mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-13 12:51:16 +00:00
a6367c5971
Removed port allocation test from the media suite. The port allocation failure is now in the stream suite. rtspserver: Make sure that the media is suspended after the DESCRIBE request before reconfiguring the UDP sinks. rtspclientsink: In the RECORD case we have to set async property to false for the appsink element in the test in order to make sure that the media pipeline doesn't hang in start_preroll(). https://bugzilla.gnome.org/show_bug.cgi?id=757488
1934 lines
57 KiB
C
1934 lines
57 KiB
C
/* GStreamer unit test for GstRTSPServer
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* Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
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* @author David Svensson Fors <davidsf at axis dot com>
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* Copyright (C) 2015 Centricular Ltd
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* @author Tim-Philipp Müller <tim@centricular.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/check/gstcheck.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <stdio.h>
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#include <netinet/in.h>
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#include "rtsp-server.h"
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#define VIDEO_PIPELINE "videotestsrc ! " \
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"video/x-raw,width=352,height=288 ! " \
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"rtpgstpay name=pay0 pt=96"
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#define AUDIO_PIPELINE "audiotestsrc ! " \
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"audio/x-raw,rate=8000 ! " \
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"rtpgstpay name=pay1 pt=97"
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#define TEST_MOUNT_POINT "/test"
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#define TEST_PROTO "RTP/AVP"
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#define TEST_ENCODING "X-GST"
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#define TEST_CLOCK_RATE "90000"
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/* tested rtsp server */
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static GstRTSPServer *server = NULL;
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/* tcp port that the test server listens for rtsp requests on */
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static gint test_port = 0;
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/* id of the server's source within the GMainContext */
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static guint source_id;
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/* iterate the default main loop until there are no events to dispatch */
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static void
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iterate (void)
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{
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while (g_main_context_iteration (NULL, FALSE)) {
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GST_DEBUG ("iteration");
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}
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}
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static void
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get_client_ports_full (GstRTSPRange * range, GSocket ** rtp_socket,
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GSocket ** rtcp_socket)
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{
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GSocket *rtp = NULL;
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GSocket *rtcp = NULL;
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gint rtp_port = 0;
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gint rtcp_port;
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GInetAddress *anyaddr = g_inet_address_new_any (G_SOCKET_FAMILY_IPV4);
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GSocketAddress *sockaddr;
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gboolean bound;
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for (;;) {
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if (rtp_port != 0)
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rtp_port += 2;
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rtp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
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G_SOCKET_PROTOCOL_UDP, NULL);
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fail_unless (rtp != NULL);
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sockaddr = g_inet_socket_address_new (anyaddr, rtp_port);
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fail_unless (sockaddr != NULL);
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bound = g_socket_bind (rtp, sockaddr, FALSE, NULL);
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g_object_unref (sockaddr);
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if (!bound) {
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g_object_unref (rtp);
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continue;
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}
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sockaddr = g_socket_get_local_address (rtp, NULL);
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fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
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rtp_port =
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g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
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g_object_unref (sockaddr);
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if (rtp_port % 2 != 0) {
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rtp_port += 1;
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g_object_unref (rtp);
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continue;
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}
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rtcp_port = rtp_port + 1;
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rtcp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
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G_SOCKET_PROTOCOL_UDP, NULL);
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fail_unless (rtcp != NULL);
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sockaddr = g_inet_socket_address_new (anyaddr, rtcp_port);
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fail_unless (sockaddr != NULL);
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bound = g_socket_bind (rtcp, sockaddr, FALSE, NULL);
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g_object_unref (sockaddr);
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if (!bound) {
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g_object_unref (rtp);
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g_object_unref (rtcp);
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continue;
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}
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sockaddr = g_socket_get_local_address (rtcp, NULL);
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fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
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fail_unless (rtcp_port ==
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g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr)));
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g_object_unref (sockaddr);
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break;
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}
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range->min = rtp_port;
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range->max = rtcp_port;
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if (rtp_socket)
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*rtp_socket = rtp;
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else
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g_object_unref (rtp);
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if (rtcp_socket)
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*rtcp_socket = rtcp;
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else
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g_object_unref (rtcp);
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GST_DEBUG ("client_port=%d-%d", range->min, range->max);
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g_object_unref (anyaddr);
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}
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/* get a free rtp/rtcp client port pair */
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static void
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get_client_ports (GstRTSPRange * range)
