gstreamer/gst/rtsp-server/rtsp-stream-transport.c
Branko Subasic 2218510cae rtsp-*: Treat sending packets to clients as keepalive
As long as gst-rtsp-server can successfully send RTP/RTCP data to
clients then the client must be reading. This change makes the server
timeout the connection if the client stops reading.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-24 10:37:59 +03:00

541 lines
14 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:rtsp-stream-transport
* @short_description: A media stream transport configuration
* @see_also: #GstRTSPStream, #GstRTSPSessionMedia
*
* The #GstRTSPStreamTransport configures the transport used by a
* #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
*
* With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
* to handle the RTP and RTCP packets from the stream, for example when they
* need to be sent over TCP.
*
* With gst_rtsp_stream_transport_set_active() the transports are added and
* removed from the stream.
*
* A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
* is received from the client. It will also call
* gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
*
* Last reviewed on 2013-07-16 (1.0.0)
*/
#include <string.h>
#include <stdlib.h>
#include "rtsp-stream-transport.h"
#define GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportPrivate))
struct _GstRTSPStreamTransportPrivate
{
GstRTSPStream *stream;
GstRTSPSendFunc send_rtp;
GstRTSPSendFunc send_rtcp;
gpointer user_data;
GDestroyNotify notify;
GstRTSPKeepAliveFunc keep_alive;
gpointer ka_user_data;
GDestroyNotify ka_notify;
gboolean active;
gboolean timed_out;
GstRTSPTransport *transport;
GstRTSPUrl *url;
GObject *rtpsource;
};
enum
{
PROP_0,
PROP_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
#define GST_CAT_DEFAULT rtsp_stream_transport_debug
static void gst_rtsp_stream_transport_finalize (GObject * obj);
G_DEFINE_TYPE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
G_TYPE_OBJECT);
static void
gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
{
GObjectClass *gobject_class;
g_type_class_add_private (klass, sizeof (GstRTSPStreamTransportPrivate));
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtsp_stream_transport_finalize;
GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
0, "GstRTSPStreamTransport");
}
static void
gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
{
GstRTSPStreamTransportPrivate *priv =
GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE (trans);
trans->priv = priv;
}
static void
gst_rtsp_stream_transport_finalize (GObject * obj)
{
GstRTSPStreamTransportPrivate *priv;
GstRTSPStreamTransport *trans;
trans = GST_RTSP_STREAM_TRANSPORT (obj);
priv = trans->priv;
/* remove callbacks now */
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
if (priv->transport)
gst_rtsp_transport_free (priv->transport);
if (priv->url)
gst_rtsp_url_free (priv->url);
G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
}
/**
* gst_rtsp_stream_transport_new:
* @stream: a #GstRTSPStream
* @tr: (transfer full): a GstRTSPTransport
*
* Create a new #GstRTSPStreamTransport that can be used to manage
* @stream with transport @tr.
*
* Returns: (transfer full): a new #GstRTSPStreamTransport
*/
GstRTSPStreamTransport *
gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
{
GstRTSPStreamTransportPrivate *priv;
GstRTSPStreamTransport *trans;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (tr != NULL, NULL);
trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
priv = trans->priv;
priv->stream = stream;
priv->transport = tr;
return trans;
}
/**
* gst_rtsp_stream_transport_get_stream:
* @trans: a #GstRTSPStreamTransport
*
* Get the #GstRTSPStream used when constructing @trans.
*
* Returns: (transfer none): the stream used when constructing @trans.
