mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-13 12:51:16 +00:00
fde25cd9c3
Fix caps. Remove bufferlist stuff Update for new API. Add queue before appsink now that preroll-queue-len is gone. Update for request pad changes.
616 lines
16 KiB
C
616 lines
16 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <string.h>
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#include "rtsp-session.h"
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#undef DEBUG
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#define DEFAULT_TIMEOUT 60
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enum
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{
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PROP_0,
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PROP_SESSIONID,
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PROP_TIMEOUT,
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PROP_LAST
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};
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GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
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#define GST_CAT_DEFAULT rtsp_session_debug
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static void gst_rtsp_session_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec);
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static void gst_rtsp_session_set_property (GObject * object, guint propid,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtsp_session_finalize (GObject * obj);
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G_DEFINE_TYPE (GstRTSPSession, gst_rtsp_session, G_TYPE_OBJECT);
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static void
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gst_rtsp_session_class_init (GstRTSPSessionClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->get_property = gst_rtsp_session_get_property;
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gobject_class->set_property = gst_rtsp_session_set_property;
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gobject_class->finalize = gst_rtsp_session_finalize;
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g_object_class_install_property (gobject_class, PROP_SESSIONID,
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g_param_spec_string ("sessionid", "Sessionid", "the session id",
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NULL, G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TIMEOUT,
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g_param_spec_uint ("timeout", "timeout",
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"the timeout of the session (0 = never)", 0, G_MAXUINT,
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DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsession", 0,
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"GstRTSPSession");
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}
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static void
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gst_rtsp_session_init (GstRTSPSession * session)
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{
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session->timeout = DEFAULT_TIMEOUT;
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g_get_current_time (&session->create_time);
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gst_rtsp_session_touch (session);
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}
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static void
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gst_rtsp_session_free_stream (GstRTSPSessionStream * stream)
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{
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GST_INFO ("free session stream %p", stream);
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/* remove callbacks now */
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gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
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gst_rtsp_session_stream_set_keepalive (stream, NULL, NULL, NULL);
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gst_rtsp_media_trans_cleanup (&stream->trans);
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g_free (stream);
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}
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static void
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gst_rtsp_session_free_media (GstRTSPSessionMedia * media,
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GstRTSPSession * session)
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{
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guint size, i;
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size = media->streams->len;
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GST_INFO ("free session media %p", media);
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gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
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for (i = 0; i < size; i++) {
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GstRTSPSessionStream *stream;
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stream = g_array_index (media->streams, GstRTSPSessionStream *, i);
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if (stream)
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gst_rtsp_session_free_stream (stream);
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}
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g_array_free (media->streams, TRUE);
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if (media->url)
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gst_rtsp_url_free (media->url);
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if (media->media)
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g_object_unref (media->media);
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g_free (media);
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}
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static void
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gst_rtsp_session_finalize (GObject * obj)
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{
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GstRTSPSession *session;
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session = GST_RTSP_SESSION (obj);
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GST_INFO ("finalize session %p", session);
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/* free all media */
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g_list_foreach (session->medias, (GFunc) gst_rtsp_session_free_media,
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session);
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g_list_free (session->medias);
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/* free session id */
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g_free (session->sessionid);
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G_OBJECT_CLASS (gst_rtsp_session_parent_class)->finalize (obj);
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}
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static void
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gst_rtsp_session_get_property (GObject * object, guint propid,
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GValue * value, GParamSpec * pspec)
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{
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GstRTSPSession *session = GST_RTSP_SESSION (object);
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switch (propid) {
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case PROP_SESSIONID:
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g_value_set_string (value, session->sessionid);
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break;
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case PROP_TIMEOUT:
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g_value_set_uint (value, gst_rtsp_session_get_timeout (session));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
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}
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}
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static void
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gst_rtsp_session_set_property (GObject * object, guint propid,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTSPSession *session = GST_RTSP_SESSION (object);
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switch (propid) {
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case PROP_SESSIONID:
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g_free (session->sessionid);
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session->sessionid = g_value_dup_string (value);
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break;
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case PROP_TIMEOUT:
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gst_rtsp_session_set_timeout (session, g_value_get_uint (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
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}
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}
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/**
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* gst_rtsp_session_manage_media:
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* @sess: a #GstRTSPSession
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* @uri: the uri for the media
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* @media: a #GstRTSPMedia
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*
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* Manage the media object @obj in @sess. @uri will be used to retrieve this
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* media from the session with gst_rtsp_session_get_media().
