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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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4645 lines
132 KiB
C
4645 lines
132 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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* Copyright (C) 2015 Centricular Ltd
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* Author: Sebastian Dröge <sebastian@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:rtsp-client
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* @short_description: A client connection state
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* @see_also: #GstRTSPServer, #GstRTSPThreadPool
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*
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* The client object handles the connection with a client for as long as a TCP
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* connection is open.
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*
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* A #GstRTSPClient is created by #GstRTSPServer when a new connection is
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* accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
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* #GstRTSPAuth and #GstRTSPThreadPool from the server.
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*
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* The client connection should be configured with the #GstRTSPConnection using
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* gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
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* using gst_rtsp_client_attach(). From then on the client will handle requests
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* on the connection.
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*
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* Use gst_rtsp_client_session_filter() to iterate or modify all the
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* #GstRTSPSession objects managed by the client object.
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*
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* Last reviewed on 2013-07-11 (1.0.0)
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*/
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#include <stdio.h>
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#include <string.h>
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#include <gst/sdp/gstmikey.h>
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#include <gst/rtsp/gstrtsp-enumtypes.h>
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#include "rtsp-client.h"
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#include "rtsp-sdp.h"
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#include "rtsp-params.h"
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#define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
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/* locking order:
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* send_lock, lock, tunnels_lock
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*/
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struct _GstRTSPClientPrivate
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{
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GMutex lock; /* protects everything else */
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GMutex send_lock;
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GMutex watch_lock;
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GstRTSPConnection *connection;
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GstRTSPWatch *watch;
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GMainContext *watch_context;
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guint close_seq;
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gchar *server_ip;
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gboolean is_ipv6;
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GstRTSPClientSendFunc send_func; /* protected by send_lock */
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gpointer send_data; /* protected by send_lock */
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GDestroyNotify send_notify; /* protected by send_lock */
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GstRTSPSessionPool *session_pool;
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gulong session_removed_id;
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GstRTSPMountPoints *mount_points;
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GstRTSPAuth *auth;
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GstRTSPThreadPool *thread_pool;
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/* used to cache the media in the last requested DESCRIBE so that
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* we can pick it up in the next SETUP immediately */
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gchar *path;
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GstRTSPMedia *media;
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GHashTable *transports;
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GList *sessions;
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guint sessions_cookie;
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gboolean drop_backlog;
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guint rtsp_ctrl_timeout_id;
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guint rtsp_ctrl_timeout_cnt;
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/* The version currently being used */
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GstRTSPVersion version;
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GHashTable *pipelined_requests; /* pipelined_request_id -> session_id */
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};
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static GMutex tunnels_lock;
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static GHashTable *tunnels; /* protected by tunnels_lock */
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/* FIXME make this configurable. We don't want to do this yet because it will
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* be superceeded by a cache object later */
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#define WATCH_BACKLOG_SIZE 100
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#define DEFAULT_SESSION_POOL NULL
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#define DEFAULT_MOUNT_POINTS NULL
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#define DEFAULT_DROP_BACKLOG TRUE
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#define RTSP_CTRL_CB_INTERVAL 1
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#define RTSP_CTRL_TIMEOUT_VALUE 60
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enum
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{
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PROP_0,
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PROP_SESSION_POOL,
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PROP_MOUNT_POINTS,
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PROP_DROP_BACKLOG,
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PROP_LAST
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};
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enum
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{
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SIGNAL_CLOSED,
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SIGNAL_NEW_SESSION,
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SIGNAL_PRE_OPTIONS_REQUEST,
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SIGNAL_OPTIONS_REQUEST,
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SIGNAL_PRE_DESCRIBE_REQUEST,
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SIGNAL_DESCRIBE_REQUEST,
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SIGNAL_PRE_SETUP_REQUEST,
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SIGNAL_SETUP_REQUEST,
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SIGNAL_PRE_PLAY_REQUEST,
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SIGNAL_PLAY_REQUEST,
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SIGNAL_PRE_PAUSE_REQUEST,
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SIGNAL_PAUSE_REQUEST,
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SIGNAL_PRE_TEARDOWN_REQUEST,
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SIGNAL_TEARDOWN_REQUEST,
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SIGNAL_PRE_SET_PARAMETER_REQUEST,
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SIGNAL_SET_PARAMETER_REQUEST,
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SIGNAL_PRE_GET_PARAMETER_REQUEST,
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SIGNAL_GET_PARAMETER_REQUEST,
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SIGNAL_HANDLE_RESPONSE,
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SIGNAL_SEND_MESSAGE,
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SIGNAL_PRE_ANNOUNCE_REQUEST,
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SIGNAL_ANNOUNCE_REQUEST,
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SIGNAL_PRE_RECORD_REQUEST,
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SIGNAL_RECORD_REQUEST,
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SIGNAL_CHECK_REQUIREMENTS,
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SIGNAL_LAST
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};
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GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
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#define GST_CAT_DEFAULT rtsp_client_debug
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static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
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static void gst_rtsp_client_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec);
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static void gst_rtsp_client_set_property (GObject * object, guint propid,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtsp_client_finalize (GObject * obj);
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static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
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static gboolean handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx,
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GstRTSPMedia * media, GstSDPMessage * sdp);
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static gboolean default_configure_client_media (GstRTSPClient * client,
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GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
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static gboolean default_configure_client_transport (GstRTSPClient * client,
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GstRTSPContext * ctx, GstRTSPTransport * ct);
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static GstRTSPResult default_params_set (GstRTSPClient * client,
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GstRTSPContext * ctx);
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static GstRTSPResult default_params_get (GstRTSPClient * client,
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GstRTSPContext * ctx);
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static gchar *default_make_path_from_uri (GstRTSPClient * client,
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const GstRTSPUrl * uri);
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static void client_session_removed (GstRTSPSessionPool * pool,
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GstRTSPSession * session, GstRTSPClient * client);
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static GstRTSPStatusCode default_pre_signal_handler (GstRTSPClient * client,
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GstRTSPContext * ctx);
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static gboolean pre_signal_accumulator (GSignalInvocationHint * ihint,
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GValue * return_accu, const GValue * handler_return, gpointer data);
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G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
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static void
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gst_rtsp_client_class_init (GstRTSPClientClass * klass)
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{
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GObjectClass *gobject_class;
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g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
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gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->get_property = gst_rtsp_client_get_property;
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gobject_class->set_property = gst_rtsp_client_set_property;
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gobject_class->finalize = gst_rtsp_client_finalize;
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klass->create_sdp = create_sdp;
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klass->handle_sdp = handle_sdp;
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klass->configure_client_media = default_configure_client_media;
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klass->configure_client_transport = default_configure_client_transport;
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klass->params_set = default_params_set;
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klass->params_get = default_params_get;
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klass->make_path_from_uri = default_make_path_from_uri;
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klass->pre_options_request = default_pre_signal_handler;
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klass->pre_describe_request = default_pre_signal_handler;
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klass->pre_setup_request = default_pre_signal_handler;
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klass->pre_play_request = default_pre_signal_handler;
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klass->pre_pause_request = default_pre_signal_handler;
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klass->pre_teardown_request = default_pre_signal_handler;
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klass->pre_set_parameter_request = default_pre_signal_handler;
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klass->pre_get_parameter_request = default_pre_signal_handler;
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klass->pre_announce_request = default_pre_signal_handler;
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klass->pre_record_request = default_pre_signal_handler;
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g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
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g_param_spec_object ("session-pool", "Session Pool",
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"The session pool to use for client session",
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GST_TYPE_RTSP_SESSION_POOL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
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g_param_spec_object ("mount-points", "Mount Points",
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"The mount points to use for client session",
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GST_TYPE_RTSP_MOUNT_POINTS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
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g_param_spec_boolean ("drop-backlog", "Drop Backlog",
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"Drop data when the backlog queue is full",
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DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_rtsp_client_signals[SIGNAL_CLOSED] =
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g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
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g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
