gstreamer/gst/rtsp-server/rtsp-session-media.c
Wim Taymans a64cb68164 media: add method to get the base_time of the pipeline
Together with a shared clock, this base-time could eventually be sent to
the client so that it can reconstruct the exact running-time of the clock
on the server.
2013-04-12 11:34:38 +01:00

361 lines
9.2 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"
#define GST_RTSP_SESSION_MEDIA_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaPrivate))
struct _GstRTSPSessionMediaPrivate
{
GMutex lock;
GstRTSPUrl *url; /* unmutable */
GstRTSPMedia *media; /* unmutable */
GstRTSPState state; /* protected by lock */
guint counter; /* protected by lock */
GPtrArray *transports; /* protected by lock */
};
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;
g_type_class_add_private (klass, sizeof (GstRTSPSessionMediaPrivate));
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)
{
GstRTSPSessionMediaPrivate *priv = GST_RTSP_SESSION_MEDIA_GET_PRIVATE (media);
media->priv = priv;
g_mutex_init (&priv->lock);
priv->state = GST_RTSP_STATE_INIT;
}
static void
gst_rtsp_session_media_finalize (GObject * obj)
{
GstRTSPSessionMedia *media;
GstRTSPSessionMediaPrivate *priv;
media = GST_RTSP_SESSION_MEDIA (obj);
priv = media->priv;
GST_INFO ("free session media %p", media);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
g_ptr_array_unref (priv->transports);
gst_rtsp_url_free (priv->url);
g_object_unref (priv->media);
g_mutex_clear (&priv->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)
{
GstRTSPSessionMediaPrivate *priv;
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 (gst_rtsp_media_get_status (media) ==
GST_RTSP_MEDIA_STATUS_PREPARED, NULL);
result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
priv = result->priv;
priv->url = gst_rtsp_url_copy ((GstRTSPUrl *) url);
priv->media = media;
/* prealloc the streams now, filled with NULL */
n_streams = gst_rtsp_media_n_streams (media);
priv->transports = g_ptr_array_new_full (n_streams, free_session_media);
g_ptr_array_set_size (priv->transports, n_streams);
return result;
}
/**
* gst_rtsp_session_media_matches_url:
* @media: a #GstRTSPSessionMedia
* @url: a #GstRTSPUrl
*
* Check if the url of @media matches @url.
*
* Returns: %TRUE when @url matches the url of @media.
*/
gboolean
gst_rtsp_session_media_matches_url (GstRTSPSessionMedia * media,
const GstRTSPUrl * url)
{
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
g_return_val_if_fail (url != NULL, FALSE);
return g_str_equal (media->priv->url->abspath, url->abspath);
}
/**
* gst_rtsp_session_media_get_media:
* @media: a #GstRTSPSessionMedia
*
* Get the #GstRTSPMedia that was used when constructing @media
*
* Returns: (transfer none): the #GstRTSPMedia of @media. Remains valid as long
* as @media is valid.
*/
GstRTSPMedia *
gst_rtsp_session_media_get_media (GstRTSPSessionMedia * media)
{
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
return media->priv->media;
}
/**
* gst_rtsp_session_media_get_base_time:
* @media: a #GstRTSPSessionMedia
*
* Get the base_time of the #GstRTSPMedia in @media
*
* Returns: the base_time of the media.
*/
GstClockTime
gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia * media)
{
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), GST_CLOCK_TIME_NONE);
return gst_rtsp_media_get_base_time (media->priv->media);
}
/**
* gst_rtsp_session_media_set_transport:
* @media: a #GstRTSPSessionMedia
* @stream: a #GstRTSPStream
* @tr: a #GstRTSPTransport
*
* 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)
{
GstRTSPSessionMediaPrivate *priv;
GstRTSPStreamTransport *result;
guint idx;
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 (tr != NULL, NULL);
priv = media->priv;
idx = gst_rtsp_stream_get_index (stream);
g_return_val_if_fail (idx < priv->transports->len, NULL);
g_mutex_lock (&priv->lock);
result = g_ptr_array_index (priv->transports, idx);
if (result == NULL) {
result = gst_rtsp_stream_transport_new (stream, tr);
g_ptr_array_index (priv->transports, idx) = result;
g_mutex_unlock (&priv->lock);
} else {
gst_rtsp_stream_transport_set_transport (result, tr);
g_mutex_unlock (&priv->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)
{
GstRTSPSessionMediaPrivate *priv;
GstRTSPStreamTransport *result;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
priv = media->priv;
g_return_val_if_fail (idx < priv->transports->len, NULL);
g_mutex_lock (&priv->lock);
result = g_ptr_array_index (priv->transports, idx);
g_mutex_unlock (&priv->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)
{
GstRTSPSessionMediaPrivate *priv;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
priv = media->priv;
g_mutex_lock (&priv->lock);
range->min = priv->counter++;
range->max = priv->counter++;
g_mutex_unlock (&priv->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)
{
GstRTSPSessionMediaPrivate *priv;
gboolean ret;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
priv = media->priv;
g_mutex_lock (&priv->lock);
ret = gst_rtsp_media_set_state (priv->media, state, priv->transports);
g_mutex_unlock (&priv->lock);
return ret;
}
/**
* gst_rtsp_session_media_set_rtsp_state:
* @media: a #GstRTSPSessionMedia
* @state: a #GstRTSPState
*
* Set the RTSP state of @media to @state.
*/
void
gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia * media,
GstRTSPState state)
{
GstRTSPSessionMediaPrivate *priv;
g_return_if_fail (GST_IS_RTSP_SESSION_MEDIA (media));
priv = media->priv;
g_mutex_lock (&priv->lock);
priv->state = state;
g_mutex_unlock (&priv->lock);
}
/**
* gst_rtsp_session_media_set_rtsp_state:
* @media: a #GstRTSPSessionMedia
*
* Get the current RTSP state of @media.
*
* Returns: the current RTSP state of @media.
*/
GstRTSPState
gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia * media)
{
GstRTSPSessionMediaPrivate *priv;
GstRTSPState ret;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media),
GST_RTSP_STATE_INVALID);
priv = media->priv;
g_mutex_lock (&priv->lock);
ret = priv->state;
g_mutex_unlock (&priv->lock);
return ret;
}