mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-01 06:01:04 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtsp-server/rtsp-client.c
Wim Taymans 6fa73b2552 client: use method to access property
2011-08-16 16:07:04 +02:00

1981 lines
51 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <string.h>
#include <sys/time.h>
#include <sys/types.h>
#include <netinet/in.h>
#include <netdb.h>
#include <sys/socket.h>
#include <sys/wait.h>
#include <fcntl.h>
#include <arpa/inet.h>
#include <sys/ioctl.h>
#include "rtsp-client.h"
#include "rtsp-sdp.h"
#include "rtsp-params.h"
static GMutex *tunnels_lock;
static GHashTable *tunnels;
enum
{
PROP_0,
PROP_SESSION_POOL,
PROP_MEDIA_MAPPING,
PROP_LAST
};
enum
{
SIGNAL_CLOSED,
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 void client_session_finalized (GstRTSPClient * client,
GstRTSPSession * session);
static void unlink_session_streams (GstRTSPClient * client,
GstRTSPSession * session, GstRTSPSessionMedia * media);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
static void
gst_rtsp_client_class_init (GstRTSPClientClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_client_get_property;
gobject_class->set_property = gst_rtsp_client_set_property;
gobject_class->finalize = gst_rtsp_client_finalize;
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_MEDIA_MAPPING,
g_param_spec_object ("media-mapping", "Media Mapping",
"The media mapping to use for client session",
GST_TYPE_RTSP_MEDIA_MAPPING,
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_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
tunnels =
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
tunnels_lock = g_mutex_new ();
GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
}
static void
client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
{
GList *medias;
/* unlink all media managed in this session */
for (medias = session->medias; medias; medias = g_list_next (medias)) {
GstRTSPSessionMedia *media = medias->data;
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
unlink_session_streams (client, session, media);
/* unmanage the media in the session. */
gst_rtsp_session_release_media (session, media);
}
}
static void
client_cleanup_sessions (GstRTSPClient * client)
{
GList *sessions;
/* remove weak-ref from sessions */
for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
GstRTSPSession *session = (GstRTSPSession *) sessions->data;
g_object_weak_unref (G_OBJECT (session),
(GWeakNotify) client_session_finalized, client);
client_unlink_session (client, session);
}
g_list_free (client->sessions);
client->sessions = NULL;
}
/* A client is finalized when the connection is broken */
static void
gst_rtsp_client_finalize (GObject * obj)
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
GST_INFO ("finalize client %p", client);
client_cleanup_sessions (client);
gst_rtsp_connection_free (client->connection);
if (client->session_pool)
g_object_unref (client->session_pool);
if (client->media_mapping)
g_object_unref (client->media_mapping);
if (client->auth)
g_object_unref (client->auth);
if (client->uri)
gst_rtsp_url_free (client->uri);
if (client->media)
g_object_unref (client->media);
g_free (client->server_ip);
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
static void
gst_rtsp_client_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
switch (propid) {
case PROP_SESSION_POOL:
g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
break;
case PROP_MEDIA_MAPPING:
g_value_take_object (value, gst_rtsp_client_get_media_mapping (client));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_client_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
switch (propid) {
case PROP_SESSION_POOL:
gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
break;
case PROP_MEDIA_MAPPING:
gst_rtsp_client_set_media_mapping (client, g_value_get_object (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_client_new:
*
* Create a new #GstRTSPClient instance.