*/
GstRTSPStream *
gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
return trans->priv->stream;
}
/**
* gst_rtsp_stream_transport_set_callbacks:
* @trans: a #GstRTSPStreamTransport
* @send_rtp: (scope notified): a callback called when RTP should be sent
* @send_rtcp: (scope notified): a callback called when RTCP should be sent
* @user_data: (closure): user data passed to callbacks
* @notify: (allow-none): 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_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
gpointer user_data, GDestroyNotify notify)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->send_rtp = send_rtp;
priv->send_rtcp = send_rtcp;
if (priv->notify)
priv->notify (priv->user_data);
priv->user_data = user_data;
priv->notify = notify;
}
/**
* gst_rtsp_stream_transport_set_keepalive:
* @trans: a #GstRTSPStreamTransport
* @keep_alive: (scope notified): a callback called when the receiver is active
* @user_data: (closure): user data passed to callback
* @notify: (allow-none): called with the user_data when no longer needed.
*
* Install callbacks that will be called when RTCP packets are received from the
* receiver of @trans.
*/
void
gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->keep_alive = keep_alive;
if (priv->ka_notify)
priv->ka_notify (priv->ka_user_data);
priv->ka_user_data = user_data;
priv->ka_notify = notify;
}
/**
* gst_rtsp_stream_transport_set_transport:
* @trans: a #GstRTSPStreamTransport
* @tr: (transfer full): a client #GstRTSPTransport
*
* Set @tr as the client transport. This function takes ownership of the
* passed @tr.
*/
void
gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
GstRTSPTransport * tr)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
g_return_if_fail (tr != NULL);
priv = trans->priv;
/* keep track of the transports in the stream. */
if (priv->transport)
gst_rtsp_transport_free (priv->transport);
priv->transport = tr;
}
/**
* gst_rtsp_stream_transport_get_transport:
* @trans: a #GstRTSPStreamTransport
*
* Get the transport configured in @trans.
*
* Returns: (transfer none): the transport configured in @trans. It remains
* valid for as long as @trans is valid.
*/
const GstRTSPTransport *
gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
return trans->priv->transport;
}
/**
* gst_rtsp_stream_transport_set_url:
* @trans: a #GstRTSPStreamTransport
* @url: (transfer none): a client #GstRTSPUrl
*
* Set @url as the client url.
*/
void
gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
const GstRTSPUrl * url)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
/* keep track of the transports in the stream. */
if (priv->url)
gst_rtsp_url_free (priv->url);
priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
}
/**
* gst_rtsp_stream_transport_get_url:
* @trans: a #GstRTSPStreamTransport
*
* Get the url configured in @trans.
*
* Returns: (transfer none): the url configured in @trans. It remains
* valid for as long as @trans is valid.
*/
const GstRTSPUrl *
gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
return trans->priv->url;
}
/**
* gst_rtsp_stream_transport_get_rtpinfo:
* @trans: a #GstRTSPStreamTransport
* @start_time: a star time
*
* Get the RTP-Info string for @trans and @start_time.
*
* Returns: (transfer full) (nullable): the RTPInfo string for @trans
* and @start_time or %NULL when the RTP-Info could not be
* determined. g_free() after usage.
*/
gchar *
gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
GstClockTime start_time)
{
GstRTSPStreamTransportPrivate *priv;
gchar *url_str;
GString *rtpinfo;
guint rtptime, seq, clock_rate;
GstClockTime running_time = GST_CLOCK_TIME_NONE;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
priv = trans->priv;
if (!gst_rtsp_stream_get_rtpinfo (priv->stream, &rtptime, &seq, &clock_rate,
&running_time))
return NULL;
GST_DEBUG ("RTP time %u, seq %u, rate %u, running-time %" GST_TIME_FORMAT,
rtptime, seq, clock_rate, GST_TIME_ARGS (running_time));
if (GST_CLOCK_TIME_IS_VALID (running_time)
&& GST_CLOCK_TIME_IS_VALID (start_time)) {
if (running_time > start_time) {
rtptime -=
gst_util_uint64_scale_int (running_time - start_time, clock_rate,
GST_SECOND);
} else {
rtptime +=
gst_util_uint64_scale_int (start_time - running_time, clock_rate,
GST_SECOND);
}
}
GST_DEBUG ("RTP time %u, for start-time %" GST_TIME_FORMAT,
rtptime, GST_TIME_ARGS (start_time));
rtpinfo = g_string_new ("");
url_str = gst_rtsp_url_get_request_uri (trans->priv->url);
g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
url_str, seq, rtptime);
g_free (url_str);
return g_string_free (rtpinfo, FALSE);
}
/**
* gst_rtsp_stream_transport_set_active:
* @trans: a #GstRTSPStreamTransport
* @active: new state of @trans
*
* Activate or deactivate datatransfer configured in @trans.
