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gstreamer/subprojects/gst-rtsp-server/tests/check/gst/stream.c

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stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
/* GStreamer
* Copyright (C) 2013 Axis Communications AB <dev-gstreamer at axis dot 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 <rtsp-stream.h>
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
#include <rtsp-address-pool.h>
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
static void
get_sockets (GstRTSPLowerTrans lower_transport, GSocketFamily socket_family)
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstBin *bin;
GstElement *rtpbin;
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
GstRTSPAddressPool *pool;
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
GSocket *socket;
gboolean have_ipv4;
gboolean have_ipv6;
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
GstRTSPTransport *transport;
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
/* configure address pool for IPv4 and IPv6 unicast addresses */
pool = gst_rtsp_address_pool_new ();
rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak. - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function. - Create unit test for shared media. https://bugzilla.gnome.org/show_bug.cgi?id=764744
2016-04-25 12:55:25 +00:00
fail_unless (gst_rtsp_address_pool_add_range (pool,
GST_RTSP_ADDRESS_POOL_ANY_IPV4, GST_RTSP_ADDRESS_POOL_ANY_IPV4, 50000,
60000, 0));
fail_unless (gst_rtsp_address_pool_add_range (pool,
GST_RTSP_ADDRESS_POOL_ANY_IPV6, GST_RTSP_ADDRESS_POOL_ANY_IPV6, 50000,
60000, 0));
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
fail_unless (gst_rtsp_address_pool_add_range (pool, "233.252.0.0",
"233.252.0.0", 50000, 60000, 1));
fail_unless (gst_rtsp_address_pool_add_range (pool, "FF11:DB8::1",
"FF11:DB8::1", 50000, 60000, 1));
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
gst_rtsp_stream_set_address_pool (stream, pool);
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
/* allocate udp ports first */
fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
transport->lower_transport = lower_transport;
/* no ports allocated, complete stream should fail */
fail_if (gst_rtsp_stream_complete_stream (stream, transport));
/* allocate ports */
fail_unless (gst_rtsp_stream_allocate_udp_sockets (stream,
tests: fix indentation Fix and "fix".
2018-01-13 15:02:28 +00:00
socket_family, transport, FALSE));
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
fail_unless (gst_rtsp_stream_complete_stream (stream, transport));
fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV4);
else
socket = gst_rtsp_stream_get_rtp_multicast_socket (stream,
G_SOCKET_FAMILY_IPV4);
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
have_ipv4 = (socket != NULL);
if (have_ipv4) {
fail_unless (g_socket_get_fd (socket) >= 0);
g_object_unref (socket);
}
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV4);
else
socket = gst_rtsp_stream_get_rtcp_multicast_socket (stream,
G_SOCKET_FAMILY_IPV4);
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
if (have_ipv4) {
fail_unless (socket != NULL);
fail_unless (g_socket_get_fd (socket) >= 0);
g_object_unref (socket);
} else {
fail_unless (socket == NULL);
}
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV6);
else
socket = gst_rtsp_stream_get_rtp_multicast_socket (stream,
G_SOCKET_FAMILY_IPV6);
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
have_ipv6 = (socket != NULL);
if (have_ipv6) {
fail_unless (g_socket_get_fd (socket) >= 0);
g_object_unref (socket);
}
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
if (lower_transport == GST_RTSP_LOWER_TRANS_UDP)
socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV6);
else
socket = gst_rtsp_stream_get_rtcp_multicast_socket (stream,
G_SOCKET_FAMILY_IPV6);
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
if (have_ipv6) {
fail_unless (socket != NULL);
fail_unless (g_socket_get_fd (socket) >= 0);
g_object_unref (socket);
} else {
fail_unless (socket == NULL);
}
/* check that at least one family is available */
fail_unless (have_ipv4 || have_ipv6);
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
g_object_unref (pool);
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst_object_unref (bin);
gst_object_unref (stream);
}
tests: Don't fail tests if IPv6 not available. On computers with IPv6 disabled it shouldn't result in a test failure. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 03:26:02 +00:00
GST_START_TEST (test_get_sockets_udp_ipv4)
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
{
get_sockets (GST_RTSP_LOWER_TRANS_UDP, G_SOCKET_FAMILY_IPV4);
tests: Don't fail tests if IPv6 not available. On computers with IPv6 disabled it shouldn't result in a test failure. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 03:26:02 +00:00