*
* Returns: 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_response (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPMessage * response)
{
gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
"GStreamer RTSP server");
/* remove any previous header */
gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
/* add the new session header for new session ids */
if (session) {
gchar *str;
if (session->timeout != 60)
str =
g_strdup_printf ("%s; timeout=%d", session->sessionid,
session->timeout);
else
str = g_strdup (session->sessionid);
gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, str);
}
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (response);
}
gst_rtsp_watch_send_message (client->watch, response, NULL);
gst_rtsp_message_unset (response);
}
static void
send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
GstRTSPClientState * state)
{
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
send_response (client, NULL, state->response);
}
static void
handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
GstRTSPClientState * state)
{
gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
if (auth) {
/* and let the authentication manager setup the auth tokens */
gst_rtsp_auth_setup_auth (auth, client, 0, state);
}
send_response (client, state->session, state->response);
}
static gboolean
compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
{
if (uri1 == NULL || uri2 == NULL)
return FALSE;
if (strcmp (uri1->abspath, uri2->abspath))
return FALSE;
return TRUE;
}
/* this function is called to initially find the media for the DESCRIBE request
* but is cached for when the same client (without breaking the connection) is
* doing a setup for the exact same url. */
static GstRTSPMedia *
find_media (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
GstRTSPAuth *auth;
if (!compare_uri (client->uri, state->uri)) {
/* remove any previously cached values before we try to construct a new
* media for uri */
if (client->uri)
gst_rtsp_url_free (client->uri);
client->uri = NULL;
if (client->media)
g_object_unref (client->media);
client->media = NULL;
if (!client->media_mapping)
goto no_mapping;
/* find the factory for the uri first */
if (!(factory =
gst_rtsp_media_mapping_find_factory (client->media_mapping,
state->uri)))
goto no_factory;
state->factory = factory;
/* check if we have access to the factory */
if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
if (!gst_rtsp_auth_check (auth, client, 0, state))
goto not_allowed;
g_object_unref (auth);
}
/* prepare the media and add it to the pipeline */
if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
goto no_media;
g_object_unref (factory);
factory = NULL;
state->factory = NULL;
/* set ipv6 on the media before preparing */
media->is_ipv6 = client->is_ipv6;
state->media = media;
/* prepare the media */
if (!(gst_rtsp_media_prepare (media)))
goto no_prepare;
/* now keep track of the uri and the media */
client->uri = gst_rtsp_url_copy (state->uri);
client->media = media;
} else {
/* we have seen this uri before, used cached media */
media = client->media;
state->media = media;
GST_INFO ("reusing cached media %p", media);
}
if (media)
g_object_ref (media);
return media;
/* ERRORS */
no_mapping:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
no_factory:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
not_allowed:
{
handle_unauthorized_request (client, auth, state);
g_object_unref (factory);
g_object_unref (auth);
return NULL;
}
no_media:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (factory);
return NULL;
}
no_prepare:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
return NULL;
}
}
static gboolean
do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
GstRTSPMessage message = { 0 };
guint8 *data;
guint usize;
gsize size;
gst_rtsp_message_init_data (&message, channel);
data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
usize = size;
gst_rtsp_message_take_body (&message, data, usize);
/* FIXME, client->watch could have been finalized here, we need to keep an
* extra refcount to the watch. */
gst_rtsp_watch_send_message (client->watch, &message, NULL);
gst_rtsp_message_steal_body (&message, &data, &usize);
gst_buffer_unmap (buffer, data, size);
gst_rtsp_message_unset (&message);
return TRUE;
}
static gboolean
do_send_data_list (GstBufferList * blist, guint8 channel,
GstRTSPClient * client)
{
guint len, i;
len = gst_buffer_list_len (blist);
for (i = 0; i < len; i++) {
GstBuffer *group = gst_buffer_list_get (blist, i);
do_send_data (group, channel, client);
}
return TRUE;
}
static void
link_stream (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPSessionStream * stream)
{
GST_DEBUG ("client %p: linking stream %p", client, stream);
gst_rtsp_session_stream_set_callbacks (stream, (GstRTSPSendFunc) do_send_data,
(GstRTSPSendFunc) do_send_data, (GstRTSPSendListFunc) do_send_data_list,
(GstRTSPSendListFunc) do_send_data_list, client, NULL);
client->streams = g_list_prepend (client->streams, stream);
/* make sure our session can't expire */
gst_rtsp_session_prevent_expire (session);
}
static void
unlink_stream (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPSessionStream * stream)
{
GST_DEBUG ("client %p: unlinking stream %p", client, stream);
gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL, NULL,
NULL);
client->streams = g_list_remove (client->streams, stream);
/* our session can now expire */