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gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
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g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
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g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
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/**
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* GstRTSPClient::pre-options-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
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* Since: 1.12
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*/
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gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST] =
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g_signal_new ("pre-options-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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pre_options_request), pre_signal_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
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GST_TYPE_RTSP_CONTEXT);
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gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
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g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
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NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
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GST_TYPE_RTSP_CONTEXT);
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/**
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* GstRTSPClient::pre-describe-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
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* Since: 1.12
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*/
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gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST] =
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g_signal_new ("pre-describe-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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pre_describe_request), pre_signal_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
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GST_TYPE_RTSP_CONTEXT);
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gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
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g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
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NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
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GST_TYPE_RTSP_CONTEXT);
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/**
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* GstRTSPClient::pre-setup-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
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* Since: 1.12
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*/
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gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST] =
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g_signal_new ("pre-setup-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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pre_setup_request), pre_signal_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
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GST_TYPE_RTSP_CONTEXT);
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gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
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g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
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NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
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GST_TYPE_RTSP_CONTEXT);
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/**
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* GstRTSPClient::pre-play-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
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* Since: 1.12
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*/
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gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST] =
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g_signal_new ("pre-play-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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pre_play_request), pre_signal_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
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GST_TYPE_RTSP_CONTEXT);
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gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
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g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
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NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
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GST_TYPE_RTSP_CONTEXT);
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/**
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* GstRTSPClient::pre-pause-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
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* Since: 1.12
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*/
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gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST] =
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g_signal_new ("pre-pause-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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pre_pause_request), pre_signal_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
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GST_TYPE_RTSP_CONTEXT);
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gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
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g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
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NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
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GST_TYPE_RTSP_CONTEXT);
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/**
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* GstRTSPClient::pre-teardown-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
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* Since: 1.12
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*/
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gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST] =
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g_signal_new ("pre-teardown-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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pre_teardown_request), pre_signal_accumulator, NULL,
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g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
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GST_TYPE_RTSP_CONTEXT);
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gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
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g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
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NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
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GST_TYPE_RTSP_CONTEXT);
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/**
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* GstRTSPClient::pre-set-parameter-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
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* Since: 1.12
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*/
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gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST] =
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g_signal_new ("pre-set-parameter-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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pre_set_parameter_request), pre_signal_accumulator, NULL,
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g_cclosure_marshal_generic,
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GST_TYPE_RTSP_STATUS_CODE, 1, GST_TYPE_RTSP_CONTEXT);
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gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
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g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
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G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
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/**
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* GstRTSPClient::pre-get-parameter-request:
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* @client: a #GstRTSPClient
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* @ctx: a #GstRTSPContext
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*
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* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
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* otherwise an appropriate return code
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*
|
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* Since: 1.12
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*/
|
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gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST] =
|
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g_signal_new ("pre-get-parameter-request", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
|
|
pre_get_parameter_request), pre_signal_accumulator, NULL,
|
|
g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 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_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_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);
|
|
|
|
/**
|
|
* GstRTSPClient::pre-announce-request:
|
|
* @client: a #GstRTSPClient
|
|
* @ctx: a #GstRTSPContext
|
|
*
|
|
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
|
|
* otherwise an appropriate return code
|
|
*
|
|
* Since: 1.12
|
|
*/
|
|
gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST] =
|
|
g_signal_new ("pre-announce-request", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
|
|
pre_announce_request), pre_signal_accumulator, NULL,
|
|
g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 1,
|
|
GST_TYPE_RTSP_CONTEXT);
|
|
|
|
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);
|
|
|
|
/**
|
|
* GstRTSPClient::pre-record-request:
|
|
* @client: a #GstRTSPClient
|
|
* @ctx: a #GstRTSPContext
|
|
*
|
|
* Returns: a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success,
|
|
* otherwise an appropriate return code
|
|
*
|
|
* Since: 1.12
|
|
*/
|
|
gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST] =
|
|
g_signal_new ("pre-record-request", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
|
|
pre_record_request), pre_signal_accumulator, NULL,
|
|
g_cclosure_marshal_generic, GST_TYPE_RTSP_STATUS_CODE, 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);
|
|
|
|
/**
|
|
* GstRTSPClient::check-requirements:
|
|
* @client: a #GstRTSPClient
|
|
* @ctx: a #GstRTSPContext
|
|
* @arr: a NULL-terminated array of strings
|
|
*
|
|
* Returns: a newly allocated string with comma-separated list of
|
|
* unsupported options. An empty string must be returned if
|
|
* all options are supported.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS] =
|
|
g_signal_new ("check-requirements", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
|
|
check_requirements), NULL, NULL, g_cclosure_marshal_generic,
|
|
G_TYPE_STRING, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_STRV);
|
|
|
|
tunnels =
|
|
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
|
|
g_mutex_init (&tunnels_lock);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_init (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
|
|
|
|
client->priv = priv;
|
|
|
|
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);
|
|
priv->pipelined_requests = g_hash_table_new_full (g_str_hash,
|
|
g_str_equal, g_free, g_free);
|
|
}
|
|
|
|
static GstRTSPFilterResult
|
|
filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
|
|
gpointer user_data)
|
|
{
|
|
gboolean *closed = user_data;
|
|
GstRTSPMedia *media;
|
|
guint i, n_streams;
|
|
gboolean is_all_udp = TRUE;
|
|
|
|
media = gst_rtsp_session_media_get_media (sessmedia);
|
|
n_streams = gst_rtsp_media_n_streams (media);
|
|
|
|
for (i = 0; i < n_streams; i++) {
|
|
GstRTSPStreamTransport *transport =
|
|
gst_rtsp_session_media_get_transport (sessmedia, i);
|
|
const GstRTSPTransport *rtsp_transport;
|
|
|
|
if (!transport)
|
|
continue;
|
|
|
|
rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
|
|
if (rtsp_transport
|
|
&& rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP
|
|
&& rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
is_all_udp = FALSE;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!is_all_udp || gst_rtsp_media_is_stop_on_disconnect (media)) {
|
|
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
|
|
return GST_RTSP_FILTER_REMOVE;
|
|
} else {
|
|
*closed = FALSE;
|
|
return GST_RTSP_FILTER_KEEP;
|
|
}
|
|
}
|
|
|
|
static void
|
|
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
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);
|
|
|
|
priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
|
|
priv->sessions_cookie++;
|
|
|
|
/* 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);
|
|
|
|
return;
|
|
}
|
|
|
|
/* should be called with lock */
|
|
static void
|
|
client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
|
|
GList * link)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
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;
|
|
}
|
|
|
|
if (!priv->drop_backlog) {
|
|
/* unlink all media managed in this session */
|
|
gst_rtsp_session_filter (session, filter_session_media, client);
|
|
}
|
|
|
|
/* remove the session */
|
|
g_object_unref (session);
|
|
}
|
|
|
|
static GstRTSPFilterResult
|
|
cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
|
|
gpointer user_data)
|
|
{
|
|
gboolean *closed = user_data;
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
if (priv->drop_backlog) {
|
|
/* 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, user_data);
|
|
}
|
|
|
|
if (*closed)
|
|
return GST_RTSP_FILTER_REMOVE;
|
|
else
|
|
return GST_RTSP_FILTER_KEEP;
|
|
}
|
|
|
|
static void
|
|
clean_cached_media (GstRTSPClient * client, gboolean unprepare)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
/* A client is finalized when the connection is broken */
|
|
static void
|
|
gst_rtsp_client_finalize (GObject * obj)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
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);
|
|
|
|
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);
|
|
g_hash_table_unref (priv->pipelined_requests);
|
|
|
|
if (priv->connection)
|
|
gst_rtsp_connection_free (priv->connection);
|
|
if (priv->session_pool) {
|
|
g_object_unref (priv->session_pool);
|
|
}
|
|
if (priv->mount_points)
|
|
g_object_unref (priv->mount_points);
|
|
if (priv->auth)
|
|
g_object_unref (priv->auth);
|
|
if (priv->thread_pool)
|
|
g_object_unref (priv->thread_pool);
|
|
|
|
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);
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_get_property (GObject * object, guint propid,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (object);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
switch (propid) {
|
|
case PROP_SESSION_POOL:
|
|
g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
|
|
break;
|
|
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);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtsp_client_set_property (GObject * object, guint propid,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (object);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
switch (propid) {
|
|
case PROP_SESSION_POOL:
|
|
gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
|
|
break;
|
|
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);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_new:
|
|
*
|
|
* Create a new #GstRTSPClient instance.
