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

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tests: add mediafactory test
2012-11-28 11:40:46 +00:00
/* GStreamer
* Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <gst/check/gstcheck.h>
#include <rtsp-media-factory.h>
GST_START_TEST (test_parse_error)
{
GstRTSPMediaFactory *factory;
GstRTSPUrl *url;
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory, "foo");
tests: Check gst_rtsp_url_parse return value See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:01:38 +00:00
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
ASSERT_CRITICAL (gst_rtsp_media_factory_create_element (factory, url));
ASSERT_CRITICAL (gst_rtsp_media_factory_construct (factory, url));
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
GST_START_TEST (test_launch)
{
GstRTSPMediaFactory *factory;
GstElement *element;
GstRTSPUrl *url;
factory = gst_rtsp_media_factory_new ();
fail_if (gst_rtsp_media_factory_is_shared (factory));
tests: Check gst_rtsp_url_parse return value See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:01:38 +00:00
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
element = gst_rtsp_media_factory_create_element (factory, url);
fail_unless (GST_IS_BIN (element));
fail_if (GST_IS_PIPELINE (element));
gst_object_unref (element);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
GST_START_TEST (test_launch_construct)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media, *media2;
GstRTSPUrl *url;
factory = gst_rtsp_media_factory_new ();
fail_if (gst_rtsp_media_factory_is_shared (factory));
tests: Check gst_rtsp_url_parse return value See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:01:38 +00:00
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
media2 = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media2));
fail_if (media == media2);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media2);
tests: Hold ref while creating second media To test if the media aren't shared, make sure we keep the first one while creating a second otherwise the same memory address may be reused.
2013-03-13 21:46:58 +00:00
g_object_unref (media);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
g_object_unref (media2);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
GST_START_TEST (test_shared)
{
GstRTSPMediaFactory *factory;
GstElement *element;
GstRTSPMedia *media, *media2;
GstRTSPUrl *url;
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_shared (factory, TRUE);
fail_unless (gst_rtsp_media_factory_is_shared (factory));
tests: Check gst_rtsp_url_parse return value See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:01:38 +00:00
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
element = gst_rtsp_media_factory_create_element (factory, url);
fail_unless (GST_IS_BIN (element));
fail_if (GST_IS_PIPELINE (element));
gst_object_unref (element);
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
media2 = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media2));
fail_unless (media == media2);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media2);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
g_object_unref (media);
g_object_unref (media2);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
tests: Check the passing around of a RTSPAddressPool Make sure the RTSPAddressPool is propagated from the MediaFactory all the way down to the stream.
2013-02-13 00:34:24 +00:00
GST_START_TEST (test_addresspool)
{
GstRTSPMediaFactory *factory;
GstElement *element;
GstRTSPMedia *media;
GstRTSPUrl *url;
GstRTSPAddressPool *pool, *tmppool;
GstRTSPStream *stream;
GstRTSPAddress *addr;
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_shared (factory, TRUE);
tests: Check gst_rtsp_url_parse return value See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:01:38 +00:00
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
tests: Check the passing around of a RTSPAddressPool Make sure the RTSPAddressPool is propagated from the MediaFactory all the way down to the stream.
2013-02-13 00:34:24 +00:00
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
" audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
pool = gst_rtsp_address_pool_new ();
fail_unless (gst_rtsp_address_pool_add_range (pool,
"233.252.0.1", "233.252.0.1", 5000, 5001, 3));
gst_rtsp_media_factory_set_address_pool (factory, pool);
tmppool = gst_rtsp_media_factory_get_address_pool (factory);
fail_unless (pool == tmppool);
g_object_unref (tmppool);
element = gst_rtsp_media_factory_create_element (factory, url);
fail_unless (GST_IS_BIN (element));
fail_if (GST_IS_PIPELINE (element));
gst_object_unref (element);
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
tmppool = gst_rtsp_media_get_address_pool (media);
fail_unless (pool == tmppool);
g_object_unref (tmppool);
fail_unless (gst_rtsp_media_n_streams (media) == 2);
stream = gst_rtsp_media_get_stream (media, 0);
fail_unless (stream != NULL);
tmppool = gst_rtsp_stream_get_address_pool (stream);
fail_unless (pool == tmppool);
g_object_unref (tmppool);
tests: fix compilation
2013-07-02 13:59:16 +00:00
addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
tests: Check the passing around of a RTSPAddressPool Make sure the RTSPAddressPool is propagated from the MediaFactory all the way down to the stream.