gst_rtsp_session_allow_expire (session);
}
static void
unlink_session_streams (GstRTSPClient * client, GstRTSPSession * session,
GstRTSPSessionMedia * media)
{
guint n_streams, i;
n_streams = gst_rtsp_media_n_streams (media->media);
for (i = 0; i < n_streams; i++) {
GstRTSPSessionStream *sstream;
GstRTSPTransport *tr;
/* get the stream as configured in the session */
sstream = gst_rtsp_session_media_get_stream (media, i);
/* get the transport, if there is no transport configured, skip this stream */
if (!(tr = sstream->trans.transport))
continue;
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, unlink the stream from the TCP connection of the client */
unlink_stream (client, session, sstream);
}
}
}
static void
close_connection (GstRTSPClient * client)
{
const gchar *tunnelid;
GST_DEBUG ("client %p: closing connection", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
g_mutex_lock (tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (tunnels_lock);
}
gst_rtsp_connection_close (client->connection);
if (client->watchid) {
g_source_destroy ((GSource *) client->watch);
client->watchid = 0;
client->watch = NULL;
}
}
static gboolean
handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPSession *session;
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
if (!state->session)
goto no_session;
session = state->session;
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
state->sessmedia = media;
/* unlink the all TCP callbacks */
unlink_session_streams (client, session, media);
/* remove the session from the watched sessions */
g_object_weak_unref (G_OBJECT (session),
(GWeakNotify) client_session_finalized, client);
client->sessions = g_list_remove (client->sessions, session);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
/* unmanage the media in the session, returns false if all media session
* are torn down. */
if (!gst_rtsp_session_release_media (session, media)) {
/* remove the session */
gst_rtsp_session_pool_remove (client->session_pool, session);
}
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
"close");
send_response (client, session, state->response);
close_connection (client);
return TRUE;
/* ERRORS */
no_session:
{
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
}
static gboolean
handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
guint8 *data;
guint size;
res = gst_rtsp_message_get_body (state->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0) {
/* no body, keep-alive request */
send_generic_response (client, GST_RTSP_STS_OK, state);
} else {
/* there is a body, handle the params */
res = gst_rtsp_params_get (client, state);
if (res != GST_RTSP_OK)
goto bad_request;
send_response (client, state->session, state->response);
}
return TRUE;
/* ERRORS */
bad_request:
{
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
}
static gboolean
handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
guint8 *data;
guint size;
res = gst_rtsp_message_get_body (state->request, &data, &size);
if (res != GST_RTSP_OK)
goto bad_request;
if (size == 0) {
/* no body, keep-alive request */
send_generic_response (client, GST_RTSP_STS_OK, state);
} else {
/* there is a body, handle the params */
res = gst_rtsp_params_set (client, state);
if (res != GST_RTSP_OK)
goto bad_request;
send_response (client, state->session, state->response);
}
return TRUE;
/* ERRORS */
bad_request:
{
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
}
static gboolean
handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPSession *session;
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
if (!(session = state->session))
goto no_session;
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
state->sessmedia = media;
/* the session state must be playing or recording */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != GST_RTSP_STATE_RECORDING)
goto invalid_state;
/* unlink the all TCP callbacks */
unlink_session_streams (client, session, media);
/* then pause sending */
gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
send_response (client, session, state->response);
/* the state is now READY */
media->state = GST_RTSP_STATE_READY;
return TRUE;
/* ERRORS */
no_session:
{
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
state);
return FALSE;
}
}
static gboolean
handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPSession *session;
GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
GString *rtpinfo;
guint n_streams, i, infocount;
guint timestamp, seqnum;
gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
if (!(session = state->session))
goto no_session;
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
state->sessmedia = media;
/* the session state must be playing or ready */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != GST_RTSP_STATE_READY)
goto invalid_state;
/* parse the range header if we have one */
res =
gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
if (res == GST_RTSP_OK) {
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
/* we have a range, seek to the position */
gst_rtsp_media_seek (media->media, range);
gst_rtsp_range_free (range);
}
}
/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
n_streams = gst_rtsp_media_n_streams (media->media);
for (i = 0, infocount = 0; i < n_streams; i++) {
GstRTSPSessionStream *sstream;
GstRTSPMediaStream *stream;
GstRTSPTransport *tr;
GObjectClass *payobjclass;
gchar *uristr;