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{
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get_client_ports_full (range, NULL, NULL);
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}
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/* start the tested rtsp server */
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static void
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start_server (void)
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{
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GstRTSPMountPoints *mounts;
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gchar *service;
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GstRTSPMediaFactory *factory;
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GstRTSPAddressPool *pool;
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mounts = gst_rtsp_server_get_mount_points (server);
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factory = gst_rtsp_media_factory_new ();
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gst_rtsp_media_factory_set_launch (factory,
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"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
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gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
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g_object_unref (mounts);
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/* use an address pool for multicast */
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pool = gst_rtsp_address_pool_new ();
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gst_rtsp_address_pool_add_range (pool,
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"224.3.0.0", "224.3.0.10", 5000, 5010, 16);
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gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
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GST_RTSP_ADDRESS_POOL_ANY_IPV4, 6000, 6010, 0);
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gst_rtsp_media_factory_set_address_pool (factory, pool);
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gst_object_unref (pool);
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/* set port to any */
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gst_rtsp_server_set_service (server, "0");
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/* attach to default main context */
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source_id = gst_rtsp_server_attach (server, NULL);
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fail_if (source_id == 0);
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/* get port */
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service = gst_rtsp_server_get_service (server);
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test_port = atoi (service);
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fail_unless (test_port != 0);
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g_free (service);
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GST_DEBUG ("rtsp server listening on port %d", test_port);
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}
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/* start the testing rtsp server for RECORD mode */
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static GstRTSPMediaFactory *
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start_record_server (const gchar * launch_line)
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{
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GstRTSPMediaFactory *factory;
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GstRTSPMountPoints *mounts;
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gchar *service;
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mounts = gst_rtsp_server_get_mount_points (server);
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factory = gst_rtsp_media_factory_new ();
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gst_rtsp_media_factory_set_transport_mode (factory,
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GST_RTSP_TRANSPORT_MODE_RECORD);
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gst_rtsp_media_factory_set_launch (factory, launch_line);
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gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
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g_object_unref (mounts);
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/* set port to any */
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gst_rtsp_server_set_service (server, "0");
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/* attach to default main context */
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source_id = gst_rtsp_server_attach (server, NULL);
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fail_if (source_id == 0);
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/* get port */
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service = gst_rtsp_server_get_service (server);
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test_port = atoi (service);
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fail_unless (test_port != 0);
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g_free (service);
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GST_DEBUG ("rtsp server listening on port %d", test_port);
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return factory;
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}
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/* stop the tested rtsp server */
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static void
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stop_server (void)
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{
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g_source_remove (source_id);
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source_id = 0;
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GST_DEBUG ("rtsp server stopped");
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}
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/* create an rtsp connection to the server on test_port */
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static GstRTSPConnection *
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connect_to_server (gint port, const gchar * mount_point)
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{
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GstRTSPConnection *conn = NULL;
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gchar *address;
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gchar *uri_string;
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GstRTSPUrl *url = NULL;
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address = gst_rtsp_server_get_address (server);
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uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
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g_free (address);
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fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
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g_free (uri_string);
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fail_unless (gst_rtsp_connection_create (url, &conn) == GST_RTSP_OK);
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gst_rtsp_url_free (url);
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fail_unless (gst_rtsp_connection_connect (conn, NULL) == GST_RTSP_OK);
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return conn;
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}
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/* create an rtsp request */
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static GstRTSPMessage *
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create_request (GstRTSPConnection * conn, GstRTSPMethod method,
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const gchar * control)
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{
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GstRTSPMessage *request = NULL;
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gchar *base_uri;
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gchar *full_uri;
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base_uri = gst_rtsp_url_get_request_uri (gst_rtsp_connection_get_url (conn));
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full_uri = g_strdup_printf ("%s/%s", base_uri, control ? control : "");
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g_free (base_uri);
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if (gst_rtsp_message_new_request (&request, method, full_uri) != GST_RTSP_OK) {
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GST_DEBUG ("failed to create request object");
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g_free (full_uri);
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return NULL;
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}
