gstreamer/gst/rtsp-server/rtsp-client.c

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2008-10-09 12:29:12 +00:00
/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
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*/
#include <stdio.h>
#include <string.h>
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#include "rtsp-client.h"
#include "rtsp-sdp.h"
#include "rtsp-params.h"
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static GMutex tunnels_lock;
static GHashTable *tunnels;
#define DEFAULT_SESSION_POOL NULL
#define DEFAULT_MOUNT_POINTS NULL
#define DEFAULT_USE_CLIENT_SETTINGS FALSE
enum
{
PROP_0,
PROP_SESSION_POOL,
PROP_MOUNT_POINTS,
PROP_USE_CLIENT_SETTINGS,
PROP_LAST
};
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enum
{
SIGNAL_CLOSED,
SIGNAL_NEW_SESSION,
SIGNAL_OPTIONS_REQUEST,
SIGNAL_DESCRIBE_REQUEST,
SIGNAL_SETUP_REQUEST,
SIGNAL_PLAY_REQUEST,
SIGNAL_PAUSE_REQUEST,
SIGNAL_TEARDOWN_REQUEST,
SIGNAL_SET_PARAMETER_REQUEST,
SIGNAL_GET_PARAMETER_REQUEST,
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SIGNAL_LAST
};
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GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
#define GST_CAT_DEFAULT rtsp_client_debug
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static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
static void gst_rtsp_client_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_finalize (GObject * obj);
static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
static void client_session_finalized (GstRTSPClient * client,
GstRTSPSession * session);
static void unlink_session_transports (GstRTSPClient * client,
GstRTSPSession * session, GstRTSPSessionMedia * media);
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G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
static void
gst_rtsp_client_class_init (GstRTSPClientClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_client_get_property;
gobject_class->set_property = gst_rtsp_client_set_property;
gobject_class->finalize = gst_rtsp_client_finalize;
klass->create_sdp = create_sdp;
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
"The session pool to use for client session",
GST_TYPE_RTSP_SESSION_POOL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
g_param_spec_object ("mount-points", "Mount Points",
"The mount points to use for client session",
GST_TYPE_RTSP_MOUNT_POINTS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_USE_CLIENT_SETTINGS,
g_param_spec_boolean ("use-client-settings", "Use Client Settings",
"Use client settings for ttl and destination in multicast",
DEFAULT_USE_CLIENT_SETTINGS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_rtsp_client_signals[SIGNAL_CLOSED] =
g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
NULL, NULL, g_cclosure_marshal_VOID__POINTER, G_TYPE_NONE, 1,
G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
set_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
G_TYPE_NONE, 1, G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
get_parameter_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
G_TYPE_NONE, 1, G_TYPE_POINTER);
tunnels =
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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}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
g_mutex_init (&client->lock);
client->use_client_settings = DEFAULT_USE_CLIENT_SETTINGS;
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client->close_seq = 0;
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}
static void
client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
{
/* unlink all media managed in this session */
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while (session->medias) {
GstRTSPSessionMedia *media = session->medias->data;
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
unlink_session_transports (client, session, media);
/* unmanage the media in the session. this will modify session->medias */
gst_rtsp_session_release_media (session, media);
}
}
static void
client_cleanup_sessions (GstRTSPClient * client)
{
GList *sessions;
/* remove weak-ref from sessions */
for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
GstRTSPSession *session = (GstRTSPSession *) sessions->data;
g_object_weak_unref (G_OBJECT (session),
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(GWeakNotify) client_session_finalized, client);
client_unlink_session (client, session);
}
g_list_free (client->sessions);
client->sessions = NULL;
}
/* A client is finalized when the connection is broken */
static void
gst_rtsp_client_finalize (GObject * obj)
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
GST_INFO ("finalize client %p", client);
if (client->watch)
g_source_destroy ((GSource *) client->watch);
if (client->send_notify)
client->send_notify (client->send_data);
client_cleanup_sessions (client);
gst_rtsp_connection_free (client->connection);
if (client->session_pool)
g_object_unref (client->session_pool);
if (client->mount_points)
g_object_unref (client->mount_points);
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if (client->auth)
g_object_unref (client->auth);
if (client->uri)
gst_rtsp_url_free (client->uri);
if (client->media) {
gst_rtsp_media_unprepare (client->media);
g_object_unref (client->media);
}
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g_free (client->server_ip);
g_mutex_clear (&client->lock);
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G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
static void
gst_rtsp_client_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
switch (propid) {
case PROP_SESSION_POOL:
g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
break;
case PROP_MOUNT_POINTS:
g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
break;
case PROP_USE_CLIENT_SETTINGS:
g_value_set_boolean (value,
gst_rtsp_client_get_use_client_settings (client));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_client_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
switch (propid) {
case PROP_SESSION_POOL:
gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
break;
case PROP_MOUNT_POINTS:
gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
break;
case PROP_USE_CLIENT_SETTINGS:
gst_rtsp_client_set_use_client_settings (client,
g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
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/**
* gst_rtsp_client_new:
*
* Create a new #GstRTSPClient instance.
