gstreamer/tests/check/gst/rtspclientsink.c

226 lines
5.9 KiB
C
Raw Normal View History

/* GStreamer unit test for rtspclientsink
* Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
* @author David Svensson Fors <davidsf at axis dot com>
* Copyright (C) 2015 Centricular Ltd
* @author Tim-Philipp Müller <tim@centricular.com>
* @author Jan Schmidt <jan@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.
*/
#include <gst/check/gstcheck.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <stdio.h>
#include <netinet/in.h>
#include "rtsp-server.h"
#define TEST_MOUNT_POINT "/test"
/* tested rtsp server */
static GstRTSPServer *server = NULL;
/* tcp port that the test server listens for rtsp requests on */
static gint test_port = 0;
/* id of the server's source within the GMainContext */
static guint source_id;
/* iterate the default main context until there are no events to dispatch */
static void
iterate (void)
{
while (g_main_context_iteration (NULL, FALSE)) {
GST_DEBUG ("iteration");
}
}
/* start the testing rtsp server for RECORD mode */
static GstRTSPMediaFactory *
start_record_server (const gchar * launch_line)
{
GstRTSPMediaFactory *factory;
GstRTSPMountPoints *mounts;
gchar *service;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_transport_mode (factory,
GST_RTSP_TRANSPORT_MODE_RECORD);
gst_rtsp_media_factory_set_launch (factory, launch_line);
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
GST_DEBUG ("rtsp server listening on port %d", test_port);
return factory;
}
/* stop the tested rtsp server */
static void
stop_server (void)
{
g_source_remove (source_id);
source_id = 0;
GST_DEBUG ("rtsp server stopped");
}
/* fixture setup function */
static void
setup (void)
{
server = gst_rtsp_server_new ();
}
/* fixture clean-up function */
static void
teardown (void)
{
if (server) {
g_object_unref (server);
server = NULL;
}
test_port = 0;
}
/* create an rtsp connection to the server on test_port */
static gchar *
get_server_uri (gint port, const gchar * mount_point)
{
gchar *address;
gchar *uri_string;
GstRTSPUrl *url = NULL;
address = gst_rtsp_server_get_address (server);
uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
g_free (address);
fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
gst_rtsp_url_free (url);
return uri_string;
}
static void
media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
gpointer user_data)
{
GstElement **p_sink = user_data;
GstElement *bin;
bin = gst_rtsp_media_get_element (media);
*p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink);
}
#define AUDIO_PIPELINE "audiotestsrc num-buffers=%d ! " \
"audio/x-raw,rate=8000 ! alawenc ! rtspclientsink name=sink location=%s"
#define RECORD_N_BUFS 10
GST_START_TEST (test_record)
{
GstRTSPMediaFactory *mfactory;
GstElement *server_sink = NULL;
gint i;
mfactory =
start_record_server ("( rtppcmadepay name=depay0 ! appsink name=sink async=false )");
g_signal_connect (mfactory, "media-constructed",
G_CALLBACK (media_constructed_cb), &server_sink);
/* Create an rtspclientsink and send some data */
{
gchar *uri = get_server_uri (test_port, TEST_MOUNT_POINT);
gchar *pipe_str;
GstMessage *msg;
GstElement *pipeline;
GstBus *bus;
pipe_str = g_strdup_printf (AUDIO_PIPELINE, RECORD_N_BUFS, uri);
g_free (uri);
pipeline = gst_parse_launch (pipe_str, NULL);
g_free (pipe_str);
fail_unless (pipeline != NULL);
bus = gst_element_get_bus (pipeline);
fail_if (bus == NULL);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
msg = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
fail_if (GST_MESSAGE_TYPE (msg) != GST_MESSAGE_EOS);
gst_message_unref (msg);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
iterate ();
/* check received data (we assume every buffer created by audiotestsrc and
* subsequently encoded by mulawenc results in exactly one RTP packet) */
for (i = 0; i < RECORD_N_BUFS; ++i) {
GstSample *sample = NULL;
g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample);
GST_INFO ("%2d recv sample: %p", i, sample);
if (sample)
gst_sample_unref (sample);
}
/* clean up and iterate so the clean-up can finish */
stop_server ();
iterate ();
}
GST_END_TEST;
static Suite *
rtspclientsink_suite (void)
{
Suite *s = suite_create ("rtspclientsink");
TCase *tc = tcase_create ("general");
suite_add_tcase (s, tc);
tcase_add_checked_fixture (tc, setup, teardown);
tcase_set_timeout (tc, 120);
tcase_add_test (tc, test_record);
return s;
}
GST_CHECK_MAIN (rtspclientsink);