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g_free (full_uri);
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return request;
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}
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/* send an rtsp request */
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static gboolean
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send_request (GstRTSPConnection * conn, GstRTSPMessage * request)
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{
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if (gst_rtsp_connection_send (conn, request, NULL) != GST_RTSP_OK) {
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GST_DEBUG ("failed to send request");
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return FALSE;
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}
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return TRUE;
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}
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/* read rtsp response. response must be freed by the caller */
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static GstRTSPMessage *
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read_response (GstRTSPConnection * conn)
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{
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GstRTSPMessage *response = NULL;
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if (gst_rtsp_message_new (&response) != GST_RTSP_OK) {
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GST_DEBUG ("failed to create response object");
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return NULL;
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}
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if (gst_rtsp_connection_receive (conn, response, NULL) != GST_RTSP_OK) {
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GST_DEBUG ("failed to read response");
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gst_rtsp_message_free (response);
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return NULL;
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}
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fail_unless (gst_rtsp_message_get_type (response) ==
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GST_RTSP_MESSAGE_RESPONSE);
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return response;
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}
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/* send an rtsp request and receive response. gchar** parameters are out
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* parameters that have to be freed by the caller */
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static GstRTSPStatusCode
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do_request_full (GstRTSPConnection * conn, GstRTSPMethod method,
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const gchar * control, const gchar * session_in, const gchar * transport_in,
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const gchar * range_in, const gchar * require_in,
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gchar ** content_type, gchar ** content_base, gchar ** body,
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gchar ** session_out, gchar ** transport_out, gchar ** range_out,
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gchar ** unsupported_out)
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{
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GstRTSPMessage *request;
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GstRTSPMessage *response;
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GstRTSPStatusCode code;
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gchar *value;
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/* create request */
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request = create_request (conn, method, control);
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/* add headers */
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if (session_in) {
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session_in);
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}
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if (transport_in) {
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT, transport_in);
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}
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if (range_in) {
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_RANGE, range_in);
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}
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if (require_in) {
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gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE, require_in);
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}
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/* send request */
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fail_unless (send_request (conn, request));
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gst_rtsp_message_free (request);
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iterate ();
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/* read response */
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response = read_response (conn);
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/* check status line */
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gst_rtsp_message_parse_response (response, &code, NULL, NULL);
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if (code != GST_RTSP_STS_OK) {
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if (unsupported_out != NULL && code == GST_RTSP_STS_OPTION_NOT_SUPPORTED) {
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gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED,
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&value, 0);
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*unsupported_out = g_strdup (value);
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}
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gst_rtsp_message_free (response);
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return code;
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}
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/* get information from response */
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if (content_type) {
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gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_TYPE,
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&value, 0);
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*content_type = g_strdup (value);
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}
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if (content_base) {
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gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
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&value, 0);
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*content_base = g_strdup (value);
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}
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if (body) {
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*body = g_malloc (response->body_size + 1);
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strncpy (*body, (gchar *) response->body, response->body_size);
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}
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if (session_out) {
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gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &value, 0);
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value = g_strdup (value);
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/* Remove the timeout */
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if (value) {
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char *pos = strchr (value, ';');
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if (pos)
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*pos = 0;
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}
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if (session_in) {
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/* check that we got the same session back */
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fail_unless (!g_strcmp0 (value, session_in));
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}
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*session_out = value;
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}
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if (transport_out) {
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gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &value, 0);
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*transport_out = g_strdup (value);
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}
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if (range_out) {
