/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#include <stdio.h>
#include <string.h>
+#include <gst/sdp/gstmikey.h>
+
#include "rtsp-client.h"
#include "rtsp-sdp.h"
#include "rtsp-params.h"
{
GMutex lock; /* protects everything else */
GMutex send_lock;
+ GMutex watch_lock;
GstRTSPConnection *connection;
GstRTSPWatch *watch;
+ GMainContext *watch_context;
guint close_seq;
gchar *server_ip;
gboolean is_ipv6;
GDestroyNotify send_notify; /* protected by send_lock */
GstRTSPSessionPool *session_pool;
+ gulong session_removed_id;
GstRTSPMountPoints *mount_points;
GstRTSPAuth *auth;
GstRTSPThreadPool *thread_pool;
gchar *path;
GstRTSPMedia *media;
- GList *transports;
+ GHashTable *transports;
GList *sessions;
+ guint sessions_cookie;
+
+ gboolean drop_backlog;
};
static GMutex tunnels_lock;
static GHashTable *tunnels; /* protected by tunnels_lock */
+/* FIXME make this configurable. We don't want to do this yet because it will
+ * be superceeded by a cache object later */
+#define WATCH_BACKLOG_SIZE 100
+
#define DEFAULT_SESSION_POOL NULL
#define DEFAULT_MOUNT_POINTS NULL
+#define DEFAULT_DROP_BACKLOG TRUE
enum
{
PROP_0,
PROP_SESSION_POOL,
PROP_MOUNT_POINTS,
+ PROP_DROP_BACKLOG,
PROP_LAST
};
SIGNAL_SET_PARAMETER_REQUEST,
SIGNAL_GET_PARAMETER_REQUEST,
SIGNAL_HANDLE_RESPONSE,
+ SIGNAL_SEND_MESSAGE,
+ SIGNAL_ANNOUNCE_REQUEST,
+ SIGNAL_RECORD_REQUEST,
SIGNAL_LAST
};
static void gst_rtsp_client_finalize (GObject * obj);
static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
-static void client_session_finalized (GstRTSPClient * client,
- GstRTSPSession * session);
-static void unlink_session_transports (GstRTSPClient * client,
- GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
+static gboolean handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx,
+ GstRTSPMedia * media, GstSDPMessage * sdp);
+static gboolean default_configure_client_media (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
static gboolean default_configure_client_transport (GstRTSPClient * client,
GstRTSPContext * ctx, GstRTSPTransport * ct);
static GstRTSPResult default_params_set (GstRTSPClient * client,
GstRTSPContext * ctx);
static GstRTSPResult default_params_get (GstRTSPClient * client,
GstRTSPContext * ctx);
+static gchar *default_make_path_from_uri (GstRTSPClient * client,
+ const GstRTSPUrl * uri);
+static void client_session_removed (GstRTSPSessionPool * pool,
+ GstRTSPSession * session, GstRTSPClient * client);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
gobject_class->finalize = gst_rtsp_client_finalize;
klass->create_sdp = create_sdp;
+ klass->handle_sdp = handle_sdp;
+ klass->configure_client_media = default_configure_client_media;
klass->configure_client_transport = default_configure_client_transport;
klass->params_set = default_params_set;
klass->params_get = default_params_get;
+ klass->make_path_from_uri = default_make_path_from_uri;
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
GST_TYPE_RTSP_MOUNT_POINTS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
+ g_param_spec_boolean ("drop-backlog", "Drop Backlog",
+ "Drop data when the backlog queue is full",
+ DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gst_rtsp_client_signals[SIGNAL_CLOSED] =
g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
- g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+ g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
- g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
+ g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
- NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
- G_TYPE_POINTER);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
- NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
- G_TYPE_POINTER);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
- NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
- G_TYPE_POINTER);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
- NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
- G_TYPE_POINTER);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
- NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
- G_TYPE_POINTER);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
- NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
- G_TYPE_POINTER);
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
- set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
- G_TYPE_NONE, 1, G_TYPE_POINTER);
+ set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
- get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
- G_TYPE_NONE, 1, G_TYPE_POINTER);
+ get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
- handle_response), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
- G_TYPE_NONE, 1, G_TYPE_POINTER);
+ handle_response), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+ /**
+ * GstRTSPClient::send-message:
+ * @client: The RTSP client
+ * @session: (type GstRtspServer.RTSPSession): The session
+ * @message: (type GstRtsp.RTSPMessage): The message
+ */
+ gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
+ g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+ send_message), NULL, NULL, g_cclosure_marshal_generic,
+ G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
+
+ gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST] =
+ g_signal_new ("announce-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, announce_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
+
+ gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST] =
+ g_signal_new ("record-request", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, record_request),
+ NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+ GST_TYPE_RTSP_CONTEXT);
tunnels =
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
g_mutex_init (&priv->lock);
g_mutex_init (&priv->send_lock);
+ g_mutex_init (&priv->watch_lock);
priv->close_seq = 0;
+ priv->drop_backlog = DEFAULT_DROP_BACKLOG;
+ priv->transports =
+ g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
+ g_object_unref);
}
static GstRTSPFilterResult
-filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
+filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
gpointer user_data)
{
- GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
-
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
- unlink_session_transports (client, sess, sessmedia);
- /* unmanage the media in the session */
return GST_RTSP_FILTER_REMOVE;
}
static void
-client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
-{
- /* unlink all media managed in this session */
- gst_rtsp_session_filter (session, filter_session, client);
-}
-
-static void
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
{
GstRTSPClientPrivate *priv = client->priv;
- GList *walk;
- for (walk = priv->sessions; walk; walk = g_list_next (walk)) {
- GstRTSPSession *msession = (GstRTSPSession *) walk->data;
+ g_mutex_lock (&priv->lock);
+ /* check if we already know about this session */
+ if (g_list_find (priv->sessions, session) == NULL) {
+ GST_INFO ("watching session %p", session);
- /* we already know about this session */
- if (msession == session)
- return;
- }
+ priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
+ priv->sessions_cookie++;
- GST_INFO ("watching session %p", session);
+ /* connect removed session handler, it will be disconnected when the last
+ * session gets removed */
+ if (priv->session_removed_id == 0)
+ priv->session_removed_id = g_signal_connect_data (priv->session_pool,
+ "session-removed", G_CALLBACK (client_session_removed),
+ g_object_ref (client), (GClosureNotify) g_object_unref, 0);
+ }
+ g_mutex_unlock (&priv->lock);
- g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
- client);
- priv->sessions = g_list_prepend (priv->sessions, session);
+ return;
}
+/* should be called with lock */
static void
-client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session)
+client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
+ GList * link)
{
GstRTSPClientPrivate *priv = client->priv;
- GST_INFO ("unwatching session %p", session);
+ GST_INFO ("client %p: unwatch session %p", client, session);
+
+ if (link == NULL) {
+ link = g_list_find (priv->sessions, session);
+ if (link == NULL)
+ return;
+ }
+
+ priv->sessions = g_list_delete_link (priv->sessions, link);
+ priv->sessions_cookie++;
+
+ /* if this was the last session, disconnect the handler.
+ * This will also drop the extra client ref */
+ if (!priv->sessions) {
+ g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
+ priv->session_removed_id = 0;
+ }
- g_object_weak_unref (G_OBJECT (session),
- (GWeakNotify) client_session_finalized, client);
- priv->sessions = g_list_remove (priv->sessions, session);
+ /* remove the session */
+ g_object_unref (session);
}
-static void
-client_cleanup_session (GstRTSPClient * client, GstRTSPSession * session)
+static GstRTSPFilterResult
+cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
+ gpointer user_data)
{
- g_object_weak_unref (G_OBJECT (session),
- (GWeakNotify) client_session_finalized, client);
- client_unlink_session (client, session);
+ /* unlink all media managed in this session. This needs to happen
+ * without the client lock, so we really want to do it here. */
+ gst_rtsp_session_filter (sess, filter_session_media, client);
+
+ return GST_RTSP_FILTER_REMOVE;
}
static void
-client_cleanup_sessions (GstRTSPClient * client)
+clean_cached_media (GstRTSPClient * client, gboolean unprepare)
{
GstRTSPClientPrivate *priv = client->priv;
- GList *sessions;
- /* remove weak-ref from sessions */
- for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) {
- client_cleanup_session (client, (GstRTSPSession *) sessions->data);
+ if (priv->path) {
+ g_free (priv->path);
+ priv->path = NULL;
+ }
+ if (priv->media) {
+ if (unprepare)
+ gst_rtsp_media_unprepare (priv->media);
+ g_object_unref (priv->media);
+ priv->media = NULL;
}
- g_list_free (priv->sessions);
- priv->sessions = NULL;
}
/* A client is finalized when the connection is broken */
GST_INFO ("finalize client %p", client);
+ if (priv->watch)
+ gst_rtsp_watch_set_flushing (priv->watch, TRUE);
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
if (priv->watch)
g_source_destroy ((GSource *) priv->watch);
- client_cleanup_sessions (client);
+ if (priv->watch_context)
+ g_main_context_unref (priv->watch_context);
+
+ /* all sessions should have been removed by now. We keep a ref to
+ * the client object for the session removed handler. The ref is
+ * dropped when the last session is removed from the list. */
+ g_assert (priv->sessions == NULL);
+ g_assert (priv->session_removed_id == 0);
+
+ g_hash_table_unref (priv->transports);
if (priv->connection)
gst_rtsp_connection_free (priv->connection);
- if (priv->session_pool)
+ if (priv->session_pool) {
g_object_unref (priv->session_pool);
+ }
if (priv->mount_points)
g_object_unref (priv->mount_points);
if (priv->auth)
if (priv->thread_pool)
g_object_unref (priv->thread_pool);
- if (priv->path)
- g_free (priv->path);
- if (priv->media) {
- gst_rtsp_media_unprepare (priv->media);
- g_object_unref (priv->media);
- }
+ clean_cached_media (client, TRUE);
g_free (priv->server_ip);
g_mutex_clear (&priv->lock);
g_mutex_clear (&priv->send_lock);
+ g_mutex_clear (&priv->watch_lock);
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
+ GstRTSPClientPrivate *priv = client->priv;
switch (propid) {
case PROP_SESSION_POOL:
case PROP_MOUNT_POINTS:
g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
break;
+ case PROP_DROP_BACKLOG:
+ g_value_set_boolean (value, priv->drop_backlog);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
const GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
+ GstRTSPClientPrivate *priv = client->priv;
switch (propid) {
case PROP_SESSION_POOL:
case PROP_MOUNT_POINTS:
gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
break;
+ case PROP_DROP_BACKLOG:
+ g_mutex_lock (&priv->lock);
+ priv->drop_backlog = g_value_get_boolean (value);
+ g_mutex_unlock (&priv->lock);
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
*
* Create a new #GstRTSPClient instance.
