTHORS COPYING NEWS README RELEASE REQUIREMENTS \
 	gst-rtsp.spec docs/design/gst-rtp-server-design
 
-ACLOCAL_AMFLAGS = -I common/m4 -I m4
+ACLOCAL_AMFLAGS = -I m4 -I common/m4
 
 DISTCLEANFILES = _stdint.h gst-rtsp.spec
 
 include $(top_srcdir)/common/release.mak
 include $(top_srcdir)/common/po.mak
 
+include $(top_srcdir)/common/coverage/lcov.mak
+
 check-valgrind:
 	cd tests/check && make check-valgrind
 
@@ -37,7 +39,41 @@ endif
 # cruft: plugins that have been merged or moved or renamed
 CRUFT_FILES = \
 	$(top_builddir)/common/shave \
- $(top_builddir)/common/shave-libtool
+	$(top_builddir)/common/shave-libtool \
+	$(top_builddir)/common/m4/codeset.m4 \
+	$(top_builddir)/common/m4/gettext.m4 \
+	$(top_builddir)/common/m4/glibc2.m4 \
+	$(top_builddir)/common/m4/glibc21.m4 \
+	$(top_builddir)/common/m4/iconv.m4 \
+	$(top_builddir)/common/m4/intdiv0.m4 \
+	$(top_builddir)/common/m4/intl.m4 \
+	$(top_builddir)/common/m4/intldir.m4 \
+	$(top_builddir)/common/m4/intlmacosx.m4 \
+	$(top_builddir)/common/m4/intmax.m4 \
+	$(top_builddir)/common/m4/inttypes-pri.m4 \
+	$(top_builddir)/common/m4/inttypes_h.m4 \
+	$(top_builddir)/common/m4/lcmessage.m4 \
+	$(top_builddir)/common/m4/lib-ld.m4 \
+	$(top_builddir)/common/m4/lib-link.m4 \
+	$(top_builddir)/common/m4/lib-prefix.m4 \
+	$(top_builddir)/common/m4/libtool.m4 \
+	$(top_builddir)/common/m4/lock.m4 \
+	$(top_builddir)/common/m4/longlong.m4 \
+	$(top_builddir)/common/m4/ltoptions.m4 \
+	$(top_builddir)/common/m4/ltsugar.m4 \
+	$(top_builddir)/common/m4/ltversion.m4 \
+	$(top_builddir)/common/m4/lt~obsolete.m4 \
+	$(top_builddir)/common/m4/nls.m4 \
+	$(top_builddir)/common/m4/po.m4 \
+	$(top_builddir)/common/m4/printf-posix.m4 \
+	$(top_builddir)/common/m4/progtest.m4 \
+	$(top_builddir)/common/m4/size_max.m4 \
+	$(top_builddir)/common/m4/stdint_h.m4 \
+	$(top_builddir)/common/m4/uintmax_t.m4 \
+	$(top_builddir)/common/m4/visibility.m4 \
+	$(top_builddir)/common/m4/wchar_t.m4 \
+	$(top_builddir)/common/m4/wint_t.m4 \
+	$(top_builddir)/common/m4/xsize.m4
 
 include $(top_srcdir)/common/cruft.mak
 

autogen.sh

@@ -83,7 +83,7 @@ tool_run "$libtoolize" "--copy --force"
 tool_run "$aclocal" "-I m4 -I common/m4 $ACLOCAL_FLAGS"
 tool_run "$autoheader"
 
-# touch the stamp-h.in build stamp so we don't re-run autoheader in maintainer mode -- wingo
+# touch the stamp-h.in build stamp so we don't re-run autoheader in maintainer mode
 echo timestamp > stamp-h.in 2> /dev/null
 
 tool_run "$autoconf"

bindings/python/Makefile.am

@@ -37,6 +37,7 @@ rtspserver.c: $(DEFS) $(OVERRIDES) arg-types.py
 	($(PYTHON) $(srcdir)/codegen/codegen.py \
 		--load-types $(srcdir)/arg-types.py \
 		--register $(srcdir)/rtspserver-types.defs \
+		--register $(PYGST_DEFSDIR)/gst-types.defs \
 		--override $(srcdir)/$*.override \
 		--extendpath $(top_builddir)/gst/ \
 		--extendpath $(srcdir)/ \

bindings/python/arg-types.py

@@ -145,6 +145,54 @@ class GstIteratorArg(ArgType):
 		info.varlist.add('GstIterator', '*ret')
 		info.codeafter.append('    return pygst_iterator_new(ret);')
 
+class GstRTSPUrlArg(ArgType):
+	"""GstRTSPUrl node generator"""
+
+	before = ('    parse_result = gst_rtsp_url_parse (py_%(name)s, &%(name)s);\n'
+		  '    if (parse_result != GST_RTSP_OK) {\n'
+          '      PyErr_SetString(PyExc_TypeError, "invalid url");\n'
+		  '      return NULL;\n'
+          '    }')
+	beforenull = ('    if (py_%(name)s == NULL)\n'
+		      '        %(name)s = NULL;\n'
+		      '    else\n'
+		      '  ' + before)
+	after = ('    if (%(name)s)\n'
+		 '        gst_rtsp_url_free (%(name)s);\n')
+
+	def write_param(self, ptype, pname, pdflt, pnull, keeprefcount, info):
+		if ptype in ('const-GstRTSPUrl*', 'GstRTSPUrl*'):
+			self.write_normal_param(pname, pdflt, pnull, info)
+		else:
+			raise RuntimeError, "write_param not implemented for %s" % ptype
+
+	def write_normal_param(self, pname, pdflt, pnull, info):
+		info.varlist.add('GstRTSPResult', 'parse_result')
+		if pdflt:
+			assert pdflt == 'NULL'
+			info.varlist.add('const char', '*py_' + pname + ' = NULL')
+		else:
+			info.varlist.add('const char', '*py_' + pname)
+		info.varlist.add('GstRTSPUrl', '*'+pname)
+		info.add_parselist('s', ['&py_'+pname], [pname])
+		info.arglist.append(pname)
+		if pnull:
+			info.codebefore.append (self.beforenull % { 'name' : pname  })
+		else:
+			info.codebefore.append (self.before % { 'name' : pname })
+		info.codeafter.append (self.after % { 'name' : pname })
+
+	def write_return(self, ptype, ownsreturn, info):
+		if ptype == 'GstRTSPUrl*':
+			info.varlist.add('GstRTSPUrl', '*ret')
+			copyval = 'FALSE'
+		elif ptype == 'const-GstRTSPUrl*':
+			info.varlist.add('const GstRTSPUrl', '*ret')
+			copyval = 'TRUE'
+		else:
+			raise RuntimeError, "write_return not implemented for %s" % ptype
+		info.codeafter.append('    return pyg_boxed_new (GST_TYPE_RTSP_URL, ret, '+copyval+', TRUE);')
+
 class GstMiniObjectParam(Parameter):
 
 	def get_c_type(self):
@@ -165,6 +213,31 @@ class GstMiniObjectParam(Parameter):
 
 matcher.register_reverse('GstMiniObject*', GstMiniObjectParam)
 
+class GstRTSPUrlParam(Parameter):
+
+	def get_c_type(self):
+		c_type = self.props.get('c_type', None)
+        if c_type and c_type.startswith('const'):
+            return 'const GstRTSPUrl *'
+		return 'GstRTSPUrl *'
+
+	def convert_c2py(self):
+		self.wrapper.add_declaration("char *%s_str = NULL;" % self.name)
+		self.wrapper.add_declaration("PyObject *py_%s = NULL;" % self.name)
+		self.wrapper.write_code(code=("if (%(name)s) {\n"
+					      "    %(name)s_str = gst_rtsp_url_get_request_uri ((GstRTSPUrl*) %(name)s);\n"
+					      "    py_%(name)s = PyString_FromString (%(name)s_str);\n"
+                          "    g_free (%(name)s_str);\n"
+					      "} else {\n"
+					      "    Py_INCREF(Py_None);\n"
+					      "    py_%(name)s = Py_None;\n"
+					      "}" % {'name': self.name}),
+					cleanup=("Py_DECREF(py_%s);" % self.name))
+		self.wrapper.add_pyargv_item("py_%s" % self.name)
+
+matcher.register_reverse('const-GstRTSPUrl*', GstRTSPUrlParam)
+matcher.register_reverse('GstRTSPUrl*', GstRTSPUrlParam)
+
 class GstMiniObjectReturn(ReturnType):
 
 	def get_c_type(self):
@@ -343,6 +416,8 @@ matcher.register('GstCaps', GstCapsArg()) #FIXME: does this work?
 matcher.register('GstCaps*', GstCapsArg()) #FIXME: does this work?
 matcher.register('const-GstCaps*', GstCapsArg())
 matcher.register('GstIterator*', GstIteratorArg())
+matcher.register('const-GstRTSPUrl*', GstRTSPUrlArg())
+matcher.register('GstRTSPUrl*', GstRTSPUrlArg())
 
 arg = PointerArg('gpointer', 'G_TYPE_POINTER')
 matcher.register('GstClockID', arg)

bindings/python/rtspserver-types.defs

@@ -1,6 +1,63 @@
+;; From gst/rtsp-server/rtsp-server.h
+
 (define-object Server
   (in-module "Gst.RTSPServer")
   (parent "GObject")
   (c-name "GstRTSPServer")
   (gtype-id "GST_TYPE_RTSP_SERVER")
 )
+
+;; From gst/rtsp-server/rtsp-media-mapping.h
+
+(define-object MediaMapping
+  (in-module "Gst.RTSPServer")
+  (parent "GObject")
+  (c-name "GstRTSPMediaMapping")
+  (gtype-id "GST_TYPE_RTSP_MEDIA_MAPPING")
+)
+
+;; From gst/rtsp-server/rtsp-media-factory.h
+
+(define-object MediaFactory
+  (in-module "Gst.RTSPServer")
+  (parent "GObject")
+  (c-name "GstRTSPMediaFactory")
+  (gtype-id "GST_TYPE_RTSP_MEDIA_FACTORY")
+)
+
+;; From gst/rtsp-server/rtsp-media.h
+
+(define-object Media
+  (in-module "Gst.RTSPServer")
+  (parent "GObject")
+  (c-name "GstRTSPMedia")
+  (gtype-id "GST_TYPE_RTSP_MEDIA")
+)
+
+;; From gst/rtsp-server/rtsp-session-pool.h
+
+(define-object SessionPool
+  (in-module "Gst")
+  (parent "GObject")
+  (c-name "GstRTSPSessionPool")
+  (gtype-id "GST_TYPE_RTSP_SESSION_POOL")
+)
+
+;; From gst/rtsp-server/rtsp-session.h
+
+(define-object Session
+  (in-module "Gst")
+  (parent "GObject")
+  (c-name "GstRTSPSession")
+  (gtype-id "GST_TYPE_RTSP_SESSION")
+)
+
+;; From gst/rtsp-server/rtsp-client.h
+
+(define-object Client
+  (in-module "Gst")
+  (parent "GObject")
+  (c-name "GstRTSPClient")
+  (gtype-id "GST_TYPE_RTSP_CLIENT")
+)
+

bindings/python/rtspserver.defs

@@ -1,11 +1,99 @@
 (include "rtspserver-types.defs")
 
+;; From gst/rtsp-server/rtsp-server.h
+
 (define-function rtsp_server_new
   (c-name "gst_rtsp_server_new")
   (is-constructor-of "GstRTSPServer")
   (return-type "GstRTSPServer*")
 )
 
+(define-method set_address
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_set_address")
+  (parameters
+    '("const-gchar*" "address")
+  )
+)
+
+(define-method get_address
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_get_address")
+  (return-type "const-gchar*")
+)
+
+(define-method set_service
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_set_service")
+  (parameters
+    '("const-gchar*" "service")
+  )
+)
+
+(define-method get_service
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_get_service")
+  (return-type "const-gchar*")
+)
+
+(define-method set_backlog
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_set_backlog")
+  (parameters
+    '("gint" "backlog")
+  )
+)
+
+(define-method get_backlog
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_get_backlog")
+  (return-type "gint")
+)
+
+(define-method set_session_pool
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_set_session_pool")
+  (parameters
+    '("GstRTSPSessionPool*" "pool")
+  )
+)
+
+(define-method get_session_pool
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_get_session_pool")
+  (return-type "GstRTSPSessionPool*")
+)
+
+(define-method set_media_mapping
+ (of-object "GstRTSPServer")
+ (c-name "gst_rtsp_server_set_media_mapping")
+  (parameters
+    '("GstRTSPMediaMapping*" "mapping")
+  )
+)
+
+(define-method get_media_mapping
+ (of-object "GstRTSPServer")
+ (c-name "gst_rtsp_server_get_media_mapping")
+ (return-type "GstRTSPMediaMapping*")
+)
+
+(define-function io_func
+ (c-name "gst_rtsp_server_io_func")
+ (return-type "gboolean")
+  (parameters
+    '("GIOChannel*" "channel")
+    '("GIOCondition" "condition")
+    '("GstRTSPServer*" "server")
+  )
+)
+
+(define-method get_io_channel
+ (of-object "GstRTSPServer")
+ (c-name "gst_rtsp_server_get_io_channel")
+ (return-type "GIOChannel*")
+)
+
 (define-method attach
   (of-object "GstRTSPServer")
   (c-name "gst_rtsp_server_attach")
@@ -14,3 +102,298 @@
     '("GMainContext*" "context")
   )
 )
+
+(define-method create_watch
+  (of-object "GstRTSPServer")
+  (c-name "gst_rtsp_server_create_watch")
+  (return-type "GSource*")
+)
+
+;; From gst/rtsp-server/rtsp-media-mapping.h
+
+(define-function rtsp_media_mapping_new
+  (c-name "gst_rtsp_media_mapping_new")
+  (is-constructor-of "GstRTSPMediaMapping")
+  (return-type "GstRTSPMediaMapping*")
+)
+
+
+;; TODO define const-GstRTSPUrl* on arg-types.py
+(define-method find_factory
+  (of-object "GstRTSPMediaMapping")
+  (c-name "gst_rtsp_media_mapping_find_factory")
+  (return-type "GstRTSPMediaFactory*")
+  (parameters
+   '("const-GstRTSPUrl*" "url")
+  )
+)
+
+(define-method add_factory
+  (of-object "GstRTSPMediaMapping")
+  (c-name "gst_rtsp_media_mapping_add_factory")
+  (parameters
+   '("const-gchar*" "path")
+   '("GstRTSPMediaFactory*" "factory")
+  )
+)
+
+(define-method remove_factory
+  (of-object "GstRTSPMediaMapping")
+  (c-name "gst_rtsp_media_mapping_remove_factory")
+  (parameters
+   '("const-gchar*" "path")
+  )
+)
+
+;; From gst/rtsp-server/rtsp-media-factory.h
+
+(define-function rtsp_media_factory_new
+  (c-name "gst_rtsp_media_factory_new")
+  (is-constructor-of "GstRTSMediaFactory")
+  (return-type "GstRTSPMediaFactory*")
+)
+
+(define-method set_launch
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_set_launch")
+  (parameters
+   '("gchar*" "launch")
+  )
+)
+
+(define-method get_launch
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_get_launch")
+  (return-type "gchar*")
+)
+
+(define-method set_shared
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_set_shared")
+  (parameters
+   '("gboolean" "shared")
+  )
+)
+
+(define-method is_shared
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_is_shared")
+  (return-type "gboolean")
+)
+
+(define-method set_eos_shutdown
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_set_eos_shutdown")
+  (parameters
+   '("gboolean" "eos_shutdown")
+  )
+)
+
+(define-method is_eos_shutdown
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_is_eos_shutdown")
+  (return-type "gboolean")
+)
+
+;; TODO define const-GstRTSPUrl* on arg-types.py
+(define-method construct
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_construct")
+  (return-type "GstRTSPMedia*")
+  (parameters
+   '("const-GstRTSPUrl*" "url")
+  )
+)
+
+(define-method collect_streams
+  (of-object "GstRTSPMediaFactory")
+  (c-name "gst_rtsp_media_factory_collect_streams")
+  (parameters
+   '("const-GstRTSPUrl*" "url")
+   '("GstRTSPMedia*" "media")
+  )
+)
+
+(define-virtual get_element
+  (of-object "GstRTSPMediaFactory")
+  (parameters
+   '("const-GstRTSPUrl*" "url")
+  )
+  (return-type "GstElement*")
+)
+
+
+(define-virtual construct
+  (of-object "GstRTSPMediaFactory")
+  (parameters
+   '("const-GstRTSPUrl*" "url")
+  )
+  (return-type "GstRTSPMedia*")
+)
+
+
+(define-virtual configure
+  (of-object "GstRTSPMediaFactory")
+  (parameters
+   '("GstRTSPMedia*" "media")
+  )
+  (return-type "none")
+)
+
+
+(define-virtual create_pipeline
+  (of-object "GstRTSPMediaFactory")
+  (parameters
+   '("GstRTSPMedia*" "media")
+  )
+  (return-type "GstElement*")
+)
+
+
+;; From gst/rtsp-server/rtsp-session-pool.h
+
+(define-function gst_rtsp_session_pool_new
+  (c-name "gst_rtsp_session_pool_new")
+  (is-constructor-of "GstRTSPSessionPool")
+  (return-type "GstRTSPSessionPool*")
+)
+
+(define-method set_max_sessions
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_set_max_sessions")
+  (return-type "none")
+  (parameters
+    '("guint" "max")
+  )
+)
+
+(define-method get_max_sessions
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_get_max_sessions")
+  (return-type "guint")
+)
+
+(define-method get_n_sessions
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_get_n_sessions")
+  (return-type "guint")
+)
+
+(define-method create
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_create")
+  (return-type "GstRTSPSession*")
+)
+
+(define-method find
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_find")
+  (return-type "GstRTSPSession*")
+  (parameters
+    '("const-gchar*" "sessionid")
+  )
+)
+
+(define-method remove
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_remove")
+  (return-type "gboolean")
+  (parameters
+    '("GstRTSPSession*" "sess")
+  )
+)
+
+(define-method filter
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_filter")
+  (return-type "GList*")
+  (parameters
+    '("GstRTSPSessionFilterFunc" "func")
+    '("gpointer" "user_data")
+  )
+)
+
+(define-method cleanup
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_cleanup")
+  (return-type "guint")
+)
+
+(define-method create_watch
+  (of-object "GstRTSPSessionPool")
+  (c-name "gst_rtsp_session_pool_create_watch")
+  (return-type "GSource*")
+)
+
+;; From gst/rtsp-server/rtsp-client.h
+
+(define-function gst_rtsp_client_new
+  (c-name "gst_rtsp_client_new")
+  (is-constructor-of "GstRTSPClient")
+  (return-type "GstRTSPClient*")
+)
+
+(define-method set_session_pool
+  (of-object "GstRTSPClient")
+  (c-name "gst_rtsp_client_set_session_pool")
+  (return-type "none")
+  (parameters
+    '("GstRTSPSessionPool*" "pool")
+  )
+)
+
+(define-method get_session_pool
+  (of-object "GstRTSPClient")
+  (c-name "gst_rtsp_client_get_session_pool")
+  (return-type "GstRTSPSessionPool*")
+)
+
+(define-method set_media_mapping
+  (of-object "GstRTSPClient")
+  (c-name "gst_rtsp_client_set_media_mapping")
+  (return-type "none")
+  (parameters
+    '("GstRTSPMediaMapping*" "mapping")
+  )
+)
+
+(define-method get_media_mapping
+  (of-object "GstRTSPClient")
+  (c-name "gst_rtsp_client_get_media_mapping")
+  (return-type "GstRTSPMediaMapping*")
+)
+
+(define-method accept
+  (of-object "GstRTSPClient")
+  (c-name "gst_rtsp_client_accept")
+  (return-type "gboolean")
+  (parameters
+    '("GIOChannel*" "channel")
+  )
+)
+
+;; From bindings/python/rtsp-params.h
+
+(define-function gst_rtsp_params_set
+  (c-name "gst_rtsp_params_set")
+  (return-type "GstRTSPResult")
+  (parameters
+    '("GstRTSPClient*" "client")
+    '("GstRTSPUrl*" "uri")
+    '("GstRTSPSession*" "session")
+    '("GstRTSPMessage*" "request")
+    '("GstRTSPMessage*" "response")
+  )
+)
+
+(define-function gst_rtsp_params_get
+  (c-name "gst_rtsp_params_get")
+  (return-type "GstRTSPResult")
+  (parameters
+    '("GstRTSPClient*" "client")
+    '("GstRTSPUrl*" "uri")
+    '("GstRTSPSession*" "session")
+    '("GstRTSPMessage*" "request")
+    '("GstRTSPMessage*" "response")
+  )
+)

