mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-09-22 12:00:11 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/tests/check/gst/client.c
David Svensson Fors a2e182c3b4 rtsp-client: Avoid reuse of channel numbers for interleaved 
If a (strange) client would reuse interleaved channel numbers in
multiple SETUP requests, we should not accept them. The channel
numbers are used for looking up stream transports in the
priv->transports hash table, and transports disappear from the table
if channel numbers are reused.

RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
server to change the channel numbers suggested by the client.

https://bugzilla.gnome.org/show_bug.cgi?id=796988
2018-08-29 14:46:01 +03:00

1502 lines
50 KiB
C
Raw Blame History

/* 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-client.h>
#define VIDEO_PIPELINE "videotestsrc ! " \
"video/x-raw,width=352,height=288 ! " \
"rtpgstpay name=pay0 pt=96"
#define AUDIO_PIPELINE "audiotestsrc ! " \
"audio/x-raw,rate=8000 ! " \
"rtpgstpay name=pay1 pt=97"
static gchar *session_id;
static gint cseq;
static guint expected_session_timeout = 60;
static const gchar *expected_unsupported_header;
static gboolean
test_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_OK);
fail_unless (g_str_equal (reason, "OK"));
fail_unless (version == GST_RTSP_VERSION_1_0);
return TRUE;
}
static gboolean
test_response_400 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_BAD_REQUEST);
fail_unless (g_str_equal (reason, "Bad Request"));
fail_unless (version == GST_RTSP_VERSION_1_0);
return TRUE;
}
static gboolean
test_response_404 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_NOT_FOUND);
fail_unless (g_str_equal (reason, "Not Found"));
fail_unless (version == GST_RTSP_VERSION_1_0);
return TRUE;
}
static gboolean
test_response_454 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_SESSION_NOT_FOUND);
fail_unless (g_str_equal (reason, "Session Not Found"));
fail_unless (version == GST_RTSP_VERSION_1_0);
return TRUE;
}
static gboolean
test_response_551 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *options;
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_OPTION_NOT_SUPPORTED);
fail_unless (g_str_equal (reason, "Option not supported"));
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED,
&options, 0) == GST_RTSP_OK);
fail_unless (!g_strcmp0 (expected_unsupported_header, options));
fail_unless (version == GST_RTSP_VERSION_1_0);
return TRUE;
}
static void
create_connection (GstRTSPConnection ** conn)
{
GSocket *sock;
GError *error = NULL;
sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
G_SOCKET_PROTOCOL_TCP, &error);
g_assert_no_error (error);
fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1", 444,
NULL, conn) == GST_RTSP_OK);
g_object_unref (sock);
}
static GstRTSPClient *
setup_client (const gchar * launch_line)
{
GstRTSPClient *client;
GstRTSPSessionPool *session_pool;
GstRTSPMountPoints *mount_points;
GstRTSPMediaFactory *factory;
GstRTSPThreadPool *thread_pool;
client = gst_rtsp_client_new ();
session_pool = gst_rtsp_session_pool_new ();
gst_rtsp_client_set_session_pool (client, session_pool);
mount_points = gst_rtsp_mount_points_new ();
factory = gst_rtsp_media_factory_new ();
if (launch_line == NULL)
gst_rtsp_media_factory_set_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
else
gst_rtsp_media_factory_set_launch (factory, launch_line);
gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
gst_rtsp_client_set_mount_points (client, mount_points);
thread_pool = gst_rtsp_thread_pool_new ();
gst_rtsp_client_set_thread_pool (client, thread_pool);
g_object_unref (mount_points);
g_object_unref (session_pool);
g_object_unref (thread_pool);
return client;
}
static void
teardown_client (GstRTSPClient * client)
{
gst_rtsp_client_set_thread_pool (client, NULL);
g_object_unref (client);
}
static gchar *
check_requirements_cb (GstRTSPClient * client, GstRTSPContext * ctx,
gchar ** req, gpointer user_data)
{
int index = 0;
GString *result = g_string_new ("");
while (req[index] != NULL) {
if (g_strcmp0 (req[index], "test-requirements")) {
if (result->len > 0)
g_string_append (result, ", ");
g_string_append (result, req[index]);
}
index++;
}
return g_string_free (result, FALSE);
}
GST_START_TEST (test_require)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
client = gst_rtsp_client_new ();
/* require header without handler */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("test-not-supported1");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
expected_unsupported_header = "test-not-supported1";
gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
g_signal_connect (G_OBJECT (client), "check-requirements",
G_CALLBACK (check_requirements_cb), NULL);
