gstreamer/gst/rtsp-server/rtsp-session-media.c
Linus Svensson a455181aff client: Stop caching media in client when doing setup
If the media has been managed by a session media, it should not be
cached in the client any longer. The GstRTSPSessionMedia object is now
responsible for unpreparing the GstRTSPMedia object using
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
session media.

https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-11-07 12:34:23 +01:00

507 lines
13 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-session-media
* @short_description: Media managed in a session
* @see_also: #GstRTSPMedia, #GstRTSPSession
*
* The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path.
*
* With gst_rtsp_session_media_get_transport() and
* gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of
* the managed #GstRTSPMedia can be retrieved and configured.
*
* Use gst_rtsp_session_media_set_state() to control the media state and
* transports.
*
* Last reviewed on 2013-07-16 (1.0.0)
*/
#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;
gchar *path; /* unmutable */
gint path_len; /* 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);
gst_rtsp_media_unprepare (priv->media);
g_ptr_array_unref (priv->transports);
g_free (priv->path);
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:
* @path: the path
* @media: (transfer full): the #GstRTSPMedia
*
* Create a new #GstRTSPSessionMedia that manages the streams
* in @media for @path. @media should be prepared.
*
* Ownership is taken of @media.
*
* Returns: (transfer full): a new #GstRTSPSessionMedia.
*/
GstRTSPSessionMedia *
gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media)
{
GstRTSPSessionMediaPrivate *priv;
GstRTSPSessionMedia *result;
guint n_streams;
GstRTSPMediaStatus status;
g_return_val_if_fail (path != NULL, NULL);
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
status = gst_rtsp_media_get_status (media);
g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
priv = result->priv;
priv->path = g_strdup (path);
priv->path_len = strlen (path);
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:
* @media: a #GstRTSPSessionMedia
* @path: a path
* @matched: (out): the amount of matched characters of @path
*
* Check if the path of @media matches @path. It @path matches, the amount of
* matched characters is returned in @matched.
*
* Returns: %TRUE when @path matches the path of @media.
*/
gboolean
gst_rtsp_session_media_matches (GstRTSPSessionMedia * media,
const gchar * path, gint * matched)
{
GstRTSPSessionMediaPrivate *priv;
gint len;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
g_return_val_if_fail (path != NULL, FALSE);
g_return_val_if_fail (matched != NULL, FALSE);
priv = media->priv;
len = strlen (path);
/* path needs to be smaller than the media path */
if (len < priv->path_len)
return FALSE;
/* if media path is larger, it there should be a / following the path */
if (len > priv->path_len && path[priv->path_len] != '/')
return FALSE;
*matched = priv->path_len;
return strncmp (path, priv->path, priv->path_len) == 0;
}
/**
* 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_get_rtpinfo:
* @media: a #GstRTSPSessionMedia
*
* Retrieve the RTP-Info header string for all streams in @media
* with configured transports.
*
* Returns: (transfer full) (nullable): The RTP-Info as a string or
* %NULL when no RTP-Info could be generated, g_free() after usage.
*/
gchar *
gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media)
{
GstRTSPSessionMediaPrivate *priv;
GString *rtpinfo = NULL;
GstRTSPStreamTransport *transport;
GstRTSPStream *stream;
guint i, n_streams;
GstClockTime earliest = GST_CLOCK_TIME_NONE;
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
priv = media->priv;
g_mutex_lock (&priv->lock);
if (gst_rtsp_media_get_status (priv->media) != GST_RTSP_MEDIA_STATUS_PREPARED)
goto not_prepared;
n_streams = priv->transports->len;
/* first step, take lowest running-time from all streams */
GST_LOG_OBJECT (media, "determining start time among %d transports",
n_streams);
for (i = 0; i < n_streams; i++) {
GstClockTime running_time;
transport = g_ptr_array_index (priv->transports, i);
if (transport == NULL) {
GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
continue;
}
stream = gst_rtsp_stream_transport_get_stream (transport);
if (!gst_rtsp_stream_get_rtpinfo (stream, NULL, NULL, NULL, &running_time))
continue;
GST_LOG_OBJECT (media, "running time of %d stream: %" GST_TIME_FORMAT, i,
GST_TIME_ARGS (running_time));
if (!GST_CLOCK_TIME_IS_VALID (earliest)) {
earliest = running_time;
} else {
earliest = MIN (earliest, running_time);
}
}
GST_LOG_OBJECT (media, "media start time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (earliest));
/* next step, scale all rtptime of all streams to lowest running-time */
GST_LOG_OBJECT (media, "collecting RTP info for %d transports", n_streams);
for (i = 0; i < n_streams; i++) {
gchar *stream_rtpinfo;
transport = g_ptr_array_index (priv->transports, i);
if (transport == NULL) {
GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
continue;
}
stream_rtpinfo =
gst_rtsp_stream_transport_get_rtpinfo (transport, earliest);
if (stream_rtpinfo == NULL) {
GST_DEBUG_OBJECT (media, "ignoring unknown RTPInfo %d", i);
continue;
}
if (rtpinfo == NULL)
rtpinfo = g_string_new ("");
else
g_string_append (rtpinfo, ", ");
g_string_append (rtpinfo, stream_rtpinfo);
g_free (stream_rtpinfo);
}
g_mutex_unlock (&priv->lock);
if (rtpinfo == NULL) {
GST_WARNING_OBJECT (media, "RTP info is empty");
return NULL;
}
return g_string_free (rtpinfo, FALSE);
/* ERRORS */
not_prepared:
{
g_mutex_unlock (&priv->lock);
GST_ERROR_OBJECT (media, "media was not prepared");
return NULL;
}
}
/**
* gst_rtsp_session_media_set_transport:
* @media: a #GstRTSPSessionMedia
* @stream: a #GstRTSPStream
* @tr: (transfer full): 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: (out): 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_get_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;
}