}
GST_END_TEST;
GST_START_TEST (test_get_sockets_udp_ipv6)
{
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
get_sockets (GST_RTSP_LOWER_TRANS_UDP, G_SOCKET_FAMILY_IPV6);
}
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
GST_END_TEST;
tests: Don't fail tests if IPv6 not available. On computers with IPv6 disabled it shouldn't result in a test failure. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 03:26:02 +00:00
GST_START_TEST (test_get_sockets_mcast_ipv4)
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
{
get_sockets (GST_RTSP_LOWER_TRANS_UDP_MCAST, G_SOCKET_FAMILY_IPV4);
tests: Don't fail tests if IPv6 not available. On computers with IPv6 disabled it shouldn't result in a test failure. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 03:26:02 +00:00
}
GST_END_TEST;
GST_START_TEST (test_get_sockets_mcast_ipv6)
{
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
get_sockets (GST_RTSP_LOWER_TRANS_UDP_MCAST, G_SOCKET_FAMILY_IPV6);
}
GST_END_TEST;
/* The purpose of this test is to make sure that it's not possible to allocate
* multicast UDP ports if the address pool does not contain multicast UDP
* addresses. */
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
GST_START_TEST (test_allocate_udp_ports_fail)
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstBin *bin;
GstElement *rtpbin;
GstRTSPAddressPool *pool;
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
GstRTSPTransport *transport;
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
pool = gst_rtsp_address_pool_new ();
fail_unless (gst_rtsp_address_pool_add_range (pool, "192.168.1.1",
rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak. - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function. - Create unit test for shared media. https://bugzilla.gnome.org/show_bug.cgi?id=764744
2016-04-25 12:55:25 +00:00
"192.168.1.1", 6000, 6001, 0));
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
gst_rtsp_stream_set_address_pool (stream, pool);
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
transport->lower_transport = GST_RTSP_LOWER_TRANS_UDP_MCAST;
fail_if (gst_rtsp_stream_allocate_udp_sockets (stream, G_SOCKET_FAMILY_IPV4,
tests: fix indentation Fix and "fix".
2018-01-13 15:02:28 +00:00
transport, FALSE));
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
g_object_unref (pool);
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst_object_unref (bin);
gst_object_unref (stream);
}
GST_END_TEST;
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
GST_START_TEST (test_get_multicast_address)
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstRTSPAddressPool *pool;
GstRTSPAddress *addr1;
GstRTSPAddress *addr2;
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
pool = gst_rtsp_address_pool_new ();
fail_unless (gst_rtsp_address_pool_add_range (pool,
tests: try to avoid using the same ports in different tests Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:38:01 +00:00
"233.252.0.0", "233.252.0.0", 5100, 5101, 1));
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
fail_unless (gst_rtsp_address_pool_add_range (pool,
tests: try to avoid using the same ports in different tests Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:38:01 +00:00
"FF11:DB8::1", "FF11:DB8::1", 5102, 5103, 1));
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
gst_rtsp_stream_set_address_pool (stream, pool);
addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
fail_unless (addr1 != NULL);
fail_unless_equals_string (addr1->address, "233.252.0.0");
tests: try to avoid using the same ports in different tests Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:38:01 +00:00
fail_unless_equals_int (addr1->port, 5100);
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
fail_unless_equals_int (addr1->n_ports, 2);
addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
fail_unless (addr2 != NULL);
fail_unless_equals_string (addr2->address, "233.252.0.0");
tests: try to avoid using the same ports in different tests Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:38:01 +00:00
fail_unless_equals_int (addr2->port, 5100);
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
fail_unless_equals_int (addr2->n_ports, 2);
gst_rtsp_address_free (addr1);
gst_rtsp_address_free (addr2);
addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
fail_unless (addr1 != NULL);
fail_unless (!g_ascii_strcasecmp (addr1->address, "FF11:DB8::1"));
tests: try to avoid using the same ports in different tests Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:38:01 +00:00
fail_unless_equals_int (addr1->port, 5102);
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
fail_unless_equals_int (addr1->n_ports, 2);
addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
fail_unless (addr2 != NULL);
fail_unless (!g_ascii_strcasecmp (addr2->address, "FF11:DB8::1"));
