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

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/* 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 const gchar *expected_scale_header;
static const gchar *expected_speed_header;
static gdouble fake_rate_value = 0;
static gdouble fake_applied_rate_value = 0;
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_play_200 (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *str;
gchar **session_hdr_params;
gchar *pattern;
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);
/* Verify mandatory headers according to RFC 2326 */
/* verify mandatory CSeq header */
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
0) == GST_RTSP_OK);
fail_unless (atoi (str) == cseq++);
/* verify mandatory Session header */
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
&str, 0) == GST_RTSP_OK);
session_hdr_params = g_strsplit (str, ";", -1);
fail_unless (session_hdr_params[0] != NULL);
g_strfreev (session_hdr_params);
/* verify mandatory RTP-Info header */
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_RTP_INFO,
&str, 0) == GST_RTSP_OK);
pattern = g_strdup_printf ("^url=rtsp://.+;seq=[0-9]+;rtptime=[0-9]+");
fail_unless (g_regex_match_simple (pattern, str, 0, 0),
"GST_RTSP_HDR_RTP_INFO '%s' doesn't match pattern '%s'", str, pattern);
g_free (pattern);
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, const gchar * mount_point,
gboolean enable_rtcp)
{
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_media_factory_set_enable_rtcp (factory, enable_rtcp);
gst_rtsp_mount_points_add_factory (mount_points, mount_point, 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;
static void
test_describe_sub (const gchar * mount_point, const gchar * url)
{
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,
url) == 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, mount_point, TRUE);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
url) == 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_START_TEST (test_describe)
{
test_describe_sub ("/test", "rtsp://localhost/test");
}
GST_END_TEST;
GST_START_TEST (test_describe_root_mount_point)
{
test_describe_sub ("/", "rtsp://localhost");
}
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, const gchar * url)
{
GstRTSPMessage request = { 0, };
gchar *str;
fail_unless (session_id != NULL);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN,
url) == 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;
}
static void
test_setup_tcp_sub (const gchar * mount_point, const gchar * url1,
const gchar * url2)
{
GstRTSPClient *client;
GstRTSPConnection *conn;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_client (NULL, mount_point, TRUE);
create_connection (&conn);
fail_unless (gst_rtsp_client_set_connection (client, conn));
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
url1) == 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, url2);
teardown_client (client);
}
GST_START_TEST (test_setup_tcp)
{
test_setup_tcp_sub ("/test", "rtsp://localhost/test/stream=0",
"rtsp://localhost/test");
}
GST_END_TEST;
GST_START_TEST (test_setup_tcp_root_mount_point)
{
test_setup_tcp_sub ("/", "rtsp://localhost/stream=0", "rtsp://localhost");
}
GST_END_TEST;
GST_START_TEST (test_setup_no_rtcp)
{
GstRTSPClient *client;
GstRTSPConnection *conn;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_client (NULL, "/test", FALSE);
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;unicast;client_port=5000-5001");
gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
/* We want to verify that server_port holds a single number, not a range */
expected_transport =
"RTP/AVP;unicast;client_port=5000-5001;server_port=[0-9]+;ssrc=.*;mode=\"PLAY\"";
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
send_teardown (client, "rtsp://localhost/test");
teardown_client (client);
}
GST_END_TEST;
static void
test_setup_tcp_two_streams_same_channels_sub (const gchar * mount_point,
const gchar * url1, const gchar * url2, const gchar * url3)
{
GstRTSPClient *client;
GstRTSPConnection *conn;
GstRTSPMessage request = { 0, };
gchar *str;
client = setup_client (NULL, mount_point, TRUE);
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,
url1) == 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,
url2) == 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, url3);
teardown_client (client);
}
GST_START_TEST (test_setup_tcp_two_streams_same_channels)
{
test_setup_tcp_two_streams_same_channels_sub ("/test",
"rtsp://localhost/test/stream=0", "rtsp://localhost/test/stream=1",
"rtsp://localhost/test");
}
GST_END_TEST;
GST_START_TEST (test_setup_tcp_two_streams_same_channels_root_mount_point)
{
test_setup_tcp_two_streams_same_channels_sub ("/",
"rtsp://localhost/stream=0", "rtsp://localhost/stream=1",
"rtsp://localhost");
}
GST_END_TEST;
static GstRTSPClient *
setup_multicast_client (guint max_ttl, const gchar * mount_point)
{
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, mount_point, 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, "/test");
/* 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, "/test");
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, "rtsp://localhost/test");
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, "/test");
/* 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, "rtsp://localhost/test");
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, "/test");
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 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, "rtsp://localhost/test");
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=10002-10004;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, "/test", TRUE);
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, gboolean bind_mcast_address)
{
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_bind_mcast_address (factory, bind_mcast_address);
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, "rtsp://localhost/test");
gst_rtsp_context_pop_current (&ctx2);
gst_rtsp_context_push_current (&ctx);
session_id = session_id1;
send_teardown (client1, "rtsp://localhost/test");
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);
}
/* CASE: media is shared.
