mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-14 05:12:09 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtsp-server/rtsp-client.c
Sebastian Dröge 9fab555cc5 rtsp-server: Implement clock signalling according to RFC7273 
For NTP and PTP clocks we signal the actual clock that is used and signal
the direct media clock offset.

For all other clocks we at least signal that it's the local sender clock.

This allows receivers to know which clock was used to generate the media and
its RTP timestamps. Receivers can then implement network synchronization,
either absolute or at least relative by getting the sender clock rate directly
via NTP/PTP instead of estimating it from RTP timestamps and packet receive
times.

https://bugzilla.gnome.org/show_bug.cgi?id=760005
2016-04-03 11:22:31 +03:00

3996 lines
110 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
* Copyright (C) 2015 Centricular Ltd
* Author: Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* 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
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:rtsp-client
* @short_description: A client connection state
* @see_also: #GstRTSPServer, #GstRTSPThreadPool
*
* The client object handles the connection with a client for as long as a TCP
* connection is open.
*
* A #GstRTSPClient is created by #GstRTSPServer when a new connection is
* accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
* #GstRTSPAuth and #GstRTSPThreadPool from the server.
*
* The client connection should be configured with the #GstRTSPConnection using
* gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
* using gst_rtsp_client_attach(). From then on the client will handle requests
* on the connection.
*
* Use gst_rtsp_client_session_filter() to iterate or modify all the
* #GstRTSPSession objects managed by the client object.
*
* Last reviewed on 2013-07-11 (1.0.0)
*/
#include <stdio.h>
#include <string.h>
#include <gst/sdp/gstmikey.h>
#include "rtsp-client.h"
#include "rtsp-sdp.h"
#include "rtsp-params.h"
#define GST_RTSP_CLIENT_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
/* locking order:
* send_lock, lock, tunnels_lock
*/
struct _GstRTSPClientPrivate
{
GMutex lock; /* protects everything else */
GMutex send_lock;
GMutex watch_lock;
GstRTSPConnection *connection;
GstRTSPWatch *watch;
GMainContext *watch_context;
guint close_seq;
gchar *server_ip;
gboolean is_ipv6;
GstRTSPClientSendFunc send_func; /* protected by send_lock */
gpointer send_data; /* protected by send_lock */
GDestroyNotify send_notify; /* protected by send_lock */
GstRTSPSessionPool *session_pool;
gulong session_removed_id;
GstRTSPMountPoints *mount_points;
GstRTSPAuth *auth;
GstRTSPThreadPool *thread_pool;
/* used to cache the media in the last requested DESCRIBE so that
* we can pick it up in the next SETUP immediately */
gchar *path;
GstRTSPMedia *media;
GHashTable *transports;
GList *sessions;
guint sessions_cookie;
gboolean drop_backlog;
};
static GMutex tunnels_lock;
static GHashTable *tunnels; /* protected by tunnels_lock */
/* FIXME make this configurable. We don't want to do this yet because it will
* be superceeded by a cache object later */
#define WATCH_BACKLOG_SIZE 100
#define DEFAULT_SESSION_POOL NULL
#define DEFAULT_MOUNT_POINTS NULL
#define DEFAULT_DROP_BACKLOG TRUE
enum
{
PROP_0,
PROP_SESSION_POOL,
PROP_MOUNT_POINTS,
PROP_DROP_BACKLOG,
PROP_LAST
};
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,
SIGNAL_HANDLE_RESPONSE,
SIGNAL_SEND_MESSAGE,
SIGNAL_ANNOUNCE_REQUEST,
SIGNAL_RECORD_REQUEST,
SIGNAL_CHECK_REQUIREMENTS,
SIGNAL_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
#define GST_CAT_DEFAULT rtsp_client_debug
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 gboolean handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPMedia * media, GstSDPMessage * sdp);
static gboolean default_configure_client_media (GstRTSPClient * client,
GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
static gboolean default_configure_client_transport (GstRTSPClient * client,
GstRTSPContext * ctx, GstRTSPTransport * ct);
static GstRTSPResult default_params_set (GstRTSPClient * client,
GstRTSPContext * ctx);
static GstRTSPResult default_params_get (GstRTSPClient * client,
GstRTSPContext * ctx);
static gchar *default_make_path_from_uri (GstRTSPClient * client,
const GstRTSPUrl * uri);
static void client_session_removed (GstRTSPSessionPool * pool,
GstRTSPSession * session, GstRTSPClient * client);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
static void
gst_rtsp_client_class_init (GstRTSPClientClass * klass)
{
GObjectClass *gobject_class;
g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
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;
klass->handle_sdp = handle_sdp;
klass->configure_client_media = default_configure_client_media;
klass->configure_client_transport = default_configure_client_transport;
klass->params_set = default_params_set;
klass->params_get = default_params_get;
klass->make_path_from_uri = default_make_path_from_uri;
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
"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_DROP_BACKLOG,
g_param_spec_boolean ("drop-backlog", "Drop Backlog",
"Drop data when the backlog queue is full",
DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_rtsp_client_signals[SIGNAL_CLOSED] =
g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
handle_response), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::send-message:
* @client: The RTSP client
* @session: (type GstRtspServer.RTSPSession): The session
* @message: (type GstRtsp.RTSPMessage): The message
*/
gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
send_message), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST] =
g_signal_new ("announce-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, announce_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST] =
g_signal_new ("record-request", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, record_request),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_RTSP_CONTEXT);
/**
* GstRTSPClient::check-requirements:
* @client: a #GstRTSPClient
* @ctx: a #GstRTSPContext
* @arr: a NULL-terminated array of strings
*
* Returns: a newly allocated string with comma-separated list of
* unsupported options. An empty string must be returned if
* all options are supported.
*
* Since: 1.6
*/
gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS] =
g_signal_new ("check-requirements", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
check_requirements), NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_STRING, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_STRV);
tunnels =
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
g_mutex_init (&tunnels_lock);
GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
client->priv = priv;
g_mutex_init (&priv->lock);
g_mutex_init (&priv->send_lock);
g_mutex_init (&priv->watch_lock);
priv->close_seq = 0;
priv->drop_backlog = DEFAULT_DROP_BACKLOG;
priv->transports =
g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL,
g_object_unref);
}
static GstRTSPFilterResult
filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
gpointer user_data)
{
gboolean *closed = user_data;
GstRTSPMedia *media;
guint i, n_streams;
gboolean is_all_udp = TRUE;
media = gst_rtsp_session_media_get_media (sessmedia);
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPStreamTransport *transport =
gst_rtsp_session_media_get_transport (sessmedia, i);
const GstRTSPTransport *rtsp_transport;
if (!transport)
continue;
rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
if (rtsp_transport
&& rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP
&& rtsp_transport->lower_transport != GST_RTSP_LOWER_TRANS_UDP_MCAST) {
is_all_udp = FALSE;
break;
}
}
if (!is_all_udp || gst_rtsp_media_is_stop_on_disconnect (media)) {
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
return GST_RTSP_FILTER_REMOVE;
} else {
*closed = FALSE;
return GST_RTSP_FILTER_KEEP;
}
}
static void
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
{
GstRTSPClientPrivate *priv = client->priv;
g_mutex_lock (&priv->lock);
/* check if we already know about this session */
if (g_list_find (priv->sessions, session) == NULL) {
GST_INFO ("watching session %p", session);
priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
priv->sessions_cookie++;
/* connect removed session handler, it will be disconnected when the last
* session gets removed */
if (priv->session_removed_id == 0)
priv->session_removed_id = g_signal_connect_data (priv->session_pool,
"session-removed", G_CALLBACK (client_session_removed),
g_object_ref (client), (GClosureNotify) g_object_unref, 0);
}
g_mutex_unlock (&priv->lock);
return;
}
/* should be called with lock */
static void
client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
GList * link)
{
GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("client %p: unwatch session %p", client, session);
if (link == NULL) {
link = g_list_find (priv->sessions, session);
if (link == NULL)
return;
}
priv->sessions = g_list_delete_link (priv->sessions, link);
priv->sessions_cookie++;
/* if this was the last session, disconnect the handler.