bindings/python/rtspserver.override

@@ -5,6 +5,11 @@ headers
 #define NO_IMPORT_PYGOBJECT
 #include <pygobject.h>
 
+#include <gst/gst.h>
+#include <gst/rtsp-server/rtsp-server.h>
+
+#include <glib.h>
+
 #ifdef HAVE_CONFIG_H
 # include <config.h>
 #endif
@@ -24,7 +29,6 @@ typedef struct {
 %%
 import gobject.GObject as PyGObject_Type
 import gobject.MainContext as PyGMainContext_Type
-
 %%
 override gst_rtsp_server_attach kwargs
 static PyObject *
@@ -50,3 +54,14 @@ _wrap_gst_rtsp_server_attach (PyGObject *self,
 
   return PyLong_FromLong (res);
 }
+%%
+override gst_rtsp_server_create_watch kwargs
+static PyObject *
+_wrap_gst_rtsp_server_create_watch(PyGObject *self, PyObject *args, PyObject *keywords)
+{
+    GSource *ret;
+    pyg_begin_allow_threads;
+    ret = gst_rtsp_server_create_watch(GST_RTSP_SERVER(self->obj));
+    pyg_end_allow_threads;
+    return pygobject_new((GObject *)ret);
+}

bindings/python/test.py

@@ -0,0 +1,130 @@
+import unittest
+import rtspserver
+
+
+def pubdir(obj):
+    return [d for d in dir(obj) if not d.startswith('_')]
+
+#print 'Module listing:', pubdir(rtspserver)
+
+from rtspserver import Server, SessionPool, Session, MediaMapping, MediaFactory
+
+#print 'Server listing: ', pubdir(Server)
+#print 'MediaMapping listing: ', pubdir(MediaMapping)
+#print 'MediaFactory listing: ', pubdir(MediaFactory)
+#print 'SessionPool listing: ', pubdir(SessionPool)
+#print 'Session listing: ', pubdir(Session)
+
+
+class ServerTestCase(unittest.TestCase):
+
+    def setUp(self):
+        self.server = Server()
+
+    def tearDown(self):
+        del self.server
+
+    def test_address(self):
+        """ Server address set/get """
+        addr = '1.2.3.4'
+        self.server.set_address(addr)
+        self.assertEquals(addr, self.server.get_address())
+
+    def test_service(self):
+        """ Server service set/get """
+        service = '12345'
+        self.server.set_service(service)
+        self.assertEquals(service, self.server.get_service())
+
+    def test_backlog(self):
+        """ Server backlog set/get """
+        backlog = 1234
+        self.server.set_backlog(backlog)
+        self.assertEquals(backlog, self.server.get_backlog())
+
+    def test_session_pool(self):
+        """ Server session pool set/get """
+        pool = SessionPool()
+        self.server.set_session_pool(pool)
+        self.assertEquals(pool, self.server.get_session_pool())
+
+    def test_media_mapping(self):
+        """ Server media mapping set/get """
+        mmap = MediaMapping()
+        self.server.set_media_mapping(mmap)
+        self.assertEquals(mmap, self.server.get_media_mapping())
+
+
+class MediaMappingTestCase(unittest.TestCase):
+
+    def setUp(self):
+        self.mmap = MediaMapping()
+
+    def tearDown(self):
+        del self.mmap
+
+    def test_factory(self):
+        """ MediaMapping factory add/remove """
+        self.factory = MediaFactory()
+        self.mmap.add_factory("/test", self.factory)
+        self.mmap.remove_factory("/test")
+
+
+class MediaFactoryTestCase(unittest.TestCase):
+
+    def setUp(self):
+        self.factory = MediaFactory()
+
+    def tearDown(self):
+        del self.factory
+
+    def test_launch(self):
+        """ MediaFactory launch set/get """
+        launch = "videotestsrc ! xvimagesink"
+        self.factory.set_launch(launch)
+        self.assertEquals(launch, self.factory.get_launch())
+
+    def test_shared(self):
+        """ MediaFactory shared set/is """
+        self.factory.set_shared(True)
+        self.assert_(self.factory.is_shared())
+        self.factory.set_shared(False)
+        self.assert_(not self.factory.is_shared())
+
+    def test_eos_shutdown(self):
+        """ MediaFactory eos_shutdown set/is """
+        self.factory.set_eos_shutdown(True)
+        self.assert_(self.factory.is_eos_shutdown())
+        self.factory.set_eos_shutdown(False)
+        self.assert_(not self.factory.is_eos_shutdown())
+
+
+
+def alltests():
+    tests = []
+
+    for p in dir(ServerTestCase):
+        try:
+            if 'test_' in p:
+                tests.append(ServerTestCase(p))
+        except:
+            pass
+
+    for p in dir(MediaMappingTestCase):
+        try:
+            if 'test_' in p:
+                tests.append(MediaMappingTestCase(p))
+        except:
+            pass
+
+    for p in dir(MediaFactoryTestCase):
+        try:
+            if 'test_' in p:
+                tests.append(MediaFactoryTestCase(p))
+        except:
+            pass
+
+    return unittest.TestSuite(tests)
+
+
+unittest.TextTestRunner(verbosity=2).run(alltests())

configure.ac

@@ -52,6 +52,7 @@ AS_AUTOTOOLS_ALTERNATE
 
 dnl Add parameters for aclocal
 AC_SUBST(ACLOCAL_AMFLAGS, "-I m4 -I common/m4")
+AC_CONFIG_MACRO_DIR([m4])
 
 dnl set up gettext
 dnl the version check needs to stay here because autopoint greps for it
@@ -67,6 +68,8 @@ AG_GST_ARG_GCOV
 AG_GST_ARG_WITH_PACKAGE_NAME
 AG_GST_ARG_WITH_PACKAGE_ORIGIN
 
+AG_GST_PKG_CONFIG_PATH
+
 dnl *** checks for platform ***
 
 dnl * hardware/architecture *
@@ -77,6 +80,16 @@ dnl find a compiler
 AC_PROG_CC
 AM_PROG_CC_C_O
 
+AC_PATH_PROG(VALGRIND_PATH, valgrind, no)
+AM_CONDITIONAL(HAVE_VALGRIND, test ! "x$VALGRIND_PATH" = "xno")
+
+dnl check for gobject-introspection
+GOBJECT_INTROSPECTION_CHECK([0.6.3])
+
+dnl check for documentation tools
+AG_GST_DOCBOOK_CHECK
+GTK_DOC_CHECK([1.3])
+
 dnl check for python
 AM_PATH_PYTHON
 AC_MSG_CHECKING(for python >= 2.3)
@@ -89,49 +102,33 @@ sys.exit(0)"
 
 if $PYTHON -c "$prog" 1>&AC_FD_CC 2>&AC_FD_CC
 then
+  HAVE_PYTHON=yes
   AC_MSG_RESULT(okay)
 else
-  AC_MSG_ERROR(too old)
+  HAVE_PYTHON=no
+  AC_MSG_RESULT(no python)
 fi
-AM_CHECK_PYTHON_HEADERS(,[AC_MSG_ERROR(could not find Python headers)])
 
-AC_PATH_PROG(VALGRIND_PATH, valgrind, no)
+AM_CHECK_PYTHON_HEADERS([HAVE_PYTHON_HEADERS=yes],[HAVE_PYTHON_HEADERS=no])
-AM_CONDITIONAL(HAVE_VALGRIND, test ! "x$VALGRIND_PATH" = "xno")
-
-dnl check for gobject-introspection
-GOBJECT_INTROSPECTION_CHECK([0.6.3])
-
-dnl check for documentation tools
-AG_GST_DOCBOOK_CHECK
-GTK_DOC_CHECK([1.3])
-
-AC_SUBST(PYGOBJECT_REQ, 2.11.2)
 
 dnl check for pygobject (optional, used in the bindings)
+PYGOBJECT_REQ=2.11.2
 PKG_CHECK_MODULES(PYGOBJECT, pygobject-2.0 >= $PYGOBJECT_REQ,
-  [
+  [HAVE_PYGOBJECT="yes"], [HAVE_PYGOBJECT="no"])
-    HAVE_PYGOBJECT="yes"
-  ],
-  [
-    HAVE_PYGOBJECT="no"
-  ])
 AC_SUBST(PYGOBJECT_CFLAGS)
 
 dnl check for gst-python
 PKG_CHECK_MODULES(PYGST, gst-python-0.10,
-  [
+  [HAVE_PYGST="yes"], [HAVE_PYGST="no"])
-    HAVE_PYGST="yes"
-  ],
-  [
-    HAVE_PYGST="no"
-  ])
 
 if test "x$HAVE_PYGST" = "xyes"; then
   PYGST_DEFSDIR=`pkg-config gst-python-0.10 --variable=defsdir`
 fi
 AC_SUBST(PYGST_DEFSDIR, $PYGST_DEFSDIR)
 
-if test "x$HAVE_PYTHON_HEADERS" = "xyes" -a \
+if test \
+    "x$HAVE_PYTHON" = "xyes" -a \
+    "x$HAVE_PYTHON_HEADERS" = "xyes" -a \
     "x$HAVE_PYGOBJECT" = "xyes" -a \
     "x$HAVE_PYGST" = "xyes"; then
   HAVE_PYTHON_BINDINGS="yes"
@@ -286,10 +283,10 @@ Makefile
 gst-rtsp.spec
 common/Makefile
 common/m4/Makefile
-m4/Makefile
 gst/Makefile
 gst/rtsp-server/Makefile
 examples/Makefile
+tests/Makefile
 bindings/Makefile
 bindings/python/Makefile
 bindings/python/codegen/Makefile

docs/.gitignore

@@ -0,0 +1 @@
+version.entities

docs/Makefile.am

@@ -9,3 +9,12 @@ DIST_SUBDIRS = libs
 
 EXTRA_DIST = \
 	version.entities.in
+
+upload:
+	@if test "x$(SUBDIRS)" != x; then \
+	  for a in $(SUBDIRS); do \
+	    cd $$a; \
+	    make upload; \
+	    cd ..; \
+	  done; \
+	fi

docs/libs/.gitignore

@@ -0,0 +1,22 @@
+*.stamp
+html
+tmpl
+xml
+Makefile
+Makefile.in
+*-decl.txt
+*-decl-list.txt
+*-presed-scan.c
+*-undeclared.txt
+*-undocumented.txt
+*-unused.txt
+*-overrides.txt
+*.args
+*.hierarchy
+*.interfaces
+*.prerequisites
+*.signals
+doc-registry.xml
+
+*-unused.sgml
+*.bak

docs/libs/Makefile.am

@@ -8,7 +8,7 @@ DOC_MODULE=$(MODULE)
 
 # for upload-doc.mak
 DOC=$(MODULE)
-FORMATS=html ps pdf
+FORMATS=html
 html: html-build.stamp
 include $(top_srcdir)/common/upload-doc.mak
 

docs/libs/gst-rtsp-server-docs.sgml

@@ -9,15 +9,17 @@
   <bookinfo>
     <title>GStreamer RTSP Server Reference Manual</title>
     <releaseinfo>
-      for GStreamer RTSP Server &GST_MAJORMINOR;
+      for GStreamer RTSP Server &GST_VERSION;
     </releaseinfo>
   </bookinfo>
 
   <chapter>
     <xi:include href="xml/rtsp-client.xml"/>
     <xi:include href="xml/rtsp-media-factory.xml"/>
+    <xi:include href="xml/rtsp-media-factory-uri.xml"/>
     <xi:include href="xml/rtsp-media-mapping.xml"/>
     <xi:include href="xml/rtsp-media.xml"/>
+    <xi:include href="xml/rtsp-auth.xml"/>
     <xi:include href="xml/rtsp-params.xml"/>
     <xi:include href="xml/rtsp-sdp.xml"/>
     <xi:include href="xml/rtsp-server.xml"/>

docs/libs/gst-rtsp-server-sections.txt

@@ -45,6 +45,30 @@ GST_IS_RTSP_MEDIA_FACTORY_CLASS
 GST_RTSP_MEDIA_FACTORY_GET_CLASS
 </SECTION>
 
+<SECTION>
+<FILE>rtsp-media-factory-uri</FILE>
+<TITLE>GstRTSPMediaFactoryURI</TITLE>
+GST_RTSP_MEDIA_FACTORY_GET_LOCK
+GST_RTSP_MEDIA_FACTORY_LOCK
+GST_RTSP_MEDIA_FACTORY_UNLOCK
+GstRTSPMediaFactoryURI
+GstRTSPMediaFactoryURIClass
+gst_rtsp_media_factory_uri_new
+gst_rtsp_media_factory_uri_set_uri
+gst_rtsp_media_factory_uri_get_uri
+<SUBSECTION Standard>
+GST_IS_RTSP_MEDIA_FACTORY_URI
+GST_IS_RTSP_MEDIA_FACTORY_URI_CLASS
+GST_RTSP_MEDIA_FACTORY_URI
+GST_RTSP_MEDIA_FACTORY_URI_CAST
+GST_RTSP_MEDIA_FACTORY_URI_CLASS
+GST_RTSP_MEDIA_FACTORY_URI_CLASS_CAST
+GST_RTSP_MEDIA_FACTORY_URI_GET_CLASS
+GST_TYPE_RTSP_MEDIA_FACTORY_URI
+gst_rtsp_media_factory_uri_get_type
+</SECTION>
+
+
 <SECTION>
 <FILE>rtsp-media</FILE>
 <TITLE>GstRTSPMedia</TITLE>
@@ -53,6 +77,7 @@ GstRTSPMedia
 GstRTSPMediaClass
 GstRTSPMediaTrans
 GstRTSPSendFunc
+GstRTSPSendListFunc
 GstRTSPKeepAliveFunc
 GstRTSPMediaStatus
 gst_rtsp_media_new
@@ -70,6 +95,7 @@ gst_rtsp_media_unprepare
 gst_rtsp_media_n_streams
 gst_rtsp_media_get_stream
 gst_rtsp_media_seek
+gst_rtsp_media_get_range_string
 gst_rtsp_media_stream_rtp
 gst_rtsp_media_stream_rtcp
 gst_rtsp_media_set_state
@@ -103,6 +129,8 @@ gst_rtsp_server_set_session_pool
 gst_rtsp_server_get_session_pool
 gst_rtsp_server_set_media_mapping
 gst_rtsp_server_get_media_mapping
+gst_rtsp_server_get_auth
+gst_rtsp_server_set_auth
 gst_rtsp_server_io_func
 gst_rtsp_server_get_io_channel
 gst_rtsp_server_create_watch
@@ -186,16 +214,42 @@ GST_IS_RTSP_SESSION_CLASS
 GST_RTSP_SESSION_GET_CLASS
 </SECTION>
 