/* get the stream as configured in the session */
sstream = gst_rtsp_session_media_get_stream (media, i);
/* get the transport, if there is no transport configured, skip this stream */
if (!(tr = sstream->trans.transport)) {
GST_INFO ("stream %d is not configured", i);
continue;
}
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, link the stream to the TCP connection of the client */
link_stream (client, session, sstream);
}
stream = sstream->media_stream;
payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
if (g_object_class_find_property (payobjclass, "seqnum") &&
g_object_class_find_property (payobjclass, "timestamp")) {
GObject *payobj;
payobj = G_OBJECT (stream->payloader);
/* only add RTP-Info for streams with seqnum and timestamp */
g_object_get (payobj, "seqnum", &seqnum, "timestamp", &timestamp, NULL);
if (infocount > 0)
g_string_append (rtpinfo, ", ");
uristr = gst_rtsp_url_get_request_uri (state->uri);
g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
uristr, i, seqnum, timestamp);
g_free (uristr);
infocount++;
} else {
GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
}
}
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
/* add the RTP-Info header */
if (infocount > 0) {
str = g_string_free (rtpinfo, FALSE);
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
} else {
g_string_free (rtpinfo, TRUE);
}
/* add the range */
str = gst_rtsp_media_get_range_string (media->media, TRUE);
gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
send_response (client, session, state->response);
/* start playing after sending the request */
gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
media->state = GST_RTSP_STATE_PLAYING;
return TRUE;
/* ERRORS */
no_session:
{
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
state);
return FALSE;
}
}
static void
do_keepalive (GstRTSPSession * session)
{
GST_INFO ("keep session %p alive", session);
gst_rtsp_session_touch (session);
}
static gboolean
handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
GstRTSPUrl *uri;
gchar *transport;
gchar **transports;
gboolean have_transport;
GstRTSPTransport *ct, *st;
gint i;
GstRTSPLowerTrans supported;
GstRTSPStatusCode code;
GstRTSPSession *session;
GstRTSPSessionStream *stream;
gchar *trans_str, *pos;
guint streamid;
GstRTSPSessionMedia *media;
uri = state->uri;
/* the uri contains the stream number we added in the SDP config, which is
* always /stream=%d so we need to strip that off
* parse the stream we need to configure, look for the stream in the abspath
* first and then in the query. */
if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
goto bad_request;
}
/* we can mofify the parse uri in place */
*pos = '\0';
pos += strlen ("/stream=");
if (sscanf (pos, "%u", &streamid) != 1)
goto bad_request;
/* parse the transport */
res =
gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
&transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
transports = g_strsplit (transport, ",", 0);
gst_rtsp_transport_new (&ct);
/* init transports */
have_transport = FALSE;
gst_rtsp_transport_init (ct);
/* our supported transports */
supported = GST_RTSP_LOWER_TRANS_UDP |
GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
/* loop through the transports, try to parse */
for (i = 0; transports[i]; i++) {
res = gst_rtsp_transport_parse (transports[i], ct);
if (res != GST_RTSP_OK) {
/* no valid transport, search some more */
GST_WARNING ("could not parse transport %s", transports[i]);
goto next;
}
/* we have a transport, see if it's RTP/AVP */
if (ct->trans != GST_RTSP_TRANS_RTP || ct->profile != GST_RTSP_PROFILE_AVP) {
GST_WARNING ("invalid transport %s", transports[i]);
goto next;
}
if (!(ct->lower_transport & supported)) {
GST_WARNING ("unsupported transport %s", transports[i]);
goto next;
}
/* we have a valid transport */
GST_INFO ("found valid transport %s", transports[i]);
have_transport = TRUE;
break;
next:
gst_rtsp_transport_init (ct);
}
g_strfreev (transports);
/* we have not found anything usable, error out */
if (!have_transport)
goto unsupported_transports;
if (client->session_pool == NULL)
goto no_pool;
session = state->session;
if (session) {
g_object_ref (session);
/* get a handle to the configuration of the media in the session, this can
* return NULL if this is a new url to manage in this session. */
media = gst_rtsp_session_get_media (session, uri);
} else {
/* create a session if this fails we probably reached our session limit or
* something. */
if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
goto service_unavailable;
state->session = session;
/* we need a new media configuration in this session */
media = NULL;
}
/* we have no media, find one and manage it */
if (media == NULL) {
GstRTSPMedia *m;
/* get a handle to the configuration of the media in the session */
if ((m = find_media (client, state))) {
/* manage the media in our session now */
media = gst_rtsp_session_manage_media (session, uri, m);
}
}
/* if we stil have no media, error */
if (media == NULL)
goto not_found;
state->sessmedia = media;
/* we have a valid transport now, set the destination of the client. */
g_free (ct->destination);
if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
ct->destination = gst_rtsp_media_get_multicast_group (media->media);
} else {
GstRTSPUrl *url;
url = gst_rtsp_connection_get_url (client->connection);
ct->destination = g_strdup (url->host);
