gstreamer/gst/rtsp-server/rtsp-media.c
Wim Taymans d5a00f1f23 Rework the way we handle transports for streams
Make the media accept an array of transports for the streams that we have
configured for the play/pause requests.

Implement server states for a client and its media.

Require 0.10.22.1 (git HEAD) of gstreamer.
2009-02-03 19:32:38 +01:00

662 lines
17 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 "rtsp-media.h"
#define DEFAULT_SHARED FALSE
enum
{
PROP_0,
PROP_SHARED,
PROP_LAST
};
static void gst_rtsp_media_get_property (GObject *object, guint propid,
GValue *value, GParamSpec *pspec);
static void gst_rtsp_media_set_property (GObject *object, guint propid,
const GValue *value, GParamSpec *pspec);
static void gst_rtsp_media_finalize (GObject * obj);
G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
static void
gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_media_get_property;
gobject_class->set_property = gst_rtsp_media_set_property;
gobject_class->finalize = gst_rtsp_media_finalize;
g_object_class_install_property (gobject_class, PROP_SHARED,
g_param_spec_boolean ("shared", "Shared", "If this media pipeline can be shared",
DEFAULT_SHARED, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_rtsp_media_init (GstRTSPMedia * media)
{
media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
media->complete = FALSE;
}
static void
gst_rtsp_media_stream_free (GstRTSPMediaStream *stream)
{
if (stream->caps)
gst_caps_unref (stream->caps);
g_free (stream);
}
static void
gst_rtsp_media_finalize (GObject * obj)
{
GstRTSPMedia *media;
guint i;
media = GST_RTSP_MEDIA (obj);
for (i = 0; i < media->streams->len; i++) {
GstRTSPMediaStream *stream;
stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
gst_rtsp_media_stream_free (stream);
}
g_array_free (media->streams, TRUE);
if (media->pipeline)
gst_object_unref (media->pipeline);
G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
}
static void
gst_rtsp_media_get_property (GObject *object, guint propid,
GValue *value, GParamSpec *pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
switch (propid) {
case PROP_SHARED:
g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_media_set_property (GObject *object, guint propid,
const GValue *value, GParamSpec *pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
switch (propid) {
case PROP_SHARED:
gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
* gst_rtsp_media_new:
*
* Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
* element to produde RTP data for one or more related (audio/video/..)
* streams.
*
* Returns: a new #GstRTSPMedia object.
*/
GstRTSPMedia *
gst_rtsp_media_new (void)
{
GstRTSPMedia *result;
result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);
return result;
}
/**
* gst_rtsp_media_set_shared:
* @media: a #GstRTSPMedia
* @shared: the new value
*
* Set or unset if the pipeline for @media can be shared will multiple clients.
* When @shared is %TRUE, client requests for this media will share the media
* pipeline.
*/
void
gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
media->shared = shared;
}
/**
* gst_rtsp_media_is_shared:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media can be shared between multiple clients.
*
* Returns: %TRUE if the media can be shared between clients.
*/
gboolean
gst_rtsp_media_is_shared (GstRTSPMedia *media)
{
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
return media->shared;
}
/**
* gst_rtsp_media_n_streams:
* @media: a #GstRTSPMedia
*
* Get the number of streams in this media.
*
* Returns: The number of streams.
*/
guint
gst_rtsp_media_n_streams (GstRTSPMedia *media)
{
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
return media->streams->len;
}
/**
* gst_rtsp_media_get_stream:
* @media: a #GstRTSPMedia
* @idx: the stream index
*
* Retrieve the stream with index @idx from @media.
*
* Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with
* that index did not exist.