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*
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* Ownership is taken from @media.
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*
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* Returns: a new @GstRTSPSessionMedia object.
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*/
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GstRTSPSessionMedia *
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gst_rtsp_session_manage_media (GstRTSPSession * sess, const GstRTSPUrl * uri,
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GstRTSPMedia * media)
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{
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GstRTSPSessionMedia *result;
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guint n_streams;
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g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
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g_return_val_if_fail (uri != NULL, NULL);
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g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
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g_return_val_if_fail (media->status == GST_RTSP_MEDIA_STATUS_PREPARED, NULL);
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result = g_new0 (GstRTSPSessionMedia, 1);
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result->media = media;
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result->url = gst_rtsp_url_copy ((GstRTSPUrl *) uri);
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result->state = GST_RTSP_STATE_INIT;
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/* prealloc the streams now, filled with NULL */
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n_streams = gst_rtsp_media_n_streams (media);
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result->streams =
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g_array_sized_new (FALSE, TRUE, sizeof (GstRTSPSessionStream *),
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n_streams);
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g_array_set_size (result->streams, n_streams);
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sess->medias = g_list_prepend (sess->medias, result);
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GST_INFO ("manage new media %p in session %p", media, result);
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return result;
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}
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/**
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* gst_rtsp_session_release_media:
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* @sess: a #GstRTSPSession
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* @media: a #GstRTSPMedia
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*
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* Release the managed @media in @sess, freeing the memory allocated by it.
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*
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* Returns: %TRUE if there are more media session left in @sess.
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*/
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gboolean
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gst_rtsp_session_release_media (GstRTSPSession * sess,
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GstRTSPSessionMedia * media)
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{
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GList *walk, *next;
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g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
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g_return_val_if_fail (media != NULL, FALSE);
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for (walk = sess->medias; walk;) {
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GstRTSPSessionMedia *find;
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find = (GstRTSPSessionMedia *) walk->data;
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next = g_list_next (walk);
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if (find == media) {
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sess->medias = g_list_delete_link (sess->medias, walk);
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gst_rtsp_session_free_media (find, sess);
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break;
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}
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walk = next;
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}
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return (sess->medias != NULL);
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}
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/**
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* gst_rtsp_session_get_media:
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* @sess: a #GstRTSPSession
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* @url: the url for the media
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*
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* Get the session media of the @url.
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*
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* Returns: the configuration for @url in @sess.
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*/
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GstRTSPSessionMedia *
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gst_rtsp_session_get_media (GstRTSPSession * sess, const GstRTSPUrl * url)
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{
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GstRTSPSessionMedia *result;
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GList *walk;
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g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
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g_return_val_if_fail (url != NULL, NULL);
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result = NULL;
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for (walk = sess->medias; walk; walk = g_list_next (walk)) {
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result = (GstRTSPSessionMedia *) walk->data;
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if (strcmp (result->url->abspath, url->abspath) == 0)
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break;
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result = NULL;
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}
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return result;
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}
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/**
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* gst_rtsp_session_media_get_stream:
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* @media: a #GstRTSPSessionMedia
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* @idx: the stream index
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*
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* Get a previously created or create a new #GstRTSPSessionStream at @idx.
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*
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* Returns: a #GstRTSPSessionStream that is valid until the session of @media
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* is unreffed.
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*/
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GstRTSPSessionStream *
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gst_rtsp_session_media_get_stream (GstRTSPSessionMedia * media, guint idx)
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{
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GstRTSPSessionStream *result;
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g_return_val_if_fail (media != NULL, NULL);
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g_return_val_if_fail (media->media != NULL, NULL);
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if (idx >= media->streams->len)
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return NULL;
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result = g_array_index (media->streams, GstRTSPSessionStream *, idx);
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if (result == NULL) {
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GstRTSPMediaStream *media_stream;
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media_stream = gst_rtsp_media_get_stream (media->media, idx);
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if (media_stream == NULL)
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goto no_media;
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result = g_new0 (GstRTSPSessionStream, 1);
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result->trans.idx = idx;
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result->trans.transport = NULL;
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result->media_stream = media_stream;
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g_array_index (media->streams, GstRTSPSessionStream *, idx) = result;
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}
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return result;
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/* ERRORS */
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no_media:
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{
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return NULL;
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}
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}
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gboolean
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gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
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GstRTSPRange * range)
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{
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range->min = media->counter++;
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range->max = media->counter++;
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return TRUE;
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}
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/**
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* gst_rtsp_session_new:
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*
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* Create a new #GstRTSPSession instance.
