mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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4e9c4d8bb7
Get a thread from the configured threadpool and pass it to the prepare method of the media.
2813 lines
76 KiB
C
2813 lines
76 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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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#include <stdio.h>
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#include <string.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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GstRTSPConnection *connection;
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GstRTSPWatch *watch;
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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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gboolean use_client_settings;
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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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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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GList *transports;
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GList *sessions;
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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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#define DEFAULT_SESSION_POOL NULL
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#define DEFAULT_MOUNT_POINTS NULL
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#define DEFAULT_USE_CLIENT_SETTINGS FALSE
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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_USE_CLIENT_SETTINGS,
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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_OPTIONS_REQUEST,
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SIGNAL_DESCRIBE_REQUEST,
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SIGNAL_SETUP_REQUEST,
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SIGNAL_PLAY_REQUEST,
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SIGNAL_PAUSE_REQUEST,
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SIGNAL_TEARDOWN_REQUEST,
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SIGNAL_SET_PARAMETER_REQUEST,
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SIGNAL_GET_PARAMETER_REQUEST,
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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 void client_session_finalized (GstRTSPClient * client,
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GstRTSPSession * session);
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static void unlink_session_transports (GstRTSPClient * client,
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GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
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static gboolean default_configure_client_transport (GstRTSPClient * client,
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GstRTSPClientState * state, GstRTSPTransport * ct);
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static GstRTSPResult default_params_set (GstRTSPClient * client,
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GstRTSPClientState * state);
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static GstRTSPResult default_params_get (GstRTSPClient * client,
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GstRTSPClientState * state);
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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->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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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_USE_CLIENT_SETTINGS,
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g_param_spec_boolean ("use-client-settings", "Use Client Settings",
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"Use client settings for ttl and destination in multicast",
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DEFAULT_USE_CLIENT_SETTINGS,
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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_VOID__VOID, 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_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
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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_VOID__POINTER, G_TYPE_NONE, 1,
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G_TYPE_POINTER);
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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_VOID__POINTER, G_TYPE_NONE, 1,
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G_TYPE_POINTER);
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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_VOID__POINTER, G_TYPE_NONE, 1,
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G_TYPE_POINTER);
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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_VOID__POINTER, G_TYPE_NONE, 1,
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G_TYPE_POINTER);
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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_VOID__POINTER, G_TYPE_NONE, 1,
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G_TYPE_POINTER);
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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_VOID__POINTER, G_TYPE_NONE, 1,
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G_TYPE_POINTER);
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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_VOID__POINTER,
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G_TYPE_NONE, 1, G_TYPE_POINTER);
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gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
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g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
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get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
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G_TYPE_NONE, 1, G_TYPE_POINTER);
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tunnels =
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g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
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g_mutex_init (&tunnels_lock);
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GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
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}
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static void
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gst_rtsp_client_init (GstRTSPClient * client)
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{
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GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
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client->priv = priv;
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g_mutex_init (&priv->lock);
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g_mutex_init (&priv->send_lock);
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priv->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
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priv->close_seq = 0;
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}
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static GstRTSPFilterResult
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filter_session (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
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gpointer user_data)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
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gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
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unlink_session_transports (client, sess, sessmedia);
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/* unmanage the media in the session */
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return GST_RTSP_FILTER_REMOVE;
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}
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static void
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client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
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{
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/* unlink all media managed in this session */
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gst_rtsp_session_filter (session, filter_session, client);
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}
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static void
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client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
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{
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GstRTSPClientPrivate *priv = client->priv;
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GList *walk;
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for (walk = priv->sessions; walk; walk = g_list_next (walk)) {
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GstRTSPSession *msession = (GstRTSPSession *) walk->data;
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/* we already know about this session */
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if (msession == session)
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return;
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}
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GST_INFO ("watching session %p", session);
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g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
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client);
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priv->sessions = g_list_prepend (priv->sessions, session);
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}
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static void
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client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session)
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{
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GstRTSPClientPrivate *priv = client->priv;
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GST_INFO ("unwatching session %p", session);
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g_object_weak_unref (G_OBJECT (session),
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(GWeakNotify) client_session_finalized, client);
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priv->sessions = g_list_remove (priv->sessions, session);
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}
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static void
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client_cleanup_session (GstRTSPClient * client, GstRTSPSession * session)
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{
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g_object_weak_unref (G_OBJECT (session),
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(GWeakNotify) client_session_finalized, client);
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client_unlink_session (client, session);
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}
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static void
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client_cleanup_sessions (GstRTSPClient * client)
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{
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GstRTSPClientPrivate *priv = client->priv;
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GList *sessions;
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/* remove weak-ref from sessions */
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for (sessions = priv->sessions; sessions; sessions = g_list_next (sessions)) {
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client_cleanup_session (client, (GstRTSPSession *) sessions->data);
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}
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g_list_free (priv->sessions);
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priv->sessions = NULL;
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}
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/* A client is finalized when the connection is broken */
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static void
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gst_rtsp_client_finalize (GObject * obj)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (obj);
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GstRTSPClientPrivate *priv = client->priv;
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GST_INFO ("finalize client %p", client);
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gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
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if (priv->watch)
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g_source_destroy ((GSource *) priv->watch);
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client_cleanup_sessions (client);
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if (priv->connection)
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gst_rtsp_connection_free (priv->connection);
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if (priv->session_pool)
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g_object_unref (priv->session_pool);
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if (priv->mount_points)
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g_object_unref (priv->mount_points);
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if (priv->auth)
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g_object_unref (priv->auth);
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if (priv->path)
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g_free (priv->path);
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if (priv->media) {
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gst_rtsp_media_unprepare (priv->media);
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g_object_unref (priv->media);
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}
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g_free (priv->server_ip);
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g_mutex_clear (&priv->lock);
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g_mutex_clear (&priv->send_lock);
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G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
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}
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static void
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gst_rtsp_client_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (object);
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switch (propid) {
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case PROP_SESSION_POOL:
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g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
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break;
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case PROP_MOUNT_POINTS:
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g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
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break;
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case PROP_USE_CLIENT_SETTINGS:
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g_value_set_boolean (value,
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gst_rtsp_client_get_use_client_settings (client));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
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}
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}
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static void
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gst_rtsp_client_set_property (GObject * object, guint propid,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTSPClient *client = GST_RTSP_CLIENT (object);
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switch (propid) {
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case PROP_SESSION_POOL:
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gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
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break;
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case PROP_MOUNT_POINTS:
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gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
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break;
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case PROP_USE_CLIENT_SETTINGS:
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gst_rtsp_client_set_use_client_settings (client,
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g_value_get_boolean (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
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}
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}
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/**
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* gst_rtsp_client_new:
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*
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* Create a new #GstRTSPClient instance.
