ClientState -> Context

Rename the clientstate to context and put the code in a separate file.

diff --git a/examples/test-cgroups.c b/examples/test-cgroups.c
index 8188588..dcd700b 100644 (file)
--- a/examples/test-cgroups.c
+++ b/examples/test-cgroups.c
@@ -54,7 +54,7 @@ static void gst_rtsp_cgroup_pool_finalize (GObject * obj);
 static void default_thread_enter (GstRTSPThreadPool * pool,
     GstRTSPThread * thread);
 static void default_configure_thread (GstRTSPThreadPool * pool,
-    GstRTSPThread * thread, GstRTSPClientState * state);
+    GstRTSPThread * thread, GstRTSPContext * ctx);
 
 G_DEFINE_TYPE (GstRTSPCGroupPool, gst_rtsp_cgroup_pool,
     GST_TYPE_RTSP_THREAD_POOL);
@@ -118,14 +118,14 @@ default_thread_enter (GstRTSPThreadPool * pool, GstRTSPThread * thread)
 
 static void
 default_configure_thread (GstRTSPThreadPool * pool,
-    GstRTSPThread * thread, GstRTSPClientState * state)
+    GstRTSPThread * thread, GstRTSPContext * ctx)
 {
   GstRTSPCGroupPool *cpool = GST_RTSP_CGROUP_POOL (pool);
   const gchar *cls;
   struct cgroup *cgroup;
 
-  if (state->token)
-    cls = gst_rtsp_token_get_string (state->token, "cgroup.pool.media.class");
+  if (ctx->token)
+    cls = gst_rtsp_token_get_string (ctx->token, "cgroup.pool.media.class");
   else
     cls = NULL;
 

diff --git a/gst/rtsp-server/Makefile.am b/gst/rtsp-server/Makefile.am
index 725537e..638b2c6 100644 (file)
--- a/gst/rtsp-server/Makefile.am
+++ b/gst/rtsp-server/Makefile.am
@@ -1,6 +1,7 @@
 public_headers = \
                rtsp-auth.h \
                rtsp-address-pool.h \
+               rtsp-context.h \
                rtsp-params.h \
                rtsp-sdp.h \
                rtsp-thread-pool.h \
@@ -21,6 +22,7 @@ public_headers = \
 c_sources = \
        rtsp-auth.c \
        rtsp-address-pool.c \
+       rtsp-context.c \
        rtsp-params.c \
        rtsp-sdp.c \
        rtsp-thread-pool.c \

diff --git a/gst/rtsp-server/rtsp-auth.c b/gst/rtsp-server/rtsp-auth.c
index 68ab719..7264128 100644 (file)
--- a/gst/rtsp-server/rtsp-auth.c
+++ b/gst/rtsp-server/rtsp-auth.c
@@ -31,7 +31,7 @@
  * The RTSP server will call gst_rtsp_auth_check() with a string describing the
  * check to perform. The possible checks are prefixed with
  * #GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation
- * will use the current #GstRTSPToken, #GstRTSPClientState and
+ * will use the current #GstRTSPToken, #GstRTSPContext and
  * #GstRTSPPermissions on the object to check if an operation is allowed.
  *
  * The default #GstRTSPAuth object has support for basic authentication. With
@@ -79,9 +79,8 @@ 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_authenticate (GstRTSPAuth * auth,
-    GstRTSPClientState * state);
-static gboolean default_check (GstRTSPAuth * auth, GstRTSPClientState * state,
+static gboolean default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx);
+static gboolean default_check (GstRTSPAuth * auth, GstRTSPContext * ctx,
     const gchar * check);
 
 G_DEFINE_TYPE (GstRTSPAuth, gst_rtsp_auth, G_TYPE_OBJECT);
@@ -340,7 +339,7 @@ gst_rtsp_auth_remove_basic (GstRTSPAuth * auth, const gchar * basic)
 }
 
 static gboolean
-default_authenticate (GstRTSPAuth * auth, GstRTSPClientState * state)
+default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx)
 {
   GstRTSPAuthPrivate *priv = auth->priv;
   GstRTSPResult res;
@@ -349,13 +348,13 @@ default_authenticate (GstRTSPAuth * auth, GstRTSPClientState * state)
   GST_DEBUG_OBJECT (auth, "authenticate");
 
   g_mutex_lock (&priv->lock);
-  /* FIXME, need to ref but we have no way to unref when the state is
+  /* FIXME, need to ref but we have no way to unref when the ctx is
    * popped */
-  state->token = priv->default_token;
+  ctx->token = priv->default_token;
   g_mutex_unlock (&priv->lock);
 
   res =
-      gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_AUTHORIZATION,
+      gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_AUTHORIZATION,
       &authorization, 0);
   if (res < 0)
     goto no_auth;
@@ -368,7 +367,7 @@ default_authenticate (GstRTSPAuth * auth, GstRTSPClientState * state)
     g_mutex_lock (&priv->lock);
     if ((token = g_hash_table_lookup (priv->basic, &authorization[6]))) {
       GST_DEBUG_OBJECT (auth, "setting token %p", token);
-      state->token = token;
+      ctx->token = token;
     }
     g_mutex_unlock (&priv->lock);
   } else if (g_ascii_strncasecmp (authorization, "digest ", 7) == 0) {
@@ -385,35 +384,34 @@ no_auth:
 }
 
 static void
-send_response (GstRTSPAuth * auth, GstRTSPStatusCode code,
-    GstRTSPClientState * state)
+send_response (GstRTSPAuth * auth, GstRTSPStatusCode code, GstRTSPContext * ctx)
 {
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
   if (code == GST_RTSP_STS_UNAUTHORIZED) {
     /* we only have Basic for now */
-    gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+    gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_WWW_AUTHENTICATE,
         "Basic realm=\"GStreamer RTSP Server\"");
   }
-  gst_rtsp_client_send_message (state->client, state->session, state->response);
+  gst_rtsp_client_send_message (ctx->client, ctx->session, ctx->response);
 }
 
 static gboolean
-ensure_authenticated (GstRTSPAuth * auth, GstRTSPClientState * state)
+ensure_authenticated (GstRTSPAuth * auth, GstRTSPContext * ctx)
 {
   GstRTSPAuthClass *klass;
 
   klass = GST_RTSP_AUTH_GET_CLASS (auth);
 
   /* we need a token to check */
-  if (state->token == NULL) {
+  if (ctx->token == NULL) {
     if (klass->authenticate) {
-      if (!klass->authenticate (auth, state))
+      if (!klass->authenticate (auth, ctx))
         goto authenticate_failed;
     }
   }
-  if (state->token == NULL)
+  if (ctx->token == NULL)
     goto no_auth;
 
   return TRUE;
@@ -422,21 +420,20 @@ ensure_authenticated (GstRTSPAuth * auth, GstRTSPClientState * state)
 authenticate_failed:
   {
     GST_DEBUG_OBJECT (auth, "authenticate failed");
-    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, state);
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
     return FALSE;
   }
 no_auth:
   {
     GST_DEBUG_OBJECT (auth, "no authorization token found");
-    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, state);
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
     return FALSE;
   }
 }
 
 /* new connection */
 static gboolean
-check_connect (GstRTSPAuth * auth, GstRTSPClientState * state,
-    const gchar * check)
+check_connect (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
 {
   GstRTSPAuthPrivate *priv = auth->priv;
 
@@ -444,7 +441,7 @@ check_connect (GstRTSPAuth * auth, GstRTSPClientState * state,
     GTlsConnection *tls;
 
     /* configure the connection */
-    tls = gst_rtsp_connection_get_tls (state->conn, NULL);
+    tls = gst_rtsp_connection_get_tls (ctx->conn, NULL);
     g_tls_connection_set_certificate (tls, priv->certificate);
   }
   return TRUE;
@@ -452,12 +449,12 @@ check_connect (GstRTSPAuth * auth, GstRTSPClientState * state,
 
 /* check url and methods */
 static gboolean
-check_url (GstRTSPAuth * auth, GstRTSPClientState * state, const gchar * check)
+check_url (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
 {
   GstRTSPAuthPrivate *priv = auth->priv;
 