* This will also drop the extra client ref */
if (!priv->sessions) {
g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
priv->session_removed_id = 0;
}
/* remove the session */
g_object_unref (session);
}
static GstRTSPFilterResult
cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
gpointer user_data)
{
gboolean *closed = user_data;
/* unlink all media managed in this session. This needs to happen
* without the client lock, so we really want to do it here. */
gst_rtsp_session_filter (sess, filter_session_media, user_data);
if (*closed)
return GST_RTSP_FILTER_REMOVE;
else
return GST_RTSP_FILTER_KEEP;
}
static void
clean_cached_media (GstRTSPClient * client, gboolean unprepare)
{
GstRTSPClientPrivate *priv = client->priv;
if (priv->path) {
g_free (priv->path);
priv->path = NULL;
}
if (priv->media) {
if (unprepare)
gst_rtsp_media_unprepare (priv->media);
g_object_unref (priv->media);
priv->media = NULL;
}
}
/* A client is finalized when the connection is broken */
static void
gst_rtsp_client_finalize (GObject * obj)
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("finalize client %p", client);
if (priv->watch)
gst_rtsp_watch_set_flushing (priv->watch, TRUE);
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
if (priv->watch)
g_source_destroy ((GSource *) priv->watch);
if (priv->watch_context)
g_main_context_unref (priv->watch_context);
/* all sessions should have been removed by now. We keep a ref to
* the client object for the session removed handler. The ref is
* dropped when the last session is removed from the list. */
g_assert (priv->sessions == NULL);
g_assert (priv->session_removed_id == 0);
g_hash_table_unref (priv->transports);
if (priv->connection)
gst_rtsp_connection_free (priv->connection);
if (priv->session_pool) {
g_object_unref (priv->session_pool);
}
if (priv->mount_points)
g_object_unref (priv->mount_points);
if (priv->auth)
g_object_unref (priv->auth);
if (priv->thread_pool)
g_object_unref (priv->thread_pool);
clean_cached_media (client, TRUE);
g_free (priv->server_ip);
g_mutex_clear (&priv->lock);
g_mutex_clear (&priv->send_lock);
g_mutex_clear (&priv->watch_lock);
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
static void
gst_rtsp_client_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
GstRTSPClientPrivate *priv = client->priv;
switch (propid) {
case PROP_SESSION_POOL:
g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
break;
case PROP_MOUNT_POINTS:
g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
break;
case PROP_DROP_BACKLOG:
g_value_set_boolean (value, priv->drop_backlog);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_client_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
GstRTSPClientPrivate *priv = client->priv;
switch (propid) {
case PROP_SESSION_POOL:
gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
break;
case PROP_MOUNT_POINTS:
gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
break;
case PROP_DROP_BACKLOG:
g_mutex_lock (&priv->lock);
priv->drop_backlog = g_value_get_boolean (value);
g_mutex_unlock (&priv->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_client_new:
*
* Create a new #GstRTSPClient instance.
*
* Returns: (transfer full): a new #GstRTSPClient
*/
GstRTSPClient *
gst_rtsp_client_new (void)
{
GstRTSPClient *result;
result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
return result;
}
static void
send_message (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPMessage * message, gboolean close)
{
GstRTSPClientPrivate *priv = client->priv;
gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
"GStreamer RTSP server");
/* remove any previous header */
gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
/* add the new session header for new session ids */
if (ctx->session) {
gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
gst_rtsp_session_get_header (ctx->session));
}
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (message);
}
if (close)
gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
0, ctx, message);
g_mutex_lock (&priv->send_lock);
if (priv->send_func)
priv->send_func (client, message, close, priv->send_data);
g_mutex_unlock (&priv->send_lock);
gst_rtsp_message_unset (message);
}
static void
send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
GstRTSPContext * ctx)
{
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
ctx->session = NULL;
send_message (client, ctx, ctx->response, FALSE);
}
static void
send_option_not_supported_response (GstRTSPClient * client,
GstRTSPContext * ctx, const gchar * unsupported_options)
{
GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
if (unsupported_options != NULL) {
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
unsupported_options);
}
ctx->session = NULL;
send_message (client, ctx, ctx->response, FALSE);
}
static gboolean
paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
{
if (path1 == NULL || path2 == NULL)
return FALSE;
if (strlen (path1) != len2)
return FALSE;
if (strncmp (path1, path2, len2))
return FALSE;
return TRUE;
}
/* this function is called to initially find the media for the DESCRIBE request
* but is cached for when the same client (without breaking the connection) is
* doing a setup for the exact same url. */
static GstRTSPMedia *
find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
gint * matched)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
gint path_len;
/* find the longest matching factory for the uri first */
if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
path, matched)))
goto no_factory;
ctx->factory = factory;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
goto no_factory_access;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
goto not_authorized;
if (matched)
path_len = *matched;
else
path_len = strlen (path);
if (!paths_are_equal (priv->path, path, path_len)) {
/* remove any previously cached values before we try to construct a new
* media for uri */
clean_cached_media (client, TRUE);
/* prepare the media and add it to the pipeline */
if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
goto no_media;
ctx->media = media;
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD)) {
GstRTSPThread *thread;
thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
GST_RTSP_THREAD_TYPE_MEDIA, ctx);
if (thread == NULL)
goto no_thread;
/* prepare the media */
if (!gst_rtsp_media_prepare (media, thread))
goto no_prepare;
}
/* now keep track of the uri and the media */
priv->path = g_strndup (path, path_len);
priv->media = media;
} else {
/* we have seen this path before, used cached media */
media = priv->media;
ctx->media = media;
GST_INFO ("reusing cached media %p for path %s", media, priv->path);
}
g_object_unref (factory);
ctx->factory = NULL;
if (media)
g_object_ref (media);
return media;
/* ERRORS */
no_factory:
{
GST_ERROR ("client %p: no factory for path %s", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return NULL;
}
no_factory_access:
{
GST_ERROR ("client %p: not authorized to see factory path %s", client,
path);
/* error reply is already sent */
return NULL;
}
not_authorized:
{
GST_ERROR ("client %p: not authorized for factory path %s", client, path);
/* error reply is already sent */
return NULL;
}
no_media:
{
GST_ERROR ("client %p: can't create media", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
g_object_unref (factory);
ctx->factory = NULL;
return NULL;
}
no_thread:
{
GST_ERROR ("client %p: can't create thread", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
ctx->media = NULL;
g_object_unref (factory);
ctx->factory = NULL;
return NULL;
}
no_prepare:
{
GST_ERROR ("client %p: can't prepare media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
ctx->media = NULL;
g_object_unref (factory);
ctx->factory = NULL;
return NULL;
}
}
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMessage message = { 0 };
GstRTSPResult res = GST_RTSP_OK;
GstMapInfo map_info;
guint8 *data;
guint usize;
gst_rtsp_message_init_data (&message, channel);
/* FIXME, need some sort of iovec RTSPMessage here */
if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
return FALSE;
gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
g_mutex_lock (&priv->send_lock);
if (priv->send_func)
res = priv->send_func (client, &message, FALSE, priv->send_data);
g_mutex_unlock (&priv->send_lock);
gst_rtsp_message_steal_body (&message, &data, &usize);
gst_buffer_unmap (buffer, &map_info);
gst_rtsp_message_unset (&message);
return res == GST_RTSP_OK;
}
/**
* gst_rtsp_client_close:
* @client: a #GstRTSPClient
*
* Close the connection of @client and remove all media it was managing.
*
* Since: 1.4
*/
void
gst_rtsp_client_close (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_DEBUG ("client %p: closing connection", client);
if (priv->connection) {
if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
gst_rtsp_connection_close (priv->connection);
}
/* connection is now closed, destroy the watch which will also cause the
* closed signal to be emitted */
if (priv->watch) {
GST_DEBUG ("client %p: destroying watch", client);
g_source_destroy ((GSource *) priv->watch);
priv->watch = NULL;
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
g_main_context_unref (priv->watch_context);
priv->watch_context = NULL;
}
}
static gchar *
default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
{
gchar *path;
if (uri->query)
path = g_strconcat (uri->abspath, "?", uri->query, NULL);
else
path = g_strdup (uri->abspath);
return path;
}
static gboolean
handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPClientClass *klass;
GstRTSPSession *session;
GstRTSPSessionMedia *sessmedia;
GstRTSPStatusCode code;
gchar *path;
gint matched;
gboolean keep_session;
if (!ctx->session)
goto no_session;
session = ctx->session;
if (!ctx->uri)
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (!sessmedia)
goto not_found;
/* only aggregate control for now.. */
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
ctx->sessmedia = sessmedia;
/* we emit the signal before closing the connection */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
0, ctx);
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
/* unmanage the media in the session, returns false if all media session
* are torn down. */
keep_session = gst_rtsp_session_release_media (session, sessmedia);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
send_message (client, ctx, ctx->response, TRUE);
if (!keep_session) {
/* remove the session */
gst_rtsp_session_pool_remove (priv->session_pool, session);
}
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_free (path);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_free (path);
return FALSE;
}
}
static GstRTSPResult
default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
res = gst_rtsp_params_set (client, ctx);
return res;
}
static GstRTSPResult
default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
res = gst_rtsp_params_get (client, ctx);
return res;
}
static gboolean
handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
res = gst_rtsp_message_get_body (ctx->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, ctx);
} else {
/* there is a body, handle the params */
res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
send_message (client, ctx, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
0, ctx);
return TRUE;
/* ERRORS */
bad_request:
{
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPResult res;
guint8 *data;
guint size;
res = gst_rtsp_message_get_body (ctx->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, ctx);
} else {
/* there is a body, handle the params */
res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
if (res != GST_RTSP_OK)
goto bad_request;
send_message (client, ctx, ctx->response, FALSE);
}
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
0, ctx);
return TRUE;
/* ERRORS */
bad_request:
{
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
GstRTSPState rtspstate;
gchar *path;
gint matched;
guint i, n;
if (!(session = ctx->session))
goto no_session;
if (!ctx->uri)
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, ctx->uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (!sessmedia)