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*
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* Returns: a new #GstRTSPClient
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*/
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GstRTSPClient *
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gst_rtsp_client_new (void)
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{
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GstRTSPClient *result;
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result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
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return result;
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}
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static void
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send_message (GstRTSPClient * client, GstRTSPSession * session,
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GstRTSPMessage * message, gboolean close)
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{
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GstRTSPClientPrivate *priv = client->priv;
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gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
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"GStreamer RTSP server");
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/* remove any previous header */
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gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
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/* add the new session header for new session ids */
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if (session) {
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gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
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gst_rtsp_session_get_header (session));
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}
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if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
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gst_rtsp_message_dump (message);
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}
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if (close)
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gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
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g_mutex_lock (&priv->send_lock);
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if (priv->send_func)
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priv->send_func (client, message, close, priv->send_data);
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g_mutex_unlock (&priv->send_lock);
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gst_rtsp_message_unset (message);
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}
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static void
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send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
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GstRTSPClientState * state)
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{
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gst_rtsp_message_init_response (state->response, code,
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gst_rtsp_status_as_text (code), state->request);
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send_message (client, NULL, state->response, FALSE);
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}
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static void
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handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
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GstRTSPClientState * state)
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{
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gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
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gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
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|
if (auth) {
|
|
/* and let the authentication manager setup the auth tokens */
|
|
gst_rtsp_auth_setup (auth, state);
|
|
}
|
|
|
|
send_message (client, state->session, state->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, GstRTSPClientState * state, gint * matched)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPMediaFactory *factory;
|
|
GstRTSPMedia *media;
|
|
gchar *path;
|
|
gint path_len;
|
|
|
|
if (!priv->mount_points)
|
|
goto no_mount_points;
|
|
|
|
path = state->uri->abspath;
|
|
|
|
/* find the longest matching factory for the uri first */
|
|
if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
|
|
path, matched)))
|
|
goto no_factory;
|
|
|
|
state->factory = factory;
|
|
|
|
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)) {
|
|
GstRTSPThread *thread;
|
|
|
|
/* remove any previously cached values before we try to construct a new
|
|
* media for uri */
|
|
if (priv->path)
|
|
g_free (priv->path);
|
|
priv->path = NULL;
|
|
if (priv->media) {
|
|
gst_rtsp_media_unprepare (priv->media);
|
|
g_object_unref (priv->media);
|
|
}
|
|
priv->media = NULL;
|
|
|
|
/* prepare the media and add it to the pipeline */
|
|
if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
|
|
goto no_media;
|
|
|
|
state->media = media;
|
|
|
|
thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
|
|
GST_RTSP_THREAD_TYPE_MEDIA, state);
|
|
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;
|
|
state->media = media;
|
|
GST_INFO ("reusing cached media %p for path %s", media, priv->path);
|
|
}
|
|
|
|
g_object_unref (factory);
|
|
state->factory = NULL;
|
|
|
|
if (media)
|
|
g_object_ref (media);
|
|
|
|
return media;
|
|
|
|
/* ERRORS */
|
|
no_mount_points:
|
|
{
|
|
GST_ERROR ("client %p: no mount points configured", client);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
return NULL;
|
|
}
|
|
no_factory:
|
|
{
|
|
GST_ERROR ("client %p: no factory for uri %s", client, path);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
return NULL;
|
|
}
|
|
no_factory_access:
|
|
{
|
|
GST_ERROR ("client %p: not authorized to see factory uri %s", client, path);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
return NULL;
|
|
}
|
|
not_authorized:
|
|
{
|
|
GST_ERROR ("client %p: not authorized for factory uri %s", client, path);
|
|
handle_unauthorized_request (client, priv->auth, state);