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*
* Returns: a new #GstRTSPClient
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*/
GstRTSPClient *
gst_rtsp_client_new (void)
{
GstRTSPClient *result;
result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
return result;
}
static void
send_response (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPMessage * response, gboolean close)
{
gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
"GStreamer RTSP server");
/* remove any previous header */
gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
/* add the new session header for new session ids */
if (session) {
gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION,
gst_rtsp_session_get_header (session));
}
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (response);
}
if (close)
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gst_rtsp_message_add_header (response, GST_RTSP_HDR_CONNECTION, "close");
if (client->send_func)
client->send_func (client, response, close, client->send_data);
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gst_rtsp_message_unset (response);
}
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static void
send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
GstRTSPClientState * state)
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{
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
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send_response (client, NULL, state->response, FALSE);
}
static void
handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
GstRTSPClientState * state)
{
gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
if (auth) {
/* and let the authentication manager setup the auth tokens */
gst_rtsp_auth_setup_auth (auth, client, 0, state);
}
send_response (client, state->session, state->response, FALSE);
}
static gboolean
compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
{
if (uri1 == NULL || uri2 == NULL)
return FALSE;
if (strcmp (uri1->abspath, uri2->abspath))
return FALSE;
return TRUE;
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}
/* 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)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
GstRTSPAuth *auth;
if (!compare_uri (client->uri, state->uri)) {
/* remove any previously cached values before we try to construct a new
* media for uri */
if (client->uri)
gst_rtsp_url_free (client->uri);
client->uri = NULL;
if (client->media) {
gst_rtsp_media_unprepare (client->media);
g_object_unref (client->media);
}
client->media = NULL;
if (!client->mount_points)
goto no_mount_points;
/* find the factory for the uri first */
if (!(factory =
gst_rtsp_mount_points_find_factory (client->mount_points,
state->uri)))
goto no_factory;
state->factory = factory;
/* check if we have access to the factory */
if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
if (!gst_rtsp_auth_check (auth, client, 0, state))
goto not_allowed;
g_object_unref (auth);
}
/* prepare the media and add it to the pipeline */
if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
goto no_media;
g_object_unref (factory);
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factory = NULL;
state->factory = NULL;
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/* set ipv6 on the media before preparing */
media->is_ipv6 = client->is_ipv6;
state->media = media;
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/* prepare the media */
if (!(gst_rtsp_media_prepare (media)))
goto no_prepare;
/* now keep track of the uri and the media */
client->uri = gst_rtsp_url_copy (state->uri);
client->media = media;
} else {
/* we have seen this uri before, used cached media */
media = client->media;
state->media = media;
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GST_INFO ("reusing cached media %p", media);
}
if (media)
g_object_ref (media);
return media;
/* ERRORS */
no_mount_points:
{
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GST_ERROR ("client %p: no mount points configured", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
no_factory:
{
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GST_ERROR ("client %p: no factory for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
not_allowed:
{
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GST_ERROR ("client %p: unauthorized request", client);
handle_unauthorized_request (client, auth, state);
g_object_unref (factory);
g_object_unref (auth);
return NULL;
}
no_media:
{
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GST_ERROR ("client %p: can't create media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (factory);
return NULL;
}
no_prepare:
{
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GST_ERROR ("client %p: can't prepare media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
return NULL;
}
}
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
GstRTSPMessage message = { 0 };
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GstMapInfo map_info;
guint8 *data;
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guint usize;
gst_rtsp_message_init_data (&message, channel);
/* FIXME, need some sort of iovec RTSPMessage here */
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if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
return FALSE;
gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
if (client->send_func)
client->send_func (client, &message, FALSE, client->send_data);
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gst_rtsp_message_steal_body (&message, &data, &usize);
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gst_buffer_unmap (buffer, &map_info);
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gst_rtsp_message_unset (&message);