if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
/* check if the client selected channels for TCP */
if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
gst_rtsp_session_media_alloc_channels (media, &ct->interleaved);
}
}
}
/* get a handle to the stream in the media */
if (!(stream = gst_rtsp_session_media_get_stream (media, streamid)))
goto no_stream;
st = gst_rtsp_session_stream_set_transport (stream, ct);
/* configure keepalive for this transport */
gst_rtsp_session_stream_set_keepalive (stream,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
/* serialize the server transport */
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (state->response, code,
gst_rtsp_status_as_text (code), state->request);
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
trans_str);
g_free (trans_str);
send_response (client, session, state->response);
/* update the state */
switch (media->state) {
case GST_RTSP_STATE_PLAYING:
case GST_RTSP_STATE_RECORDING:
case GST_RTSP_STATE_READY:
/* no state change */
break;
default:
media->state = GST_RTSP_STATE_READY;
break;
}
g_object_unref (session);
return TRUE;
/* ERRORS */
bad_request:
{
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
not_found:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
return FALSE;
}
no_stream:
{
send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (media);
g_object_unref (session);
return FALSE;
}
no_transport:
{
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
return FALSE;
}
unsupported_transports:
{
send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
gst_rtsp_transport_free (ct);
return FALSE;
}
no_pool:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
return FALSE;
}
service_unavailable:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
return FALSE;
}
}
static GstSDPMessage *
create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
{
GstSDPMessage *sdp;
GstSDPInfo info;
const gchar *proto;
GstRTSPLowerTrans protocols;
gst_sdp_message_new (&sdp);
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
if (client->is_ipv6)
proto = "IP6";
else
proto = "IP4";
gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
client->server_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtsp-server");
gst_sdp_message_add_time (sdp, "0", "0", NULL);
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
gst_sdp_message_add_attribute (sdp, "type", "broadcast");
gst_sdp_message_add_attribute (sdp, "control", "*");
info.server_proto = proto;
protocols = gst_rtsp_media_get_protocols (media);
if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
info.server_ip = gst_rtsp_media_get_multicast_group (media);
else
info.server_ip = g_strdup (client->server_ip);
/* create an SDP for the media object */
if (!gst_rtsp_sdp_from_media (sdp, &info, media))
goto no_sdp;
g_free (info.server_ip);
return sdp;
/* ERRORS */
no_sdp:
{
g_free (info.server_ip);
gst_sdp_message_free (sdp);
return NULL;
}
}
/* for the describe we must generate an SDP */
static gboolean
handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
GstSDPMessage *sdp;
guint i, str_len;
gchar *str, *content_base;
GstRTSPMedia *media;
/* check what kind of format is accepted, we don't really do anything with it
* and always return SDP for now. */
for (i = 0; i++;) {
gchar *accept;
res =
gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
&accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
break;
}
/* find the media object for the uri */
if (!(media = find_media (client, state)))
goto no_media;
/* create an SDP for the media object on this client */
if (!(sdp = create_sdp (client, media)))
goto no_sdp;
g_object_unref (media);
gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* content base for some clients that might screw up creating the setup uri */
str = gst_rtsp_url_get_request_uri (state->uri);
str_len = strlen (str);
/* check for trailing '/' and append one */
if (str[str_len - 1] != '/') {
content_base = g_malloc (str_len + 2);
memcpy (content_base, str, str_len);
content_base[str_len] = '/';
content_base[str_len + 1] = '\0';
g_free (str);
} else {
content_base = str;
}
GST_INFO ("adding content-base: %s", content_base);
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
content_base);
g_free (content_base);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
send_response (client, state->session, state->response);
return TRUE;
/* ERRORS */
no_media:
{
/* error reply is already sent */
return FALSE;
}
no_sdp:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
return FALSE;
}
}
static gboolean
handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPMethod options;
gchar *str;
options = GST_RTSP_DESCRIBE |
GST_RTSP_OPTIONS |
GST_RTSP_PAUSE |
GST_RTSP_PLAY |
GST_RTSP_SETUP |
GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
str = gst_rtsp_options_as_text (options);
gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
send_response (client, state->session, state->response);
return TRUE;
}
/* remove duplicate and trailing '/' */
static void
sanitize_uri (GstRTSPUrl * uri)
{
gint i, len;
gchar *s, *d;
gboolean have_slash, prev_slash;
s = d = uri->abspath;
len = strlen (uri->abspath);
prev_slash = FALSE;
for (i = 0; i < len; i++) {
have_slash = s[i] == '/';
*d = s[i];
if (!have_slash || !prev_slash)
d++;
prev_slash = have_slash;
}
len = d - uri->abspath;
/* don't remove the first slash if that's the only thing left */
if (len > 1 && *(d - 1) == '/')