*/
GstRTSPMediaStream *
gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx)
{
GstRTSPMediaStream *res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
if (idx < media->streams->len)
res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
else
res = NULL;
return res;
}
/* Allocate the udp ports and sockets */
static gboolean
alloc_udp_ports (GstRTSPMediaStream * stream)
{
GstStateChangeReturn ret;
GstElement *udpsrc0, *udpsrc1;
GstElement *udpsink0, *udpsink1;
gint tmp_rtp, tmp_rtcp;
guint count;
gint rtpport, rtcpport, sockfd;
udpsrc0 = NULL;
udpsrc1 = NULL;
udpsink0 = NULL;
udpsink1 = NULL;
count = 0;
/* Start with random port */
tmp_rtp = 0;
/* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
if (udpsrc0 == NULL)
goto no_udp_protocol;
g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE) {
if (tmp_rtp != 0) {
tmp_rtp += 2;
if (++count > 20)
goto no_ports;
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
goto again;
}
goto no_udp_protocol;
}
g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
/* check if port is even */
if ((tmp_rtp & 1) != 0) {
/* port not even, close and allocate another */
if (++count > 20)
goto no_ports;
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
tmp_rtp++;
goto again;
}
/* allocate port+1 for RTCP now */
udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
if (udpsrc1 == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
/* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
if (ret == GST_STATE_CHANGE_FAILURE) {
if (++count > 20)
goto no_ports;
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
tmp_rtp += 2;
goto again;
}
/* all fine, do port check */
g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
/* this should not happen... */
if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
goto port_error;
udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink0)
goto no_udp_protocol;
g_object_get (G_OBJECT (udpsrc0), "sock", &sockfd, NULL);
g_object_set (G_OBJECT (udpsink0), "sockfd", sockfd, NULL);
g_object_set (G_OBJECT (udpsink0), "closefd", FALSE, NULL);
udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
if (!udpsink1)
goto no_udp_protocol;
g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL);
g_object_set (G_OBJECT (udpsink1), "sockfd", sockfd, NULL);
g_object_set (G_OBJECT (udpsink1), "closefd", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
/* we keep these elements, we configure all in configure_transport when the
* server told us to really use the UDP ports. */
stream->udpsrc[0] = udpsrc0;
stream->udpsrc[1] = udpsrc1;
stream->udpsink[0] = udpsink0;
stream->udpsink[1] = udpsink1;
stream->server_port.min = rtpport;
stream->server_port.max = rtcpport;
return TRUE;
/* ERRORS */
no_udp_protocol:
{
goto cleanup;
}
no_ports:
{
goto cleanup;
}
no_udp_rtcp_protocol:
{
goto cleanup;
}
port_error:
{
goto cleanup;
}
cleanup:
{
if (udpsrc0) {
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
}
if (udpsrc1) {
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
}
if (udpsink0) {
gst_element_set_state (udpsink0, GST_STATE_NULL);
gst_object_unref (udpsink0);
}
if (udpsink1) {
gst_element_set_state (udpsink1, GST_STATE_NULL);
gst_object_unref (udpsink1);
}
return FALSE;
}
}
static void
caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
{
gchar *capsstr;
if (stream->caps)
gst_caps_unref (stream->caps);
if ((stream->caps = GST_PAD_CAPS (pad)))
gst_caps_ref (stream->caps);
capsstr = gst_caps_to_string (stream->caps);
g_message ("stream %p received caps %s", stream, capsstr);
g_free (capsstr);
}
/* prepare the pipeline objects to handle @stream in @media */
static gboolean
setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
{
gchar *name;
GstPad *pad;
alloc_udp_ports (stream);
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[0]);
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[1]);
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[0]);
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[1]);
/* hook up the stream to the RTP session elements. */
name = g_strdup_printf ("send_rtp_sink_%d", idx);
stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
g_free (name);
name = g_strdup_printf ("send_rtp_src_%d", idx);
stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name);
g_free (name);
name = g_strdup_printf ("send_rtcp_src_%d", idx);
stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name);
g_free (name);
name = g_strdup_printf ("recv_rtcp_sink_%d", idx);
stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
g_free (name);
/* link the RTP pad to the session manager */
gst_pad_link (stream->srcpad, stream->send_rtp_sink);
/* link udp elements */
pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
gst_pad_link (stream->send_rtp_src, pad);
gst_object_unref (pad);
pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
gst_pad_link (stream->send_rtcp_src, pad);
gst_object_unref (pad);
pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
gst_pad_link (pad, stream->recv_rtcp_sink);
gst_object_unref (pad);
/* we set and keep these to playing so that they don't cause NO_PREROLL return
* values */
gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
