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/* Farstream ad-hoc test for the rtp codec discovery
*
* Copyright (C) 2007 Collabora, Nokia
* @author: Olivier Crete <olivier.crete@collabora.co.uk>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <gst/gst.h>
#include <farstream/fs-codec.h>
#include "fs-rtp-discover-codecs.h"
#include "fs-rtp-conference.h"
static void
debug_pipeline (GList *pipeline)
{
GList *walk;
for (walk = pipeline; walk; walk = g_list_next (walk))
{
GList *walk2;
for (walk2 = g_list_first (walk->data); walk2; walk2 = g_list_next (walk2))
g_message ("%p:%d:%s ", walk2->data,
GST_OBJECT_REFCOUNT_VALUE(GST_OBJECT (walk2->data)),
gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (walk2->data)));
g_message ("--");
}
}
static void
debug_blueprint (CodecBlueprint *blueprint)
{
gchar *str;
str = fs_codec_to_string (blueprint->codec);
g_message ("Codec: %s", str);
g_free (str);
str = gst_caps_to_string (blueprint->media_caps);
g_message ("media_caps: %s", str);
g_free (str);
str = gst_caps_to_string (blueprint->rtp_caps);
g_message ("rtp_caps: %s", str);
g_free (str);
g_message ("send pipeline:");
debug_pipeline (blueprint->send_pipeline_factory);
g_message ("recv pipeline:");
debug_pipeline (blueprint->receive_pipeline_factory);
g_message ("================================");
}
int main (int argc, char **argv)
{
GList *elements = NULL;
GError *error = NULL;
gst_init (&argc, &argv);
GST_DEBUG_CATEGORY_INIT (fsrtpconference_debug, "fsrtpconference", 0,
"Farstream RTP Conference Element");
GST_DEBUG_CATEGORY_INIT (fsrtpconference_disco, "fsrtpconference_disco",
0, "Farstream RTP Codec Discovery");
GST_DEBUG_CATEGORY_INIT (fsrtpconference_nego, "fsrtpconference_nego",
0, "Farstream RTP Codec Negotiation");
gst_debug_set_default_threshold (GST_LEVEL_WARNING);
g_message ("AUDIO STARTING!!");
elements = fs_rtp_blueprints_get (FS_MEDIA_TYPE_AUDIO, &error);
if (error)
g_message ("Error: %s", error->message);
else
g_list_foreach (elements, (GFunc) debug_blueprint, NULL);
g_clear_error (&error);
fs_rtp_blueprints_unref (FS_MEDIA_TYPE_AUDIO);
g_message ("AUDIO FINISHED!!");
g_message ("VIDEO STARTING!!");
elements = fs_rtp_blueprints_get (FS_MEDIA_TYPE_VIDEO, &error);
if (error)
g_message ("Error: %s", error->message);
else
g_list_foreach (elements, (GFunc) debug_blueprint, NULL);
g_clear_error (&error);
fs_rtp_blueprints_unref (FS_MEDIA_TYPE_VIDEO);
g_message ("VIDEO FINISHED!!");
return 0;
}
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