*
- * Returns: a new #GstRTSPClient
+ * Returns: (transfer full): a new #GstRTSPClient
*/
GstRTSPClient *
gst_rtsp_client_new (void)
}
static void
-send_message (GstRTSPClient * client, GstRTSPSession * session,
+send_message (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPMessage * message, gboolean close)
{
GstRTSPClientPrivate *priv = client->priv;
gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
/* add the new session header for new session ids */
- if (session) {
+ if (ctx->session) {
gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
- gst_rtsp_session_get_header (session));
+ gst_rtsp_session_get_header (ctx->session));
}
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
if (close)
gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
+ 0, ctx, message);
+
g_mutex_lock (&priv->send_lock);
if (priv->send_func)
priv->send_func (client, message, close, priv->send_data);
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
- send_message (client, NULL, ctx->response, FALSE);
+ ctx->session = NULL;
+
+ send_message (client, ctx, ctx->response, FALSE);
+}
+
+static void
+send_option_not_supported_response (GstRTSPClient * client,
+ GstRTSPContext * ctx, const gchar * unsupported_options)
+{
+ GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
+
+ gst_rtsp_message_init_response (ctx->response, code,
+ gst_rtsp_status_as_text (code), ctx->request);
+
+ if (unsupported_options != NULL) {
+ gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
+ unsupported_options);
+ }
+
+ ctx->session = NULL;
+
+ send_message (client, ctx, ctx->response, FALSE);
}
static gboolean
path_len = strlen (path);
if (!paths_are_equal (priv->path, path, path_len)) {
- GstRTSPThread *thread;
-
/* remove any previously cached values before we try to construct a new
* media for uri */
- if (priv->path)
- g_free (priv->path);
- priv->path = NULL;
- if (priv->media) {
- gst_rtsp_media_unprepare (priv->media);
- g_object_unref (priv->media);
- }
- priv->media = NULL;
+ clean_cached_media (client, TRUE);
/* prepare the media and add it to the pipeline */
if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
ctx->media = media;
- thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
- GST_RTSP_THREAD_TYPE_MEDIA, ctx);
- if (thread == NULL)
- goto no_thread;
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD)) {
+ GstRTSPThread *thread;
+
+ thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, ctx);
+ if (thread == NULL)
+ goto no_thread;
- /* prepare the media */
- if (!(gst_rtsp_media_prepare (media, thread)))
- goto no_prepare;
+ /* prepare the media */
+ if (!gst_rtsp_media_prepare (media, thread))
+ goto no_prepare;
+ }
/* now keep track of the uri and the media */
priv->path = g_strndup (path, path_len);
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMessage message = { 0 };
+ GstRTSPResult res = GST_RTSP_OK;
GstMapInfo map_info;
guint8 *data;
guint usize;
g_mutex_lock (&priv->send_lock);
if (priv->send_func)
- priv->send_func (client, &message, FALSE, priv->send_data);
+ res = priv->send_func (client, &message, FALSE, priv->send_data);
g_mutex_unlock (&priv->send_lock);
gst_rtsp_message_steal_body (&message, &data, &usize);
gst_rtsp_message_unset (&message);
- return TRUE;
-}
-
-static void
-link_transport (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPStreamTransport * trans)
-{
- GstRTSPClientPrivate *priv = client->priv;
-
- GST_DEBUG ("client %p: linking transport %p", client, trans);
-
- gst_rtsp_stream_transport_set_callbacks (trans,
- (GstRTSPSendFunc) do_send_data,
- (GstRTSPSendFunc) do_send_data, client, NULL);
-
- priv->transports = g_list_prepend (priv->transports, trans);
-
- /* make sure our session can't expire */
- gst_rtsp_session_prevent_expire (session);
+ return res == GST_RTSP_OK;
}
-static void
-unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPStreamTransport * trans)
+/**
+ * gst_rtsp_client_close:
+ * @client: a #GstRTSPClient
+ *
+ * Close the connection of @client and remove all media it was managing.
+ *
+ * Since: 1.4
+ */
+void
+gst_rtsp_client_close (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
+ const gchar *tunnelid;
- GST_DEBUG ("client %p: unlinking transport %p", client, trans);
-
- gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
-
- priv->transports = g_list_remove (priv->transports, trans);
-
- /* our session can now expire */
- gst_rtsp_session_allow_expire (session);
-}
-
-static void
-unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
- GstRTSPSessionMedia * sessmedia)
-{
- guint n_streams, i;
-
- n_streams =
- gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
- for (i = 0; i < n_streams; i++) {
- GstRTSPStreamTransport *trans;
- const GstRTSPTransport *tr;
-
- /* get the transport, if there is no transport configured, skip this stream */
- trans = gst_rtsp_session_media_get_transport (sessmedia, i);
- if (trans == NULL)
- continue;
-
- tr = gst_rtsp_stream_transport_get_transport (trans);
+ GST_DEBUG ("client %p: closing connection", client);
- if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
- /* for TCP, unlink the stream from the TCP connection of the client */
- unlink_transport (client, session, trans);
+ if (priv->connection) {
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
+ g_mutex_lock (&tunnels_lock);
+ /* remove from tunnelids */
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
}
+ gst_rtsp_connection_close (priv->connection);
+ }
+
+ /* connection is now closed, destroy the watch which will also cause the
+ * closed signal to be emitted */
+ if (priv->watch) {
+ GST_DEBUG ("client %p: destroying watch", client);
+ g_source_destroy ((GSource *) priv->watch);
+ priv->watch = NULL;
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ g_main_context_unref (priv->watch_context);
+ priv->watch_context = NULL;
}
}
-static void
-close_connection (GstRTSPClient * client)
+static gchar *
+default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
{
- GstRTSPClientPrivate *priv = client->priv;
- const gchar *tunnelid;
-
- GST_DEBUG ("client %p: closing connection", client);
+ gchar *path;
- if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
- g_mutex_lock (&tunnels_lock);
- /* remove from tunnelids */
- g_hash_table_remove (tunnels, tunnelid);
- g_mutex_unlock (&tunnels_lock);
- }
+ if (uri->query)
+ path = g_strconcat (uri->abspath, "?", uri->query, NULL);
+ else
+ path = g_strdup (uri->abspath);
- gst_rtsp_connection_close (priv->connection);
+ return path;
}
static gboolean
handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClientClass *klass;
GstRTSPSession *session;
GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code;
- const gchar *path;
+ gchar *path;
gint matched;
+ gboolean keep_session;
if (!ctx->session)
goto no_session;
if (!ctx->uri)
goto no_uri;
- path = ctx->uri->abspath;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (path[matched] != '\0')
goto no_aggregate;
+ g_free (path);
+
ctx->sessmedia = sessmedia;
/* we emit the signal before closing the connection */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
0, ctx);
- /* unlink the all TCP callbacks */
- unlink_session_transports (client, session, sessmedia);
-
- /* remove the session from the watched sessions */
- client_unwatch_session (client, session);
-
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
/* unmanage the media in the session, returns false if all media session
* are torn down. */
- if (!gst_rtsp_session_release_media (session, sessmedia)) {
- /* remove the session */
- gst_rtsp_session_pool_remove (priv->session_pool, session);
- }
+ keep_session = gst_rtsp_session_release_media (session, sessmedia);
+
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
- send_message (client, session, ctx->response, TRUE);
+ send_message (client, ctx, ctx->response, TRUE);
+
+ if (!keep_session) {
+ /* remove the session */
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
+ }
return TRUE;
{
GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
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, ctx);
+ g_free (path);
return FALSE;
}
}
if (res != GST_RTSP_OK)
goto bad_request;
- send_message (client, ctx->session, ctx->response, FALSE);
+ send_message (client, ctx, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
if (res != GST_RTSP_OK)
goto bad_request;
- send_message (client, ctx->session, ctx->response, FALSE);
+ send_message (client, ctx, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
+ GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code;
GstRTSPState rtspstate;
- const gchar *path;
+ gchar *path;
gint matched;
if (!(session = ctx->session))
if (!ctx->uri)
goto no_uri;
- path = ctx->uri->abspath;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (path[matched] != '\0')
goto no_aggregate;
+ g_free (path);
+
ctx->sessmedia = sessmedia;
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
rtspstate != GST_RTSP_STATE_RECORDING)
goto invalid_state;
- /* unlink the all TCP callbacks */
- unlink_session_transports (client, session, sessmedia);
-
/* then pause sending */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
- send_message (client, session, ctx->response, FALSE);
+ send_message (client, ctx, ctx->response, FALSE);
/* the state is now READY */
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
{
GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ g_free (path);
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, ctx);
+ g_free (path);
return FALSE;
}
invalid_state:
/* convert @url and @path to a URL used as a content base for the factory
* located at @path */
static gchar *