|
|
return NULL;
|
|
}
|
|
no_media:
|
|
{
|
|
GST_ERROR ("client %p: can't create media", client);
|
|
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
|
|
g_object_unref (factory);
|
|
state->factory = NULL;
|
|
return NULL;
|
|
}
|
|
no_thread:
|
|
{
|
|
GST_ERROR ("client %p: can't create thread", client);
|
|
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
|
|
g_object_unref (media);
|
|
state->media = NULL;
|
|
g_object_unref (factory);
|
|
state->factory = NULL;
|
|
return NULL;
|
|
}
|
|
no_prepare:
|
|
{
|
|
GST_ERROR ("client %p: can't prepare media", client);
|
|
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
|
|
g_object_unref (media);
|
|
state->media = NULL;
|
|
g_object_unref (factory);
|
|
state->factory = NULL;
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPMessage message = { 0 };
|
|
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)
|
|
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 TRUE;
|
|
}
|
|
|
|
static void
|
|
link_transport (GstRTSPClient * client, GstRTSPSession * session,
|
|
GstRTSPStreamTransport * trans)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
GST_DEBUG ("client %p: linking transport %p", client, trans);
|
|
|
|
gst_rtsp_stream_transport_set_callbacks (trans,
|
|
(GstRTSPSendFunc) do_send_data,
|
|
(GstRTSPSendFunc) do_send_data, client, NULL);
|
|
|
|
priv->transports = g_list_prepend (priv->transports, trans);
|
|
|
|
/* make sure our session can't expire */
|
|
gst_rtsp_session_prevent_expire (session);
|
|
}
|
|
|
|
static void
|
|
unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
|
|
GstRTSPStreamTransport * trans)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
GST_DEBUG ("client %p: unlinking transport %p", client, trans);
|
|
|
|
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
|
|
|
|
priv->transports = g_list_remove (priv->transports, trans);
|
|
|
|
/* our session can now expire */
|
|
gst_rtsp_session_allow_expire (session);
|
|
}
|
|
|
|
static void
|
|
unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
|
|
GstRTSPSessionMedia * sessmedia)
|
|
{
|
|
guint n_streams, i;
|
|
|
|
n_streams =
|
|
gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
|
|
for (i = 0; i < n_streams; i++) {
|
|
GstRTSPStreamTransport *trans;
|
|
const GstRTSPTransport *tr;
|
|
|
|
/* get the transport, if there is no transport configured, skip this stream */
|
|
trans = gst_rtsp_session_media_get_transport (sessmedia, i);
|
|
if (trans == NULL)
|
|
continue;
|
|
|
|
tr = gst_rtsp_stream_transport_get_transport (trans);
|
|
|
|
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
|
|
/* for TCP, unlink the stream from the TCP connection of the client */
|
|
unlink_transport (client, session, trans);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void
|
|
close_connection (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
const gchar *tunnelid;
|
|
|
|
GST_DEBUG ("client %p: closing connection", 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_connection_close (priv->connection);
|
|
}
|
|
|
|
static gboolean
|
|
handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPSession *session;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPStatusCode code;
|
|
const gchar *path;
|
|
gint matched;
|
|
|
|
if (!state->session)
|
|
goto no_session;
|
|
|
|
session = state->session;
|
|
|
|
if (!state->uri)
|
|
goto no_uri;
|
|
|
|
path = state->uri->abspath;
|
|
|
|
/* 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;
|
|
|
|
state->sessmedia = sessmedia;
|
|
|
|
/* we emit the signal before closing the connection */
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
|
|
0, state);
|
|
|
|
/* unlink the all TCP callbacks */
|
|
unlink_session_transports (client, session, sessmedia);
|
|
|
|
/* remove the session from the watched sessions */
|
|
client_unwatch_session (client, session);
|
|
|
|
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
|
|
|
|
/* unmanage the media in the session, returns false if all media session
|
|
* are torn down. */
|
|
if (!gst_rtsp_session_release_media (session, sessmedia)) {
|
|
/* remove the session */
|
|
gst_rtsp_session_pool_remove (priv->session_pool, session);
|
|
}
|
|
/* construct the response now */
|
|
code = GST_RTSP_STS_OK;
|
|
gst_rtsp_message_init_response (state->response, code,
|
|
gst_rtsp_status_as_text (code), state->request);
|
|
|
|
send_message (client, session, state->response, TRUE);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_session:
|
|
{
|
|
GST_ERROR ("client %p: no session", client);
|
|
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
no_uri:
|
|
{
|
|
GST_ERROR ("client %p: no uri supplied", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
return FALSE;
|
|
}
|
|
not_found:
|
|
{
|
|
GST_ERROR ("client %p: no media for uri", client);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
no_aggregate:
|
|
{
|
|
GST_ERROR ("client %p: no aggregate path %s", client, path);
|
|
send_generic_response (client,
|
|
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
default_params_set (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
res = gst_rtsp_params_set (client, state);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
default_params_get (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPResult res;
|
|
|
|
res = gst_rtsp_params_get (client, state);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPResult res;
|
|
guint8 *data;
|
|
guint size;
|
|
|
|
res = gst_rtsp_message_get_body (state->request, &data, &size);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
if (size == 0) {
|
|
/* no body, keep-alive request */
|
|
send_generic_response (client, GST_RTSP_STS_OK, state);
|
|
} else {
|
|
/* there is a body, handle the params */
|
|
res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, state);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
send_message (client, state->session, state->response, FALSE);
|
|
}
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
|
|
0, state);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
bad_request:
|
|
{
|
|
GST_ERROR ("client %p: bad request", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPResult res;
|
|
guint8 *data;
|
|
guint size;
|
|
|
|
res = gst_rtsp_message_get_body (state->request, &data, &size);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
if (size == 0) {
|
|
/* no body, keep-alive request */
|
|
send_generic_response (client, GST_RTSP_STS_OK, state);
|
|
} else {
|
|