/* one supported option */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("test-requirements");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* unsupported option */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("test-not-supported1");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
expected_unsupported_header = "test-not-supported1";
gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* more than one unsupported options */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("test-not-supported1");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
str = g_strdup_printf ("test-not-supported2");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
expected_unsupported_header = "test-not-supported1, test-not-supported2";
gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* supported and unsupported together */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("test-not-supported1");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
str = g_strdup_printf ("test-requirements");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
str = g_strdup_printf ("test-not-supported2");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_REQUIRE, str);
g_free (str);
expected_unsupported_header = "test-not-supported1, test-not-supported2";
gst_rtsp_client_set_send_func (client, test_response_551, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
g_object_unref (client);
}
GST_END_TEST;
GST_START_TEST (test_request)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPConnection *conn;
client = gst_rtsp_client_new ();
/* OPTIONS with invalid url */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"foopy://padoop/") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* OPTIONS with unknown session id */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, "foobar");
gst_rtsp_client_set_send_func (client, test_response_454, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* OPTIONS with an absolute path instead of an absolute url */
/* set host information */
create_connection (&conn);
fail_unless (gst_rtsp_client_set_connection (client, conn));
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* OPTIONS with an absolute path instead of an absolute url with invalid
* host information */
g_object_unref (client);
client = gst_rtsp_client_new ();
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
g_object_unref (client);
}
GST_END_TEST;
static gboolean
test_option_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *str;
GstRTSPMethod methods;
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_OK);
fail_unless (g_str_equal (reason, "OK"));
fail_unless (version == GST_RTSP_VERSION_1_0);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
0) == GST_RTSP_OK);
fail_unless (atoi (str) == cseq++);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_PUBLIC, &str,
0) == GST_RTSP_OK);
methods = gst_rtsp_options_from_text (str);
fail_if (methods == 0);
fail_unless (methods == (GST_RTSP_DESCRIBE |
GST_RTSP_ANNOUNCE |
GST_RTSP_OPTIONS |
GST_RTSP_PAUSE |
GST_RTSP_PLAY |
GST_RTSP_RECORD |
GST_RTSP_SETUP |
GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN));
return TRUE;
}
GST_START_TEST (test_options)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
client = gst_rtsp_client_new ();
/* simple OPTIONS */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_option_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
g_object_unref (client);
}
GST_END_TEST;
GST_START_TEST (test_describe)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
client = gst_rtsp_client_new ();
/* simple DESCRIBE for non-existing url */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
g_object_unref (client);
/* simple DESCRIBE for an existing url */
client = setup_client (NULL);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
teardown_client (client);
}
GST_END_TEST;
static const gchar *expected_transport = NULL;
static gboolean
test_setup_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *str;
gchar *pattern;
GstRTSPSessionPool *session_pool;
GstRTSPSession *session;
gchar **session_hdr_params;
fail_unless (expected_transport != NULL);
fail_unless_equals_int (gst_rtsp_message_get_type (response),
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless_equals_int (code, GST_RTSP_STS_OK);
fail_unless_equals_string (reason, "OK");
fail_unless_equals_int (version, GST_RTSP_VERSION_1_0);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
0) == GST_RTSP_OK);
fail_unless (atoi (str) == cseq++);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT,
&str, 0) == GST_RTSP_OK);
pattern = g_strdup_printf ("^%s$", expected_transport);
fail_unless (g_regex_match_simple (pattern, str, 0, 0),
"Transport '%s' doesn't match pattern '%s'", str, pattern);
g_free (pattern);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
&str, 0) == GST_RTSP_OK);
session_hdr_params = g_strsplit (str, ";", -1);
