gstreamer/gst/rtsp-server/rtsp-session.c
Wim Taymans 50b4c8de98 rtsp-server: add support for buffer lists
Add support for sending bufferlists received from appsink.

Fixes #635832
2010-12-29 16:26:41 +01:00

617 lines
16 KiB
C

/* 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 <string.h>
#include "rtsp-session.h"
#undef DEBUG
#define DEFAULT_TIMEOUT 60
enum
{
PROP_0,
PROP_SESSIONID,
PROP_TIMEOUT,
PROP_LAST
};
GST_DEBUG_CATEGORY_EXTERN (rtsp_session_debug);
#define GST_CAT_DEFAULT rtsp_session_debug
static void gst_rtsp_session_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_session_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_session_finalize (GObject * obj);
G_DEFINE_TYPE (GstRTSPSession, gst_rtsp_session, G_TYPE_OBJECT);
static void
gst_rtsp_session_class_init (GstRTSPSessionClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_session_get_property;
gobject_class->set_property = gst_rtsp_session_set_property;
gobject_class->finalize = gst_rtsp_session_finalize;
g_object_class_install_property (gobject_class, PROP_SESSIONID,
g_param_spec_string ("sessionid", "Sessionid", "the session id",
NULL, G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TIMEOUT,
g_param_spec_uint ("timeout", "timeout",
"the timeout of the session (0 = never)", 0, G_MAXUINT,
DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_rtsp_session_init (GstRTSPSession * session)
{
session->timeout = DEFAULT_TIMEOUT;
g_get_current_time (&session->create_time);
gst_rtsp_session_touch (session);
}
static void
gst_rtsp_session_free_stream (GstRTSPSessionStream * stream)
{
GST_INFO ("free session stream %p", stream);
/* remove callbacks now */
gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL, NULL,
NULL);
gst_rtsp_session_stream_set_keepalive (stream, NULL, NULL, NULL);
gst_rtsp_media_trans_cleanup (&stream->trans);
g_free (stream);
}
static void
gst_rtsp_session_free_media (GstRTSPSessionMedia * media,
GstRTSPSession * session)
{
guint size, i;
size = media->streams->len;
GST_INFO ("free session media %p", media);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
for (i = 0; i < size; i++) {
GstRTSPSessionStream *stream;
stream = g_array_index (media->streams, GstRTSPSessionStream *, i);
if (stream)
gst_rtsp_session_free_stream (stream);
}
g_array_free (media->streams, TRUE);
if (media->url)
gst_rtsp_url_free (media->url);
if (media->media)
g_object_unref (media->media);
g_free (media);
}
static void
gst_rtsp_session_finalize (GObject * obj)
{
GstRTSPSession *session;
session = GST_RTSP_SESSION (obj);
GST_INFO ("finalize session %p", session);
/* free all media */
g_list_foreach (session->medias, (GFunc) gst_rtsp_session_free_media,
session);
g_list_free (session->medias);
/* free session id */
g_free (session->sessionid);
G_OBJECT_CLASS (gst_rtsp_session_parent_class)->finalize (obj);
}
static void
gst_rtsp_session_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPSession *session = GST_RTSP_SESSION (object);
switch (propid) {
case PROP_SESSIONID:
g_value_set_string (value, session->sessionid);
break;
case PROP_TIMEOUT:
g_value_set_uint (value, gst_rtsp_session_get_timeout (session));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_session_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPSession *session = GST_RTSP_SESSION (object);
switch (propid) {
case PROP_SESSIONID:
g_free (session->sessionid);
session->sessionid = g_value_dup_string (value);
break;
case PROP_TIMEOUT:
gst_rtsp_session_set_timeout (session, g_value_get_uint (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_session_manage_media:
* @sess: a #GstRTSPSession
* @uri: the uri for the media
* @media: a #GstRTSPMedia
*
* Manage the media object @obj in @sess. @uri will be used to retrieve this
* media from the session with gst_rtsp_session_get_media().