/* there is a body, handle the params */
|
|
res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, state);
|
|
if (res != GST_RTSP_OK)
|
|
goto bad_request;
|
|
|
|
send_message (client, state->session, state->response, FALSE);
|
|
}
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
|
|
0, state);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
bad_request:
|
|
{
|
|
GST_ERROR ("client %p: bad request", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPSession *session;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPStatusCode code;
|
|
GstRTSPState rtspstate;
|
|
const gchar *path;
|
|
gint matched;
|
|
|
|
if (!(session = state->session))
|
|
goto no_session;
|
|
|
|
if (!state->uri)
|
|
goto no_uri;
|
|
|
|
path = state->uri->abspath;
|
|
|
|
/* 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;
|
|
|
|
state->sessmedia = sessmedia;
|
|
|
|
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;
|
|
|
|
/* unlink the all TCP callbacks */
|
|
unlink_session_transports (client, session, sessmedia);
|
|
|
|
/* 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 (state->response, code,
|
|
gst_rtsp_status_as_text (code), state->request);
|
|
|
|
send_message (client, session, state->response, FALSE);
|
|
|
|
/* the state is now READY */
|
|
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
|
|
0, state);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_session:
|
|
{
|
|
GST_ERROR ("client %p: no seesion", client);
|
|
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
no_uri:
|
|
{
|
|
GST_ERROR ("client %p: no uri supplied", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
return FALSE;
|
|
}
|
|
not_found:
|
|
{
|
|
GST_ERROR ("client %p: no media for uri", client);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
no_aggregate:
|
|
{
|
|
GST_ERROR ("client %p: no aggregate path %s", client, path);
|
|
send_generic_response (client,
|
|
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
|
|
return FALSE;
|
|
}
|
|
invalid_state:
|
|
{
|
|
GST_ERROR ("client %p: not PLAYING or RECORDING", client);
|
|
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
|
|
state);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPSession *session;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPMedia *media;
|
|
GstRTSPStatusCode code;
|
|
GString *rtpinfo;
|
|
guint n_streams, i, infocount;
|
|
gchar *str;
|
|
GstRTSPTimeRange *range;
|
|
GstRTSPResult res;
|
|
GstRTSPState rtspstate;
|
|
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
|
|
const gchar *path;
|
|
gint matched;
|
|
|
|
if (!(session = state->session))
|
|
goto no_session;
|
|
|
|
if (!state->uri)
|
|
goto no_uri;
|
|
|
|
path = state->uri->abspath;
|
|
|
|
/* 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;
|
|
|
|
state->sessmedia = sessmedia;
|
|
state->media = media = gst_rtsp_session_media_get_media (sessmedia);
|
|
|
|
/* the session state must be playing or ready */
|
|
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
|
|
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
|
|
goto invalid_state;
|
|
|
|
/* parse the range header if we have one */
|
|
res =
|
|
gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
|
|
if (res == GST_RTSP_OK) {
|
|
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
|
|
/* we have a range, seek to the position */
|
|
unit = range->unit;
|
|
gst_rtsp_media_seek (media, range);
|
|
gst_rtsp_range_free (range);
|
|
}
|
|
}
|
|
|
|
/* grab RTPInfo from the payloaders now */
|
|
rtpinfo = g_string_new ("");
|
|
|
|
n_streams = gst_rtsp_media_n_streams (media);
|
|
for (i = 0, infocount = 0; i < n_streams; i++) {
|
|
GstRTSPStreamTransport *trans;
|
|
GstRTSPStream *stream;
|
|
const GstRTSPTransport *tr;
|
|
gchar *uristr;
|
|
guint rtptime, seq;
|
|
|
|
/* get the transport, if there is no transport configured, skip this stream */
|
|
trans = gst_rtsp_session_media_get_transport (sessmedia, i);
|
|
if (trans == NULL) {
|
|
GST_INFO ("stream %d is not configured", i);
|
|
continue;
|
|
}
|
|
tr = gst_rtsp_stream_transport_get_transport (trans);
|
|
|
|
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
|
|
/* for TCP, link the stream to the TCP connection of the client */
|
|
link_transport (client, session, trans);
|
|
}
|
|
|
|
stream = gst_rtsp_stream_transport_get_stream (trans);
|
|
if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
|
|
if (infocount > 0)
|
|
g_string_append (rtpinfo, ", ");
|
|
|
|
uristr = gst_rtsp_url_get_request_uri (state->uri);
|
|
g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
|
|
uristr, i, seq, rtptime);
|
|
g_free (uristr);
|
|
|
|
infocount++;
|
|
} else {
|
|
GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
|
|
}
|
|
}
|
|
|
|
/* construct the response now */
|
|
code = GST_RTSP_STS_OK;
|
|
gst_rtsp_message_init_response (state->response, code,
|
|
gst_rtsp_status_as_text (code), state->request);
|
|
|
|
/* add the RTP-Info header */
|
|
if (infocount > 0) {
|
|
str = g_string_free (rtpinfo, FALSE);
|
|
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
|
|
} else {
|
|
g_string_free (rtpinfo, TRUE);
|
|
}
|
|
|
|
/* add the range */
|
|
str = gst_rtsp_media_get_range_string (media, TRUE, unit);
|
|
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
|
|
|
|
send_message (client, session, state->response, FALSE);
|
|
|
|
/* start playing after sending the request */
|
|
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
|
|
|
|
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
|
|
0, state);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_session:
|
|
{
|
|
GST_ERROR ("client %p: no session", client);
|
|
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
no_uri:
|
|
{
|
|
GST_ERROR ("client %p: no uri supplied", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
return FALSE;
|
|
}
|
|
not_found:
|
|
{
|
|
GST_ERROR ("client %p: media not found", client);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
no_aggregate:
|
|
{
|
|
GST_ERROR ("client %p: no aggregate path %s", client, path);
|
|
send_generic_response (client,
|
|
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, state);
|
|
return FALSE;
|
|
}
|
|
invalid_state:
|
|
{
|
|
GST_ERROR ("client %p: not PLAYING or READY", client);
|
|
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
|
|
state);
|
|
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
|
|
* from @supported are returned. Returns FALSE if no valid transport
|
|
* was found. */
|
|
static gboolean
|
|