/* session-id value */
fail_unless (session_hdr_params[0] != NULL);
if (expected_session_timeout != 60) {
/* session timeout param */
gchar *timeout_str = g_strdup_printf ("timeout=%u",
expected_session_timeout);
fail_unless (session_hdr_params[1] != NULL);
g_strstrip (session_hdr_params[1]);
fail_unless (g_strcmp0 (session_hdr_params[1], timeout_str) == 0);
g_free (timeout_str);
}
session_pool = gst_rtsp_client_get_session_pool (client);
fail_unless (session_pool != NULL);
session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
g_strfreev (session_hdr_params);
/* remember session id to be able to send teardown */
if (session_id)
g_free (session_id);
session_id = g_strdup (gst_rtsp_session_get_sessionid (session));
fail_unless (session_id != NULL);
fail_unless (session != NULL);
g_object_unref (session);
g_object_unref (session_pool);
return TRUE;
}
static gboolean
test_setup_response_461 (GstRTSPClient * client,
GstRTSPMessage * response, gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *str;
fail_unless (expected_transport == NULL);
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_UNSUPPORTED_TRANSPORT);
fail_unless (g_str_equal (reason, "Unsupported transport"));
fail_unless (version == GST_RTSP_VERSION_1_0);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
0) == GST_RTSP_OK);
fail_unless (atoi (str) == cseq++);
return TRUE;
}
static gboolean
test_teardown_response_200 (GstRTSPClient * client,
GstRTSPMessage * response, gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
&version)
== GST_RTSP_OK);
fail_unless (code == GST_RTSP_STS_OK);
fail_unless (g_str_equal (reason, "OK"));
fail_unless (version == GST_RTSP_VERSION_1_0);
return TRUE;
}
static void
send_teardown (GstRTSPClient * client)
{
GstRTSPMessage request = { 0, };
gchar *str;
fail_unless (session_id != NULL);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
gst_rtsp_client_set_send_func (client, test_teardown_response_200,
NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
g_free (session_id);
session_id = NULL;
}
GST_START_TEST (test_setup_tcp)
{
GstRTSPClient *client;
GstRTSPConnection *conn;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_client (NULL);
create_connection (&conn);
fail_unless (gst_rtsp_client_set_connection (client, conn));
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP/TCP;unicast");
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
expected_transport =
"RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\"";
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
send_teardown (client);
teardown_client (client);
}
GST_END_TEST;
GST_START_TEST (test_setup_tcp_two_streams_same_channels)
{
GstRTSPClient *client;
GstRTSPConnection *conn;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_client (NULL);
create_connection (&conn);
fail_unless (gst_rtsp_client_set_connection (client, conn));
/* test SETUP of a video stream with 0-1 as interleaved channels */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP/TCP;unicast;interleaved=0-1");
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
expected_transport =
"RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\"";
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* test SETUP of an audio stream with *the same* interleaved channels.
* we expect the server to allocate new channel numbers */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=1") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP/TCP;unicast;interleaved=0-1");
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
expected_transport =
"RTP/AVP/TCP;unicast;interleaved=2-3;ssrc=.*;mode=\"PLAY\"";
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
send_teardown (client);
teardown_client (client);
}
GST_END_TEST;
static GstRTSPClient *
setup_multicast_client (guint max_ttl)
{
GstRTSPClient *client;
GstRTSPSessionPool *session_pool;
GstRTSPMountPoints *mount_points;
GstRTSPMediaFactory *factory;
GstRTSPAddressPool *address_pool;
GstRTSPThreadPool *thread_pool;
client = gst_rtsp_client_new ();
session_pool = gst_rtsp_session_pool_new ();
gst_rtsp_client_set_session_pool (client, session_pool);
mount_points = gst_rtsp_mount_points_new ();
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory,
"audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
address_pool = gst_rtsp_address_pool_new ();
fail_unless (gst_rtsp_address_pool_add_range (address_pool,
"233.252.0.1", "233.252.0.1", 5000, 5010, 1));
gst_rtsp_media_factory_set_address_pool (factory, address_pool);
gst_rtsp_media_factory_add_role (factory, "user",
"media.factory.access", G_TYPE_BOOLEAN, TRUE,
"media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
gst_rtsp_client_set_mount_points (client, mount_points);
gst_rtsp_media_factory_set_max_mcast_ttl (factory, max_ttl);