-  if ((state->method & priv->methods) != 0)
-    if (!ensure_authenticated (auth, state))
+  if ((ctx->method & priv->methods) != 0)
+    if (!ensure_authenticated (auth, ctx))
       goto not_authenticated;
 
   return TRUE;
@@ -471,19 +468,18 @@ not_authenticated:
 
 /* check access to media factory */
 static gboolean
-check_factory (GstRTSPAuth * auth, GstRTSPClientState * state,
-    const gchar * check)
+check_factory (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
 {
   const gchar *role;
   GstRTSPPermissions *perms;
 
-  if (!ensure_authenticated (auth, state))
+  if (!ensure_authenticated (auth, ctx))
     return FALSE;
 
-  if (!(role = gst_rtsp_token_get_string (state->token,
+  if (!(role = gst_rtsp_token_get_string (ctx->token,
               GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE)))
     goto no_media_role;
-  if (!(perms = gst_rtsp_media_factory_get_permissions (state->factory)))
+  if (!(perms = gst_rtsp_media_factory_get_permissions (ctx->factory)))
     goto no_permissions;
 
   if (g_str_equal (check, GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS)) {
@@ -501,55 +497,54 @@ check_factory (GstRTSPAuth * auth, GstRTSPClientState * state,
 no_media_role:
   {
     GST_DEBUG_OBJECT (auth, "no media factory role found");
-    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, state);
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
     return FALSE;
   }
 no_permissions:
   {
     GST_DEBUG_OBJECT (auth, "no permissions on media factory found");
-    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, state);
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
     return FALSE;
   }
 no_access:
   {
     GST_DEBUG_OBJECT (auth, "no permissions to access media factory");
-    send_response (auth, GST_RTSP_STS_NOT_FOUND, state);
+    send_response (auth, GST_RTSP_STS_NOT_FOUND, ctx);
     return FALSE;
   }
 no_construct:
   {
     GST_DEBUG_OBJECT (auth, "no permissions to construct media factory");
-    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, state);
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
     return FALSE;
   }
 }
 
 static gboolean
-check_client_settings (GstRTSPAuth * auth, GstRTSPClientState * state,
+check_client_settings (GstRTSPAuth * auth, GstRTSPContext * ctx,
     const gchar * check)
 {
-  if (!ensure_authenticated (auth, state))
+  if (!ensure_authenticated (auth, ctx))
     return FALSE;
 
-  return gst_rtsp_token_is_allowed (state->token,
+  return gst_rtsp_token_is_allowed (ctx->token,
       GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS);
 }
 
 static gboolean
-default_check (GstRTSPAuth * auth, GstRTSPClientState * state,
-    const gchar * check)
+default_check (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
 {
   gboolean res = FALSE;
 
   /* FIXME, use hastable or so */
   if (g_str_equal (check, GST_RTSP_AUTH_CHECK_CONNECT)) {
-    res = check_connect (auth, state, check);
+    res = check_connect (auth, ctx, check);
   } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_URL)) {
-    res = check_url (auth, state, check);
+    res = check_url (auth, ctx, check);
   } else if (g_str_has_prefix (check, "auth.check.media.factory.")) {
-    res = check_factory (auth, state, check);
+    res = check_factory (auth, ctx, check);
   } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS)) {
-    res = check_client_settings (auth, state, check);
+    res = check_client_settings (auth, ctx, check);
   }
   return res;
 }
@@ -580,16 +575,16 @@ gst_rtsp_auth_check (const gchar * check)
 {
   gboolean result = FALSE;
   GstRTSPAuthClass *klass;
-  GstRTSPClientState *state;
+  GstRTSPContext *ctx;
   GstRTSPAuth *auth;
 
   g_return_val_if_fail (check != NULL, FALSE);
 
-  if (!(state = gst_rtsp_client_state_get_current ()))
-    goto no_state;
+  if (!(ctx = gst_rtsp_context_get_current ()))
+    goto no_context;
 
   /* no auth, we don't need to check */
-  if (!(auth = state->auth))
+  if (!(auth = ctx->auth))
     return no_auth_check (check);
 
   klass = GST_RTSP_AUTH_GET_CLASS (auth);
@@ -597,14 +592,14 @@ gst_rtsp_auth_check (const gchar * check)
   GST_DEBUG_OBJECT (auth, "check authorization '%s'", check);
 
   if (klass->check)
-    result = klass->check (auth, state, check);
+    result = klass->check (auth, ctx, check);
 
   return result;
 
   /* ERRORS */
-no_state:
+no_context:
   {
-    GST_ERROR ("no clientstate found");
+    GST_ERROR ("no context found");
     return FALSE;
   }
 }

diff --git a/gst/rtsp-server/rtsp-auth.h b/gst/rtsp-server/rtsp-auth.h
index fd0da62..c2dd971 100644 (file)
--- a/gst/rtsp-server/rtsp-auth.h
+++ b/gst/rtsp-server/rtsp-auth.h
@@ -56,7 +56,7 @@ struct _GstRTSPAuth {
  * @authenticate: check the authentication of a client. The default implementation
  *         checks if the authentication in the header matches one of the basic
  *         authentication tokens. This function should set the authgroup field
- *         in the state.
+ *         in the context.
  * @check: check if a resource can be accessed. this function should
  *         call authenticate to authenticate the client when needed. The method
  *         should also construct and send an appropriate response message on
@@ -67,8 +67,8 @@ struct _GstRTSPAuth {
 struct _GstRTSPAuthClass {
   GObjectClass  parent_class;
 
-  gboolean           (*authenticate) (GstRTSPAuth *auth, GstRTSPClientState *state);
-  gboolean           (*check)        (GstRTSPAuth *auth, GstRTSPClientState *state,
+  gboolean           (*authenticate) (GstRTSPAuth *auth, GstRTSPContext *ctx);
+  gboolean           (*check)        (GstRTSPAuth *auth, GstRTSPContext *ctx,
                                       const gchar *check);
 };
 

diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index e67386a..dcaffaa 100644 (file)
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -127,11 +127,11 @@ static void client_session_finalized (GstRTSPClient * client,
 static void unlink_session_transports (GstRTSPClient * client,
     GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
 static gboolean default_configure_client_transport (GstRTSPClient * client,
-    GstRTSPClientState * state, GstRTSPTransport * ct);
+    GstRTSPContext * ctx, GstRTSPTransport * ct);
 static GstRTSPResult default_params_set (GstRTSPClient * client,
-    GstRTSPClientState * state);
+    GstRTSPContext * ctx);
 static GstRTSPResult default_params_get (GstRTSPClient * client,
-    GstRTSPClientState * state);
+    GstRTSPContext * ctx);
 
 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
 
@@ -444,12 +444,12 @@ send_message (GstRTSPClient * client, GstRTSPSession * session,
 
 static void
 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
-    GstRTSPClientState * state)
+    GstRTSPContext * ctx)
 {
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
-  send_message (client, NULL, state->response, FALSE);
+  send_message (client, NULL, ctx->response, FALSE);
 }
 
 static gboolean
@@ -471,7 +471,7 @@ paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
  * but is cached for when the same client (without breaking the connection) is
  * doing a setup for the exact same url. */
 static GstRTSPMedia *
-find_media (GstRTSPClient * client, GstRTSPClientState * state, gint * matched)
+find_media (GstRTSPClient * client, GstRTSPContext * ctx, gint * matched)
 {
   GstRTSPClientPrivate *priv = client->priv;
   GstRTSPMediaFactory *factory;
@@ -482,14 +482,14 @@ find_media (GstRTSPClient * client, GstRTSPClientState * state, gint * matched)
   if (!priv->mount_points)
     goto no_mount_points;
 
-  path = state->uri->abspath;
+  path = ctx->uri->abspath;
 
   /* find the longest matching factory for the uri first */
   if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
               path, matched)))
     goto no_factory;
 
-  state->factory = factory;
+  ctx->factory = factory;
 
   if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
     goto no_factory_access;
@@ -517,13 +517,13 @@ find_media (GstRTSPClient * client, GstRTSPClientState * state, gint * matched)
     priv->media = NULL;
 
     /* prepare the media and add it to the pipeline */
-    if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
+    if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
       goto no_media;
 