parse_transport (const char *transport, GstRTSPLowerTrans supported,
|
|
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 RTP/AVP */
|
|
if (tr->trans != GST_RTSP_TRANS_RTP || tr->profile != GST_RTSP_PROFILE_AVP) {
|
|
GST_WARNING ("invalid transport %s", transports[i]);
|
|
goto next;
|
|
}
|
|
|
|
if (!(tr->lower_transport & supported)) {
|
|
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
|
|
handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
|
|
GstRTSPMessage * request)
|
|
{
|
|
gchar *blocksize_str;
|
|
gboolean ret = TRUE;
|
|
|
|
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) {
|
|
GST_ERROR ("failed to parse blocksize");
|
|
ret = FALSE;
|
|
} else {
|
|
/* we don't want to change the mtu when this media
|
|
* can be shared because it impacts other clients */
|
|
if (gst_rtsp_media_is_shared (media))
|
|
return TRUE;
|
|
|
|
if (blocksize > G_MAXUINT)
|
|
blocksize = G_MAXUINT;
|
|
gst_rtsp_stream_set_mtu (stream, blocksize);
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
default_configure_client_transport (GstRTSPClient * client,
|
|
GstRTSPClientState * state, 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) {
|
|
if (ct->destination && priv->use_client_settings) {
|
|
GstRTSPAddress *addr;
|
|
|
|
addr = gst_rtsp_stream_reserve_address (state->stream, ct->destination,
|
|
ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
|
|
|
|
if (addr == NULL)
|
|
goto no_address;
|
|
|
|
gst_rtsp_address_free (addr);
|
|
} else {
|
|
GstRTSPAddress *addr;
|
|
GSocketFamily family;
|
|
|
|
family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
|
|
|
|
addr = gst_rtsp_stream_get_multicast_address (state->stream, family);
|
|
if (addr == NULL)
|
|
goto no_address;
|
|
|
|
g_free (ct->destination);
|
|
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);
|
|
|
|
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
|
|
/* check if the client selected channels for TCP */
|
|
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
|
|
gst_rtsp_session_media_alloc_channels (state->sessmedia,
|
|
&ct->interleaved);
|
|
}
|
|
}
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_address:
|
|
{
|
|
GST_ERROR_OBJECT (client, "failed to acquire address for stream");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstRTSPTransport *
|
|
make_server_transport (GstRTSPClient * client, GstRTSPClientState * state,
|
|
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;
|
|
|
|
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 (state->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;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
gst_rtsp_stream_get_ssrc (state->stream, &st->ssrc);
|
|
|
|
return st;
|
|
}
|
|
|
|
static gboolean
|
|
handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPResult res;
|
|
GstRTSPUrl *uri;
|
|
gchar *transport;
|
|
GstRTSPTransport *ct, *st;
|
|
GstRTSPLowerTrans supported;
|
|
GstRTSPStatusCode code;
|
|
GstRTSPSession *session;
|
|
GstRTSPStreamTransport *trans;
|
|
gchar *trans_str;
|
|
GstRTSPSessionMedia *sessmedia;
|
|
GstRTSPMedia *media;
|
|
GstRTSPStream *stream;
|
|
GstRTSPState rtspstate;
|
|
GstRTSPClientClass *klass;
|
|
gchar *path, *control;
|
|
gint matched;
|
|
|
|
if (!state->uri)
|
|
goto no_uri;
|
|
|
|
uri = state->uri;
|
|
path = uri->abspath;
|
|
|
|
/* parse the transport */
|
|
res =
|
|
gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
|
|
&transport, 0);
|
|
if (res != GST_RTSP_OK)
|
|
goto no_transport;
|
|
|
|
/* we create the session after parsing stuff so that we don't make
|
|
* a session for malformed requests */
|
|
if (priv->session_pool == NULL)
|
|
goto no_pool;
|
|
|
|
session = state->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, state, &matched);
|
|
} else {
|
|
if ((media = gst_rtsp_session_media_get_media (sessmedia)))
|
|
g_object_ref (media);
|
|
}
|
|
/* no media, not found then */
|
|
if (media == NULL)
|
|
goto media_not_found;
|
|
|
|
/* path is what matched. We can modify the parsed uri in place */
|
|
path[matched] = '\0';
|
|
/* control is remainder */
|
|
control = &path[matched + 1];
|
|
|
|
/* find the stream now using the control part */
|
|
stream = gst_rtsp_media_find_stream (media, control);
|
|
if (stream == NULL)
|
|
goto stream_not_found;
|
|
|
|
/* now we have a uri identifying a valid media and stream */
|
|
state->stream = stream;
|
|
state->media = media;
|
|
|
|
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;
|
|
|
|
/* make sure this client is closed when the session is closed */
|
|
client_watch_session (client, session);
|
|
|
|
/* signal new session */
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
|
|
session);
|
|
|
|
state->session = session;
|
|
}
|
|
|
|
if (sessmedia == NULL) {
|
|
/* manage the media in our session now, if not done already */
|
|
sessmedia = gst_rtsp_session_manage_media (session, path, media);
|
|
/* if we stil have no media, error */
|
|
if (sessmedia == NULL)
|
|
goto sessmedia_unavailable;
|
|
} else {
|
|
g_object_unref (media);
|
|
}
|
|
|
|
state->sessmedia = sessmedia;
|
|
|
|
/* set blocksize on this stream */
|
|
if (!handle_blocksize (media, stream, state->request))
|
|
goto invalid_blocksize;
|
|
|
|
gst_rtsp_transport_new (&ct);
|
|
|
|
/* our supported transports */
|
|
supported = GST_RTSP_LOWER_TRANS_UDP |
|
|
GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
|
|
|
|
/* parse and find a usable supported transport */
|
|
if (!parse_transport (transport, supported, ct))
|
|
goto unsupported_transports;
|
|
|
|
/* update the client transport */
|
|
klass = GST_RTSP_CLIENT_GET_CLASS (client);
|
|
if (!klass->configure_client_transport (client, state, ct))
|
|
goto unsupported_client_transport;
|
|
|
|
/* set in the session media transport */
|
|
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
|
|
|
|
/* configure keepalive for this transport */
|
|
gst_rtsp_stream_transport_set_keepalive (trans,
|
|
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
|
|
|
|
/* create and serialize the server transport */
|
|
st = make_server_transport (client, state, ct);
|
|
trans_str = gst_rtsp_transport_as_text (st);
|
|
gst_rtsp_transport_free (st);
|
|
|
|
/* construct the response now */
|
|
code = GST_RTSP_STS_OK;
|
|
gst_rtsp_message_init_response (state->response, code,
|
|
gst_rtsp_status_as_text (code), state->request);
|
|
|
|
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
|
|
trans_str);
|
|
g_free (trans_str);
|
|
|
|