-make_base_url (GstRTSPClient * client, GstRTSPUrl * url, gchar * path)
+make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
{
GstRTSPUrl tmp;
- gchar *result, *trail;
+ gchar *result;
+ const gchar *trail;
/* check for trailing '/' and append one */
trail = (path[strlen (path) - 1] != '/' ? "/" : "");
handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
+ GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
GstRTSPUrl *uri;
- GString *rtpinfo;
- guint n_streams, i, infocount;
- gchar *str, *base_url;
+ gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
GstRTSPState rtspstate;
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
- gchar *path;
+ gchar *path, *rtpinfo;
gint matched;
if (!(session = ctx->session))
if (!(uri = ctx->uri))
goto no_uri;
- path = uri->abspath;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (path[matched] != '\0')
goto no_aggregate;
+ g_free (path);
+
ctx->sessmedia = sessmedia;
ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY))
+ goto unsupported_mode;
+
/* the session state must be playing or ready */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
+ /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
+
/* parse the range header if we have one */
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) {
+ GstRTSPMediaStatus media_status;
+
/* we have a range, seek to the position */
unit = range->unit;
gst_rtsp_media_seek (media, range);
gst_rtsp_range_free (range);
- }
- }
-
- /* grab RTPInfo from the payloaders now */
- rtpinfo = g_string_new ("");
-
- base_url = make_base_url (client, uri, path);
- n_streams = gst_rtsp_media_n_streams (media);
- for (i = 0, infocount = 0; i < n_streams; i++) {
- GstRTSPStreamTransport *trans;
- GstRTSPStream *stream;
- const GstRTSPTransport *tr;
- guint rtptime, seq;
-
- /* get the transport, if there is no transport configured, skip this stream */
- trans = gst_rtsp_session_media_get_transport (sessmedia, i);
- if (trans == NULL) {
- GST_INFO ("stream %d is not configured", i);
- continue;
- }
- tr = gst_rtsp_stream_transport_get_transport (trans);
-
- if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
- /* for TCP, link the stream to the TCP connection of the client */
- link_transport (client, session, trans);
- }
-
- stream = gst_rtsp_stream_transport_get_stream (trans);
- if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
- gchar *control;
-
- if (infocount > 0)
- g_string_append (rtpinfo, ", ");
-
- control = gst_rtsp_stream_get_control (stream);
- g_string_append_printf (rtpinfo, "url=%s%s;seq=%u;rtptime=%u",
- base_url, control, seq, rtptime);
- g_free (control);
-
- infocount++;
- } else {
- GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
+ media_status = gst_rtsp_media_get_status (media);
+ if (media_status == GST_RTSP_MEDIA_STATUS_ERROR)
+ goto seek_failed;
}
}
- g_free (base_url);
+
+ /* grab RTPInfo from the media now */
+ rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
/* construct the response now */
code = GST_RTSP_STS_OK;
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 (ctx->response, GST_RTSP_HDR_RTP_INFO, str);
- } else {
- g_string_free (rtpinfo, TRUE);
- }
+ if (rtpinfo)
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
+ rtpinfo);
/* add the range */
str = gst_rtsp_media_get_range_string (media, TRUE, unit);
if (str)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
- send_message (client, session, ctx->response, FALSE);
+ send_message (client, ctx, ctx->response, FALSE);
- /* start playing after sending the request */
+ /* start playing after sending the response */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
return FALSE;
}
invalid_state:
ctx);
return FALSE;
}
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ return FALSE;
+ }
+seek_failed:
+ {
+ GST_ERROR ("client %p: seek failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ return FALSE;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: media does not support PLAY", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+ return FALSE;
+ }
}
static void
}
/* parse @transport and return a valid transport in @tr. only transports
- * from @supported are returned. Returns FALSE if no valid transport
+ * supported by @stream are returned. Returns FALSE if no valid transport
* was found. */
static gboolean
-parse_transport (const char *transport, GstRTSPLowerTrans supported,
+parse_transport (const char *transport, GstRTSPStream * stream,
GstRTSPTransport * tr)
{
gint i;
goto next;
}
- /* we have a transport, see if it's RTP/AVP */
- if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) {
- GST_WARNING ("invalid transport %s", transports[i]);
- goto next;
- }
-
- if (!(tr->lower_transport & supported)) {
+ /* we have a transport, see if it's supported */
+ if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
GST_WARNING ("unsupported transport %s", transports[i]);
goto next;
}
}
static gboolean
-handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
- GstRTSPMessage * request)
+default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx)
{
+ GstRTSPMessage *request = ctx->request;
gchar *blocksize_str;
- gboolean ret = TRUE;
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
&blocksize_str, 0) == GST_RTSP_OK) {
gchar *end;
blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
- if (end == blocksize_str) {
- GST_ERROR ("failed to parse blocksize");
- ret = FALSE;
- } else {
- /* we don't want to change the mtu when this media
- * can be shared because it impacts other clients */
- if (gst_rtsp_media_is_shared (media))
- return TRUE;
-
- if (blocksize > G_MAXUINT)
- blocksize = G_MAXUINT;
- gst_rtsp_stream_set_mtu (stream, blocksize);
- }
+ if (end == blocksize_str)
+ goto parse_failed;
+
+ /* we don't want to change the mtu when this media
+ * can be shared because it impacts other clients */
+ if (gst_rtsp_media_is_shared (media))
+ goto done;
+
+ if (blocksize > G_MAXUINT)
+ blocksize = G_MAXUINT;
+
+ gst_rtsp_stream_set_mtu (stream, blocksize);
+ }
+done:
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_ERROR_OBJECT (client, "failed to parse blocksize");
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
}
- return ret;
}
static gboolean
ct->destination = g_strdup (url->host);
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
+ GSocket *sock;
+ GSocketAddress *addr;
+
+ sock = gst_rtsp_connection_get_read_socket (priv->connection);
+ if ((addr = g_socket_get_remote_address (sock, NULL))) {
+ /* our read port is the sender port of client */
+ ct->client_port.min =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ if ((addr = g_socket_get_local_address (sock, NULL))) {
+ ct->server_port.max =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ sock = gst_rtsp_connection_get_write_socket (priv->connection);
+ if ((addr = g_socket_get_remote_address (sock, NULL))) {
+ /* our write port is the receiver port of client */
+ ct->client_port.max =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
+ if ((addr = g_socket_get_local_address (sock, NULL))) {
+ ct->server_port.min =
+ g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+ g_object_unref (addr);
+ }
/* check if the client selected channels for TCP */
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
}
static GstRTSPTransport *
-make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
- GstRTSPTransport * ct)
+make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPContext * ctx, GstRTSPTransport * ct)
{
GstRTSPTransport *st;
GInetAddress *addr;
st->trans = ct->trans;
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
+ st->mode_play = ct->mode_play;
+ st->mode_record = ct->mode_record;
addr = g_inet_address_new_from_string (ct->destination);
break;
case GST_RTSP_LOWER_TRANS_TCP:
st->interleaved = ct->interleaved;
+ st->client_port = ct->client_port;
+ st->server_port = ct->server_port;
default:
break;
}
- gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
+ if ((gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY))
+ gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
return st;
}
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
static gboolean
-handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
+mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
{
- GstRTSPClientPrivate *priv = client->priv;
- GstRTSPResult res;
- GstRTSPUrl *uri;
- gchar *transport;
- GstRTSPTransport *ct, *st;
- GstRTSPLowerTrans supported;
- GstRTSPStatusCode code;
- GstRTSPSession *session;
- GstRTSPStreamTransport *trans;
- gchar *trans_str;
- GstRTSPSessionMedia *sessmedia;
- GstRTSPMedia *media;
- GstRTSPStream *stream;
- GstRTSPState rtspstate;
- GstRTSPClientClass *klass;
- gchar *path, *control;
- gint matched;
+ const gchar *srtp_cipher;
+ const gchar *srtp_auth;
+ const GstMIKEYPayload *sp;
+ guint i;
- if (!ctx->uri)
- goto no_uri;
+ /* loop over Security policy until we find one containing policy */
+ for (i = 0;; i++) {
+ if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
+ break;
- uri = ctx->uri;
- if (uri->query)
- path = g_strconcat (uri->abspath, "?", uri->query, NULL);
- else
- path = g_strdup (uri->abspath);
+ if (((GstMIKEYPayloadSP *) sp)->policy == policy)
+ break;
+ }
- /* parse the transport */
- res =
- gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
- &transport, 0);
- if (res != GST_RTSP_OK)
- goto no_transport;
+ /* the default ciphers */
+ srtp_cipher = "aes-128-icm";
+ srtp_auth = "hmac-sha1-80";
+
+ /* now override the defaults with what is in the Security Policy */
+ if (sp != NULL) {
+ guint len;
+
+ /* collect all the params and go over them */
+ len = gst_mikey_payload_sp_get_n_params (sp);
+ for (i = 0; i < len; i++) {