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gst_rtsp_message_get_header (response, GST_RTSP_HDR_RANGE, &value, 0);
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*range_out = g_strdup (value);
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}
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gst_rtsp_message_free (response);
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return code;
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}
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/* send an rtsp request and receive response. gchar** parameters are out
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* parameters that have to be freed by the caller */
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static GstRTSPStatusCode
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do_request (GstRTSPConnection * conn, GstRTSPMethod method,
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const gchar * control, const gchar * session_in,
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const gchar * transport_in, const gchar * range_in,
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gchar ** content_type, gchar ** content_base, gchar ** body,
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gchar ** session_out, gchar ** transport_out, gchar ** range_out)
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{
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return do_request_full (conn, method, control, session_in, transport_in,
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range_in, NULL, content_type, content_base, body, session_out,
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transport_out, range_out, NULL);
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}
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/* send an rtsp request with a method and a session, and receive response */
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static GstRTSPStatusCode
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do_simple_request (GstRTSPConnection * conn, GstRTSPMethod method,
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const gchar * session)
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{
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return do_request (conn, method, NULL, session, NULL, NULL, NULL,
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NULL, NULL, NULL, NULL, NULL);
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}
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/* send a DESCRIBE request and receive response. returns a received
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* GstSDPMessage that must be freed by the caller */
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static GstSDPMessage *
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do_describe (GstRTSPConnection * conn, const gchar * mount_point)
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{
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GstSDPMessage *sdp_message;
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gchar *content_type = NULL;
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gchar *content_base = NULL;
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gchar *body = NULL;
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gchar *address;
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gchar *expected_content_base;
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/* send DESCRIBE request */
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fail_unless (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL, NULL,
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&content_type, &content_base, &body, NULL, NULL, NULL) ==
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GST_RTSP_STS_OK);
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/* check response values */
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fail_unless (!g_strcmp0 (content_type, "application/sdp"));
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address = gst_rtsp_server_get_address (server);
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expected_content_base =
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g_strdup_printf ("rtsp://%s:%d%s/", address, test_port, mount_point);
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fail_unless (!g_strcmp0 (content_base, expected_content_base));
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/* create sdp message */
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fail_unless (gst_sdp_message_new (&sdp_message) == GST_SDP_OK);
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fail_unless (gst_sdp_message_parse_buffer ((guint8 *) body,
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strlen (body), sdp_message) == GST_SDP_OK);
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/* clean up */
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g_free (content_type);
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g_free (content_base);
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g_free (body);
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g_free (address);
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g_free (expected_content_base);
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return sdp_message;
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}
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/* send a SETUP request and receive response. if *session is not NULL,
|
|
* it is used in the request. otherwise, *session is set to a returned
|
|
* session string that must be freed by the caller. the returned
|
|
* transport must be freed by the caller. */
|
|
static GstRTSPStatusCode
|
|
do_setup_full (GstRTSPConnection * conn, const gchar * control,
|
|
GstRTSPLowerTrans lower_transport, const GstRTSPRange * client_ports,
|
|
const gchar * require, gchar ** session, GstRTSPTransport ** transport,
|
|
gchar ** unsupported)
|
|
{
|
|
GstRTSPStatusCode code;
|
|
gchar *session_in = NULL;
|
|
GString *transport_string_in = NULL;
|
|
gchar **session_out = NULL;
|
|
gchar *transport_string_out = NULL;
|
|
|
|
/* prepare and send SETUP request */
|
|
if (session) {
|
|
if (*session) {
|
|
session_in = *session;
|
|
} else {
|
|
session_out = session;
|
|
}
|
|
}
|
|
|
|
transport_string_in = g_string_new (TEST_PROTO);
|
|
switch (lower_transport) {
|
|
case GST_RTSP_LOWER_TRANS_UDP:
|
|
transport_string_in =
|
|
g_string_append (transport_string_in, "/UDP;unicast");
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
|
|
transport_string_in =
|
|
g_string_append (transport_string_in, "/UDP;multicast");
|
|
break;
|
|
case GST_RTSP_LOWER_TRANS_TCP:
|
|
transport_string_in =
|
|
g_string_append (transport_string_in, "/TCP;unicast");
|
|
break;
|
|
default:
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
|
|
if (client_ports) {
|
|
g_string_append_printf (transport_string_in, ";client_port=%d-%d",
|
|
client_ports->min, client_ports->max);
|
|
}
|
|
|
|
code =
|
|
do_request_full (conn, GST_RTSP_SETUP, control, session_in,
|
|
transport_string_in->str, NULL, require, NULL, NULL, NULL, session_out,
|
|
&transport_string_out, NULL, unsupported);
|
|
g_string_free (transport_string_in, TRUE);
|
|
|
|
if (transport_string_out) {
|
|
/* create transport */
|
|
fail_unless (gst_rtsp_transport_new (transport) == GST_RTSP_OK);
|
|
fail_unless (gst_rtsp_transport_parse (transport_string_out,
|
|
*transport) == GST_RTSP_OK);
|
|
g_free (transport_string_out);
|
|
}
|
|
GST_INFO ("code=%d", code);
|
|
return code;
|
|
}
|
|
|
|
/* send a SETUP request and receive response. if *session is not NULL,
|
|
* it is used in the request. otherwise, *session is set to a returned
|
|
* session string that must be freed by the caller. the returned
|
|
* transport must be freed by the caller. */
|
|
static GstRTSPStatusCode
|
|
do_setup (GstRTSPConnection * conn, const gchar * control,
|
|
const GstRTSPRange * client_ports, gchar ** session,
|
|
GstRTSPTransport ** transport)
|
|
{
|
|
return do_setup_full (conn, control, GST_RTSP_LOWER_TRANS_UDP, client_ports,
|
|
NULL, session, transport, NULL);
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
|
|
GST_START_TEST (test_connect)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
|
|
start_server ();
|
|
|
|
/* connect to server */
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
/* clean up */
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
|
|
/* iterate so the clean-up can finish */
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_describe)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
gint32 format;
|
|
gchar *expected_rtpmap;
|
|
const gchar *rtpmap;
|
|
const gchar *control_video;
|
|
const gchar *control_audio;
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
/* send DESCRIBE request */
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
|
|
/* check video sdp */
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