return TRUE;
}
static void
link_transport (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPStreamTransport * trans)
{
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);
client->transports = g_list_prepend (client->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)
{
GST_DEBUG ("client %p: unlinking transport %p", client, trans);
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
client->transports = g_list_remove (client->transports, trans);
/* our session can now expire */
gst_rtsp_session_allow_expire (session);
}
static void
unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPSessionMedia * media)
{
guint n_streams, i;
n_streams = gst_rtsp_media_n_streams (media->media);
for (i = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans;
GstRTSPTransport *tr;
/* get the transport, if there is no transport configured, skip this stream */
trans = gst_rtsp_session_media_get_transport (media, i);
if (trans == NULL)
continue;
tr = trans->transport;
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)
{
const gchar *tunnelid;
GST_DEBUG ("client %p: closing connection", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
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g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
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g_mutex_unlock (&tunnels_lock);
}
gst_rtsp_connection_close (client->connection);
}
static gboolean
handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPSession *session;
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
if (!state->session)
goto no_session;
session = state->session;
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/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
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state->sessmedia = media;
/* unlink the all TCP callbacks */
unlink_session_transports (client, session, media);
/* remove the session from the watched sessions */
g_object_weak_unref (G_OBJECT (session),
(GWeakNotify) client_session_finalized, client);
client->sessions = g_list_remove (client->sessions, session);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
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/* unmanage the media in the session, returns false if all media session
* are torn down. */
if (!gst_rtsp_session_release_media (session, media)) {
/* remove the session */
gst_rtsp_session_pool_remove (client->session_pool, session);
}
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/* 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);
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send_response (client, session, state->response, TRUE);
/* we emit the signal before closing the connection */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
0, state);
return TRUE;
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/* ERRORS */
no_session:
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{
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GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
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return FALSE;
}
not_found:
{
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GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
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}
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_params_get (client, state);
if (res != GST_RTSP_OK)
goto bad_request;
send_response (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:
{
2012-11-26 12:37:20 +00:00
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_params_set (client, state);
if (res != GST_RTSP_OK)
goto bad_request;
send_response (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:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
}
2008-10-09 12:29:12 +00:00
static gboolean
handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
2008-10-09 12:29:12 +00:00
{
GstRTSPSession *session;
2008-10-09 12:29:12 +00:00
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
if (!(session = state->session))
goto no_session;
2008-10-09 12:29:12 +00:00
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
2008-10-09 12:29:12 +00:00
state->sessmedia = media;
/* the session state must be playing or recording */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != GST_RTSP_STATE_RECORDING)
goto invalid_state;
/* unlink the all TCP callbacks */
unlink_session_transports (client, session, media);
/* then pause sending */
gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
2008-10-09 12:29:12 +00:00
/* 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);
2008-10-09 12:29:12 +00:00
send_response (client, session, state->response, FALSE);
2008-10-09 12:29:12 +00:00
/* the state is now READY */
media->state = GST_RTSP_STATE_READY;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST],
0, state);
return TRUE;
2008-10-09 12:29:12 +00:00
/* ERRORS */
no_session:
2008-10-09 12:29:12 +00:00
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: no seesion", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
2008-10-09 12:29:12 +00:00
return FALSE;
}
not_found:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
2012-11-26 12:37:20 +00:00
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;
}
2008-10-09 12:29:12 +00:00
}
static gboolean
handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
2008-10-09 12:29:12 +00:00
{
GstRTSPSession *session;
2008-10-09 12:29:12 +00:00
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
GString *rtpinfo;
guint n_streams, i, infocount;
gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
2008-10-09 12:29:12 +00:00
if (!(session = state->session))