+<SECTION>
+<FILE>rtsp-auth</FILE>
+<TITLE>GstRTSPAuth</TITLE>
+GstRTSPAuth
+GstRTSPAuthClass
+gst_rtsp_auth_new
+gst_rtsp_auth_set_basic
+gst_rtsp_auth_setup_auth
+gst_rtsp_auth_check_method
+gst_rtsp_auth_make_basic
+<SUBSECTION Standard>
+GST_IS_RTSP_AUTH
+GST_IS_RTSP_AUTH_CLASS
+GST_RTSP_AUTH
+GST_RTSP_AUTH_CAST
+GST_RTSP_AUTH_CLASS
+GST_RTSP_AUTH_CLASS_CAST
+GST_RTSP_AUTH_GET_CLASS
+GST_TYPE_RTSP_AUTH
+gst_rtsp_auth_get_type
+</SECTION>
+
 <SECTION>
 <FILE>rtsp-client</FILE>
 <TITLE>GstRTSPClient</TITLE>
 GstRTSPClient
 GstRTSPClientClass
 gst_rtsp_client_new
+gst_rtsp_client_set_server
+gst_rtsp_client_get_server
 gst_rtsp_client_set_session_pool
 gst_rtsp_client_get_session_pool
 gst_rtsp_client_set_media_mapping
 gst_rtsp_client_get_media_mapping
+gst_rtsp_client_set_auth
+gst_rtsp_client_get_auth
 gst_rtsp_client_accept
 <SUBSECTION Standard>
 GST_RTSP_CLIENT_CLASS

docs/libs/gst-rtsp-server.types

@@ -1,4 +1,5 @@
 #include <gst/gst.h>
+gst_rtsp_auth_get_type
 gst_rtsp_media_mapping_get_type
 gst_rtsp_media_factory_get_type
 gst_rtsp_media_get_type

docs/version.entities.in

@@ -1 +1,2 @@
 <!ENTITY GST_MAJORMINOR "@GST_MAJORMINOR@">
+<!ENTITY GST_VERSION "@VERSION@">

examples/.gitignore

@@ -4,3 +4,5 @@ test-ogg
 test-readme
 test-sdp
 test-video
+test-uri
+test-auth

examples/Makefile.am

@@ -1,4 +1,4 @@
-noinst_PROGRAMS = test-video test-ogg test-mp4 test-readme test-launch test-sdp test-uri
+noinst_PROGRAMS = test-video test-ogg test-mp4 test-readme test-launch test-sdp test-uri test-auth
 
 INCLUDES = -I$(top_srcdir) -I$(srcdir)
 

examples/test-auth.c

@@ -0,0 +1,115 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMediaMapping *mapping;
+  GstRTSPMediaFactory *factory;
+  GstRTSPAuth *auth;
+  gchar *basic;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mapping for this server, every server has a default mapper object
+   * that be used to map uri mount points to media factories */
+  mapping = gst_rtsp_server_get_media_mapping (server);
+
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw-yuv,width=352,height=288,framerate=15/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 "
+      "audiotestsrc ! audio/x-raw-int,rate=8000 ! "
+      "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+
+  /* make a new authentication manager */
+  auth = gst_rtsp_auth_new ();
+  basic = gst_rtsp_auth_make_basic ("user", "admin");
+  gst_rtsp_auth_set_basic (auth, basic);
+  g_free (basic);
+  gst_rtsp_media_factory_set_auth (factory, auth);
+  g_object_unref (auth);
+  /* attach the test factory to the /test url */
+  gst_rtsp_media_mapping_add_factory (mapping, "/test", factory);
+
+  /* make another factory */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw-yuv,width=352,height=288,framerate=30/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 )");
+  /* make a new authentication manager */
+  auth = gst_rtsp_auth_new ();
+  basic = gst_rtsp_auth_make_basic ("user2", "admin2");
+  gst_rtsp_auth_set_basic (auth, basic);
+  g_free (basic);
+  gst_rtsp_media_factory_set_auth (factory, auth);
+  g_object_unref (auth);
+  /* attach the test factory to the /test url */
+  gst_rtsp_media_mapping_add_factory (mapping, "/test2", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mapping);
+
+  /* attach the server to the default maincontext */
+  if (gst_rtsp_server_attach (server, NULL) == 0)
+    goto failed;
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+  /* start serving */
+  g_main_loop_run (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}

examples/test-uri.c

@@ -60,7 +60,13 @@ main (int argc, char *argv[])
 
   /* make a URI media factory for a test stream. */
   factory = gst_rtsp_media_factory_uri_new ();
+  /* when using GStreamer as a client, one can use the gst payloader, which is
+   * more efficient when there is no payloader for the compressed format */
+  /* g_object_set (factory, "use-gstpay", TRUE, NULL); */
   gst_rtsp_media_factory_uri_set_uri (factory, argv[1]);
+  /* if you want multiple clients to see the same video, set the shared property
+   * to TRUE */
+  /* gst_rtsp_media_factory_set_shared ( GST_RTSP_MEDIA_FACTORY (factory), TRUE); */
 
   /* attach the test factory to the /test url */
   gst_rtsp_media_mapping_add_factory (mapping, "/test",

examples/test-video.c

@@ -21,7 +21,13 @@
 
 #include <gst/rtsp-server/rtsp-server.h>
 
+/* define this if you want the resource to only be available when using
+ * user/admin as the password */
+#undef WITH_AUTH
 
+/* this timeout is periodically run to clean up the expired sessions from the
+ * pool. This needs to be run explicitly currently but might be done
+ * automatically as part of the mainloop. */
 static gboolean
 timeout (GstRTSPServer * server, gboolean ignored)
 {
@@ -41,6 +47,10 @@ main (int argc, char *argv[])
   GstRTSPServer *server;
   GstRTSPMediaMapping *mapping;
   GstRTSPMediaFactory *factory;
+#ifdef WITH_AUTH
+  GstRTSPAuth *auth;
+  gchar *basic;
+#endif
 
   gst_init (&argc, &argv);
 
@@ -53,6 +63,17 @@ main (int argc, char *argv[])
    * that be used to map uri mount points to media factories */
   mapping = gst_rtsp_server_get_media_mapping (server);
 
+#ifdef WITH_AUTH
+  /* make a new authentication manager. it can be added to control access to all
+   * the factories on the server or on individual factories. */
+  auth = gst_rtsp_auth_new ();
+  basic = gst_rtsp_auth_make_basic ("user", "admin");
+  gst_rtsp_auth_set_basic (auth, basic);
+  g_free (basic);
+  /* configure in the server */
+  gst_rtsp_server_set_auth (server, auth);
+#endif
+
   /* make a media factory for a test stream. The default media factory can use
    * gst-launch syntax to create pipelines. 
    * any launch line works as long as it contains elements named pay%d. Each
@@ -74,9 +95,10 @@ main (int argc, char *argv[])
   if (gst_rtsp_server_attach (server, NULL) == 0)
     goto failed;
 
+  /* add a timeout for the session cleanup */
   g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
 
-  /* start serving */
+  /* start serving, this never stops */
   g_main_loop_run (loop);
 
   return 0;

gst/rtsp-server/.gitignore

@@ -0,0 +1,2 @@
+GstRtspServer-0.10.gir
+GstRtspServer-0.10.typelib

gst/rtsp-server/Makefile.am

@@ -1,4 +1,5 @@
 public_headers = \
+		rtsp-auth.h \
 		rtsp-params.h \
 		rtsp-sdp.h \
 		rtsp-media.h \
@@ -11,6 +12,8 @@ public_headers = \
 		rtsp-server.h
 
 c_sources = \
+	rtsp-funnel.c \
+	rtsp-auth.c \
 	rtsp-params.c \
 	rtsp-sdp.c \
 	rtsp-media.c \
@@ -22,6 +25,8 @@ c_sources = \
 	rtsp-client.c \
 	rtsp-server.c
 
+noinst_HEADERS = rtsp-funnel.h
+
 lib_LTLIBRARIES = \
 	libgstrtspserver-@GST_MAJORMINOR@.la
 
@@ -60,6 +65,8 @@ GstRtspServer-@GST_MAJORMINOR@.gir: $(INTROSPECTION_SCANNER) libgstrtspserver-@G
 		-DIN_GOBJECT_INTROSPECTION=1 \
 		--c-include='gst/gst.h' \
 		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-0.10` \
+		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-0.10` \
+		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-0.10` \
 		--library=libgstrtspserver-0.10.la \
 		--include=Gst-0.10 \
 		--include=GstRtsp-0.10 \
@@ -81,7 +88,14 @@ typelibsdir = $(libdir)/girepository-1.0/
 typelibs_DATA = $(BUILT_GIRSOURCES:.gir=.typelib)
 
 %.typelib: %.gir $(INTROSPECTION_COMPILER)
-	$(AM_V_GEN)$(INTROSPECTION_COMPILER) --includedir=$(srcdir) --includedir=$(builddir) $(INTROSPECTION_COMPILER_OPTS) $< -o $(@F)
+	$(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+	$(INTROSPECTION_COMPILER) \
+	--includedir=$(srcdir) \
+	--includedir=$(builddir) \
+	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-0.10` \
+	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-0.10` \
+	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-0.10` \
+	$(INTROSPECTION_COMPILER_OPTS) $< -o $(@F)
 
 CLEANFILES += $(BUILT_GIRSOURCES) $(typelibs_DATA)
 endif

gst/rtsp-server/rtsp-auth.c

@@ -0,0 +1,272 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <string.h>
+
+#include "rtsp-auth.h"
+
+enum
+{
+  PROP_0,
+  PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_auth_debug);
+#define GST_CAT_DEFAULT rtsp_auth_debug
+
+static void gst_rtsp_auth_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_finalize (GObject * obj);
+
+static gboolean default_setup_auth (GstRTSPAuth * auth, GstRTSPClient * client,
+    GQuark hint, GstRTSPClientState * state);
+static gboolean default_check_method (GstRTSPAuth * auth,
+    GstRTSPClient * client, GQuark hint, GstRTSPClientState * state);
+
+G_DEFINE_TYPE (GstRTSPAuth, gst_rtsp_auth, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_auth_class_init (GstRTSPAuthClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_auth_get_property;
+  gobject_class->set_property = gst_rtsp_auth_set_property;
+  gobject_class->finalize = gst_rtsp_auth_finalize;
+
+  klass->setup_auth = default_setup_auth;
+  klass->check_method = default_check_method;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_auth_debug, "rtspauth", 0, "GstRTSPAuth");
+}
+
+static void
+gst_rtsp_auth_init (GstRTSPAuth * auth)
+{
+  /* bitwise or of all methods that need authentication */
+  auth->methods = GST_RTSP_DESCRIBE |
+      GST_RTSP_ANNOUNCE |
+      GST_RTSP_GET_PARAMETER |
+      GST_RTSP_SET_PARAMETER |
+      GST_RTSP_PAUSE |
+      GST_RTSP_PLAY | GST_RTSP_RECORD | GST_RTSP_SETUP | GST_RTSP_TEARDOWN;
+}
+
+static void
+gst_rtsp_auth_finalize (GObject * obj)
+{
+  GstRTSPAuth *auth = GST_RTSP_AUTH (obj);
+
+  GST_INFO ("finalize auth %p", auth);
+  g_free (auth->basic);
+
+  G_OBJECT_CLASS (gst_rtsp_auth_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_auth_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPAuth *auth = GST_RTSP_AUTH (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_auth_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPAuth *auth = GST_RTSP_AUTH (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_auth_new:
+ *
+ * Create a new #GstRTSPAuth instance.
+ *
+ * Returns: a new #GstRTSPAuth
+ */
+GstRTSPAuth *
+gst_rtsp_auth_new (void)
+{
+  GstRTSPAuth *result;
+
+  result = g_object_new (GST_TYPE_RTSP_AUTH, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_auth_set_basic:
+ * @auth: a #GstRTSPAuth
+ * @basic: the basic token
+ *
+ * Set the basic token for the default authentication algorithm.
+ */
+void
+gst_rtsp_auth_set_basic (GstRTSPAuth * auth, const gchar * basic)
+{
+  g_free (auth->basic);
+  auth->basic = g_strdup (basic);
+}
+
+static gboolean
+default_setup_auth (GstRTSPAuth * auth, GstRTSPClient * client,
+    GQuark hint, GstRTSPClientState * state)
+{
+  if (state->response == NULL)
+    return FALSE;
+
+  /* we only have Basic for now */
+  gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+      "Basic realm=\"GStreamer RTSP Server\"");
+
+  return TRUE;
+}
+
+/**
+ * gst_rtsp_auth_setup_auth:
+ * @auth: a #GstRTSPAuth
+ * @client: the client
+ * @uri: the requested uri
+ * @session: the session
+ * @request: the request
+ * @response: the response
+ *
+ * Add authentication tokens to @response.
+ *
+ * Returns: FALSE if something is wrong.
+ */
+gboolean
+gst_rtsp_auth_setup_auth (GstRTSPAuth * auth, GstRTSPClient * client,
+    GQuark hint, GstRTSPClientState * state)
+{
+  gboolean result = FALSE;
+  GstRTSPAuthClass *klass;
+
+  klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+  GST_DEBUG_OBJECT (auth, "setup auth");
+
+  if (klass->setup_auth)
+    result = klass->setup_auth (auth, client, hint, state);
+
+  return result;
+}
+
+static gboolean
+default_check_method (GstRTSPAuth * auth, GstRTSPClient * client,
+    GQuark hint, GstRTSPClientState * state)
+{
+  gboolean result = TRUE;
+  GstRTSPResult res;
+
+  if (state->method & auth->methods != 0) {
+    gchar *authorization;
+
+    result = FALSE;
+
+    res =
+        gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_AUTHORIZATION,
+        &authorization, 0);
+    if (res < 0)
+      goto no_auth;
+
+    /* parse type */
+    if (g_ascii_strncasecmp (authorization, "basic ", 6) == 0) {
+      GST_DEBUG_OBJECT (auth, "check Basic auth");
+      if (auth->basic && strcmp (&authorization[6], auth->basic) == 0)
+        result = TRUE;
+    } else if (g_ascii_strncasecmp (authorization, "digest ", 7) == 0) {
+      GST_DEBUG_OBJECT (auth, "check Digest auth");
+      /* not implemented yet */
+      result = FALSE;
+    }
+  }
+  return result;
+
+no_auth:
+  {
+    GST_DEBUG_OBJECT (auth, "no authorization header found");
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_auth_check_method:
+ * @auth: a #GstRTSPAuth
+ * @client: the client
+ * @hint: a hint
+ * @state: client state
+ *
+ * Check if @client is allowed to perform the actions of @state.
+ *
+ * Returns: FALSE if the action is not allowed.
+ */
+gboolean
+gst_rtsp_auth_check (GstRTSPAuth * auth, GstRTSPClient * client,
+    GQuark hint, GstRTSPClientState * state)
+{
+  gboolean result = FALSE;
+  GstRTSPAuthClass *klass;
+
+  klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+  GST_DEBUG_OBJECT (auth, "check state");
+
+  if (klass->check_method)
+    result = klass->check_method (auth, client, hint, state);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_auth_make_basic:
+ * @user: a userid
+ * @pass: a password
+ *
+ * Construct a Basic authorisation token from @user and @pass.
+ *
+ * Returns: the base64 encoding of the string @user:@pass. g_free()
+ *    after usage.
+ */
+gchar *
+gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass)
+{
+  gchar *user_pass;
+  gchar *result;
+
+  user_pass = g_strjoin (":", user, pass, NULL);
+  result = g_base64_encode ((guchar *) user_pass, strlen (user_pass));
+  g_free (user_pass);
+
+  return result;
+}

gst/rtsp-server/rtsp-auth.h

@@ -0,0 +1,80 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <gst/gst.h>
+
+#ifndef __GST_RTSP_AUTH_H__
+#define __GST_RTSP_AUTH_H__
+
+typedef struct _GstRTSPAuth GstRTSPAuth;
+typedef struct _GstRTSPAuthClass GstRTSPAuthClass;
+
+#include "rtsp-client.h"
+#include "rtsp-media-mapping.h"
+#include "rtsp-session-pool.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_AUTH              (gst_rtsp_auth_get_type ())
+#define GST_IS_RTSP_AUTH(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_AUTH))
+#define GST_IS_RTSP_AUTH_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_AUTH))
+#define GST_RTSP_AUTH_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuth))
+#define GST_RTSP_AUTH_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH_CAST(obj)         ((GstRTSPAuth*)(obj))
+#define GST_RTSP_AUTH_CLASS_CAST(klass) ((GstRTSPAuthClass*)(klass))
+
+/**
+ * GstRTSPAuth:
+ *
+ * The authentication structure.
+ */
+struct _GstRTSPAuth {
+  GObject       parent;
+
+  /*< private >*/
+  gchar        *basic;
+  GstRTSPMethod methods;
+};
+
+struct _GstRTSPAuthClass {
+  GObjectClass  parent_class;
+
+  gboolean (*setup_auth)   (GstRTSPAuth *auth, GstRTSPClient * client,
+                            GQuark hint, GstRTSPClientState *state);
+  gboolean (*check_method) (GstRTSPAuth *auth, GstRTSPClient * client,
+                            GQuark hint, GstRTSPClientState *state);
+};
+
+GType               gst_rtsp_auth_get_type          (void);
+
+GstRTSPAuth *       gst_rtsp_auth_new               (void);
+
+void                gst_rtsp_auth_set_basic         (GstRTSPAuth *auth, const gchar * basic);
+
+gboolean            gst_rtsp_auth_setup_auth        (GstRTSPAuth *auth, GstRTSPClient * client,
+                                                     GQuark hint, GstRTSPClientState *state);
+gboolean            gst_rtsp_auth_check_method      (GstRTSPAuth *auth, GstRTSPClient * client,
+                                                     GQuark hint, GstRTSPClientState *state);
+/* helpers */
+gchar *             gst_rtsp_auth_make_basic        (const gchar * user, const gchar * pass);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_AUTH_H__ */

gst/rtsp-server/rtsp-client.h

@@ -17,31 +17,23 @@
  * Boston, MA 02111-1307, USA.
  */
 
-#include <stdio.h>
-#include <stdlib.h>
-#include <unistd.h>
-#include <errno.h>
-#include <string.h>
-#include <sys/time.h>
-#include <sys/types.h>
-#include <netinet/in.h>
-#include <netdb.h>
-#include <sys/socket.h>
-#include <sys/wait.h>
-#include <fcntl.h>
-#include <arpa/inet.h>
-
 #include <gst/gst.h>
 #include <gst/rtsp/gstrtspconnection.h>
 