goto not_found;
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
media = gst_rtsp_session_media_get_media (sessmedia);
n = gst_rtsp_media_n_streams (media);
for (i = 0; i < n; i++) {
GstRTSPStream *stream = gst_rtsp_media_get_stream (media, i);
if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET)
goto not_supported;
}
ctx->sessmedia = sessmedia;
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
/* the session state must be playing or recording */
if (rtspstate != GST_RTSP_STATE_PLAYING &&
rtspstate != GST_RTSP_STATE_RECORDING)
goto invalid_state;
/* then pause sending */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
send_message (client, ctx, ctx->response, FALSE);
/* the state is now READY */
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no seesion", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: no media for uri", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_free (path);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_free (path);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or RECORDING", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
return FALSE;
}
not_supported:
{
GST_ERROR ("client %p: pausing not supported", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
/* convert @url and @path to a URL used as a content base for the factory
* located at @path */
static gchar *
make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
{
GstRTSPUrl tmp;
gchar *result;
const gchar *trail;
/* check for trailing '/' and append one */
trail = (path[strlen (path) - 1] != '/' ? "/" : "");
tmp = *url;
tmp.user = NULL;
tmp.passwd = NULL;
tmp.abspath = g_strdup_printf ("%s%s", path, trail);
tmp.query = NULL;
result = gst_rtsp_url_get_request_uri (&tmp);
g_free (tmp.abspath);
return result;
}
static gboolean
handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
GstRTSPUrl *uri;
gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
GstRTSPState rtspstate;
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
gchar *path, *rtpinfo;
gint matched;
if (!(session = ctx->session))
goto no_session;
if (!(uri = ctx->uri))
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (!sessmedia)
goto not_found;
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
ctx->sessmedia = sessmedia;
ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY))
goto unsupported_mode;
/* the session state must be playing or ready */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
/* in play we first unsuspend, media could be suspended from SDP or PAUSED */
if (!gst_rtsp_media_unsuspend (media))
goto unsuspend_failed;
/* parse the range header if we have one */
res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
if (res == GST_RTSP_OK) {
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
GstRTSPMediaStatus media_status;
/* we have a range, seek to the position */
unit = range->unit;
gst_rtsp_media_seek (media, range);
gst_rtsp_range_free (range);
media_status = gst_rtsp_media_get_status (media);
if (media_status == GST_RTSP_MEDIA_STATUS_ERROR)
goto seek_failed;
}
}
/* grab RTPInfo from the media now */
rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
/* add the RTP-Info header */
if (rtpinfo)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
rtpinfo);
/* add the range */
str = gst_rtsp_media_get_range_string (media, TRUE, unit);
if (str)
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
send_message (client, ctx, ctx->response, FALSE);
/* start playing after sending the response */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_free (path);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or READY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
return FALSE;
}
unsuspend_failed:
{
GST_ERROR ("client %p: unsuspend failed", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
return FALSE;
}
seek_failed:
{
GST_ERROR ("client %p: seek failed", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support PLAY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
return FALSE;
}
}
static void
do_keepalive (GstRTSPSession * session)
{
GST_INFO ("keep session %p alive", session);
gst_rtsp_session_touch (session);
}
/* parse @transport and return a valid transport in @tr. only transports
* supported by @stream are returned. Returns FALSE if no valid transport
* was found. */
static gboolean
parse_transport (const char *transport, GstRTSPStream * stream,
GstRTSPTransport * tr)
{
gint i;
gboolean res;
gchar **transports;
res = FALSE;
gst_rtsp_transport_init (tr);
GST_DEBUG ("parsing transports %s", transport);
transports = g_strsplit (transport, ",", 0);
/* loop through the transports, try to parse */
for (i = 0; transports[i]; i++) {
res = gst_rtsp_transport_parse (transports[i], tr);
if (res != GST_RTSP_OK) {
/* no valid transport, search some more */
GST_WARNING ("could not parse transport %s", transports[i]);
goto next;
}
/* we have a transport, see if it's supported */
if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
GST_WARNING ("unsupported transport %s", transports[i]);
goto next;
}
/* we have a valid transport */
GST_INFO ("found valid transport %s", transports[i]);
res = TRUE;
break;
next:
gst_rtsp_transport_init (tr);
}
g_strfreev (transports);
return res;
}
static gboolean
default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
GstRTSPStream * stream, GstRTSPContext * ctx)
{
GstRTSPMessage *request = ctx->request;
gchar *blocksize_str;
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
&blocksize_str, 0) == GST_RTSP_OK) {
guint64 blocksize;
gchar *end;
blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
if (end == blocksize_str)
goto parse_failed;
/* we don't want to change the mtu when this media
* can be shared because it impacts other clients */
if (gst_rtsp_media_is_shared (media))
goto done;
if (blocksize > G_MAXUINT)
blocksize = G_MAXUINT;
gst_rtsp_stream_set_mtu (stream, blocksize);
}
done:
return TRUE;
/* ERRORS */
parse_failed:
{
GST_ERROR_OBJECT (client, "failed to parse blocksize");
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
}
static gboolean
default_configure_client_transport (GstRTSPClient * client,
GstRTSPContext * ctx, GstRTSPTransport * ct)
{
GstRTSPClientPrivate *priv = client->priv;
/* we have a valid transport now, set the destination of the client. */
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
gboolean use_client_settings;
GSocketFamily family;
use_client_settings =
gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
if (ct->destination && use_client_settings) {
family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, family, ct, TRUE))
goto no_udp_protocol;
} else {
GstRTSPAddress *addr;
family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, family, ct,
FALSE))
goto no_udp_protocol;
gst_rtsp_stream_get_server_port (ctx->stream, &ct->port, family);
addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
if (addr == NULL)
goto no_address;
g_free (ct->destination);
ct->destination = g_strdup (addr->address);
ct->port.min = addr->port;
ct->port.max = addr->port + addr->n_ports - 1;
ct->ttl = addr->ttl;
gst_rtsp_address_free (addr);
}
} else {
GstRTSPUrl *url;
url = gst_rtsp_connection_get_url (priv->connection);
g_free (ct->destination);
ct->destination = g_strdup (url->host);
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
GSocket *sock;
GSocketAddress *addr;
sock = gst_rtsp_connection_get_read_socket (priv->connection);
if ((addr = g_socket_get_remote_address (sock, NULL))) {
/* our read port is the sender port of client */
ct->client_port.min =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
if ((addr = g_socket_get_local_address (sock, NULL))) {
ct->server_port.max =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
sock = gst_rtsp_connection_get_write_socket (priv->connection);
if ((addr = g_socket_get_remote_address (sock, NULL))) {
/* our write port is the receiver port of client */
ct->client_port.max =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
if ((addr = g_socket_get_local_address (sock, NULL))) {
ct->server_port.min =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
g_object_unref (addr);
}
/* check if the client selected channels for TCP */
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
&ct->interleaved);
}
} else if (ct->lower_transport & GST_RTSP_LOWER_TRANS_UDP) {
GSocketFamily family;
family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, family, ct,
FALSE))
goto no_udp_protocol;
}
}
return TRUE;
/* ERRORS */
no_address:
{
GST_ERROR_OBJECT (client, "failed to acquire address for stream");
return FALSE;
}
no_udp_protocol:
{
GST_ERROR_OBJECT (client, "failed to allocate udp ports");
return FALSE;
}
}
static GstRTSPTransport *
make_server_transport (GstRTSPClient * client, GstRTSPMedia * media,
GstRTSPContext * ctx, GstRTSPTransport * ct)
{
GstRTSPTransport *st;
GInetAddress *addr;
GSocketFamily family;
/* prepare the server transport */
gst_rtsp_transport_new (&st);
st->trans = ct->trans;
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
st->mode_play = ct->mode_play;
st->mode_record = ct->mode_record;
addr = g_inet_address_new_from_string (ct->destination);
if (!addr) {
GST_ERROR ("failed to get inet addr from client destination");
family = G_SOCKET_FAMILY_IPV4;
} else {
family = g_inet_address_get_family (addr);
g_object_unref (addr);
addr = NULL;
}
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
st->port = ct->port;
st->destination = g_strdup (ct->destination);
st->ttl = ct->ttl;
break;
case GST_RTSP_LOWER_TRANS_TCP:
st->interleaved = ct->interleaved;
st->client_port = ct->client_port;
st->server_port = ct->server_port;
default:
break;
}
if ((gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY))
gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
return st;
}
#define AES_128_KEY_LEN 16
#define AES_256_KEY_LEN 32
#define HMAC_32_KEY_LEN 4
#define HMAC_80_KEY_LEN 10
static gboolean
mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
{
const gchar *srtp_cipher;
const gchar *srtp_auth;
const GstMIKEYPayload *sp;
guint i;
/* loop over Security policy until we find one containing policy */
for (i = 0;; i++) {
if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
break;
if (((GstMIKEYPayloadSP *) sp)->policy == policy)
break;
}
/* the default ciphers */
srtp_cipher = "aes-128-icm";
srtp_auth = "hmac-sha1-80";
/* now override the defaults with what is in the Security Policy */
if (sp != NULL) {
guint len;
/* collect all the params and go over them */
len = gst_mikey_payload_sp_get_n_params (sp);
for (i = 0; i < len; i++) {
const GstMIKEYPayloadSPParam *param =
gst_mikey_payload_sp_get_param (sp, i);
switch (param->type) {
case GST_MIKEY_SP_SRTP_ENC_ALG:
switch (param->val[0]) {
case 0:
srtp_cipher = "null";
break;
case 2:
case 1:
srtp_cipher = "aes-128-icm";
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
switch (param->val[0]) {
case AES_128_KEY_LEN:
srtp_cipher = "aes-128-icm";
break;
case AES_256_KEY_LEN:
srtp_cipher = "aes-256-icm";
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_AUTH_ALG:
switch (param->val[0]) {
case 0:
srtp_auth = "null";
break;
case 2:
case 1:
srtp_auth = "hmac-sha1-80";
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
switch (param->val[0]) {
case HMAC_32_KEY_LEN:
srtp_auth = "hmac-sha1-32";
break;
case HMAC_80_KEY_LEN:
srtp_auth = "hmac-sha1-80";
break;
default:
break;
}
break;
case GST_MIKEY_SP_SRTP_SRTP_ENC:
break;
case GST_MIKEY_SP_SRTP_SRTCP_ENC:
break;
default:
break;
}
}
}
/* now configure the SRTP parameters */
gst_caps_set_simple (caps,
"srtp-cipher", G_TYPE_STRING, srtp_cipher,
"srtp-auth", G_TYPE_STRING, srtp_auth,
"srtcp-cipher", G_TYPE_STRING, srtp_cipher,
"srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
return TRUE;
}
static gboolean
handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
guint8 * data, gsize size)
{
GstMIKEYMessage *msg;
guint i, n_cs;
GstCaps *caps = NULL;
GstMIKEYPayloadKEMAC *kemac;
const GstMIKEYPayloadKeyData *pkd;
GstBuffer *key;
/* the MIKEY message contains a CSB or crypto session bundle. It is a
* set of Crypto Sessions protected with the same master key.