tests: try to avoid using the same ports in different tests Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:38:01 +00:00
fail_unless_equals_int (addr2->port, 5102);
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
fail_unless_equals_int (addr2->n_ports, 2);
gst_rtsp_address_free (addr1);
gst_rtsp_address_free (addr2);
g_object_unref (pool);
gst_object_unref (stream);
}
GST_END_TEST;
stream tests: added new tests Test a case when the address pool only contains multicast addresses and the client is requesting unicast udp. Added tests for multicast ports allocation. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 14:07:06 +00:00
/* test case: address pool only contains multicast addresses,
* but the client is requesting unicast udp */
GST_START_TEST (test_multicast_address_and_unicast_udp)
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstBin *bin;
GstElement *rtpbin;
GstRTSPAddressPool *pool;
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
pool = gst_rtsp_address_pool_new ();
/* add a multicast addres to the address pool */
fail_unless (gst_rtsp_address_pool_add_range (pool,
tests: try to avoid using the same ports in different tests Causes problems with client multicast tests otherwise if tests are run in parallel. https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:38:01 +00:00
"233.252.0.0", "233.252.0.0", 5200, 5201, 1));
stream tests: added new tests Test a case when the address pool only contains multicast addresses and the client is requesting unicast udp. Added tests for multicast ports allocation. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 14:07:06 +00:00
gst_rtsp_stream_set_address_pool (stream, pool);
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
g_object_unref (pool);
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst_object_unref (bin);
gst_object_unref (stream);
}
GST_END_TEST;
GST_START_TEST (test_allocate_udp_ports_multicast)
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstBin *bin;
GstElement *rtpbin;
GstRTSPAddressPool *pool;
GstRTSPAddress *addr;
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
pool = gst_rtsp_address_pool_new ();
/* add multicast addresses to the address pool */
fail_unless (gst_rtsp_address_pool_add_range (pool,
"233.252.0.1", "233.252.0.1", 6000, 6001, 1));
fail_unless (gst_rtsp_address_pool_add_range (pool,
"FF11:DB8::1", "FF11:DB8::1", 6002, 6003, 1));
gst_rtsp_stream_set_address_pool (stream, pool);
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
/* check the multicast address and ports for IPv4 */
addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
fail_unless (addr != NULL);
fail_unless_equals_string (addr->address, "233.252.0.1");
fail_unless_equals_int (addr->port, 6000);
fail_unless_equals_int (addr->n_ports, 2);
gst_rtsp_address_free (addr);
/* check the multicast address and ports for IPv6 */
addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
fail_unless (addr != NULL);
fail_unless (!g_ascii_strcasecmp (addr->address, "FF11:DB8::1"));
fail_unless_equals_int (addr->port, 6002);
fail_unless_equals_int (addr->n_ports, 2);
gst_rtsp_address_free (addr);
g_object_unref (pool);
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst_object_unref (bin);
gst_object_unref (stream);
}
GST_END_TEST;
GST_START_TEST (test_allocate_udp_ports_client_settings)
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstBin *bin;
GstElement *rtpbin;
GstRTSPAddressPool *pool;
GstRTSPAddress *addr;
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
pool = gst_rtsp_address_pool_new ();
/* add multicast addresses to the address pool */
fail_unless (gst_rtsp_address_pool_add_range (pool,
"233.252.0.1", "233.252.0.1", 6000, 6001, 1));
fail_unless (gst_rtsp_address_pool_add_range (pool,
"FF11:DB7::1", "FF11:DB7::1", 6004, 6005, 1));
/* multicast address specified by the client */
fail_unless (gst_rtsp_address_pool_add_range (pool,
"233.252.0.2", "233.252.0.2", 6002, 6003, 1));
fail_unless (gst_rtsp_address_pool_add_range (pool,
"FF11:DB8::1", "FF11:DB8::1", 6006, 6007, 1));
gst_rtsp_stream_set_address_pool (stream, pool);
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast This is basically reverting changes introduced in commit f62a9a7, because it was introducing various regressions: - It introduces a leak of udpsrc elements that got wrongly fixed by adding an hash table in commit cba045e. We should have at most 4 udpsrc for unicast: ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients. - If a mcast client connects, it creates a new socket in SETUP to try to respect the destination/port given by the client in the transport, and overrides the socket already set on the udpsink element. That means that if we already had a client connected, the source address on the udp packets it receives suddenly changes. - If a 2nd mcast client connects, the destination/port in its transport is ignored but its transport wasn't updated. What this patch does: - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE. - Always have a tee+queue when udp is enabled. This could be optimized again in a later patch, but is more complicated. If no unicast clients connects then those elements are useless, this could be also optimized in a later patch. - When mcast transport is added, it creates a new set of udpsrc/udpsink, seperated from those for unicast clients. Since we already support only one mcast address, we also create only one set of elements. https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-28 19:33:05 +00:00