 #ifndef __GST_RTSP_CLIENT_H__
 #define __GST_RTSP_CLIENT_H__
 
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPClient GstRTSPClient;
+typedef struct _GstRTSPClientClass GstRTSPClientClass;
+typedef struct _GstRTSPClientState GstRTSPClientState;
+
+#include "rtsp-server.h"
 #include "rtsp-media.h"
 #include "rtsp-media-mapping.h"
 #include "rtsp-session-pool.h"
-
+#include "rtsp-auth.h"
-G_BEGIN_DECLS
 
 #define GST_TYPE_RTSP_CLIENT              (gst_rtsp_client_get_type ())
 #define GST_IS_RTSP_CLIENT(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT))
@@ -52,8 +44,29 @@ G_BEGIN_DECLS
 #define GST_RTSP_CLIENT_CAST(obj)         ((GstRTSPClient*)(obj))
 #define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
 
-typedef struct _GstRTSPClient GstRTSPClient;
+/**
-typedef struct _GstRTSPClientClass GstRTSPClientClass;
+ * GstRTSPClientState:
+ * @request: the complete request
+ * @uri: the complete url parsed from @request
+ * @method: the parsed method of @uri
+ * @session: the session, can be NULL
+ * @sessmedia: the session media for the url can be NULL
+ * @factory: the media factory for the url, can be NULL.
+ * @media: the session media for the url can be NULL
+ * @response: the response
+ *
+ * Information passed around containing the client state of a request.
+ */
+struct _GstRTSPClientState{
+  GstRTSPMessage      *request;
+  GstRTSPUrl          *uri;
+  GstRTSPMethod        method;
+  GstRTSPSession      *session;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia        *media;
+  GstRTSPMessage      *response;
+};
 
 /**
  * GstRTSPClient:
@@ -80,8 +93,10 @@ struct _GstRTSPClient {
   gchar             *server_ip;
   gboolean           is_ipv6;
 
+  GstRTSPServer        *server;
   GstRTSPSessionPool   *session_pool;
   GstRTSPMediaMapping  *media_mapping;
+  GstRTSPAuth          *auth;
 
   GstRTSPUrl     *uri;
   GstRTSPMedia   *media;
@@ -92,12 +107,18 @@ struct _GstRTSPClient {
 
 struct _GstRTSPClientClass {
   GObjectClass  parent_class;
+
+  /* signals */
+  void     (*closed)        (GstRTSPClient *client);
 };
 
 GType                 gst_rtsp_client_get_type          (void);
 
 GstRTSPClient *       gst_rtsp_client_new               (void);
 
+void                  gst_rtsp_client_set_server        (GstRTSPClient * client, GstRTSPServer * server);
+GstRTSPServer *       gst_rtsp_client_get_server        (GstRTSPClient * client);
+
 void                  gst_rtsp_client_set_session_pool  (GstRTSPClient *client,
                                                          GstRTSPSessionPool *pool);
 GstRTSPSessionPool *  gst_rtsp_client_get_session_pool  (GstRTSPClient *client);
@@ -106,6 +127,10 @@ void                  gst_rtsp_client_set_media_mapping (GstRTSPClient *client,
                                                          GstRTSPMediaMapping *mapping);
 GstRTSPMediaMapping * gst_rtsp_client_get_media_mapping (GstRTSPClient *client);
 
+void                  gst_rtsp_client_set_auth          (GstRTSPClient *client, GstRTSPAuth *auth);
+GstRTSPAuth *         gst_rtsp_client_get_auth          (GstRTSPClient *client);
+
+
 gboolean              gst_rtsp_client_accept            (GstRTSPClient *client,
                                                          GIOChannel *channel);
 

gst/rtsp-server/rtsp-funnel.c

@@ -0,0 +1,407 @@
+/*
+ * Farsight2 - Farsight Funnel element
+ *
+ * Copyright 2007 Collabora Ltd.
+ *  @author: Olivier Crete <olivier.crete@collabora.co.uk>
+ * Copyright 2007 Nokia Corp.
+ *
+ * rtsp-funnel.c: Simple Funnel element
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+ */
+
+/**
+ * SECTION:element-rtspfunnel
+ * @short_description: N-to-1 simple funnel
+ *
+ * Takes packets from various input sinks into one output source
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include "rtsp-funnel.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_funnel_debug);
+#define GST_CAT_DEFAULT rtsp_funnel_debug
+
+static const GstElementDetails rtsp_funnel_details =
+GST_ELEMENT_DETAILS ("Farsight Funnel pipe fitting",
+    "Generic",
+    "N-to-1 pipe fitting",
+    "Olivier Crete <olivier.crete@collabora.co.uk>");
+
+static GstStaticPadTemplate funnel_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink%d",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS_ANY);
+
+static GstStaticPadTemplate funnel_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS_ANY);
+
+
+static void
+_do_init (GType type)
+{
+  GST_DEBUG_CATEGORY_INIT (rtsp_funnel_debug, "rtspfunnel", 0,
+      "rtsp funnel element");
+}
+
+GST_BOILERPLATE_FULL (RTSPFunnel, rtsp_funnel, GstElement, GST_TYPE_ELEMENT,
+    _do_init);
+
+
+
+static GstStateChangeReturn rtsp_funnel_change_state (GstElement * element,
+    GstStateChange transition);
+
+static GstPad *rtsp_funnel_request_new_pad (GstElement * element,
+    GstPadTemplate * templ, const gchar * name);
+static void rtsp_funnel_release_pad (GstElement * element, GstPad * pad);
+
+static GstFlowReturn rtsp_funnel_chain (GstPad * pad, GstBuffer * buffer);
+static gboolean rtsp_funnel_event (GstPad * pad, GstEvent * event);
+static gboolean rtsp_funnel_src_event (GstPad * pad, GstEvent * event);
+static GstCaps *rtsp_funnel_getcaps (GstPad * pad);
+
+
+typedef struct
+{
+  GstSegment segment;
+} RTSPFunnelPadPrivate;
+
+static void
+rtsp_funnel_base_init (gpointer g_class)
+{
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_set_details (gstelement_class, &rtsp_funnel_details);
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&funnel_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&funnel_src_template));
+}
+
+
+static void
+rtsp_funnel_dispose (GObject * object)
+{
+  GList *item;
+
+restart:
+  for (item = GST_ELEMENT_PADS (object); item; item = g_list_next (item)) {
+    GstPad *pad = GST_PAD (item->data);
+
+    if (GST_PAD_IS_SINK (pad)) {
+      gst_element_release_request_pad (GST_ELEMENT (object), pad);
+      goto restart;
+    }
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+rtsp_funnel_class_init (RTSPFunnelClass * klass)
+{
+  GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+  GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+  gobject_class->dispose = GST_DEBUG_FUNCPTR (rtsp_funnel_dispose);
+
+  gstelement_class->request_new_pad =
+      GST_DEBUG_FUNCPTR (rtsp_funnel_request_new_pad);
+  gstelement_class->release_pad = GST_DEBUG_FUNCPTR (rtsp_funnel_release_pad);
+  gstelement_class->change_state = GST_DEBUG_FUNCPTR (rtsp_funnel_change_state);
+}
+
+static void
+rtsp_funnel_init (RTSPFunnel * funnel, RTSPFunnelClass * g_class)
+{
+  funnel->srcpad = gst_pad_new_from_static_template (&funnel_src_template,
+      "src");
+  gst_pad_set_event_function (funnel->srcpad, rtsp_funnel_src_event);
+  gst_pad_use_fixed_caps (funnel->srcpad);
+  gst_element_add_pad (GST_ELEMENT (funnel), funnel->srcpad);
+}
+
+
+static GstPad *
+rtsp_funnel_request_new_pad (GstElement * element, GstPadTemplate * templ,
+    const gchar * name)
+{
+  GstPad *sinkpad;
+  RTSPFunnelPadPrivate *priv = g_slice_alloc0 (sizeof (RTSPFunnelPadPrivate));
+
+  GST_DEBUG_OBJECT (element, "requesting pad");
+
+  sinkpad = gst_pad_new_from_template (templ, name);
+
+  gst_pad_set_chain_function (sinkpad, GST_DEBUG_FUNCPTR (rtsp_funnel_chain));
+  gst_pad_set_event_function (sinkpad, GST_DEBUG_FUNCPTR (rtsp_funnel_event));
+  gst_pad_set_getcaps_function (sinkpad,
+      GST_DEBUG_FUNCPTR (rtsp_funnel_getcaps));
+
+  gst_segment_init (&priv->segment, GST_FORMAT_UNDEFINED);
+  gst_pad_set_element_private (sinkpad, priv);
+
+  gst_pad_set_active (sinkpad, TRUE);
+
+  gst_element_add_pad (element, sinkpad);
+
+  return sinkpad;
+}
+
+static void
+rtsp_funnel_release_pad (GstElement * element, GstPad * pad)
+{
+  RTSPFunnel *funnel = RTSP_FUNNEL (element);
+  RTSPFunnelPadPrivate *priv = gst_pad_get_element_private (pad);
+
+  GST_DEBUG_OBJECT (funnel, "releasing pad");
+
+  gst_pad_set_active (pad, FALSE);
+
+  if (priv)
+    g_slice_free1 (sizeof (RTSPFunnelPadPrivate), priv);
+
+  gst_element_remove_pad (GST_ELEMENT_CAST (funnel), pad);
+}
+
+static GstCaps *
+rtsp_funnel_getcaps (GstPad * pad)
+{
+  RTSPFunnel *funnel = RTSP_FUNNEL (gst_pad_get_parent (pad));
+  GstCaps *caps;
+
+  caps = gst_pad_peer_get_caps (funnel->srcpad);
+  if (caps == NULL)
+    caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+
+  gst_object_unref (funnel);
+
+  return caps;
+}
+
+static GstFlowReturn
+rtsp_funnel_chain (GstPad * pad, GstBuffer * buffer)
+{
+  GstFlowReturn res;
+  RTSPFunnel *funnel = RTSP_FUNNEL (gst_pad_get_parent (pad));
+  RTSPFunnelPadPrivate *priv = gst_pad_get_element_private (pad);
+  GstEvent *event = NULL;
+  GstClockTime newts;
+  GstCaps *padcaps;
+
+  GST_DEBUG_OBJECT (funnel, "received buffer %p", buffer);
+
+  GST_OBJECT_LOCK (funnel);
+  if (priv->segment.format == GST_FORMAT_UNDEFINED) {
+    GST_WARNING_OBJECT (funnel, "Got buffer without segment,"
+        " setting segment [0,inf[");
+    gst_segment_set_newsegment_full (&priv->segment, FALSE, 1.0, 1.0,
+        GST_FORMAT_TIME, 0, -1, 0);
+  }
+
+  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buffer)))
+    gst_segment_set_last_stop (&priv->segment, priv->segment.format,
+        GST_BUFFER_TIMESTAMP (buffer));
+
+  newts = gst_segment_to_running_time (&priv->segment,
+      priv->segment.format, GST_BUFFER_TIMESTAMP (buffer));
+  if (newts != GST_BUFFER_TIMESTAMP (buffer)) {
+    buffer = gst_buffer_make_metadata_writable (buffer);
+    GST_BUFFER_TIMESTAMP (buffer) = newts;
+  }
+
+  if (!funnel->has_segment) {
+    event = gst_event_new_new_segment_full (FALSE, 1.0, 1.0, GST_FORMAT_TIME,
+        0, -1, 0);
+    funnel->has_segment = TRUE;
+  }
+  GST_OBJECT_UNLOCK (funnel);
+
+  if (event) {
+    if (!gst_pad_push_event (funnel->srcpad, event)) {
+      GST_WARNING_OBJECT (funnel, "Could not push out newsegment event");
+      res = GST_FLOW_ERROR;
+      goto out;
+    }
+  }
+
+
+  GST_OBJECT_LOCK (pad);
+  padcaps = GST_PAD_CAPS (funnel->srcpad);
+  GST_OBJECT_UNLOCK (pad);
+
+  if (GST_BUFFER_CAPS (buffer) && GST_BUFFER_CAPS (buffer) != padcaps) {
+    if (!gst_pad_set_caps (funnel->srcpad, GST_BUFFER_CAPS (buffer))) {
+      res = GST_FLOW_NOT_NEGOTIATED;
+      goto out;
+    }
+  }
+
+  res = gst_pad_push (funnel->srcpad, buffer);
+
+  GST_LOG_OBJECT (funnel, "handled buffer %s", gst_flow_get_name (res));
+
+out:
+  gst_object_unref (funnel);
+
+  return res;
+}
+
+static gboolean
+rtsp_funnel_event (GstPad * pad, GstEvent * event)
+{
+  RTSPFunnel *funnel = RTSP_FUNNEL (gst_pad_get_parent (pad));
+  RTSPFunnelPadPrivate *priv = gst_pad_get_element_private (pad);
+  gboolean forward = TRUE;
+  gboolean res = TRUE;
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_NEWSEGMENT:
+    {
+      gboolean update;
+      gdouble rate, arate;
+      GstFormat format;
+      gint64 start;
+      gint64 stop;
+      gint64 time;
+
+      gst_event_parse_new_segment_full (event, &update, &rate, &arate,
+          &format, &start, &stop, &time);
+
+
+      GST_OBJECT_LOCK (funnel);
+      gst_segment_set_newsegment_full (&priv->segment, update, rate, arate,
+          format, start, stop, time);
+      GST_OBJECT_UNLOCK (funnel);
+
+      forward = FALSE;
+      gst_event_unref (event);
+    }
+      break;
+    case GST_EVENT_FLUSH_STOP:
+    {
+      GST_OBJECT_LOCK (funnel);
+      gst_segment_init (&priv->segment, GST_FORMAT_UNDEFINED);
+      GST_OBJECT_UNLOCK (funnel);
+    }
+      break;
+    default:
+      break;
+  }
+
+
+  if (forward)
+    res = gst_pad_push_event (funnel->srcpad, event);
+
+  gst_object_unref (funnel);
+
+  return res;
+}
+
+static gboolean
+rtsp_funnel_src_event (GstPad * pad, GstEvent * event)
+{
+  GstElement *funnel;
+  GstIterator *iter;
+  GstPad *sinkpad;
+  gboolean result = FALSE;
+  gboolean done = FALSE;
+
+  funnel = gst_pad_get_parent_element (pad);
+  g_return_val_if_fail (funnel != NULL, FALSE);
+
+  iter = gst_element_iterate_sink_pads (funnel);
+
+  while (!done) {
+    switch (gst_iterator_next (iter, (gpointer) & sinkpad)) {
+      case GST_ITERATOR_OK:
+        gst_event_ref (event);
+        result |= gst_pad_push_event (sinkpad, event);
+        gst_object_unref (sinkpad);
+        break;
+      case GST_ITERATOR_RESYNC:
+        gst_iterator_resync (iter);
+        result = FALSE;
+        break;
+      case GST_ITERATOR_ERROR:
+        GST_WARNING_OBJECT (funnel, "Error iterating sinkpads");
+      case GST_ITERATOR_DONE:
+        done = TRUE;
+        break;
+    }
+  }
+  gst_iterator_free (iter);
+  gst_object_unref (funnel);
+  gst_event_unref (event);
+
+  return result;
+}
+
+static void
+reset_pad (gpointer data, gpointer user_data)
+{
+  GstPad *pad = data;
+  RTSPFunnelPadPrivate *priv = gst_pad_get_element_private (pad);
+
+  GST_OBJECT_LOCK (pad);
+  gst_segment_init (&priv->segment, GST_FORMAT_UNDEFINED);
+  GST_OBJECT_UNLOCK (pad);
+  gst_object_unref (pad);
+}
+
+static GstStateChangeReturn
+rtsp_funnel_change_state (GstElement * element, GstStateChange transition)
+{
+  RTSPFunnel *funnel = RTSP_FUNNEL (element);
+  GstStateChangeReturn ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+    {
+      GstIterator *iter = gst_element_iterate_sink_pads (element);
+      GstIteratorResult res;
+
+      do {
+        res = gst_iterator_foreach (iter, reset_pad, NULL);
+      } while (res == GST_ITERATOR_RESYNC);
+
+      gst_iterator_free (iter);
+
+      if (res == GST_ITERATOR_ERROR)
+        return GST_STATE_CHANGE_FAILURE;
+
+      GST_OBJECT_LOCK (funnel);
+      funnel->has_segment = FALSE;
+      GST_OBJECT_UNLOCK (funnel);
+    }
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  return ret;
+}

gst/rtsp-server/rtsp-funnel.h

@@ -0,0 +1,69 @@
+/*
+ * Farsight2 - Farsight Funnel element
+ *
+ * Copyright 2007 Collabora Ltd.
+ *  @author: Olivier Crete <olivier.crete@collabora.co.uk>
+ * Copyright 2007 Nokia Corp.
+ *
+ * rtsp-funnel.h: Simple Funnel element
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+ */
+
+
+#ifndef __RTSP_FUNNEL_H__
+#define __RTSP_FUNNEL_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define RTSP_TYPE_FUNNEL \
+  (rtsp_funnel_get_type ())
+#define RTSP_FUNNEL(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),RTSP_TYPE_FUNNEL,RTSPFunnel))
+#define RTSP_FUNNEL_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),RTSP_TYPE_FUNNEL,RTSPFunnelClass))
+#define RTSP_IS_FUNNEL(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),RTSP_TYPE_FUNNEL))
+#define RTSP_IS_FUNNEL_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),RTSP_TYPE_FUNNEL))
+
+typedef struct _RTSPFunnel          RTSPFunnel;
+typedef struct _RTSPFunnelClass     RTSPFunnelClass;
+
+/**
+ * RTSPFunnel:
+ *
+ * Opaque #RTSPFunnel data structure.
+ */
+struct _RTSPFunnel {
+  GstElement      element;
+
+  /*< private >*/
+  GstPad         *srcpad;
+
+  gboolean has_segment;
+};
+
+struct _RTSPFunnelClass {
+  GstElementClass parent_class;
+};
+
+GType   rtsp_funnel_get_type        (void);
+
+G_END_DECLS
+
+#endif /* __RTSP_FUNNEL_H__ */

gst/rtsp-server/rtsp-media-factory-uri.c

@@ -17,14 +17,18 @@
  * Boston, MA 02111-1307, USA.
  */
 
+#include <string.h>
+
 #include "rtsp-media-factory-uri.h"
 