-    state->media = media;
+    ctx->media = media;
 
     thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
-        GST_RTSP_THREAD_TYPE_MEDIA, state);
+        GST_RTSP_THREAD_TYPE_MEDIA, ctx);
     if (thread == NULL)
       goto no_thread;
 
@@ -537,12 +537,12 @@ find_media (GstRTSPClient * client, GstRTSPClientState * state, gint * matched)
   } else {
     /* we have seen this path before, used cached media */
     media = priv->media;
-    state->media = media;
+    ctx->media = media;
     GST_INFO ("reusing cached media %p for path %s", media, priv->path);
   }
 
   g_object_unref (factory);
-  state->factory = NULL;
+  ctx->factory = NULL;
 
   if (media)
     g_object_ref (media);
@@ -553,13 +553,13 @@ find_media (GstRTSPClient * client, GstRTSPClientState * state, gint * matched)
 no_mount_points:
   {
     GST_ERROR ("client %p: no mount points configured", client);
-    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
     return NULL;
   }
 no_factory:
   {
     GST_ERROR ("client %p: no factory for uri %s", client, path);
-    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
     return NULL;
   }
 no_factory_access:
@@ -575,29 +575,29 @@ not_authorized:
 no_media:
   {
     GST_ERROR ("client %p: can't create media", client);
-    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
     g_object_unref (factory);
-    state->factory = NULL;
+    ctx->factory = NULL;
     return NULL;
   }
 no_thread:
   {
     GST_ERROR ("client %p: can't create thread", client);
-    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
     g_object_unref (media);
-    state->media = NULL;
+    ctx->media = NULL;
     g_object_unref (factory);
-    state->factory = NULL;
+    ctx->factory = NULL;
     return NULL;
   }
 no_prepare:
   {
     GST_ERROR ("client %p: can't prepare media", client);
-    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
     g_object_unref (media);
-    state->media = NULL;
+    ctx->media = NULL;
     g_object_unref (factory);
-    state->factory = NULL;
+    ctx->factory = NULL;
     return NULL;
   }
 }
@@ -711,7 +711,7 @@ close_connection (GstRTSPClient * client)
 }
 
 static gboolean
-handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPClientPrivate *priv = client->priv;
   GstRTSPSession *session;
@@ -720,15 +720,15 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
   const gchar *path;
   gint matched;
 
-  if (!state->session)
+  if (!ctx->session)
     goto no_session;
 
-  session = state->session;
+  session = ctx->session;
 
-  if (!state->uri)
+  if (!ctx->uri)
     goto no_uri;
 
-  path = state->uri->abspath;
+  path = ctx->uri->abspath;
 
   /* get a handle to the configuration of the media in the session */
   sessmedia = gst_rtsp_session_get_media (session, path, &matched);
@@ -739,11 +739,11 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
   if (path[matched] != '\0')
     goto no_aggregate;
 
-  state->sessmedia = sessmedia;
+  ctx->sessmedia = sessmedia;
 
   /* we emit the signal before closing the connection */
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
-      0, state);
+      0, ctx);
 
   /* unlink the all TCP callbacks */
   unlink_session_transports (client, session, sessmedia);
@@ -761,10 +761,10 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
   }
   /* construct the response now */
   code = GST_RTSP_STS_OK;
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
-  send_message (client, session, state->response, TRUE);
+  send_message (client, session, ctx->response, TRUE);
 
   return TRUE;
 
@@ -772,75 +772,75 @@ handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
 no_session:
   {
     GST_ERROR ("client %p: no session", client);
-    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
     return FALSE;
   }
 no_uri:
   {
     GST_ERROR ("client %p: no uri supplied", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     return FALSE;
   }
 not_found:
   {
     GST_ERROR ("client %p: no media for uri", client);
-    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
     return FALSE;
   }
 no_aggregate:
   {
     GST_ERROR ("client %p: no aggregate path %s", client, path);
     send_generic_response (client,
-        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
+        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
     return FALSE;
   }
 }
 
 static GstRTSPResult
-default_params_set (GstRTSPClient * client, GstRTSPClientState * state)
+default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPResult res;
 
-  res = gst_rtsp_params_set (client, state);
+  res = gst_rtsp_params_set (client, ctx);
 
   return res;
 }
 
 static GstRTSPResult
-default_params_get (GstRTSPClient * client, GstRTSPClientState * state)
+default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPResult res;
 
-  res = gst_rtsp_params_get (client, state);
+  res = gst_rtsp_params_get (client, ctx);
 
   return res;
 }
 
 static gboolean
-handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPResult res;
   guint8 *data;
   guint size;
 
-  res = gst_rtsp_message_get_body (state->request, &data, &size);
+  res = gst_rtsp_message_get_body (ctx->request, &data, &size);
   if (res != GST_RTSP_OK)
     goto bad_request;
 
   if (size == 0) {
     /* no body, keep-alive request */
-    send_generic_response (client, GST_RTSP_STS_OK, state);
+    send_generic_response (client, GST_RTSP_STS_OK, ctx);
   } else {
     /* there is a body, handle the params */
-    res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, state);
+    res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
     if (res != GST_RTSP_OK)
       goto bad_request;
 
-    send_message (client, state->session, state->response, FALSE);
+    send_message (client, ctx->session, ctx->response, FALSE);
   }
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
-      0, state);
+      0, ctx);
 
   return TRUE;
 
@@ -848,36 +848,36 @@ handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
 bad_request:
   {
     GST_ERROR ("client %p: bad request", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     return FALSE;
   }
 }
 
 static gboolean
-handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPResult res;
   guint8 *data;
   guint size;
 
-  res = gst_rtsp_message_get_body (state->request, &data, &size);
+  res = gst_rtsp_message_get_body (ctx->request, &data, &size);
   if (res != GST_RTSP_OK)
     goto bad_request;
 
   if (size == 0) {
     /* no body, keep-alive request */
-    send_generic_response (client, GST_RTSP_STS_OK, state);
+    send_generic_response (client, GST_RTSP_STS_OK, ctx);
   } else {
     /* there is a body, handle the params */
-    res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, state);
+    res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
     if (res != GST_RTSP_OK)
       goto bad_request;
 
-    send_message (client, state->session, state->response, FALSE);
+    send_message (client, ctx->session, ctx->response, FALSE);
   }
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
-      0, state);
+      0, ctx);
 
   return TRUE;
 
@@ -885,13 +885,13 @@ handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
 bad_request:
   {
     GST_ERROR ("client %p: bad request", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     return FALSE;
   }
 }
 
 static gboolean
-handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPSession *session;
   GstRTSPSessionMedia *sessmedia;
@@ -900,13 +900,13 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
   const gchar *path;
   gint matched;
 
-  if (!(session = state->session))
+  if (!(session = ctx->session))
     goto no_session;
 
-  if (!state->uri)
+  if (!ctx->uri)
     goto no_uri;
 
-  path = state->uri->abspath;
+  path = ctx->uri->abspath;
 
   /* get a handle to the configuration of the media in the session */
   sessmedia = gst_rtsp_session_get_media (session, path, &matched);
@@ -916,7 +916,7 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
   if (path[matched] != '\0')
     goto no_aggregate;
 
-  state->sessmedia = sessmedia;
+  ctx->sessmedia = sessmedia;
 
   rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
   /* the session state must be playing or recording */
@@ -932,16 +932,15 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
 
   /* construct the response now */
   code = GST_RTSP_STS_OK;
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
-  send_message (client, session, state->response, FALSE);
+  send_message (client, session, ctx->response, FALSE);
 
   /* the state is now READY */
   gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
 
-  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
-      0, state);
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
 
   return TRUE;
 
@@ -949,39 +948,39 @@ handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
 no_session:
   {
     GST_ERROR ("client %p: no seesion", client);
-    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
     return FALSE;
   }
 no_uri:
   {
     GST_ERROR ("client %p: no uri supplied", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     return FALSE;
   }
 not_found:
   {
     GST_ERROR ("client %p: no media for uri", client);
-    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
     return FALSE;
   }
 no_aggregate:
   {
     GST_ERROR ("client %p: no aggregate path %s", client, path);
     send_generic_response (client,
-        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
+        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
     return FALSE;
   }
 invalid_state:
   {
     GST_ERROR ("client %p: not PLAYING or RECORDING", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
-        state);
+        ctx);
     return FALSE;
   }
 }
 
 static gboolean
-handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPSession *session;
   GstRTSPSessionMedia *sessmedia;
@@ -997,13 +996,13 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
   const gchar *path;
   gint matched;
 