*
* Ownership is taken from @media.
*
* Returns: a new @GstRTSPSessionMedia object.
*/
GstRTSPSessionMedia *
gst_rtsp_session_manage_media (GstRTSPSession * sess, const GstRTSPUrl * uri,
GstRTSPMedia * media)
{
GstRTSPSessionMedia *result;
guint n_streams;
g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
g_return_val_if_fail (uri != NULL, NULL);
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_return_val_if_fail (media->status == GST_RTSP_MEDIA_STATUS_PREPARED, NULL);
result = g_new0 (GstRTSPSessionMedia, 1);
result->media = media;
result->url = gst_rtsp_url_copy ((GstRTSPUrl *) uri);
result->state = GST_RTSP_STATE_INIT;
/* prealloc the streams now, filled with NULL */
n_streams = gst_rtsp_media_n_streams (media);
result->streams =
g_array_sized_new (FALSE, TRUE, sizeof (GstRTSPSessionStream *),
n_streams);
g_array_set_size (result->streams, n_streams);
sess->medias = g_list_prepend (sess->medias, result);
GST_INFO ("manage new media %p in session %p", media, result);
return result;
}
/**
* gst_rtsp_session_release_media:
* @sess: a #GstRTSPSession
* @media: a #GstRTSPMedia
*
* Release the managed @media in @sess, freeing the memory allocated by it.
*
* Returns: %TRUE if there are more media session left in @sess.
*/
gboolean
gst_rtsp_session_release_media (GstRTSPSession * sess,
GstRTSPSessionMedia * media)
{
GList *walk, *next;
g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
g_return_val_if_fail (media != NULL, FALSE);
for (walk = sess->medias; walk;) {
GstRTSPSessionMedia *find;
find = (GstRTSPSessionMedia *) walk->data;
next = g_list_next (walk);
if (find == media) {
sess->medias = g_list_delete_link (sess->medias, walk);
gst_rtsp_session_free_media (find, sess);
break;
}
walk = next;
}
return (sess->medias != NULL);
}
/**
* gst_rtsp_session_get_media:
* @sess: a #GstRTSPSession
* @url: the url for the media
*
* Get the session media of the @url.
*
* Returns: the configuration for @url in @sess.
*/
GstRTSPSessionMedia *
gst_rtsp_session_get_media (GstRTSPSession * sess, const GstRTSPUrl * url)
{
GstRTSPSessionMedia *result;
GList *walk;
g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
g_return_val_if_fail (url != NULL, NULL);
result = NULL;
for (walk = sess->medias; walk; walk = g_list_next (walk)) {
result = (GstRTSPSessionMedia *) walk->data;
if (strcmp (result->url->abspath, url->abspath) == 0)
break;
result = NULL;
}
return result;
}
/**
* gst_rtsp_session_media_get_stream:
* @media: a #GstRTSPSessionMedia
* @idx: the stream index
*
* Get a previously created or create a new #GstRTSPSessionStream at @idx.
*
* Returns: a #GstRTSPSessionStream that is valid until the session of @media
* is unreffed.
*/
GstRTSPSessionStream *
gst_rtsp_session_media_get_stream (GstRTSPSessionMedia * media, guint idx)
{
GstRTSPSessionStream *result;
g_return_val_if_fail (media != NULL, NULL);
g_return_val_if_fail (media->media != NULL, NULL);
if (idx >= media->streams->len)
return NULL;
result = g_array_index (media->streams, GstRTSPSessionStream *, idx);
if (result == NULL) {
GstRTSPMediaStream *media_stream;
media_stream = gst_rtsp_media_get_stream (media->media, idx);
if (media_stream == NULL)
goto no_media;
result = g_new0 (GstRTSPSessionStream, 1);
result->trans.idx = idx;
result->trans.transport = NULL;
result->media_stream = media_stream;
g_array_index (media->streams, GstRTSPSessionStream *, idx) = result;
}
return result;
/* ERRORS */
no_media:
{
return NULL;
}
}
gboolean
gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
GstRTSPRange * range)
{
range->min = media->counter++;
range->max = media->counter++;
return TRUE;
}
/**
* gst_rtsp_session_new:
*
* Create a new #GstRTSPSession instance.