 #define DEFAULT_URI         NULL
+#define DEFAULT_USE_GSTPAY  FALSE
 
 enum
 {
   PROP_0,
   PROP_URI,
+  PROP_USE_GSTPAY,
   PROP_LAST
 };
 
@@ -56,7 +60,7 @@ free_data (FactoryData * data)
 
 static const gchar *factory_key = "GstRTSPMediaFactoryURI";
 
-GST_DEBUG_CATEGORY (rtsp_media_factory_uri_debug);
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_factory_uri_debug);
 #define GST_CAT_DEFAULT rtsp_media_factory_uri_debug
 
 static void gst_rtsp_media_factory_uri_get_property (GObject * object,
@@ -93,6 +97,16 @@ gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
       g_param_spec_string ("uri", "URI",
           "The URI of the resource to stream", DEFAULT_URI,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPMediaFactoryURI::use-gstpay
+   *
+   * Allow the usage of gstpay in order to avoid decoding of compressed formats
+   * without a payloader.
+   */
+  g_object_class_install_property (gobject_class, PROP_USE_GSTPAY,
+      g_param_spec_boolean ("use-gstpay", "Use gstpay",
+          "Use the gstpay payloader to avoid decoding", DEFAULT_USE_GSTPAY,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
   mediafactory_class->get_element = rtsp_media_factory_uri_get_element;
 
@@ -100,13 +114,68 @@ gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
       0, "GstRTSPMediaFactoryUri");
 }
 
+typedef struct
+{
+  GList *demux;
+  GList *payload;
+  GList *decode;
+} FilterData;
+
+static gboolean
+payloader_filter (GstPluginFeature * feature, FilterData * data)
+{
+  gboolean res;
+  const gchar *klass;
+  GstElementFactory *fact;
+  GList **list = NULL;
+
+  /* we only care about element factories */
+  if (G_UNLIKELY (!GST_IS_ELEMENT_FACTORY (feature)))
+    return FALSE;
+
+  if (gst_plugin_feature_get_rank (feature) < GST_RANK_MARGINAL)
+    return FALSE;
+
+  fact = GST_ELEMENT_FACTORY_CAST (feature);
+
+  klass = gst_element_factory_get_klass (fact);
+
+  if (strstr (klass, "Decoder"))
+    list = &data->decode;
+  else if (strstr (klass, "Demux"))
+    list = &data->demux;
+  else if (strstr (klass, "Parser") && strstr (klass, "Codec"))
+    list = &data->demux;
+  else if (strstr (klass, "Payloader") && strstr (klass, "RTP"))
+    list = &data->payload;
+
+  if (list) {
+    GST_DEBUG ("adding %s", GST_PLUGIN_FEATURE_NAME (fact));
+    *list = g_list_prepend (*list, fact);
+  }
+
+  return FALSE;
+}
+
 static void
 gst_rtsp_media_factory_uri_init (GstRTSPMediaFactoryURI * factory)
 {
+  FilterData data = { NULL, NULL, NULL };
+
   factory->uri = g_strdup (DEFAULT_URI);
-  factory->factories =
+  factory->use_gstpay = DEFAULT_USE_GSTPAY;
-      gst_element_factory_list_get_elements (GST_ELEMENT_FACTORY_TYPE_PAYLOADER,
+
-      GST_RANK_NONE);
+  /* get the feature list using the filter */
+  gst_default_registry_feature_filter ((GstPluginFeatureFilter)
+      payloader_filter, FALSE, &data);
+  /* sort */
+  factory->demuxers =
+      g_list_sort (data.demux, gst_plugin_feature_rank_compare_func);
+  factory->payloaders =
+      g_list_sort (data.payload, gst_plugin_feature_rank_compare_func);
+  factory->decoders =
+      g_list_sort (data.decode, gst_plugin_feature_rank_compare_func);
+
   factory->raw_vcaps = gst_static_caps_get (&raw_video_caps);
   factory->raw_acaps = gst_static_caps_get (&raw_audio_caps);
 }
@@ -117,7 +186,9 @@ gst_rtsp_media_factory_uri_finalize (GObject * obj)
   GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (obj);
 
   g_free (factory->uri);
-  gst_plugin_feature_list_free (factory->factories);
+  gst_plugin_feature_list_free (factory->demuxers);
+  gst_plugin_feature_list_free (factory->payloaders);
+  gst_plugin_feature_list_free (factory->decoders);
   gst_caps_unref (factory->raw_vcaps);
   gst_caps_unref (factory->raw_acaps);
 
@@ -134,6 +205,9 @@ gst_rtsp_media_factory_uri_get_property (GObject * object, guint propid,
     case PROP_URI:
       g_value_take_string (value, gst_rtsp_media_factory_uri_get_uri (factory));
       break;
+    case PROP_USE_GSTPAY:
+      g_value_set_boolean (value, factory->use_gstpay);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -149,6 +223,9 @@ gst_rtsp_media_factory_uri_set_property (GObject * object, guint propid,
     case PROP_URI:
       gst_rtsp_media_factory_uri_set_uri (factory, g_value_get_string (value));
       break;
+    case PROP_USE_GSTPAY:
+      factory->use_gstpay = g_value_get_boolean (value);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -219,26 +296,43 @@ find_payloader (GstRTSPMediaFactoryURI * urifact, GstCaps * caps)
   GList *list, *tmp;
   GstElementFactory *factory = NULL;
 
-  /* find payloader that can link */
+  /* first find a demuxer that can link */
-  list = gst_element_factory_list_filter (urifact->factories, caps,
+  list = gst_element_factory_list_filter (urifact->demuxers, caps,
       GST_PAD_SINK, FALSE);
 
-  for (tmp = list; tmp; tmp = g_list_next (tmp)) {
+  if (list != NULL) {
-    GstElementFactory *f = GST_ELEMENT_FACTORY_CAST (tmp->data);
+    /* we have a demuxer, try that one first */
-    const gchar *name;
-
-    name = gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (f));
-    if (strcmp (name, "gdppay") == 0)
-      continue;
-
-    factory = f;
-    break;
-  }
-  if (factory)
-    g_object_ref (factory);
-
     gst_plugin_feature_list_free (list);
+    return NULL;
+  }
 
+  /* no demuxer try a depayloader */
+  list = gst_element_factory_list_filter (urifact->payloaders, caps,
+      GST_PAD_SINK, FALSE);
+
+  if (list == NULL) {
+    if (urifact->use_gstpay) {
+      /* no depayloader or parser/demuxer, use gstpay when allowed */
+      factory = gst_element_factory_find ("rtpgstpay");
+    } else {
+      /* no depayloader, try a decoder, we'll get to a payloader for a decoded
+       * video or audio format, worst case. */
+      list = gst_element_factory_list_filter (urifact->decoders, caps,
+          GST_PAD_SINK, FALSE);
+
+      if (list != NULL) {
+        /* we have a decoder, try that one first */
+        gst_plugin_feature_list_free (list);
+        return NULL;
+      }
+    }
+  }
+
+  if (list != NULL) {
+    factory = GST_ELEMENT_FACTORY_CAST (list->data);
+    g_object_ref (factory);
+    gst_plugin_feature_list_free (list);
+  }
   return factory;
 }
 
@@ -259,8 +353,9 @@ autoplug_continue_cb (GstElement * uribin, GstPad * pad, GstCaps * caps,
   if (!(factory = find_payloader (data->factory, caps)))
     goto no_factory;
 
-  /* we found a payloader, stop autoplugging */
+  /* we found a payloader, stop autoplugging so we can plug the
-  GST_DEBUG ("found payloader factory %s",
+   * payloader. */
+  GST_DEBUG ("found factory %s",
       gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
   gst_object_unref (factory);
 
@@ -349,6 +444,10 @@ pad_added_cb (GstElement * uribin, GstPad * pad, GstElement * element)
   g_object_set (payloader, "pt", data->pt, NULL);
   data->pt++;
 
+  if (g_object_class_find_property (G_OBJECT_GET_CLASS (payloader),
+          "buffer-list"))
+    g_object_set (payloader, "buffer-list", TRUE, NULL);
+
   /* add the payloader to the pipeline */
   gst_bin_add (GST_BIN_CAST (element), payloader);
   gst_element_set_state (payloader, GST_STATE_PLAYING);

gst/rtsp-server/rtsp-media-factory-uri.h

@@ -49,9 +49,13 @@ struct _GstRTSPMediaFactoryURI {
   GstRTSPMediaFactory   parent;
 
   gchar *uri;
+  gboolean use_gstpay;
+
   GstCaps *raw_vcaps;
   GstCaps *raw_acaps;
-  GList *factories;
+  GList *demuxers;
+  GList *payloaders;
+  GList *decoders;
 };
 
 /**

gst/rtsp-server/rtsp-media-factory.c

@@ -22,6 +22,7 @@
 #define DEFAULT_LAUNCH         NULL
 #define DEFAULT_SHARED         FALSE
 #define DEFAULT_EOS_SHUTDOWN   FALSE
+#define DEFAULT_BUFFER_SIZE    0x80000
 
 enum
 {
@@ -29,10 +30,11 @@ enum
   PROP_LAUNCH,
   PROP_SHARED,
   PROP_EOS_SHUTDOWN,
+  PROP_BUFFER_SIZE,
   PROP_LAST
 };
 
-GST_DEBUG_CATEGORY (rtsp_media_debug);
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
 #define GST_CAT_DEFAULT rtsp_media_debug
 
 static void gst_rtsp_media_factory_get_property (GObject * object, guint propid,
@@ -96,13 +98,19 @@ gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
           "Send EOS down the pipeline before shutting down",
           DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+      g_param_spec_uint ("buffer-size", "Buffer Size",
+          "The kernel UDP buffer size to use", 0, G_MAXUINT,
+          DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
   klass->gen_key = default_gen_key;
   klass->get_element = default_get_element;
   klass->construct = default_construct;
   klass->configure = default_configure;
   klass->create_pipeline = default_create_pipeline;
 
-  GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmediafactory", 0,
+      "GstRTSPMediaFactory");
 }
 
 static void
@@ -111,6 +119,7 @@ gst_rtsp_media_factory_init (GstRTSPMediaFactory * factory)
   factory->launch = g_strdup (DEFAULT_LAUNCH);
   factory->shared = DEFAULT_SHARED;
   factory->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+  factory->buffer_size = DEFAULT_BUFFER_SIZE;
 
   factory->lock = g_mutex_new ();
   factory->medias_lock = g_mutex_new ();
@@ -127,6 +136,8 @@ gst_rtsp_media_factory_finalize (GObject * obj)
   g_mutex_free (factory->medias_lock);
   g_free (factory->launch);
   g_mutex_free (factory->lock);
+  if (factory->auth)
+    g_object_unref (factory->auth);
 
   G_OBJECT_CLASS (gst_rtsp_media_factory_parent_class)->finalize (obj);
 }
@@ -148,6 +159,10 @@ gst_rtsp_media_factory_get_property (GObject * object, guint propid,
       g_value_set_boolean (value,
           gst_rtsp_media_factory_is_eos_shutdown (factory));
       break;
+    case PROP_BUFFER_SIZE:
+      g_value_set_uint (value,
+          gst_rtsp_media_factory_get_buffer_size (factory));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -170,6 +185,10 @@ gst_rtsp_media_factory_set_property (GObject * object, guint propid,
       gst_rtsp_media_factory_set_eos_shutdown (factory,
           g_value_get_boolean (value));
       break;
+    case PROP_BUFFER_SIZE:
+      gst_rtsp_media_factory_set_buffer_size (factory,
+          g_value_get_uint (value));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -326,6 +345,94 @@ gst_rtsp_media_factory_is_eos_shutdown (GstRTSPMediaFactory * factory)
   return result;
 }
 
+/**
+ * gst_rtsp_media_factory_set_buffer_size:
+ * @factory: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_factory_set_buffer_size (GstRTSPMediaFactory * factory,
+    guint size)
+{
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  factory->buffer_size = size;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_buffer_size:
+ * @factory: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory)
+{
+  guint result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  result = factory->buffer_size;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_auth:
+ * @factory: a #GstRTSPMediaFactory
+ * @auth: a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @factory.
+ */
+void
+gst_rtsp_media_factory_set_auth (GstRTSPMediaFactory * factory,
+    GstRTSPAuth * auth)
+{
+  GstRTSPAuth *old;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  old = factory->auth;
+
+  if (old != auth) {
+    if (auth)
+      g_object_ref (auth);
+    factory->auth = auth;
+    if (old)
+      g_object_unref (old);
+  }
+}
+
+/**
+ * gst_rtsp_media_factory_get_auth:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @factory.
+ *
+ * Returns: the #GstRTSPAuth of @factory. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_media_factory_get_auth (GstRTSPMediaFactory * factory)
+{
+  GstRTSPAuth *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+  if ((result = factory->auth))
+    g_object_ref (result);
+
+  return result;
+}
+
 static gboolean
 compare_media (gpointer key, GstRTSPMedia * media1, GstRTSPMedia * media2)
 {
@@ -608,13 +715,22 @@ static void
 default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
 {
   gboolean shared, eos_shutdown;
+  guint size;
+  GstRTSPAuth *auth;
 
   /* configure the sharedness */
   GST_RTSP_MEDIA_FACTORY_LOCK (factory);
   shared = factory->shared;
   eos_shutdown = factory->eos_shutdown;
+  size = factory->buffer_size;
   GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
 
   gst_rtsp_media_set_shared (media, shared);
   gst_rtsp_media_set_eos_shutdown (media, eos_shutdown);
+  gst_rtsp_media_set_buffer_size (media, size);
+
+  if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
+    gst_rtsp_media_set_auth (media, auth);
+    g_object_unref (auth);
+  }
 }

gst/rtsp-server/rtsp-media-factory.h

@@ -21,6 +21,7 @@
 #include <gst/rtsp/gstrtspurl.h>
 
 #include "rtsp-media.h"
+#include "rtsp-auth.h"
 
 #ifndef __GST_RTSP_MEDIA_FACTORY_H__
 #define __GST_RTSP_MEDIA_FACTORY_H__
@@ -62,6 +63,8 @@ struct _GstRTSPMediaFactory {
   gchar        *launch;
   gboolean      shared;
   gboolean      eos_shutdown;
+  GstRTSPAuth  *auth;
+  guint         buffer_size;
 
   GMutex       *medias_lock;
   GHashTable   *medias;
@@ -116,6 +119,13 @@ void                  gst_rtsp_media_factory_set_eos_shutdown   (GstRTSPMediaFac
                                                                  gboolean eos_shutdown);
 gboolean              gst_rtsp_media_factory_is_eos_shutdown    (GstRTSPMediaFactory *factory);
 
+void                  gst_rtsp_media_factory_set_auth     (GstRTSPMediaFactory *factory, GstRTSPAuth *auth);
+GstRTSPAuth *         gst_rtsp_media_factory_get_auth     (GstRTSPMediaFactory *factory);
+
+void                  gst_rtsp_media_factory_set_buffer_size    (GstRTSPMediaFactory * factory, guint size);
+guint                 gst_rtsp_media_factory_get_buffer_size    (GstRTSPMediaFactory * factory);
+
+
 /* creating the media from the factory and a url */
 GstRTSPMedia *        gst_rtsp_media_factory_construct    (GstRTSPMediaFactory *factory,
                                                            const GstRTSPUrl *url);

gst/rtsp-server/rtsp-media-mapping.c

@@ -21,7 +21,7 @@
 
 G_DEFINE_TYPE (GstRTSPMediaMapping, gst_rtsp_media_mapping, G_TYPE_OBJECT);
 
-GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug);
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
 #define GST_CAT_DEFAULT rtsp_media_debug
 
 static void gst_rtsp_media_mapping_finalize (GObject * obj);
@@ -39,11 +39,16 @@ gst_rtsp_media_mapping_class_init (GstRTSPMediaMappingClass * klass)
   gobject_class->finalize = gst_rtsp_media_mapping_finalize;
 
   klass->find_media = find_media;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmediamapping", 0,
+      "GstRTSPMediaMapping");
 }
 
 static void
 gst_rtsp_media_mapping_init (GstRTSPMediaMapping * mapping)
 {
+  GST_DEBUG_OBJECT (mapping, "created");
+
   mapping->mappings = g_hash_table_new_full (g_str_hash, g_str_equal,
       g_free, g_object_unref);
 }
@@ -53,6 +58,8 @@ gst_rtsp_media_mapping_finalize (GObject * obj)
 {
   GstRTSPMediaMapping *mapping = GST_RTSP_MEDIA_MAPPING (obj);
 