/* Reserve IPV4 mcast address */
addr = gst_rtsp_stream_reserve_address (stream, "233.252.0.2", 6002, 2, 1);
fail_unless (addr != NULL);
gst_rtsp_address_free (addr);
stream tests: added new tests Test a case when the address pool only contains multicast addresses and the client is requesting unicast udp. Added tests for multicast ports allocation. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 14:07:06 +00:00
/* verify that the multicast address and ports correspond to the requested client
* transport information for IPv4 */
addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
fail_unless (addr != NULL);
fail_unless_equals_string (addr->address, "233.252.0.2");
fail_unless_equals_int (addr->port, 6002);
fail_unless_equals_int (addr->n_ports, 2);
gst_rtsp_address_free (addr);
stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast This is basically reverting changes introduced in commit f62a9a7, because it was introducing various regressions: - It introduces a leak of udpsrc elements that got wrongly fixed by adding an hash table in commit cba045e. We should have at most 4 udpsrc for unicast: ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients. - If a mcast client connects, it creates a new socket in SETUP to try to respect the destination/port given by the client in the transport, and overrides the socket already set on the udpsink element. That means that if we already had a client connected, the source address on the udp packets it receives suddenly changes. - If a 2nd mcast client connects, the destination/port in its transport is ignored but its transport wasn't updated. What this patch does: - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE. - Always have a tee+queue when udp is enabled. This could be optimized again in a later patch, but is more complicated. If no unicast clients connects then those elements are useless, this could be also optimized in a later patch. - When mcast transport is added, it creates a new set of udpsrc/udpsink, seperated from those for unicast clients. Since we already support only one mcast address, we also create only one set of elements. https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-28 19:33:05 +00:00
/* Reserve IPV6 mcast address */
addr = gst_rtsp_stream_reserve_address (stream, "FF11:DB8::1", 6006, 2, 1);
fail_unless (addr != NULL);
gst_rtsp_address_free (addr);
stream tests: added new tests Test a case when the address pool only contains multicast addresses and the client is requesting unicast udp. Added tests for multicast ports allocation. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 14:07:06 +00:00
/* verify that the multicast address and ports correspond to the requested client
* transport information for IPv6 */
addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
fail_unless (addr != NULL);
fail_unless (!g_ascii_strcasecmp (addr->address, "FF11:DB8::1"));
fail_unless_equals_int (addr->port, 6006);
fail_unless_equals_int (addr->n_ports, 2);
gst_rtsp_address_free (addr);
g_object_unref (pool);
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst_object_unref (bin);
gst_object_unref (stream);
}
GST_END_TEST;
rtsp-stream: fixed segmenation fault in _get_server_port() Calling function gst_rtsp_stream_get_server_port() results in segmenation fault in the RTP/RTSP/TCP case. Port that the server will use to receive RTCP makes only sense in the UDP case, however the function should handle the TCP case in a nicer way. https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 13:12:05 +00:00
GST_START_TEST (test_tcp_transport)
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstBin *bin;
GstElement *rtpbin;
GstRTSPRange server_port;
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
/* TCP transport */
gst_rtsp_stream_set_protocols (stream, GST_RTSP_LOWER_TRANS_TCP);
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
/* port that the server will use to receive RTCP makes only sense in the UDP
* case so verify that the received server port is 0 in the TCP case */
gst_rtsp_stream_get_server_port (stream, &server_port, G_SOCKET_FAMILY_IPV4);
fail_unless_equals_int (server_port.min, 0);
fail_unless_equals_int (server_port.max, 0);