+ const GstMIKEYPayloadSPParam *param =
+ gst_mikey_payload_sp_get_param (sp, i);
+
+ switch (param->type) {
+ case GST_MIKEY_SP_SRTP_ENC_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_cipher = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_cipher = "aes-128-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
+ switch (param->val[0]) {
+ case AES_128_KEY_LEN:
+ srtp_cipher = "aes-128-icm";
+ break;
+ case AES_256_KEY_LEN:
+ srtp_cipher = "aes-256-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_auth = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
+ switch (param->val[0]) {
+ case HMAC_32_KEY_LEN:
+ srtp_auth = "hmac-sha1-32";
+ break;
+ case HMAC_80_KEY_LEN:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_SRTP_ENC:
+ break;
+ case GST_MIKEY_SP_SRTP_SRTCP_ENC:
+ break;
+ default:
+ break;
+ }
+ }
+ }
+ /* now configure the SRTP parameters */
+ gst_caps_set_simple (caps,
+ "srtp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtp-auth", G_TYPE_STRING, srtp_auth,
+ "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
- /* we create the session after parsing stuff so that we don't make
- * a session for malformed requests */
- if (priv->session_pool == NULL)
- goto no_pool;
+ return TRUE;
+}
- session = ctx->session;
+static gboolean
+handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
+ guint8 * data, gsize size)
+{
+ GstMIKEYMessage *msg;
+ guint i, n_cs;
+ GstCaps *caps = NULL;
+ GstMIKEYPayloadKEMAC *kemac;
+ const GstMIKEYPayloadKeyData *pkd;
+ GstBuffer *key;
+
+ /* the MIKEY message contains a CSB or crypto session bundle. It is a
+ * set of Crypto Sessions protected with the same master key.
+ * In the context of SRTP, an RTP and its RTCP stream is part of a
+ * crypto session */
+ if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
+ goto parse_failed;
+
+ /* we can only handle SRTP crypto sessions for now */
+ if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
+ goto invalid_map_type;
+
+ /* get the number of crypto sessions. This maps SSRC to its
+ * security parameters */
+ n_cs = gst_mikey_message_get_n_cs (msg);
+ if (n_cs == 0)
+ goto no_crypto_sessions;
+
+ /* we also need keys */
+ if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
+ (msg, GST_MIKEY_PT_KEMAC, 0)))
+ goto no_keys;
+
+ /* we don't support encrypted keys */
+ if (kemac->enc_alg != GST_MIKEY_ENC_NULL
+ || kemac->mac_alg != GST_MIKEY_MAC_NULL)
+ goto unsupported_encryption;
+
+ /* get Key data sub-payload */
+ pkd = (const GstMIKEYPayloadKeyData *)
+ gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
+
+ key =
+ gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
+ pkd->key_len);
+
+ /* go over all crypto sessions and create the security policy for each
+ * SSRC */
+ for (i = 0; i < n_cs; i++) {
+ const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
+
+ caps = gst_caps_new_simple ("application/x-srtp",
+ "ssrc", G_TYPE_UINT, map->ssrc,
+ "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
+ mikey_apply_policy (caps, msg, map->policy);
+
+ gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
+ gst_caps_unref (caps);
+ }
+ gst_mikey_message_unref (msg);
+ gst_buffer_unref (key);
- if (session) {
- g_object_ref (session);
- /* get a handle to the configuration of the media in the session, this can
- * return NULL if this is a new url to manage in this session. */
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
+ return FALSE;
+ }
+invalid_map_type:
+ {
+ GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
+ goto cleanup_message;
+ }
+no_crypto_sessions:
+ {
+ GST_DEBUG_OBJECT (client, "no crypto sessions");
+ goto cleanup_message;
+ }
+no_keys:
+ {
+ GST_DEBUG_OBJECT (client, "no keys found");
+ goto cleanup_message;
+ }
+unsupported_encryption:
+ {
+ GST_DEBUG_OBJECT (client, "unsupported key encryption");
+ goto cleanup_message;
+ }
+cleanup_message:
+ {
+ gst_mikey_message_unref (msg);
+ return FALSE;
+ }
+}
+
+#define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
+
+static void
+strip_chars (gchar * str)
+{
+ gchar *s;
+ gsize len;
+
+ len = strlen (str);
+ while (len--) {
+ if (!IS_STRIP_CHAR (str[len]))
+ break;
+ str[len] = '\0';
+ }
+ for (s = str; *s && IS_STRIP_CHAR (*s); s++);
+ memmove (str, s, len + 1);
+}
+
+/* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
+ * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
+ */
+static gboolean
+handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
+{
+ gchar **specs;
+ gint i, j;
+
+ specs = g_strsplit (keymgmt, ",", 0);
+ for (i = 0; specs[i]; i++) {
+ gchar **split;
+
+ split = g_strsplit (specs[i], ";", 0);
+ for (j = 0; split[j]; j++) {
+ g_strstrip (split[j]);
+ if (g_str_has_prefix (split[j], "prot=")) {
+ g_strstrip (split[j] + 5);
+ if (!g_str_equal (split[j] + 5, "mikey"))
+ break;
+ GST_DEBUG ("found mikey");
+ } else if (g_str_has_prefix (split[j], "uri=")) {
+ strip_chars (split[j] + 4);
+ GST_DEBUG ("found uri '%s'", split[j] + 4);
+ } else if (g_str_has_prefix (split[j], "data=")) {
+ guchar *data;
+ gsize size;
+ strip_chars (split[j] + 5);
+ GST_DEBUG ("found data '%s'", split[j] + 5);
+ data = g_base64_decode_inplace (split[j] + 5, &size);
+ handle_mikey_data (client, ctx, data, size);
+ }
+ }
+ g_strfreev (split);
+ }
+ g_strfreev (specs);
+ return TRUE;
+}
+
+static gboolean
+handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult res;
+ GstRTSPUrl *uri;
+ gchar *transport, *keymgmt;
+ GstRTSPTransport *ct, *st;
+ GstRTSPStatusCode code;
+ GstRTSPSession *session;
+ GstRTSPStreamTransport *trans;
+ gchar *trans_str;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPStream *stream;
+ GstRTSPState rtspstate;
+ GstRTSPClientClass *klass;
+ gchar *path, *control = NULL;
+ gint matched;
+ gboolean new_session = FALSE;
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ uri = ctx->uri;
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
+
+ /* parse the transport */
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
+ &transport, 0);
+ if (res != GST_RTSP_OK)
+ goto no_transport;
+
+ /* we create the session after parsing stuff so that we don't make
+ * a session for malformed requests */
+ if (priv->session_pool == NULL)
+ goto no_pool;
+
+ session = ctx->session;
+
+ if (session) {
+ g_object_ref (session);
+ /* get a handle to the configuration of the media in the session, this can
+ * return NULL if this is a new url to manage in this session. */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
} else {
/* we need a new media configuration in this session */
if (media == NULL)
goto media_not_found_no_reply;
- if (path[matched] == '\0')
- goto control_not_found;
+ if (path[matched] == '\0') {
+ if (gst_rtsp_media_n_streams (media) == 1) {
+ stream = gst_rtsp_media_get_stream (media, 0);
+ } else {
+ goto control_not_found;
+ }
+ } else {
+ /* path is what matched. */
+ path[matched] = '\0';
+ /* control is remainder */
+ control = &path[matched + 1];
- /* path is what matched. We can modify the parsed uri in place */
- path[matched] = '\0';
- /* control is remainder */
- control = &path[matched + 1];
+ /* find the stream now using the control part */
+ stream = gst_rtsp_media_find_stream (media, control);
+ }
- /* find the stream now using the control part */
- stream = gst_rtsp_media_find_stream (media, control);
if (stream == NULL)
goto stream_not_found;
/* make sure this client is closed when the session is closed */
client_watch_session (client, session);
+ new_session = TRUE;
/* signal new session */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
session);
ctx->session = session;
}
+ if (!klass->configure_client_media (client, media, stream, ctx))
+ goto configure_media_failed_no_reply;
+
+ gst_rtsp_transport_new (&ct);
+
+ /* parse and find a usable supported transport */
+ if (!parse_transport (transport, stream, ct))
+ goto unsupported_transports;
+
+ if ((ct->mode_play
+ && !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY)) || (ct->mode_record
+ && !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD)))
+ goto unsupported_mode;
+
+ /* parse the keymgmt */
+ if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
+ &keymgmt, 0) == GST_RTSP_OK) {
+ if (!handle_keymgmt (client, ctx, keymgmt))
+ goto keymgmt_error;
+ }
+
if (sessmedia == NULL) {
/* manage the media in our session now, if not done already */
sessmedia = gst_rtsp_session_manage_media (session, path, media);
/* if we stil have no media, error */
if (sessmedia == NULL)
goto sessmedia_unavailable;
+
+ /* don't cache media anymore */
+ clean_cached_media (client, FALSE);
} else {
g_object_unref (media);
}
ctx->sessmedia = sessmedia;
- /* set blocksize on this stream */
- if (!handle_blocksize (media, stream, ctx->request))
- goto invalid_blocksize;
-
- gst_rtsp_transport_new (&ct);
-
- /* our supported transports */
- supported = gst_rtsp_stream_get_protocols (stream);
-
- /* parse and find a usable supported transport */
- if (!parse_transport (transport, supported, ct))
- goto unsupported_transports;
-
/* update the client transport */
- klass = GST_RTSP_CLIENT_GET_CLASS (client);
if (!klass->configure_client_transport (client, ctx, ct))
goto unsupported_client_transport;
/* set in the session media transport */
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+ ctx->trans = trans;
+
+ /* configure the url used to set this transport, this we will use when
+ * generating the response for the PLAY request */
+ gst_rtsp_stream_transport_set_url (trans, uri);
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ /* our callbacks to send data on this TCP connection */
+ gst_rtsp_stream_transport_set_callbacks (trans,