send_message (client, session, state->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 (session);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
|
|
0, state);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_uri:
|
|
{
|
|
GST_ERROR ("client %p: no uri", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
return FALSE;
|
|
}
|
|
no_transport:
|
|
{
|
|
GST_ERROR ("client %p: no transport", client);
|
|
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
|
|
return FALSE;
|
|
}
|
|
no_pool:
|
|
{
|
|
GST_ERROR ("client %p: no session pool configured", client);
|
|
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
media_not_found:
|
|
{
|
|
GST_ERROR ("client %p: media '%s' not found", client, path);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
return FALSE;
|
|
}
|
|
stream_not_found:
|
|
{
|
|
GST_ERROR ("client %p: stream '%s' not found", client, control);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
|
|
g_object_unref (media);
|
|
return FALSE;
|
|
}
|
|
service_unavailable:
|
|
{
|
|
GST_ERROR ("client %p: can't create session", client);
|
|
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
|
|
g_object_unref (media);
|
|
return FALSE;
|
|
}
|
|
sessmedia_unavailable:
|
|
{
|
|
GST_ERROR ("client %p: can't create session media", client);
|
|
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
|
|
g_object_unref (media);
|
|
g_object_unref (session);
|
|
return FALSE;
|
|
}
|
|
invalid_blocksize:
|
|
{
|
|
GST_ERROR ("client %p: invalid blocksize", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
g_object_unref (session);
|
|
return FALSE;
|
|
}
|
|
unsupported_transports:
|
|
{
|
|
GST_ERROR ("client %p: unsupported transports", client);
|
|
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
|
|
gst_rtsp_transport_free (ct);
|
|
g_object_unref (session);
|
|
return FALSE;
|
|
}
|
|
unsupported_client_transport:
|
|
{
|
|
GST_ERROR ("client %p: unsupported client transport", client);
|
|
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
|
|
gst_rtsp_transport_free (ct);
|
|
g_object_unref (session);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static GstSDPMessage *
|
|
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstSDPMessage *sdp;
|
|
GstSDPInfo info;
|
|
const gchar *proto;
|
|
|
|
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";
|
|
|
|
gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "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_sdp_from_media (sdp, &info, media))
|
|
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, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPResult res;
|
|
GstSDPMessage *sdp;
|
|
guint i, str_len;
|
|
gchar *str, *content_base;
|
|
GstRTSPMedia *media;
|
|
GstRTSPClientClass *klass;
|
|
|
|
klass = GST_RTSP_CLIENT_GET_CLASS (client);
|
|
|
|
if (!state->uri)
|
|
goto no_uri;
|
|
|
|
/* check what kind of format is accepted, we don't really do anything with it
|
|
* and always return SDP for now. */
|
|
for (i = 0; i++;) {
|
|
gchar *accept;
|
|
|
|
res =
|
|
gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
|
|
&accept, i);
|
|
if (res == GST_RTSP_ENOTIMPL)
|
|
break;
|
|
|
|
if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
|
|
break;
|
|
}
|
|
|
|
/* find the media object for the uri */
|
|
if (!(media = find_media (client, state, NULL)))
|
|
goto no_media;
|
|
|
|
/* create an SDP for the media object on this client */
|
|
if (!(sdp = klass->create_sdp (client, media)))
|
|
goto no_sdp;
|
|
|
|
g_object_unref (media);
|
|
|
|
gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
|
|
gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
|
|
|
|
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
|
|
"application/sdp");
|
|
|
|
/* content base for some clients that might screw up creating the setup uri */
|
|
str = gst_rtsp_url_get_request_uri (state->uri);
|
|
str_len = strlen (str);
|
|
|
|
/* check for trailing '/' and append one */
|
|
if (str[str_len - 1] != '/') {
|
|
content_base = g_malloc (str_len + 2);
|
|
memcpy (content_base, str, str_len);
|
|
content_base[str_len] = '/';
|
|
content_base[str_len + 1] = '\0';
|
|
g_free (str);
|
|
} else {
|
|
content_base = str;
|
|
}
|
|
|
|
GST_INFO ("adding content-base: %s", content_base);
|
|
|
|
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
|
|
content_base);
|
|
g_free (content_base);
|
|
|
|
/* add SDP to the response body */
|
|
str = gst_sdp_message_as_text (sdp);
|
|
gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
|
|
gst_sdp_message_free (sdp);
|
|
|
|
send_message (client, state->session, state->response, FALSE);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
|
|
0, state);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_uri:
|
|
{
|
|
GST_ERROR ("client %p: no uri", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
|
|
return FALSE;
|
|
}
|
|
no_media:
|
|
{
|
|
GST_ERROR ("client %p: no media", client);
|
|
/* error reply is already sent */
|
|
return FALSE;
|
|
}
|
|
no_sdp:
|
|
{
|
|
GST_ERROR ("client %p: can't create SDP", client);
|
|
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
|
|
g_object_unref (media);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
|
|
{
|
|
GstRTSPMethod options;
|
|
gchar *str;
|
|
|
|
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;
|
|
|
|
str = gst_rtsp_options_as_text (options);
|
|
|
|
gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
|
|
gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
|
|
|
|
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
|
|
g_free (str);
|
|
|
|
send_message (client, state->session, state->response, FALSE);
|
|
|
|
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
|
|
0, state);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* 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';
|
|
}
|
|
|
|
static void
|
|
client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
GST_INFO ("client %p: session %p finished", client, session);
|
|
|
|
/* unlink all media managed in this session */
|
|
client_unlink_session (client, session);
|
|
|
|
/* remove the session */
|
|
if (!(priv->sessions = g_list_remove (priv->sessions, session))) {
|
|
GST_INFO ("client %p: all sessions finalized, close the connection",
|
|
client);
|
|
close_connection (client);
|
|
}
|
|
}
|
|
|
|
static GPrivate state_key;
|
|
|
|
/**
|
|
* gst_rtsp_client_state_get_current:
|
|
*
|
|
* Get the current #GstRTSPClientState. This object is retrieved from the
|
|
* current thread that is handling the request for a client.