|
|
*
|
|
* Returns: (transfer full): a new #GstRTSPClient
|
|
*/
|
|
GstRTSPClient *
|
|
gst_rtsp_client_new (void)
|
|
{
|
|
GstRTSPClient *result;
|
|
|
|
result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
|
|
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
send_message (GstRTSPClient * client, GstRTSPContext * ctx,
|
|
GstRTSPMessage * message, gboolean close)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
|
|
"GStreamer RTSP server");
|
|
|
|
/* remove any previous header */
|
|
gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
|
|
|
|
/* add the new session header for new session ids */
|
|
if (ctx->session) {
|
|
gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
|
|
gst_rtsp_session_get_header (ctx->session));
|
|
}
|
|
|
|
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
|
|
gst_rtsp_message_dump (message);
|
|
}
|
|
|
|
if (close)
|
|
gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
|
|
|
|
if (ctx->request)
|
|
message->type_data.response.version =
|
|
ctx->request->type_data.request.version;
|
|
|
|
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);
|
|
g_mutex_unlock (&priv->send_lock);
|
|
|
|
gst_rtsp_message_unset (message);
|
|
}
|
|
|
|
static void
|
|
send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
|
|
GstRTSPContext * ctx)
|
|
{
|
|
gst_rtsp_message_init_response (ctx->response, code,
|
|
gst_rtsp_status_as_text (code), ctx->request);
|
|
|
|
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
|
|
paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
|
|
{
|
|
if (path1 == NULL || path2 == NULL)
|
|
return FALSE;
|
|
|
|
if (strlen (path1) != len2)
|
|
return FALSE;
|
|
|
|
if (strncmp (path1, path2, len2))
|
|
return FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* this function is called to initially find the media for the DESCRIBE request
|
|
* 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, GstRTSPContext * ctx, gchar * path,
|
|
gint * matched)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPMediaFactory *factory;
|
|
GstRTSPMedia *media;
|
|
gint path_len;
|
|
|
|
/* find the longest matching factory for the uri first */
|
|
if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
|
|
path, matched)))
|
|
goto no_factory;
|
|
|
|
ctx->factory = factory;
|
|
|
|
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
|
|
goto no_factory_access;
|
|
|
|
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
|
|
goto not_authorized;
|
|
|
|
if (matched)
|
|
path_len = *matched;
|
|
else
|
|
path_len = strlen (path);
|
|
|
|
if (!paths_are_equal (priv->path, path, path_len)) {
|
|
/* remove any previously cached values before we try to construct a new
|
|
* media for uri */
|
|
clean_cached_media (client, TRUE);
|
|
|
|
/* prepare the media and add it to the pipeline */
|
|
if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
|
|
goto no_media;
|
|
|
|
ctx->media = media;
|
|
|
|
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;
|
|
}
|
|
|
|
/* now keep track of the uri and the media */
|
|
priv->path = g_strndup (path, path_len);
|
|
priv->media = media;
|
|
} else {
|
|
/* we have seen this path before, used cached media */
|
|
media = priv->media;
|
|
ctx->media = media;
|
|
GST_INFO ("reusing cached media %p for path %s", media, priv->path);
|
|
}
|
|
|
|
g_object_unref (factory);
|
|
ctx->factory = NULL;
|
|
|
|
if (media)
|
|
g_object_ref (media);
|
|
|
|
return media;
|
|
|
|
/* ERRORS */
|
|
no_factory:
|
|
{
|
|
GST_ERROR ("client %p: no factory for path %s", client, path);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
|
|
return NULL;
|
|
}
|
|
no_factory_access:
|
|
{
|
|
g_object_unref (factory);
|
|
ctx->factory = NULL;
|
|
GST_ERROR ("client %p: not authorized to see factory path %s", client,
|
|
path);
|
|
/* error reply is already sent */
|
|
return NULL;
|
|
}
|
|
not_authorized:
|
|
{
|
|
g_object_unref (factory);
|
|
ctx->factory = NULL;
|
|
GST_ERROR ("client %p: not authorized for factory path %s", client, path);
|
|
/* error reply is already sent */
|
|
return NULL;
|
|
}
|
|
no_media:
|
|
{
|
|
GST_ERROR ("client %p: can't create media", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
|
|
g_object_unref (factory);
|
|
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, ctx);
|
|
g_object_unref (media);
|
|
ctx->media = NULL;
|
|
g_object_unref (factory);
|
|
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, ctx);
|
|
g_object_unref (media);
|
|
ctx->media = NULL;
|
|
g_object_unref (factory);
|
|
ctx->factory = NULL;
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPMessage message = { 0 };
|
|
GstRTSPResult res = GST_RTSP_OK;
|
|
GstMapInfo map_info;
|
|
guint8 *data;
|
|
guint usize;
|
|
|
|
gst_rtsp_message_init_data (&message, channel);
|
|
|
|
/* FIXME, need some sort of iovec RTSPMessage here */
|
|
if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
|
|
return FALSE;
|
|
|
|
gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
|
|
|
|
g_mutex_lock (&priv->send_lock);
|
|
if (priv->send_func)
|
|
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_buffer_unmap (buffer, &map_info);
|
|
|
|
gst_rtsp_message_unset (&message);
|
|
|
|
return res == GST_RTSP_OK;
|
|
}
|
|
|
|
/**
|
|
* 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: closing connection", client);
|
|
|
|
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 gchar *
|
|
default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
|
|
{
|
|
gchar *path;
|
|
|
|
if (uri->query)
|
|
path = g_strconcat (uri->abspath, "?", uri->query, NULL);
|
|
else
|
|
path = g_strdup (uri->abspath);
|
|
|
|
return path;
|
|
}
|
|
|
|
/* Default signal handler function for all "pre-command" signals, like
|
|
* pre-options-request. It just returns the RTSP return code 200.
|
|
* Subclasses can override this to get another default behaviour.
|
|
*/
|
|
static GstRTSPStatusCode
|
|
default_pre_signal_handler (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GST_LOG_OBJECT (client, "returning GST_RTSP_STS_OK");
|
|
return GST_RTSP_STS_OK;
|
|
}
|
|
|
|
/* The pre-signal accumulator function checks the return value of the signal
|
|
* handlers. If any of them returns an RTSP status code that does not start
|
|
* with 2 it will return FALSE, no more signal handlers will be called, and
|
|
* this last RTSP status code will be the result of the signal emission.
|
|
*/
|
|
static gboolean
|
|
pre_signal_accumulator (GSignalInvocationHint * ihint, GValue * return_accu,
|
|
const GValue * handler_return, gpointer data)
|
|
{
|
|
GstRTSPStatusCode handler_value = g_value_get_enum (handler_return);
|
|
GstRTSPStatusCode accumulated_value = g_value_get_enum (return_accu);
|
|
|
|
if (handler_value < 200 || handler_value > 299) {
|
|
GST_DEBUG ("handler_value : %d, returning FALSE", handler_value);
|
|
g_value_set_enum (return_accu, handler_value);
|
|
return FALSE;
|
|
}
|
|
|
|
/* the accumulated value is initiated to 0 by GLib. if current handler value is
|
|
* bigger then use that instead
|
|
*
|
|
* FIXME: Should we prioritize the 2xx codes in a smarter way?
|
|
* Like, "201 Created" > "250 Low On Storage Space" > "200 OK"?
|
|
*/
|
|
if (handler_value > accumulated_value)
|
|
g_value_set_enum (return_accu, handler_value);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPClientClass *klass;
|
|
GstRTSPSession *session;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPStatusCode code;
|
|
gchar *path;
|
|
gint matched;
|
|
gboolean keep_session;
|
|
GstRTSPStatusCode sig_result;
|
|
|
|
if (!ctx->session)
|
|
goto no_session;
|
|
|
|
session = ctx->session;
|
|
|
|
if (!ctx->uri)
|
|
goto no_uri;
|
|
|
|
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 (!sessmedia)
|
|
goto not_found;
|
|
|
|
/* only aggregate control for now.. */
|
|
if (path[matched] != '\0')
|
|
goto no_aggregate;
|
|
|
|
g_free (path);
|
|
|
|
ctx->sessmedia = sessmedia;
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_TEARDOWN_REQUEST],
|
|
0, ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
/* we emit the signal before closing the connection */
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
|
|
0, ctx);
|
|
|
|
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. */
|
|
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, ctx, ctx->response, TRUE);
|
|
|
|
if (!keep_session) {
|
|
/* remove the session */
|
|
gst_rtsp_session_pool_remove (priv->session_pool, session);
|
|
}
|
|
|
|
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 supplied", client);
|
|
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, 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;
|
|
}
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
res = gst_rtsp_params_set (client, ctx);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
res = gst_rtsp_params_get (client, ctx);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPResult res;
|
|
guint8 *data;
|
|
guint size;
|
|
GstRTSPStatusCode sig_result;
|
|
|
|
g_signal_emit (client,
|
|
gst_rtsp_client_signals[SIGNAL_PRE_GET_PARAMETER_REQUEST], 0, ctx,
|
|
&sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
res = gst_rtsp_message_get_body (ctx->request, &data, &size);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