+ (GstRTSPSendFunc) do_send_data,
+ (GstRTSPSendFunc) do_send_data, client, NULL);
+
+ g_hash_table_insert (priv->transports,
+ GINT_TO_POINTER (ct->interleaved.min), trans);
+ g_object_ref (trans);
+ g_hash_table_insert (priv->transports,
+ GINT_TO_POINTER (ct->interleaved.max), trans);
+ g_object_ref (trans);
+ }
+
/* create and serialize the server transport */
- st = make_server_transport (client, ctx, ct);
+ st = make_server_transport (client, media, ctx, ct);
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
trans_str);
g_free (trans_str);
- send_message (client, session, ctx->response, FALSE);
+ send_message (client, ctx, ctx->response, FALSE);
/* update the state */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
{
GST_ERROR ("client %p: no transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
- g_free (path);
- return FALSE;
+ goto cleanup_path;
}
no_pool:
{
GST_ERROR ("client %p: no session pool configured", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
- g_free (path);
- return FALSE;
+ goto cleanup_path;
}
media_not_found_no_reply:
{
GST_ERROR ("client %p: media '%s' not found", client, path);
- g_free (path);
/* error reply is already sent */
- return FALSE;
+ goto cleanup_path;
}
media_not_found:
{
GST_ERROR ("client %p: media '%s' not found", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
- g_free (path);
- return FALSE;
+ goto cleanup_path;
}
control_not_found:
{
GST_ERROR ("client %p: no control in path '%s'", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_object_unref (media);
- g_free (path);
- return FALSE;
+ goto cleanup_path;
}
stream_not_found:
{
- GST_ERROR ("client %p: stream '%s' not found", client, control);
+ GST_ERROR ("client %p: stream '%s' not found", client,
+ GST_STR_NULL (control));
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_object_unref (media);
- g_free (path);
- return FALSE;
+ goto cleanup_path;
}
service_unavailable:
{
GST_ERROR ("client %p: can't create session", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
- g_free (path);
- return FALSE;
+ goto cleanup_path;
}
sessmedia_unavailable:
{
GST_ERROR ("client %p: can't create session media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
- g_object_unref (session);
- g_free (path);
- return FALSE;
+ goto cleanup_session;
}
-invalid_blocksize:
+configure_media_failed_no_reply:
{
- GST_ERROR ("client %p: invalid blocksize", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
- g_object_unref (session);
- g_free (path);
- return FALSE;
+ GST_ERROR ("client %p: configure_media failed", client);
+ /* error reply is already sent */
+ goto cleanup_session;
}
unsupported_transports:
{
GST_ERROR ("client %p: unsupported transports", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
- gst_rtsp_transport_free (ct);
- g_object_unref (session);
- g_free (path);
- return FALSE;
+ goto cleanup_transport;
}
unsupported_client_transport:
{
GST_ERROR ("client %p: unsupported client transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_transport;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: unsupported mode (media play: %d, media record: %d, "
+ "mode play: %d, mode record: %d)", client,
+ ! !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY),
+ ! !(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD), ct->mode_play, ct->mode_record);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+ goto cleanup_transport;
+ }
+keymgmt_error:
+ {
+ GST_ERROR ("client %p: keymgmt error", client);
+ send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
+ goto cleanup_transport;
+ }
+ {
+ cleanup_transport:
gst_rtsp_transport_free (ct);
+ cleanup_session:
+ if (new_session)
+ gst_rtsp_session_pool_remove (priv->session_pool, session);
g_object_unref (session);
+ cleanup_path:
g_free (path);
return FALSE;
}
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
+ guint64 session_id_tmp;
+ gchar session_id[21];
gst_sdp_message_new (&sdp);
else
proto = "IP4";
- gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
+ session_id_tmp = (((guint64) g_random_int ()) << 32) | g_random_int ();
+ g_snprintf (session_id, sizeof (session_id), "%" G_GUINT64_FORMAT,
+ session_id_tmp);
+
+ gst_sdp_message_set_origin (sdp, "-", session_id, "1", "IN", proto,
priv->server_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
info.server_ip = priv->server_ip;
/* create an SDP for the media object */
- if (!gst_rtsp_sdp_from_media (sdp, &info, media))
+ if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
goto no_sdp;
return sdp;
if (!ctx->uri)
goto no_uri;
- /* check what kind of format is accepted, we don't really do anything with it
- * and always return SDP for now. */
- for (i = 0; i++;) {
- gchar *accept;
+ /* check what kind of format is accepted, we don't really do anything with it
+ * and always return SDP for now. */
+ for (i = 0;; i++) {
+ gchar *accept;
+
+ res =
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
+ &accept, i);
+ if (res == GST_RTSP_ENOTIMPL)
+ break;
+
+ if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
+ break;
+ }
+
+ if (!priv->mount_points)
+ goto no_mount_points;
+
+ if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+ goto no_path;
+
+ /* find the media object for the uri */
+ if (!(media = find_media (client, ctx, path, NULL)))
+ goto no_media;
+
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_PLAY))
+ goto unsupported_mode;
+
+ /* create an SDP for the media object on this client */
+ if (!(sdp = klass->create_sdp (client, media)))
+ goto no_sdp;
+
+ /* we suspend after the describe */
+ gst_rtsp_media_suspend (media);
+ g_object_unref (media);
+
+ 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 (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
+
+ /* content base for some clients that might screw up creating the setup uri */
+ str = make_base_url (client, ctx->uri, path);
+ g_free (path);
+
+ GST_INFO ("adding content-base: %s", str);
+ gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
+
+ /* add SDP to the response body */
+ str = gst_sdp_message_as_text (sdp);
+ gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
+ gst_sdp_message_free (sdp);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
+ 0, ctx);
+
+ return TRUE;
+
+ /* ERRORS */
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_mount_points:
+ {
+ GST_ERROR ("client %p: no mount points configured", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_path:
+ {
+ GST_ERROR ("client %p: can't find path for url", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_media:
+ {
+ GST_ERROR ("client %p: no media", client);
+ g_free (path);
+ /* error reply is already sent */
+ return FALSE;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: media does not support DESCRIBE", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+ g_free (path);
+ g_object_unref (media);
+ return FALSE;
+ }
+no_sdp:
+ {
+ GST_ERROR ("client %p: can't create SDP", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
+ g_object_unref (media);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMedia * media,
+ GstSDPMessage * sdp)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPThread *thread;
+
+ /* create an SDP for the media object */
+ if (!gst_rtsp_media_handle_sdp (media, sdp))
+ goto unhandled_sdp;
+
+ thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+ GST_RTSP_THREAD_TYPE_MEDIA, ctx);
+ if (thread == NULL)
+ goto no_thread;
+
+ /* prepare the media */
+ if (!gst_rtsp_media_prepare (media, thread))
+ goto no_prepare;
+
+ return TRUE;
+
+ /* ERRORS */
+unhandled_sdp:
+ {
+ GST_ERROR ("client %p: could not handle SDP", client);
+ return FALSE;
+ }
+no_thread:
+ {
+ GST_ERROR ("client %p: can't create thread", client);
+ return FALSE;
+ }
+no_prepare:
+ {
+ GST_ERROR ("client %p: can't prepare media", client);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPClientClass *klass;
+ GstSDPResult sres;
+ GstSDPMessage *sdp;
+ GstRTSPMedia *media;
+ gchar *path, *cont = NULL;
+ guint8 *data;
+ guint size;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (!ctx->uri)
+ goto no_uri;
+
+ if (!priv->mount_points)
+ goto no_mount_points;
+
+ /* check if reply is SDP */
+ gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_CONTENT_TYPE, &cont,
+ 0);
+ /* could not be set but since the request returned OK, we assume it
+ * was SDP, else check it. */
+ if (cont) {
+ if (g_ascii_strcasecmp (cont, "application/sdp") != 0)
+ goto wrong_content_type;
+ }
+
+ /* get message body and parse as SDP */
+ gst_rtsp_message_get_body (ctx->request, &data, &size);
+ if (data == NULL || size == 0)
+ goto no_message;
+
+ GST_DEBUG ("client %p: parse SDP...", client);
+ gst_sdp_message_new (&sdp);
+ sres = gst_sdp_message_parse_buffer (data, size, sdp);
+ if (sres != GST_SDP_OK)
+ goto sdp_parse_failed;
+
+ if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+ goto no_path;
+
+ /* find the media object for the uri */
+ if (!(media = find_media (client, ctx, path, NULL)))
+ goto no_media;
+
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD))
+ goto unsupported_mode;
+
+ /* Tell client subclass about the media */
+ if (!klass->handle_sdp (client, ctx, media, sdp))
+ goto unhandled_sdp;
+
+ /* we suspend after the announce */
+ gst_rtsp_media_suspend (media);
+ g_object_unref (media);
+
+ gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+ send_message (client, ctx, ctx->response, FALSE);
+
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],
+ 0, ctx);
+
+ gst_sdp_message_free (sdp);
+ g_free (path);
+ return TRUE;
+
+no_uri:
+ {
+ GST_ERROR ("client %p: no uri", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_mount_points:
+ {
+ GST_ERROR ("client %p: no mount points configured", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ return FALSE;
+ }
+no_path:
+ {
+ GST_ERROR ("client %p: can't find path for url", client);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+wrong_content_type:
+ {
+ GST_ERROR ("client %p: unknown content type", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+no_message:
+ {
+ GST_ERROR ("client %p: can't find SDP message", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
+ }
+sdp_parse_failed:
+ {
+ GST_ERROR ("client %p: failed to parse SDP message", client);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+no_media:
+ {
+ GST_ERROR ("client %p: no media", client);
+ g_free (path);
+ /* error reply is already sent */
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+unsupported_mode:
+ {
+ GST_ERROR ("client %p: media does not support ANNOUNCE", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
+ g_free (path);
+ g_object_unref (media);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+unhandled_sdp:
+ {
+ GST_ERROR ("client %p: can't handle SDP", client);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
+ g_free (path);
+ g_object_unref (media);
+ gst_sdp_message_free (sdp);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+ GstRTSPSession *session;
+ GstRTSPClientClass *klass;
+ GstRTSPSessionMedia *sessmedia;
+ GstRTSPMedia *media;
+ GstRTSPUrl *uri;
+ GstRTSPState rtspstate;
+ gchar *path;
+ gint matched;
+
+ if (!(session = ctx->session))
+ goto no_session;
+
+ if (!(uri = ctx->uri))
+ goto no_uri;
+
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+ path = klass->make_path_from_uri (client, uri);
- res =
- gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
- &accept, i);
- if (res == GST_RTSP_ENOTIMPL)
- break;
+ /* get a handle to the configuration of the media in the session */
+ sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+ if (!sessmedia)
+ goto not_found;
- if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
- break;
- }
+ if (path[matched] != '\0')
+ goto no_aggregate;
- if (!priv->mount_points)
- goto no_mount_points;
+ g_free (path);
- if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
- goto no_path;
+ ctx->sessmedia = sessmedia;
+ ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
- /* find the media object for the uri */
- if (!(media = find_media (client, ctx, path, NULL)))
- goto no_media;
+ if (!(gst_rtsp_media_get_transport_mode (media) &
+ GST_RTSP_TRANSPORT_MODE_RECORD))
+ goto unsupported_mode;
- /* create an SDP for the media object on this client */
- if (!(sdp = klass->create_sdp (client, media)))
- goto no_sdp;
+ /* the session state must be playing or ready */
+ rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+ if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
+ goto invalid_state;
- g_object_unref (media);
+ /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
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 (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
- "application/sdp");
-
- /* content base for some clients that might screw up creating the setup uri */
- str = make_base_url (client, ctx->uri, path);
- g_free (path);
-
- GST_INFO ("adding content-base: %s", str);
- gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
+ send_message (client, ctx, ctx->response, FALSE);
- /* add SDP to the response body */
- str = gst_sdp_message_as_text (sdp);
- gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
- gst_sdp_message_free (sdp);
+ /* start playing after sending the response */
+ gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
- send_message (client, ctx->session, ctx->response, FALSE);
+ gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
- g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
- 0, ctx);
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST], 0,
+ ctx);
return TRUE;
/* ERRORS */
+no_session:
+ {
+ GST_ERROR ("client %p: no session", client);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+ return FALSE;
+ }
no_uri:
{
- GST_ERROR ("client %p: no uri", client);
+ GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
-no_mount_points:
+not_found:
{
- GST_ERROR ("client %p: no mount points configured", client);
+ GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
-no_path:
+no_aggregate:
{
- GST_ERROR ("client %p: can't find path for url", client);
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+ GST_ERROR ("client %p: no aggregate path %s", client, path);
+ send_generic_response (client,
+ GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+ g_free (path);
return FALSE;
}
-no_media:
+unsupported_mode:
{
- GST_ERROR ("client %p: no media", client);
- g_free (path);
- /* error reply is already sent */
+ GST_ERROR ("client %p: media does not support RECORD", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
return FALSE;
}
-no_sdp:
+invalid_state:
{
- GST_ERROR ("client %p: can't create SDP", client);
+ GST_ERROR ("client %p: not PLAYING or READY", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ return FALSE;
+ }
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
- g_free (path);
- g_object_unref (media);
return FALSE;
}
}
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
- send_message (client, ctx->session, ctx->response, FALSE);
+ send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
0, ctx);
*d = '\0';
}
+/* is called when the session is removed from its session pool. */
static void
-client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
+client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
+ GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
- GST_INFO ("client %p: session %p finished", client, session);
+ GST_INFO ("client %p: session %p removed", client, session);
- /* unlink all media managed in this session */
- client_unlink_session (client, session);
+ g_mutex_lock (&priv->lock);
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
+ client_unwatch_session (client, session, NULL);
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
+ g_mutex_unlock (&priv->lock);
+}
- /* remove the session */
- if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
- GST_INFO ("client %p: all sessions finalized, close the connection",
- client);
- close_connection (client);
+/* Returns TRUE if there are no Require headers, otherwise returns FALSE
+ * and also returns a newly-allocated string of (comma-separated) unsupported
+ * options in the unsupported_reqs variable .
+ *
+ * There may be multiple Require headers, but we must send one single
+ * Unsupported header with all the unsupported options as response. If
+ * an incoming Require header contained a comma-separated list of options
+ * GstRtspConnection will already have split that list up into multiple
+ * headers.
+ *
+ * TODO: allow the application to decide what features are supported
+ */
+static gboolean
+check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
+{
+ GstRTSPResult res;
+ GPtrArray *arr = NULL;
+ gchar *reqs = NULL;
+ gint i;
+
+ i = 0;
+ do {
+ res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
+
+ if (res == GST_RTSP_ENOTIMPL)
+ break;
+
+ if (arr == NULL)
+ arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
+
+ g_ptr_array_add (arr, g_strdup (reqs));
}
+ while (TRUE);
+
+ /* if we don't have any Require headers at all, all is fine */
+ if (i == 1)
+ return TRUE;
+
+ /* otherwise we've now processed at all the Require headers */
+ g_ptr_array_add (arr, NULL);
+
+ /* for now we don't commit to supporting anything, so will just report
+ * all of the required options as unsupported */
+ *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
+
+ g_ptr_array_unref (arr);
+ return FALSE;
}
static void
GstRTSPSession *session = NULL;
GstRTSPContext sctx = { NULL }, *ctx;
GstRTSPMessage response = { 0 };
+ gchar *unsupported_reqs = NULL;
gchar *sessid;
if (!(ctx = gst_rtsp_context_get_current ())) {
gst_rtsp_message_dump (request);
}
- GST_INFO ("client %p: received a request", client);
-
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
+ GST_INFO ("client %p: received a request %s %s %s", client,
+ gst_rtsp_method_as_text (method), uristr,
+ gst_rtsp_version_as_text (version));
+
/* we can only handle 1.0 requests */
if (version != GST_RTSP_VERSION_1_0)
goto not_supported;
/* we always try to parse the url first */
if (strcmp (uristr, "*") == 0) {
/* special case where we have * as uri, keep uri = NULL */
- } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK)
- goto bad_request;
+ } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ /* check if the uristr is an absolute path <=> scheme and host information
+ * is missing */
+ gchar *scheme;
+
+ scheme = g_uri_parse_scheme (uristr);
+ if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
+ gchar *absolute_uristr = NULL;
+
+ GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
+ if (priv->server_ip == NULL) {
+ GST_WARNING_OBJECT (client, "host information missing");
+ goto bad_request;
+ }
+
+ absolute_uristr =
+ g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
+
+ GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
+ if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
+ g_free (absolute_uristr);
+ goto bad_request;
+ }
+ g_free (absolute_uristr);
+ } else {
+ g_free (scheme);
+ goto bad_request;
+ }
+ }
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
goto not_authorized;
+ /* handle any 'Require' headers */
+ if (!check_request_requirements (ctx->request, &unsupported_reqs))
+ goto unsupported_requirement;
+
+ /* the backlog must be unlimited while processing requests.