*
* Returns: %TRUE when the state was changed.
*/
gboolean
gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
gboolean active)
{
GstRTSPStreamTransportPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
priv = trans->priv;
if (priv->active == active)
return FALSE;
if (active)
res = gst_rtsp_stream_add_transport (priv->stream, trans);
else
res = gst_rtsp_stream_remove_transport (priv->stream, trans);
if (res)
priv->active = active;
return res;
}
/**
* gst_rtsp_stream_transport_set_timed_out:
* @trans: a #GstRTSPStreamTransport
* @timedout: timed out value
*
* Set the timed out state of @trans to @timedout
*/
void
gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
gboolean timedout)
{
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
trans->priv->timed_out = timedout;
}
/**
* gst_rtsp_stream_transport_is_timed_out:
* @trans: a #GstRTSPStreamTransport
*
* Check if @trans is timed out.
*
* Returns: %TRUE if @trans timed out.
*/
gboolean
gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
return trans->priv->timed_out;
}
/**
* gst_rtsp_stream_transport_send_rtp:
* @trans: a #GstRTSPStreamTransport
* @buffer: (transfer none): a #GstBuffer
*
* Send @buffer to the installed RTP callback for @trans.
*
* Returns: %TRUE on success
*/
gboolean
gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
GstRTSPStreamTransportPrivate *priv;
gboolean res = FALSE;
priv = trans->priv;
if (priv->send_rtp)
res =
priv->send_rtp (buffer, priv->transport->interleaved.min,
priv->user_data);
if (res)
gst_rtsp_stream_transport_keep_alive (trans);
return res;
}
/**
* gst_rtsp_stream_transport_send_rtcp:
* @trans: a #GstRTSPStreamTransport
* @buffer: (transfer none): a #GstBuffer
*
* Send @buffer to the installed RTCP callback for @trans.
*
* Returns: %TRUE on success
*/
gboolean
gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
GstRTSPStreamTransportPrivate *priv;
gboolean res = FALSE;
priv = trans->priv;
if (priv->send_rtcp)
res =
priv->send_rtcp (buffer, priv->transport->interleaved.max,
priv->user_data);
if (res)
gst_rtsp_stream_transport_keep_alive (trans);
return res;
}
/**
* gst_rtsp_stream_transport_keep_alive:
* @trans: a #GstRTSPStreamTransport
*
* Signal the installed keep_alive callback for @trans.
*/
void
gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
{
GstRTSPStreamTransportPrivate *priv;
priv = trans->priv;
if (priv->keep_alive)
priv->keep_alive (priv->ka_user_data);
}
/**
* gst_rtsp_stream_transport_recv_data:
* @trans: a #GstRTSPStreamTransport
* @channel: a channel
* @buffer: (transfer full): a #GstBuffer
*
* Receive @buffer on @channel @trans.
*
* Returns: a #GstFlowReturn. Returns GST_FLOW_NOT_LINKED when @channel is not
* configured in the transport of @trans.
*/
GstFlowReturn
gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport * trans,
guint channel, GstBuffer * buffer)
{
GstRTSPStreamTransportPrivate *priv;
const GstRTSPTransport *tr;
GstFlowReturn res;
priv = trans->priv;
tr = priv->transport;
if (tr->interleaved.min == channel) {
res = gst_rtsp_stream_recv_rtp (priv->stream, buffer);
} else if (tr->interleaved.max == channel) {
res = gst_rtsp_stream_recv_rtcp (priv->stream, buffer);
} else {
res = GST_FLOW_NOT_LINKED;
}
return res;
}