Gst-rtsp-server gst_rtsp_client_close

General Discussion
1

I am using gst-rtsp-server and I want to be able to close an ongoing stream from the server. I am trying to use the gst_rtsp_client_close() function. However, when I call it the application sometimes becomes unresponsive. In the log I can see: GLib-CRITICAL **: g_main_context_unref: assertion 'g_atomic_int_get (&context->ref_count) > 0' failed​.

I have modified the test-uri example to trigger this issue. Whenever a client connects, a timer will be started. When the timer runs out, gst_rtsp_client_close() will be called for that client.

/* 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 <gst/gst.h>

#include <gst/rtsp-server/rtsp-server.h>
#include <gst/rtsp-server/rtsp-media-factory-uri.h>

#define DEFAULT_RTSP_PORT "8554"

static char *port = (char *) DEFAULT_RTSP_PORT;

static GOptionEntry entries[] = {
  {"port", 'p', 0, G_OPTION_ARG_STRING, &port,
      "Port to listen on (default: " DEFAULT_RTSP_PORT ")", "PORT"},
  {NULL}
};

static void
client_closed(GstRTSPClient *client,
              gpointer user_data)
{
  g_print("client closed %p\n", client);
}

static gboolean
disconnect (GstRTSPClient * client)
{
  g_print("disconnect client %p\n", client);
  gst_rtsp_client_close(client);
  return FALSE;
}

static void
client_connected(GstRTSPServer *server,
                 GstRTSPClient *client,
                 gpointer user_data)
{
  g_print("client connected %p\n", client);
  g_signal_connect(client, "closed", (GCallback)client_closed,
                   NULL);
  g_timeout_add_seconds (8, (GSourceFunc) disconnect, client);
}

int
main (int argc, gchar * argv[])
{
  GMainLoop *loop;
  GstRTSPServer *server;
  GstRTSPMountPoints *mounts;
  GstRTSPMediaFactoryURI *factory;
  GOptionContext *optctx;
  GError *error = NULL;
  gchar *uri;


  guint major;
  guint minor;
  guint micro;
  guint nano;
  gst_version(&major, &minor, &micro, &nano);
  g_print("%s started. GST version %u.%u.%u.%u\n", argv[0], major, minor, micro,     nano);
  optctx = g_option_context_new ("<uri> - Test RTSP Server, URI");
  g_option_context_add_main_entries (optctx, entries, NULL);
  g_option_context_add_group (optctx, gst_init_get_option_group ());
  if (!g_option_context_parse (optctx, &argc, &argv, &error)) {
    g_printerr ("Error parsing options: %s\n", error->message);
    g_option_context_free (optctx);
    g_clear_error (&error);
    return -1;
  }
  g_option_context_free (optctx);

  if (argc < 2) {
    g_printerr ("Please pass an URI or file as argument!\n");
    return -1;
  }

  loop = g_main_loop_new (NULL, FALSE);

  /* create a server instance */
  server = gst_rtsp_server_new ();
  g_object_set (server, "service", port, NULL);
  g_signal_connect(server, "client-connected", (GCallback)client_connected,
                   NULL);
  /* get the mount points for this server, every server has a default object
   * that be used to map uri mount points to media factories */
  mounts = gst_rtsp_server_get_mount_points (server);

  /* make a URI media factory for a test stream. */
  factory = gst_rtsp_media_factory_uri_new ();

  /* check if URI is valid, otherwise convert filename to URI if it's a file */
  if (gst_uri_is_valid (argv[1])) {
    uri = g_strdup (argv[1]);
  } else if (g_file_test (argv[1], G_FILE_TEST_EXISTS)) {
    uri = gst_filename_to_uri (argv[1], NULL);
  } else {
    g_printerr ("Unrecognised command line argument '%s'.\n"
        "Please pass an URI or file as argument!\n", argv[1]);
    return -1;
  }

  gst_rtsp_media_factory_uri_set_uri (factory, uri);
  g_free (uri);

  /* attach the test factory to the /test url */
  gst_rtsp_mount_points_add_factory (mounts, "/test",
      GST_RTSP_MEDIA_FACTORY (factory));

  /* don't need the ref to the mapper anymore */
  g_object_unref (mounts);

  /* attach the server to the default maincontext */
  if (gst_rtsp_server_attach (server, NULL) == 0)
    goto failed;

  /* start serving */
  g_print ("stream ready at rtsp://127.0.0.1:%s/test\n", port);
  g_main_loop_run (loop);

  return 0;

  /* ERRORS */
failed:
  {
    g_print ("failed to attach the server\n");
    return -1;
  }
}

I start the application with: ./close-test-uri.out ~/Videos/big_buck_bunny.mp4 and then I connect to the RTSP stream using a client (VLC): vlc -vvv "rtsp://127.0.0.1:8554/test" The video will start playing and after 8 seconds it will (as expected), be disconnected. However, as mentioned before I get the critical error and the application becomes unresponsive (sometimes it crashes).

Am I using gst_rtsp_client_close() it the wrong way?

Thank you in advance!