* client 1: SETUP --->
* client 1: PLAY --->
* client 2: SETUP --->
* client 1: TEARDOWN --->
* client 2: PLAY --->
* client 2: TEARDOWN --->
*/
static void
mcast_transport_two_clients_teardown_play (const gchar * transport1,
const gchar * expected_transport1, const gchar * transport2,
const gchar * expected_transport2, gboolean bind_mcast_address,
gboolean is_shared)
{
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, *session_id2;
mount_points = gst_rtsp_mount_points_new ();
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_shared (factory, is_shared);
gst_rtsp_media_factory_set_max_mcast_ttl (factory, 5);
gst_rtsp_media_factory_set_bind_mcast_address (factory, bind_mcast_address);
gst_rtsp_media_factory_set_launch (factory,
"audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
address_pool = gst_rtsp_address_pool_new ();
if (is_shared)
fail_unless (gst_rtsp_address_pool_add_range (address_pool,
"233.252.0.1", "233.252.0.1", 5000, 5001, 1));
else
fail_unless (gst_rtsp_address_pool_add_range (address_pool,
"233.252.0.1", "233.252.0.1", 5000, 5003, 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 ();
/* client 1 configuration */
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;
/* client 1 sends 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;
/* client 1 sends 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);
gst_rtsp_context_pop_current (&ctx);
session_id1 = g_strdup (session_id);
/* client 2 configuration */
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;
/* client 2 sends 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;
session_id2 = g_strdup (session_id);
g_free (session_id);
gst_rtsp_context_pop_current (&ctx2);
/* the first client sends TEARDOWN request */
gst_rtsp_context_push_current (&ctx);
session_id = session_id1;
send_teardown (client1, "rtsp://localhost/test");
gst_rtsp_context_pop_current (&ctx);
teardown_client (client1);
/* the second client sends PLAY request */
gst_rtsp_context_push_current (&ctx2);
session_id = session_id2;
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);
/* client 2 sends TEARDOWN request */
send_teardown (client2, "rtsp://localhost/test");
gst_rtsp_context_pop_current (&ctx2);
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, FALSE);
}
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, FALSE);
}
GST_END_TEST;
/* test if two multicast clients can choose the same ports but different
* multicast destinations
* CASE: media is not shared */
GST_START_TEST (test_client_multicast_transport_specific_two_clients_same_ports)
{
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=9000-9001;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.1:9000";
const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=1;port=9000-9001;mode=\"PLAY\"";
const gchar *expected_transport_2 = transport_client_2;
const gchar *addr_client_2 = "233.252.0.2:9000";
/* configure the multicast socket to be bound to the requested multicast address instead of INADDR_ANY.
* The clients request the same rtp/rtcp borts and having the socket that are bound to ANY would result
* in bind() failure */
gboolean allow_bind_mcast_address = TRUE;
mcast_transport_two_clients (FALSE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2, allow_bind_mcast_address);
}
GST_END_TEST;
/* test if two multicast clients can choose the same multicast destination but different
* ports
* CASE: media is not shared */
GST_START_TEST
(test_client_multicast_transport_specific_two_clients_same_destination) {
const gchar *transport_client_1 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=1;port=9002-9003;mode=\"PLAY\"";
const gchar *expected_transport_1 = transport_client_1;
const gchar *addr_client_1 = "233.252.0.2:9002";
const gchar *transport_client_2 = "RTP/AVP;multicast;destination=233.252.0.2;"
"ttl=1;port=9004-9005;mode=\"PLAY\"";
const gchar *expected_transport_2 = transport_client_2;
const gchar *addr_client_2 = "233.252.0.2:9004";
mcast_transport_two_clients (FALSE, transport_client_1,
expected_transport_1, addr_client_1, transport_client_2,
expected_transport_2, addr_client_2, FALSE);
}
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, FALSE);
}
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, FALSE);
}
GST_END_TEST;
/* test if it's possible to play the shared media, after one of the clients
* has terminated its session.