thread_pool = gst_rtsp_thread_pool_new ();
gst_rtsp_client_set_thread_pool (client, thread_pool);
g_object_unref (mount_points);
g_object_unref (session_pool);
g_object_unref (address_pool);
g_object_unref (thread_pool);
return client;
}
GST_START_TEST (test_client_multicast_transport_404)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_multicast_client (1);
/* simple SETUP for non-existing url */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test2/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP;multicast");
gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
teardown_client (client);
}
GST_END_TEST;
static void
new_session_cb (GObject * client, GstRTSPSession * session, gpointer user_data)
{
GST_DEBUG ("%p: new session %p", client, session);
gst_rtsp_session_set_timeout (session, expected_session_timeout);
}
GST_START_TEST (test_client_multicast_transport)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_multicast_client (1);
expected_session_timeout = 20;
g_signal_connect (G_OBJECT (client), "new-session",
G_CALLBACK (new_session_cb), NULL);
/* simple SETUP with a valid URI and multicast */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP;multicast");
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
expected_transport = NULL;
expected_session_timeout = 60;
send_teardown (client);
teardown_client (client);
}
GST_END_TEST;
GST_START_TEST (test_client_multicast_ignore_transport_specific)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_multicast_client (1);
/* simple SETUP with a valid URI and multicast and a specific dest,
* but ignore it */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP;multicast;destination=233.252.0.2;ttl=2;port=5001-5006;");
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
expected_transport = NULL;
send_teardown (client);
teardown_client (client);
}
GST_END_TEST;
static void
multicast_transport_specific (void)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPSessionPool *session_pool;
GstRTSPContext ctx = { NULL };
client = setup_multicast_client (1);
ctx.client = client;
ctx.auth = gst_rtsp_auth_new ();
ctx.token =
gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
"user", NULL);
gst_rtsp_context_push_current (&ctx);
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
/* simple SETUP with a valid URI */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
expected_transport);
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
session_pool = gst_rtsp_client_get_session_pool (client);
fail_unless (session_pool != NULL);
fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
g_object_unref (session_pool);
/* send PLAY request */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
send_teardown (client);
teardown_client (client);
g_object_unref (ctx.auth);
gst_rtsp_token_unref (ctx.token);
gst_rtsp_context_pop_current (&ctx);
}
/* CASE: multicast address requested by the client exists in the address pool */
GST_START_TEST (test_client_multicast_transport_specific)
{
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
multicast_transport_specific ();
expected_transport = NULL;
}
GST_END_TEST;
/* CASE: multicast address requested by the client does not exist in the address pool */
GST_START_TEST (test_client_multicast_transport_specific_no_address_in_pool)
{
expected_transport = "RTP/AVP;multicast;destination=234.252.0.3;"
"ttl=1;port=6000-6001;mode=\"PLAY\"";
multicast_transport_specific ();
expected_transport = NULL;
}
GST_END_TEST;
static gboolean
test_response_sdp (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
guint8 *data;
guint size;
GstSDPMessage *sdp_msg;
const GstSDPMedia *sdp_media;
const GstSDPBandwidth *bw;
gint bandwidth_val = GPOINTER_TO_INT (user_data);
fail_unless (gst_rtsp_message_get_body (response, &data, &size)
== GST_RTSP_OK);
gst_sdp_message_new (&sdp_msg);
fail_unless (gst_sdp_message_parse_buffer (data, size, sdp_msg)
== GST_SDP_OK);
/* session description */
/* v= */
fail_unless (gst_sdp_message_get_version (sdp_msg) != NULL);
/* o= */
fail_unless (gst_sdp_message_get_origin (sdp_msg) != NULL);
/* s= */
fail_unless (gst_sdp_message_get_session_name (sdp_msg) != NULL);
/* t=0 0 */
fail_unless (gst_sdp_message_times_len (sdp_msg) == 0);
/* verify number of medias */
fail_unless (gst_sdp_message_medias_len (sdp_msg) == 1);
/* media description */
sdp_media = gst_sdp_message_get_media (sdp_msg, 0);
fail_unless (sdp_media != NULL);
/* m= */
fail_unless (gst_sdp_media_get_media (sdp_media) != NULL);
/* media bandwidth */
if (bandwidth_val) {
fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 1);
bw = gst_sdp_media_get_bandwidth (sdp_media, 0);
fail_unless (bw != NULL);
fail_unless (g_strcmp0 (bw->bwtype, "AS") == 0);
fail_unless (bw->bandwidth == bandwidth_val);
} else {
fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 0);