2013-02-13 00:34:24 +00:00
fail_unless (addr != NULL);
fail_unless (addr->port == 5000);
fail_unless (addr->n_ports == 2);
fail_unless (addr->ttl == 3);
gst_rtsp_address_free (addr);
stream = gst_rtsp_media_get_stream (media, 1);
fail_unless (stream != NULL);
tmppool = gst_rtsp_stream_get_address_pool (stream);
fail_unless (pool == tmppool);
g_object_unref (tmppool);
tests: fix compilation
2013-07-02 13:59:16 +00:00
addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
tests: Check the passing around of a RTSPAddressPool Make sure the RTSPAddressPool is propagated from the MediaFactory all the way down to the stream.
2013-02-13 00:34:24 +00:00
fail_unless (addr == NULL);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
tests: Check the passing around of a RTSPAddressPool Make sure the RTSPAddressPool is propagated from the MediaFactory all the way down to the stream.
2013-02-13 00:34:24 +00:00
g_object_unref (media);
g_object_unref (pool);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
tests: Test mediafactory permissions See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:50:47 +00:00
GST_START_TEST (test_permissions)
{
GstRTSPMediaFactory *factory;
GstRTSPPermissions *perms;
GstRTSPMedia *media;
GstRTSPUrl *url;
factory = gst_rtsp_media_factory_new ();
fail_if (gst_rtsp_media_factory_is_shared (factory));
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
gst_rtsp_media_factory_add_role (factory, "admin",
"media.factory.access", G_TYPE_BOOLEAN, TRUE,
"media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
perms = gst_rtsp_media_factory_get_permissions (factory);
fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
"media.factory.access"));
fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
"media.factory.construct"));
fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
"media.factory.access"));
gst_rtsp_permissions_unref (perms);
perms = gst_rtsp_permissions_new ();
gst_rtsp_permissions_add_role (perms, "user",
"media.factory.access", G_TYPE_BOOLEAN, TRUE,
"media.factory.construct", G_TYPE_BOOLEAN, FALSE, NULL);
gst_rtsp_media_factory_set_permissions (factory, perms);
gst_rtsp_permissions_unref (perms);
perms = gst_rtsp_media_factory_get_permissions (factory);
fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
"media.factory.access"));
fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
"media.factory.construct"));
fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
"media.factory.access"));
fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
"media.factory.construct"));
fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
"media.factory.access"));
gst_rtsp_permissions_unref (perms);
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
perms = gst_rtsp_media_get_permissions (media);
fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
"media.factory.access"));
fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
"media.factory.construct"));
fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
"media.factory.access"));
fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
"media.factory.construct"));
fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
"media.factory.access"));
gst_rtsp_permissions_unref (perms);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
tests: Test mediafactory permissions See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:50:47 +00:00
g_object_unref (media);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
GST_START_TEST (test_reset)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
GstRTSPUrl *url;
factory = gst_rtsp_media_factory_new ();
fail_if (gst_rtsp_media_factory_is_shared (factory));
gst_rtsp_url_parse ("rtsp://localhost:8554/test", &url);
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
fail_if (gst_rtsp_media_get_suspend_mode (media) !=
GST_RTSP_SUSPEND_MODE_NONE);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
g_object_unref (media);
gst_rtsp_media_factory_set_suspend_mode (factory,
GST_RTSP_SUSPEND_MODE_RESET);
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
fail_if (gst_rtsp_media_get_suspend_mode (media) !=
GST_RTSP_SUSPEND_MODE_RESET);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
g_object_unref (media);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
Add new API for setting/getting maximum multicast ttl value Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 07:35:46 +00:00
GST_START_TEST (test_mcast_ttl)
{
GstRTSPMediaFactory *factory;
GstElement *element;
GstRTSPMedia *media;
GstRTSPUrl *url;
GstRTSPStream *stream;
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_shared (factory, TRUE);
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
" audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
/* try to set an invalid ttl and make sure that the default ttl value (255) is
* set */
gst_rtsp_media_factory_set_max_mcast_ttl (factory, 0);
fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 255);
gst_rtsp_media_factory_set_max_mcast_ttl (factory, 300);
fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 255);
/* set a valid ttl value */
gst_rtsp_media_factory_set_max_mcast_ttl (factory, 3);