*/
GstRTSPSession *
gst_rtsp_session_new (const gchar * sessionid)
{
GstRTSPSession *result;
g_return_val_if_fail (sessionid != NULL, NULL);
result = g_object_new (GST_TYPE_RTSP_SESSION, "sessionid", sessionid, NULL);
return result;
}
/**
* gst_rtsp_session_get_sessionid:
* @session: a #GstRTSPSession
*
* Get the sessionid of @session.
*
* Returns: the sessionid of @session. The value remains valid as long as
* @session is alive.
*/
const gchar *
gst_rtsp_session_get_sessionid (GstRTSPSession * session)
{
g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
return session->sessionid;
}
/**
* gst_rtsp_session_set_timeout:
* @session: a #GstRTSPSession
* @timeout: the new timeout
*
* Configure @session for a timeout of @timeout seconds. The session will be
* cleaned up when there is no activity for @timeout seconds.
*/
void
gst_rtsp_session_set_timeout (GstRTSPSession * session, guint timeout)
{
g_return_if_fail (GST_IS_RTSP_SESSION (session));
session->timeout = timeout;
}
/**
* gst_rtsp_session_get_timeout:
* @session: a #GstRTSPSession
*
* Get the timeout value of @session.
*
* Returns: the timeout of @session in seconds.
*/
guint
gst_rtsp_session_get_timeout (GstRTSPSession * session)
{
g_return_val_if_fail (GST_IS_RTSP_SESSION (session), 0);
return session->timeout;
}
/**
* gst_rtsp_session_touch:
* @session: a #GstRTSPSession
*
* Update the last_access time of the session to the current time.
*/
void
gst_rtsp_session_touch (GstRTSPSession * session)
{
g_return_if_fail (GST_IS_RTSP_SESSION (session));
g_get_current_time (&session->last_access);
}
void
gst_rtsp_session_prevent_expire (GstRTSPSession * session)
{
g_return_if_fail (GST_IS_RTSP_SESSION (session));
g_atomic_int_add (&session->expire_count, 1);
}
void
gst_rtsp_session_allow_expire (GstRTSPSession * session)
{
g_atomic_int_add (&session->expire_count, -1);
}
/**
* gst_rtsp_session_next_timeout:
* @session: a #GstRTSPSession
* @now: the current system time
*
* Get the amount of milliseconds till the session will expire.
*
* Returns: the amount of milliseconds since the session will time out.
*/
gint
gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
{
gint res;
GstClockTime last_access, now_ns;
g_return_val_if_fail (GST_IS_RTSP_SESSION (session), -1);
g_return_val_if_fail (now != NULL, -1);
if (g_atomic_int_get (&session->expire_count) != 0) {
/* touch session when the expire count is not 0 */
g_get_current_time (&session->last_access);
}
last_access = GST_TIMEVAL_TO_TIME (session->last_access);
/* add timeout allow for 5 seconds of extra time */
last_access += session->timeout * GST_SECOND + (5 * GST_SECOND);
now_ns = GST_TIMEVAL_TO_TIME (*now);
if (last_access > now_ns)
res = GST_TIME_AS_MSECONDS (last_access - now_ns);
else
res = 0;
return res;
}
/**
* gst_rtsp_session_is_expired:
* @session: a #GstRTSPSession
* @now: the current system time
*
* Check if @session timeout out.