+  GST_DEBUG_OBJECT (mapping, "finalized");
+
   g_hash_table_unref (mapping->mappings);
 
   G_OBJECT_CLASS (gst_rtsp_media_mapping_parent_class)->finalize (obj);

gst/rtsp-server/rtsp-media.c

@@ -23,6 +23,7 @@
 #include <gst/app/gstappsrc.h>
 #include <gst/app/gstappsink.h>
 
+#include "rtsp-funnel.h"
 #include "rtsp-media.h"
 
 #define DEFAULT_SHARED         FALSE
@@ -30,6 +31,7 @@
 #define DEFAULT_PROTOCOLS      GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
 //#define DEFAULT_PROTOCOLS      GST_RTSP_LOWER_TRANS_UDP_MCAST
 #define DEFAULT_EOS_SHUTDOWN   FALSE
+#define DEFAULT_BUFFER_SIZE    0x80000
 
 /* define to dump received RTCP packets */
 #undef DUMP_STATS
@@ -41,16 +43,19 @@ enum
   PROP_REUSABLE,
   PROP_PROTOCOLS,
   PROP_EOS_SHUTDOWN,
+  PROP_BUFFER_SIZE,
   PROP_LAST
 };
 
 enum
 {
+  SIGNAL_PREPARED,
   SIGNAL_UNPREPARED,
+  SIGNAL_NEW_STATE,
   SIGNAL_LAST
 };
 
-GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug);
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
 #define GST_CAT_DEFAULT rtsp_media_debug
 
 static GQuark ssrc_stream_map_key;
@@ -103,14 +108,31 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
           "Send an EOS event to the pipeline before unpreparing",
           DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+      g_param_spec_uint ("buffer-size", "Buffer Size",
+          "The kernel UDP buffer size to use", 0, G_MAXUINT,
+          DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_rtsp_media_signals[SIGNAL_PREPARED] =
+      g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
+      g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+
   gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
       g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
       G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
       g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
 
+  gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
+      g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
+      g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
+
   klass->context = g_main_context_new ();
   klass->loop = g_main_loop_new (klass->context, TRUE);
 
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
+
   klass->thread = g_thread_create ((GThreadFunc) do_loop, klass, TRUE, &error);
   if (error != NULL) {
     g_critical ("could not start bus thread: %s", error->message);
@@ -119,6 +141,9 @@ gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
   klass->unprepare = default_unprepare;
 
   ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
+
+  gst_element_register (NULL, "rtspfunnel", GST_RANK_NONE, RTSP_TYPE_FUNNEL);
+
 }
 
 static void
@@ -132,45 +157,7 @@ gst_rtsp_media_init (GstRTSPMedia * media)
   media->reusable = DEFAULT_REUSABLE;
   media->protocols = DEFAULT_PROTOCOLS;
   media->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
-}
+  media->buffer_size = DEFAULT_BUFFER_SIZE;
-
-/* FIXME. this should be done in multiudpsink */
-typedef struct
-{
-  gint count;
-  gchar *dest;
-  gint min, max;
-} RTSPDestination;
-
-static gint
-dest_compare (RTSPDestination * a, RTSPDestination * b)
-{
-  if ((a->min == b->min) && (a->max == b->max)
-      && (strcmp (a->dest, b->dest) == 0))
-    return 0;
-
-  return 1;
-}
-
-static RTSPDestination *
-create_destination (const gchar * dest, gint min, gint max)
-{
-  RTSPDestination *res;
-
-  res = g_slice_new (RTSPDestination);
-  res->count = 1;
-  res->dest = g_strdup (dest);
-  res->min = min;
-  res->max = max;
-
-  return res;
-}
-
-static void
-free_destination (RTSPDestination * dest)
-{
-  g_free (dest->dest);
-  g_slice_free (RTSPDestination, dest);
 }
 
 void
@@ -208,9 +195,6 @@ gst_rtsp_media_stream_free (GstRTSPMediaStream * stream)
 
   g_list_free (stream->transports);
 
-  g_list_foreach (stream->destinations, (GFunc) free_destination, NULL);
-  g_list_free (stream->destinations);
-
   g_free (stream);
 }
 
@@ -271,6 +255,9 @@ gst_rtsp_media_get_property (GObject * object, guint propid,
     case PROP_EOS_SHUTDOWN:
       g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
       break;
+    case PROP_BUFFER_SIZE:
+      g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -295,6 +282,9 @@ gst_rtsp_media_set_property (GObject * object, guint propid,
     case PROP_EOS_SHUTDOWN:
       gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
       break;
+    case PROP_BUFFER_SIZE:
+      gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
   }
@@ -341,7 +331,7 @@ collect_media_stats (GstRTSPMedia * media)
     GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
         GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
 
-    if (position == -1 || media->active > 0) {
+    if (position == -1) {
       media->range.min.type = GST_RTSP_TIME_NOW;
       media->range.min.seconds = -1;
     } else {
@@ -507,6 +497,85 @@ gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
   return media->eos_shutdown;
 }
 
+/**
+ * gst_rtsp_media_set_buffer_size:
+ * @media: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
+{
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  media->buffer_size = size;
+}
+
+/**
+ * gst_rtsp_media_get_buffer_size:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
+{
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  return media->buffer_size;
+}
+
+/**
+ * gst_rtsp_media_set_auth:
+ * @media: a #GstRTSPMedia
+ * @auth: a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @media.
+ */
+void
+gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
+{
+  GstRTSPAuth *old;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  old = media->auth;
+
+  if (old != auth) {
+    if (auth)
+      g_object_ref (auth);
+    media->auth = auth;
+    if (old)
+      g_object_unref (old);
+  }
+}
+
+/**
+ * gst_rtsp_media_get_auth:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @media.
+ *
+ * Returns: the #GstRTSPAuth of @media. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_media_get_auth (GstRTSPMedia * media)
+{
+  GstRTSPAuth *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+  if ((result = media->auth))
+    g_object_ref (result);
+
+  return result;
+}
+
+
 /**
  * gst_rtsp_media_n_streams:
  * @media: a #GstRTSPMedia
@@ -548,6 +617,34 @@ gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
   return res;
 }
 
+/**
+ * gst_rtsp_media_get_range_string:
+ * @media: a #GstRTSPMedia
+ * @play: for the PLAY request
+ *
+ * Get the current range as a string.
+ *
+ * Returns: The range as a string, g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play)
+{
+  gchar *result;
+  GstRTSPTimeRange range;
+
+  /* make copy */
+  range = media->range;
+
+  if (!play && media->active > 0) {
+    range.min.type = GST_RTSP_TIME_NOW;
+    range.min.seconds = -1;
+  }
+
+  result = gst_rtsp_range_to_string (&range);
+
+  return result;
+}
+
 /**
  * gst_rtsp_media_seek:
  * @media: a #GstRTSPMedia
@@ -804,10 +901,16 @@ again:
           "send-duplicates")) {
     g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
     g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
-    stream->filter_duplicates = FALSE;
   } else {
-    GST_WARNING ("multiudpsink version found without send-duplicates property");
+    g_warning
-    stream->filter_duplicates = TRUE;
+        ("old multiudpsink version found without send-duplicates property");
+  }
+
+  if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
+          "buffer-size")) {
+    g_object_set (G_OBJECT (udpsink0), "buffer-size", media->buffer_size, NULL);
+  } else {
+    GST_WARNING ("multiudpsink version found without buffer-size property");
   }
 
   g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL);
@@ -1068,10 +1171,42 @@ handle_new_buffer (GstAppSink * sink, gpointer user_data)
   return GST_FLOW_OK;
 }
 
+static GstFlowReturn
+handle_new_buffer_list (GstAppSink * sink, gpointer user_data)
+{
+  GList *walk;
+  GstBufferList *blist;
+  GstRTSPMediaStream *stream;
+
+  blist = gst_app_sink_pull_buffer_list (sink);
+  if (!blist)
+    return GST_FLOW_OK;
+
+  stream = (GstRTSPMediaStream *) user_data;
+
+  for (walk = stream->transports; walk; walk = g_list_next (walk)) {
+    GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
+
+    if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
+      if (tr->send_rtp_list)
+        tr->send_rtp_list (blist, tr->transport->interleaved.min,
+            tr->user_data);
+    } else {
+      if (tr->send_rtcp_list)
+        tr->send_rtcp_list (blist, tr->transport->interleaved.max,
+            tr->user_data);
+    }
+  }
+  gst_buffer_list_unref (blist);
+
+  return GST_FLOW_OK;
+}
+
 static GstAppSinkCallbacks sink_cb = {
   NULL,                         /* not interested in EOS */
   NULL,                         /* not interested in preroll buffers */
-  handle_new_buffer
+  handle_new_buffer,
+  handle_new_buffer_list
 };
 
 /* prepare the pipeline objects to handle @stream in @media */
@@ -1190,8 +1325,7 @@ setup_stream (GstRTSPMediaStream * stream, guint idx, GstRTSPMedia * media)
   gst_object_unref (teepad);
 
   /* make selector for the RTP receivers */
-  stream->selector[0] = gst_element_factory_make ("input-selector", NULL);
+  stream->selector[0] = gst_element_factory_make ("rtspfunnel", NULL);
-  g_object_set (stream->selector[0], "select-all", TRUE, NULL);
   gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[0]);
 
   pad = gst_element_get_static_pad (stream->selector[0], "src");
@@ -1211,8 +1345,7 @@ setup_stream (GstRTSPMediaStream * stream, guint idx, GstRTSPMedia * media)
   gst_object_unref (selpad);
 
   /* make selector for the RTCP receivers */
-  stream->selector[1] = gst_element_factory_make ("input-selector", NULL);
+  stream->selector[1] = gst_element_factory_make ("rtspfunnel", NULL);
-  g_object_set (stream->selector[1], "select-all", TRUE, NULL);
   gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[1]);
 
   pad = gst_element_get_static_pad (stream->selector[1], "src");
@@ -1594,6 +1727,8 @@ gst_rtsp_media_prepare (GstRTSPMedia * media)
   if (status == GST_RTSP_MEDIA_STATUS_ERROR)
     goto state_failed;
 
+  g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
+
   GST_INFO ("object %p is prerolled", media);
 
   return TRUE;
@@ -1686,83 +1821,18 @@ static void
 add_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
     gchar * dest, gint min, gint max)
 {
-  gboolean do_add = TRUE;
-  RTSPDestination *ndest;
-
-  if (stream->filter_duplicates) {
-    RTSPDestination fdest;
-    GList *find;
-
-    fdest.dest = dest;
-    fdest.min = min;
-    fdest.max = max;
-
-    /* first see if we already added this destination */
-    find =
-        g_list_find_custom (stream->destinations, &fdest,
-        (GCompareFunc) dest_compare);
-    if (find) {
-      ndest = (RTSPDestination *) find->data;
-
-      GST_INFO ("already streaming to %s:%d-%d with %d clients", dest, min, max,
-          ndest->count);
-      ndest->count++;
-      do_add = FALSE;
-    }
-  }
-
-  if (do_add) {
   GST_INFO ("adding %s:%d-%d", dest, min, max);
   g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
   g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
-
-    if (stream->filter_duplicates) {
-      ndest = create_destination (dest, min, max);
-      stream->destinations = g_list_prepend (stream->destinations, ndest);
-    }
-  }
 }
 
 static void
 remove_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
     gchar * dest, gint min, gint max)
 {
-  gboolean do_remove = TRUE;
-  RTSPDestination *ndest = NULL;
-  GList *find = NULL;
-
-  if (stream->filter_duplicates) {
-    RTSPDestination fdest;
-
-    fdest.dest = dest;
-    fdest.min = min;
-    fdest.max = max;
-
-    /* first see if we already added this destination */
-    find =
-        g_list_find_custom (stream->destinations, &fdest,
-        (GCompareFunc) dest_compare);
-    if (!find)
-      return;
-
-    ndest = (RTSPDestination *) find->data;
-    if (--ndest->count > 0) {
-      do_remove = FALSE;
-      GST_INFO ("still streaming to %s:%d-%d with %d clients", dest, min, max,
-          ndest->count);
-    }
-  }
-
-  if (do_remove) {
   GST_INFO ("removing %s:%d-%d", dest, min, max);
   g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
   g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
-
-    if (stream->filter_duplicates) {
-      stream->destinations = g_list_delete_link (stream->destinations, find);
-      free_destination (ndest);
-    }
-  }
 }
 
 /**
@@ -1888,17 +1958,23 @@ gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
   else
     do_state = FALSE;
 
-  GST_INFO ("active %d media %p", media->active, media);
+  GST_INFO ("state %d active %d media %p do_state %d", state, media->active,
+      media, do_state);
 
-  if (do_state && media->target_state != state) {
+  if (media->target_state != state) {
+    if (do_state) {
       if (state == GST_STATE_NULL) {
         gst_rtsp_media_unprepare (media);
       } else {
-      GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
+        GST_INFO ("state %s media %p", gst_element_state_get_name (state),
+            media);
         media->target_state = state;
         ret = gst_element_set_state (media->pipeline, state);
       }
     }
+    g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
+        NULL);
+  }
 
   /* remember where we are */
   if (state == GST_STATE_PAUSED || old_active != media->active)

gst/rtsp-server/rtsp-media.h

@@ -42,6 +42,7 @@ typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
 typedef struct _GstRTSPMediaTrans GstRTSPMediaTrans;
 
 typedef gboolean (*GstRTSPSendFunc)      (GstBuffer *buffer, guint8 channel, gpointer user_data);
+typedef gboolean (*GstRTSPSendListFunc)  (GstBufferList *blist, guint8 channel, gpointer user_data);
 typedef void     (*GstRTSPKeepAliveFunc) (gpointer user_data);
 
 /**
@@ -49,6 +50,8 @@ typedef void     (*GstRTSPKeepAliveFunc) (gpointer user_data);
  * @idx: a stream index
  * @send_rtp: callback for sending RTP messages
  * @send_rtcp: callback for sending RTCP messages
+ * @send_rtp_list: callback for sending RTP messages
+ * @send_rtcp_list: callback for sending RTCP messages
  * @user_data: user data passed in the callbacks
  * @notify: free function for the user_data.
  * @keep_alive: keep alive callback
@@ -66,6 +69,8 @@ struct _GstRTSPMediaTrans {
 
   GstRTSPSendFunc      send_rtp;
   GstRTSPSendFunc      send_rtcp;
+  GstRTSPSendListFunc  send_rtp_list;
+  GstRTSPSendListFunc  send_rtcp_list;
   gpointer             user_data;
   GDestroyNotify       notify;
 
@@ -80,16 +85,17 @@ struct _GstRTSPMediaTrans {
   GObject             *rtpsource;
 };
 
+#include "rtsp-auth.h"
+
 /**
  * GstRTSPMediaStream:
  * @srcpad: the srcpad of the stream
  * @payloader: the payloader of the format
  * @prepared: if the stream is prepared for streaming
- * @server_port: the server udp ports
  * @recv_rtp_sink: sinkpad for RTP buffers
  * @recv_rtcp_sink: sinkpad for RTCP buffers
- * @recv_rtp_src: srcpad for RTP buffers
+ * @send_rtp_src: srcpad for RTP buffers
- * @recv_rtcp_src: srcpad for RTCP buffers
+ * @send_rtcp_src: srcpad for RTCP buffers
  * @udpsrc: the udp source elements for RTP/RTCP
  * @udpsink: the udp sink elements for RTP/RTCP
  * @appsrc: the app source elements for RTP/RTCP
@@ -136,11 +142,6 @@ struct _GstRTSPMediaStream {
 
   /* transports we stream to */
   GList        *transports;
-
-  /* to filter out duplicate destinations in case multiudpsink is too old to do
-   * this for us */
-  gboolean      filter_duplicates;
-  GList        *destinations;
 };
 
 /**
@@ -200,6 +201,8 @@ struct _GstRTSPMedia {
   gboolean           reused;
   gboolean           is_ipv6;
   gboolean           eos_shutdown;
+  guint              buffer_size;
+  GstRTSPAuth       *auth;
 
   GstElement        *element;
   GArray            *streams;
@@ -250,7 +253,10 @@ struct _GstRTSPMediaClass {
   gboolean     (*unprepare)       (GstRTSPMedia *media);
 
   /* signals */
+  gboolean     (*prepared)        (GstRTSPMedia *media);
   gboolean     (*unprepared)      (GstRTSPMedia *media);
+
+  gboolean     (*new_state)       (GstRTSPMedia *media, GstState state);
 };
 
 GType                 gst_rtsp_media_get_type         (void);
@@ -270,6 +276,12 @@ GstRTSPLowerTrans     gst_rtsp_media_get_protocols    (GstRTSPMedia *media);
 void                  gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
 gboolean              gst_rtsp_media_is_eos_shutdown  (GstRTSPMedia *media);
 