}
gst_sdp_message_free (sdp_msg);
return TRUE;
}
static void
test_client_sdp (const gchar * launch_line, guint * bandwidth_val)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
/* simple DESCRIBE for an existing url */
client = setup_client (launch_line);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
g_free (str);
gst_rtsp_client_set_send_func (client, test_response_sdp,
(gpointer) bandwidth_val, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
teardown_client (client);
}
GST_START_TEST (test_client_sdp_with_max_bitrate_tag)
{
test_client_sdp ("videotestsrc "
"! taginject tags=\"maximum-bitrate=(uint)50000000\" "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
GUINT_TO_POINTER (50000));
/* max-bitrate=0: no bandwidth line */
test_client_sdp ("videotestsrc "
"! taginject tags=\"maximum-bitrate=(uint)0\" "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
GUINT_TO_POINTER (0));
}
GST_END_TEST;
GST_START_TEST (test_client_sdp_with_bitrate_tag)
{
test_client_sdp ("videotestsrc "
"! taginject tags=\"bitrate=(uint)7000000\" "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
GUINT_TO_POINTER (7000));
/* bitrate=0: no bandwdith line */
test_client_sdp ("videotestsrc "
"! taginject tags=\"bitrate=(uint)0\" "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
GUINT_TO_POINTER (0));
}
GST_END_TEST;
GST_START_TEST (test_client_sdp_with_max_bitrate_and_bitrate_tags)
{
test_client_sdp ("videotestsrc "
"! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)50000000\" "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
GUINT_TO_POINTER (50000));
/* max-bitrate is zero: fallback to bitrate */
test_client_sdp ("videotestsrc "
"! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)0\" "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
GUINT_TO_POINTER (7000));
/* max-bitrate=bitrate=0o: no bandwidth line */
test_client_sdp ("videotestsrc "
"! taginject tags=\"bitrate=(uint)0,maximum-bitrate=(uint)0\" "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
GUINT_TO_POINTER (0));
}
GST_END_TEST;
GST_START_TEST (test_client_sdp_with_no_bitrate_tags)
{
test_client_sdp ("videotestsrc "
"! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", NULL);
}
GST_END_TEST;
static void
mcast_transport_two_clients (gboolean shared, const gchar * transport1,
const gchar * expected_transport1, const gchar * addr1,
const gchar * transport2, const gchar * expected_transport2,
const gchar * addr2)
{
GstRTSPClient *client1, *client2;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPSessionPool *session_pool;
GstRTSPContext ctx = { NULL };
GstRTSPContext ctx2 = { NULL };
GstRTSPMountPoints *mount_points;
GstRTSPMediaFactory *factory;
GstRTSPAddressPool *address_pool;
GstRTSPThreadPool *thread_pool;
gchar *session_id1;
gchar *client_addr = NULL;
mount_points = gst_rtsp_mount_points_new ();
factory = gst_rtsp_media_factory_new ();
if (shared)
gst_rtsp_media_factory_set_shared (factory, TRUE);
gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5);
gst_rtsp_media_factory_set_launch (factory,
"audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
address_pool = gst_rtsp_address_pool_new ();
fail_unless (gst_rtsp_address_pool_add_range (address_pool,
"233.252.0.1", "233.252.0.1", 5000, 5001, 1));
gst_rtsp_media_factory_set_address_pool (factory, address_pool);
gst_rtsp_media_factory_add_role (factory, "user",
"media.factory.access", G_TYPE_BOOLEAN, TRUE,
"media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
session_pool = gst_rtsp_session_pool_new ();
thread_pool = gst_rtsp_thread_pool_new ();
/* first multicast client with transport specific request */
client1 = gst_rtsp_client_new ();
gst_rtsp_client_set_session_pool (client1, session_pool);
gst_rtsp_client_set_mount_points (client1, mount_points);
gst_rtsp_client_set_thread_pool (client1, thread_pool);
ctx.client = client1;
ctx.auth = gst_rtsp_auth_new ();
ctx.token =
gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
"user", NULL);
gst_rtsp_context_push_current (&ctx);
expected_transport = expected_transport1;
/* send SETUP request */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport1);
gst_rtsp_client_set_send_func (client1, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client1,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
expected_transport = NULL;
/* send PLAY request */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
gst_rtsp_client_set_send_func (client1, test_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client1,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* check address */
client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx.stream);
fail_if (client_addr == NULL);
fail_unless (g_str_equal (client_addr, addr1));
g_free (client_addr);
gst_rtsp_context_pop_current (&ctx);
session_id1 = g_strdup (session_id);