+ * the causes of this are two cases of deadlocks while streaming over TCP:
+ *
+ * 1. consider the scenario where the media pipeline's streaming thread
+ * is blocking in the appsink (taking the appsink's preroll lock) because
+ * the backlog is full. when a PAUSE request is received by the RTSP
+ * client thread then the the state of the session media ought to change
+ * to PAUSED. while most elements in the pipeline can change state this
+ * can never happen for the appsink since its preroll lock is taken by
+ * another thread.
+ *
+ * 2. consider the scenario where the media pipeline's streaming thread
+ * is blocking in the appsink new_sample callback (taking the send lock
+ * in RTSP client) because the backlog is full. when e.g. a GET request
+ * is received by the RTSP client thread then a response ought to be sent
+ * but this can never happen since it requires taking the send lock
+ * already taken by another thread.
+ *
+ * the reason that the backlog is never emptied is that the source used
+ * for dequeing messages from the backlog is never dispatched because it
+ * is attached to the same mainloop as the source receving RTSP requests and
+ * therefore run by the RTSP client thread which is alreayd blocking.
+ *
+ * without significant changes the easiest way to cope with this is to
+ * not block indefinitely when the backlog is full, but rather let the
+ * backlog grow in size. this in effect means that there can not be any
+ * upper boundary on its size.
+ */
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
+
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
handle_get_param_request (client, ctx);
break;
case GST_RTSP_ANNOUNCE:
+ handle_announce_request (client, ctx);
+ break;
case GST_RTSP_RECORD:
+ handle_record_request (client, ctx);
+ break;
case GST_RTSP_REDIRECT:
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
goto not_implemented;
case GST_RTSP_INVALID:
default:
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
goto bad_request;
}
+ if (priv->watch != NULL)
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
+
done:
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
/* error reply is already sent */
goto done;
}
+unsupported_requirement:
+ {
+ GST_ERROR ("client %p: Required option is not supported (%s)", client,
+ unsupported_reqs);
+ send_option_not_supported_response (client, ctx, unsupported_reqs);
+ g_free (unsupported_reqs);
+ goto done;
+ }
not_implemented:
{
GST_ERROR ("client %p: method %d not implemented", client, method);
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
guint8 channel;
- GList *walk;
guint8 *data;
guint size;
GstBuffer *buffer;
- gboolean handled;
+ GstRTSPStreamTransport *trans;
/* find the stream for this message */
res = gst_rtsp_message_parse_data (message, &channel);
if (res != GST_RTSP_OK)
return;
+ gst_rtsp_message_get_body (message, &data, &size);
+ if (size < 2)
+ goto invalid_length;
+
gst_rtsp_message_steal_body (message, &data, &size);
- buffer = gst_buffer_new_wrapped (data, size);
+ /* Strip trailing \0 (which GstRTSPConnection adds) */
+ --size;
- handled = FALSE;
- for (walk = priv->transports; walk; walk = g_list_next (walk)) {
- GstRTSPStreamTransport *trans;
- GstRTSPStream *stream;
- const GstRTSPTransport *tr;
+ buffer = gst_buffer_new_wrapped (data, size);
- trans = walk->data;
+ trans =
+ g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
+ if (trans) {
+ /* dispatch to the stream based on the channel number */
+ GST_LOG_OBJECT (client, "%u bytes of data on channel %u", size, channel);
+ gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
+ } else {
+ GST_DEBUG_OBJECT (client, "received %u bytes of data for "
+ "unknown channel %u", size, channel);
+ gst_buffer_unref (buffer);
+ }
- tr = gst_rtsp_stream_transport_get_transport (trans);
- stream = gst_rtsp_stream_transport_get_stream (trans);
+ return;
- /* check for TCP transport */
- if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
- /* dispatch to the stream based on the channel number */
- if (tr->interleaved.min == channel) {
- gst_rtsp_stream_recv_rtp (stream, buffer);
- handled = TRUE;
- break;
- } else if (tr->interleaved.max == channel) {
- gst_rtsp_stream_recv_rtcp (stream, buffer);
- handled = TRUE;
- break;
- }
- }
+/* ERRORS */
+invalid_length:
+ {
+ GST_DEBUG ("client %p: Short message received, ignoring", client);
+ return;
}
- if (!handled)
- gst_buffer_unref (buffer);
}
/**
* gst_rtsp_client_set_session_pool:
* @client: a #GstRTSPClient
- * @pool: a #GstRTSPSessionPool
+ * @pool: (transfer none): a #GstRTSPSessionPool
*
* Set @pool as the sessionpool for @client which it will use to find
* or allocate sessions. the sessionpool is usually inherited from the server
g_mutex_lock (&priv->lock);
old = priv->session_pool;
priv->session_pool = pool;
+
+ if (priv->session_removed_id) {
+ g_signal_handler_disconnect (old, priv->session_removed_id);
+ priv->session_removed_id = 0;
+ }
g_mutex_unlock (&priv->lock);
+ /* FIXME, should remove all sessions from the old pool for this client */
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_set_mount_points:
* @client: a #GstRTSPClient
- * @mounts: a #GstRTSPMountPoints
+ * @mounts: (transfer none): a #GstRTSPMountPoints
*
* Set @mounts as the mount points for @client which it will use to map urls
* to media streams. These mount points are usually inherited from the server that
/**
* gst_rtsp_client_set_auth:
* @client: a #GstRTSPClient
- * @auth: a #GstRTSPAuth
+ * @auth: (transfer none): a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @client.
*/
/**
* gst_rtsp_client_set_thread_pool:
* @client: a #GstRTSPClient
- * @pool: a #GstRTSPThreadPool
+ * @pool: (transfer none): a #GstRTSPThreadPool
*
* configure @pool to be used as the thread pool of @client.
*/
/**
* gst_rtsp_client_set_send_func:
* @client: a #GstRTSPClient
- * @func: a #GstRTSPClientSendFunc
- * @user_data: user data passed to @func
- * @notify: called when @user_data is no longer in use
+ * @func: (scope notified): a #GstRTSPClientSendFunc
+ * @user_data: (closure): user data passed to @func
+ * @notify: (allow-none): called when @user_data is no longer in use
*
* Set @func as the callback that will be called when a new message needs to be
* sent to the client. @user_data is passed to @func and @notify is called when
/**
* gst_rtsp_client_handle_message:
* @client: a #GstRTSPClient
- * @message: an #GstRTSPMessage
+ * @message: (transfer none): an #GstRTSPMessage
*
* Let the client handle @message.
*
/**
* gst_rtsp_client_send_message:
* @client: a #GstRTSPClient
- * @session: a #GstRTSPSession to send the message to or %NULL
- * @message: The #GstRTSPMessage to send
+ * @session: (allow-none) (transfer none): a #GstRTSPSession to send
+ * the message to or %NULL
+ * @message: (transfer none): The #GstRTSPMessage to send
*
* Send a message message to the remote end. @message must be a
* #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPMessage * message)
{
+ GstRTSPContext sctx = { NULL }
+ , *ctx;
+ GstRTSPClientPrivate *priv;
+
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
- send_message (client, session, message, FALSE);
+ priv = client->priv;
+
+ if (!(ctx = gst_rtsp_context_get_current ())) {
+ ctx = &sctx;
+ ctx->auth = priv->auth;
+ gst_rtsp_context_push_current (ctx);
+ }
+
+ ctx->conn = priv->connection;
+ ctx->client = client;
+ ctx->session = session;
+
+ send_message (client, ctx, message, FALSE);
+
+ if (ctx == &sctx)
+ gst_rtsp_context_pop_current (ctx);
return GST_RTSP_OK;
}
gboolean close, gpointer user_data)
{
GstRTSPClientPrivate *priv = client->priv;
+ GstRTSPResult ret;
+ GTimeVal time;
+
+ time.tv_sec = 1;
+ time.tv_usec = 0;
+
+ do {
+ /* send the response and store the seq number so we can wait until it's
+ * written to the client to close the connection */
+ ret =
+ gst_rtsp_watch_send_message (priv->watch, message,
+ close ? &priv->close_seq : NULL);
+ if (ret == GST_RTSP_OK)
+ break;
+
+ if (ret != GST_RTSP_ENOMEM)
+ goto error;
+
+ /* drop backlog */
+ if (priv->drop_backlog)
+ break;
+
+ /* queue was full, wait for more space */
+ GST_DEBUG_OBJECT (client, "waiting for backlog");
+ ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
+ GST_DEBUG_OBJECT (client, "Resend due to backlog full");
+ } while (ret != GST_RTSP_EINTR);
- /* send the response and store the seq number so we can wait until it's
- * written to the client to close the connection */
- return gst_rtsp_watch_send_message (priv->watch, message, close ?