goto no_session;
2008-10-09 12:29:12 +00:00
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
2008-10-09 12:29:12 +00:00
state->sessmedia = media;
/* the session state must be playing or ready */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != 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 */
gst_rtsp_media_seek (media->media, range);
gst_rtsp_range_free (range);
}
}
2008-10-09 12:29:12 +00:00
/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
n_streams = gst_rtsp_media_n_streams (media->media);
for (i = 0, infocount = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans;
GstRTSPTransport *tr;
gchar *uristr;
guint rtptime, seq;
2008-10-09 12:29:12 +00:00
/* get the transport, if there is no transport configured, skip this stream */
trans = gst_rtsp_session_media_get_transport (media, i);
if (trans == NULL) {
2009-11-21 18:20:23 +00:00
GST_INFO ("stream %d is not configured", i);
continue;
}
tr = trans->transport;
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);
}
if (gst_rtsp_stream_get_rtpinfo (trans->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 {
2009-11-21 18:20:23 +00:00
GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
2009-05-23 14:30:55 +00:00
}
2008-10-09 12:29:12 +00:00
}
/* 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);
2008-10-09 12:29:12 +00:00
/* add the RTP-Info header */
2009-05-23 14:30:55 +00:00
if (infocount > 0) {
str = g_string_free (rtpinfo, FALSE);
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
} else {
2009-05-23 14:30:55 +00:00
g_string_free (rtpinfo, TRUE);
}
/* add the range */
str = gst_rtsp_media_get_range_string (media->media, TRUE);
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
2008-10-09 12:29:12 +00:00
send_response (client, session, state->response, FALSE);
2008-10-09 12:29:12 +00:00
/* start playing after sending the request */
gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
media->state = GST_RTSP_STATE_PLAYING;
2008-10-09 12:29:12 +00:00
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST],
0, state);
return TRUE;
2008-10-09 12:29:12 +00:00
/* ERRORS */
no_session:
2008-10-09 12:29:12 +00:00
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
2008-10-09 12:29:12 +00:00
return FALSE;
}
not_found:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
2012-11-26 12:37:20 +00:00
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;
}
2008-10-09 12:29:12 +00:00
}
static void
do_keepalive (GstRTSPSession * session)
{
2009-11-21 18:20:23 +00:00
GST_INFO ("keep session %p alive", session);
gst_rtsp_session_touch (session);
}
2012-10-27 19:26:55 +00:00
/* 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);
2012-11-15 15:15:20 +00:00
GST_DEBUG ("parsing transports %s", transport);
2012-10-27 19:26:55 +00:00
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,
2012-07-10 09:39:58 +00:00
&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
configure_client_transport (GstRTSPClient * client, GstRTSPClientState * state,
GstRTSPTransport * ct)
{
/* 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 == NULL || !client->use_client_settings) {
GstRTSPAddress *addr;
addr = gst_rtsp_stream_get_address (state->stream);
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;
}
} else {
GstRTSPUrl *url;
url = gst_rtsp_connection_get_url (client->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;
/* prepare the server transport */
gst_rtsp_transport_new (&st);
st->trans = ct->trans;
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
st->server_port = state->stream->server_port;
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;
}
if (state->stream->session)
g_object_get (state->stream->session, "internal-ssrc", &st->ssrc, NULL);
return st;
}
2008-10-09 12:29:12 +00:00
static gboolean
handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
2008-10-09 12:29:12 +00:00
{
GstRTSPResult res;
GstRTSPUrl *uri;
2008-10-09 12:29:12 +00:00
gchar *transport;
GstRTSPTransport *ct, *st;
GstRTSPLowerTrans supported;
GstRTSPStatusCode code;
GstRTSPSession *session;
GstRTSPStreamTransport *trans;
2008-10-09 12:29:12 +00:00
gchar *trans_str, *pos;
guint streamid;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStream *stream;
2008-10-09 12:29:12 +00:00
uri = state->uri;
/* the uri contains the stream number we added in the SDP config, which is
2012-10-27 19:26:55 +00:00
* always /stream=%d so we need to strip that off
* parse the stream we need to configure, look for the stream in the abspath
* first and then in the query. */
2010-03-09 12:42:50 +00:00
if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
goto bad_request;
}
2012-10-27 19:26:55 +00:00
/* we can mofify the parsed uri in place */
*pos = '\0';
pos += strlen ("/stream=");
if (sscanf (pos, "%u", &streamid) != 1)
goto bad_request;
2008-10-09 12:29:12 +00:00
/* parse the transport */
res =
gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
&transport, 0);
2008-10-09 12:29:12 +00:00
if (res != GST_RTSP_OK)
goto no_transport;
2008-10-09 12:29:12 +00:00
gst_rtsp_transport_new (&ct);
2008-10-09 12:29:12 +00:00
/* our supported transports */
supported = GST_RTSP_LOWER_TRANS_UDP |
2010-03-16 17:33:23 +00:00
GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
2012-10-27 19:26:55 +00:00
/* parse and find a usable supported transport */
if (!parse_transport (transport, supported, ct))
2008-10-09 12:29:12 +00:00
goto unsupported_transports;
/* we create the session after parsing stuff so that we don't make
* a session for malformed requests */
if (client->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, uri);
} else {
2008-10-09 12:29:12 +00:00
/* create a session if this fails we probably reached our session limit or