if (size == 0 || !data || strlen ((char *) data) == 0) {
|
|
if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
|
|
GST_ERROR_OBJECT (client, "Using PLAY request for keep-alive is forbidden"
|
|
" in RTSP 2.0");
|
|
goto bad_request;
|
|
}
|
|
|
|
/* no body (or only '\0'), keep-alive request */
|
|
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, ctx);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
send_message (client, ctx, ctx->response, FALSE);
|
|
}
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
|
|
0, ctx);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
return FALSE;
|
|
}
|
|
bad_request:
|
|
{
|
|
GST_ERROR ("client %p: bad request", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPResult res;
|
|
guint8 *data;
|
|
guint size;
|
|
GstRTSPStatusCode sig_result;
|
|
|
|
g_signal_emit (client,
|
|
gst_rtsp_client_signals[SIGNAL_PRE_SET_PARAMETER_REQUEST], 0, ctx,
|
|
&sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
res = gst_rtsp_message_get_body (ctx->request, &data, &size);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
if (size == 0 || !data || strlen ((char *) data) == 0) {
|
|
/* no body (or only '\0'), keep-alive request */
|
|
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, ctx);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
send_message (client, ctx, ctx->response, FALSE);
|
|
}
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
|
|
0, ctx);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
return FALSE;
|
|
}
|
|
bad_request:
|
|
{
|
|
GST_ERROR ("client %p: bad request", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPSession *session;
|
|
GstRTSPClientClass *klass;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPMedia *media;
|
|
GstRTSPStatusCode code;
|
|
GstRTSPState rtspstate;
|
|
gchar *path;
|
|
gint matched;
|
|
GstRTSPStatusCode sig_result;
|
|
guint i, n;
|
|
|
|
if (!(session = ctx->session))
|
|
goto no_session;
|
|
|
|
if (!ctx->uri)
|
|
goto no_uri;
|
|
|
|
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 (!sessmedia)
|
|
goto not_found;
|
|
|
|
if (path[matched] != '\0')
|
|
goto no_aggregate;
|
|
|
|
g_free (path);
|
|
|
|
media = gst_rtsp_session_media_get_media (sessmedia);
|
|
n = gst_rtsp_media_n_streams (media);
|
|
for (i = 0; i < n; i++) {
|
|
GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
|
|
|
|
if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
|
|
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET)
|
|
goto not_supported;
|
|
}
|
|
|
|
ctx->sessmedia = sessmedia;
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PAUSE_REQUEST], 0,
|
|
ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
|
|
/* the session state must be playing or recording */
|
|
if (rtspstate != GST_RTSP_STATE_PLAYING &&
|
|
rtspstate != GST_RTSP_STATE_RECORDING)
|
|
goto invalid_state;
|
|
|
|
/* then pause sending */
|
|
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
|
|
|
|
/* 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, ctx, 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, 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 supplied", client);
|
|
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, 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;
|
|
}
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, 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,
|
|
ctx);
|
|
return FALSE;
|
|
}
|
|
not_supported:
|
|
{
|
|
GST_ERROR ("client %p: pausing not supported", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* 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, const gchar * path)
|
|
{
|
|
GstRTSPUrl tmp;
|
|
gchar *result;
|
|
const gchar *trail;
|
|
|
|
/* check for trailing '/' and append one */
|
|
trail = (path[strlen (path) - 1] != '/' ? "/" : "");
|
|
|
|
tmp = *url;
|
|
tmp.user = NULL;
|
|
tmp.passwd = NULL;
|
|
tmp.abspath = g_strdup_printf ("%s%s", path, trail);
|
|
tmp.query = NULL;
|
|
result = gst_rtsp_url_get_request_uri (&tmp);
|
|
g_free (tmp.abspath);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPSession *session;
|
|
GstRTSPClientClass *klass;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPMedia *media;
|
|
GstRTSPStatusCode code;
|
|
GstRTSPUrl *uri;
|
|
gchar *str;
|
|
GstRTSPTimeRange *range;
|
|
GstRTSPResult res;
|
|
GstRTSPState rtspstate;
|
|
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
|
|
gchar *path, *rtpinfo;
|
|
gint matched;
|
|
gchar *seek_style = NULL;
|
|
GstRTSPStatusCode sig_result;
|
|
GPtrArray *transports;
|
|
|
|
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);
|
|
|
|
/* 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 (path[matched] != '\0')
|
|
goto no_aggregate;
|
|
|
|
g_free (path);
|
|
|
|
ctx->sessmedia = sessmedia;
|
|
ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_PLAY_REQUEST], 0,
|
|
ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
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;
|
|
|
|
/* update the pipeline */
|
|
transports = gst_rtsp_session_media_get_transports (sessmedia);
|
|
if (!gst_rtsp_media_complete_pipeline (media, transports)) {
|
|
g_ptr_array_unref (transports);
|
|
goto pipeline_error;
|
|
}
|
|
g_ptr_array_unref (transports);
|
|
|
|
/* 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;
|
|
GstSeekFlags flags = 0;
|
|
|
|
if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_SEEK_STYLE,
|
|
&seek_style, 0)) {
|
|
if (g_strcmp0 (seek_style, "RAP") == 0)
|
|
flags = GST_SEEK_FLAG_ACCURATE;
|
|
else if (g_strcmp0 (seek_style, "CoRAP") == 0)
|
|
flags = GST_SEEK_FLAG_KEY_UNIT;
|
|
else if (g_strcmp0 (seek_style, "First-Prior") == 0)
|
|
flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_BEFORE;
|
|
else if (g_strcmp0 (seek_style, "Next") == 0)
|
|
flags = GST_SEEK_FLAG_KEY_UNIT & GST_SEEK_FLAG_SNAP_AFTER;
|
|
else
|
|
GST_FIXME_OBJECT (client, "Add support for seek style %s",
|
|
seek_style);
|
|
}
|
|
|
|
/* we have a range, seek to the position */
|
|
unit = range->unit;
|
|
gst_rtsp_media_seek_full (media, range, flags);
|
|
gst_rtsp_range_free (range);
|
|
|
|
media_status = gst_rtsp_media_get_status (media);
|
|
if (media_status == GST_RTSP_MEDIA_STATUS_ERROR)
|
|
goto seek_failed;
|
|
}
|
|
}
|
|
|
|
/* 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_message_init_response (ctx->response, code,
|
|
gst_rtsp_status_as_text (code), ctx->request);
|
|
|
|
/* add the RTP-Info header */
|
|
if (rtpinfo)
|
|
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
|
|
rtpinfo);
|
|
if (seek_style)
|
|
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_SEEK_STYLE,
|
|
seek_style);
|
|
|
|
/* 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, ctx, ctx->response, FALSE);
|
|
|
|
/* 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);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_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 supplied", client);
|
|
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, 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, ctx);
|
|
g_free (path);
|
|
return FALSE;
|
|
}
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, 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,
|
|
ctx);
|
|
return FALSE;
|
|
}
|
|
pipeline_error:
|
|
{
|
|
GST_ERROR ("client %p: failed to configure the pipeline", 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);
|
|
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
|
|
do_keepalive (GstRTSPSession * session)
|
|
{
|
|
GST_INFO ("keep session %p alive", session);
|
|
gst_rtsp_session_touch (session);
|
|
}
|
|
|
|
/* parse @transport and return a valid transport in @tr. only transports
|
|
* supported by @stream are returned. Returns FALSE if no valid transport
|
|
* was found. */
|
|
static gboolean
|
|
parse_transport (const char *transport, GstRTSPStream * stream,
|
|
GstRTSPTransport * tr)
|
|
{
|
|
gint i;
|
|
gboolean res;
|
|
gchar **transports;
|
|
|
|
res = FALSE;
|
|
gst_rtsp_transport_init (tr);
|
|
|
|
GST_DEBUG ("parsing transports %s", transport);
|
|
|
|
transports = g_strsplit (transport, ",", 0);
|
|
|
|
/* loop through the transports, try to parse */
|
|
for (i = 0; transports[i]; i++) {
|
|
res = gst_rtsp_transport_parse (transports[i], tr);
|
|
if (res != GST_RTSP_OK) {
|
|
/* no valid transport, search some more */
|
|
GST_WARNING ("could not parse transport %s", transports[i]);
|
|
goto next;
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
|
|
/* we have a valid transport */
|
|
GST_INFO ("found valid transport %s", transports[i]);
|
|
res = TRUE;
|
|
break;
|
|
|
|
next:
|
|
gst_rtsp_transport_init (tr);
|
|
}
|
|
g_strfreev (transports);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
|
|
GstRTSPStream * stream, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPMessage *request = ctx->request;
|
|
gchar *blocksize_str;
|
|
|
|
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
|
|
&blocksize_str, 0) == GST_RTSP_OK) {
|
|
guint64 blocksize;
|
|
gchar *end;
|
|
|
|
blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
|
|
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;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
default_configure_client_transport (GstRTSPClient * client,
|
|
GstRTSPContext * ctx, GstRTSPTransport * ct)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