/* second multicast client with transport specific request */
cseq = 0;
client2 = gst_rtsp_client_new ();
gst_rtsp_client_set_session_pool (client2, session_pool);
gst_rtsp_client_set_mount_points (client2, mount_points);
gst_rtsp_client_set_thread_pool (client2, thread_pool);
ctx2.client = client2;
ctx2.auth = gst_rtsp_auth_new ();
ctx2.token =
gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
"user", NULL);
gst_rtsp_context_push_current (&ctx2);
expected_transport = expected_transport2;
/* send SETUP request */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport2);
gst_rtsp_client_set_send_func (client2, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client2,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
expected_transport = NULL;
/* send PLAY request */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
"rtsp://localhost/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
gst_rtsp_client_set_send_func (client2, test_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client2,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
/* check addresses */
client_addr = gst_rtsp_stream_get_multicast_client_addresses (ctx2.stream);
fail_if (client_addr == NULL);
if (shared) {
if (g_str_equal (addr1, addr2)) {
fail_unless (g_str_equal (client_addr, addr1));
} else {
gchar *addr_str = g_strdup_printf ("%s,%s", addr2, addr1);
fail_unless (g_str_equal (client_addr, addr_str));
g_free (addr_str);
}
} else {
fail_unless (g_str_equal (client_addr, addr2));
}
g_free (client_addr);
send_teardown (client2);
gst_rtsp_context_pop_current (&ctx2);
gst_rtsp_context_push_current (&ctx);
session_id = session_id1;
send_teardown (client1);
gst_rtsp_context_pop_current (&ctx);
teardown_client (client1);
teardown_client (client2);
g_object_unref (ctx.auth);
g_object_unref (ctx2.auth);
gst_rtsp_token_unref (ctx.token);
gst_rtsp_token_unref (ctx2.token);
g_object_unref (mount_points);
g_object_unref (session_pool);
g_object_unref (address_pool);
g_object_unref (thread_pool);
}
/* test if two multicast clients can choose different transport settings
* CASE: media is shared */
GST_START_TEST
(test_client_multicast_transport_specific_two_clients_shared_media) {
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=1;port=5002-5003;mode=\"PLAY\"";
const gchar *expected_transport_2 = transport_client_2;
const gchar *addr_client_2 = "233.252.0.2:5002";
mcast_transport_two_clients (TRUE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
/* test if two multicast clients can choose different transport settings
* CASE: media is not shared */
GST_START_TEST (test_client_multicast_transport_specific_two_clients)
{
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=1;port=5002-5003;mode=\"PLAY\"";
const gchar *expected_transport_2 = transport_client_2;
const gchar *addr_client_2 = "233.252.0.2:5002";
mcast_transport_two_clients (FALSE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
/* test if two multicast clients can choose the same transport settings.
* CASE: media is shared */
GST_START_TEST
(test_client_multicast_transport_specific_two_clients_shared_media_same_transport)
{
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = transport_client_1;
const gchar *expected_transport_2 = expected_transport_1;
const gchar *addr_client_2 = addr_client_1;
mcast_transport_two_clients (TRUE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
/* test if two multicast clients get the same transport settings without
* requesting specific transport.
* CASE: media is shared */
GST_START_TEST (test_client_multicast_two_clients_shared_media)
{
const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
const gchar *expected_transport_1 =
"RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = transport_client_1;
const gchar *expected_transport_2 = expected_transport_1;
const gchar *addr_client_2 = addr_client_1;
mcast_transport_two_clients (TRUE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
/* test if two multicast clients get the different transport settings: the first client
* requests the specific transport configuration while the second client lets
* the server select the multicast address and the ports.
* CASE: media is shared */
GST_START_TEST
(test_client_multicast_two_clients_first_specific_transport_shared_media) {
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = "RTP/AVP;multicast;mode=\"PLAY\"";
const gchar *expected_transport_2 = expected_transport_1;
const gchar *addr_client_2 = addr_client_1;
mcast_transport_two_clients (TRUE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
/* test if two multicast clients get the different transport settings: the first client lets
* the server select the multicast address and the ports while the second client requests
* the specific transport configuration.