|
|
fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
|
|
sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
|
|
&format);
|
|
expected_rtpmap =
|
|
g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
|
|
rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
|
|
fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
|
|
g_free (expected_rtpmap);
|
|
control_video = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
fail_unless (!g_strcmp0 (control_video, "stream=0"));
|
|
|
|
/* check audio sdp */
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
|
|
fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
|
|
fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
|
|
sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
|
|
&format);
|
|
expected_rtpmap =
|
|
g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
|
|
rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
|
|
fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
|
|
g_free (expected_rtpmap);
|
|
control_audio = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
fail_unless (!g_strcmp0 (control_audio, "stream=1"));
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_describe_record_media)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
|
|
start_record_server ("( fakesink name=depay0 )");
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
/* send DESCRIBE request */
|
|
fail_unless_equals_int (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL,
|
|
NULL, NULL, NULL, NULL, NULL, NULL, NULL),
|
|
GST_RTSP_STS_METHOD_NOT_ALLOWED);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_describe_non_existing_mount_point)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
|
|
start_server ();
|
|
|
|
/* send DESCRIBE request for a non-existing mount point
|
|
* and check that we get a 404 Not Found */
|
|
conn = connect_to_server (test_port, "/non-existing");
|
|
fail_unless (do_simple_request (conn, GST_RTSP_DESCRIBE, NULL)
|
|
== GST_RTSP_STS_NOT_FOUND);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
do_test_setup (GstRTSPLowerTrans lower_transport)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
const gchar *audio_control;
|
|
GstRTSPRange client_ports = { 0 };
|
|
gchar *session = NULL;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
GstRTSPTransport *audio_transport = NULL;
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get control strings from DESCRIBE response */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
|
|
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports (&client_ports);
|
|
|
|
/* send SETUP request for video */
|
|
fail_unless (do_setup_full (conn, video_control, lower_transport,
|
|
&client_ports, NULL, &session, &video_transport,
|
|
NULL) == GST_RTSP_STS_OK);
|
|
GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
|
|
|
|
/* check response from SETUP */
|
|
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
|
|
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
|
|
fail_unless (video_transport->lower_transport == lower_transport);
|
|
fail_unless (video_transport->mode_play);
|
|
gst_rtsp_transport_free (video_transport);
|
|
|
|
/* send SETUP request for audio */
|
|
fail_unless (do_setup_full (conn, audio_control, lower_transport,
|
|
&client_ports, NULL, &session, &audio_transport,
|
|
NULL) == GST_RTSP_STS_OK);
|
|
GST_DEBUG ("set up audio %s with session '%s'", audio_control, session);
|
|
|
|
/* check response from SETUP */
|
|
fail_unless (audio_transport->trans == GST_RTSP_TRANS_RTP);
|
|
fail_unless (audio_transport->profile == GST_RTSP_PROFILE_AVP);
|
|
fail_unless (audio_transport->lower_transport == lower_transport);
|
|
fail_unless (audio_transport->mode_play);
|
|
gst_rtsp_transport_free (audio_transport);
|
|
|
|
/* send TEARDOWN request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session) == GST_RTSP_STS_OK);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
g_free (session);
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_START_TEST (test_setup_udp)
|
|
{
|
|
do_test_setup (GST_RTSP_LOWER_TRANS_UDP);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_setup_tcp)
|
|
{
|
|
do_test_setup (GST_RTSP_LOWER_TRANS_TCP);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_setup_udp_mcast)
|
|
{
|
|
do_test_setup (GST_RTSP_LOWER_TRANS_UDP_MCAST);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_setup_twice)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
GstRTSPRange client_ports;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
gchar *session1 = NULL;
|
|
gchar *session2 = NULL;
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
/* we wan't more than one session for this connection */
|
|
gst_rtsp_connection_set_remember_session_id (conn, FALSE);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get the control url */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports (&client_ports);
|
|
|
|
/* send SETUP request for one session */
|
|
fail_unless (do_setup (conn, video_control, &client_ports, &session1,
|
|
&video_transport) == GST_RTSP_STS_OK);
|
|
GST_DEBUG ("set up video %s, got session '%s'", video_control, session1);
|
|
|
|
/* check response from SETUP */
|
|
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
|
|
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
|
|
fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
|
|
fail_unless (video_transport->mode_play);
|
|
gst_rtsp_transport_free (video_transport);
|
|
|
|
/* send SETUP request for another session */
|
|
fail_unless (do_setup (conn, video_control, &client_ports, &session2,
|
|
&video_transport) == GST_RTSP_STS_OK);
|
|
GST_DEBUG ("set up video %s, got session '%s'", video_control, session2);
|
|
|
|
/* check response from SETUP */
|
|
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
|
|
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
|
|
fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
|
|
fail_unless (video_transport->mode_play);
|
|
gst_rtsp_transport_free (video_transport);
|
|
|
|
/* session can not be the same */
|
|
fail_unless (strcmp (session1, session2));
|
|
|
|
/* send TEARDOWN request for the first session */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session1) == GST_RTSP_STS_OK);
|
|
|
|
/* send TEARDOWN request for the second session */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session2) == GST_RTSP_STS_OK);
|
|
|
|
g_free (session1);
|
|
g_free (session2);
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_setup_with_require_header)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
GstRTSPRange client_ports;
|
|
gchar *session = NULL;
|
|
gchar *unsupported = NULL;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get control strings from DESCRIBE response */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports (&client_ports);
|
|
|
|
/* send SETUP request for video, with single Require header */
|
|
fail_unless_equals_int (do_setup_full (conn, video_control,
|
|
GST_RTSP_LOWER_TRANS_UDP, &client_ports, "funky-feature", &session,
|
|
&video_transport, &unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
|
|
fail_unless_equals_string (unsupported, "funky-feature");
|
|
g_free (unsupported);
|
|
unsupported = NULL;
|
|
|
|
/* send SETUP request for video, with multiple Require headers */
|
|
fail_unless_equals_int (do_setup_full (conn, video_control,
|
|
GST_RTSP_LOWER_TRANS_UDP, &client_ports,
|
|
"funky-feature, foo-bar, superburst", &session, &video_transport,
|
|
&unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
|
|
fail_unless_equals_string (unsupported, "funky-feature, foo-bar, superburst");
|
|
g_free (unsupported);
|
|
unsupported = NULL;
|
|
|
|
/* ok, just do a normal setup then (make sure that still works) */
|
|
fail_unless_equals_int (do_setup (conn, video_control, &client_ports,
|
|
&session, &video_transport), GST_RTSP_STS_OK);
|
|
|
|
GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
|
|
|
|
/* check response from SETUP */
|
|
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
|
|
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
|
|
fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
|
|
fail_unless (video_transport->mode_play);
|
|
gst_rtsp_transport_free (video_transport);
|
|
|
|
/* send TEARDOWN request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session) == GST_RTSP_STS_OK);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
g_free (session);
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_setup_non_existing_stream)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstRTSPRange client_ports;
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
get_client_ports (&client_ports);
|
|
|
|
/* send SETUP request with a non-existing stream and check that we get a
|
|
* 404 Not Found */
|
|
fail_unless (do_setup (conn, "stream=7", &client_ports, NULL,
|
|
NULL) == GST_RTSP_STS_NOT_FOUND);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static void
|
|
receive_rtp (GSocket * socket, GSocketAddress ** addr)
|
|
{
|
|
GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
|
|
|
|
for (;;) {
|
|
gssize bytes;
|
|
GstMapInfo map = GST_MAP_INFO_INIT;
|
|
GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
|
|
bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
|
|
map.maxsize, NULL, NULL);
|
|
fail_unless (bytes > 0);
|
|
gst_buffer_unmap (buffer, &map);
|
|
gst_buffer_set_size (buffer, bytes);
|
|
|
|
if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer)) {
|
|
gst_rtp_buffer_unmap (&rtpbuffer);
|
|
break;
|
|
}
|
|
|
|
if (addr)
|
|
g_clear_object (addr);
|
|
}
|
|
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
static void
|
|
receive_rtcp (GSocket * socket, GSocketAddress ** addr, GstRTCPType type)
|
|
{
|
|
GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
|
|
|
|
for (;;) {
|
|
gssize bytes;
|
|
GstMapInfo map = GST_MAP_INFO_INIT;
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
|
|
bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
|
|
map.maxsize, NULL, NULL);
|
|
fail_unless (bytes > 0);
|
|
gst_buffer_unmap (buffer, &map);
|
|
gst_buffer_set_size (buffer, bytes);
|
|
|
|
if (gst_rtcp_buffer_validate (buffer)) {
|
|
GstRTCPBuffer rtcpbuffer = GST_RTCP_BUFFER_INIT;
|
|
GstRTCPPacket packet;
|
|
|
|
if (type) {
|
|
fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcpbuffer));
|
|
fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuffer, &packet));
|
|
do {
|
|
if (gst_rtcp_packet_get_type (&packet) == type) {
|
|
gst_rtcp_buffer_unmap (&rtcpbuffer);
|
|
goto done;
|
|
}
|
|
} while (gst_rtcp_packet_move_to_next (&packet));
|
|
gst_rtcp_buffer_unmap (&rtcpbuffer);
|
|
} else {
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (addr)
|
|
g_clear_object (addr);
|
|
}
|
|
|
|
done:
|
|
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
|
|
static void
|
|
do_test_play_full (const gchar * range, GstRTSPLowerTrans lower_transport)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
const gchar *audio_control;
|
|
GstRTSPRange client_port;
|
|
gchar *session = NULL;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
GstRTSPTransport *audio_transport = NULL;
|
|
GSocket *rtp_socket, *rtcp_socket;
|
|
gchar *range_out = NULL;
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get control strings from DESCRIBE response */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
|
|
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
|
|
|
|
/* do SETUP for video and audio */
|
|
fail_unless (do_setup_full (conn, video_control, lower_transport,
|
|
&client_port, NULL, &session, &video_transport,
|
|
NULL) == GST_RTSP_STS_OK);
|
|
fail_unless (do_setup_full (conn, audio_control, lower_transport,
|
|
&client_port, NULL, &session, &audio_transport,
|
|
NULL) == GST_RTSP_STS_OK);
|
|
|
|
/* send PLAY request and check that we get 200 OK */
|
|
fail_unless (do_request (conn, GST_RTSP_PLAY, NULL, session, NULL, range,
|
|
NULL, NULL, NULL, NULL, NULL, &range_out) == GST_RTSP_STS_OK);
|
|
if (range)
|
|
fail_unless_equals_string (range, range_out);
|
|
g_free (range_out);
|
|
|
|
receive_rtp (rtp_socket, NULL);
|
|
receive_rtcp (rtcp_socket, NULL, 0);
|
|
|
|
/* send TEARDOWN request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session) == GST_RTSP_STS_OK);
|
|
|
|
/* FIXME: The rtsp-server always disconnects the transport before
|
|
* sending the RTCP BYE
|
|
* receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
|
|
*/
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
g_object_unref (rtp_socket);
|
|
g_object_unref (rtcp_socket);
|
|
g_free (session);
|
|
gst_rtsp_transport_free (video_transport);
|
|
gst_rtsp_transport_free (audio_transport);
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
}
|
|
|
|
static void
|
|
do_test_play (const gchar * range)
|
|
{
|
|
do_test_play_full (range, GST_RTSP_LOWER_TRANS_UDP);
|
|
}
|
|
|
|
GST_START_TEST (test_play)
|
|
{
|
|
start_server ();
|
|
|
|
do_test_play (NULL);
|
|
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_play_without_session)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
/* send PLAY request without a session and check that we get a
|
|
* 454 Session Not Found */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
|
|
NULL) == GST_RTSP_STS_SESSION_NOT_FOUND);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_bind_already_in_use)
|
|
{
|
|
GstRTSPServer *serv;
|
|
GSocketService *service;
|
|
GError *error = NULL;
|
|
guint16 port;
|
|
gchar *port_str;
|
|
|
|
serv = gst_rtsp_server_new ();
|
|
service = g_socket_service_new ();
|
|
|
|
/* bind service to port */
|
|
port =
|
|
g_socket_listener_add_any_inet_port (G_SOCKET_LISTENER (service), NULL,
|
|
&error);
|
|
g_assert_no_error (error);
|
|
|
|
port_str = g_strdup_printf ("%d\n", port);
|
|
|
|
/* try to bind server to the same port */
|
|
g_object_set (serv, "service", port_str, NULL);
|
|
g_free (port_str);
|
|
|
|
/* attach to default main context */
|
|
fail_unless (gst_rtsp_server_attach (serv, NULL) == 0);
|
|
|
|
/* cleanup */
|
|
g_object_unref (serv);
|
|
g_socket_service_stop (service);
|
|
g_object_unref (service);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
GST_START_TEST (test_play_multithreaded)
|
|
{
|
|
GstRTSPThreadPool *pool;
|
|
|
|
pool = gst_rtsp_server_get_thread_pool (server);
|
|
gst_rtsp_thread_pool_set_max_threads (pool, 2);
|
|
g_object_unref (pool);
|
|
|
|
start_server ();
|
|
|
|
do_test_play (NULL);
|
|
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
enum
|
|
{
|
|
BLOCK_ME,
|
|
BLOCKED,
|
|
UNBLOCK
|
|
};
|
|
|
|
|
|
static void
|
|
media_constructed_block (GstRTSPMediaFactory * factory,
|
|
GstRTSPMedia * media, gpointer user_data)
|
|
{
|
|
gint *block_state = user_data;
|
|
|
|
g_mutex_lock (&check_mutex);
|
|
|
|
*block_state = BLOCKED;
|
|
g_cond_broadcast (&check_cond);
|
|
|
|
while (*block_state != UNBLOCK)
|
|
g_cond_wait (&check_cond, &check_mutex);
|
|
g_mutex_unlock (&check_mutex);
|
|
}
|
|
|
|
|
|
GST_START_TEST (test_play_multithreaded_block_in_describe)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstRTSPMountPoints *mounts;
|
|
GstRTSPMediaFactory *factory;
|
|
gint block_state = BLOCK_ME;
|
|
GstRTSPMessage *request;
|
|
GstRTSPMessage *response;
|
|
GstRTSPStatusCode code;
|
|
GstRTSPThreadPool *pool;
|
|
|
|
pool = gst_rtsp_server_get_thread_pool (server);
|
|
gst_rtsp_thread_pool_set_max_threads (pool, 2);
|
|
g_object_unref (pool);
|
|
|
|
mounts = gst_rtsp_server_get_mount_points (server);
|
|
fail_unless (mounts != NULL);
|
|
factory = gst_rtsp_media_factory_new ();
|
|
gst_rtsp_media_factory_set_launch (factory,
|
|
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
|
|
g_signal_connect (factory, "media-constructed",
|
|
G_CALLBACK (media_constructed_block), &block_state);
|
|
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT "2", factory);
|
|
g_object_unref (mounts);
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT "2");
|
|
iterate ();
|
|
|
|
/* do describe, it will not return now as we've blocked it */
|
|
request = create_request (conn, GST_RTSP_DESCRIBE, NULL);
|
|
fail_unless (send_request (conn, request));
|
|
gst_rtsp_message_free (request);
|
|
|
|
g_mutex_lock (&check_mutex);
|
|
while (block_state != BLOCKED)
|
|
g_cond_wait (&check_cond, &check_mutex);
|
|
g_mutex_unlock (&check_mutex);
|
|
|
|
/* Do a second connection while the first one is blocked */
|
|
do_test_play (NULL);
|
|
|
|
/* Now unblock the describe */
|
|
g_mutex_lock (&check_mutex);
|
|
block_state = UNBLOCK;
|
|
g_cond_broadcast (&check_cond);
|
|
g_mutex_unlock (&check_mutex);
|
|
|
|
response = read_response (conn);
|
|
gst_rtsp_message_parse_response (response, &code, NULL, NULL);
|
|
fail_unless (code == GST_RTSP_STS_OK);
|
|
gst_rtsp_message_free (response);
|
|
|
|
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
static void
|
|
new_session_timeout_one (GstRTSPClient * client,
|
|
GstRTSPSession * session, gpointer user_data)
|
|
{
|
|
gst_rtsp_session_set_timeout (session, 1);
|
|
|
|
g_signal_handlers_disconnect_by_func (client, new_session_timeout_one,
|
|
user_data);
|
|
}
|
|
|
|
static void
|
|
session_connected_new_session_cb (GstRTSPServer * server,
|
|
GstRTSPClient * client, gpointer user_data)
|
|
{
|
|
|
|
g_signal_connect (client, "new-session", user_data, NULL);
|
|
}
|
|
|
|
GST_START_TEST (test_play_multithreaded_timeout_client)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
const gchar *audio_control;
|
|
GstRTSPRange client_port;
|
|
gchar *session = NULL;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
GstRTSPTransport *audio_transport = NULL;
|
|
GstRTSPSessionPool *pool;
|
|
GstRTSPThreadPool *thread_pool;
|
|
|
|
thread_pool = gst_rtsp_server_get_thread_pool (server);
|
|
gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
|
|
g_object_unref (thread_pool);
|
|
|
|
pool = gst_rtsp_server_get_session_pool (server);
|
|
g_signal_connect (server, "client-connected",
|
|
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
|
|
|
|
start_server ();
|
|