fail_unless (gst_rtsp_media_factory_get_max_mcast_ttl (factory) == 3);
element = gst_rtsp_media_factory_create_element (factory, url);
fail_unless (GST_IS_BIN (element));
fail_if (GST_IS_PIPELINE (element));
gst_object_unref (element);
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
fail_unless (gst_rtsp_media_n_streams (media) == 2);
fail_unless (gst_rtsp_media_get_max_mcast_ttl (media) == 3);
/* verify that the correct ttl value has been propageted to the media
* streams */
stream = gst_rtsp_media_get_stream (media, 0);
fail_unless (stream != NULL);
fail_unless (gst_rtsp_stream_get_max_mcast_ttl (stream) == 3);
stream = gst_rtsp_media_get_stream (media, 1);
fail_unless (stream != NULL);
fail_unless (gst_rtsp_stream_get_max_mcast_ttl (stream) == 3);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
Add new API for setting/getting maximum multicast ttl value Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 07:35:46 +00:00
g_object_unref (media);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
New property for socket binding to mcast addresses By default the multicast sockets are bound to INADDR_ANY, as it's not allowed to bind sockets to multicast addresses in Windows. This default behaviour can be changed by setting bind-mcast-address property on the media-factory object. https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-06 14:17:33 +00:00
GST_START_TEST (test_allow_bind_mcast)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
GstRTSPUrl *url;
GstRTSPStream *stream;
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_shared (factory, TRUE);
fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
&url) == GST_RTSP_OK);
gst_rtsp_media_factory_set_launch (factory,
"( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
" audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
/* verify that by default binding sockets to multicast addresses is not enabled */
fail_unless (gst_rtsp_media_factory_is_bind_mcast_address (factory) == FALSE);
/* allow multicast sockets to be bound to multicast addresses */
gst_rtsp_media_factory_set_bind_mcast_address (factory, TRUE);
/* verify that the socket binding to multicast address has been enabled */
fail_unless (gst_rtsp_media_factory_is_bind_mcast_address (factory) == TRUE);
media = gst_rtsp_media_factory_construct (factory, url);
fail_unless (GST_IS_RTSP_MEDIA (media));
/* verify that the correct socket binding configuration has been propageted to the media */
fail_unless (gst_rtsp_media_is_bind_mcast_address (media) == TRUE);
fail_unless (gst_rtsp_media_n_streams (media) == 2);
/* verify that the correct socket binding configuration has been propageted to the media streams */
stream = gst_rtsp_media_get_stream (media, 0);
fail_unless (stream != NULL);
fail_unless (gst_rtsp_stream_is_bind_mcast_address (stream) == TRUE);
stream = gst_rtsp_media_get_stream (media, 1);
fail_unless (stream != NULL);
fail_unless (gst_rtsp_stream_is_bind_mcast_address (stream) == TRUE);
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
gst_rtsp_media_unlock (media);
New property for socket binding to mcast addresses By default the multicast sockets are bound to INADDR_ANY, as it's not allowed to bind sockets to multicast addresses in Windows. This default behaviour can be changed by setting bind-mcast-address property on the media-factory object. https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-06 14:17:33 +00:00
g_object_unref (media);
gst_rtsp_url_free (url);
g_object_unref (factory);
}
GST_END_TEST;
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
static Suite *
rtspmediafactory_suite (void)
{
Suite *s = suite_create ("rtspmediafactory");
TCase *tc = tcase_create ("general");
suite_add_tcase (s, tc);
tcase_set_timeout (tc, 20);
tcase_add_test (tc, test_parse_error);
tcase_add_test (tc, test_launch);
tcase_add_test (tc, test_launch_construct);
tcase_add_test (tc, test_shared);
tests: Check the passing around of a RTSPAddressPool Make sure the RTSPAddressPool is propagated from the MediaFactory all the way down to the stream.
2013-02-13 00:34:24 +00:00
tcase_add_test (tc, test_addresspool);
tests: Test mediafactory permissions See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 16:50:47 +00:00
tcase_add_test (tc, test_permissions);
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
tcase_add_test (tc, test_reset);
Add new API for setting/getting maximum multicast ttl value Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 07:35:46 +00:00
tcase_add_test (tc, test_mcast_ttl);
New property for socket binding to mcast addresses By default the multicast sockets are bound to INADDR_ANY, as it's not allowed to bind sockets to multicast addresses in Windows. This default behaviour can be changed by setting bind-mcast-address property on the media-factory object. https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-06 14:17:33 +00:00
tcase_add_test (tc, test_allow_bind_mcast);
tests: add mediafactory test
2012-11-28 11:40:46 +00:00
return s;
}
GST_CHECK_MAIN (rtspmediafactory);
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