d--;
*d = '\0';
}
static void
client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
{
GST_INFO ("client %p: session %p finished", client, session);
/* unlink all media managed in this session */
client_unlink_session (client, session);
/* remove the session */
if (!(client->sessions = g_list_remove (client->sessions, session))) {
GST_INFO ("client %p: all sessions finalized, close the connection",
client);
close_connection (client);
}
}
static void
client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
{
GList *walk;
for (walk = client->sessions; walk; walk = g_list_next (walk)) {
GstRTSPSession *msession = (GstRTSPSession *) walk->data;
/* we already know about this session */
if (msession == session)
return;
}
GST_INFO ("watching session %p", session);
g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
client);
client->sessions = g_list_prepend (client->sessions, session);
}
static void
handle_request (GstRTSPClient * client, GstRTSPMessage * request)
{
GstRTSPMethod method;
const gchar *uristr;
GstRTSPUrl *uri;
GstRTSPVersion version;
GstRTSPResult res;
GstRTSPSession *session;
GstRTSPClientState state = { NULL };
GstRTSPMessage response = { 0 };
gchar *sessid;
state.request = request;
state.response = &response;
if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
gst_rtsp_message_dump (request);
}
GST_INFO ("client %p: received a request", client);
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
if (version != GST_RTSP_VERSION_1_0) {
/* we can only handle 1.0 requests */
send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
&state);
return;
}
state.method = method;
/* we always try to parse the url first */
if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
return;
}
/* sanitize the uri */
sanitize_uri (uri);
state.uri = uri;
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
if (client->session_pool == NULL)
goto no_pool;
/* we had a session in the request, find it again */
if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
goto session_not_found;
/* we add the session to the client list of watched sessions. When a session
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
} else
session = NULL;
state.session = session;
if (client->auth) {
if (!gst_rtsp_auth_check (client->auth, client, &state))
goto not_authorized;
}
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
handle_options_request (client, &state);
break;
case GST_RTSP_DESCRIBE:
handle_describe_request (client, &state);
break;
case GST_RTSP_SETUP:
handle_setup_request (client, &state);
break;
case GST_RTSP_PLAY:
handle_play_request (client, &state);
break;
case GST_RTSP_PAUSE:
handle_pause_request (client, &state);
break;
case GST_RTSP_TEARDOWN:
handle_teardown_request (client, &state);
break;
case GST_RTSP_SET_PARAMETER:
handle_set_param_request (client, &state);
break;
case GST_RTSP_GET_PARAMETER:
handle_get_param_request (client, &state);
break;
case GST_RTSP_ANNOUNCE:
case GST_RTSP_RECORD:
case GST_RTSP_REDIRECT:
send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
break;
case GST_RTSP_INVALID:
default:
send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
break;
}
if (session)
g_object_unref (session);
gst_rtsp_url_free (uri);
return;
/* ERRORS */
no_pool:
{
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
return;
}
session_not_found:
{
send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
return;
}
not_authorized:
{
handle_unauthorized_request (client, client->auth, &state);
return;
}
}
static void
handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
GstRTSPResult res;
guint8 channel;
GList *walk;
guint8 *data;
guint size;
GstBuffer *buffer;
gboolean handled;
/* find the stream for this message */
res = gst_rtsp_message_parse_data (message, &channel);
if (res != GST_RTSP_OK)
return;
gst_rtsp_message_steal_body (message, &data, &size);
buffer = gst_buffer_new ();
gst_buffer_take_memory (buffer, -1,
gst_memory_new_wrapped (0, data, g_free, size, 0, size));
handled = FALSE;
for (walk = client->streams; walk; walk = g_list_next (walk)) {
GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
GstRTSPMediaStream *mstream;
GstRTSPTransport *tr;
/* get the transport, if there is no transport configured, skip this stream */
if (!(tr = stream->trans.transport))
continue;
/* we also need a media stream */
if (!(mstream = stream->media_stream))
continue;
/* check for TCP transport */
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* dispatch to the stream based on the channel number */
if (tr->interleaved.min == channel) {
gst_rtsp_media_stream_rtp (mstream, buffer);
handled = TRUE;
break;
} else if (tr->interleaved.max == channel) {
gst_rtsp_media_stream_rtcp (mstream, buffer);
handled = TRUE;
break;
}
}
}
if (!handled)
gst_buffer_unref (buffer);
}
/**
* gst_rtsp_client_set_session_pool:
* @client: a #GstRTSPClient
* @pool: a #GstRTSPSessionPool
*
* Set @pool as the sessionpool for @client which it will use to find
* or allocate sessions. the sessionpool is usually inherited from the server
* that created the client but can be overridden later.
*/
void
gst_rtsp_client_set_session_pool (GstRTSPClient * client,
GstRTSPSessionPool * pool)
{
GstRTSPSessionPool *old;
old = client->session_pool;
if (old != pool) {
if (pool)
g_object_ref (pool);
client->session_pool = pool;
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: a #GstRTSPSessionPool, unref after usage.