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*/
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GstRTSPSession *
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gst_rtsp_session_new (const gchar * sessionid)
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{
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GstRTSPSession *result;
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g_return_val_if_fail (sessionid != NULL, NULL);
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result = g_object_new (GST_TYPE_RTSP_SESSION, "sessionid", sessionid, NULL);
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return result;
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}
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/**
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* gst_rtsp_session_get_sessionid:
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* @session: a #GstRTSPSession
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*
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* Get the sessionid of @session.
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*
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* Returns: the sessionid of @session. The value remains valid as long as
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* @session is alive.
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*/
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const gchar *
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gst_rtsp_session_get_sessionid (GstRTSPSession * session)
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{
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g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
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return session->sessionid;
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}
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/**
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* gst_rtsp_session_set_timeout:
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* @session: a #GstRTSPSession
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* @timeout: the new timeout
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*
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* Configure @session for a timeout of @timeout seconds. The session will be
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* cleaned up when there is no activity for @timeout seconds.
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*/
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void
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gst_rtsp_session_set_timeout (GstRTSPSession * session, guint timeout)
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{
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g_return_if_fail (GST_IS_RTSP_SESSION (session));
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session->timeout = timeout;
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}
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/**
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* gst_rtsp_session_get_timeout:
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* @session: a #GstRTSPSession
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*
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* Get the timeout value of @session.
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*
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* Returns: the timeout of @session in seconds.
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*/
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guint
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gst_rtsp_session_get_timeout (GstRTSPSession * session)
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{
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g_return_val_if_fail (GST_IS_RTSP_SESSION (session), 0);
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return session->timeout;
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}
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/**
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* gst_rtsp_session_touch:
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* @session: a #GstRTSPSession
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*
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* Update the last_access time of the session to the current time.
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*/
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void
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gst_rtsp_session_touch (GstRTSPSession * session)
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{
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g_return_if_fail (GST_IS_RTSP_SESSION (session));
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g_get_current_time (&session->last_access);
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}
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void
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gst_rtsp_session_prevent_expire (GstRTSPSession * session)
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{
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g_return_if_fail (GST_IS_RTSP_SESSION (session));
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g_atomic_int_add (&session->expire_count, 1);
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}
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void
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gst_rtsp_session_allow_expire (GstRTSPSession * session)
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{
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g_atomic_int_add (&session->expire_count, -1);
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}
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/**
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* gst_rtsp_session_next_timeout:
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* @session: a #GstRTSPSession
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* @now: the current system time
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*
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* Get the amount of milliseconds till the session will expire.
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*
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* Returns: the amount of milliseconds since the session will time out.
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*/
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gint
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gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
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{
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gint res;
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GstClockTime last_access, now_ns;
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g_return_val_if_fail (GST_IS_RTSP_SESSION (session), -1);
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g_return_val_if_fail (now != NULL, -1);
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if (g_atomic_int_get (&session->expire_count) != 0) {
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/* touch session when the expire count is not 0 */
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g_get_current_time (&session->last_access);
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}
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last_access = GST_TIMEVAL_TO_TIME (session->last_access);
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/* add timeout allow for 5 seconds of extra time */
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last_access += session->timeout * GST_SECOND + (5 * GST_SECOND);
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now_ns = GST_TIMEVAL_TO_TIME (*now);
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if (last_access > now_ns)
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res = GST_TIME_AS_MSECONDS (last_access - now_ns);
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else
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res = 0;
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return res;
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}
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/**
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* gst_rtsp_session_is_expired:
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* @session: a #GstRTSPSession
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* @now: the current system time
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*
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* Check if @session timeout out.