-  if (!(session = state->session))
+  if (!(session = ctx->session))
     goto no_session;
 
-  if (!state->uri)
+  if (!ctx->uri)
     goto no_uri;
 
-  path = state->uri->abspath;
+  path = ctx->uri->abspath;
 
   /* get a handle to the configuration of the media in the session */
   sessmedia = gst_rtsp_session_get_media (session, path, &matched);
@@ -1013,8 +1012,8 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
   if (path[matched] != '\0')
     goto no_aggregate;
 
-  state->sessmedia = sessmedia;
-  state->media = media = gst_rtsp_session_media_get_media (sessmedia);
+  ctx->sessmedia = sessmedia;
+  ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
 
   /* the session state must be playing or ready */
   rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
@@ -1022,8 +1021,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
     goto invalid_state;
 
   /* parse the range header if we have one */
-  res =
-      gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
+  res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
   if (res == GST_RTSP_OK) {
     if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
       /* we have a range, seek to the position */
@@ -1062,7 +1060,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
       if (infocount > 0)
         g_string_append (rtpinfo, ", ");
 
-      uristr = gst_rtsp_url_get_request_uri (state->uri);
+      uristr = gst_rtsp_url_get_request_uri (ctx->uri);
       g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
           uristr, i, seq, rtptime);
       g_free (uristr);
@@ -1075,30 +1073,29 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
 
   /* construct the response now */
   code = GST_RTSP_STS_OK;
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
   /* add the RTP-Info header */
   if (infocount > 0) {
     str = g_string_free (rtpinfo, FALSE);
-    gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
+    gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO, str);
   } else {
     g_string_free (rtpinfo, TRUE);
   }
 
   /* add the range */
   str = gst_rtsp_media_get_range_string (media, TRUE, unit);
-  gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
+  gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
 
-  send_message (client, session, state->response, FALSE);
+  send_message (client, session, ctx->response, FALSE);
 
   /* start playing after sending the request */
   gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
 
   gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
 
-  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
-      0, state);
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
 
   return TRUE;
 
@@ -1106,33 +1103,33 @@ handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
 no_session:
   {
     GST_ERROR ("client %p: no session", client);
-    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
     return FALSE;
   }
 no_uri:
   {
     GST_ERROR ("client %p: no uri supplied", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     return FALSE;
   }
 not_found:
   {
     GST_ERROR ("client %p: media not found", client);
-    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
     return FALSE;
   }
 no_aggregate:
   {
     GST_ERROR ("client %p: no aggregate path %s", client, path);
     send_generic_response (client,
-        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
+        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
     return FALSE;
   }
 invalid_state:
   {
     GST_ERROR ("client %p: not PLAYING or READY", client);
     send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
-        state);
+        ctx);
     return FALSE;
   }
 }
@@ -1227,7 +1224,7 @@ handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
 
 static gboolean
 default_configure_client_transport (GstRTSPClient * client,
-    GstRTSPClientState * state, GstRTSPTransport * ct)
+    GstRTSPContext * ctx, GstRTSPTransport * ct)
 {
   GstRTSPClientPrivate *priv = client->priv;
 
@@ -1241,7 +1238,7 @@ default_configure_client_transport (GstRTSPClient * client,
     if (ct->destination && use_client_settings) {
       GstRTSPAddress *addr;
 
-      addr = gst_rtsp_stream_reserve_address (state->stream, ct->destination,
+      addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
           ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
 
       if (addr == NULL)
@@ -1254,7 +1251,7 @@ default_configure_client_transport (GstRTSPClient * client,
 
       family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
 
-      addr = gst_rtsp_stream_get_multicast_address (state->stream, family);
+      addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
       if (addr == NULL)
         goto no_address;
 
@@ -1276,7 +1273,7 @@ default_configure_client_transport (GstRTSPClient * client,
     if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
       /* check if the client selected channels for TCP */
       if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
-        gst_rtsp_session_media_alloc_channels (state->sessmedia,
+        gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
             &ct->interleaved);
       }
     }
@@ -1292,7 +1289,7 @@ no_address:
 }
 
 static GstRTSPTransport *
-make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
+make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
     GstRTSPTransport * ct)
 {
   GstRTSPTransport *st;
@@ -1320,7 +1317,7 @@ make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
   switch (st->lower_transport) {
     case GST_RTSP_LOWER_TRANS_UDP:
       st->client_port = ct->client_port;
-      gst_rtsp_stream_get_server_port (state->stream, &st->server_port, family);
+      gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
       break;
     case GST_RTSP_LOWER_TRANS_UDP_MCAST:
       st->port = ct->port;
@@ -1333,13 +1330,13 @@ make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
       break;
   }
 
-  gst_rtsp_stream_get_ssrc (state->stream, &st->ssrc);
+  gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
 
   return st;
 }
 
 static gboolean
-handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPClientPrivate *priv = client->priv;
   GstRTSPResult res;
@@ -1359,15 +1356,15 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
   gchar *path, *control;
   gint matched;
 
-  if (!state->uri)
+  if (!ctx->uri)
     goto no_uri;
 
-  uri = state->uri;
+  uri = ctx->uri;
   path = uri->abspath;
 
   /* parse the transport */
   res =
-      gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
+      gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
       &transport, 0);
   if (res != GST_RTSP_OK)
     goto no_transport;
@@ -1377,7 +1374,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
   if (priv->session_pool == NULL)
     goto no_pool;
 
-  session = state->session;
+  session = ctx->session;
 
   if (session) {
     g_object_ref (session);
@@ -1392,7 +1389,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
   /* we have no session media, find one and manage it */
   if (sessmedia == NULL) {
     /* get a handle to the configuration of the media in the session */
-    media = find_media (client, state, &matched);
+    media = find_media (client, ctx, &matched);
   } else {
     if ((media = gst_rtsp_session_media_get_media (sessmedia)))
       g_object_ref (media);
@@ -1412,8 +1409,8 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
     goto stream_not_found;
 
   /* now we have a uri identifying a valid media and stream */
-  state->stream = stream;
-  state->media = media;
+  ctx->stream = stream;
+  ctx->media = media;
 
   if (session == NULL) {
     /* create a session if this fails we probably reached our session limit or
@@ -1428,7 +1425,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
     g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
         session);
 
-    state->session = session;
+    ctx->session = session;
   }
 
   if (sessmedia == NULL) {
@@ -1441,10 +1438,10 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
     g_object_unref (media);
   }
 
-  state->sessmedia = sessmedia;
+  ctx->sessmedia = sessmedia;
 
   /* set blocksize on this stream */
-  if (!handle_blocksize (media, stream, state->request))
+  if (!handle_blocksize (media, stream, ctx->request))
     goto invalid_blocksize;
 
   gst_rtsp_transport_new (&ct);
@@ -1459,7 +1456,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
 
   /* update the client transport */
   klass = GST_RTSP_CLIENT_GET_CLASS (client);
-  if (!klass->configure_client_transport (client, state, ct))
+  if (!klass->configure_client_transport (client, ctx, ct))
     goto unsupported_client_transport;
 
   /* set in the session media transport */
@@ -1470,20 +1467,20 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
       (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
 
   /* create and serialize the server transport */
-  st = make_server_transport (client, state, ct);
+  st = make_server_transport (client, ctx, ct);
   trans_str = gst_rtsp_transport_as_text (st);
   gst_rtsp_transport_free (st);
 
   /* construct the response now */
   code = GST_RTSP_STS_OK;
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
-  gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
       trans_str);
   g_free (trans_str);
 
-  send_message (client, session, state->response, FALSE);
+  send_message (client, session, ctx->response, FALSE);
 
   /* update the state */
   rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
@@ -1499,8 +1496,7 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
   }
   g_object_unref (session);
 
-  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
-      0, state);
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
 
   return TRUE;
 