|
|
*
|
|
* Returns: a #GstRTSPClientState
|
|
*/
|
|
GstRTSPClientState *
|
|
gst_rtsp_client_state_get_current (void)
|
|
{
|
|
return g_private_get (&state_key);
|
|
}
|
|
|
|
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;
|
|
GstRTSPClientState state = { NULL };
|
|
GstRTSPMessage response = { 0 };
|
|
gchar *sessid;
|
|
|
|
state.client = client;
|
|
state.request = request;
|
|
state.response = &response;
|
|
state.auth = priv->auth;
|
|
g_private_set (&state_key, &state);
|
|
|
|
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
|
|
gst_rtsp_message_dump (request);
|
|
}
|
|
|
|
GST_INFO ("client %p: received a request", client);
|
|
|
|
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
|
|
|
|
/* we can only handle 1.0 requests */
|
|
if (version != GST_RTSP_VERSION_1_0)
|
|
goto not_supported;
|
|
|
|
state.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)
|
|
goto bad_request;
|
|
|
|
/* get the session if there is any */
|
|
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);
|
|
state.uri = uri;
|
|
state.session = session;
|
|
|
|
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
|
|
goto not_authorized;
|
|
|
|
/* now see what is asked and dispatch to a dedicated handler */
|
|
switch (method) {
|
|
case GST_RTSP_OPTIONS:
|
|
handle_options_request (client, &state);
|
|
break;
|
|
case GST_RTSP_DESCRIBE:
|
|
handle_describe_request (client, &state);
|
|
break;
|
|
case GST_RTSP_SETUP:
|
|
handle_setup_request (client, &state);
|
|
break;
|
|
case GST_RTSP_PLAY:
|
|
handle_play_request (client, &state);
|
|
break;
|
|
case GST_RTSP_PAUSE:
|
|
handle_pause_request (client, &state);
|
|
break;
|
|
case GST_RTSP_TEARDOWN:
|
|
handle_teardown_request (client, &state);
|
|
break;
|
|
case GST_RTSP_SET_PARAMETER:
|
|
handle_set_param_request (client, &state);
|
|
break;
|
|
case GST_RTSP_GET_PARAMETER:
|
|
handle_get_param_request (client, &state);
|
|
break;
|
|
case GST_RTSP_ANNOUNCE:
|
|
case GST_RTSP_RECORD:
|
|
case GST_RTSP_REDIRECT:
|
|
goto not_implemented;
|
|
case GST_RTSP_INVALID:
|
|
default:
|
|
goto bad_request;
|
|
}
|
|
|
|
done:
|
|
g_private_set (&state_key, NULL);
|
|
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,
|
|
&state);
|
|
goto done;
|
|
}
|
|
bad_request:
|
|
{
|
|
GST_ERROR ("client %p: bad request", client);
|
|
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
|
|
goto done;
|
|
}
|
|
no_pool:
|
|
{
|
|
GST_ERROR ("client %p: no pool configured", client);
|
|
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
|
|
goto done;
|
|
}
|
|
session_not_found:
|
|
{
|
|
GST_ERROR ("client %p: session not found", client);
|
|
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
|
|
goto done;
|
|
}
|
|
not_authorized:
|
|
{
|
|
GST_ERROR ("client %p: not allowed", client);
|
|
handle_unauthorized_request (client, priv->auth, &state);
|
|
goto done;
|
|
}
|
|
not_implemented:
|
|
{
|
|
GST_ERROR ("client %p: method %d not implemented", client, method);
|
|
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static void
|
|
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPResult res;
|
|
guint8 channel;
|
|
GList *walk;
|
|
guint8 *data;
|
|
guint size;
|
|
GstBuffer *buffer;
|
|
gboolean handled;
|
|
|
|
/* find the stream for this message */
|
|
res = gst_rtsp_message_parse_data (message, &channel);
|
|
if (res != GST_RTSP_OK)
|
|
return;
|
|
|
|
gst_rtsp_message_steal_body (message, &data, &size);
|
|
|
|
buffer = gst_buffer_new_wrapped (data, size);
|
|
|
|
handled = FALSE;
|
|
for (walk = priv->transports; walk; walk = g_list_next (walk)) {
|
|
GstRTSPStreamTransport *trans;
|
|
GstRTSPStream *stream;
|
|
const GstRTSPTransport *tr;
|
|
|
|
trans = walk->data;
|
|
|
|
tr = gst_rtsp_stream_transport_get_transport (trans);
|
|
stream = gst_rtsp_stream_transport_get_stream (trans);
|
|
|
|
/* check for TCP transport */
|
|
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
|
|
/* dispatch to the stream based on the channel number */
|
|
if (tr->interleaved.min == channel) {
|
|
gst_rtsp_stream_recv_rtp (stream, buffer);
|
|
handled = TRUE;
|
|
break;
|
|
} else if (tr->interleaved.max == channel) {
|
|
gst_rtsp_stream_recv_rtcp (stream, buffer);
|
|
handled = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
if (!handled)
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_session_pool:
|
|
* @client: a #GstRTSPClient
|
|
* @pool: 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;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
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: 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_use_client_settings:
|
|
* @client: a #GstRTSPClient
|
|
* @use_client_settings: whether to use client settings for multicast
|
|
*
|
|
* Use client transport settings (destination and ttl) for multicast.
|
|
* When @use_client_settings is %FALSE, the server settings will be
|
|
* used.
|
|
*/
|
|
void
|
|
gst_rtsp_client_set_use_client_settings (GstRTSPClient * client,
|
|
gboolean use_client_settings)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
|
|
|
|
priv = client->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
priv->use_client_settings = use_client_settings;
|
|
g_mutex_unlock (&priv->lock);
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_get_use_client_settings:
|
|
* @client: a #GstRTSPClient
|
|
*
|
|
* Check if client transport settings (destination and ttl) for multicast
|
|
* will be used.
|
|
*/
|
|
gboolean
|
|
gst_rtsp_client_get_use_client_settings (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv;
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
|
|
|
|
priv = client->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
res = priv->use_client_settings;
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_set_auth:
|
|
* @client: a #GstRTSPClient
|
|
* @auth: 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: 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: a #GstRTSPClientSendFunc
|
|
* @user_data: user data passed to @func
|
|
* @notify: 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.