* CASE: media is shared */
GST_START_TEST
(test_client_multicast_two_clients_second_specific_transport_shared_media) {
const gchar *transport_client_1 = "RTP/AVP;multicast;mode=\"PLAY\"";
const gchar *expected_transport_1 =
"RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=2;port=5004-5005;mode=\"PLAY\"";
const gchar *expected_transport_2 = transport_client_2;
const gchar *addr_client_2 = "233.252.0.2:5004";
mcast_transport_two_clients (TRUE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
/* test if the maximum ttl multicast value is chosen by the server
* CASE: the first client provides the highest ttl value */
GST_START_TEST (test_client_multicast_max_ttl_first_client)
{
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=3;port=5000-5001;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=1;port=5002-5003;mode=\"PLAY\"";
const gchar *expected_transport_2 =
"RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=3;port=5002-5003;mode=\"PLAY\"";
const gchar *addr_client_2 = "233.252.0.2:5002";
mcast_transport_two_clients (TRUE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
/* test if the maximum ttl multicast value is chosen by the server
* CASE: the second client provides the highest ttl value */
GST_START_TEST (test_client_multicast_max_ttl_second_client)
{
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=2;port=5000-5001;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.1:5000";
const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=4;port=5002-5003;mode=\"PLAY\"";
const gchar *expected_transport_2 = transport_client_2;
const gchar *addr_client_2 = "233.252.0.2:5002";
mcast_transport_two_clients (TRUE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2);
}
GST_END_TEST;
GST_START_TEST (test_client_multicast_invalid_ttl)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPSessionPool *session_pool;
GstRTSPContext ctx = { NULL };
client = setup_multicast_client (3);
ctx.client = client;
ctx.auth = gst_rtsp_auth_new ();
ctx.token =
gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
"user", NULL);
gst_rtsp_context_push_current (&ctx);
/* simple SETUP with an invalid ttl=0 */
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
"rtsp://localhost/test/stream=0") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP;multicast;destination=233.252.0.1;ttl=0;port=5000-5001;");
gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
session_pool = gst_rtsp_client_get_session_pool (client);
fail_unless (session_pool != NULL);
fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
g_object_unref (session_pool);
teardown_client (client);
g_object_unref (ctx.auth);
gst_rtsp_token_unref (ctx.token);
gst_rtsp_context_pop_current (&ctx);
}
GST_END_TEST;
static Suite *
rtspclient_suite (void)
{
Suite *s = suite_create ("rtspclient");
TCase *tc = tcase_create ("general");
suite_add_tcase (s, tc);
tcase_set_timeout (tc, 20);
tcase_add_test (tc, test_require);
tcase_add_test (tc, test_request);
tcase_add_test (tc, test_options);
tcase_add_test (tc, test_describe);
tcase_add_test (tc, test_setup_tcp);
tcase_add_test (tc, test_setup_tcp_two_streams_same_channels);
tcase_add_test (tc, test_client_multicast_transport_404);
tcase_add_test (tc, test_client_multicast_transport);
tcase_add_test (tc, test_client_multicast_ignore_transport_specific);
tcase_add_test (tc, test_client_multicast_transport_specific);
tcase_add_test (tc, test_client_sdp_with_max_bitrate_tag);
tcase_add_test (tc, test_client_sdp_with_bitrate_tag);
tcase_add_test (tc, test_client_sdp_with_max_bitrate_and_bitrate_tags);
tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags);
tcase_add_test (tc,
test_client_multicast_transport_specific_two_clients_shared_media);
tcase_add_test (tc, test_client_multicast_transport_specific_two_clients);
tcase_add_test (tc,
test_client_multicast_transport_specific_two_clients_shared_media_same_transport);
tcase_add_test (tc, test_client_multicast_two_clients_shared_media);
tcase_add_test (tc,
test_client_multicast_two_clients_first_specific_transport_shared_media);
tcase_add_test (tc,
test_client_multicast_two_clients_second_specific_transport_shared_media);
tcase_add_test (tc,
test_client_multicast_transport_specific_no_address_in_pool);
tcase_add_test (tc, test_client_multicast_max_ttl_first_client);
tcase_add_test (tc, test_client_multicast_max_ttl_second_client);
tcase_add_test (tc, test_client_multicast_invalid_ttl);
return s;
}
GST_CHECK_MAIN (rtspclient);
Reference in a new issue View git blame Copy permalink