@@ -1508,45 +1504,45 @@ handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
 no_uri:
   {
     GST_ERROR ("client %p: no uri", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     return FALSE;
   }
 no_transport:
   {
     GST_ERROR ("client %p: no transport", client);
-    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
     return FALSE;
   }
 no_pool:
   {
     GST_ERROR ("client %p: no session pool configured", client);
-    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
     return FALSE;
   }
 media_not_found:
   {
     GST_ERROR ("client %p: media '%s' not found", client, path);
-    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
     return FALSE;
   }
 stream_not_found:
   {
     GST_ERROR ("client %p: stream '%s' not found", client, control);
-    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
     g_object_unref (media);
     return FALSE;
   }
 service_unavailable:
   {
     GST_ERROR ("client %p: can't create session", client);
-    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
     g_object_unref (media);
     return FALSE;
   }
 sessmedia_unavailable:
   {
     GST_ERROR ("client %p: can't create session media", client);
-    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
     g_object_unref (media);
     g_object_unref (session);
     return FALSE;
@@ -1554,14 +1550,14 @@ sessmedia_unavailable:
 invalid_blocksize:
   {
     GST_ERROR ("client %p: invalid blocksize", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     g_object_unref (session);
     return FALSE;
   }
 unsupported_transports:
   {
     GST_ERROR ("client %p: unsupported transports", client);
-    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
     gst_rtsp_transport_free (ct);
     g_object_unref (session);
     return FALSE;
@@ -1569,7 +1565,7 @@ unsupported_transports:
 unsupported_client_transport:
   {
     GST_ERROR ("client %p: unsupported client transport", client);
-    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
     gst_rtsp_transport_free (ct);
     g_object_unref (session);
     return FALSE;
@@ -1624,7 +1620,7 @@ no_sdp:
 
 /* for the describe we must generate an SDP */
 static gboolean
-handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPResult res;
   GstSDPMessage *sdp;
@@ -1635,7 +1631,7 @@ handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
 
   klass = GST_RTSP_CLIENT_GET_CLASS (client);
 
-  if (!state->uri)
+  if (!ctx->uri)
     goto no_uri;
 
   /* check what kind of format is accepted, we don't really do anything with it
@@ -1644,7 +1640,7 @@ handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
     gchar *accept;
 
     res =
-        gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
+        gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
         &accept, i);
     if (res == GST_RTSP_ENOTIMPL)
       break;
@@ -1654,7 +1650,7 @@ handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
   }
 
   /* find the media object for the uri */
-  if (!(media = find_media (client, state, NULL)))
+  if (!(media = find_media (client, ctx, NULL)))
     goto no_media;
 
   /* create an SDP for the media object on this client */
@@ -1663,14 +1659,14 @@ handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
 
   g_object_unref (media);
 
-  gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
-      gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
+  gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+      gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
 
-  gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
       "application/sdp");
 
   /* content base for some clients that might screw up creating the setup uri */
-  str = gst_rtsp_url_get_request_uri (state->uri);
+  str = gst_rtsp_url_get_request_uri (ctx->uri);
   str_len = strlen (str);
 
   /* check for trailing '/' and append one */
@@ -1686,19 +1682,19 @@ handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
 
   GST_INFO ("adding content-base: %s", content_base);
 
-  gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE,
       content_base);
   g_free (content_base);
 
   /* add SDP to the response body */
   str = gst_sdp_message_as_text (sdp);
-  gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
+  gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
   gst_sdp_message_free (sdp);
 
-  send_message (client, state->session, state->response, FALSE);
+  send_message (client, ctx->session, ctx->response, FALSE);
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
-      0, state);
+      0, ctx);
 
   return TRUE;
 
@@ -1706,7 +1702,7 @@ handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
 no_uri:
   {
     GST_ERROR ("client %p: no uri", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     return FALSE;
   }
 no_media:
@@ -1718,14 +1714,14 @@ no_media:
 no_sdp:
   {
     GST_ERROR ("client %p: can't create SDP", client);
-    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
     g_object_unref (media);
     return FALSE;
   }
 }
 
 static gboolean
-handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
+handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPMethod options;
   gchar *str;
@@ -1739,16 +1735,16 @@ handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
 
   str = gst_rtsp_options_as_text (options);
 
-  gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
-      gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
+  gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+      gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
 
-  gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
   g_free (str);
 
-  send_message (client, state->session, state->response, FALSE);
+  send_message (client, ctx->session, ctx->response, FALSE);
 
   g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
-      0, state);
+      0, ctx);
 
   return TRUE;
 }
@@ -1798,69 +1794,6 @@ client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
   }
 }
 
-static GPrivate current_state;
-
-/**
- * gst_rtsp_client_state_get_current:
- *
- * Get the current #GstRTSPClientState. This object is retrieved from the
- * current thread that is handling the request for a client.
- *
- * Returns: a #GstRTSPClientState
- */
-GstRTSPClientState *
-gst_rtsp_client_state_get_current (void)
-{
-  GSList *l;
-
-  l = g_private_get (&current_state);
-  if (l == NULL)
-    return NULL;
-
-  return (GstRTSPClientState *) (l->data);
-
-}
-
-/**
- * gst_rtsp_client_state_push_current:
- * @state: a ##GstRTSPClientState
- *
- * Pushes @state onto the state stack. The current
- * state can then be received using gst_rtsp_client_state_get_current().
- **/
-void
-gst_rtsp_client_state_push_current (GstRTSPClientState * state)
-{
-  GSList *l;
-
-  g_return_if_fail (state != NULL);
-
-  l = g_private_get (&current_state);
-  l = g_slist_prepend (l, state);
-  g_private_set (&current_state, l);
-}
-
-/**
- * gst_rtsp_client_state_pop_current:
- * @state: a #GstRTSPClientState
- *
- * Pops @state off the state stack (verifying that @state
- * is on the top of the stack).
- **/
-void
-gst_rtsp_client_state_pop_current (GstRTSPClientState * state)
-{
-  GSList *l;
-
-  l = g_private_get (&current_state);
-
-  g_return_if_fail (l != NULL);
-  g_return_if_fail (l->data == state);
-
-  l = g_slist_delete_link (l, l);
-  g_private_set (&current_state, l);
-}
-
 static void
 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
 {
@@ -1871,20 +1804,20 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   GstRTSPVersion version;
   GstRTSPResult res;
   GstRTSPSession *session = NULL;
-  GstRTSPClientState sstate = { NULL }, *state;
+  GstRTSPContext sctx = { NULL }, *ctx;
   GstRTSPMessage response = { 0 };
   gchar *sessid;
 
-  if (!(state = gst_rtsp_client_state_get_current ())) {
-    state = &sstate;
-    state->auth = priv->auth;
-    gst_rtsp_client_state_push_current (state);
+  if (!(ctx = gst_rtsp_context_get_current ())) {
+    ctx = &sctx;
+    ctx->auth = priv->auth;
+    gst_rtsp_context_push_current (ctx);
   }
 
-  state->conn = priv->connection;
-  state->client = client;
-  state->request = request;
-  state->response = &response;
+  ctx->conn = priv->connection;
+  ctx->client = client;
+  ctx->request = request;
+  ctx->response = &response;
 
   if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
     gst_rtsp_message_dump (request);
@@ -1898,7 +1831,7 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   if (version != GST_RTSP_VERSION_1_0)
     goto not_supported;
 
-  state->method = method;
+  ctx->method = method;
 