* In the context of SRTP, an RTP and its RTCP stream is part of a
* crypto session */
if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
goto parse_failed;
/* we can only handle SRTP crypto sessions for now */
if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
goto invalid_map_type;
/* get the number of crypto sessions. This maps SSRC to its
* security parameters */
n_cs = gst_mikey_message_get_n_cs (msg);
if (n_cs == 0)
goto no_crypto_sessions;
/* we also need keys */
if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
(msg, GST_MIKEY_PT_KEMAC, 0)))
goto no_keys;
/* we don't support encrypted keys */
if (kemac->enc_alg != GST_MIKEY_ENC_NULL
|| kemac->mac_alg != GST_MIKEY_MAC_NULL)
goto unsupported_encryption;
/* get Key data sub-payload */
pkd = (const GstMIKEYPayloadKeyData *)
gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
key =
gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
pkd->key_len);
/* go over all crypto sessions and create the security policy for each
* SSRC */
for (i = 0; i < n_cs; i++) {
const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
caps = gst_caps_new_simple ("application/x-srtp",
"ssrc", G_TYPE_UINT, map->ssrc,
"roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
mikey_apply_policy (caps, msg, map->policy);
gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
gst_caps_unref (caps);
}
gst_mikey_message_unref (msg);
gst_buffer_unref (key);
return TRUE;
/* ERRORS */
parse_failed:
{
GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
return FALSE;
}
invalid_map_type:
{
GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
goto cleanup_message;
}
no_crypto_sessions:
{
GST_DEBUG_OBJECT (client, "no crypto sessions");
goto cleanup_message;
}
no_keys:
{
GST_DEBUG_OBJECT (client, "no keys found");
goto cleanup_message;
}
unsupported_encryption:
{
GST_DEBUG_OBJECT (client, "unsupported key encryption");
goto cleanup_message;
}
cleanup_message:
{
gst_mikey_message_unref (msg);
return FALSE;
}
}
#define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
static void
strip_chars (gchar * str)
{
gchar *s;
gsize len;
len = strlen (str);
while (len--) {
if (!IS_STRIP_CHAR (str[len]))
break;
str[len] = '\0';
}
for (s = str; *s && IS_STRIP_CHAR (*s); s++);
memmove (str, s, len + 1);
}
/* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
* key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
*/
static gboolean
handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
{
gchar **specs;
gint i, j;
specs = g_strsplit (keymgmt, ",", 0);
for (i = 0; specs[i]; i++) {
gchar **split;
split = g_strsplit (specs[i], ";", 0);
for (j = 0; split[j]; j++) {
g_strstrip (split[j]);
if (g_str_has_prefix (split[j], "prot=")) {
g_strstrip (split[j] + 5);
if (!g_str_equal (split[j] + 5, "mikey"))
break;
GST_DEBUG ("found mikey");
} else if (g_str_has_prefix (split[j], "uri=")) {
strip_chars (split[j] + 4);
GST_DEBUG ("found uri '%s'", split[j] + 4);
} else if (g_str_has_prefix (split[j], "data=")) {
guchar *data;
gsize size;
strip_chars (split[j] + 5);
GST_DEBUG ("found data '%s'", split[j] + 5);
data = g_base64_decode_inplace (split[j] + 5, &size);
handle_mikey_data (client, ctx, data, size);
}
}
g_strfreev (split);
}
g_strfreev (specs);
return TRUE;
}
static gboolean
handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstRTSPUrl *uri;
gchar *transport, *keymgmt;
GstRTSPTransport *ct, *st;
GstRTSPStatusCode code;
GstRTSPSession *session;
GstRTSPStreamTransport *trans;
gchar *trans_str;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStream *stream;
GstRTSPState rtspstate;
GstRTSPClientClass *klass;
gchar *path, *control = NULL;
gint matched;
gboolean new_session = FALSE;
if (!ctx->uri)
goto no_uri;
uri = ctx->uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, uri);
/* parse the transport */
res =
gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
&transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
/* we create the session after parsing stuff so that we don't make
* a session for malformed requests */
if (priv->session_pool == NULL)
goto no_pool;
session = ctx->session;
if (session) {
g_object_ref (session);
/* get a handle to the configuration of the media in the session, this can
* return NULL if this is a new url to manage in this session. */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
} else {
/* we need a new media configuration in this session */
sessmedia = NULL;
}
/* we have no session media, find one and manage it */
if (sessmedia == NULL) {
/* get a handle to the configuration of the media in the session */
media = find_media (client, ctx, path, &matched);
/* need to suspend the media, if the protocol has changed */
if (media != NULL)
gst_rtsp_media_suspend (media);
} else {
if ((media = gst_rtsp_session_media_get_media (sessmedia)))
g_object_ref (media);
else
goto media_not_found;
}
/* no media, not found then */
if (media == NULL)
goto media_not_found_no_reply;
if (path[matched] == '\0') {
if (gst_rtsp_media_n_streams (media) == 1) {
stream = gst_rtsp_media_get_stream (media, 0);
} else {
goto control_not_found;
}
} else {
/* path is what matched. */
path[matched] = '\0';
/* control is remainder */
control = &path[matched + 1];
/* find the stream now using the control part */
stream = gst_rtsp_media_find_stream (media, control);
}
if (stream == NULL)
goto stream_not_found;
/* now we have a uri identifying a valid media and stream */
ctx->stream = stream;
ctx->media = media;
if (session == NULL) {
/* create a session if this fails we probably reached our session limit or
* something. */
if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
goto service_unavailable;
/* make sure this client is closed when the session is closed */
client_watch_session (client, session);
new_session = TRUE;
/* signal new session */
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
session);
ctx->session = session;
}
if (!klass->configure_client_media (client, media, stream, ctx))
goto configure_media_failed_no_reply;
gst_rtsp_transport_new (&ct);
/* parse and find a usable supported transport */
if (!parse_transport (transport, stream, ct))
goto unsupported_transports;
if ((ct->mode_play
&& !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY)) || (ct->mode_record
&& !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD)))
goto unsupported_mode;
/* parse the keymgmt */
if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
&keymgmt, 0) == GST_RTSP_OK) {
if (!handle_keymgmt (client, ctx, keymgmt))
goto keymgmt_error;
}
if (sessmedia == NULL) {
/* manage the media in our session now, if not done already */
sessmedia = gst_rtsp_session_manage_media (session, path, media);
/* if we stil have no media, error */
if (sessmedia == NULL)
goto sessmedia_unavailable;
/* don't cache media anymore */
clean_cached_media (client, FALSE);
} else {
g_object_unref (media);
}
ctx->sessmedia = sessmedia;
/* update the client transport */
if (!klass->configure_client_transport (client, ctx, ct))
goto unsupported_client_transport;
/* set in the session media transport */
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
ctx->trans = trans;
/* configure the url used to set this transport, this we will use when
* generating the response for the PLAY request */
gst_rtsp_stream_transport_set_url (trans, uri);
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* our callbacks to send data on this TCP connection */
gst_rtsp_stream_transport_set_callbacks (trans,
(GstRTSPSendFunc) do_send_data,
(GstRTSPSendFunc) do_send_data, client, NULL);
g_hash_table_insert (priv->transports,
GINT_TO_POINTER (ct->interleaved.min), trans);
g_object_ref (trans);
g_hash_table_insert (priv->transports,
GINT_TO_POINTER (ct->interleaved.max), trans);
g_object_ref (trans);
}
/* create and serialize the server transport */
st = make_server_transport (client, media, ctx, ct);
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (ctx->response, code,
gst_rtsp_status_as_text (code), ctx->request);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
trans_str);
g_free (trans_str);
send_message (client, ctx, ctx->response, FALSE);
/* update the state */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
switch (rtspstate) {
case GST_RTSP_STATE_PLAYING:
case GST_RTSP_STATE_RECORDING:
case GST_RTSP_STATE_READY:
/* no state change */
break;
default:
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
break;
}
g_object_unref (session);
g_free (path);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
return TRUE;
/* ERRORS */
no_uri:
{
GST_ERROR ("client %p: no uri", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_transport:
{
GST_ERROR ("client %p: no transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_path;
}
no_pool:
{
GST_ERROR ("client %p: no session pool configured", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto cleanup_path;
}
media_not_found_no_reply:
{
GST_ERROR ("client %p: media '%s' not found", client, path);
/* error reply is already sent */
goto cleanup_path;
}
media_not_found:
{
GST_ERROR ("client %p: media '%s' not found", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
goto cleanup_path;
}
control_not_found:
{
GST_ERROR ("client %p: no control in path '%s'", client, path);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_object_unref (media);
goto cleanup_path;
}
stream_not_found:
{
GST_ERROR ("client %p: stream '%s' not found", client,
GST_STR_NULL (control));
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
g_object_unref (media);
goto cleanup_path;
}
service_unavailable:
{
GST_ERROR ("client %p: can't create session", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
goto cleanup_path;
}
sessmedia_unavailable:
{
GST_ERROR ("client %p: can't create session media", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_object_unref (media);
goto cleanup_session;
}
configure_media_failed_no_reply:
{
GST_ERROR ("client %p: configure_media failed", client);
/* error reply is already sent */
goto cleanup_session;
}
unsupported_transports:
{
GST_ERROR ("client %p: unsupported transports", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_transport;
}
unsupported_client_transport:
{
GST_ERROR ("client %p: unsupported client transport", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_transport;
}
unsupported_mode:
{
GST_ERROR ("client %p: unsupported mode (media play: %d, media record: %d, "
"mode play: %d, mode record: %d)", client,
! !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY),
! !(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD), ct->mode_play, ct->mode_record);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
goto cleanup_transport;
}
keymgmt_error:
{
GST_ERROR ("client %p: keymgmt error", client);
send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
goto cleanup_transport;
}
{
cleanup_transport:
gst_rtsp_transport_free (ct);
cleanup_session:
if (new_session)
gst_rtsp_session_pool_remove (priv->session_pool, session);
g_object_unref (session);