*/
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient * client)
{
GstRTSPSessionPool *result;
if ((result = client->session_pool))
g_object_ref (result);
return result;
}
/**
* gst_rtsp_client_set_server:
* @client: a #GstRTSPClient
* @server: a #GstRTSPServer
*
* Set @server as the server that created @client.
*/
void
gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
{
GstRTSPServer *old;
old = client->server;
if (old != server) {
if (server)
g_object_ref (server);
client->server = server;
if (old)
g_object_unref (old);
}
}
/**
* gst_rtsp_client_get_server:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPServer object that @client was created from.
*
* Returns: a #GstRTSPServer, unref after usage.
*/
GstRTSPServer *
gst_rtsp_client_get_server (GstRTSPClient * client)
{
GstRTSPServer *result;
if ((result = client->server))
g_object_ref (result);
return result;
}
/**
* gst_rtsp_client_set_media_mapping:
* @client: a #GstRTSPClient
* @mapping: a #GstRTSPMediaMapping
*
* Set @mapping as the media mapping for @client which it will use to map urls
* to media streams. These mapping is usually inherited from the server that
* created the client but can be overriden later.
*/
void
gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
GstRTSPMediaMapping * mapping)
{
GstRTSPMediaMapping *old;
old = client->media_mapping;
if (old != mapping) {
if (mapping)
g_object_ref (mapping);
client->media_mapping = mapping;
if (old)
g_object_unref (old);
}
}
/**
* gst_rtsp_client_get_media_mapping:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
*
* Returns: a #GstRTSPMediaMapping, unref after usage.
*/
GstRTSPMediaMapping *
gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
{
GstRTSPMediaMapping *result;
if ((result = client->media_mapping))
g_object_ref (result);
return result;
}
/**
* gst_rtsp_client_set_auth:
* @client: a #GstRTSPClient
* @auth: a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @client.
*/
void
gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
{
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_CLIENT (client));
old = client->auth;
if (old != auth) {
if (auth)
g_object_ref (auth);
client->auth = auth;
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: the #GstRTSPAuth of @client. g_object_unref() after
* usage.
*/
GstRTSPAuth *
gst_rtsp_client_get_auth (GstRTSPClient * client)
{
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
if ((result = client->auth))
g_object_ref (result);
return result;
}
static GstRTSPResult
message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
switch (message->type) {
case GST_RTSP_MESSAGE_REQUEST:
handle_request (client, message);
break;
case GST_RTSP_MESSAGE_RESPONSE:
break;
case GST_RTSP_MESSAGE_DATA:
handle_data (client, message);
break;
default:
break;
}
return GST_RTSP_OK;
}
static GstRTSPResult
message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
/* GstRTSPClient *client; */
/* client = GST_RTSP_CLIENT (user_data); */
/* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
return GST_RTSP_OK;
}
static GstRTSPResult
closed (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
const gchar *tunnelid;
GST_INFO ("client %p: connection closed", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
g_mutex_lock (tunnels_lock);
/* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
g_mutex_unlock (tunnels_lock);
}
return GST_RTSP_OK;
}
static GstRTSPResult
error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gchar *str;
str = gst_rtsp_strresult (result);
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: received an error %s when handling message %p with id %d",
client, str, message, id);
g_free (str);
return GST_RTSP_OK;
}
static gboolean
remember_tunnel (GstRTSPClient * client)
{
const gchar *tunnelid;
/* store client in the pending tunnels */
tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
if (tunnelid == NULL)
goto no_tunnelid;
GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
/* we can't have two clients connecting with the same tunnelid */
g_mutex_lock (tunnels_lock);
if (g_hash_table_lookup (tunnels, tunnelid))
goto tunnel_existed;
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
g_mutex_unlock (tunnels_lock);
return TRUE;
/* ERRORS */
no_tunnelid:
{
GST_ERROR ("client %p: no tunnelid provided", client);
return FALSE;
}
tunnel_existed:
{
g_mutex_unlock (tunnels_lock);
GST_ERROR ("client %p: tunnel session %s already existed", client,
tunnelid);
return FALSE;
}
}
static GstRTSPStatusCode
tunnel_start (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client;
client = GST_RTSP_CLIENT (user_data);
GST_INFO ("client %p: tunnel start (connection %p)", client,
client->connection);
if (!remember_tunnel (client))