   /* we always try to parse the url first */
   if (strcmp (uristr, "*") == 0) {
@@ -1925,8 +1858,8 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   /* sanitize the uri */
   if (uri)
     sanitize_uri (uri);
-  state->uri = uri;
-  state->session = session;
+  ctx->uri = uri;
+  ctx->session = session;
 
   if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
     goto not_authorized;
@@ -1934,28 +1867,28 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   /* now see what is asked and dispatch to a dedicated handler */
   switch (method) {
     case GST_RTSP_OPTIONS:
-      handle_options_request (client, state);
+      handle_options_request (client, ctx);
       break;
     case GST_RTSP_DESCRIBE:
-      handle_describe_request (client, state);
+      handle_describe_request (client, ctx);
       break;
     case GST_RTSP_SETUP:
-      handle_setup_request (client, state);
+      handle_setup_request (client, ctx);
       break;
     case GST_RTSP_PLAY:
-      handle_play_request (client, state);
+      handle_play_request (client, ctx);
       break;
     case GST_RTSP_PAUSE:
-      handle_pause_request (client, state);
+      handle_pause_request (client, ctx);
       break;
     case GST_RTSP_TEARDOWN:
-      handle_teardown_request (client, state);
+      handle_teardown_request (client, ctx);
       break;
     case GST_RTSP_SET_PARAMETER:
-      handle_set_param_request (client, state);
+      handle_set_param_request (client, ctx);
       break;
     case GST_RTSP_GET_PARAMETER:
-      handle_get_param_request (client, state);
+      handle_get_param_request (client, ctx);
       break;
     case GST_RTSP_ANNOUNCE:
     case GST_RTSP_RECORD:
@@ -1967,8 +1900,8 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   }
 
 done:
-  if (state == &sstate)
-    gst_rtsp_client_state_pop_current (state);
+  if (ctx == &sctx)
+    gst_rtsp_context_pop_current (ctx);
   if (session)
     g_object_unref (session);
   if (uri)
@@ -1980,25 +1913,25 @@ not_supported:
   {
     GST_ERROR ("client %p: version %d not supported", client, version);
     send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
-        state);
+        ctx);
     goto done;
   }
 bad_request:
   {
     GST_ERROR ("client %p: bad request", client);
-    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
     goto done;
   }
 no_pool:
   {
     GST_ERROR ("client %p: no pool configured", client);
-    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
     goto done;
   }
 session_not_found:
   {
     GST_ERROR ("client %p: session not found", client);
-    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
     goto done;
   }
 not_authorized:
@@ -2009,7 +1942,7 @@ not_authorized:
 not_implemented:
   {
     GST_ERROR ("client %p: method %d not implemented", client, method);
-    send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, state);
+    send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
     goto done;
   }
 }

diff --git a/gst/rtsp-server/rtsp-client.h b/gst/rtsp-server/rtsp-client.h
index 5ea6071..72c5187 100644 (file)
--- a/gst/rtsp-server/rtsp-client.h
+++ b/gst/rtsp-server/rtsp-client.h
@@ -27,18 +27,12 @@ G_BEGIN_DECLS
 
 typedef struct _GstRTSPClient GstRTSPClient;
 typedef struct _GstRTSPClientClass GstRTSPClientClass;
-typedef struct _GstRTSPClientState GstRTSPClientState;
 typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
 
-#include "rtsp-server.h"
-#include "rtsp-media.h"
+#include "rtsp-context.h"
 #include "rtsp-mount-points.h"
-#include "rtsp-session-pool.h"
-#include "rtsp-session-media.h"
-#include "rtsp-auth.h"
-#include "rtsp-thread-pool.h"
-#include "rtsp-token.h"
 #include "rtsp-sdp.h"
+#include "rtsp-auth.h"
 
 #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))
@@ -50,47 +44,6 @@ typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
 #define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
 
 /**
- * GstRTSPClientState:
- * @server: the server
- * @conn: the connection
- * @client: the client
- * @request: the complete request
- * @uri: the complete url parsed from @request
- * @method: the parsed method of @uri
- * @auth: the current auth object or NULL
- * @token: authorisation token
- * @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 media for the url can be NULL
- * @stream: the stream for the url can be NULL
- * @response: the response
- *
- * Information passed around containing the client state of a request.
- */
-struct _GstRTSPClientState {
-  GstRTSPServer       *server;
-  GstRTSPConnection   *conn;
-  GstRTSPClient       *client;
-  GstRTSPMessage      *request;
-  GstRTSPUrl          *uri;
-  GstRTSPMethod        method;
-  GstRTSPAuth         *auth;
-  GstRTSPToken        *token;
-  GstRTSPSession      *session;
-  GstRTSPSessionMedia *sessmedia;
-  GstRTSPMediaFactory *factory;
-  GstRTSPMedia        *media;
-  GstRTSPStream       *stream;
-  GstRTSPMessage      *response;
-};
-
-GstRTSPClientState * gst_rtsp_client_state_get_current   (void);
-void                 gst_rtsp_client_state_push_current  (GstRTSPClientState * state);
-void                 gst_rtsp_client_state_pop_current   (GstRTSPClientState * state);
-
-
-/**
  * GstRTSPClientSendFunc:
  * @client: a #GstRTSPClient
  * @message: a #GstRTSPMessage
@@ -124,9 +77,9 @@ struct _GstRTSPClient {
  * @configure_client_transport: called when the client transport needs to be
  *    configured.
  * @params_set: set parameters. This function should also initialize the
- *    RTSP response(state->response) via a call to gst_rtsp_message_init_response()
+ *    RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
  * @params_get: get parameters. This function should also initialize the
- *    RTSP response(state->response) via a call to gst_rtsp_message_init_response()
+ *    RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
  *
  * The client class structure.
  */
@@ -135,22 +88,22 @@ struct _GstRTSPClientClass {
 
   GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
   gboolean        (*configure_client_transport) (GstRTSPClient * client,
-                                                 GstRTSPClientState * state,
+                                                 GstRTSPContext * ctx,
                                                  GstRTSPTransport * ct);
-  GstRTSPResult   (*params_set) (GstRTSPClient *client, GstRTSPClientState *state);
-  GstRTSPResult   (*params_get) (GstRTSPClient *client, GstRTSPClientState *state);
+  GstRTSPResult   (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
+  GstRTSPResult   (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
 
   /* signals */
   void     (*closed)                  (GstRTSPClient *client);
   void     (*new_session)             (GstRTSPClient *client, GstRTSPSession *session);
-  void     (*options_request)         (GstRTSPClient *client, GstRTSPClientState *state);
-  void     (*describe_request)        (GstRTSPClient *client, GstRTSPClientState *state);
-  void     (*setup_request)           (GstRTSPClient *client, GstRTSPClientState *state);
-  void     (*play_request)            (GstRTSPClient *client, GstRTSPClientState *state);
-  void     (*pause_request)           (GstRTSPClient *client, GstRTSPClientState *state);
-  void     (*teardown_request)        (GstRTSPClient *client, GstRTSPClientState *state);
-  void     (*set_parameter_request)   (GstRTSPClient *client, GstRTSPClientState *state);
-  void     (*get_parameter_request)   (GstRTSPClient *client, GstRTSPClientState *state);
+  void     (*options_request)         (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*describe_request)        (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*setup_request)           (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*play_request)            (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*pause_request)           (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*teardown_request)        (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*set_parameter_request)   (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*get_parameter_request)   (GstRTSPClient *client, GstRTSPContext *ctx);
 };
 
 GType                 gst_rtsp_client_get_type          (void);

diff --git a/gst/rtsp-server/rtsp-context.c b/gst/rtsp-server/rtsp-context.c
new file mode 100644 (file)
index 0000000..3b8c0a1
--- /dev/null
+++ b/gst/rtsp-server/rtsp-context.c
@@ -0,0 +1,90 @@
+/* GStreamer
+ * Copyright (C) 2013 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-context
+ * @short_description: A client request context
+ * @see_also: #GstRTSPServer, #GstRTSPClient
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include "rtsp-context.h"
+
+static GPrivate current_context;
+
+/**
+ * gst_rtsp_context_get_current:
+ *
+ * Get the current #GstRTSPContext. This object is retrieved from the
+ * current thread that is handling the request for a client.
+ *
+ * Returns: a #GstRTSPContext
+ */
+GstRTSPContext *
+gst_rtsp_context_get_current (void)
+{
+  GSList *l;
+
+  l = g_private_get (&current_context);
+  if (l == NULL)
+    return NULL;
+
+  return (GstRTSPContext *) (l->data);
+
+}
+
+/**
+ * gst_rtsp_context_push_current:
+ * @ctx: a ##GstRTSPContext
+ *
+ * Pushes @ctx onto the context stack. The current
+ * context can then be received using gst_rtsp_context_get_current().
+ **/
+void
+gst_rtsp_context_push_current (GstRTSPContext * ctx)
+{
+  GSList *l;
+
+  g_return_if_fail (ctx != NULL);
+
+  l = g_private_get (&current_context);
+  l = g_slist_prepend (l, ctx);
+  g_private_set (&current_context, l);
+}
+
+/**
+ * gst_rtsp_context_pop_current:
+ * @ctx: a #GstRTSPContext
+ *
+ * Pops @ctx off the context stack (verifying that @ctx
+ * is on the top of the stack).
+ **/
+void
+gst_rtsp_context_pop_current (GstRTSPContext * ctx)
+{
+  GSList *l;
+
+  l = g_private_get (&current_context);
+
+  g_return_if_fail (l != NULL);
+  g_return_if_fail (l->data == ctx);
+
+  l = g_slist_delete_link (l, l);
+  g_private_set (&current_context, l);
+}