/* we have a valid transport now, set the destination of the client. */
|
|
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST ||
|
|
ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP) {
|
|
|
|
/* allocate UDP ports */
|
|
GSocketFamily family;
|
|
gboolean use_client_settings = FALSE;
|
|
|
|
family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
|
|
if ((ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) &&
|
|
gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS) &&
|
|
(ct->destination != NULL))
|
|
use_client_settings = TRUE;
|
|
|
|
if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, family, ct,
|
|
use_client_settings))
|
|
goto error_allocating_ports;
|
|
|
|
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
|
|
GstRTSPAddress *addr = NULL;
|
|
|
|
if (use_client_settings) {
|
|
/* the address has been successfully allocated, let's check if it's
|
|
* the one requested by the client */
|
|
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)
|
|
goto no_address;
|
|
} else {
|
|
g_free (ct->destination);
|
|
addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
|
|
if (addr == NULL)
|
|
goto no_address;
|
|
ct->destination = g_strdup (addr->address);
|
|
ct->port.min = addr->port;
|
|
ct->port.max = addr->port + addr->n_ports - 1;
|
|
ct->ttl = addr->ttl;
|
|
}
|
|
|
|
gst_rtsp_address_free (addr);
|
|
} else {
|
|
GstRTSPUrl *url;
|
|
|
|
url = gst_rtsp_connection_get_url (priv->connection);
|
|
g_free (ct->destination);
|
|
ct->destination = g_strdup (url->host);
|
|
}
|
|
} else {
|
|
GstRTSPUrl *url;
|
|
|
|
url = gst_rtsp_connection_get_url (priv->connection);
|
|
g_free (ct->destination);
|
|
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,
|
|
&ct->interleaved);
|
|
}
|
|
}
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error_allocating_ports:
|
|
{
|
|
GST_ERROR_OBJECT (client, "Failed to allocate UDP ports");
|
|
return FALSE;
|
|
}
|
|
no_address:
|
|
{
|
|
GST_ERROR_OBJECT (client, "Failed to acquire address for stream");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstRTSPTransport *
|
|
make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
|
|
GstRTSPContext * ctx, GstRTSPTransport * ct)
|
|
{
|
|
GstRTSPTransport *st;
|
|
GInetAddress *addr;
|
|
GSocketFamily family;
|
|
|
|
/* prepare the server transport */
|
|
gst_rtsp_transport_new (&st);
|
|
|
|
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);
|
|
|
|
if (!addr) {
|
|
GST_ERROR ("failed to get inet addr from client destination");
|
|
family = G_SOCKET_FAMILY_IPV4;
|
|
} else {
|
|
family = g_inet_address_get_family (addr);
|
|
g_object_unref (addr);
|
|
addr = NULL;
|
|
}
|
|
|
|
switch (st->lower_transport) {
|
|
case GST_RTSP_LOWER_TRANS_UDP:
|
|
st->client_port = ct->client_port;
|
|
gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
|
|
st->port = ct->port;
|
|
st->destination = g_strdup (ct->destination);
|
|
st->ttl = ct->ttl;
|
|
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;
|
|
}
|
|
|
|
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
|
|
mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
|
|
{
|
|
const gchar *srtp_cipher;
|
|
const gchar *srtp_auth;
|
|
const GstMIKEYPayload *sp;
|
|
guint i;
|
|
|
|
/* 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;
|
|
|
|
if (((GstMIKEYPayloadSP *) sp)->policy == policy)
|
|
break;
|
|
}
|
|
|
|
/* 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);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
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);
|
|
|
|
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
|
|
rtsp_ctrl_timeout_cb (gpointer user_data)
|
|
{
|
|
gboolean res = G_SOURCE_CONTINUE;
|
|
GstRTSPClient *client = (GstRTSPClient *) user_data;
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
priv->rtsp_ctrl_timeout_cnt += RTSP_CTRL_CB_INTERVAL;
|
|
|
|
if (priv->rtsp_ctrl_timeout_cnt > RTSP_CTRL_TIMEOUT_VALUE) {
|
|
GST_DEBUG ("rtsp control session timeout id=%u expired, closing client.",
|
|
priv->rtsp_ctrl_timeout_id);
|
|
g_mutex_lock (&priv->lock);
|
|
priv->rtsp_ctrl_timeout_id = 0;
|
|
priv->rtsp_ctrl_timeout_cnt = 0;
|
|
g_mutex_unlock (&priv->lock);
|
|
gst_rtsp_client_close (client);
|
|
|
|
res = G_SOURCE_REMOVE;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
rtsp_ctrl_timeout_remove (GstRTSPClientPrivate * priv)
|
|
{
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
if (priv->rtsp_ctrl_timeout_id != 0) {
|
|
g_source_destroy (g_main_context_find_source_by_id (priv->watch_context,
|
|
priv->rtsp_ctrl_timeout_id));
|
|
GST_DEBUG ("rtsp control session removed timeout id=%u.",
|
|
priv->rtsp_ctrl_timeout_id);
|
|
priv->rtsp_ctrl_timeout_id = 0;
|
|
priv->rtsp_ctrl_timeout_cnt = 0;
|
|
}
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
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;
|
|
GstRTSPStatusCode sig_result;
|
|
gchar *pipelined_request_id = NULL, *accept_range = NULL;
|
|
|
|
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;
|
|
|
|
/* Handle Pipelined-requests if using >= 2.0 */
|
|
if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0)
|
|
gst_rtsp_message_get_header (ctx->request,
|
|
GST_RTSP_HDR_PIPELINED_REQUESTS, &pipelined_request_id, 0);
|
|
|
|
/* 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 */
|
|
sessmedia = NULL;
|
|
}
|
|
|
|
/* 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, ctx, path, &matched);
|
|
/* need to suspend the media, if the protocol has changed */
|
|
if (media != NULL)
|
|
gst_rtsp_media_suspend (media);
|
|
} else {
|
|
if ((media = gst_rtsp_session_media_get_media (sessmedia)))
|
|
g_object_ref (media);
|
|
else
|
|
goto media_not_found;
|
|
}
|
|
/* no media, not found then */
|
|
if (media == NULL)
|
|
goto media_not_found_no_reply;
|
|
|
|
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];
|
|
|
|
/* find the stream now using the control part */
|
|
stream = gst_rtsp_media_find_stream (media, control);
|
|
}
|
|
|
|
if (stream == NULL)
|
|
goto stream_not_found;
|
|
|
|
/* now we have a uri identifying a valid media and stream */
|
|
ctx->stream = stream;
|
|
ctx->media = media;
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_SETUP_REQUEST], 0,
|
|
ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
if (session == NULL) {
|
|
/* create a session if this fails we probably reached our session limit or
|
|
* something. */
|
|
if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
|
|
goto service_unavailable;
|
|
|
|
/* Pipelined requests should be cleared between sessions */
|
|
g_hash_table_remove_all (priv->pipelined_requests);
|
|
|
|
/* 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 (pipelined_request_id) {
|
|
g_hash_table_insert (client->priv->pipelined_requests,
|
|
g_strdup (pipelined_request_id),
|
|
g_strdup (gst_rtsp_session_get_sessionid (session)));
|
|
}
|
|
rtsp_ctrl_timeout_remove (priv);
|
|
|
|
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 (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
|
|
&accept_range, 0) == GST_RTSP_OK) {
|
|
GEnumValue *runit = NULL;
|
|
gint i;
|
|
gchar **valid_ranges;
|
|
GEnumClass *runit_class = g_type_class_ref (GST_TYPE_RTSP_RANGE_UNIT);
|
|
|
|
gst_rtsp_message_dump (ctx->request);
|
|
valid_ranges = g_strsplit (accept_range, ",", -1);
|
|
|
|
for (i = 0; valid_ranges[i]; i++) {
|
|
gchar *range = valid_ranges[i];
|
|
|
|
while (*range == ' ')
|
|
range++;
|
|
|
|
runit = g_enum_get_value_by_nick (runit_class, range);
|
|
if (runit)
|
|
break;
|
|
}
|
|
g_strfreev (valid_ranges);
|
|
g_type_class_unref (runit_class);
|
|
|
|
if (!runit)
|
|
goto unsupported_range_unit;
|
|
}
|
|
|
|
if (sessmedia == NULL) {
|
|
/* manage the media in our session now, if not done already */
|
|
sessmedia =
|
|
gst_rtsp_session_manage_media (session, path, g_object_ref (media));
|
|
/* if we stil have no media, error */
|
|
if (sessmedia == NULL)
|
|
goto sessmedia_unavailable;
|
|
|
|
/* don't cache media anymore */
|
|
clean_cached_media (client, FALSE);
|
|
}
|
|
|
|
ctx->sessmedia = sessmedia;
|
|
|
|
/* update the client transport */
|
|
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, media, 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 (ctx->response, code,
|
|
gst_rtsp_status_as_text (code), ctx->request);
|
|
|
|
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
|
|
trans_str);
|
|
g_free (trans_str);
|
|
|
|
if (pipelined_request_id)
|
|
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PIPELINED_REQUESTS,
|
|
pipelined_request_id);
|
|
|
|
if (ctx->request->type_data.request.version >= GST_RTSP_VERSION_2_0) {
|
|
GstClockTimeDiff seekable = gst_rtsp_media_seekable (media);
|
|
GString *media_properties = g_string_new (NULL);
|
|
|
|
if (seekable == -1)
|
|
g_string_append (media_properties,
|
|
"No-Seeking,Time-Progressing,Time-Duration=0.0");
|
|
else if (seekable == 0)
|
|
g_string_append (media_properties, "Beginning-Only");
|
|
else if (seekable == G_MAXINT64)
|
|
g_string_append (media_properties, "Random-Access");
|
|
else
|
|
g_string_append_printf (media_properties,
|
|
"Random-Access=%f, Unlimited, Immutable",
|
|
(gdouble) seekable / GST_SECOND);
|
|
|
|
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_MEDIA_PROPERTIES,