|
|
*/
|
|
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: 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:
|
|
break;
|
|
case GST_RTSP_MESSAGE_DATA:
|
|
handle_data (client, message);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_send_request:
|
|
* @client: a #GstRTSPClient
|
|
* @session: a #GstRTSPSession to send the request to or %NULL
|
|
* @request: The request #GstRTSPMessage to send
|
|
*
|
|
* Send a request message to the client.
|
|
*/
|
|
GstRTSPResult
|
|
gst_rtsp_client_send_request (GstRTSPClient * client, GstRTSPSession * session,
|
|
GstRTSPMessage * request)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (request != NULL, GST_RTSP_EINVAL);
|
|
g_return_val_if_fail (request->type == GST_RTSP_MESSAGE_REQUEST,
|
|
GST_RTSP_EINVAL);
|
|
|
|
send_message (client, session, request, FALSE);
|
|
|
|
return GST_RTSP_OK;
|
|
}
|
|
|
|
static GstRTSPResult
|
|
do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
|
|
gboolean close, gpointer user_data)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
/* send the response and store the seq number so we can wait until it's
|
|
* written to the client to close the connection */
|
|
return gst_rtsp_watch_send_message (priv->watch, message, close ?
|
|
&priv->close_seq : NULL);
|
|
}
|
|
|
|
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) {
|
|
priv->close_seq = 0;
|
|
close_connection (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_client_set_send_func (client, NULL, NULL, NULL);
|
|
|
|
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 GstRTSPStatusCode
|
|
tunnel_start (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
GST_INFO ("client %p: tunnel start (connection %p)", client,
|
|
priv->connection);
|
|
|
|
if (!remember_tunnel (client))
|
|
goto tunnel_error;
|
|
|
|
return GST_RTSP_STS_OK;
|
|
|
|
/* ERRORS */
|
|
tunnel_error:
|
|
{
|
|
GST_ERROR ("client %p: error starting tunnel", client);
|
|
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
|
|
}
|
|
}
|
|
|
|
static GstRTSPResult
|
|
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
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 GstRTSPResult
|
|
tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
|
|
{
|
|
const gchar *tunnelid;
|
|
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
GstRTSPClient *oclient;
|
|
GstRTSPClientPrivate *opriv;
|
|
|
|
GST_INFO ("client %p: tunnel complete", client);
|
|
|
|
/* find previous tunnel */
|
|
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
|
|
if (tunnelid == NULL)
|
|
goto no_tunnelid;
|
|
|
|
g_mutex_lock (&tunnels_lock);
|
|
if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
|
|
goto no_tunnel;
|
|
|
|
/* 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);
|
|
|
|
opriv = oclient->priv;
|
|
|
|
if (opriv->watch == NULL)
|
|
goto tunnel_closed;
|
|
g_mutex_unlock (&tunnels_lock);
|
|
|
|
GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
|
|
opriv->connection, priv->connection);
|
|
|
|
/* merge the tunnels into the first client */
|
|
gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
|
|
gst_rtsp_watch_reset (opriv->watch);
|
|
g_object_unref (oclient);
|
|
|
|
return GST_RTSP_OK;
|
|
|
|
/* ERRORS */
|
|
no_tunnelid:
|
|
{
|
|
GST_ERROR ("client %p: no tunnelid provided", client);
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
no_tunnel:
|
|
{
|
|
g_mutex_unlock (&tunnels_lock);
|
|
GST_ERROR ("client %p: tunnel session %s not found", client, tunnelid);
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
tunnel_closed:
|
|
{
|
|
g_mutex_unlock (&tunnels_lock);
|
|
GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
|
|
g_object_unref (oclient);
|
|
return GST_RTSP_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstRTSPWatchFuncs watch_funcs = {
|
|
message_received,
|
|
message_sent,
|
|
closed,
|
|
error,
|
|
tunnel_start,
|
|
tunnel_complete,
|
|
error_full,
|
|
tunnel_lost
|
|
};
|
|
|
|
static void
|
|
client_watch_notify (GstRTSPClient * client)
|
|
{
|
|
GstRTSPClientPrivate *priv = client->priv;
|
|
|
|
GST_INFO ("client %p: watch destroyed", client);
|
|
priv->watch = NULL;
|
|
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;
|
|
guint res;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
|
|
priv = client->priv;
|
|
g_return_val_if_fail (priv->watch == NULL, 0);
|
|
|
|
/* create watch for the connection and attach */
|
|
priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
|
|
g_object_ref (client), (GDestroyNotify) client_watch_notify);
|
|
gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
|
|
(GDestroyNotify) gst_rtsp_watch_unref);
|
|
|
|
/* FIXME make this configurable. We don't want to do this yet because it will
|
|
* be superceeded by a cache object later */
|
|
gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
|
|
|
|
GST_INFO ("attaching to context %p", context);
|
|
res = gst_rtsp_watch_attach (priv->watch, context);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_client_session_filter:
|
|
* @client: a #GstRTSPclient
|
|
* @func: (scope call): 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..
|
|
*
|
|
* 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;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
|
|
g_return_val_if_fail (func != NULL, NULL);
|
|
|
|
priv = client->priv;
|
|
|
|
result = NULL;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
for (walk = priv->sessions; walk; walk = next) {
|
|
GstRTSPSession *sess = walk->data;
|
|
|
|
next = g_list_next (walk);
|
|
|
|
switch (func (client, sess, user_data)) {
|
|
case GST_RTSP_FILTER_REMOVE:
|
|
/* stop watching the session and pretent it went away */
|
|
client_cleanup_session (client, sess);
|
|
break;
|
|
case GST_RTSP_FILTER_REF:
|
|
result = g_list_prepend (result, g_object_ref (sess));
|
|
break;
|
|
case GST_RTSP_FILTER_KEEP:
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
g_mutex_unlock (&priv->lock);
|
|
|
|
return result;
|
|
}
|