*
* Returns: %TRUE if @session timed out
*/
gboolean
gst_rtsp_session_is_expired (GstRTSPSession * session, GTimeVal * now)
{
gboolean res;
res = (gst_rtsp_session_next_timeout (session, now) == 0);
return res;
}
/**
* gst_rtsp_session_stream_init_udp:
* @stream: a #GstRTSPSessionStream
* @ct: a client #GstRTSPTransport
*
* Set @ct as the client transport and create and return a matching server
* transport. This function takes ownership of the passed @ct.
*
* Returns: a server transport or NULL if something went wrong. Use
* gst_rtsp_transport_free () after usage.
*/
GstRTSPTransport *
gst_rtsp_session_stream_set_transport (GstRTSPSessionStream * stream,
GstRTSPTransport * ct)
{
GstRTSPTransport *st;
g_return_val_if_fail (stream != NULL, NULL);
g_return_val_if_fail (ct != NULL, NULL);
/* prepare the server transport */
gst_rtsp_transport_new (&st);
st->trans = ct->trans;
st->profile = ct->profile;
st->lower_transport = ct->lower_transport;
switch (st->lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
st->client_port = ct->client_port;
st->server_port = stream->media_stream->server_port;
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
ct->port = st->port = stream->media_stream->server_port;
st->destination = g_strdup (ct->destination);
break;
case GST_RTSP_LOWER_TRANS_TCP:
st->interleaved = ct->interleaved;
default:
break;
}
if (stream->media_stream->session)
g_object_get (stream->media_stream->session, "internal-ssrc", &st->ssrc,
NULL);
/* keep track of the transports in the stream. */
if (stream->trans.transport)
gst_rtsp_transport_free (stream->trans.transport);
stream->trans.transport = ct;
return st;
}
/**
* gst_rtsp_session_stream_set_callbacks:
* @stream: a #GstRTSPSessionStream
* @send_rtp: a callback called when RTP should be sent
* @send_rtcp: a callback called when RTCP should be sent
* @send_rtp_list: a callback called when RTP should be sent
* @send_rtcp_list: a callback called when RTCP should be sent
* @user_data: user data passed to callbacks
* @notify: called with the user_data when no longer needed.
*
* Install callbacks that will be called when data for a stream should be sent
* to a client. This is usually used when sending RTP/RTCP over TCP.
*/
void
gst_rtsp_session_stream_set_callbacks (GstRTSPSessionStream * stream,
GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
GstRTSPSendListFunc send_rtp_list, GstRTSPSendListFunc send_rtcp_list,
gpointer user_data, GDestroyNotify notify)
{
stream->trans.send_rtp = send_rtp;
stream->trans.send_rtcp = send_rtcp;
stream->trans.send_rtp_list = send_rtp_list;
stream->trans.send_rtcp_list = send_rtcp_list;
if (stream->trans.notify)
stream->trans.notify (stream->trans.user_data);
stream->trans.user_data = user_data;
stream->trans.notify = notify;
}
/**
* gst_rtsp_session_stream_set_keepalive:
* @stream: a #GstRTSPSessionStream
* @keep_alive: a callback called when the receiver is active
* @user_data: user data passed to callback
* @notify: called with the user_data when no longer needed.
*
* Install callbacks that will be called when RTCP packets are received from the
* receiver of @stream.
*/
void
gst_rtsp_session_stream_set_keepalive (GstRTSPSessionStream * stream,
GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
{
stream->trans.keep_alive = keep_alive;
if (stream->trans.ka_notify)
stream->trans.ka_notify (stream->trans.ka_user_data);
stream->trans.ka_user_data = user_data;
stream->trans.ka_notify = notify;
}
/**
* gst_rtsp_session_media_set_state:
* @media: a #GstRTSPSessionMedia
* @state: the new state
*
* Tell the media object @media to change to @state.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state)
{
gboolean ret;
g_return_val_if_fail (media != NULL, FALSE);
ret = gst_rtsp_media_set_state (media->media, state, media->streams);
return ret;
}