cleanup_path:
g_free (path);
return FALSE;
}
}
static GstSDPMessage *
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
{
GstRTSPClientPrivate *priv = client->priv;
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
guint64 session_id_tmp;
gchar session_id[21];
gst_sdp_message_new (&sdp);
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
if (priv->is_ipv6)
proto = "IP6";
else
proto = "IP4";
session_id_tmp = (((guint64) g_random_int ()) << 32) | g_random_int ();
g_snprintf (session_id, sizeof (session_id), "%" G_GUINT64_FORMAT,
session_id_tmp);
gst_sdp_message_set_origin (sdp, "-", session_id, "1", "IN", proto,
priv->server_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtsp-server");
gst_sdp_message_add_time (sdp, "0", "0", NULL);
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
gst_sdp_message_add_attribute (sdp, "type", "broadcast");
gst_sdp_message_add_attribute (sdp, "control", "*");
info.is_ipv6 = priv->is_ipv6;
info.server_ip = priv->server_ip;
/* create an SDP for the media object */
if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
goto no_sdp;
return sdp;
/* ERRORS */
no_sdp:
{
GST_ERROR ("client %p: could not create SDP", client);
gst_sdp_message_free (sdp);
return NULL;
}
}
/* for the describe we must generate an SDP */
static gboolean
handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstSDPMessage *sdp;
guint i;
gchar *path, *str;
GstRTSPMedia *media;
GstRTSPClientClass *klass;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
if (!ctx->uri)
goto no_uri;
/* check what kind of format is accepted, we don't really do anything with it
* and always return SDP for now. */
for (i = 0;; i++) {
gchar *accept;
res =
gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
&accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
break;
}
if (!priv->mount_points)
goto no_mount_points;
if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
goto no_path;
/* find the media object for the uri */
if (!(media = find_media (client, ctx, path, NULL)))
goto no_media;
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_PLAY))
goto unsupported_mode;
/* create an SDP for the media object on this client */
if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
/* we suspend after the describe */
gst_rtsp_media_suspend (media);
g_object_unref (media);
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* content base for some clients that might screw up creating the setup uri */
str = make_base_url (client, ctx->uri, path);
g_free (path);
GST_INFO ("adding content-base: %s", str);
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
0, ctx);
return TRUE;
/* ERRORS */
no_uri:
{
GST_ERROR ("client %p: no uri", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_mount_points:
{
GST_ERROR ("client %p: no mount points configured", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_path:
{
GST_ERROR ("client %p: can't find path for url", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_media:
{
GST_ERROR ("client %p: no media", client);
g_free (path);
/* error reply is already sent */
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support DESCRIBE", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
g_free (path);
g_object_unref (media);
return FALSE;
}
no_sdp:
{
GST_ERROR ("client %p: can't create SDP", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_free (path);
g_object_unref (media);
return FALSE;
}
}
static gboolean
handle_sdp (GstRTSPClient * client, GstRTSPContext * ctx, GstRTSPMedia * media,
GstSDPMessage * sdp)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPThread *thread;
/* create an SDP for the media object */
if (!gst_rtsp_media_handle_sdp (media, sdp))
goto unhandled_sdp;
thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
GST_RTSP_THREAD_TYPE_MEDIA, ctx);
if (thread == NULL)
goto no_thread;
/* prepare the media */
if (!gst_rtsp_media_prepare (media, thread))
goto no_prepare;
return TRUE;
/* ERRORS */
unhandled_sdp:
{
GST_ERROR ("client %p: could not handle SDP", client);
return FALSE;
}
no_thread:
{
GST_ERROR ("client %p: can't create thread", client);
return FALSE;
}
no_prepare:
{
GST_ERROR ("client %p: can't prepare media", client);
return FALSE;
}
}
static gboolean
handle_announce_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPClientClass *klass;
GstSDPResult sres;
GstSDPMessage *sdp;
GstRTSPMedia *media;
gchar *path, *cont = NULL;
guint8 *data;
guint size;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
if (!ctx->uri)
goto no_uri;
if (!priv->mount_points)
goto no_mount_points;
/* check if reply is SDP */
gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_CONTENT_TYPE, &cont,
0);
/* could not be set but since the request returned OK, we assume it
* was SDP, else check it. */
if (cont) {
if (g_ascii_strcasecmp (cont, "application/sdp") != 0)
goto wrong_content_type;
}
/* get message body and parse as SDP */
gst_rtsp_message_get_body (ctx->request, &data, &size);
if (data == NULL || size == 0)
goto no_message;
GST_DEBUG ("client %p: parse SDP...", client);
gst_sdp_message_new (&sdp);
sres = gst_sdp_message_parse_buffer (data, size, sdp);
if (sres != GST_SDP_OK)
goto sdp_parse_failed;
if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
goto no_path;
/* find the media object for the uri */
if (!(media = find_media (client, ctx, path, NULL)))
goto no_media;
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD))
goto unsupported_mode;
/* Tell client subclass about the media */
if (!klass->handle_sdp (client, ctx, media, sdp))
goto unhandled_sdp;
/* we suspend after the announce */
gst_rtsp_media_suspend (media);
g_object_unref (media);
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_ANNOUNCE_REQUEST],
0, ctx);
gst_sdp_message_free (sdp);
g_free (path);
return TRUE;
no_uri:
{
GST_ERROR ("client %p: no uri", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_mount_points:
{
GST_ERROR ("client %p: no mount points configured", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_path:
{
GST_ERROR ("client %p: can't find path for url", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
gst_sdp_message_free (sdp);
return FALSE;
}
wrong_content_type:
{
GST_ERROR ("client %p: unknown content type", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
no_message:
{
GST_ERROR ("client %p: can't find SDP message", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
sdp_parse_failed:
{
GST_ERROR ("client %p: failed to parse SDP message", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
gst_sdp_message_free (sdp);
return FALSE;
}
no_media:
{
GST_ERROR ("client %p: no media", client);
g_free (path);
/* error reply is already sent */
gst_sdp_message_free (sdp);
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support ANNOUNCE", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
g_free (path);
g_object_unref (media);
gst_sdp_message_free (sdp);
return FALSE;
}
unhandled_sdp:
{
GST_ERROR ("client %p: can't handle SDP", client);
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE, ctx);
g_free (path);
g_object_unref (media);
gst_sdp_message_free (sdp);
return FALSE;
}
}
static gboolean
handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPSession *session;
GstRTSPClientClass *klass;
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPUrl *uri;
GstRTSPState rtspstate;
gchar *path;
gint matched;
if (!(session = ctx->session))
goto no_session;
if (!(uri = ctx->uri))
goto no_uri;
klass = GST_RTSP_CLIENT_GET_CLASS (client);
path = klass->make_path_from_uri (client, uri);
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
if (!sessmedia)
goto not_found;
if (path[matched] != '\0')
goto no_aggregate;
g_free (path);
ctx->sessmedia = sessmedia;
ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
if (!(gst_rtsp_media_get_transport_mode (media) &
GST_RTSP_TRANSPORT_MODE_RECORD))
goto unsupported_mode;
/* the session state must be playing or ready */
rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
/* in play we first unsuspend, media could be suspended from SDP or PAUSED */
if (!gst_rtsp_media_unsuspend (media))
goto unsuspend_failed;
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
send_message (client, ctx, ctx->response, FALSE);
/* start playing after sending the response */
gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_RECORD_REQUEST], 0,
ctx);
return TRUE;
/* ERRORS */
no_session:
{
GST_ERROR ("client %p: no session", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
return FALSE;
}
no_uri:
{
GST_ERROR ("client %p: no uri supplied", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
return FALSE;
}
not_found:
{
GST_ERROR ("client %p: media not found", client);
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
return FALSE;
}
no_aggregate:
{
GST_ERROR ("client %p: no aggregate path %s", client, path);
send_generic_response (client,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
g_free (path);
return FALSE;
}
unsupported_mode:
{
GST_ERROR ("client %p: media does not support RECORD", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_ALLOWED, ctx);
return FALSE;
}
invalid_state:
{
GST_ERROR ("client %p: not PLAYING or READY", client);
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
ctx);
return FALSE;
}
unsuspend_failed:
{
GST_ERROR ("client %p: unsuspend failed", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
return FALSE;
}
}
static gboolean
handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
GstRTSPMethod options;
gchar *str;
options = GST_RTSP_DESCRIBE |
GST_RTSP_OPTIONS |
GST_RTSP_PAUSE |
GST_RTSP_PLAY |
GST_RTSP_RECORD | GST_RTSP_ANNOUNCE |
GST_RTSP_SETUP |
GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
str = gst_rtsp_options_as_text (options);
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
send_message (client, ctx, ctx->response, FALSE);
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
0, ctx);
return TRUE;
}
/* remove duplicate and trailing '/' */
static void
sanitize_uri (GstRTSPUrl * uri)
{
gint i, len;
gchar *s, *d;
gboolean have_slash, prev_slash;
s = d = uri->abspath;
len = strlen (uri->abspath);
prev_slash = FALSE;
for (i = 0; i < len; i++) {
have_slash = s[i] == '/';
*d = s[i];
if (!have_slash || !prev_slash)
d++;
prev_slash = have_slash;
}
len = d - uri->abspath;
/* don't remove the first slash if that's the only thing left */
if (len > 1 && *(d - 1) == '/')
d--;
*d = '\0';
}
/* is called when the session is removed from its session pool. */
static void
client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
GST_INFO ("client %p: session %p removed", client, session);
g_mutex_lock (&priv->lock);
if (priv->watch != NULL)
gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
client_unwatch_session (client, session, NULL);
if (priv->watch != NULL)
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
g_mutex_unlock (&priv->lock);
}
/* Check for Require headers. Returns TRUE if there are no Require headers,
* otherwise lets the application decide which headers are supported.