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get control strings from DESCRIBE response */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
|
|
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports (&client_port);
|
|
|
|
/* do SETUP for video and audio */
|
|
fail_unless (do_setup_full (conn, video_control, GST_RTSP_LOWER_TRANS_UDP,
|
|
&client_port, NULL, &session, &video_transport,
|
|
NULL) == GST_RTSP_STS_OK);
|
|
fail_unless (do_setup_full (conn, audio_control, GST_RTSP_LOWER_TRANS_UDP,
|
|
&client_port, NULL, &session, &audio_transport,
|
|
NULL) == GST_RTSP_STS_OK);
|
|
|
|
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
|
|
|
|
/* send PLAY request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
|
|
session) == GST_RTSP_STS_OK);
|
|
|
|
sleep (7);
|
|
|
|
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
|
|
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 0);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
g_object_unref (pool);
|
|
g_free (session);
|
|
gst_rtsp_transport_free (video_transport);
|
|
gst_rtsp_transport_free (audio_transport);
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
GST_START_TEST (test_play_multithreaded_timeout_session)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
const gchar *audio_control;
|
|
GstRTSPRange client_port;
|
|
gchar *session1 = NULL;
|
|
gchar *session2 = NULL;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
GstRTSPTransport *audio_transport = NULL;
|
|
GstRTSPSessionPool *pool;
|
|
GstRTSPThreadPool *thread_pool;
|
|
|
|
thread_pool = gst_rtsp_server_get_thread_pool (server);
|
|
gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
|
|
g_object_unref (thread_pool);
|
|
|
|
pool = gst_rtsp_server_get_session_pool (server);
|
|
g_signal_connect (server, "client-connected",
|
|
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
|
|
|
|
start_server ();
|
|
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
gst_rtsp_connection_set_remember_session_id (conn, FALSE);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get control strings from DESCRIBE response */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
|
|
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports (&client_port);
|
|
|
|
/* do SETUP for video and audio */
|
|
fail_unless (do_setup (conn, video_control, &client_port, &session1,
|
|
&video_transport) == GST_RTSP_STS_OK);
|
|
fail_unless (do_setup (conn, audio_control, &client_port, &session2,
|
|
&audio_transport) == GST_RTSP_STS_OK);
|
|
|
|
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 2);
|
|
|
|
/* send PLAY request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
|
|
session1) == GST_RTSP_STS_OK);
|
|
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
|
|
session2) == GST_RTSP_STS_OK);
|
|
|
|
sleep (7);
|
|
|
|
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
|
|
|
|
/* send TEARDOWN request and check that we get 454 Session Not found */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session1) == GST_RTSP_STS_SESSION_NOT_FOUND);
|
|
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session2) == GST_RTSP_STS_OK);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
g_object_unref (pool);
|
|
g_free (session1);
|
|
g_free (session2);
|
|
gst_rtsp_transport_free (video_transport);
|
|
gst_rtsp_transport_free (audio_transport);
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
GST_START_TEST (test_play_disconnect)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
const gchar *audio_control;
|
|
GstRTSPRange client_port;
|
|
gchar *session = NULL;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
GstRTSPTransport *audio_transport = NULL;
|
|
GstRTSPSessionPool *pool;
|
|
|
|
pool = gst_rtsp_server_get_session_pool (server);
|
|
g_signal_connect (server, "client-connected",
|
|
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
|
|
|
|
start_server ();
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get control strings from DESCRIBE response */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
|
|
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports (&client_port);
|
|
|
|
/* do SETUP for video and audio */
|
|
fail_unless (do_setup (conn, video_control, &client_port, &session,
|
|
&video_transport) == GST_RTSP_STS_OK);
|
|
fail_unless (do_setup (conn, audio_control, &client_port, &session,
|
|
&audio_transport) == GST_RTSP_STS_OK);
|
|
|
|
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
|
|
|
|
/* send PLAY request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
|
|
session) == GST_RTSP_STS_OK);
|
|
|
|
gst_rtsp_connection_free (conn);
|
|
|
|
sleep (7);
|
|
|
|
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
|
|
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
|
|
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
g_object_unref (pool);
|
|
g_free (session);
|
|
gst_rtsp_transport_free (video_transport);
|
|
gst_rtsp_transport_free (audio_transport);
|
|
gst_sdp_message_free (sdp_message);
|
|
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
/* Only different with test_play is the specific ports selected */
|
|
|
|
GST_START_TEST (test_play_specific_server_port)
|
|
{
|
|
GstRTSPMountPoints *mounts;
|
|
gchar *service;
|
|
GstRTSPMediaFactory *factory;
|
|
GstRTSPAddressPool *pool;
|
|
GstRTSPConnection *conn;
|
|
GstSDPMessage *sdp_message = NULL;
|
|
const GstSDPMedia *sdp_media;
|
|
const gchar *video_control;
|
|
GstRTSPRange client_port;
|
|
gchar *session = NULL;
|
|
GstRTSPTransport *video_transport = NULL;
|
|
GSocket *rtp_socket, *rtcp_socket;
|
|
GSocketAddress *rtp_address, *rtcp_address;
|
|
guint16 rtp_port, rtcp_port;
|
|
|
|
mounts = gst_rtsp_server_get_mount_points (server);
|
|
|
|
factory = gst_rtsp_media_factory_new ();
|
|
/* we have to suspend media after SDP in order to make sure that
|
|
* we can reconfigure UDP sink with new UDP ports */
|
|
gst_rtsp_media_factory_set_suspend_mode (factory, GST_RTSP_SUSPEND_MODE_RESET);
|
|
pool = gst_rtsp_address_pool_new ();
|
|
gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
|
|
GST_RTSP_ADDRESS_POOL_ANY_IPV4, 7770, 7780, 0);
|
|
gst_rtsp_media_factory_set_address_pool (factory, pool);
|
|
g_object_unref (pool);
|
|
gst_rtsp_media_factory_set_launch (factory, "( " VIDEO_PIPELINE " )");
|
|
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);
|
|
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
|
|
|
|
/* get control strings from DESCRIBE response */
|
|
fail_unless (gst_sdp_message_medias_len (sdp_message) == 1);
|
|
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
|
|
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
|
|
|
|
get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
|
|
|
|
/* do SETUP for video */
|
|
fail_unless (do_setup (conn, video_control, &client_port, &session,
|
|
&video_transport) == GST_RTSP_STS_OK);
|
|
|
|
/* send PLAY request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
|
|
session) == GST_RTSP_STS_OK);
|
|
|
|
receive_rtp (rtp_socket, &rtp_address);
|
|
receive_rtcp (rtcp_socket, &rtcp_address, 0);
|
|
|
|
fail_unless (G_IS_INET_SOCKET_ADDRESS (rtp_address));
|
|
fail_unless (G_IS_INET_SOCKET_ADDRESS (rtcp_address));
|
|
rtp_port =
|
|
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_address));
|
|
rtcp_port =
|
|
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtcp_address));
|
|
fail_unless (rtp_port >= 7770 && rtp_port <= 7780 && rtp_port % 2 == 0);
|
|
fail_unless (rtcp_port >= 7770 && rtcp_port <= 7780 && rtcp_port % 2 == 1);
|
|
fail_unless (rtp_port + 1 == rtcp_port);
|
|
|
|
g_object_unref (rtp_address);
|
|
g_object_unref (rtcp_address);
|
|
|
|
/* send TEARDOWN request and check that we get 200 OK */
|
|
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
|
|
session) == GST_RTSP_STS_OK);
|
|
|
|
/* FIXME: The rtsp-server always disconnects the transport before
|
|
* sending the RTCP BYE
|
|
* receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
|
|
*/
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
g_object_unref (rtp_socket);
|
|
g_object_unref (rtcp_socket);
|
|
g_free (session);
|
|
gst_rtsp_transport_free (video_transport);
|
|
gst_sdp_message_free (sdp_message);
|
|
gst_rtsp_connection_free (conn);
|
|
|
|
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
GST_START_TEST (test_play_smpte_range)
|
|
{
|
|
start_server ();
|
|
|
|
do_test_play ("npt=5-");
|
|
do_test_play ("smpte=0:00:00-");
|
|
do_test_play ("smpte=1:00:00-");
|
|
do_test_play ("smpte=1:00:03-");
|
|
do_test_play ("clock=20120321T152256Z-");
|
|
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
GST_START_TEST (test_announce_without_sdp)
|
|
{
|
|
GstRTSPConnection *conn;
|
|
GstRTSPStatusCode status;
|
|
GstRTSPMessage *request;
|
|
GstRTSPMessage *response;
|
|
|
|
start_record_server ("( fakesink name=depay0 )");
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
/* create and send ANNOUNCE request */
|
|
request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
|
|
|
|
fail_unless (send_request (conn, request));
|
|
|
|
iterate ();
