mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ebc28a47da
Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP connection.
613 lines
16 KiB
C
613 lines
16 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <sys/ioctl.h>
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#include "rtsp-server.h"
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#include "rtsp-client.h"
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#define DEFAULT_BACKLOG 5
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#define DEFAULT_PORT 8554
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enum
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{
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PROP_0,
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PROP_BACKLOG,
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PROP_PORT,
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PROP_SESSION_POOL,
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PROP_MEDIA_MAPPING,
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PROP_LAST
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};
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G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
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static void gst_rtsp_server_get_property (GObject *object, guint propid,
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GValue *value, GParamSpec *pspec);
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static void gst_rtsp_server_set_property (GObject *object, guint propid,
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const GValue *value, GParamSpec *pspec);
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static GstRTSPClient * default_accept_client (GstRTSPServer *server,
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GIOChannel *channel);
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static void
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gst_rtsp_server_class_init (GstRTSPServerClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->get_property = gst_rtsp_server_get_property;
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gobject_class->set_property = gst_rtsp_server_set_property;
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/**
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* GstRTSPServer::backlog
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*
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* The backlog argument defines the maximum length to which the queue of
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* pending connections for the server may grow. If a connection request arrives
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* when the queue is full, the client may receive an error with an indication of
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* ECONNREFUSED or, if the underlying protocol supports retransmission, the
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* request may be ignored so that a later reattempt at connection succeeds.
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*/
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g_object_class_install_property (gobject_class, PROP_BACKLOG,
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g_param_spec_int ("backlog", "Backlog", "The maximum length to which the queue "
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"of pending connections may grow",
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0, G_MAXINT, DEFAULT_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::port
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*
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* The session port of the server. This is the port where the server will
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* listen on.
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*/
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g_object_class_install_property (gobject_class, PROP_PORT,
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g_param_spec_int ("port", "Port", "The port the server uses to listen on",
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1, 65535, DEFAULT_PORT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::session-pool
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*
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* The session pool of the server. By default each server has a separate
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* session pool but sessions can be shared between servers by setting the same
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* session pool on multiple servers.
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*/
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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, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRTSPServer::media-mapping
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*
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* The media mapping to use for this server. By default the server has no
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* media mapping and thus cannot map urls to media streams.
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*/
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g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
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g_param_spec_object ("media-mapping", "Media Mapping",
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"The media mapping to use for client session",
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GST_TYPE_RTSP_MEDIA_MAPPING, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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klass->accept_client = default_accept_client;
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}
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static void
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gst_rtsp_server_init (GstRTSPServer * server)
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{
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server->port = DEFAULT_PORT;
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server->backlog = DEFAULT_BACKLOG;
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server->session_pool = gst_rtsp_session_pool_new ();
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server->media_mapping = gst_rtsp_media_mapping_new ();
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}
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/**
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* gst_rtsp_server_new:
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*
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* Create a new #GstRTSPServer instance.
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*/
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GstRTSPServer *
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gst_rtsp_server_new (void)
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{
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GstRTSPServer *result;
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result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
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return result;
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}
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/**
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* gst_rtsp_server_set_port:
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* @server: a #GstRTSPServer
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* @port: the port
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*
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* Configure @server to accept connections on the given port.
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* @port should be a port number between 1 and 65535.
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*
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* This function must be called before the server is bound.
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*/
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void
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gst_rtsp_server_set_port (GstRTSPServer *server, gint port)
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{
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g_return_if_fail (GST_IS_RTSP_SERVER (server));
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g_return_if_fail (port >= 1 && port <= 65535);
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server->port = port;
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}
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/**
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* gst_rtsp_server_get_port:
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* @server: a #GstRTSPServer
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*
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* Get the port number on which the server will accept connections.
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*
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* Returns: the server port.
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*/
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gint
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gst_rtsp_server_get_port (GstRTSPServer *server)
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{
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g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
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return server->port;
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}
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/**
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* gst_rtsp_server_set_backlog:
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* @server: a #GstRTSPServer
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* @backlog: the backlog
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*
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* configure the maximum amount of requests that may be queued for the
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* server.
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*
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* This function must be called before the server is bound.