|
|
g_string_free (media_properties, FALSE));
|
|
/* TODO Check how Accept-Ranges should be filled */
|
|
gst_rtsp_message_add_header (ctx->request, GST_RTSP_HDR_ACCEPT_RANGES,
|
|
"npt, clock, smpte, clock");
|
|
}
|
|
|
|
send_message (client, ctx, ctx->response, FALSE);
|
|
|
|
/* update the state */
|
|
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
|
|
switch (rtspstate) {
|
|
case GST_RTSP_STATE_PLAYING:
|
|
case GST_RTSP_STATE_RECORDING:
|
|
case GST_RTSP_STATE_READY:
|
|
/* no state change */
|
|
break;
|
|
default:
|
|
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
|
|
break;
|
|
}
|
|
g_object_unref (media);
|
|
g_object_unref (session);
|
|
g_free (path);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_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_transport:
|
|
{
|
|
GST_ERROR ("client %p: no transport", client);
|
|
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
|
|
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);
|
|
goto cleanup_path;
|
|
}
|
|
media_not_found_no_reply:
|
|
{
|
|
GST_ERROR ("client %p: media '%s' not found", client, path);
|
|
/* error reply is already sent */
|
|
goto cleanup_session;
|
|
}
|
|
media_not_found:
|
|
{
|
|
GST_ERROR ("client %p: media '%s' not found", client, path);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
|
|
goto cleanup_session;
|
|
}
|
|
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);
|
|
goto cleanup_session;
|
|
}
|
|
stream_not_found:
|
|
{
|
|
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);
|
|
goto cleanup_session;
|
|
}
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
g_object_unref (media);
|
|
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);
|
|
goto cleanup_session;
|
|
}
|
|
sessmedia_unavailable:
|
|
{
|
|
GST_ERROR ("client %p: can't create session media", client);
|
|
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
|
|
goto cleanup_transport;
|
|
}
|
|
configure_media_failed_no_reply:
|
|
{
|
|
GST_ERROR ("client %p: configure_media failed", client);
|
|
g_object_unref (media);
|
|
/* 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);
|
|
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;
|
|
}
|
|
unsupported_range_unit:
|
|
{
|
|
GST_ERROR ("Client %p: does not support any range format we support",
|
|
client);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, 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);
|
|
if (media)
|
|
g_object_unref (media);
|
|
cleanup_session:
|
|
if (new_session)
|
|
gst_rtsp_session_pool_remove (priv->session_pool, session);
|
|
if (session)
|
|
g_object_unref (session);
|
|
cleanup_path:
|
|
g_free (path);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstSDPMessage *
|
|
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstSDPMessage *sdp;
|
|
GstSDPInfo info;
|
|
const gchar *proto;
|
|
guint64 session_id_tmp;
|
|
gchar session_id[21];
|
|
|
|
gst_sdp_message_new (&sdp);
|
|
|
|
/* some standard things first */
|
|
gst_sdp_message_set_version (sdp, "0");
|
|
|
|
if (priv->is_ipv6)
|
|
proto = "IP6";
|
|
else
|
|
proto = "IP4";
|
|
|
|
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");
|
|
gst_sdp_message_set_information (sdp, "rtsp-server");
|
|
gst_sdp_message_add_time (sdp, "0", "0", NULL);
|
|
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
|
|
gst_sdp_message_add_attribute (sdp, "type", "broadcast");
|
|
gst_sdp_message_add_attribute (sdp, "control", "*");
|
|
|
|
info.is_ipv6 = priv->is_ipv6;
|
|
info.server_ip = priv->server_ip;
|
|
|
|
/* create an SDP for the media object */
|
|
if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
|
|
goto no_sdp;
|
|
|
|
return sdp;
|
|
|
|
/* ERRORS */
|
|
no_sdp:
|
|
{
|
|
GST_ERROR ("client %p: could not create SDP", client);
|
|
gst_sdp_message_free (sdp);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* for the describe we must generate an SDP */
|
|
static gboolean
|
|
handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPResult res;
|
|
GstSDPMessage *sdp;
|
|
guint i;
|
|
gchar *path, *str;
|
|
GstRTSPMedia *media;
|
|
GstRTSPClientClass *klass;
|
|
GstRTSPStatusCode sig_result;
|
|
|
|
klass = GST_RTSP_CLIENT_GET_CLASS (client);
|
|
|
|
if (!ctx->uri)
|
|
goto no_uri;
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_DESCRIBE_REQUEST],
|
|
0, ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
/* 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 */
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
return FALSE;
|
|
}
|
|
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;
|
|
GstRTSPStatusCode sig_result;
|
|
|
|
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;
|
|
|
|
ctx->media = media;
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_ANNOUNCE_REQUEST],
|
|
0, ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
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;
|
|
}
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
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;
|
|
GstRTSPStatusCode sig_result;
|
|
GPtrArray *transports;
|
|
|
|
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);
|
|
|
|
/* 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 (path[matched] != '\0')
|
|
goto no_aggregate;
|
|
|
|
g_free (path);
|
|
|
|
ctx->sessmedia = sessmedia;
|
|
ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_RECORD_REQUEST], 0,
|
|
ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
if (!(gst_rtsp_media_get_transport_mode (media) &
|
|
GST_RTSP_TRANSPORT_MODE_RECORD))
|
|
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;
|
|
|
|
/* update the pipeline */
|
|
transports = gst_rtsp_session_media_get_transports (sessmedia);
|
|
if (!gst_rtsp_media_complete_pipeline (media, transports)) {
|
|
g_ptr_array_unref (transports);
|
|
goto pipeline_error;
|
|
}
|
|
g_ptr_array_unref (transports);
|
|
|
|
/* in record 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);
|
|
|
|
send_message (client, ctx, ctx->response, FALSE);
|
|
|
|
/* 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);
|
|
|
|
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 supplied", client);
|
|
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, 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, ctx);
|
|
g_free (path);
|
|
return FALSE;
|
|
}
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
return FALSE;
|
|
}
|
|
unsupported_mode:
|
|
{
|
|
GST_ERROR ("client %p: media does not support RECORD", client);
|
|
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_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,
|
|
ctx);
|
|
return FALSE;
|
|
}
|
|
pipeline_error:
|
|
{
|
|
GST_ERROR ("client %p: failed to configure the pipeline", 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);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx,
|
|
GstRTSPVersion version)
|
|
{
|
|
GstRTSPMethod options;
|
|
gchar *str;
|
|
GstRTSPStatusCode sig_result;
|
|
|
|
options = GST_RTSP_DESCRIBE |
|
|
GST_RTSP_OPTIONS |
|
|
GST_RTSP_PAUSE |
|
|
GST_RTSP_PLAY |
|
|
GST_RTSP_SETUP |
|
|
GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
|
|
|
|
if (version < GST_RTSP_VERSION_2_0) {
|
|
options |= GST_RTSP_RECORD;
|
|
options |= GST_RTSP_ANNOUNCE;
|
|
}
|
|
|
|
str = gst_rtsp_options_as_text (options);
|
|
|
|
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_PUBLIC, str);
|
|
g_free (str);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PRE_OPTIONS_REQUEST], 0,
|
|
ctx, &sig_result);
|
|
if (sig_result != GST_RTSP_STS_OK) {
|
|
goto sig_failed;
|
|
}
|
|
|
|
send_message (client, ctx, ctx->response, FALSE);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
|
|
0, ctx);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
sig_failed:
|
|
{
|
|
GST_ERROR ("client %p: pre signal returned error: %s", client,
|
|
gst_rtsp_status_as_text (sig_result));
|
|
send_generic_response (client, sig_result, ctx);
|
|
gst_rtsp_message_free (ctx->response);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* remove duplicate and trailing '/' */
|
|
static void
|
|
sanitize_uri (GstRTSPUrl * uri)
|
|
{
|
|
gint i, len;
|
|
gchar *s, *d;
|
|
gboolean have_slash, prev_slash;
|
|
|
|
s = d = uri->abspath;
|
|
len = strlen (uri->abspath);
|
|
|
|
prev_slash = FALSE;
|
|
|
|
for (i = 0; i < len; i++) {
|
|
have_slash = s[i] == '/';
|
|
*d = s[i];
|
|
if (!have_slash || !prev_slash)
|
|
d++;
|
|
prev_slash = have_slash;
|
|
}
|
|
len = d - uri->abspath;
|
|
/* don't remove the first slash if that's the only thing left */
|
|
if (len > 1 && *(d - 1) == '/')
|
|
d--;
|
|
*d = '\0';
|
|
}
|
|
|
|
/* is called when the session is removed from its session pool. */
|
|
static void
|
|
client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
|
|
GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
GST_INFO ("client %p: session %p removed", 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);
|
|
}
|
|
|
|
/* Check for Require headers. Returns TRUE if there are no Require headers,
|
|
* otherwise lets the application decide which headers are supported.
|
|
* By default all headers are unsupported.
|
|
* If there are unsupported options, FALSE will be returned together with
|
|
* 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.