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst_object_unref (bin);
gst_object_unref (stream);
}
GST_END_TEST;
stream: Added a list of multicast client addresses When media is shared, the same media stream can be sent to multiple multicast groups. Currently, there is no API to retrieve multicast addresses from the stream. When calling gst_rtsp_stream_get_multicast_address() function, only the first multicast address is returned. With this patch, each multicast destination requested in SETUP will be stored in an internal list (call to gst_rtsp_stream_add_multicast_client_address()). The list of multicast groups requested by the clients can be retrieved by calling gst_rtsp_stream_get_multicast_client_addresses(). There still exist some problems with the current implementation in the multicast case: 1) The receiving part is currently only configured with regard to the first multicast client (see https://bugzilla.gnome.org/show_bug.cgi?id=796917). 2) Secondly, of security reasons, some constraints should be put on the requested multicast destinations (see https://bugzilla.gnome.org/show_bug.cgi?id=796916). Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-26 10:01:16 +00:00
static void
check_multicast_client_address (const gchar * destination, guint port,
const gchar * expected_addr_str, gboolean expected_res)
{
GstPad *srcpad;
GstElement *pay;
GstRTSPStream *stream;
GstBin *bin;
GstElement *rtpbin;
GstRTSPTransport *transport;
GstRTSPRange ports = { 0 };
gchar *addr_str = NULL;
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
gst_pad_set_active (srcpad, TRUE);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
transport->lower_transport = GST_RTSP_LOWER_TRANS_UDP_MCAST;
transport->destination = g_strdup (destination);
transport->ttl = 1;
ports.min = port;
ports.max = port + 1;
transport->port = ports;
/* allocate ports */
fail_unless (gst_rtsp_stream_allocate_udp_sockets (stream,
G_SOCKET_FAMILY_IPV4, transport, TRUE) == expected_res);
fail_unless (gst_rtsp_stream_add_multicast_client_address (stream,
Fix indentation again
2018-08-14 11:31:55 +00:00
destination, ports.min, ports.max,
G_SOCKET_FAMILY_IPV4) == expected_res);
stream: Added a list of multicast client addresses When media is shared, the same media stream can be sent to multiple multicast groups. Currently, there is no API to retrieve multicast addresses from the stream. When calling gst_rtsp_stream_get_multicast_address() function, only the first multicast address is returned. With this patch, each multicast destination requested in SETUP will be stored in an internal list (call to gst_rtsp_stream_add_multicast_client_address()). The list of multicast groups requested by the clients can be retrieved by calling gst_rtsp_stream_get_multicast_client_addresses(). There still exist some problems with the current implementation in the multicast case: 1) The receiving part is currently only configured with regard to the first multicast client (see https://bugzilla.gnome.org/show_bug.cgi?id=796917). 2) Secondly, of security reasons, some constraints should be put on the requested multicast destinations (see https://bugzilla.gnome.org/show_bug.cgi?id=796916). Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-26 10:01:16 +00:00
Fix indentation again
2018-08-14 11:31:55 +00:00
fail_unless (gst_rtsp_stream_complete_stream (stream,
transport) == expected_res);
stream: Added a list of multicast client addresses When media is shared, the same media stream can be sent to multiple multicast groups. Currently, there is no API to retrieve multicast addresses from the stream. When calling gst_rtsp_stream_get_multicast_address() function, only the first multicast address is returned. With this patch, each multicast destination requested in SETUP will be stored in an internal list (call to gst_rtsp_stream_add_multicast_client_address()). The list of multicast groups requested by the clients can be retrieved by calling gst_rtsp_stream_get_multicast_client_addresses(). There still exist some problems with the current implementation in the multicast case: 1) The receiving part is currently only configured with regard to the first multicast client (see https://bugzilla.gnome.org/show_bug.cgi?id=796917). 2) Secondly, of security reasons, some constraints should be put on the requested multicast destinations (see https://bugzilla.gnome.org/show_bug.cgi?id=796916). Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-26 10:01:16 +00:00
fail_unless (gst_rtsp_transport_free (transport) == GST_RTSP_OK);
addr_str = gst_rtsp_stream_get_multicast_client_addresses (stream);
fail_unless (g_str_equal (addr_str, expected_addr_str));
g_free (addr_str);
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst_object_unref (bin);
gst_object_unref (stream);
}
/* test if the provided transport destination is correct.