* By default all headers are unsupported.
* If there are unsupported options, FALSE will be returned together with
* a newly-allocated string of (comma-separated) unsupported options in
* the unsupported_reqs variable.
*
* There may be multiple Require headers, but we must send one single
* Unsupported header with all the unsupported options as response. If
* an incoming Require header contained a comma-separated list of options
* GstRtspConnection will already have split that list up into multiple
* headers.
*/
static gboolean
check_request_requirements (GstRTSPContext * ctx, gchar ** unsupported_reqs)
{
GstRTSPResult res;
GPtrArray *arr = NULL;
GstRTSPMessage *msg = ctx->request;
gchar *reqs = NULL;
gint i;
gchar *sig_result = NULL;
gboolean result = TRUE;
i = 0;
do {
res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
if (res == GST_RTSP_ENOTIMPL)
break;
if (arr == NULL)
arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
g_ptr_array_add (arr, g_strdup (reqs));
}
while (TRUE);
/* if we don't have any Require headers at all, all is fine */
if (i == 1)
return TRUE;
/* otherwise we've now processed at all the Require headers */
g_ptr_array_add (arr, NULL);
g_signal_emit (ctx->client,
gst_rtsp_client_signals[SIGNAL_CHECK_REQUIREMENTS], 0, ctx,
(gchar **) arr->pdata, &sig_result);
if (sig_result == NULL) {
/* no supported options, just report all of the required ones as
* unsupported */
*unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
result = FALSE;
goto done;
}
if (strlen (sig_result) == 0)
g_free (sig_result);
else {
*unsupported_reqs = sig_result;
result = FALSE;
}
done:
g_ptr_array_unref (arr);
return result;
}
static void
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPMethod method;
const gchar *uristr;
GstRTSPUrl *uri = NULL;
GstRTSPVersion version;
GstRTSPResult res;
GstRTSPSession *session = NULL;
GstRTSPContext sctx = { NULL }, *ctx;
GstRTSPMessage response = { 0 };
gchar *unsupported_reqs = NULL;
gchar *sessid;
if (!(ctx = gst_rtsp_context_get_current ())) {
ctx = &sctx;
ctx->auth = priv->auth;
gst_rtsp_context_push_current (ctx);
}
ctx->conn = priv->connection;
ctx->client = client;
ctx->request = request;
ctx->response = &response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (request);
}
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
GST_INFO ("client %p: received a request %s %s %s", client,
gst_rtsp_method_as_text (method), uristr,
gst_rtsp_version_as_text (version));
/* we can only handle 1.0 requests */
if (version != GST_RTSP_VERSION_1_0)
goto not_supported;
ctx->method = method;
/* we always try to parse the url first */
if (strcmp (uristr, "*") == 0) {
/* special case where we have * as uri, keep uri = NULL */
} else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
/* check if the uristr is an absolute path <=> scheme and host information
* is missing */
gchar *scheme;
scheme = g_uri_parse_scheme (uristr);
if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
gchar *absolute_uristr = NULL;
GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
if (priv->server_ip == NULL) {
GST_WARNING_OBJECT (client, "host information missing");
goto bad_request;
}
absolute_uristr =
g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
g_free (absolute_uristr);
goto bad_request;
}
g_free (absolute_uristr);
} else {
g_free (scheme);
goto bad_request;
}
}
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
if (priv->session_pool == NULL)
goto no_pool;
/* we had a session in the request, find it again */
if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
goto session_not_found;
/* we add the session to the client list of watched sessions. When a session
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
}
/* sanitize the uri */
if (uri)
sanitize_uri (uri);
ctx->uri = uri;
ctx->session = session;
if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
goto not_authorized;
/* handle any 'Require' headers */
if (!check_request_requirements (ctx, &unsupported_reqs))
goto unsupported_requirement;
/* the backlog must be unlimited while processing requests.
* the causes of this are two cases of deadlocks while streaming over TCP:
*
* 1. consider the scenario where the media pipeline's streaming thread
* is blocking in the appsink (taking the appsink's preroll lock) because
* the backlog is full. when a PAUSE request is received by the RTSP
* client thread then the the state of the session media ought to change
* to PAUSED. while most elements in the pipeline can change state this
* can never happen for the appsink since its preroll lock is taken by
* another thread.
*
* 2. consider the scenario where the media pipeline's streaming thread
* is blocking in the appsink new_sample callback (taking the send lock
* in RTSP client) because the backlog is full. when e.g. a GET request
* is received by the RTSP client thread then a response ought to be sent
* but this can never happen since it requires taking the send lock
* already taken by another thread.
*
* the reason that the backlog is never emptied is that the source used
* for dequeing messages from the backlog is never dispatched because it
* is attached to the same mainloop as the source receving RTSP requests and
* therefore run by the RTSP client thread which is alreayd blocking.
*
* without significant changes the easiest way to cope with this is to
* not block indefinitely when the backlog is full, but rather let the
* backlog grow in size. this in effect means that there can not be any
* upper boundary on its size.
*/
if (priv->watch != NULL)
gst_rtsp_watch_set_send_backlog (priv->watch, 0, 0);
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
handle_options_request (client, ctx);
break;
case GST_RTSP_DESCRIBE:
handle_describe_request (client, ctx);
break;
case GST_RTSP_SETUP:
handle_setup_request (client, ctx);
break;
case GST_RTSP_PLAY:
handle_play_request (client, ctx);
break;
case GST_RTSP_PAUSE:
handle_pause_request (client, ctx);
break;
case GST_RTSP_TEARDOWN:
handle_teardown_request (client, ctx);
break;
case GST_RTSP_SET_PARAMETER:
handle_set_param_request (client, ctx);
break;
case GST_RTSP_GET_PARAMETER:
handle_get_param_request (client, ctx);
break;
case GST_RTSP_ANNOUNCE:
handle_announce_request (client, ctx);
break;
case GST_RTSP_RECORD:
handle_record_request (client, ctx);
break;
case GST_RTSP_REDIRECT:
if (priv->watch != NULL)
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
goto not_implemented;
case GST_RTSP_INVALID:
default:
if (priv->watch != NULL)
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
goto bad_request;
}
if (priv->watch != NULL)
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
done:
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
if (session)
g_object_unref (session);
if (uri)
gst_rtsp_url_free (uri);
return;
/* ERRORS */
not_supported:
{
GST_ERROR ("client %p: version %d not supported", client, version);
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
ctx);
goto done;
}
bad_request:
{
GST_ERROR ("client %p: bad request", client);
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
goto done;
}
no_pool:
{
GST_ERROR ("client %p: no pool configured", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto done;
}
session_not_found:
{
GST_ERROR ("client %p: session not found", client);
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
goto done;
}
not_authorized:
{
GST_ERROR ("client %p: not allowed", client);
/* error reply is already sent */
goto done;
}
unsupported_requirement:
{
GST_ERROR ("client %p: Required option is not supported (%s)", client,
unsupported_reqs);
send_option_not_supported_response (client, ctx, unsupported_reqs);
g_free (unsupported_reqs);
goto done;
}
not_implemented:
{
GST_ERROR ("client %p: method %d not implemented", client, method);
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
goto done;
}
}
static void
handle_response (GstRTSPClient * client, GstRTSPMessage * response)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstRTSPSession *session = NULL;
GstRTSPContext sctx = { NULL }, *ctx;
gchar *sessid;
if (!(ctx = gst_rtsp_context_get_current ())) {
ctx = &sctx;
ctx->auth = priv->auth;
gst_rtsp_context_push_current (ctx);
}
ctx->conn = priv->connection;
ctx->client = client;
ctx->request = NULL;
ctx->uri = NULL;
ctx->method = GST_RTSP_INVALID;
ctx->response = response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (response);
}
GST_INFO ("client %p: received a response", client);
/* get the session if there is any */
res =
gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
if (priv->session_pool == NULL)
goto no_pool;
/* we had a session in the request, find it again */
if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
goto session_not_found;
/* we add the session to the client list of watched sessions. When a session
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
}
ctx->session = session;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
0, ctx);
done:
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
if (session)
g_object_unref (session);
return;
no_pool:
{
GST_ERROR ("client %p: no pool configured", client);
goto done;
}
session_not_found:
{
GST_ERROR ("client %p: session not found", client);
goto done;
}
}
static void
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
guint8 channel;
guint8 *data;
guint size;
GstBuffer *buffer;
GstRTSPStreamTransport *trans;
/* find the stream for this message */
res = gst_rtsp_message_parse_data (message, &channel);
if (res != GST_RTSP_OK)
return;
gst_rtsp_message_get_body (message, &data, &size);
if (size < 2)
goto invalid_length;
gst_rtsp_message_steal_body (message, &data, &size);
/* Strip trailing \0 (which GstRTSPConnection adds) */
--size;
buffer = gst_buffer_new_wrapped (data, size);
trans =
g_hash_table_lookup (priv->transports, GINT_TO_POINTER ((gint) channel));
if (trans) {
/* dispatch to the stream based on the channel number */
GST_LOG_OBJECT (client, "%u bytes of data on channel %u", size, channel);
gst_rtsp_stream_transport_recv_data (trans, channel, buffer);
} else {
GST_DEBUG_OBJECT (client, "received %u bytes of data for "
"unknown channel %u", size, channel);
gst_buffer_unref (buffer);
}
return;
/* ERRORS */
invalid_length:
{
GST_DEBUG ("client %p: Short message received, ignoring", client);
return;
}
}
/**
* gst_rtsp_client_set_session_pool:
* @client: a #GstRTSPClient
* @pool: (transfer none): a #GstRTSPSessionPool
*
* Set @pool as the sessionpool for @client which it will use to find
* or allocate sessions. the sessionpool is usually inherited from the server
* that created the client but can be overridden later.