|
|
*/
|
|
static gboolean
|
|
check_request_requirements (GstRTSPContext * ctx, gchar ** unsupported_reqs)
|
|
{
|
|
GstRTSPResult res;
|
|
GPtrArray *arr = NULL;
|
|
GstRTSPMessage *msg = ctx->request;
|
|
gchar *reqs = NULL;
|
|
gint i;
|
|
gchar *sig_result = NULL;
|
|
gboolean result = TRUE;
|
|
|
|
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);
|
|
|
|
g_signal_emit (ctx->client,
|
|
gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS], 0, ctx,
|
|
(gchar **) arr->pdata, &sig_result);
|
|
|
|
if (sig_result == NULL) {
|
|
/* no supported options, just report all of the required ones as
|
|
* unsupported */
|
|
*unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
|
|
result = FALSE;
|
|
goto done;
|
|
}
|
|
|
|
if (strlen (sig_result) == 0)
|
|
g_free (sig_result);
|
|
else {
|
|
*unsupported_reqs = sig_result;
|
|
result = FALSE;
|
|
}
|
|
|
|
done:
|
|
g_ptr_array_unref (arr);
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPMethod method;
|
|
const gchar *uristr;
|
|
GstRTSPUrl *uri = NULL;
|
|
GstRTSPVersion version;
|
|
GstRTSPResult res;
|
|
GstRTSPSession *session = NULL;
|
|
GstRTSPContext sctx = { NULL }, *ctx;
|
|
GstRTSPMessage response = { 0 };
|
|
gchar *unsupported_reqs = NULL;
|
|
gchar *sessid = NULL, *pipelined_request_id = NULL;
|
|
|
|
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->request = request;
|
|
ctx->response = &response;
|
|
|
|
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
|
|
gst_rtsp_message_dump (request);
|
|
}
|
|
|
|
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 && version != GST_RTSP_VERSION_2_0)
|
|
goto not_supported;
|
|
|
|
ctx->method = method;
|
|
|
|
/* 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) {
|
|
/* 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_PIPELINED_REQUESTS,
|
|
&pipelined_request_id, 0);
|
|
if (res == GST_RTSP_OK) {
|
|
sessid = g_hash_table_lookup (client->priv->pipelined_requests,
|
|
pipelined_request_id);
|
|
|
|
if (!sessid)
|
|
res = GST_RTSP_ERROR;
|
|
}
|
|
|
|
if (res != GST_RTSP_OK)
|
|
res =
|
|
gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
|
|
|
|
if (res == GST_RTSP_OK) {
|
|
if (priv->session_pool == NULL)
|
|
goto no_pool;
|
|
|
|
/* we had a session in the request, find it again */
|
|
if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
|
|
goto session_not_found;
|
|
|
|
/* we add the session to the client list of watched sessions. When a session
|
|
* disappears because it times out, we will be notified. If all sessions are
|
|
* gone, we will close the connection */
|
|
client_watch_session (client, session);
|
|
}
|
|
|
|
/* sanitize the uri */
|
|
if (uri)
|
|
sanitize_uri (uri);
|
|
ctx->uri = uri;
|
|
ctx->session = session;
|
|
|
|
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
|
|
goto not_authorized;
|
|
|
|
/* handle any 'Require' headers */
|
|
if (!check_request_requirements (ctx, &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:
|
|
priv->version = version;
|
|
handle_options_request (client, ctx, version);
|
|
break;
|
|
case GST_RTSP_DESCRIBE:
|
|
handle_describe_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_SETUP:
|
|
handle_setup_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_PLAY:
|
|
handle_play_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_PAUSE:
|
|
handle_pause_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_TEARDOWN:
|
|
handle_teardown_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_SET_PARAMETER:
|
|
handle_set_param_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_GET_PARAMETER:
|
|
handle_get_param_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_ANNOUNCE:
|
|
if (version >= GST_RTSP_VERSION_2_0)
|
|
goto invalid_command_for_version;
|
|
handle_announce_request (client, ctx);
|
|
break;
|
|
case GST_RTSP_RECORD:
|
|
if (version >= GST_RTSP_VERSION_2_0)
|
|
goto invalid_command_for_version;
|
|
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);
|
|
if (session)
|
|
g_object_unref (session);
|
|
if (uri)
|
|
gst_rtsp_url_free (uri);
|
|
return;
|
|
|
|
/* ERRORS */
|
|
not_supported:
|
|
{
|
|
GST_ERROR ("client %p: version %d not supported", client, version);
|
|
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
|
|
ctx);
|
|
goto done;
|
|
}
|
|
invalid_command_for_version:
|
|
{
|
|
if (priv->watch != NULL)
|
|
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
|
|
|
|
GST_ERROR ("client %p: invalid command for version", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
|
|
goto done;
|
|
}
|
|
bad_request:
|
|
{
|
|
GST_ERROR ("client %p: bad request", client);
|
|
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, ctx);
|
|
goto done;
|
|
}
|
|
session_not_found:
|
|
{
|
|
GST_ERROR ("client %p: session not found", client);
|
|
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
|
|
goto done;
|
|
}
|
|
not_authorized:
|
|
{
|
|
GST_ERROR ("client %p: not allowed", client);
|
|
/* 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);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
|
|
static void
|
|
handle_response (GstRTSPClient * client, GstRTSPMessage * response)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPResult res;
|
|
GstRTSPSession *session = NULL;
|
|
GstRTSPContext sctx = { NULL }, *ctx;
|
|
gchar *sessid;
|
|
|
|
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->request = NULL;
|
|
ctx->uri = NULL;
|
|
ctx->method = GST_RTSP_INVALID;
|
|
ctx->response = response;
|
|
|
|
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
|
|
gst_rtsp_message_dump (response);
|
|
}
|
|
|
|
GST_INFO ("client %p: received a response", client);
|
|
|
|
/* get the session if there is any */
|
|
res =
|
|
gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
|
|
if (res == GST_RTSP_OK) {
|
|
if (priv->session_pool == NULL)
|
|
goto no_pool;
|
|
|
|
/* we had a session in the request, find it again */
|
|
if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
|
|
goto session_not_found;
|
|
|
|
/* we add the session to the client list of watched sessions. When a session
|
|
* disappears because it times out, we will be notified. If all sessions are
|
|
* gone, we will close the connection */
|
|
client_watch_session (client, session);
|
|
}
|
|
|
|
ctx->session = session;
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
|
|
0, ctx);
|
|
|
|
done:
|
|
if (ctx == &sctx)
|
|
gst_rtsp_context_pop_current (ctx);
|
|
if (session)
|
|
g_object_unref (session);
|
|
return;
|
|
|
|
no_pool:
|
|
{
|
|
GST_ERROR ("client %p: no pool configured", client);
|
|
goto done;
|
|
}
|
|
session_not_found:
|
|
{
|
|
GST_ERROR ("client %p: session not found", client);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static void
|
|
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPResult res;
|
|
guint8 channel;
|
|
guint8 *data;
|
|
guint size;
|
|
GstBuffer *buffer;
|
|
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);
|
|
|
|
/* Strip trailing \0 (which GstRTSPConnection adds) */
|
|
--size;
|
|
|
|
buffer = gst_buffer_new_wrapped (data, size);
|
|
|
|
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);
|
|
}
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
invalid_length:
|
|
{
|
|
GST_DEBUG ("client %p: Short message received, ignoring", client);
|
|
return;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_session_pool:
|
|
* @client: a #GstRTSPClient
|
|
* @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
|
|
* that created the client but can be overridden later.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_session_pool (GstRTSPClient * client,
|
|
GstRTSPSessionPool * pool)
|
|
{
|
|
GstRTSPSessionPool *old;
|
|
GstRTSPClientPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
|
|
|
|
priv = client->priv;
|
|
|
|
if (pool)
|
|
g_object_ref (pool);
|
|
|
|
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_get_session_pool:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
|
|
*
|
|
* Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
|
|
*/
|
|
GstRTSPSessionPool *
|
|
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GstRTSPSessionPool *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
|
|
|
|
priv = client->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = priv->session_pool))
|
|
g_object_ref (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_mount_points:
|
|
* @client: a #GstRTSPClient
|
|
* @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
|
|
* created the client but can be overriden later.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_mount_points (GstRTSPClient * client,
|
|
GstRTSPMountPoints * mounts)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GstRTSPMountPoints *old;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
|
|
|
|
priv = client->priv;
|
|
|
|
if (mounts)
|
|
g_object_ref (mounts);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
old = priv->mount_points;
|
|
priv->mount_points = mounts;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (old)
|
|
g_object_unref (old);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_get_mount_points:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
|
|
*
|
|
* Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
|
|
*/
|
|
GstRTSPMountPoints *
|
|
gst_rtsp_client_get_mount_points (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GstRTSPMountPoints *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
|
|
|
|
priv = client->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = priv->mount_points))
|
|
g_object_ref (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_auth:
|
|
* @client: a #GstRTSPClient
|
|
* @auth: (transfer none): a #GstRTSPAuth
|
|
*
|
|
* configure @auth to be used as the authentication manager of @client.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GstRTSPAuth *old;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
|
|
|
|
priv = client->priv;
|
|
|
|
if (auth)
|
|
g_object_ref (auth);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
old = priv->auth;
|
|
priv->auth = auth;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (old)
|
|
g_object_unref (old);
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_rtsp_client_get_auth:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Get the #GstRTSPAuth used as the authentication manager of @client.
|
|
*
|
|
* Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
|
|
* usage.
|
|
*/
|
|
GstRTSPAuth *
|
|
gst_rtsp_client_get_auth (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GstRTSPAuth *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
|
|
|
|
priv = client->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = priv->auth))
|
|
g_object_ref (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_thread_pool:
|
|
* @client: a #GstRTSPClient
|
|
* @pool: (transfer none): a #GstRTSPThreadPool
|
|
*
|
|
* configure @pool to be used as the thread pool of @client.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
|
|
GstRTSPThreadPool * pool)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GstRTSPThreadPool *old;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
|
|
|
|
priv = client->priv;
|
|
|
|
if (pool)
|
|
g_object_ref (pool);
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
old = priv->thread_pool;
|
|
priv->thread_pool = pool;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (old)
|
|
g_object_unref (old);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_get_thread_pool:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Get the #GstRTSPThreadPool used as the thread pool of @client.
|
|
*
|
|
* Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
|
|
* usage.
|
|
*/
|
|
GstRTSPThreadPool *
|
|
gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GstRTSPThreadPool *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
|
|
|
|
priv = client->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
if ((result = priv->thread_pool))
|
|
g_object_ref (result);
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_connection:
|
|
* @client: a #GstRTSPClient
|
|
* @conn: (transfer full): a #GstRTSPConnection
|
|
*
|
|
* Set the #GstRTSPConnection of @client. This function takes ownership of
|
|
* @conn.
|
|
*
|
|
* Returns: %TRUE on success.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_client_set_connection (GstRTSPClient * client,
|
|
GstRTSPConnection * conn)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GSocket *read_socket;
|
|
GSocketAddress *address;
|
|
GstRTSPUrl *url;
|
|
GError *error = NULL;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
|
|
g_return_val_if_fail (conn != NULL, FALSE);
|
|
|
|
priv = client->priv;
|
|
|
|
read_socket = gst_rtsp_connection_get_read_socket (conn);
|
|
|
|
if (!(address = g_socket_get_local_address (read_socket, &error)))
|
|
goto no_address;
|
|
|
|
g_free (priv->server_ip);
|
|
/* keep the original ip that the client connected to */
|
|
if (G_IS_INET_SOCKET_ADDRESS (address)) {
|
|
GInetAddress *iaddr;
|
|
|
|
iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
|
|
|
|
/* socket might be ipv6 but adress still ipv4 */
|
|
priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
|
|
priv->server_ip = g_inet_address_to_string (iaddr);
|
|
g_object_unref (address);
|
|
} else {
|
|
priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
|
|
priv->server_ip = g_strdup ("unknown");
|
|
}
|
|
|
|
GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
|
|
priv->server_ip, priv->is_ipv6);
|
|
|
|
url = gst_rtsp_connection_get_url (conn);
|
|
GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
|
|
|
|
priv->connection = conn;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_address:
|
|
{
|
|
GST_ERROR ("could not get local address %s", error->message);
|
|
g_error_free (error);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_get_connection:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Get the #GstRTSPConnection of @client.