goto tunnel_error;
return GST_RTSP_STS_OK;
/* ERRORS */
tunnel_error:
{
GST_ERROR ("client %p: error starting tunnel", client);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
}
static GstRTSPResult
tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client;
client = GST_RTSP_CLIENT (user_data);
GST_INFO ("client %p: tunnel lost (connection %p)", client,
client->connection);
/* ignore error, it'll only be a problem when the client does a POST again */
remember_tunnel (client);
return GST_RTSP_OK;
}
static GstRTSPResult
tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
{
const gchar *tunnelid;
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClient *oclient;
GST_INFO ("client %p: tunnel complete", client);
/* find previous tunnel */
tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
if (tunnelid == NULL)
goto no_tunnelid;
g_mutex_lock (tunnels_lock);
if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
goto no_tunnel;
/* remove the old client from the table. ref before because removing it will
* remove the ref to it. */
g_object_ref (oclient);
g_hash_table_remove (tunnels, tunnelid);
if (oclient->watch == NULL)
goto tunnel_closed;
g_mutex_unlock (tunnels_lock);
GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
oclient->connection, client->connection);
/* merge the tunnels into the first client */
gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
gst_rtsp_watch_reset (oclient->watch);
g_object_unref (oclient);
/* we don't need this watch anymore */
g_source_destroy ((GSource *) client->watch);
client->watchid = 0;
client->watch = NULL;
return GST_RTSP_OK;
/* ERRORS */
no_tunnelid:
{
GST_INFO ("client %p: no tunnelid provided", client);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
no_tunnel:
{
g_mutex_unlock (tunnels_lock);
GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
tunnel_closed:
{
g_mutex_unlock (tunnels_lock);
GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
g_object_unref (oclient);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
}
static GstRTSPWatchFuncs watch_funcs = {
message_received,
message_sent,
closed,
error,
tunnel_start,
tunnel_complete,
error_full,
tunnel_lost
};
static void
client_watch_notify (GstRTSPClient * client)
{
GST_INFO ("client %p: watch destroyed", client);
client->watchid = 0;
client->watch = NULL;
g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
g_object_unref (client);
}
/**
* gst_rtsp_client_attach:
* @client: a #GstRTSPClient
* @channel: a #GIOChannel
*
* Accept a new connection for @client on the socket in @channel.
*
* This function should be called when the client properties and urls are fully
* configured and the client is ready to start.
*
* Returns: %TRUE if the client could be accepted.
*/
gboolean
gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel)
{
int sock, fd;
GstRTSPConnection *conn;
GstRTSPResult res;
GSource *source;
GMainContext *context;
GstRTSPUrl *url;
struct sockaddr_storage addr;
socklen_t addrlen;
gchar ip[INET6_ADDRSTRLEN];
/* a new client connected. */
sock = g_io_channel_unix_get_fd (channel);
GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
fd = gst_rtsp_connection_get_readfd (conn);
addrlen = sizeof (addr);
if (getsockname (fd, (struct sockaddr *) &addr, &addrlen) < 0)
goto getpeername_failed;
client->is_ipv6 = addr.ss_family == AF_INET6;
if (getnameinfo ((struct sockaddr *) &addr, addrlen, ip, sizeof (ip), NULL, 0,
NI_NUMERICHOST) != 0)
goto getnameinfo_failed;
/* keep the original ip that the client connected to */
g_free (client->server_ip);
client->server_ip = g_strndup (ip, sizeof (ip));
GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
client->server_ip, client->is_ipv6);
url = gst_rtsp_connection_get_url (conn);
GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
client->connection = conn;
/* create watch for the connection and attach */
client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
g_object_ref (client), (GDestroyNotify) client_watch_notify);
/* find the context to add the watch */
if ((source = g_main_current_source ()))
context = g_source_get_context (source);
else
context = NULL;
GST_INFO ("attaching to context %p", context);
client->watchid = gst_rtsp_watch_attach (client->watch, context);
gst_rtsp_watch_unref (client->watch);
return TRUE;
/* ERRORS */
accept_failed:
{
gchar *str = gst_rtsp_strresult (res);
GST_ERROR ("Could not accept client on server socket %d: %s", sock, str);
g_free (str);
return FALSE;
}
getpeername_failed:
{
GST_ERROR ("getpeername failed: %s", g_strerror (errno));
return FALSE;
}
getnameinfo_failed:
{
GST_ERROR ("getnameinfo failed: %s", g_strerror (errno));
return FALSE;
}
}
Reference in a new issue View git blame Copy permalink