|
|
|
|
response = read_response (conn);
|
|
|
|
/* check response */
|
|
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
|
|
fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
|
|
gst_rtsp_message_free (response);
|
|
|
|
/* try again, this type with content-type, but still no SDP */
|
|
gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
|
|
"application/sdp");
|
|
|
|
fail_unless (send_request (conn, request));
|
|
|
|
iterate ();
|
|
|
|
response = read_response (conn);
|
|
|
|
/* check response */
|
|
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
|
|
fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
|
|
gst_rtsp_message_free (response);
|
|
|
|
/* try again, this type with an unknown content-type */
|
|
gst_rtsp_message_remove_header (request, GST_RTSP_HDR_CONTENT_TYPE, -1);
|
|
gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
|
|
"application/x-something");
|
|
|
|
fail_unless (send_request (conn, request));
|
|
|
|
iterate ();
|
|
|
|
response = read_response (conn);
|
|
|
|
/* check response */
|
|
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
|
|
fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
|
|
gst_rtsp_message_free (response);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
gst_rtsp_message_free (request);
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static GstRTSPStatusCode
|
|
do_announce (GstRTSPConnection * conn, GstSDPMessage * sdp)
|
|
{
|
|
GstRTSPMessage *request;
|
|
GstRTSPMessage *response;
|
|
GstRTSPStatusCode code;
|
|
gchar *str;
|
|
|
|
/* create request */
|
|
request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
|
|
|
|
gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
|
|
"application/sdp");
|
|
|
|
/* add SDP to the response body */
|
|
str = gst_sdp_message_as_text (sdp);
|
|
gst_rtsp_message_take_body (request, (guint8 *) str, strlen (str));
|
|
gst_sdp_message_free (sdp);
|
|
|
|
/* send request */
|
|
fail_unless (send_request (conn, request));
|
|
gst_rtsp_message_free (request);
|
|
|
|
iterate ();
|
|
|
|
/* read response */
|
|
response = read_response (conn);
|
|
|
|
/* check status line */
|
|
gst_rtsp_message_parse_response (response, &code, NULL, NULL);
|
|
|
|
gst_rtsp_message_free (response);
|
|
return code;
|
|
}
|
|
|
|
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 RECORD_N_BUFS 10
|
|
|
|
GST_START_TEST (test_record_tcp)
|
|
{
|
|
GstRTSPMediaFactory *mfactory;
|
|
GstRTSPConnection *conn;
|
|
GstRTSPStatusCode status;
|
|
GstRTSPMessage *response;
|
|
GstRTSPMessage *request;
|
|
GstSDPMessage *sdp;
|
|
GstRTSPResult rres;
|
|
GSocketAddress *sa;
|
|
GInetAddress *ia;
|
|
GstElement *server_sink = NULL;
|
|
GSocket *conn_socket;
|
|
const gchar *proto;
|
|
gchar *client_ip, *sess_id, *session = 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);
|
|
|
|
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
|
|
|
|
conn_socket = gst_rtsp_connection_get_read_socket (conn);
|
|
|
|
sa = g_socket_get_local_address (conn_socket, NULL);
|
|
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
|
|
client_ip = g_inet_address_to_string (ia);
|
|
if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6)
|
|
proto = "IP6";
|
|
else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
|
|
proto = "IP4";
|
|
else
|
|
g_assert_not_reached ();
|
|
g_object_unref (sa);
|
|
|
|
gst_sdp_message_new (&sdp);
|
|
|
|
/* some standard things first */
|
|
gst_sdp_message_set_version (sdp, "0");
|
|
|
|
/* session ID doesn't have to be super-unique in this case */
|
|
sess_id = g_strdup_printf ("%u", g_random_int ());
|
|
gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
|
|
g_free (sess_id);
|
|
g_free (client_ip);
|
|
|
|
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
|
|
gst_sdp_message_set_information (sdp, "rtsp-server-test");
|
|
gst_sdp_message_add_time (sdp, "0", "0", NULL);
|
|
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
|
|
|
|
/* add stream 0 */
|
|
{
|
|
GstSDPMedia *smedia;
|
|
|
|
gst_sdp_media_new (&smedia);
|
|
gst_sdp_media_set_media (smedia, "audio");
|
|
gst_sdp_media_add_format (smedia, "8"); /* pcma/alaw */
|
|
gst_sdp_media_set_port_info (smedia, 0, 1);
|
|
gst_sdp_media_set_proto (smedia, "RTP/AVP");
|
|
gst_sdp_media_add_attribute (smedia, "rtpmap", "8 PCMA/8000");
|
|
gst_sdp_message_add_media (sdp, smedia);
|
|
gst_sdp_media_free (smedia);
|
|
}
|
|
|
|
/* send ANNOUNCE request */
|
|
status = do_announce (conn, sdp);
|
|
fail_unless_equals_int (status, GST_RTSP_STS_OK);
|
|
|
|
/* create and send SETUP request */
|
|
request = create_request (conn, GST_RTSP_SETUP, NULL);
|
|
gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT,
|
|
"RTP/AVP/TCP;interleaved=0;mode=record");
|
|
fail_unless (send_request (conn, request));
|
|
gst_rtsp_message_free (request);
|
|
iterate ();
|
|
response = read_response (conn);
|
|
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
|
|
fail_unless_equals_int (status, GST_RTSP_STS_OK);
|
|
|
|
rres =
|
|
gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &session, 0);
|
|
session = g_strdup (session);
|
|
fail_unless_equals_int (rres, GST_RTSP_OK);
|
|
gst_rtsp_message_free (response);
|
|
|
|
/* send RECORD */
|
|
request = create_request (conn, GST_RTSP_RECORD, NULL);
|
|
gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
|
|
fail_unless (send_request (conn, request));
|
|
gst_rtsp_message_free (request);
|
|
iterate ();
|
|
response = read_response (conn);
|
|
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
|
|
fail_unless_equals_int (status, GST_RTSP_STS_OK);
|
|
gst_rtsp_message_free (response);
|
|
|
|
/* send some data */
|
|
{
|
|
GstElement *pipeline, *src, *enc, *pay, *sink;
|
|
|
|
pipeline = gst_pipeline_new ("send-pipeline");
|
|
src = gst_element_factory_make ("audiotestsrc", NULL);
|
|
g_object_set (src, "num-buffers", RECORD_N_BUFS,
|
|
"samplesperbuffer", 1000, NULL);
|
|
enc = gst_element_factory_make ("alawenc", NULL);
|
|
pay = gst_element_factory_make ("rtppcmapay", NULL);
|
|
sink = gst_element_factory_make ("appsink", NULL);
|
|
fail_unless (pipeline && src && enc && pay && sink);
|
|
gst_bin_add_many (GST_BIN (pipeline), src, enc, pay, sink, NULL);
|
|
gst_element_link_many (src, enc, pay, sink, NULL);
|
|
gst_element_set_state (pipeline, GST_STATE_PLAYING);
|
|
|
|
do {
|
|
GstRTSPMessage *data_msg;
|
|
GstMapInfo map = GST_MAP_INFO_INIT;
|
|
GstRTSPResult rres;
|
|
GstSample *sample = NULL;
|
|
GstBuffer *buf;
|
|
|
|
g_signal_emit_by_name (G_OBJECT (sink), "pull-sample", &sample);
|
|
if (sample == NULL)
|
|
break;
|
|
buf = gst_sample_get_buffer (sample);
|
|
rres = gst_rtsp_message_new_data (&data_msg, 0);
|
|
fail_unless_equals_int (rres, GST_RTSP_OK);
|
|
gst_buffer_map (buf, &map, GST_MAP_READ);
|
|
GST_INFO ("sending %u bytes of data on channel 0", (guint) map.size);
|
|
GST_MEMDUMP ("data on channel 0", map.data, map.size);
|
|
rres = gst_rtsp_message_set_body (data_msg, map.data, map.size);
|
|
fail_unless_equals_int (rres, GST_RTSP_OK);
|
|
gst_buffer_unmap (buf, &map);
|
|
rres = gst_rtsp_connection_send (conn, data_msg, NULL);
|
|
fail_unless_equals_int (rres, GST_RTSP_OK);
|
|
gst_rtsp_message_free (data_msg);
|
|
gst_sample_unref (sample);
|
|
} while (TRUE);
|
|
|
|
gst_element_set_state (pipeline, GST_STATE_NULL);
|
|
gst_object_unref (pipeline);
|
|
}
|
|
|
|
/* 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);
|
|
gst_sample_unref (sample);
|
|
}
|
|
|
|
fail_unless_equals_int (GST_STATE (server_sink), GST_STATE_PLAYING);
|
|
|
|
/* clean up and iterate so the clean-up can finish */
|
|
gst_rtsp_connection_free (conn);
|
|
stop_server ();
|
|
iterate ();
|
|
g_free (session);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
static Suite *
|
|
rtspserver_suite (void)
|
|
{
|
|
Suite *s = suite_create ("rtspserver");
|
|
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_connect);
|
|
tcase_add_test (tc, test_describe);
|
|
tcase_add_test (tc, test_describe_non_existing_mount_point);
|
|
tcase_add_test (tc, test_describe_record_media);
|
|
tcase_add_test (tc, test_setup_udp);
|
|
tcase_add_test (tc, test_setup_tcp);
|
|
tcase_add_test (tc, test_setup_udp_mcast);
|
|
tcase_add_test (tc, test_setup_twice);
|
|
tcase_add_test (tc, test_setup_with_require_header);
|
|
tcase_add_test (tc, test_setup_non_existing_stream);
|
|
tcase_add_test (tc, test_play);
|
|
tcase_add_test (tc, test_play_without_session);
|
|
tcase_add_test (tc, test_bind_already_in_use);
|
|
tcase_add_test (tc, test_play_multithreaded);
|
|
tcase_add_test (tc, test_play_multithreaded_block_in_describe);
|
|
tcase_add_test (tc, test_play_multithreaded_timeout_client);
|
|
tcase_add_test (tc, test_play_multithreaded_timeout_session);
|
|
tcase_add_test (tc, test_play_disconnect);
|
|
tcase_add_test (tc, test_play_specific_server_port);
|
|
tcase_add_test (tc, test_play_smpte_range);
|
|
tcase_add_test (tc, test_announce_without_sdp);
|
|
tcase_add_test (tc, test_record_tcp);
|
|
return s;
|
|
}
|
|
|
|
GST_CHECK_MAIN (rtspserver);
|