|
|
*
|
|
* Returns: (transfer none): the #GstRTSPConnection of @client.
|
|
* The connection object returned remains valid until the client is freed.
|
|
*/
|
|
GstRTSPConnection *
|
|
gst_rtsp_client_get_connection (GstRTSPClient * client)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
|
|
|
|
return client->priv->connection;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_send_func:
|
|
* @client: a #GstRTSPClient
|
|
* @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
|
|
* @user_data is no longer in use.
|
|
*
|
|
* By default, the client will send the messages on the #GstRTSPConnection that
|
|
* was configured with gst_rtsp_client_attach() was called.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_send_func (GstRTSPClient * client,
|
|
GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GDestroyNotify old_notify;
|
|
gpointer old_data;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
|
|
|
|
priv = client->priv;
|
|
|
|
g_mutex_lock (&priv->send_lock);
|
|
priv->send_func = func;
|
|
old_notify = priv->send_notify;
|
|
old_data = priv->send_data;
|
|
priv->send_notify = notify;
|
|
priv->send_data = user_data;
|
|
g_mutex_unlock (&priv->send_lock);
|
|
|
|
if (old_notify)
|
|
old_notify (old_data);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_handle_message:
|
|
* @client: a #GstRTSPClient
|
|
* @message: (transfer none): an #GstRTSPMessage
|
|
*
|
|
* Let the client handle @message.
|
|
*
|
|
* Returns: a #GstRTSPResult.
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_client_handle_message (GstRTSPClient * client,
|
|
GstRTSPMessage * message)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
|
|
|
|
switch (message->type) {
|
|
case GST_RTSP_MESSAGE_REQUEST:
|
|
handle_request (client, message);
|
|
break;
|
|
case GST_RTSP_MESSAGE_RESPONSE:
|
|
handle_response (client, message);
|
|
break;
|
|
case GST_RTSP_MESSAGE_DATA:
|
|
handle_data (client, message);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_send_message:
|
|
* @client: a #GstRTSPClient
|
|
* @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.
|
|
*/
|
|
GstRTSPResult
|
|
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);
|
|
|
|
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;
|
|
}
|
|
|
|
static gboolean
|
|
do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
|
|
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);
|
|
|
|
return ret == GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
error:
|
|
{
|
|
GST_DEBUG_OBJECT (client, "got error %d", ret);
|
|
return ret == GST_RTSP_OK;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
|
|
gpointer user_data)
|
|
{
|
|
return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
|
|
}
|
|
|
|
static GstRTSPResult
|
|
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
if (priv->close_seq && priv->close_seq == cseq) {
|
|
GST_INFO ("client %p: send close message", client);
|
|
priv->close_seq = 0;
|
|
gst_rtsp_client_close (client);
|
|
}
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
closed (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
const gchar *tunnelid;
|
|
|
|
GST_INFO ("client %p: connection closed", client);
|
|
|
|
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_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 GstRTSPResult
|
|
error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
|
gchar *str;
|
|
|
|
str = gst_rtsp_strresult (result);
|
|
GST_INFO ("client %p: received an error %s", client, str);
|
|
g_free (str);
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
error_full (GstRTSPWatch * watch, GstRTSPResult result,
|
|
GstRTSPMessage * message, guint id, gpointer user_data)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
|
gchar *str;
|
|
|
|
str = gst_rtsp_strresult (result);
|
|
GST_INFO
|
|
("client %p: error when handling message %p with id %d: %s",
|
|
client, message, id, str);
|
|
g_free (str);
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static gboolean
|
|
remember_tunnel (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
const gchar *tunnelid;
|
|
|
|
/* store client in the pending tunnels */
|
|
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
|
|
if (tunnelid == NULL)
|
|
goto no_tunnelid;
|
|
|
|
GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
|
|
|
|
/* we can't have two clients connecting with the same tunnelid */
|
|
g_mutex_lock (&tunnels_lock);
|
|
if (g_hash_table_lookup (tunnels, tunnelid))
|
|
goto tunnel_existed;
|
|
|
|
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
|
|
g_mutex_unlock (&tunnels_lock);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_tunnelid:
|
|
{
|
|
GST_ERROR ("client %p: no tunnelid provided", client);
|
|
return FALSE;
|
|
}
|
|
tunnel_existed:
|
|
{
|
|
g_mutex_unlock (&tunnels_lock);
|
|
GST_ERROR ("client %p: tunnel session %s already existed", client,
|
|
tunnelid);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
GST_WARNING ("client %p: tunnel lost (connection %p)", client,
|
|
priv->connection);
|
|
|
|
/* ignore error, it'll only be a problem when the client does a POST again */
|
|
remember_tunnel (client);
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static gboolean
|
|
handle_tunnel (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPClient *oclient;
|
|
GstRTSPClientPrivate *opriv;
|
|
const gchar *tunnelid;
|
|
|
|
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
|
|
if (tunnelid == NULL)
|
|
goto no_tunnelid;
|
|
|
|
/* check for previous tunnel */
|
|
g_mutex_lock (&tunnels_lock);
|
|
oclient = g_hash_table_lookup (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);
|
|
|
|
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);
|
|
|
|
opriv = oclient->priv;
|
|
|
|
g_mutex_lock (&opriv->watch_lock);
|
|
if (opriv->watch == NULL)
|
|
goto tunnel_closed;
|
|
|
|
GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
|
|
oclient, opriv->connection, priv->connection);
|
|
|
|
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 FALSE;
|
|
}
|
|
tunnel_closed:
|
|
{
|
|
GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
|
|
g_mutex_unlock (&opriv->watch_lock);
|
|
g_object_unref (oclient);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
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_received,
|
|
message_sent,
|
|
closed,
|
|
error,
|
|
tunnel_get,
|
|
tunnel_post,
|
|
error_full,
|
|
tunnel_lost,
|
|
tunnel_http_response
|
|
};
|
|
|
|
static void
|
|
client_watch_notify (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
gboolean closed = TRUE;
|
|
|
|
GST_INFO ("client %p: watch destroyed", client);
|
|
priv->watch = NULL;
|
|
/* remove all sessions if the media says so and so drop the extra client ref */
|
|
rtsp_ctrl_timeout_remove (priv);
|
|
gst_rtsp_client_session_filter (client, cleanup_session, &closed);
|
|
if (closed)
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
|
|
g_object_unref (client);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_attach:
|
|
* @client: a #GstRTSPClient
|
|
* @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
|
|
* used).
|
|
*
|
|
* This function should be called when the client properties and urls are fully
|
|
* configured and the client is ready to start.
|
|
*
|
|
* Returns: the ID (greater than 0) for the source within the GMainContext.
|
|
*/
|
|
guint
|
|
gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GSource *timer_src;
|
|
guint res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
|
|
priv = client->priv;
|
|
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);
|
|
|
|
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
|
|
|
|
GST_INFO ("client %p: attaching to context %p", client, context);
|
|
res = gst_rtsp_watch_attach (priv->watch, context);
|
|
|
|
/* Setting up a timeout for the RTSP control channel until a session
|
|
* is up where it is handling timeouts. */
|
|
rtsp_ctrl_timeout_remove (priv); /* removing old if any */
|
|
g_mutex_lock (&priv->lock);
|
|
|
|
timer_src = g_timeout_source_new_seconds (RTSP_CTRL_CB_INTERVAL);
|
|
g_source_set_callback (timer_src, rtsp_ctrl_timeout_cb, client, NULL);
|
|
priv->rtsp_ctrl_timeout_id = g_source_attach (timer_src, priv->watch_context);
|
|
g_source_unref (timer_src);
|
|
GST_DEBUG ("rtsp control setting up session timeout id=%u.",
|
|
priv->rtsp_ctrl_timeout_id);
|
|
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_session_filter:
|
|
* @client: a #GstRTSPClient
|
|
* @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
|
|
* determines what happens to the session. @func will be called with @client
|
|
* locked so no further actions on @client can be performed from @func.
|
|
*
|
|
* If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
|
|
* @client.
|
|
*
|
|
* If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
|
|
*
|
|
* If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
|
|
* will also be added with an additional ref to the result #GList of this
|
|
* 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.
|
|
*/
|
|
GList *
|
|
gst_rtsp_client_session_filter (GstRTSPClient * client,
|
|
GstRTSPClientSessionFilterFunc func, gpointer user_data)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
GList *result, *walk, *next;
|
|
GHashTable *visited;
|
|
guint cookie;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 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);
|
|
|
|
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 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));
|
|
break;
|
|
case GST_RTSP_FILTER_KEEP:
|
|
default:
|
|
break;
|
|
}
|
|
if (changed)
|
|
goto restart;
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
if (func)
|
|
g_hash_table_unref (visited);
|
|
|
|
return result;
|
|
}
|