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*
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* Returns: %TRUE if @session timed out
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*/
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gboolean
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gst_rtsp_session_is_expired (GstRTSPSession * session, GTimeVal * now)
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{
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gboolean res;
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res = (gst_rtsp_session_next_timeout (session, now) == 0);
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return res;
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}
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/**
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* gst_rtsp_session_stream_init_udp:
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* @stream: a #GstRTSPSessionStream
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* @ct: a client #GstRTSPTransport
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*
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* Set @ct as the client transport and create and return a matching server
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* transport. This function takes ownership of the passed @ct.
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*
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* Returns: a server transport or NULL if something went wrong. Use
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* gst_rtsp_transport_free () after usage.
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*/
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GstRTSPTransport *
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gst_rtsp_session_stream_set_transport (GstRTSPSessionStream * stream,
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GstRTSPTransport * ct)
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{
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GstRTSPTransport *st;
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g_return_val_if_fail (stream != NULL, NULL);
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g_return_val_if_fail (ct != NULL, NULL);
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/* prepare the server transport */
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gst_rtsp_transport_new (&st);
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st->trans = ct->trans;
|
|
st->profile = ct->profile;
|
|
st->lower_transport = ct->lower_transport;
|
|
|
|
switch (st->lower_transport) {
|
|
case GST_RTSP_LOWER_TRANS_UDP:
|
|
st->client_port = ct->client_port;
|
|
st->server_port = stream->media_stream->server_port;
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
|
|
ct->port = st->port = stream->media_stream->server_port;
|
|
st->destination = g_strdup (ct->destination);
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_TCP:
|
|
st->interleaved = ct->interleaved;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (stream->media_stream->session)
|
|
g_object_get (stream->media_stream->session, "internal-ssrc", &st->ssrc,
|
|
NULL);
|
|
|
|
/* keep track of the transports in the stream. */
|
|
if (stream->trans.transport)
|
|
gst_rtsp_transport_free (stream->trans.transport);
|
|
stream->trans.transport = ct;
|
|
|
|
return st;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_session_stream_set_callbacks:
|
|
* @stream: a #GstRTSPSessionStream
|
|
* @send_rtp: a callback called when RTP should be sent
|
|
* @send_rtcp: a callback called when RTCP should be sent
|
|
* @send_rtp_list: a callback called when RTP should be sent
|
|
* @send_rtcp_list: a callback called when RTCP should be sent
|
|
* @user_data: user data passed to callbacks
|
|
* @notify: called with the user_data when no longer needed.
|
|
*
|
|
* Install callbacks that will be called when data for a stream should be sent
|
|
* to a client. This is usually used when sending RTP/RTCP over TCP.
|
|
*/
|
|
void
|
|
gst_rtsp_session_stream_set_callbacks (GstRTSPSessionStream * stream,
|
|
GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
|
|
gpointer user_data, GDestroyNotify notify)
|
|
{
|
|
stream->trans.send_rtp = send_rtp;
|
|
stream->trans.send_rtcp = send_rtcp;
|
|
if (stream->trans.notify)
|
|
stream->trans.notify (stream->trans.user_data);
|
|
stream->trans.user_data = user_data;
|
|
stream->trans.notify = notify;
|
|
}
|
|
|
|
/**
|
|
* gst_rtsp_session_stream_set_keepalive:
|
|
* @stream: a #GstRTSPSessionStream
|
|
* @keep_alive: a callback called when the receiver is active
|
|
* @user_data: user data passed to callback
|
|
* @notify: called with the user_data when no longer needed.
|
|
*
|
|
* Install callbacks that will be called when RTCP packets are received from the
|
|
* receiver of @stream.
|
|
*/
|
|
void
|
|
gst_rtsp_session_stream_set_keepalive (GstRTSPSessionStream * stream,
|
|
GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
|
|
{
|
|
stream->trans.keep_alive = keep_alive;
|
|
if (stream->trans.ka_notify)
|
|
stream->trans.ka_notify (stream->trans.ka_user_data);
|
|
stream->trans.ka_user_data = user_data;
|
|
stream->trans.ka_notify = notify;
|
|
}
|
|
|
|
/**
|
|
* 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 (media != NULL, FALSE);
|
|
|
|
ret = gst_rtsp_media_set_state (media->media, state, media->streams);
|
|
|
|
return ret;
|
|
}
|