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*/
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void
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gst_rtsp_server_set_backlog (GstRTSPServer *server, gint backlog)
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{
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g_return_if_fail (GST_IS_RTSP_SERVER (server));
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server->backlog = backlog;
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}
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/**
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* gst_rtsp_server_get_backlog:
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* @server: a #GstRTSPServer
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*
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* The maximum amount of queued requests for the server.
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*
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* Returns: the server backlog.
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*/
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gint
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gst_rtsp_server_get_backlog (GstRTSPServer *server)
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{
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g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
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return server->backlog;
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}
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/**
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* gst_rtsp_server_set_session_pool:
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* @server: a #GstRTSPServer
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* @pool: a #GstRTSPSessionPool
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*
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* configure @pool to be used as the session pool of @server.
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*/
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void
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gst_rtsp_server_set_session_pool (GstRTSPServer *server, GstRTSPSessionPool *pool)
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{
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GstRTSPSessionPool *old;
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g_return_if_fail (GST_IS_RTSP_SERVER (server));
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old = server->session_pool;
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if (old != pool) {
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if (pool)
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g_object_ref (pool);
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server->session_pool = pool;
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if (old)
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g_object_unref (old);
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}
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}
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/**
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* gst_rtsp_server_get_session_pool:
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* @server: a #GstRTSPServer
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*
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* Get the #GstRTSPSessionPool used as the session pool of @server.
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*
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* Returns: the #GstRTSPSessionPool used for sessions. g_object_unref() after
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* usage.
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*/
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GstRTSPSessionPool *
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gst_rtsp_server_get_session_pool (GstRTSPServer *server)
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{
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GstRTSPSessionPool *result;
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g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
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if ((result = server->session_pool))
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g_object_ref (result);
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return result;
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}
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/**
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* gst_rtsp_server_set_media_mapping:
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* @server: a #GstRTSPServer
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* @mapping: a #GstRTSPMediaMapping
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*
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* configure @mapping to be used as the media mapping of @server.
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*/
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void
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gst_rtsp_server_set_media_mapping (GstRTSPServer *server, GstRTSPMediaMapping *mapping)
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{
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GstRTSPMediaMapping *old;
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g_return_if_fail (GST_IS_RTSP_SERVER (server));
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old = server->media_mapping;
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if (old != mapping) {
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if (mapping)
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g_object_ref (mapping);
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server->media_mapping = mapping;
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if (old)
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g_object_unref (old);
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}
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}
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/**
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* gst_rtsp_server_get_media_mapping:
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* @server: a #GstRTSPServer
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*
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* Get the #GstRTSPMediaMapping used as the media mapping of @server.
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*
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* Returns: the #GstRTSPMediaMapping of @server. g_object_unref() after
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* usage.
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*/
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GstRTSPMediaMapping *
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gst_rtsp_server_get_media_mapping (GstRTSPServer *server)
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{
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GstRTSPMediaMapping *result;
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g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
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if ((result = server->media_mapping))
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g_object_ref (result);
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return result;
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}
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static void
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gst_rtsp_server_get_property (GObject *object, guint propid,
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GValue *value, GParamSpec *pspec)
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{
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GstRTSPServer *server = GST_RTSP_SERVER (object);
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switch (propid) {
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case PROP_PORT:
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g_value_set_int (value, gst_rtsp_server_get_port (server));
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break;
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case PROP_BACKLOG:
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g_value_set_int (value, gst_rtsp_server_get_backlog (server));
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break;
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case PROP_SESSION_POOL:
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g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
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break;
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case PROP_MEDIA_MAPPING:
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g_value_take_object (value, gst_rtsp_server_get_media_mapping (server));
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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_server_set_property (GObject *object, guint propid,
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const GValue *value, GParamSpec *pspec)
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{
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GstRTSPServer *server = GST_RTSP_SERVER (object);
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switch (propid) {
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case PROP_PORT:
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gst_rtsp_server_set_port (server, g_value_get_int (value));
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break;
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case PROP_BACKLOG:
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gst_rtsp_server_set_backlog (server, g_value_get_int (value));
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break;
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case PROP_SESSION_POOL:
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gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
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break;
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case PROP_MEDIA_MAPPING:
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gst_rtsp_server_set_media_mapping (server, g_value_get_object (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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/* Prepare a server socket for @server and make it listen on the configured port */
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static gboolean
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gst_rtsp_server_sink_init_send (GstRTSPServer * server)
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{
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int ret;
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/* create server socket */
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if ((server->server_sock.fd = socket (AF_INET, SOCK_STREAM, 0)) == -1)
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goto no_socket;
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GST_DEBUG_OBJECT (server, "opened sending server socket with fd %d",
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server->server_sock.fd);
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/* make address reusable */
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ret = 1;
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if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_REUSEADDR,
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(void *) &ret, sizeof (ret)) < 0)
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goto reuse_failed;
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/* keep connection alive; avoids SIGPIPE during write */
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ret = 1;
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if (setsockopt (server->server_sock.fd, SOL_SOCKET, SO_KEEPALIVE,
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(void *) &ret, sizeof (ret)) < 0)
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goto keepalive_failed;
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/* name the socket */
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memset (&server->server_sin, 0, sizeof (server->server_sin));
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server->server_sin.sin_family = AF_INET; /* network socket */
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server->server_sin.sin_port = htons (server->port); /* on port */
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server->server_sin.sin_addr.s_addr = htonl (INADDR_ANY); /* for hosts */
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/* bind it */
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GST_DEBUG_OBJECT (server, "binding server socket to address");
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ret = bind (server->server_sock.fd, (struct sockaddr *) &server->server_sin,
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sizeof (server->server_sin));
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if (ret)
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goto bind_failed;
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/* set the server socket to nonblocking */
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fcntl (server->server_sock.fd, F_SETFL, O_NONBLOCK);
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GST_DEBUG_OBJECT (server, "listening on server socket %d with queue of %d",
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server->server_sock.fd, server->backlog);
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if (listen (server->server_sock.fd, server->backlog) == -1)
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goto listen_failed;
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GST_DEBUG_OBJECT (server,
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"listened on server socket %d, returning from connection setup",
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server->server_sock.fd);
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g_message ("listening on port %d", server->port);
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return TRUE;
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/* ERRORS */
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no_socket:
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{
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GST_ERROR_OBJECT (server, "failed to create socket: %s", g_strerror (errno));
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return FALSE;
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}
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reuse_failed:
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{
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if (server->server_sock.fd >= 0) {
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close (server->server_sock.fd);
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server->server_sock.fd = -1;
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}
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GST_ERROR_OBJECT (server, "failed to reuse socket: %s", g_strerror (errno));
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return FALSE;
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}
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keepalive_failed:
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{
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if (server->server_sock.fd >= 0) {
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close (server->server_sock.fd);
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server->server_sock.fd = -1;
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}
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GST_ERROR_OBJECT (server, "failed to configure keepalive socket: %s", g_strerror (errno));
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return FALSE;
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}
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listen_failed:
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{
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if (server->server_sock.fd >= 0) {
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close (server->server_sock.fd);
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server->server_sock.fd = -1;
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}
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GST_ERROR_OBJECT (server, "failed to listen on socket: %s", g_strerror (errno));
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return FALSE;
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}
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bind_failed:
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{
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if (server->server_sock.fd >= 0) {
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close (server->server_sock.fd);
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server->server_sock.fd = -1;
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}
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GST_ERROR_OBJECT (server, "failed to bind on socket: %s", g_strerror (errno));
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return FALSE;
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}
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}
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/* default method for creating a new client object in the server to accept and
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* handle a client connection on this server */
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static GstRTSPClient *
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default_accept_client (GstRTSPServer *server, GIOChannel *channel)
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{
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GstRTSPClient *client;
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/* a new client connected, create a session to handle the client. */
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client = gst_rtsp_client_new ();
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/* set the session pool that this client should use */
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gst_rtsp_client_set_session_pool (client, server->session_pool);
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/* set the session pool that this client should use */
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gst_rtsp_client_set_media_mapping (client, server->media_mapping);
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/* accept connections for that client, this function returns after accepting
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* the connection and will run the remainder of the communication with the
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* client asyncronously. */
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if (!gst_rtsp_client_accept (client, channel))
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goto accept_failed;
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return client;
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/* ERRORS */
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accept_failed:
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{
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g_error ("Could not accept client on server socket %d: %s (%d)",
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server->server_sock.fd, g_strerror (errno), errno);
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gst_object_unref (client);
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return NULL;
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}
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}
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/**
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* gst_rtsp_server_io_func:
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* @channel: a #GIOChannel
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* @condition: the condition on @source
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*
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* A default #GIOFunc that creates a new #GstRTSPClient to accept and handle a
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* new connection on @channel or @server.