* CASE: valid multicast address */
GST_START_TEST (test_multicast_client_address)
{
const gchar *addr = "233.252.0.1";
guint port = 50000;
const gchar *expected_addr_str = "233.252.0.1:50000";
gboolean expected_res = TRUE;
check_multicast_client_address (addr, port, expected_addr_str, expected_res);
}
GST_END_TEST;
/* test if the provided transport destination is correct.
* CASE: invalid multicast address */
GST_START_TEST (test_multicast_client_address_invalid)
{
const gchar *addr = "1.2.3.4";
guint port = 50000;
const gchar *expected_addr_str = "";
gboolean expected_res = FALSE;
check_multicast_client_address (addr, port, expected_addr_str, expected_res);
}
GST_END_TEST;
rtsp-stream: Removing invalid transports returns false When removing transports an assertion was that the transports passed in for removal are present in the list, however that can't be assumed. As an example if a transport was removed from a thread running send_tcp_message, the main thread can try to remove the same transport again if it gets a handle_pause_request. This will not effect the transport list but it will effect n_tcp_transports as it will be decrement and then have the wrong value.
2019-11-01 11:01:41 +00:00
static void
add_transports (gboolean add_twice)
{
GstRTSPTransport *transport;
GstRTSPStream *stream;
GstRTSPStreamTransport *tr;
GstPad *srcpad;
GstElement *pay;
GstBin *bin;
GstElement *rtpbin;
fail_unless (gst_rtsp_transport_new (&transport) == GST_RTSP_OK);
transport->lower_transport = GST_RTSP_LOWER_TRANS_TCP;
transport->destination = g_strdup ("127.0.0.1");
srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
fail_unless (srcpad != NULL);
pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
fail_unless (pay != NULL);
stream = gst_rtsp_stream_new (0, pay, srcpad);
fail_unless (stream != NULL);
gst_object_unref (pay);
gst_object_unref (srcpad);
rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
fail_unless (rtpbin != NULL);
bin = GST_BIN (gst_bin_new ("testbin"));
fail_unless (bin != NULL);
fail_unless (gst_bin_add (bin, rtpbin));
/* TCP transport */
gst_rtsp_stream_set_protocols (stream, GST_RTSP_LOWER_TRANS_TCP);
fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
tr = gst_rtsp_stream_transport_new (stream, transport);
fail_unless (tr);
if (add_twice) {
fail_unless (gst_rtsp_stream_add_transport (stream, tr));
fail_unless (gst_rtsp_stream_add_transport (stream, tr));
fail_unless (gst_rtsp_stream_remove_transport (stream, tr));
} else {
fail_unless (gst_rtsp_stream_add_transport (stream, tr));
fail_unless (gst_rtsp_stream_remove_transport (stream, tr));
fail_if (gst_rtsp_stream_remove_transport (stream, tr));
}
fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
gst-rtsp-server: Plug a few memory leaks in tests Found and fixed a few memory leaks in the gst_rtspserver, gst_onvif and gst_stream tests by running the tests in valgrind. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1742>
2022-02-18 15:05:53 +00:00
g_object_unref (tr);
rtsp-stream: Removing invalid transports returns false When removing transports an assertion was that the transports passed in for removal are present in the list, however that can't be assumed. As an example if a transport was removed from a thread running send_tcp_message, the main thread can try to remove the same transport again if it gets a handle_pause_request. This will not effect the transport list but it will effect n_tcp_transports as it will be decrement and then have the wrong value.