*/
void
gst_rtsp_client_set_session_pool (GstRTSPClient * client,
GstRTSPSessionPool * pool)
{
GstRTSPSessionPool *old;
GstRTSPClientPrivate *priv;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (pool)
g_object_ref (pool);
g_mutex_lock (&priv->lock);
old = priv->session_pool;
priv->session_pool = pool;
if (priv->session_removed_id) {
g_signal_handler_disconnect (old, priv->session_removed_id);
priv->session_removed_id = 0;
}
g_mutex_unlock (&priv->lock);
/* FIXME, should remove all sessions from the old pool for this client */
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_session_pool:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
*
* Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
*/
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPSessionPool *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->session_pool))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_mount_points:
* @client: a #GstRTSPClient
* @mounts: (transfer none): a #GstRTSPMountPoints
*
* Set @mounts as the mount points for @client which it will use to map urls
* to media streams. These mount points are usually inherited from the server that
* created the client but can be overriden later.
*/
void
gst_rtsp_client_set_mount_points (GstRTSPClient * client,
GstRTSPMountPoints * mounts)
{
GstRTSPClientPrivate *priv;
GstRTSPMountPoints *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (mounts)
g_object_ref (mounts);
g_mutex_lock (&priv->lock);
old = priv->mount_points;
priv->mount_points = mounts;
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_mount_points:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
*
* Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
*/
GstRTSPMountPoints *
gst_rtsp_client_get_mount_points (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPMountPoints *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->mount_points))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_auth:
* @client: a #GstRTSPClient
* @auth: (transfer none): a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @client.
*/
void
gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
{
GstRTSPClientPrivate *priv;
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (auth)
g_object_ref (auth);
g_mutex_lock (&priv->lock);
old = priv->auth;
priv->auth = auth;
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_auth:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPAuth used as the authentication manager of @client.
*
* Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
* usage.
*/
GstRTSPAuth *
gst_rtsp_client_get_auth (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->auth))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_thread_pool:
* @client: a #GstRTSPClient
* @pool: (transfer none): a #GstRTSPThreadPool
*
* configure @pool to be used as the thread pool of @client.
*/
void
gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
GstRTSPThreadPool * pool)
{
GstRTSPClientPrivate *priv;
GstRTSPThreadPool *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
if (pool)
g_object_ref (pool);
g_mutex_lock (&priv->lock);
old = priv->thread_pool;
priv->thread_pool = pool;
g_mutex_unlock (&priv->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_thread_pool:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPThreadPool used as the thread pool of @client.
*
* Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
* usage.
*/
GstRTSPThreadPool *
gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv;
GstRTSPThreadPool *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
if ((result = priv->thread_pool))
g_object_ref (result);
g_mutex_unlock (&priv->lock);
return result;
}
/**
* gst_rtsp_client_set_connection:
* @client: a #GstRTSPClient
* @conn: (transfer full): a #GstRTSPConnection
*
* Set the #GstRTSPConnection of @client. This function takes ownership of
* @conn.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_client_set_connection (GstRTSPClient * client,
GstRTSPConnection * conn)
{
GstRTSPClientPrivate *priv;
GSocket *read_socket;
GSocketAddress *address;
GstRTSPUrl *url;
GError *error = NULL;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
g_return_val_if_fail (conn != NULL, FALSE);
priv = client->priv;
read_socket = gst_rtsp_connection_get_read_socket (conn);
if (!(address = g_socket_get_local_address (read_socket, &error)))
goto no_address;
g_free (priv->server_ip);
/* keep the original ip that the client connected to */
if (G_IS_INET_SOCKET_ADDRESS (address)) {
GInetAddress *iaddr;
iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
/* socket might be ipv6 but adress still ipv4 */
priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
priv->server_ip = g_inet_address_to_string (iaddr);
g_object_unref (address);
} else {
priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
priv->server_ip = g_strdup ("unknown");
}
GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
priv->server_ip, priv->is_ipv6);
url = gst_rtsp_connection_get_url (conn);
GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
priv->connection = conn;
return TRUE;
/* ERRORS */
no_address:
{
GST_ERROR ("could not get local address %s", error->message);
g_error_free (error);
return FALSE;
}
}
/**
* gst_rtsp_client_get_connection:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPConnection of @client.
*
* Returns: (transfer none): the #GstRTSPConnection of @client.
* The connection object returned remains valid until the client is freed.
*/
GstRTSPConnection *
gst_rtsp_client_get_connection (GstRTSPClient * client)
{
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
return client->priv->connection;
}
/**
* gst_rtsp_client_set_send_func:
* @client: a #GstRTSPClient
* @func: (scope notified): a #GstRTSPClientSendFunc
* @user_data: (closure): user data passed to @func
* @notify: (allow-none): called when @user_data is no longer in use
*
* Set @func as the callback that will be called when a new message needs to be
* sent to the client. @user_data is passed to @func and @notify is called when
* @user_data is no longer in use.
*
* By default, the client will send the messages on the #GstRTSPConnection that
* was configured with gst_rtsp_client_attach() was called.
*/
void
gst_rtsp_client_set_send_func (GstRTSPClient * client,
GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
{
GstRTSPClientPrivate *priv;
GDestroyNotify old_notify;
gpointer old_data;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
priv = client->priv;
g_mutex_lock (&priv->send_lock);
priv->send_func = func;
old_notify = priv->send_notify;
old_data = priv->send_data;
priv->send_notify = notify;
priv->send_data = user_data;
g_mutex_unlock (&priv->send_lock);
if (old_notify)
old_notify (old_data);
}
/**
* gst_rtsp_client_handle_message:
* @client: a #GstRTSPClient
* @message: (transfer none): an #GstRTSPMessage
*
* Let the client handle @message.
*
* Returns: a #GstRTSPResult.