- &priv->close_seq : NULL);
+ return ret;
+
+ /* ERRORS */
+error:
+ {
+ GST_DEBUG_OBJECT (client, "got error %d", ret);
+ return ret;
+ }
}
static GstRTSPResult
GstRTSPClientPrivate *priv = client->priv;
if (priv->close_seq && priv->close_seq == cseq) {
+ GST_INFO ("client %p: send close message", client);
priv->close_seq = 0;
- close_connection (client);
+ gst_rtsp_client_close (client);
}
return GST_RTSP_OK;
g_mutex_unlock (&tunnels_lock);
}
+ gst_rtsp_watch_set_flushing (watch, TRUE);
+ g_mutex_lock (&priv->watch_lock);
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ g_mutex_unlock (&priv->watch_lock);
return GST_RTSP_OK;
}
}
}
-static GstRTSPStatusCode
-tunnel_start (GstRTSPWatch * watch, gpointer user_data)
-{
- GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
- GstRTSPClientPrivate *priv = client->priv;
-
- GST_INFO ("client %p: tunnel start (connection %p)", client,
- priv->connection);
-
- if (!remember_tunnel (client))
- goto tunnel_error;
-
- return GST_RTSP_STS_OK;
-
- /* ERRORS */
-tunnel_error:
- {
- GST_ERROR ("client %p: error starting tunnel", client);
- return GST_RTSP_STS_SERVICE_UNAVAILABLE;
- }
-}
-
static GstRTSPResult
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
{
return GST_RTSP_OK;
}
-static GstRTSPResult
-tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
+static gboolean
+handle_tunnel (GstRTSPClient * client)
{
- const gchar *tunnelid;
- GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClientPrivate *priv = client->priv;
GstRTSPClient *oclient;
GstRTSPClientPrivate *opriv;
+ const gchar *tunnelid;
- GST_INFO ("client %p: tunnel complete", client);
-
- /* find previous tunnel */
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
+ /* check for previous tunnel */
g_mutex_lock (&tunnels_lock);
- if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
- goto no_tunnel;
+ oclient = g_hash_table_lookup (tunnels, tunnelid);
- /* remove the old client from the table. ref before because removing it will
- * remove the ref to it. */
- g_object_ref (oclient);
- g_hash_table_remove (tunnels, tunnelid);
+ if (oclient == NULL) {
+ /* no previous tunnel, remember tunnel */
+ g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+ g_mutex_unlock (&tunnels_lock);
- opriv = oclient->priv;
+ GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
+ client, priv->connection);
+ } else {
+ /* merge both tunnels into the first client */
+ /* remove the old client from the table. ref before because removing it will
+ * remove the ref to it. */
+ g_object_ref (oclient);
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
- if (opriv->watch == NULL)
- goto tunnel_closed;
- g_mutex_unlock (&tunnels_lock);
+ opriv = oclient->priv;
- GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
- opriv->connection, priv->connection);
+ g_mutex_lock (&opriv->watch_lock);
+ if (opriv->watch == NULL)
+ goto tunnel_closed;
- /* merge the tunnels into the first client */
- gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
- gst_rtsp_watch_reset (opriv->watch);
- g_object_unref (oclient);
+ GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
+ oclient, opriv->connection, priv->connection);
- return GST_RTSP_OK;
+ gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
+ gst_rtsp_watch_reset (priv->watch);
+ gst_rtsp_watch_reset (opriv->watch);
+ g_mutex_unlock (&opriv->watch_lock);
+ g_object_unref (oclient);
+
+ /* the old client owns the tunnel now, the new one will be freed */
+ g_source_destroy ((GSource *) priv->watch);
+ priv->watch = NULL;
+ gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+ }
+
+ return TRUE;
/* ERRORS */
no_tunnelid:
{
GST_ERROR ("client %p: no tunnelid provided", client);
- return GST_RTSP_ERROR;
- }
-no_tunnel:
- {
- g_mutex_unlock (&tunnels_lock);
- GST_ERROR ("client %p: tunnel session %s not found", client, tunnelid);
- return GST_RTSP_ERROR;
+ return FALSE;
}
tunnel_closed:
{
- g_mutex_unlock (&tunnels_lock);
GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
+ g_mutex_unlock (&opriv->watch_lock);
g_object_unref (oclient);
+ return FALSE;
+ }
+}
+
+static GstRTSPStatusCode
+tunnel_get (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel get (connection %p)", client,
+ client->priv->connection);
+
+ if (!handle_tunnel (client)) {
+ return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+ }
+
+ return GST_RTSP_STS_OK;
+}
+
+static GstRTSPResult
+tunnel_post (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel post (connection %p)", client,
+ client->priv->connection);
+
+ if (!handle_tunnel (client)) {
return GST_RTSP_ERROR;
}
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
+ GstRTSPMessage * response, gpointer user_data)
+{
+ GstRTSPClientClass *klass;
+
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+ if (klass->tunnel_http_response) {
+ klass->tunnel_http_response (client, request, response);
+ }
+
+ return GST_RTSP_OK;
}
static GstRTSPWatchFuncs watch_funcs = {
message_sent,
closed,
error,
- tunnel_start,
- tunnel_complete,
+ tunnel_get,
+ tunnel_post,
error_full,
- tunnel_lost
+ tunnel_lost,
+ tunnel_http_response
};
static void
GST_INFO ("client %p: watch destroyed", client);
priv->watch = NULL;
+ /* remove all sessions and so drop the extra client ref */
+ gst_rtsp_client_session_filter (client, cleanup_session, NULL);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
* @context: (allow-none): a #GMainContext
*
* Attaches @client to @context. When the mainloop for @context is run, the
- * client will be dispatched. When @context is NULL, the default context will be
+ * client will be dispatched. When @context is %NULL, the default context will be
* used).
*
* This function should be called when the client properties and urls are fully
g_return_val_if_fail (priv->connection != NULL, 0);
g_return_val_if_fail (priv->watch == NULL, 0);
+ /* make sure noone will free the context before the watch is destroyed */
+ priv->watch_context = g_main_context_ref (context);
+
/* create watch for the connection and attach */
priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
g_object_ref (client), (GDestroyNotify) client_watch_notify);
gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
(GDestroyNotify) gst_rtsp_watch_unref);
- /* FIXME make this configurable. We don't want to do this yet because it will
- * be superceeded by a cache object later */
- gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
+ gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
- GST_INFO ("attaching to context %p", context);
+ GST_INFO ("client %p: attaching to context %p", client, context);
res = gst_rtsp_watch_attach (priv->watch, context);
return res;
/**
* gst_rtsp_client_session_filter:
* @client: a #GstRTSPClient
- * @func: (scope call): a callback
+ * @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
*
* Call @func for each session managed by @client. The result value of @func
* will also be added with an additional ref to the result #GList of this
* function..
*
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
+ *
* Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
* sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
* element in the #GList should be unreffed before the list is freed.
{
GstRTSPClientPrivate *priv;
GList *result, *walk, *next;
+ GHashTable *visited;
+ guint cookie;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- g_return_val_if_fail (func != NULL, NULL);
priv = client->priv;
result = NULL;
+ if (func)
+ visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
g_mutex_lock (&priv->lock);
+restart:
+ cookie = priv->sessions_cookie;
for (walk = priv->sessions; walk; walk = next) {
GstRTSPSession *sess = walk->data;
+ GstRTSPFilterResult res;
+ gboolean changed;
next = g_list_next (walk);
- switch (func (client, sess, user_data)) {
+ if (func) {
+ /* only visit each session once */
+ if (g_hash_table_contains (visited, sess))
+ continue;
+
+ g_hash_table_add (visited, g_object_ref (sess));
+ g_mutex_unlock (&priv->lock);
+
+ res = func (client, sess, user_data);
+
+ g_mutex_lock (&priv->lock);
+ } else
+ res = GST_RTSP_FILTER_REF;
+
+ changed = (cookie != priv->sessions_cookie);
+
+ switch (res) {
case GST_RTSP_FILTER_REMOVE:
- /* stop watching the session and pretent it went away */
- client_cleanup_session (client, sess);
+ /* stop watching the session and pretend it went away, if the list was
+ * changed, we can't use the current list position, try to see if we
+ * still have the session */
+ client_unwatch_session (client, sess, changed ? NULL : walk);
+ cookie = priv->sessions_cookie;
break;
case GST_RTSP_FILTER_REF:
result = g_list_prepend (result, g_object_ref (sess));
default:
break;
}
+ if (changed)
+ goto restart;
}
g_mutex_unlock (&priv->lock);
+ if (func)
+ g_hash_table_unref (visited);
+
return result;
}
projects / platform / upstream / gstreamer.git / blobdiff