2019-11-01 11:01:41 +00:00
gst_object_unref (bin);
gst_object_unref (stream);
}
GST_START_TEST (test_add_transport_twice)
{
add_transports (TRUE);
}
GST_END_TEST;
GST_START_TEST (test_remove_transport_twice)
{
add_transports (FALSE);
}
GST_END_TEST;
tests: Don't fail tests if IPv6 not available. On computers with IPv6 disabled it shouldn't result in a test failure. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 03:26:02 +00:00
static gboolean
is_ipv6_supported (void)
{
GError *err = NULL;
GSocket *sock;
sock =
g_socket_new (G_SOCKET_FAMILY_IPV6, G_SOCKET_TYPE_DATAGRAM,
G_SOCKET_PROTOCOL_DEFAULT, &err);
if (sock) {
g_object_unref (sock);
return TRUE;
}
if (!g_error_matches (err, G_IO_ERROR, G_IO_ERROR_NOT_SUPPORTED)) {
GST_WARNING ("Unabled to create IPv6 socket: %s", err->message);
}
g_clear_error (&err);
return FALSE;
}
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
static Suite *
rtspstream_suite (void)
{
Suite *s = suite_create ("rtspstream");
TCase *tc = tcase_create ("general");
tests: Don't fail tests if IPv6 not available. On computers with IPv6 disabled it shouldn't result in a test failure. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 03:26:02 +00:00
gboolean have_ipv6 = is_ipv6_supported ();
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
suite_add_tcase (s, tc);
tests: Don't fail tests if IPv6 not available. On computers with IPv6 disabled it shouldn't result in a test failure. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 03:26:02 +00:00
tcase_add_test (tc, test_get_sockets_udp_ipv4);
tcase_add_test (tc, test_get_sockets_mcast_ipv4);
if (have_ipv6) {
tcase_add_test (tc, test_get_sockets_udp_ipv6);
tcase_add_test (tc, test_get_sockets_mcast_ipv6);
}
tests: unit test fixes 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
2016-02-23 14:01:22 +00:00
tcase_add_test (tc, test_allocate_udp_ports_fail);
stream tests: Make sure we get right multicast address from stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-03-16 16:29:48 +00:00
tcase_add_test (tc, test_get_multicast_address);
stream tests: added new tests Test a case when the address pool only contains multicast addresses and the client is requesting unicast udp. Added tests for multicast ports allocation. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 14:07:06 +00:00
tcase_add_test (tc, test_multicast_address_and_unicast_udp);
tcase_add_test (tc, test_allocate_udp_ports_multicast);
tcase_add_test (tc, test_allocate_udp_ports_client_settings);
rtsp-stream: fixed segmenation fault in _get_server_port() Calling function gst_rtsp_stream_get_server_port() results in segmenation fault in the RTP/RTSP/TCP case. Port that the server will use to receive RTCP makes only sense in the UDP case, however the function should handle the TCP case in a nicer way. https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 13:12:05 +00:00
tcase_add_test (tc, test_tcp_transport);
stream: Added a list of multicast client addresses When media is shared, the same media stream can be sent to multiple multicast groups. Currently, there is no API to retrieve multicast addresses from the stream. When calling gst_rtsp_stream_get_multicast_address() function, only the first multicast address is returned. With this patch, each multicast destination requested in SETUP will be stored in an internal list (call to gst_rtsp_stream_add_multicast_client_address()). The list of multicast groups requested by the clients can be retrieved by calling gst_rtsp_stream_get_multicast_client_addresses(). There still exist some problems with the current implementation in the multicast case: 1) The receiving part is currently only configured with regard to the first multicast client (see https://bugzilla.gnome.org/show_bug.cgi?id=796917). 2) Secondly, of security reasons, some constraints should be put on the requested multicast destinations (see https://bugzilla.gnome.org/show_bug.cgi?id=796916). Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-26 10:01:16 +00:00
tcase_add_test (tc, test_multicast_client_address);
tcase_add_test (tc, test_multicast_client_address_invalid);
rtsp-stream: Removing invalid transports returns false When removing transports an assertion was that the transports passed in for removal are present in the list, however that can't be assumed. As an example if a transport was removed from a thread running send_tcp_message, the main thread can try to remove the same transport again if it gets a handle_pause_request. This will not effect the transport list but it will effect n_tcp_transports as it will be decrement and then have the wrong value.
2019-11-01 11:01:41 +00:00
tcase_add_test (tc, test_add_transport_twice);
tcase_add_test (tc, test_remove_transport_twice);
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
return s;
}
GST_CHECK_MAIN (rtspstream);
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