diff --git a/gst/rtsp-server/rtsp-context.h b/gst/rtsp-server/rtsp-context.h
new file mode 100644 (file)
index 0000000..60ff030
--- /dev/null
+++ b/gst/rtsp-server/rtsp-context.h
@@ -0,0 +1,81 @@
+/* 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspconnection.h>
+
+#ifndef __GST_RTSP_CONTEXT_H__
+#define __GST_RTSP_CONTEXT_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPContext GstRTSPContext;
+
+#include "rtsp-server.h"
+#include "rtsp-media.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-session-media.h"
+#include "rtsp-auth.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-token.h"
+
+/**
+ * GstRTSPContext:
+ * @server: the server
+ * @conn: the connection
+ * @client: the client
+ * @request: the complete request
+ * @uri: the complete url parsed from @request
+ * @method: the parsed method of @uri
+ * @auth: the current auth object or NULL
+ * @token: authorisation token
+ * @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 media for the url can be NULL
+ * @stream: the stream for the url can be NULL
+ * @response: the response
+ *
+ * Information passed around containing the context of a request.
+ */
+struct _GstRTSPContext {
+  GstRTSPServer       *server;
+  GstRTSPConnection   *conn;
+  GstRTSPClient       *client;
+  GstRTSPMessage      *request;
+  GstRTSPUrl          *uri;
+  GstRTSPMethod        method;
+  GstRTSPAuth         *auth;
+  GstRTSPToken        *token;
+  GstRTSPSession      *session;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia        *media;
+  GstRTSPStream       *stream;
+  GstRTSPMessage      *response;
+};
+
+GstRTSPContext *     gst_rtsp_context_get_current   (void);
+void                 gst_rtsp_context_push_current  (GstRTSPContext * ctx);
+void                 gst_rtsp_context_pop_current   (GstRTSPContext * ctx);
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CONTEXT_H__ */

diff --git a/gst/rtsp-server/rtsp-params.c b/gst/rtsp-server/rtsp-params.c
index d1f9b69..ce18eaa 100644 (file)
--- a/gst/rtsp-server/rtsp-params.c
+++ b/gst/rtsp-server/rtsp-params.c
@@ -31,14 +31,14 @@
 /**
  * gst_rtsp_params_set:
  * @client: a #GstRTSPClient
- * @state: a #GstRTSPClientState
+ * @ctx: a #GstRTSPContext
  *
  * Set parameters (not implemented yet)
  *
  * Returns: a #GstRTSPResult
  */
 GstRTSPResult
-gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
+gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPStatusCode code;
 
@@ -46,8 +46,8 @@ gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
    * with a list of the parameters */
   code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
 
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
   return GST_RTSP_OK;
 }
@@ -55,14 +55,14 @@ gst_rtsp_params_set (GstRTSPClient * client, GstRTSPClientState * state)
 /**
  * gst_rtsp_params_get:
  * @client: a #GstRTSPClient
- * @state: a #GstRTSPClientState
+ * @ctx: a #GstRTSPContext
  *
  * Get parameters (not implemented yet)
  *
  * Returns: a #GstRTSPResult
  */
 GstRTSPResult
-gst_rtsp_params_get (GstRTSPClient * client, GstRTSPClientState * state)
+gst_rtsp_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPStatusCode code;
 
@@ -70,8 +70,8 @@ gst_rtsp_params_get (GstRTSPClient * client, GstRTSPClientState * state)
    * with a list of the parameters */
   code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
 
-  gst_rtsp_message_init_response (state->response, code,
-      gst_rtsp_status_as_text (code), state->request);
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
 
   return GST_RTSP_OK;
 }

diff --git a/gst/rtsp-server/rtsp-params.h b/gst/rtsp-server/rtsp-params.h
index 93fb959..563757d 100644 (file)
--- a/gst/rtsp-server/rtsp-params.h
+++ b/gst/rtsp-server/rtsp-params.h
@@ -30,8 +30,8 @@
 
 G_BEGIN_DECLS
 
-GstRTSPResult    gst_rtsp_params_set      (GstRTSPClient * client, GstRTSPClientState * state);
-GstRTSPResult    gst_rtsp_params_get      (GstRTSPClient * client, GstRTSPClientState * state);
+GstRTSPResult    gst_rtsp_params_set      (GstRTSPClient * client, GstRTSPContext * ctx);
+GstRTSPResult    gst_rtsp_params_get      (GstRTSPClient * client, GstRTSPContext * ctx);
 
 G_END_DECLS
 

diff --git a/gst/rtsp-server/rtsp-server.c b/gst/rtsp-server/rtsp-server.c
index cd61614..538a2d2 100644 (file)
--- a/gst/rtsp-server/rtsp-server.c
+++ b/gst/rtsp-server/rtsp-server.c
@@ -991,26 +991,26 @@ unmanage_client (GstRTSPClient * client, ClientContext * ctx)
 static void
 manage_client (GstRTSPServer * server, GstRTSPClient * client)
 {
-  ClientContext *ctx;
+  ClientContext *cctx;
   GstRTSPServerPrivate *priv = server->priv;
   GMainContext *mainctx = NULL;
-  GstRTSPClientState state = { NULL };
+  GstRTSPContext ctx = { NULL };
 
   GST_DEBUG_OBJECT (server, "manage client %p", client);
 
-  ctx = g_slice_new0 (ClientContext);
-  ctx->server = g_object_ref (server);
-  ctx->client = client;
+  cctx = g_slice_new0 (ClientContext);
+  cctx->server = g_object_ref (server);
+  cctx->client = client;
 
   GST_RTSP_SERVER_LOCK (server);
 
-  state.server = server;
-  state.client = client;
+  ctx.server = server;
+  ctx.client = client;
 
-  ctx->thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
-      GST_RTSP_THREAD_TYPE_CLIENT, &state);
-  if (ctx->thread)
-    mainctx = ctx->thread->context;
+  cctx->thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+      GST_RTSP_THREAD_TYPE_CLIENT, &ctx);
+  if (cctx->thread)
+    mainctx = cctx->thread->context;
   else {
     GSource *source;
     /* find the context to add the watch */
@@ -1018,8 +1018,8 @@ manage_client (GstRTSPServer * server, GstRTSPClient * client)
       mainctx = g_source_get_context (source);
   }
 
-  g_signal_connect (client, "closed", (GCallback) unmanage_client, ctx);
-  priv->clients = g_list_prepend (priv->clients, ctx);
+  g_signal_connect (client, "closed", (GCallback) unmanage_client, cctx);
+  priv->clients = g_list_prepend (priv->clients, cctx);
 
   gst_rtsp_client_attach (client, mainctx);
 
@@ -1129,17 +1129,17 @@ gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
   GstRTSPServerClass *klass;
   GstRTSPResult res;
   GstRTSPConnection *conn = NULL;
-  GstRTSPClientState state = { NULL };
+  GstRTSPContext ctx = { NULL };
 
   if (condition & G_IO_IN) {
     /* a new client connected. */
     GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, NULL),
         accept_failed);
 