* something. */
if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
2008-10-09 12:29:12 +00:00
goto service_unavailable;
state->session = session;
/* we need a new media configuration in this session */
sessmedia = NULL;
2008-10-09 12:29:12 +00:00
}
/* we have no media, find one and manage it */
if (sessmedia == NULL) {
/* get a handle to the configuration of the media in the session */
if ((media = find_media (client, state))) {
/* manage the media in our session now */
sessmedia = gst_rtsp_session_manage_media (session, uri, media);
}
}
/* if we stil have no media, error */
if (sessmedia == NULL)
goto not_found;
2008-10-09 12:29:12 +00:00
state->sessmedia = sessmedia;
state->media = media = sessmedia->media;
/* now get the stream */
stream = gst_rtsp_media_get_stream (media, streamid);
if (stream == NULL)
goto not_found;
state->stream = stream;
/* set blocksize on this stream */
if (!handle_blocksize (media, stream, state->request))
goto invalid_blocksize;
/* update the client transport */
if (!configure_client_transport (client, state, ct))
goto unsupported_client_transport;
2008-10-09 12:29:12 +00:00
/* set in the session media transport */
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
2008-10-09 12:29:12 +00:00
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
2010-03-16 17:33:23 +00:00
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
/* create and serialize the server transport */
st = make_server_transport (client, state, ct);
2008-10-09 12:29:12 +00:00
trans_str = gst_rtsp_transport_as_text (st);
2009-01-30 16:06:26 +00:00
gst_rtsp_transport_free (st);
2008-10-09 12:29:12 +00:00
/* 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);
2008-10-09 12:29:12 +00:00
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
trans_str);
2008-10-09 12:29:12 +00:00
g_free (trans_str);
send_response (client, session, state->response, FALSE);
2008-10-09 12:29:12 +00:00
/* update the state */
switch (sessmedia->state) {
case GST_RTSP_STATE_PLAYING:
case GST_RTSP_STATE_RECORDING:
case GST_RTSP_STATE_READY:
/* no state change */
break;
default:
sessmedia->state = GST_RTSP_STATE_READY;
break;
}
g_object_unref (session);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST],
0, state);
2008-10-09 12:29:12 +00:00
return TRUE;
/* ERRORS */
bad_request:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
2008-10-09 12:29:12 +00:00
return FALSE;
}
not_found:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
2012-06-14 07:59:06 +00:00
gst_rtsp_transport_free (ct);
return FALSE;
}
invalid_blocksize:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: invalid blocksize", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
g_object_unref (session);
gst_rtsp_transport_free (ct);
return FALSE;
}
unsupported_client_transport:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: unsupported client transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
g_object_unref (session);
gst_rtsp_transport_free (ct);
return FALSE;
}
no_transport:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: no transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
return FALSE;
}
2008-10-09 12:29:12 +00:00
unsupported_transports:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: unsupported transports", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
gst_rtsp_transport_free (ct);
2008-10-09 12:29:12 +00:00
return FALSE;
}
no_pool:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: no session pool configured", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
2012-06-14 07:59:06 +00:00
gst_rtsp_transport_free (ct);
return FALSE;
}
2008-10-09 12:29:12 +00:00
service_unavailable:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: can't create session", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
2012-06-14 07:59:06 +00:00
gst_rtsp_transport_free (ct);
2008-10-09 12:29:12 +00:00
return FALSE;
}
}
2010-03-16 17:33:23 +00:00
static GstSDPMessage *
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
{
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
gst_sdp_message_new (&sdp);
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
if (client->is_ipv6)
proto = "IP6";
else
proto = "IP4";
gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
client->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.server_proto = proto;
2012-11-15 16:11:16 +00:00
info.server_ip = g_strdup (client->server_ip);
2010-03-16 17:33:23 +00:00
/* create an SDP for the media object */
if (!gst_rtsp_sdp_from_media (sdp, &info, media))
goto no_sdp;
g_free (info.server_ip);
2010-03-16 17:33:23 +00:00
return sdp;
/* ERRORS */
no_sdp:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: could not create SDP", client);
g_free (info.server_ip);
2010-03-16 17:33:23 +00:00
gst_sdp_message_free (sdp);
return NULL;
}
}
/* for the describe we must generate an SDP */
2008-10-09 12:29:12 +00:00
static gboolean
handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
2008-10-09 12:29:12 +00:00
{
GstRTSPResult res;
2008-10-09 12:29:12 +00:00
GstSDPMessage *sdp;
guint i, str_len;
gchar *str, *content_base;
GstRTSPMedia *media;
GstRTSPClientClass *klass;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
2008-10-09 12:29:12 +00:00
/* 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;
}
2008-10-09 12:29:12 +00:00
/* find the media object for the uri */
if (!(media = find_media (client, state)))
goto no_media;
2010-03-16 17:33:23 +00:00
/* create an SDP for the media object on this client */
if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
2008-10-09 12:29:12 +00:00
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);