+void                  gst_rtsp_media_set_auth         (GstRTSPMedia *media, GstRTSPAuth *auth);
+GstRTSPAuth *         gst_rtsp_media_get_auth         (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_buffer_size  (GstRTSPMedia *media, guint size);
+guint                 gst_rtsp_media_get_buffer_size  (GstRTSPMedia *media);
+
 /* prepare the media for playback */
 gboolean              gst_rtsp_media_prepare          (GstRTSPMedia *media);
 gboolean              gst_rtsp_media_is_prepared      (GstRTSPMedia *media);
@@ -280,6 +292,7 @@ guint                 gst_rtsp_media_n_streams        (GstRTSPMedia *media);
 GstRTSPMediaStream *  gst_rtsp_media_get_stream       (GstRTSPMedia *media, guint idx);
 
 gboolean              gst_rtsp_media_seek             (GstRTSPMedia *media, GstRTSPTimeRange *range);
+gchar *               gst_rtsp_media_get_range_string (GstRTSPMedia *media, gboolean play);
 
 GstFlowReturn         gst_rtsp_media_stream_rtp       (GstRTSPMediaStream *stream, GstBuffer *buffer);
 GstFlowReturn         gst_rtsp_media_stream_rtcp      (GstRTSPMediaStream *stream, GstBuffer *buffer);

gst/rtsp-server/rtsp-params.c

@@ -21,9 +21,7 @@
 #include "rtsp-params.h"
 
 GstRTSPResult
-gst_rtsp_params_set (GstRTSPClient * client, GstRTSPUrl * uri,
+gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
-    GstRTSPSession * session, GstRTSPMessage * request,
-    GstRTSPMessage * response)
 {
   GstRTSPStatusCode code;
 
@@ -31,16 +29,14 @@ gst_rtsp_params_set (GstRTSPClient * client, GstRTSPUrl * uri,
    * with a list of the parameters */
   code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
 
-  gst_rtsp_message_init_response (response, code,
+  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), request);
+      gst_rtsp_status_as_text (code), state->request);
 
   return GST_RTSP_OK;
 }
 
 GstRTSPResult
-gst_rtsp_params_get (GstRTSPClient * client, GstRTSPUrl * uri,
+gst_rtsp_params_get (GstRTSPClient * client, GstRTSPClientState * state)
-    GstRTSPSession * session, GstRTSPMessage * request,
-    GstRTSPMessage * response)
 {
   GstRTSPStatusCode code;
 
@@ -48,8 +44,8 @@ gst_rtsp_params_get (GstRTSPClient * client, GstRTSPUrl * uri,
    * with a list of the parameters */
   code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
 
-  gst_rtsp_message_init_response (response, code,
+  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), request);
+      gst_rtsp_status_as_text (code), state->request);
 
   return GST_RTSP_OK;
 }

gst/rtsp-server/rtsp-params.h

@@ -30,13 +30,8 @@
 
 G_BEGIN_DECLS
 
-GstRTSPResult    gst_rtsp_params_set      (GstRTSPClient * client, GstRTSPUrl * uri,
+GstRTSPResult    gst_rtsp_params_set      (GstRTSPClient * client, GstRTSPClientState * state);
-                                           GstRTSPSession * session, GstRTSPMessage * request,
+GstRTSPResult    gst_rtsp_params_get      (GstRTSPClient * client, GstRTSPClientState * state);
-                                           GstRTSPMessage * response);
-
-GstRTSPResult    gst_rtsp_params_get      (GstRTSPClient * client, GstRTSPUrl * uri,
-                                           GstRTSPSession * session, GstRTSPMessage * request,
-                                           GstRTSPMessage * response);
 
 G_END_DECLS
 

gst/rtsp-server/rtsp-sdp.c

@@ -43,7 +43,7 @@ gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
 
   n_streams = gst_rtsp_media_n_streams (media);
 
-  rangestr = gst_rtsp_range_to_string (&media->range);
+  rangestr = gst_rtsp_media_get_range_string (media, FALSE);
   gst_sdp_message_add_attribute (sdp, "range", rangestr);
   g_free (rangestr);
 

gst/rtsp-server/rtsp-server.c

@@ -17,6 +17,19 @@
  * Boston, MA 02111-1307, USA.
  */
 
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+#include <sys/time.h>
+#include <sys/types.h>
+#include <netinet/in.h>
+#include <netdb.h>
+#include <sys/socket.h>
+#include <sys/wait.h>
+#include <fcntl.h>
+#include <arpa/inet.h>
 #include <sys/ioctl.h>
 
 #include "rtsp-server.h"
@@ -55,8 +68,9 @@ static void gst_rtsp_server_set_property (GObject * object, guint propid,
     const GValue * value, GParamSpec * pspec);
 static void gst_rtsp_server_finalize (GObject * object);
 
-static GstRTSPClient *default_accept_client (GstRTSPServer * server,
+static GstRTSPClient *default_create_client (GstRTSPServer * server);
-    GIOChannel * channel);
+static gboolean default_accept_client (GstRTSPServer * server,
+    GstRTSPClient * client, GIOChannel * channel);
 
 static void
 gst_rtsp_server_class_init (GstRTSPServerClass * klass)
@@ -127,6 +141,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
           GST_TYPE_RTSP_MEDIA_MAPPING,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
 
+  klass->create_client = default_create_client;
   klass->accept_client = default_accept_client;
 
   GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
@@ -135,6 +150,7 @@ gst_rtsp_server_class_init (GstRTSPServerClass * klass)
 static void
 gst_rtsp_server_init (GstRTSPServer * server)
 {
+  server->lock = g_mutex_new ();
   server->address = g_strdup (DEFAULT_ADDRESS);
   server->service = g_strdup (DEFAULT_SERVICE);
   server->backlog = DEFAULT_BACKLOG;
@@ -147,11 +163,20 @@ gst_rtsp_server_finalize (GObject * object)
 {
   GstRTSPServer *server = GST_RTSP_SERVER (object);
 
+  GST_DEBUG_OBJECT (server, "finalize server");
+
   g_free (server->address);
   g_free (server->service);
 
   g_object_unref (server->session_pool);
   g_object_unref (server->media_mapping);
+
+  if (server->auth)
+    g_object_unref (server->auth);
+
+  g_mutex_free (server->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
 }
 
 /**
@@ -184,8 +209,10 @@ gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
   g_return_if_fail (GST_IS_RTSP_SERVER (server));
   g_return_if_fail (address != NULL);
 
+  GST_RTSP_SERVER_LOCK (server);
   g_free (server->address);
   server->address = g_strdup (address);
+  GST_RTSP_SERVER_UNLOCK (server);
 }
 
 /**
@@ -199,9 +226,14 @@ gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
 gchar *
 gst_rtsp_server_get_address (GstRTSPServer * server)
 {
+  gchar *result;
   g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
 
-  return g_strdup (server->address);
+  GST_RTSP_SERVER_LOCK (server);
+  result = g_strdup (server->address);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
 }
 
 /**
@@ -221,8 +253,10 @@ gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
   g_return_if_fail (GST_IS_RTSP_SERVER (server));
   g_return_if_fail (service != NULL);
 
+  GST_RTSP_SERVER_LOCK (server);
   g_free (server->service);
   server->service = g_strdup (service);
+  GST_RTSP_SERVER_UNLOCK (server);
 }
 
 /**
@@ -236,9 +270,15 @@ gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
 gchar *
 gst_rtsp_server_get_service (GstRTSPServer * server)
 {
+  gchar *result;
+
   g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
 
-  return g_strdup (server->service);
+  GST_RTSP_SERVER_LOCK (server);
+  result = g_strdup (server->service);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
 }
 
 /**
@@ -256,7 +296,9 @@ gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
 {
   g_return_if_fail (GST_IS_RTSP_SERVER (server));
 
+  GST_RTSP_SERVER_LOCK (server);
   server->backlog = backlog;
+  GST_RTSP_SERVER_UNLOCK (server);
 }
 
 /**
@@ -270,9 +312,15 @@ gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
 gint
 gst_rtsp_server_get_backlog (GstRTSPServer * server)
 {
+  gint result;
+
   g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
 
-  return server->backlog;
+  GST_RTSP_SERVER_LOCK (server);
+  result = server->backlog;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
 }
 
 /**
@@ -290,15 +338,16 @@ gst_rtsp_server_set_session_pool (GstRTSPServer * server,
 
   g_return_if_fail (GST_IS_RTSP_SERVER (server));
 
-  old = server->session_pool;
-
-  if (old != pool) {
   if (pool)
     g_object_ref (pool);
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = server->session_pool;
   server->session_pool = pool;
+  GST_RTSP_SERVER_UNLOCK (server);
+
   if (old)
     g_object_unref (old);
-  }
 }
 
 /**
@@ -317,8 +366,10 @@ gst_rtsp_server_get_session_pool (GstRTSPServer * server)
 
   g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
 
+  GST_RTSP_SERVER_LOCK (server);
   if ((result = server->session_pool))
     g_object_ref (result);
+  GST_RTSP_SERVER_UNLOCK (server);
 
   return result;
 }
@@ -338,15 +389,16 @@ gst_rtsp_server_set_media_mapping (GstRTSPServer * server,
 
   g_return_if_fail (GST_IS_RTSP_SERVER (server));
 
-  old = server->media_mapping;
-
-  if (old != mapping) {
   if (mapping)
     g_object_ref (mapping);
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = server->media_mapping;
   server->media_mapping = mapping;
+  GST_RTSP_SERVER_UNLOCK (server);
+
   if (old)
     g_object_unref (old);
-  }
 }
 
 
@@ -366,8 +418,61 @@ gst_rtsp_server_get_media_mapping (GstRTSPServer * server)
 
   g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
 
+  GST_RTSP_SERVER_LOCK (server);
   if ((result = server->media_mapping))
     g_object_ref (result);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_auth:
+ * @server: a #GstRTSPServer
+ * @auth: a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @server.
+ */
+void
+gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
+{
+  GstRTSPAuth *old;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+  if (auth)
+    g_object_ref (auth);
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = server->auth;
+  server->auth = auth;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (old)
+    g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_server_get_auth:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @server.
+ *
+ * Returns: the #GstRTSPAuth of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_server_get_auth (GstRTSPServer * server)
+{
+  GstRTSPAuth *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  GST_RTSP_SERVER_LOCK (server);
+  if ((result = server->auth))
+    g_object_ref (result);
+  GST_RTSP_SERVER_UNLOCK (server);
 
   return result;
 }
@@ -426,17 +531,28 @@ gst_rtsp_server_set_property (GObject * object, guint propid,
   }
 }
 
-/* Prepare a server socket for @server and make it listen on the configured port */
+/**
-static gboolean
+ * gst_rtsp_server_get_io_channel:
-gst_rtsp_server_sink_init_send (GstRTSPServer * server)
+ * @server: a #GstRTSPServer
+ *
+ * Create a #GIOChannel for @server. The io channel will listen on the
+ * configured service.
+ *
+ * Returns: the GIOChannel for @server or NULL when an error occured.
+ */
+GIOChannel *
+gst_rtsp_server_get_io_channel (GstRTSPServer * server)
 {
-  int ret, sockfd;
+  GIOChannel *channel;
+  int ret, sockfd = -1;
   struct addrinfo hints;
   struct addrinfo *result, *rp;
 #ifdef USE_SOLINGER
   struct linger linger;
 #endif
 
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
   memset (&hints, 0, sizeof (struct addrinfo));
   hints.ai_family = AF_UNSPEC;  /* Allow IPv4 or IPv6 */
   hints.ai_socktype = SOCK_STREAM;      /* stream socket */
@@ -449,6 +565,7 @@ gst_rtsp_server_sink_init_send (GstRTSPServer * server)
   GST_DEBUG_OBJECT (server, "getting address info of %s/%s", server->address,
       server->service);
 
+  GST_RTSP_SERVER_LOCK (server);
   /* resolve the server IP address */
   if ((ret =
           getaddrinfo (server->address, server->service, &hints, &result)) != 0)
@@ -464,6 +581,15 @@ gst_rtsp_server_sink_init_send (GstRTSPServer * server)
       continue;
     }
 
+    /* make address reusable */
+    ret = 1;
+    if (setsockopt (sockfd, SOL_SOCKET, SO_REUSEADDR,
+            (void *) &ret, sizeof (ret)) < 0) {
+      /* warn but try to bind anyway */
+      GST_WARNING_OBJECT (server, "failed to reuse socker (%s)",
+          g_strerror (errno));
+    }
+
     if (bind (sockfd, rp->ai_addr, rp->ai_addrlen) == 0) {
       GST_DEBUG_OBJECT (server, "bind on %s", rp->ai_canonname);
       break;
@@ -472,26 +598,18 @@ gst_rtsp_server_sink_init_send (GstRTSPServer * server)
     GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
         g_strerror (errno));
     close (sockfd);
+    sockfd = -1;
   }
   freeaddrinfo (result);
 
-  if (rp == NULL)
+  if (sockfd == -1)
     goto no_socket;
 
-  server->server_sock.fd = sockfd;
+  GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d", sockfd);
-
-  GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
-      server->server_sock.fd);
-
-  /* make address reusable */
-  ret = 1;
-  if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_REUSEADDR,
-          (void *) &ret, sizeof (ret)) < 0)
-    goto reuse_failed;
 
   /* keep connection alive; avoids SIGPIPE during write */
   ret = 1;
-  if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
+  if (setsockopt (sockfd, SOL_SOCKET, SO_KEEPALIVE,
           (void *) &ret, sizeof (ret)) < 0)
     goto keepalive_failed;
 
@@ -501,43 +619,42 @@ gst_rtsp_server_sink_init_send (GstRTSPServer * server)
    * client. */
   linger.l_onoff = 1;
   linger.l_linger = 5;
-  if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_LINGER,
+  if (setsockopt (sockfd, SOL_SOCKET, SO_LINGER,
           (void *) &linger, sizeof (linger)) < 0)
     goto linger_failed;
 #endif
 
   /* set the server socket to nonblocking */
-  fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
+  fcntl (sockfd, F_SETFL, O_NONBLOCK);
 
   GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
-      server->server_sock.fd, server->backlog);
+      sockfd, server->backlog);
-  if (listen (server->server_sock.fd, server->backlog) == -1)
+  if (listen (sockfd, server->backlog) == -1)
     goto listen_failed;
 
   GST_DEBUG_OBJECT (server,
-      "listened on server socket %d, returning from connection setup",
+      "listened on server socket %d, returning from connection setup", sockfd);
-      server->server_sock.fd);
+
+  /* create IO channel for the socket */
+  channel = g_io_channel_unix_new (sockfd);
+  g_io_channel_set_close_on_unref (channel, TRUE);
 
   GST_INFO_OBJECT (server, "listening on service %s", server->service);
+  GST_RTSP_SERVER_UNLOCK (server);
 
-  return TRUE;
+  return channel;
 
   /* ERRORS */
 no_address:
   {
     GST_ERROR_OBJECT (server, "failed to resolve address: %s",
         gai_strerror (ret));
-    return FALSE;
+    goto close_error;
   }
 no_socket:
   {
     GST_ERROR_OBJECT (server, "failed to create socket: %s",
         g_strerror (errno));
-    return FALSE;
-  }
-reuse_failed:
-  {
-    GST_ERROR_OBJECT (server, "failed to reuse socket: %s", g_strerror (errno));
     goto close_error;
   }
 keepalive_failed:
@@ -562,18 +679,44 @@ listen_failed:
   }
 close_error:
   {
-    if (server->server_sock.fd >= 0) {
+    if (sockfd >= 0) {
-      close (server->server_sock.fd);
+      close (sockfd);
-      server->server_sock.fd = -1;
     }
-    return FALSE;
+    GST_RTSP_SERVER_UNLOCK (server);
+    return NULL;
   }
 }
 
-/* default method for creating a new client object in the server to accept and
+static void
- * handle a client connection on this server */
+unmanage_client (GstRTSPClient * client, GstRTSPServer * server)
+{
+  GST_DEBUG_OBJECT (server, "unmanage client %p", client);
+
+  gst_rtsp_client_set_server (client, NULL);
+
+  GST_RTSP_SERVER_LOCK (server);
+  server->clients = g_list_remove (server->clients, client);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  g_object_unref (client);
+}
+
+/* add the client to the active list of clients, takes ownership of
+ * the client */
+static void
+manage_client (GstRTSPServer * server, GstRTSPClient * client)
+{
+  GST_DEBUG_OBJECT (server, "manage client %p", client);
+  gst_rtsp_client_set_server (client, server);
+
+  GST_RTSP_SERVER_LOCK (server);
+  g_signal_connect (client, "closed", (GCallback) unmanage_client, server);
+  server->clients = g_list_prepend (server->clients, client);
+  GST_RTSP_SERVER_UNLOCK (server);
+}
+
 static GstRTSPClient *
-default_accept_client (GstRTSPServer * server, GIOChannel * channel)
+default_create_client (GstRTSPServer * server)
 {
   GstRTSPClient *client;
 
@@ -581,26 +724,38 @@ default_accept_client (GstRTSPServer * server, GIOChannel * channel)
   client = gst_rtsp_client_new ();
 
   /* set the session pool that this client should use */
+  GST_RTSP_SERVER_LOCK (server);
   gst_rtsp_client_set_session_pool (client, server->session_pool);
   /* set the media mapping that this client should use */
   gst_rtsp_client_set_media_mapping (client, server->media_mapping);
+  /* set authentication manager */
+  gst_rtsp_client_set_auth (client, server->auth);
+  GST_RTSP_SERVER_UNLOCK (server);
 
+  return client;
+}
+
+/* default method for creating a new client object in the server to accept and
+ * handle a client connection on this server */
+static gboolean
+default_accept_client (GstRTSPServer * server, GstRTSPClient * client,
+    GIOChannel * channel)
+{
   /* accept connections for that client, this function returns after accepting
    * the connection and will run the remainder of the communication with the
    * client asyncronously. */
   if (!gst_rtsp_client_accept (client, channel))
     goto accept_failed;
 
-  return client;
+  return TRUE;
 