gst_element_set_locked_state (stream->udpsrc[0], TRUE);
gst_element_set_locked_state (stream->udpsrc[1], TRUE);
/* be notified of caps changes */
stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
(GCallback) caps_notify, stream);
stream->prepared = TRUE;
return TRUE;
}
/**
* gst_rtsp_media_prepare:
* @obj: a #GstRTSPMedia
*
* Prepare @media for streaming. This function will create the pipeline and
* other objects to manage the streaming.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_prepare (GstRTSPMedia *media)
{
GstStateChangeReturn ret;
guint i, n_streams;
if (media->prepared)
goto was_prepared;
g_message ("preparing media %p", media);
media->pipeline = gst_pipeline_new ("media-pipeline");
gst_bin_add (GST_BIN_CAST (media->pipeline), media->element);
media->rtpbin = gst_element_factory_make ("gstrtpbin", "rtpbin");
/* add stuf to the bin */
gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);
/* link streams we already have */
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPMediaStream *stream;
stream = gst_rtsp_media_get_stream (media, i);
setup_stream (stream, i, media);
}
/* first go to PAUSED */
ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
break;
case GST_STATE_CHANGE_ASYNC:
break;
case GST_STATE_CHANGE_NO_PREROLL:
/* we need to go to PLAYING */
g_message ("live media %p", media);
ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
break;
case GST_STATE_CHANGE_FAILURE:
goto state_failed;
}
/* now wait for all pads to be prerolled */
ret = gst_element_get_state (media->pipeline, NULL, NULL, -1);
/* and back to PAUSED for live pipelines */
ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
/* unlock the udp src elements */
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPMediaStream *stream;
stream = gst_rtsp_media_get_stream (media, i);
gst_element_set_locked_state (stream->udpsrc[0], FALSE);
gst_element_set_locked_state (stream->udpsrc[1], FALSE);
}
g_message ("object %p is prerolled", media);
media->prepared = TRUE;
return TRUE;
/* OK */
was_prepared:
{
return TRUE;
}
/* ERRORS */
state_failed:
{
g_message ("state change failed for media %p", media);
return FALSE;
}
}
/**
* gst_rtsp_media_play:
* @media: a #GstRTSPMedia
* @transports: a GArray of #GstRTSPMediaTrans pointers
*
* Start playing @media for to the transports in @transports.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_play (GstRTSPMedia *media, GArray *transports)
{
gint i;
GstStateChangeReturn ret;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (transports != NULL, FALSE);
g_return_val_if_fail (media->prepared, FALSE);
for (i = 0; i < transports->len; i++) {
GstRTSPMediaTrans *tr;
GstRTSPMediaStream *stream;
GstRTSPTransport *trans;
/* we need a non-NULL entry in the array */
tr = g_array_index (transports, GstRTSPMediaTrans *, i);
if (tr == NULL)
continue;
/* we need a transport */
if (!(trans = tr->transport))
continue;
/* get the stream and add the destinations */
stream = gst_rtsp_media_get_stream (media, tr->idx);
g_message ("adding %s:%d-%d", trans->destination, trans->client_port.min, trans->client_port.max);
g_signal_emit_by_name (stream->udpsink[0], "add", trans->destination, trans->client_port.min, NULL);
g_signal_emit_by_name (stream->udpsink[1], "add", trans->destination, trans->client_port.max, NULL);
}
g_message ("playing");
ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
return TRUE;
}
/**
* gst_rtsp_media_pause:
* @media: a #GstRTSPMedia
* @transports: a array of #GstRTSPTransport pointers
*
* Pause playing @media for to the transports in @transports.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_pause (GstRTSPMedia *media, GArray *transports)
{
gint i;
GstStateChangeReturn ret;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (transports != NULL, FALSE);
g_return_val_if_fail (media->prepared, FALSE);
for (i = 0; i < transports->len; i++) {
GstRTSPMediaTrans *tr;
GstRTSPMediaStream *stream;
GstRTSPTransport *trans;
/* we need a non-NULL entry in the array */
tr = g_array_index (transports, GstRTSPMediaTrans *, i);
if (tr == NULL)
continue;
/* we need a transport */
if (!(trans = tr->transport))
continue;
/* get the stream and add the destinations */
stream = gst_rtsp_media_get_stream (media, tr->idx);
g_message ("removing %s:%d-%d", trans->destination, trans->client_port.min, trans->client_port.max);
g_signal_emit_by_name (stream->udpsink[0], "remove", trans->destination, trans->client_port.min, NULL);
g_signal_emit_by_name (stream->udpsink[1], "remove", trans->destination, trans->client_port.max, NULL);
}
g_message ("pause");
ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
return TRUE;
}
/**
* gst_rtsp_media_stream_stop:
* @media: a #GstRTSPMedia
* @transports: a GArray of #GstRTSPMediaTrans pointers
*
* Stop playing @media for to the transports in @transports.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_stop (GstRTSPMedia *media, GArray *transports)
{
GstStateChangeReturn ret;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (transports != NULL, FALSE);
g_return_val_if_fail (media->prepared, FALSE);
gst_rtsp_media_pause (media, transports);
g_message ("stop");
ret = gst_element_set_state (media->pipeline, GST_STATE_NULL);
return TRUE;
}