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*
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* Returns: TRUE if the source could be connected, FALSE if an error occured.
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*/
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gboolean
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gst_rtsp_server_io_func (GIOChannel *channel, GIOCondition condition, GstRTSPServer *server)
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{
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GstRTSPClient *client = NULL;
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GstRTSPServerClass *klass;
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if (condition & G_IO_IN) {
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klass = GST_RTSP_SERVER_GET_CLASS (server);
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/* a new client connected, create a client object to handle the client. */
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if (klass->accept_client)
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client = klass->accept_client (server, channel);
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if (client == NULL)
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goto client_failed;
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/* can unref the client now, when the request is finished, it will be
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* unreffed async. */
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gst_object_unref (client);
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}
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else {
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g_print ("received unknown event %08x", condition);
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}
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return TRUE;
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/* ERRORS */
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client_failed:
|
|
{
|
|
GST_ERROR_OBJECT (server, "failed to create a client");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_get_io_channel:
|
|
* @server: a #GstRTSPServer
|
|
*
|
|
* Create a #GIOChannel for @server.
|
|
*
|
|
* Returns: the GIOChannel for @server or NULL when an error occured.
|
|
*/
|
|
GIOChannel *
|
|
gst_rtsp_server_get_io_channel (GstRTSPServer *server)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
|
|
|
|
if (server->io_channel == NULL) {
|
|
if (!gst_rtsp_server_sink_init_send (server))
|
|
goto init_failed;
|
|
|
|
/* create IO channel for the socket */
|
|
server->io_channel = g_io_channel_unix_new (server->server_sock.fd);
|
|
}
|
|
return server->io_channel;
|
|
|
|
init_failed:
|
|
{
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_create_watch:
|
|
* @server: a #GstRTSPServer
|
|
*
|
|
* Create a #GSource for @server. The new source will have a default
|
|
* #GIOFunc of gst_rtsp_server_io_func().
|
|
*
|
|
* Returns: the #GSource for @server or NULL when an error occured.
|
|
*/
|
|
GSource *
|
|
gst_rtsp_server_create_watch (GstRTSPServer *server)
|
|
{
|
|
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
|
|
|
|
if (server->io_watch == NULL) {
|
|
GIOChannel *channel;
|
|
|
|
channel = gst_rtsp_server_get_io_channel (server);
|
|
if (channel == NULL)
|
|
goto no_channel;
|
|
|
|
/* create a watch for reads (new connections) and possible errors */
|
|
server->io_watch = g_io_create_watch (channel, G_IO_IN |
|
|
G_IO_ERR | G_IO_HUP | G_IO_NVAL);
|
|
|
|
/* configure the callback */
|
|
g_source_set_callback (server->io_watch, (GSourceFunc) gst_rtsp_server_io_func, server, NULL);
|
|
}
|
|
return server->io_watch;
|
|
|
|
no_channel:
|
|
{
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_server_attach:
|
|
* @server: a #GstRTSPServer
|
|
* @context: a #GMainContext
|
|
*
|
|
* Attaches @server to @context. When the mainloop for @context is run, the
|
|
* server will be dispatched.
|
|
*
|
|
* This function should be called when the server properties and urls are fully
|
|
* configured and the server is ready to start.
|
|
*
|
|
* Returns: the ID (greater than 0) for the source within the GMainContext.
|
|
*/
|
|
guint
|
|
gst_rtsp_server_attach (GstRTSPServer *server, GMainContext *context)
|
|
{
|
|
guint res;
|
|
GSource *source;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
|
|
|
|
source = gst_rtsp_server_create_watch (server);
|
|
if (source == NULL)
|
|
goto no_source;
|
|
|
|
res = g_source_attach (source, context);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
no_source:
|
|
{
|
|
return 0;
|
|
}
|
|
}
|