-    state.server = server;
-    state.conn = conn;
-    state.auth = priv->auth;
-    gst_rtsp_client_state_push_current (&state);
+    ctx.server = server;
+    ctx.conn = conn;
+    ctx.auth = priv->auth;
+    gst_rtsp_context_push_current (&ctx);
 
     if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_CONNECT))
       goto connection_refused;
@@ -1163,7 +1163,7 @@ gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
     GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
   }
 exit:
-  gst_rtsp_client_state_pop_current (&state);
+  gst_rtsp_context_pop_current (&ctx);
 
   return G_SOURCE_CONTINUE;
 

diff --git a/gst/rtsp-server/rtsp-thread-pool.c b/gst/rtsp-server/rtsp-thread-pool.c
index b29f20e..87a035f 100644 (file)
--- a/gst/rtsp-server/rtsp-thread-pool.c
+++ b/gst/rtsp-server/rtsp-thread-pool.c
@@ -197,7 +197,7 @@ static void gst_rtsp_thread_pool_finalize (GObject * obj);
 
 static gpointer do_loop (GstRTSPThread * thread);
 static GstRTSPThread *default_get_thread (GstRTSPThreadPool * pool,
-    GstRTSPThreadType type, GstRTSPClientState * state);
+    GstRTSPThreadType type, GstRTSPContext * ctx);
 
 G_DEFINE_TYPE (GstRTSPThreadPool, gst_rtsp_thread_pool, G_TYPE_OBJECT);
 
@@ -393,7 +393,7 @@ gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * pool)
 
 static GstRTSPThread *
 make_thread (GstRTSPThreadPool * pool, GstRTSPThreadType type,
-    GstRTSPClientState * state)
+    GstRTSPContext * ctx)
 {
   GstRTSPThreadPoolClass *klass;
   GstRTSPThread *thread;
@@ -407,14 +407,14 @@ make_thread (GstRTSPThreadPool * pool, GstRTSPThreadType type,
   GST_DEBUG_OBJECT (pool, "new thread %p", thread);
 
   if (klass->configure_thread)
-    klass->configure_thread (pool, thread, state);
+    klass->configure_thread (pool, thread, ctx);
 
   return thread;
 }
 
 static GstRTSPThread *
 default_get_thread (GstRTSPThreadPool * pool,
-    GstRTSPThreadType type, GstRTSPClientState * state)
+    GstRTSPThreadType type, GstRTSPContext * ctx)
 {
   GstRTSPThreadPoolPrivate *priv = pool->priv;
   GstRTSPThreadPoolClass *klass;
@@ -450,7 +450,7 @@ default_get_thread (GstRTSPThreadPool * pool,
         } else {
           /* make more threads */
           GST_DEBUG_OBJECT (pool, "make new client thread");
-          thread = make_thread (pool, type, state);
+          thread = make_thread (pool, type, ctx);
 
           if (!g_thread_pool_push (klass->pool, thread, &error))
             goto thread_error;
@@ -461,7 +461,7 @@ default_get_thread (GstRTSPThreadPool * pool,
       break;
     case GST_RTSP_THREAD_TYPE_MEDIA:
       GST_DEBUG_OBJECT (pool, "make new media thread");
-      thread = make_thread (pool, type, state);
+      thread = make_thread (pool, type, ctx);
 
       if (!g_thread_pool_push (klass->pool, thread, &error))
         goto thread_error;
@@ -486,15 +486,15 @@ thread_error:
  * gst_rtsp_thread_pool_get_thread:
  * @pool: a #GstRTSPThreadPool
  * @type: the #GstRTSPThreadType
- * @state: a #GstRTSPClientState
+ * @ctx: a #GstRTSPContext
  *
- * Get a new #GstRTSPThread for @type and @state.
+ * Get a new #GstRTSPThread for @type and @ctx.
  *
  * Returns: a new #GstRTSPThread, gst_rtsp_thread_stop() after usage
  */
 GstRTSPThread *
 gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool * pool,
-    GstRTSPThreadType type, GstRTSPClientState * state)
+    GstRTSPThreadType type, GstRTSPContext * ctx)
 {
   GstRTSPThreadPoolClass *klass;
   GstRTSPThread *result = NULL;
@@ -504,7 +504,7 @@ gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool * pool,
   klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
 
   if (klass->get_thread)
-    result = klass->get_thread (pool, type, state);
+    result = klass->get_thread (pool, type, ctx);
 
   return result;
 }

diff --git a/gst/rtsp-server/rtsp-thread-pool.h b/gst/rtsp-server/rtsp-thread-pool.h
index 74f9e07..148e73f 100644 (file)
--- a/gst/rtsp-server/rtsp-thread-pool.h
+++ b/gst/rtsp-server/rtsp-thread-pool.h
@@ -148,10 +148,10 @@ struct _GstRTSPThreadPoolClass {
 
   GstRTSPThread * (*get_thread)        (GstRTSPThreadPool *pool,
                                         GstRTSPThreadType type,
-                                        GstRTSPClientState *state);
+                                        GstRTSPContext *ctx);
   void            (*configure_thread)  (GstRTSPThreadPool *pool,
                                         GstRTSPThread * thread,
-                                        GstRTSPClientState *state);
+                                        GstRTSPContext *ctx);
 
   void            (*thread_enter)      (GstRTSPThreadPool *pool,
                                         GstRTSPThread *thread);
@@ -168,7 +168,7 @@ gint                gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * po
 
 GstRTSPThread *     gst_rtsp_thread_pool_get_thread      (GstRTSPThreadPool *pool,
                                                           GstRTSPThreadType type,
-                                                          GstRTSPClientState *state);
+                                                          GstRTSPContext *ctx);
 G_END_DECLS
 
 #endif /* __GST_RTSP_THREAD_POOL_H__ */

diff --git a/tests/check/gst/client.c b/tests/check/gst/client.c
index 8a442bc..58d5d83 100644 (file)
--- a/tests/check/gst/client.c
+++ b/tests/check/gst/client.c
@@ -529,17 +529,17 @@ GST_START_TEST (test_client_multicast_invalid_transport_specific)
   GstRTSPMessage request = { 0, };
   gchar *str;
   GstRTSPSessionPool *session_pool;
-  GstRTSPClientState state = { NULL };
+  GstRTSPContext ctx = { NULL };
 
   client = setup_multicast_client ();
 
-  state.client = client;
-  state.auth = gst_rtsp_auth_new ();
-  state.token =
+  ctx.client = client;
+  ctx.auth = gst_rtsp_auth_new ();
+  ctx.token =
       gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
       G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
       "user", NULL);
-  gst_rtsp_client_state_push_current (&state);
+  gst_rtsp_context_push_current (&ctx);
 
   /* simple SETUP with a valid URI and multicast, but an invalid ip */
   fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
@@ -604,9 +604,9 @@ GST_START_TEST (test_client_multicast_invalid_transport_specific)
 
 
   g_object_unref (client);
-  g_object_unref (state.auth);
-  gst_rtsp_token_unref (state.token);
-  gst_rtsp_client_state_pop_current (&state);
+  g_object_unref (ctx.auth);
+  gst_rtsp_token_unref (ctx.token);
+  gst_rtsp_context_pop_current (&ctx);
 }
 
 GST_END_TEST;
@@ -617,17 +617,17 @@ GST_START_TEST (test_client_multicast_transport_specific)
   GstRTSPMessage request = { 0, };
   gchar *str;
   GstRTSPSessionPool *session_pool;
-  GstRTSPClientState state = { NULL };
+  GstRTSPContext ctx = { NULL };
 
   client = setup_multicast_client ();
 
-  state.client = client;
-  state.auth = gst_rtsp_auth_new ();
-  state.token =
+  ctx.client = client;
+  ctx.auth = gst_rtsp_auth_new ();
+  ctx.token =
       gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
       G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
       "user", NULL);
-  gst_rtsp_client_state_push_current (&state);
+  gst_rtsp_context_push_current (&ctx);
 
   expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
       "ttl=1;port=5000-5001;mode=\"PLAY\"";
@@ -655,9 +655,9 @@ GST_START_TEST (test_client_multicast_transport_specific)
   g_object_unref (session_pool);
 
   g_object_unref (client);
-  g_object_unref (state.auth);
-  gst_rtsp_token_unref (state.token);
-  gst_rtsp_client_state_pop_current (&state);
+  g_object_unref (ctx.auth);
+  gst_rtsp_token_unref (ctx.token);
+  gst_rtsp_context_pop_current (&ctx);
 }
 
 GST_END_TEST;