2008-10-09 12:29:12 +00:00
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] != '/') {
2010-03-16 17:33:23 +00:00
content_base = g_malloc (str_len + 2);
memcpy (content_base, str, str_len);
content_base[str_len] = '/';
2010-03-16 17:33:23 +00:00
content_base[str_len + 1] = '\0';
g_free (str);
} else {
content_base = str;
}
2010-03-16 17:33:23 +00:00
GST_INFO ("adding content-base: %s", content_base);
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
2010-03-16 17:33:23 +00:00
content_base);
g_free (content_base);
2008-10-09 12:29:12 +00:00
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
2008-10-09 12:29:12 +00:00
gst_sdp_message_free (sdp);
send_response (client, state->session, state->response, FALSE);
2008-10-09 12:29:12 +00:00
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
0, state);
2008-10-09 12:29:12 +00:00
return TRUE;
/* ERRORS */
no_media:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: no media", client);
/* error reply is already sent */
return FALSE;
}
no_sdp:
{
2012-11-26 12:37:20 +00:00
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;
}
2008-10-09 12:29:12 +00:00
}
static gboolean
handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
2008-10-09 12:29:12 +00:00
{
GstRTSPMethod options;
gchar *str;
2008-10-09 12:29:12 +00:00
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;
2008-10-09 12:29:12 +00:00
str = gst_rtsp_options_as_text (options);
2008-10-09 12:29:12 +00:00
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);
2008-10-09 12:29:12 +00:00
send_response (client, state->session, state->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
0, state);
return TRUE;
}
2008-10-09 12:29:12 +00:00
/* remove duplicate and trailing '/' */
static void
2010-12-28 11:18:41 +00:00
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';
2008-10-09 12:29:12 +00:00
}
static void
client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
{
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 (!(client->sessions = g_list_remove (client->sessions, session))) {
GST_INFO ("client %p: all sessions finalized, close the connection",
client);
close_connection (client);
}
}
static void
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
{
GList *walk;
for (walk = client->sessions; walk; walk = g_list_next (walk)) {
GstRTSPSession *msession = (GstRTSPSession *) walk->data;
/* we already know about this session */
if (msession == session)
return;
}
2009-11-21 18:20:23 +00:00
GST_INFO ("watching session %p", session);
g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
client);
client->sessions = g_list_prepend (client->sessions, session);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
session);
}
static void
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
2008-10-09 12:29:12 +00:00
{
GstRTSPMethod method;
const gchar *uristr;
GstRTSPUrl *uri;
2008-10-09 12:29:12 +00:00
GstRTSPVersion version;
GstRTSPResult res;
GstRTSPSession *session;
GstRTSPClientState state = { NULL };
GstRTSPMessage response = { 0 };
gchar *sessid;
2008-10-09 12:29:12 +00:00
state.request = request;
state.response = &response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (request);
}
2008-10-09 12:29:12 +00:00
2009-11-21 18:20:23 +00:00
GST_INFO ("client %p: received a request", client);
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
if (version != GST_RTSP_VERSION_1_0) {
/* we can only handle 1.0 requests */
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
&state);
return;
2008-10-09 12:29:12 +00:00
}
state.method = method;
2008-10-09 12:29:12 +00:00
/* we always try to parse the url first */
if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
return;
2008-10-09 12:29:12 +00:00
}
/* sanitize the uri */
2010-12-28 11:18:41 +00:00
sanitize_uri (uri);
state.uri = uri;
2008-10-09 12:29:12 +00:00
/* 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 (client->session_pool == NULL)
goto no_pool;
2008-10-09 12:29:12 +00:00
/* we had a session in the request, find it again */
if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
goto session_not_found;
2008-10-09 12:29:12 +00:00
/* 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);
} else
session = NULL;
state.session = session;
if (client->auth) {
if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
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:
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
break;
case GST_RTSP_INVALID:
default:
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
break;
}
if (session)
g_object_unref (session);
gst_rtsp_url_free (uri);
return;
2008-10-09 12:29:12 +00:00
/* ERRORS */
no_pool:
2008-10-09 12:29:12 +00:00
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: no pool configured", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
return;
}
session_not_found:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: session not found", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
return;
2008-10-09 12:29:12 +00:00
}
not_authorized:
{
2012-11-26 12:37:20 +00:00
GST_ERROR ("client %p: not allowed", client);
handle_unauthorized_request (client, client->auth, &state);
return;
}
2008-10-09 12:29:12 +00:00
}
static void
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
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 = client->transports; walk; walk = g_list_next (walk)) {
GstRTSPStreamTransport *trans;
GstRTSPStream *stream;
GstRTSPTransport *tr;
trans = walk->data;
/* we only add clients with a transport to the list */
tr = trans->transport;
stream = trans->stream;
/* 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);
}
2008-10-09 12:29:12 +00:00
/**
* 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.
