gstreamer/gst/rtsp-server/rtsp-session-media.c
Wim Taymans 45b6693b39 rtsp: make address-pool return an address object
Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
store more info in the structure and allows us to more easily return the address
to the right pool when no longer needed.
Pass the address to the StreamTransport so that we can return it to the pool
when the stream transport is freed or changed.
2012-11-15 13:25:14 +01:00

226 lines
6 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.
*/
#include <string.h>
#include "rtsp-session.h"
#undef DEBUG
#define DEFAULT_TIMEOUT 60
enum
{
PROP_0,
PROP_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug);
#define GST_CAT_DEFAULT rtsp_session_media_debug
static void gst_rtsp_session_media_finalize (GObject * obj);
G_DEFINE_TYPE (GstRTSPSessionMedia, gst_rtsp_session_media, G_TYPE_OBJECT);
static void
gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtsp_session_media_finalize;
GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0,
"GstRTSPSessionMedia");
}
static void
gst_rtsp_session_media_init (GstRTSPSessionMedia * media)
{
g_mutex_init (&media->lock);
media->state = GST_RTSP_STATE_INIT;
}
static void
gst_rtsp_session_media_finalize (GObject * obj)
{
GstRTSPSessionMedia *media;
media = GST_RTSP_SESSION_MEDIA (obj);
GST_INFO ("free session media %p", media);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
g_ptr_array_unref (media->transports);
gst_rtsp_url_free (media->url);
g_object_unref (media->media);
g_mutex_clear (&media->lock);
G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj);
}
static void
free_session_media (gpointer data)
{
if (data)
g_object_unref (data);
}
/**
* gst_rtsp_session_media_new:
* @url: the #GstRTSPUrl
* @media: the #GstRTSPMedia
*
* Create a new #GstRTPSessionMedia that manages the streams
* in @media for @url. @media should be prepared.
*
* Ownership is taken of @media.
*
* Returns: a new #GstRTSPSessionMedia.
*/
GstRTSPSessionMedia *
gst_rtsp_session_media_new (const GstRTSPUrl * url, GstRTSPMedia * media)
{
GstRTSPSessionMedia *result;
guint n_streams;
g_return_val_if_fail (url != 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_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
result->url = gst_rtsp_url_copy ((GstRTSPUrl *) url);
result->media = media;
/* prealloc the streams now, filled with NULL */
n_streams = gst_rtsp_media_n_streams (media);
result->transports = g_ptr_array_new_full (n_streams, free_session_media);
g_ptr_array_set_size (result->transports, n_streams);
return result;
}
/**
* gst_rtsp_session_media_set_transport:
* @media: a #GstRTSPSessionMedia
* @stream: a #GstRTSPStream
* @tr: a #GstRTSPTransport
* @addr: (transfer full) (allow none): an optional #GstRTSPAddress
*
* Configure the transport for @stream to @tr in @media.
*
* Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream.
*/
GstRTSPStreamTransport *
gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
GstRTSPStream * stream, GstRTSPTransport * tr, GstRTSPAddress * addr)
{
GstRTSPStreamTransport *result;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (stream->idx < media->transports->len, NULL);
g_mutex_lock (&media->lock);
result = g_ptr_array_index (media->transports, stream->idx);
if (result == NULL) {
result = gst_rtsp_stream_transport_new (stream, tr, addr);
g_ptr_array_index (media->transports, stream->idx) = result;
g_mutex_unlock (&media->lock);
} else {
gst_rtsp_stream_transport_set_transport (result, tr, addr);
g_mutex_unlock (&media->lock);
}
return result;
}
/**
* gst_rtsp_session_media_get_transport:
* @media: a #GstRTSPSessionMedia
* @idx: the stream index
*
* Get a previously created #GstRTSPStreamTransport for the stream at @idx.
*
* Returns: (transfer none): a #GstRTSPStreamTransport that is valid until the
* session of @media is unreffed.
*/
GstRTSPStreamTransport *
gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
{
GstRTSPStreamTransport *result;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
g_return_val_if_fail (idx < media->transports->len, NULL);
g_mutex_lock (&media->lock);
result = g_ptr_array_index (media->transports, idx);
g_mutex_unlock (&media->lock);
return result;
}
/**
* gst_rtsp_session_media_alloc_channels:
* @media: a #GstRTSPSessionMedia
* @range: a #GstRTSPRange
*
* Fill @range with the next available min and max channels for
* interleaved transport.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
GstRTSPRange * range)
{
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
g_mutex_lock (&media->lock);
range->min = media->counter++;
range->max = media->counter++;
g_mutex_unlock (&media->lock);
return TRUE;
}
/**
* 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 (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
g_mutex_lock (&media->lock);
ret = gst_rtsp_media_set_state (media->media, state, media->transports);
g_mutex_unlock (&media->lock);
return ret;
}