   /* ERRORS */
 accept_failed:
   {
     GST_ERROR_OBJECT (server,
-        "Could not accept client on server socket %d: %s (%d)",
+        "Could not accept client on server : %s (%d)", g_strerror (errno),
-        server->server_sock.fd, g_strerror (errno), errno);
+        errno);
-    gst_object_unref (client);
+    return FALSE;
-    return NULL;
   }
 }
 
@@ -618,21 +773,26 @@ gboolean
 gst_rtsp_server_io_func (GIOChannel * channel, GIOCondition condition,
     GstRTSPServer * server)
 {
+  gboolean result;
   GstRTSPClient *client = NULL;
   GstRTSPServerClass *klass;
 
   if (condition & G_IO_IN) {
     klass = GST_RTSP_SERVER_GET_CLASS (server);
 
-    /* a new client connected, create a client object to handle the client. */
+    if (klass->create_client)
-    if (klass->accept_client)
+      client = klass->create_client (server);
-      client = klass->accept_client (server, channel);
     if (client == NULL)
       goto client_failed;
 
-    /* can unref the client now, when the request is finished, it will be
+    /* a new client connected, create a client object to handle the client. */
-     * unreffed async. */
+    if (klass->accept_client)
-    gst_object_unref (client);
+      result = klass->accept_client (server, client, channel);
+    if (!result)
+      goto accept_failed;
+
+    /* manage the client connection */
+    manage_client (server, client);
   } else {
     GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
   }
@@ -644,35 +804,19 @@ client_failed:
     GST_ERROR_OBJECT (server, "failed to create a client");
     return FALSE;
   }
+accept_failed:
+  {
+    GST_ERROR_OBJECT (server, "failed to accept client");
+    gst_object_unref (client);
+    return FALSE;
+  }
 }
 
-/**
+static void
- * gst_rtsp_server_get_io_channel:
+watch_destroyed (GstRTSPServer * server)
- * @server: a #GstRTSPServer
- *
- * Create a #GIOChannel for @server.
- *
- * Returns: the GIOChannel for @server or NULL when an error occured.
- */
-GIOChannel *
-gst_rtsp_server_get_io_channel (GstRTSPServer * server)
 {
-  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+  GST_DEBUG_OBJECT (server, "source destroyed");
-
+  g_object_unref (server);
-  if (server->io_channel == NULL) {
-    if (!gst_rtsp_server_sink_init_send (server))
-      goto init_failed;
-
-    /* create IO channel for the socket */
-    server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
-  }
-  return server->io_channel;
-
-init_failed:
-  {
-    GST_ERROR_OBJECT (server, "failed to initialize server");
-    return NULL;
-  }
 }
 
 /**
@@ -687,24 +831,26 @@ init_failed:
 GSource *
 gst_rtsp_server_create_watch (GstRTSPServer * server)
 {
-  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
-
-  if (server->io_watch == NULL) {
   GIOChannel *channel;
+  GSource *source;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
 
   channel = gst_rtsp_server_get_io_channel (server);
   if (channel == NULL)
     goto no_channel;
 
   /* create a watch for reads (new connections) and possible errors */
-    server->io_watch = g_io_create_watch (channel, G_IO_IN |
+  source = g_io_create_watch (channel, G_IO_IN |
       G_IO_ERR | G_IO_HUP | G_IO_NVAL);
+  g_io_channel_unref (channel);
 
   /* configure the callback */
-    g_source_set_callback (server->io_watch,
+  g_source_set_callback (source,
-        (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
+      (GSourceFunc) gst_rtsp_server_io_func, g_object_ref (server),
-  }
+      (GDestroyNotify) watch_destroyed);
-  return server->io_watch;
+
+  return source;
 
 no_channel:
   {
@@ -719,7 +865,8 @@ no_channel:
  * @context: a #GMainContext
  *
  * Attaches @server to @context. When the mainloop for @context is run, the
- * server will be dispatched.
+ * server will be dispatched. When @context 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.
@@ -739,6 +886,7 @@ gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
     goto no_source;
 
   res = g_source_attach (source, context);
+  g_source_unref (source);
 
   return res;
 

gst/rtsp-server/rtsp-server.h

@@ -17,31 +17,21 @@
  * Boston, MA 02111-1307, USA.
  */
 
-#include <stdio.h>
+#ifndef __GST_RTSP_SERVER_H__
-#include <stdlib.h>
+#define __GST_RTSP_SERVER_H__
-#include <unistd.h>
-#include <errno.h>
-#include <string.h>
-#include <sys/time.h>
-#include <sys/types.h>
-#include <netinet/in.h>
-#include <netdb.h>
-#include <sys/socket.h>
-#include <sys/wait.h>
-#include <fcntl.h>
-#include <arpa/inet.h>
 
 #include <gst/gst.h>
 
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPServer GstRTSPServer;
+typedef struct _GstRTSPServerClass GstRTSPServerClass;
+
 #include "rtsp-session-pool.h"
 #include "rtsp-media-mapping.h"
 #include "rtsp-media-factory-uri.h"
 #include "rtsp-client.h"
-
+#include "rtsp-auth.h"
-#ifndef __GST_RTSP_SERVER_H__
-#define __GST_RTSP_SERVER_H__
-
-G_BEGIN_DECLS
 
 #define GST_TYPE_RTSP_SERVER              (gst_rtsp_server_get_type ())
 #define GST_IS_RTSP_SERVER(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SERVER))
@@ -52,43 +42,53 @@ G_BEGIN_DECLS
 #define GST_RTSP_SERVER_CAST(obj)         ((GstRTSPServer*)(obj))
 #define GST_RTSP_SERVER_CLASS_CAST(klass) ((GstRTSPServerClass*)(klass))
 
-typedef struct _GstRTSPServer GstRTSPServer;
+#define GST_RTSP_SERVER_GET_LOCK(server)  (GST_RTSP_SERVER_CAST(server)->lock)
-typedef struct _GstRTSPServerClass GstRTSPServerClass;
+#define GST_RTSP_SERVER_LOCK(server)      (g_mutex_lock(GST_RTSP_SERVER_GET_LOCK(server)))
+#define GST_RTSP_SERVER_UNLOCK(server)    (g_mutex_unlock(GST_RTSP_SERVER_GET_LOCK(server)))
 
+/**
+ * GstRTSPServer:
+ *
+ * This object listens on a port, creates and manages the clients connected to
+ * it.
+ */
 struct _GstRTSPServer {
   GObject      parent;
 
+  GMutex      *lock;
+
   /* server information */
   gchar       *address;
   gchar       *service;
   gint         backlog;
 
-  struct  sockaddr_in server_sin;
-
-  /* socket and channels */
-  GstPollFD    server_sock;
-  GIOChannel  *io_channel;
-  GSource     *io_watch;
-
   /* sessions on this server */
   GstRTSPSessionPool  *session_pool;
 
   /* media mapper for this server */
   GstRTSPMediaMapping *media_mapping;
+
+  /* authentication manager */
+  GstRTSPAuth *auth;
+
+  /* the clients that are connected */
+  GList   *clients;
 };
 
 /**
  * GstRTSPServerClass:
  *
- * @accept_client: Create, configure, accept and return a new GstRTSPClient 
+ * @create_client: Create, configure a new GstRTSPClient
  *          object that handles the new connection on @channel.
+ * @accept_client: accept a new GstRTSPClient
  *
  * The RTSP server class structure
  */
 struct _GstRTSPServerClass {
   GObjectClass  parent_class;
 
-  GstRTSPClient * (*accept_client) (GstRTSPServer *server, GIOChannel *channel);
+  GstRTSPClient * (*create_client) (GstRTSPServer *server);
+  gboolean        (*accept_client) (GstRTSPServer *server, GstRTSPClient *client, GIOChannel *channel);
 };
 
 GType                 gst_rtsp_server_get_type             (void);
@@ -110,6 +110,9 @@ GstRTSPSessionPool *  gst_rtsp_server_get_session_pool     (GstRTSPServer *serve
 void                  gst_rtsp_server_set_media_mapping    (GstRTSPServer *server, GstRTSPMediaMapping *mapping);
 GstRTSPMediaMapping * gst_rtsp_server_get_media_mapping    (GstRTSPServer *server);
 
+void                  gst_rtsp_server_set_auth             (GstRTSPServer *server, GstRTSPAuth *auth);
+GstRTSPAuth *         gst_rtsp_server_get_auth             (GstRTSPServer *server);
+
 gboolean              gst_rtsp_server_io_func              (GIOChannel *channel, GIOCondition condition,
                                                             GstRTSPServer *server);
 

gst/rtsp-server/rtsp-session-pool.c

@@ -30,7 +30,15 @@ enum
   PROP_LAST
 };
 
-GST_DEBUG_CATEGORY (rtsp_session_debug);
+static const gchar session_id_charset[] =
+    { 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', 'o',
+  'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z', 'A', 'B', 'C', 'D',
+  'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S',
+  'T', 'U', 'V', 'W', 'X', 'Y', 'Z', '0', '1', '2', '3', '4', '5', '6', '7',
+  '8', '9', '$', '-', '_', '.', '+'
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
 #define GST_CAT_DEFAULT rtsp_session_debug
 
 static void gst_rtsp_session_pool_get_property (GObject * object, guint propid,
@@ -62,8 +70,8 @@ gst_rtsp_session_pool_class_init (GstRTSPSessionPoolClass * klass)
 
   klass->create_session_id = create_session_id;
 
-  GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsession", 0,
+  GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsessionpool", 0,
-      "GstRTSPSession");
+      "GstRTSPSessionPool");
 }
 
 static void
@@ -235,7 +243,9 @@ create_session_id (GstRTSPSessionPool * pool)
   gint i;
 
   for (i = 0; i < 16; i++) {
-    id[i] = g_random_int_range ('a', 'z');
+    id[i] =
+        session_id_charset[g_random_int_range (0,
+            G_N_ELEMENTS (session_id_charset))];
   }
 
   return g_strndup (id, 16);

gst/rtsp-server/rtsp-session-pool.h

@@ -22,10 +22,14 @@
 #ifndef __GST_RTSP_SESSION_POOL_H__
 #define __GST_RTSP_SESSION_POOL_H__
 
-#include "rtsp-session.h"
 
 G_BEGIN_DECLS
 
+typedef struct _GstRTSPSessionPool GstRTSPSessionPool;
+typedef struct _GstRTSPSessionPoolClass GstRTSPSessionPoolClass;
+
+#include "rtsp-session.h"
+
 #define GST_TYPE_RTSP_SESSION_POOL              (gst_rtsp_session_pool_get_type ())
 #define GST_IS_RTSP_SESSION_POOL(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_POOL))
 #define GST_IS_RTSP_SESSION_POOL_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_POOL))
@@ -35,9 +39,6 @@ G_BEGIN_DECLS
 #define GST_RTSP_SESSION_POOL_CAST(obj)         ((GstRTSPSessionPool*)(obj))
 #define GST_RTSP_SESSION_POOL_CLASS_CAST(klass) ((GstRTSPSessionPoolClass*)(klass))
 
-typedef struct _GstRTSPSessionPool GstRTSPSessionPool;
-typedef struct _GstRTSPSessionPoolClass GstRTSPSessionPoolClass;
-
 /**
  * GstRTSPSessionPool:
  * @max_sessions: the maximum number of sessions.

gst/rtsp-server/rtsp-session.c

@@ -32,7 +32,7 @@ enum
   PROP_LAST
 };
 
-GST_DEBUG_CATEGORY_EXTERN (rtsp_session_debug);
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
 #define GST_CAT_DEFAULT rtsp_session_debug
 
 static void gst_rtsp_session_get_property (GObject * object, guint propid,
@@ -63,6 +63,9 @@ gst_rtsp_session_class_init (GstRTSPSessionClass * klass)
       g_param_spec_uint ("timeout", "timeout",
           "the timeout of the session (0 = never)", 0, G_MAXUINT,
           DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsession", 0,
+      "GstRTSPSession");
 }
 
 static void
@@ -79,7 +82,8 @@ gst_rtsp_session_free_stream (GstRTSPSessionStream * stream)
   GST_INFO ("free session stream %p", stream);
 
   /* remove callbacks now */
-  gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
+  gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL, NULL,
+      NULL);
   gst_rtsp_session_stream_set_keepalive (stream, NULL, NULL, NULL);
 
   gst_rtsp_media_trans_cleanup (&stream->trans);
@@ -549,6 +553,8 @@ gst_rtsp_session_stream_set_transport (GstRTSPSessionStream * stream,
  * @stream: a #GstRTSPSessionStream
  * @send_rtp: a callback called when RTP should be sent
  * @send_rtcp: a callback called when RTCP should be sent
+ * @send_rtp_list: a callback called when RTP should be sent
+ * @send_rtcp_list: a callback called when RTCP should be sent
  * @user_data: user data passed to callbacks
  * @notify: called with the user_data when no longer needed.
  *
@@ -557,11 +563,14 @@ gst_rtsp_session_stream_set_transport (GstRTSPSessionStream * stream,
  */
 void
 gst_rtsp_session_stream_set_callbacks (GstRTSPSessionStream * stream,
-    GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp, gpointer user_data,
+    GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
-    GDestroyNotify notify)
+    GstRTSPSendListFunc send_rtp_list, GstRTSPSendListFunc send_rtcp_list,
+    gpointer user_data, GDestroyNotify notify)
 {
   stream->trans.send_rtp = send_rtp;
   stream->trans.send_rtcp = send_rtcp;
+  stream->trans.send_rtp_list = send_rtp_list;
+  stream->trans.send_rtcp_list = send_rtcp_list;
   if (stream->trans.notify)
     stream->trans.notify (stream->trans.user_data);
   stream->trans.user_data = user_data;

gst/rtsp-server/rtsp-session.h

@@ -21,8 +21,6 @@
 
 #include <gst/rtsp/gstrtsptransport.h>
 
-#include "rtsp-media.h"
-
 #ifndef __GST_RTSP_SESSION_H__
 #define __GST_RTSP_SESSION_H__
 
@@ -43,6 +41,8 @@ typedef struct _GstRTSPSessionClass GstRTSPSessionClass;
 typedef struct _GstRTSPSessionStream GstRTSPSessionStream;
 typedef struct _GstRTSPSessionMedia GstRTSPSessionMedia;
 
+#include "rtsp-media.h"
+
 /**
  * GstRTSPSessionStream:
  * @trans: the media transport
@@ -154,6 +154,8 @@ GstRTSPTransport *     gst_rtsp_session_stream_set_transport (GstRTSPSessionStre
 void                   gst_rtsp_session_stream_set_callbacks (GstRTSPSessionStream *stream,
                                                               GstRTSPSendFunc send_rtp,
                                                               GstRTSPSendFunc send_rtcp,
+                                                              GstRTSPSendListFunc send_rtp_list,
+                                                              GstRTSPSendListFunc send_rtcp_list,
                                                               gpointer user_data,
                                                               GDestroyNotify  notify);
 void                   gst_rtsp_session_stream_set_keepalive (GstRTSPSessionStream *stream,

m4/Makefile.am

@@ -1,18 +0,0 @@
-EXTRA_DIST = \
-	codeset.m4
-	gettext.m4
-	glibc21.m4 \
-	iconv.m4 \
-	intdiv0.m4 \
-	inttypes-pri.m4 \
-	nttypes.m4 \
-	inttypes_h.m4 \
-	isc-posix.m4 \
-	lcmessage.m4 \
-	lib-ld.m4 \
-	lib-link.m4 \
-	lib-prefix.m4 \
-	progtest.m4 \
-	stdint_h.m4 \
-	uintmax_t.m4 \
-	ulonglong.m4

m4/codeset.m4

@@ -1,23 +0,0 @@
-# codeset.m4 serial AM1 (gettext-0.10.40)
-dnl Copyright (C) 2000-2002 Free Software Foundation, Inc.
-dnl This file is free software, distributed under the terms of the GNU
-dnl General Public License.  As a special exception to the GNU General
-dnl Public License, this file may be distributed as part of a program
-dnl that contains a configuration script generated by Autoconf, under
-dnl the same distribution terms as the rest of that program.
-
-dnl From Bruno Haible.
-
-AC_DEFUN([AM_LANGINFO_CODESET],
-[
-  AC_CACHE_CHECK([for nl_langinfo and CODESET], am_cv_langinfo_codeset,
-    [AC_TRY_LINK([#include <langinfo.h>],
-      [char* cs = nl_langinfo(CODESET);],
-      am_cv_langinfo_codeset=yes,
-      am_cv_langinfo_codeset=no)
-    ])
-  if test $am_cv_langinfo_codeset = yes; then
-    AC_DEFINE(HAVE_LANGINFO_CODESET, 1,
-      [Define if you have <langinfo.h> and nl_langinfo(CODESET).])
-  fi
-])

tests/.gitignore

@@ -0,0 +1 @@
+test-cleanup

tests/Makefile.am

@@ -0,0 +1,9 @@
+noinst_PROGRAMS = test-cleanup
+
+INCLUDES = -I$(top_srcdir) -I$(srcdir)
+
+AM_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+AM_LDFLAGS = \
+	$(GST_LIBS) \
+	$(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_MAJORMINOR@.la
+

tests/test-cleanup.c

@@ -0,0 +1,68 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#include <gst/gst.h>
+
+#include <gst/rtsp-server/rtsp-server.h>
+
+static gboolean
+timeout (GMainLoop * loop, gboolean ignored)
+{
+  g_main_loop_quit (loop);
+  return FALSE;
+}
+
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  guint id;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* attach the server to the default maincontext */
+  if ((id = gst_rtsp_server_attach (server, NULL)) == 0)
+    goto failed;
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, loop);
+
+  /* start serving */
+  g_main_loop_run (loop);
+
+  /* cleanup */
+  g_source_remove (id);
+  g_object_unref (server);
+  g_main_loop_unref (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}