mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-04 15:36:35 +00:00
62d4c0b179
Export rtsp-server library API in headers when we're building the library itself, otherwise import the API from the headers. This fixes linker warnings on Windows when building with MSVC. Fix up some missing config.h includes when building the lib which is needed to get the export api define from config.h https://bugzilla.gnome.org/show_bug.cgi?id=797185
536 lines
14 KiB
C
536 lines
14 KiB
C
/* GStreamer
|
|
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
|
|
* Copyright (C) 2015 Centricular Ltd
|
|
* Author: Sebastian Dröge <sebastian@centricular.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)
|
|
*/
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <string.h>
|
|
|
|
#include "rtsp-session.h"
|
|
|
|
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_WITH_PRIVATE (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)
|
|
{
|
|
GstRTSPSessionMediaPrivate *priv;
|
|
|
|
media->priv = priv = gst_rtsp_session_media_get_instance_private (media);
|
|
|
|
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) (nullable): 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_is_sender (stream))
|
|
continue;
|
|
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) (nullable): 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_get_transports:
|
|
* @media: a #GstRTSPSessionMedia
|
|
*
|
|
* Get a list of all available #GstRTSPStreamTransport in this session.
|
|
*
|
|
* Returns: (transfer full) (element-type GstRTSPStreamTransport): a
|
|
* list of #GstRTSPStreamTransport, g_ptr_array_unref () after usage.
|
|
*/
|
|
GPtrArray *
|
|
gst_rtsp_session_media_get_transports (GstRTSPSessionMedia * media)
|
|
{
|
|
GstRTSPSessionMediaPrivate *priv;
|
|
GPtrArray *result;
|
|
|
|
g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
|
|
priv = media->priv;
|
|
|
|
g_mutex_lock (&priv->lock);
|
|
result = g_ptr_array_ref (priv->transports);
|
|
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;
|
|
}
|