2008-10-09 12:29:12 +00:00
*/
void
gst_rtsp_client_set_session_pool (GstRTSPClient * client,
GstRTSPSessionPool * pool)
2008-10-09 12:29:12 +00:00
{
GstRTSPSessionPool *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
if (pool)
g_object_ref (pool);
g_mutex_lock (&client->lock);
old = client->session_pool;
client->session_pool = pool;
g_mutex_unlock (&client->lock);
if (old)
g_object_unref (old);
2008-10-09 12:29:12 +00:00
}
/**
* 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.
2008-10-09 12:29:12 +00:00
*/
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
2008-10-09 12:29:12 +00:00
{
GstRTSPSessionPool *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
g_mutex_lock (&client->lock);
if ((result = client->session_pool))
2008-10-09 12:29:12 +00:00
g_object_ref (result);
g_mutex_unlock (&client->lock);
2008-10-09 12:29:12 +00:00
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)
{
GstRTSPMountPoints *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
if (mounts)
g_object_ref (mounts);
g_mutex_lock (&client->lock);
old = client->mount_points;
client->mount_points = mounts;
g_mutex_unlock (&client->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)
{
GstRTSPMountPoints *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
g_mutex_lock (&client->lock);
if ((result = client->mount_points))
g_object_ref (result);
g_mutex_unlock (&client->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)
{
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
g_mutex_lock (&client->lock);
client->use_client_settings = use_client_settings;
g_mutex_unlock (&client->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)
{
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
g_mutex_lock (&client->lock);
res = client->use_client_settings;
g_mutex_unlock (&client->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)
{
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
if (auth)
g_object_ref (auth);
g_mutex_lock (&client->lock);
old = client->auth;
client->auth = auth;
g_mutex_unlock (&client->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)
{
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
g_mutex_lock (&client->lock);
if ((result = client->auth))
g_object_ref (result);
g_mutex_unlock (&client->lock);
return result;
}
/**
* 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)
{
2012-11-26 16:34:24 +00:00
GDestroyNotify old_func;
2012-11-26 16:28:29 +00:00
gpointer old_data;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
g_mutex_lock (&client->lock);
client->send_func = func;
2012-11-26 16:28:29 +00:00
old_func = client->send_notify;
old_data = client->send_data;
client->send_notify = notify;
2012-11-26 16:28:29 +00:00
client->send_data = user_data;
g_mutex_unlock (&client->lock);
2012-11-26 16:28:29 +00:00
if (old_func)
old_func (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;
}
static GstRTSPResult
do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
gboolean close, gpointer user_data)
{
/* 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 (client->watch, message, close ?
&client->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)
{
2012-11-26 15:45:04 +00:00
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
2012-11-26 15:45:04 +00:00
if (client->close_seq && client->close_seq == cseq) {
client->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);
const gchar *tunnelid;
2009-11-21 18:20:23 +00:00
GST_INFO ("client %p: connection closed", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
2012-05-11 07:42:47 +00:00
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
2012-05-11 07:42:47 +00:00
g_mutex_unlock (&tunnels_lock);
}
return GST_RTSP_OK;
}
static GstRTSPResult
error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gchar *str;
str = gst_rtsp_strresult (result);
2009-11-21 18:20:23 +00:00
GST_INFO ("client %p: received an error %s", client, str);
g_free (str);
return GST_RTSP_OK;
}
static GstRTSPResult
2010-03-16 17:33:23 +00:00
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);
2010-03-16 17:33:23 +00:00
GST_INFO
("client %p: received an error %s when handling message %p with id %d",
client, str, message, id);
g_free (str);
return GST_RTSP_OK;
}
static gboolean
remember_tunnel (GstRTSPClient * client)
{
const gchar *tunnelid;
/* store client in the pending tunnels */
tunnelid = gst_rtsp_connection_get_tunnelid (client->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 */
2012-05-11 07:42:47 +00:00
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));
2012-05-11 07:42:47 +00:00
g_mutex_unlock (&tunnels_lock);
return TRUE;
/* ERRORS */
no_tunnelid:
{
GST_ERROR ("client %p: no tunnelid provided", client);
return FALSE;
}
tunnel_existed:
{
2012-05-11 07:42:47 +00:00
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;
client = GST_RTSP_CLIENT (user_data);
GST_INFO ("client %p: tunnel start (connection %p)", client,
client->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;
client = GST_RTSP_CLIENT (user_data);
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GST_WARNING ("client %p: tunnel lost (connection %p)", client,
client->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);
GstRTSPClient *oclient;
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GST_INFO ("client %p: tunnel complete", client);
/* find previous tunnel */
tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
if (tunnelid == NULL)
goto no_tunnelid;