*/
GstRTSPResult
gst_rtsp_client_handle_message (GstRTSPClient * client,
GstRTSPMessage * message)
{
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
switch (message->type) {
case GST_RTSP_MESSAGE_REQUEST:
handle_request (client, message);
break;
case GST_RTSP_MESSAGE_RESPONSE:
handle_response (client, message);
break;
case GST_RTSP_MESSAGE_DATA:
handle_data (client, message);
break;
default:
break;
}
return GST_RTSP_OK;
}
/**
* gst_rtsp_client_send_message:
* @client: a #GstRTSPClient
* @session: (allow-none) (transfer none): a #GstRTSPSession to send
* the message to or %NULL
* @message: (transfer none): The #GstRTSPMessage to send
*
* Send a message message to the remote end. @message must be a
* #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
*/
GstRTSPResult
gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPMessage * message)
{
GstRTSPContext sctx = { NULL }
, *ctx;
GstRTSPClientPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
priv = client->priv;
if (!(ctx = gst_rtsp_context_get_current ())) {
ctx = &sctx;
ctx->auth = priv->auth;
gst_rtsp_context_push_current (ctx);
}
ctx->conn = priv->connection;
ctx->client = client;
ctx->session = session;
send_message (client, ctx, message, FALSE);
if (ctx == &sctx)
gst_rtsp_context_pop_current (ctx);
return GST_RTSP_OK;
}
static GstRTSPResult
do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
gboolean close, gpointer user_data)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult ret;
GTimeVal time;
time.tv_sec = 1;
time.tv_usec = 0;
do {
/* send the response and store the seq number so we can wait until it's
* written to the client to close the connection */
ret =
gst_rtsp_watch_send_message (priv->watch, message,
close ? &priv->close_seq : NULL);
if (ret == GST_RTSP_OK)
break;
if (ret != GST_RTSP_ENOMEM)
goto error;
/* drop backlog */
if (priv->drop_backlog)
break;
/* queue was full, wait for more space */
GST_DEBUG_OBJECT (client, "waiting for backlog");
ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
GST_DEBUG_OBJECT (client, "Resend due to backlog full");
} while (ret != GST_RTSP_EINTR);
return ret;
/* ERRORS */
error:
{
GST_DEBUG_OBJECT (client, "got error %d", ret);
return ret;
}
}
static GstRTSPResult
message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
gpointer user_data)
{
return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
}
static GstRTSPResult
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClientPrivate *priv = client->priv;
if (priv->close_seq && priv->close_seq == cseq) {
GST_INFO ("client %p: send close message", client);
priv->close_seq = 0;
gst_rtsp_client_close (client);
}
return GST_RTSP_OK;
}
static GstRTSPResult
closed (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
GST_INFO ("client %p: connection closed", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
g_mutex_lock (&tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
}
gst_rtsp_watch_set_flushing (watch, TRUE);
g_mutex_lock (&priv->watch_lock);
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
g_mutex_unlock (&priv->watch_lock);
return GST_RTSP_OK;
}
static GstRTSPResult
error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gchar *str;
str = gst_rtsp_strresult (result);
GST_INFO ("client %p: received an error %s", client, str);
g_free (str);
return GST_RTSP_OK;
}
static GstRTSPResult
error_full (GstRTSPWatch * watch, GstRTSPResult result,
GstRTSPMessage * message, guint id, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gchar *str;
str = gst_rtsp_strresult (result);
GST_INFO
("client %p: error when handling message %p with id %d: %s",
client, message, id, str);
g_free (str);
return GST_RTSP_OK;
}
static gboolean
remember_tunnel (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
const gchar *tunnelid;
/* store client in the pending tunnels */
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
/* we can't have two clients connecting with the same tunnelid */
g_mutex_lock (&tunnels_lock);
if (g_hash_table_lookup (tunnels, tunnelid))
goto tunnel_existed;
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
g_mutex_unlock (&tunnels_lock);
return TRUE;
/* ERRORS */
no_tunnelid:
{
GST_ERROR ("client %p: no tunnelid provided", client);
return FALSE;
}
tunnel_existed:
{
g_mutex_unlock (&tunnels_lock);
GST_ERROR ("client %p: tunnel session %s already existed", client,
tunnelid);
return FALSE;
}
}
static GstRTSPResult
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClientPrivate *priv = client->priv;
GST_WARNING ("client %p: tunnel lost (connection %p)", client,
priv->connection);
/* ignore error, it'll only be a problem when the client does a POST again */
remember_tunnel (client);
return GST_RTSP_OK;
}
static gboolean
handle_tunnel (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
GstRTSPClient *oclient;
GstRTSPClientPrivate *opriv;
const gchar *tunnelid;
tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
if (tunnelid == NULL)
goto no_tunnelid;
/* check for previous tunnel */
g_mutex_lock (&tunnels_lock);
oclient = g_hash_table_lookup (tunnels, tunnelid);
if (oclient == NULL) {
/* no previous tunnel, remember tunnel */
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
g_mutex_unlock (&tunnels_lock);
GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
client, priv->connection);
} else {
/* merge both tunnels into the first client */
/* remove the old client from the table. ref before because removing it will
* remove the ref to it. */
g_object_ref (oclient);
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (&tunnels_lock);
opriv = oclient->priv;
g_mutex_lock (&opriv->watch_lock);
if (opriv->watch == NULL)
goto tunnel_closed;
GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
oclient, opriv->connection, priv->connection);
gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
gst_rtsp_watch_reset (priv->watch);
gst_rtsp_watch_reset (opriv->watch);
g_mutex_unlock (&opriv->watch_lock);
g_object_unref (oclient);
/* the old client owns the tunnel now, the new one will be freed */
g_source_destroy ((GSource *) priv->watch);
priv->watch = NULL;
gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
}
return TRUE;
/* ERRORS */
no_tunnelid:
{
GST_ERROR ("client %p: no tunnelid provided", client);
return FALSE;
}
tunnel_closed:
{
GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
g_mutex_unlock (&opriv->watch_lock);
g_object_unref (oclient);
return FALSE;
}
}
static GstRTSPStatusCode
tunnel_get (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GST_INFO ("client %p: tunnel get (connection %p)", client,
client->priv->connection);
if (!handle_tunnel (client)) {
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
return GST_RTSP_STS_OK;
}
static GstRTSPResult
tunnel_post (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GST_INFO ("client %p: tunnel post (connection %p)", client,
client->priv->connection);
if (!handle_tunnel (client)) {
return GST_RTSP_ERROR;
}
return GST_RTSP_OK;
}
static GstRTSPResult
tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
GstRTSPMessage * response, gpointer user_data)
{
GstRTSPClientClass *klass;
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
klass = GST_RTSP_CLIENT_GET_CLASS (client);
if (klass->tunnel_http_response) {
klass->tunnel_http_response (client, request, response);
}
return GST_RTSP_OK;
}
static GstRTSPWatchFuncs watch_funcs = {
message_received,
message_sent,
closed,
error,
tunnel_get,
tunnel_post,
error_full,
tunnel_lost,
tunnel_http_response
};
static void
client_watch_notify (GstRTSPClient * client)
{
GstRTSPClientPrivate *priv = client->priv;
gboolean closed = TRUE;
GST_INFO ("client %p: watch destroyed", client);
priv->watch = NULL;
/* remove all sessions if the media says so and so drop the extra client ref */
gst_rtsp_client_session_filter (client, cleanup_session, &closed);
if (closed)
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
/**
* gst_rtsp_client_attach:
* @client: a #GstRTSPClient
* @context: (allow-none): a #GMainContext
*
* Attaches @client to @context. When the mainloop for @context is run, the
* client will be dispatched. When @context is %NULL, the default context will be
* used).
*
* This function should be called when the client properties and urls are fully
* configured and the client is ready to start.
*
* Returns: the ID (greater than 0) for the source within the GMainContext.
*/
guint
gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
{
GstRTSPClientPrivate *priv;
guint res;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
priv = client->priv;
g_return_val_if_fail (priv->connection != NULL, 0);
g_return_val_if_fail (priv->watch == NULL, 0);
/* make sure noone will free the context before the watch is destroyed */
priv->watch_context = g_main_context_ref (context);
/* create watch for the connection and attach */
priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
g_object_ref (client), (GDestroyNotify) client_watch_notify);
gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
(GDestroyNotify) gst_rtsp_watch_unref);
gst_rtsp_watch_set_send_backlog (priv->watch, 0, WATCH_BACKLOG_SIZE);
GST_INFO ("client %p: attaching to context %p", client, context);
res = gst_rtsp_watch_attach (priv->watch, context);
return res;
}
/**
* gst_rtsp_client_session_filter:
* @client: a #GstRTSPClient
* @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
*
* Call @func for each session managed by @client. The result value of @func
* determines what happens to the session. @func will be called with @client
* locked so no further actions on @client can be performed from @func.
*
* If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
* @client.
*
* If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
*
* If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
* will also be added with an additional ref to the result #GList of this
* function..
*
* When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
*
* Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
* sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
* element in the #GList should be unreffed before the list is freed.
*/
GList *
gst_rtsp_client_session_filter (GstRTSPClient * client,
GstRTSPClientSessionFilterFunc func, gpointer user_data)
{
GstRTSPClientPrivate *priv;
GList *result, *walk, *next;
GHashTable *visited;
guint cookie;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
priv = client->priv;
result = NULL;
if (func)
visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
g_mutex_lock (&priv->lock);
restart:
cookie = priv->sessions_cookie;
for (walk = priv->sessions; walk; walk = next) {
GstRTSPSession *sess = walk->data;
GstRTSPFilterResult res;
gboolean changed;
next = g_list_next (walk);
if (func) {
/* only visit each session once */
if (g_hash_table_contains (visited, sess))
continue;
g_hash_table_add (visited, g_object_ref (sess));
g_mutex_unlock (&priv->lock);
res = func (client, sess, user_data);
g_mutex_lock (&priv->lock);
} else
res = GST_RTSP_FILTER_REF;
changed = (cookie != priv->sessions_cookie);
switch (res) {
case GST_RTSP_FILTER_REMOVE:
/* stop watching the session and pretend it went away, if the list was
* changed, we can't use the current list position, try to see if we
* still have the session */
client_unwatch_session (client, sess, changed ? NULL : walk);
cookie = priv->sessions_cookie;
break;
case GST_RTSP_FILTER_REF:
result = g_list_prepend (result, g_object_ref (sess));
break;
case GST_RTSP_FILTER_KEEP:
default:
break;
}
if (changed)
goto restart;
}
g_mutex_unlock (&priv->lock);
if (func)
g_hash_table_unref (visited);
return result;
}
Reference in a new issue View git blame Copy permalink