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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);
if (oclient->watch == NULL)
goto tunnel_closed;
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g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
oclient->connection, client->connection);
/* merge the tunnels into the first client */
gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
gst_rtsp_watch_reset (oclient->watch);
g_object_unref (oclient);
return GST_RTSP_OK;
/* ERRORS */
no_tunnelid:
{
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GST_ERROR ("client %p: no tunnelid provided", client);
return GST_RTSP_ERROR;
}
no_tunnel:
{
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g_mutex_unlock (&tunnels_lock);
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GST_ERROR ("client %p: tunnel session %s not found", client, tunnelid);
return GST_RTSP_ERROR;
}
tunnel_closed:
{
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g_mutex_unlock (&tunnels_lock);
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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)
{
GST_INFO ("client %p: watch destroyed", client);
client->watch = NULL;
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g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
static gboolean
setup_client (GstRTSPClient * client, GSocket * socket,
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GstRTSPConnection * conn, GError ** error)
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{
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GSocket *read_socket;
GSocketAddress *address;
GstRTSPUrl *url;
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read_socket = gst_rtsp_connection_get_read_socket (conn);
client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
if (!(address = g_socket_get_remote_address (read_socket, error)))
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goto no_address;
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g_free (client->server_ip);
/* keep the original ip that the client connected to */
if (G_IS_INET_SOCKET_ADDRESS (address)) {
GInetAddress *iaddr;
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iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
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client->server_ip = g_inet_address_to_string (iaddr);
g_object_unref (address);
} else {
client->server_ip = g_strdup ("unknown");
}
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GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
client->server_ip, client->is_ipv6);
url = gst_rtsp_connection_get_url (conn);
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GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
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client->connection = conn;
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return TRUE;
/* ERRORS */
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no_address:
{
GST_ERROR ("could not get remote address %s", (*error)->message);
return FALSE;
}
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}
/**
* gst_rtsp_client_use_socket:
* @client: a #GstRTSPClient
* @socket: a #GSocket
* @ip: the IP address of the remote client
* @port: the port used by the other end
* @initial_buffer: any zero terminated initial data that was already read from
* the socket
* @error: a #GError
*
* Take an existing network socket and use it for an RTSP connection.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_client_use_socket (GstRTSPClient * client, GSocket * socket,
const gchar * ip, gint port, const gchar * initial_buffer, GError ** error)
{
GstRTSPConnection *conn;
GstRTSPResult res;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
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initial_buffer, &conn), no_connection);
return setup_client (client, socket, conn, error);
/* ERRORS */
no_connection:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR ("could not create connection from socket %p: %s", socket, str);
g_free (str);
return FALSE;
}
}
/**
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* gst_rtsp_client_accept:
* @client: a #GstRTSPClient
* @socket: a #GSocket
* @context: the context to run in
* @cancellable: a #GCancellable
* @error: a #GError
*
* Accept a new connection for @client on @socket.
*
* Returns: %TRUE if the client could be accepted.
*/
gboolean
gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
GCancellable * cancellable, GError ** error)
{
GstRTSPConnection *conn;
GstRTSPResult res;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
g_return_val_if_fail (G_IS_SOCKET (socket), FALSE);
/* a new client connected. */
GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
accept_failed);
return setup_client (client, socket, conn, error);
/* ERRORS */
accept_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
g_free (str);
return FALSE;
}
}
/**
* 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)
{
guint res;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
g_return_val_if_fail (client->watch == NULL, 0);
/* create watch for the connection and attach */
client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
g_object_ref (client), (GDestroyNotify) client_watch_notify);
gst_rtsp_client_set_send_func (client, do_send_message, NULL, NULL);
GST_INFO ("attaching to context %p", context);
res = gst_rtsp_watch_attach (client->watch, context);
gst_rtsp_watch_unref (client->watch);
return res;
}