*/
GST_START_TEST (test_client_multicast_two_clients_shared_media_teardown_play)
{
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 *transport_client_2 = transport_client_1;
const gchar *expected_transport_2 = expected_transport_1;
mcast_transport_two_clients_teardown_play (transport_client_1,
expected_transport_1, transport_client_2, expected_transport_2, FALSE,
TRUE);
}
GST_END_TEST;
/* test if it's possible to play the shared media, after one of the clients
* has terminated its session.
*/
GST_START_TEST
(test_client_multicast_two_clients_not_shared_media_teardown_play) {
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 *transport_client_2 = transport_client_1;
const gchar *expected_transport_2 =
"RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5002-5003;mode=\"PLAY\"";
mcast_transport_two_clients_teardown_play (transport_client_1,
expected_transport_1, transport_client_2, expected_transport_2, FALSE,
FALSE);
}
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, FALSE);
}
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, FALSE);
}
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, FALSE);
}
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, FALSE);
}
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, "/test");
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 gboolean
test_response_scale_speed (GstRTSPClient * client, GstRTSPMessage * response,
gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *header_value;
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_RANGE,
&header_value, 0) == GST_RTSP_OK);
if (expected_scale_header != NULL) {
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SCALE,
&header_value, 0) == GST_RTSP_OK);
ck_assert_str_eq (header_value, expected_scale_header);
} else {
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SCALE,
&header_value, 0) == GST_RTSP_ENOTIMPL);
}
if (expected_speed_header != NULL) {
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SPEED,
&header_value, 0) == GST_RTSP_OK);
ck_assert_str_eq (header_value, expected_speed_header);
} else {
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SPEED,
&header_value, 0) == GST_RTSP_ENOTIMPL);
}
return TRUE;
}
/* Probe that tweaks segment events according to the values of the
* fake_rate_value and fake_applied_rate_value variables. Used to simulate
* seek results with different combinations of rate and applied rate.
*/
static GstPadProbeReturn
rate_tweaking_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
GstEvent *event = GST_PAD_PROBE_INFO_EVENT (info);
GstSegment segment;
if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
GST_DEBUG ("got segment event %" GST_PTR_FORMAT, event);
gst_event_copy_segment (event, &segment);
if (fake_applied_rate_value)
segment.applied_rate = fake_applied_rate_value;
if (fake_rate_value)
segment.rate = fake_rate_value;
gst_event_unref (event);
info->data = gst_event_new_segment (&segment);
GST_DEBUG ("forwarding segment event %" GST_PTR_FORMAT,
GST_EVENT (info->data));
}
return GST_PAD_PROBE_OK;
}
static void
attach_rate_tweaking_probe (void)
{
GstRTSPContext *ctx;
GstRTSPMedia *media;
GstRTSPStream *stream;
GstPad *srcpad;
fail_unless ((ctx = gst_rtsp_context_get_current ()) != NULL);
media = ctx->media;
fail_unless (media != NULL);
stream = gst_rtsp_media_get_stream (media, 0);
fail_unless (stream != NULL);
srcpad = gst_rtsp_stream_get_srcpad (stream);
fail_unless (srcpad != NULL);
GST_DEBUG ("adding rate_tweaking_probe");
gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
rate_tweaking_probe, NULL, NULL);
gst_object_unref (srcpad);
}
static void
do_test_scale_and_speed (const gchar * scale, const gchar * speed,
GstRTSPStatusCode expected_response_code)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPContext ctx = { NULL };
client = setup_multicast_client (1, "/test");
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_session_timeout = 20;
g_signal_connect (G_OBJECT (client), "new-session",
G_CALLBACK (new_session_cb), NULL);
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=.*;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;
if (fake_applied_rate_value || fake_rate_value)
attach_rate_tweaking_probe ();
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);
if (scale != NULL)
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, scale);
if (speed != NULL)
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, speed);
if (expected_response_code == GST_RTSP_STS_BAD_REQUEST)
gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
else
gst_rtsp_client_set_send_func (client, test_response_scale_speed, NULL,
NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
send_teardown (client, "rtsp://localhost/test");
teardown_client (client);
g_object_unref (ctx.auth);
gst_rtsp_token_unref (ctx.token);
gst_rtsp_context_pop_current (&ctx);
}
GST_START_TEST (test_scale_and_speed)
{
/* no scale/speed requested, no scale/speed should be received */
expected_scale_header = NULL;
expected_speed_header = NULL;
do_test_scale_and_speed (NULL, NULL, GST_RTSP_STS_OK);
/* scale requested, scale should be received */
fake_applied_rate_value = 2;
fake_rate_value = 1;
expected_scale_header = "2.000";
expected_speed_header = NULL;
do_test_scale_and_speed ("2.000", NULL, GST_RTSP_STS_OK);
/* speed requested, speed should be received */
fake_applied_rate_value = 0;
fake_rate_value = 0;
expected_scale_header = NULL;
expected_speed_header = "2.000";
do_test_scale_and_speed (NULL, "2.000", GST_RTSP_STS_OK);
/* both requested, both should be received */
fake_applied_rate_value = 2;
fake_rate_value = 2;
expected_scale_header = "2.000";
expected_speed_header = "2.000";
do_test_scale_and_speed ("2", "2", GST_RTSP_STS_OK);
/* scale requested but media doesn't handle scaling so both should be
* received, with scale set to 1.000 and speed set to (requested scale
* requested speed) */
fake_applied_rate_value = 0;
fake_rate_value = 5;
expected_scale_header = "1.000";
expected_speed_header = "5.000";
do_test_scale_and_speed ("5", NULL, GST_RTSP_STS_OK);
/* both requested but media only handles scaling so both should be received,
* with scale set to (requested scale * requested speed) and speed set to 1.00
*/
fake_rate_value = 1.000;
fake_applied_rate_value = 4.000;
expected_scale_header = "4.000";
expected_speed_header = "1.000";
do_test_scale_and_speed ("2", "2", GST_RTSP_STS_OK);
/* test invalid values */
fake_applied_rate_value = 0;
fake_rate_value = 0;
expected_scale_header = NULL;
expected_speed_header = NULL;
/* scale or speed not decimal values */
do_test_scale_and_speed ("x", NULL, GST_RTSP_STS_BAD_REQUEST);
do_test_scale_and_speed (NULL, "y", GST_RTSP_STS_BAD_REQUEST);
/* scale or speed illegal decimal values */
do_test_scale_and_speed ("0", NULL, GST_RTSP_STS_BAD_REQUEST);
do_test_scale_and_speed (NULL, "0", GST_RTSP_STS_BAD_REQUEST);
do_test_scale_and_speed (NULL, "-2", GST_RTSP_STS_BAD_REQUEST);
}
GST_END_TEST static void
test_client_play_sub (const gchar * mount_point, const gchar * url1,
const gchar * url2)
{
GstRTSPClient *client;
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPContext ctx = { NULL };
client = setup_multicast_client (1, mount_point);
ctx.client = client;
ctx.auth = gst_rtsp_auth_new ();
ctx.token =
gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
"user", NULL);
gst_rtsp_context_push_current (&ctx);
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
url1) == 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 is from adress pool */
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=.*;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;
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_PLAY,
url2) == 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_play_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
send_teardown (client, url2);
teardown_client (client);
g_object_unref (ctx.auth);
gst_rtsp_token_unref (ctx.token);
gst_rtsp_context_pop_current (&ctx);
}
GST_START_TEST (test_client_play)
{
test_client_play_sub ("/test", "rtsp://localhost/test/stream=0",
"rtsp://localhost/test");
}
GST_END_TEST;
GST_START_TEST (test_client_play_root_mount_point)
{
test_client_play_sub ("/", "rtsp://localhost/stream=0", "rtsp://localhost");
}
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_describe_root_mount_point);
tcase_add_test (tc, test_setup_tcp);
tcase_add_test (tc, test_setup_tcp_root_mount_point);
tcase_add_test (tc, test_setup_no_rtcp);
tcase_add_test (tc, test_setup_tcp_two_streams_same_channels);
tcase_add_test (tc,
test_setup_tcp_two_streams_same_channels_root_mount_point);
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);
#ifndef G_OS_WIN32
tcase_add_test (tc,
test_client_multicast_transport_specific_two_clients_same_ports);
#else
/* skip the test on windows as the test restricts the multicast sockets to multicast traffic only,
* by specifying the multicast IP as the bind address and this currently doesn't work on Windows */
tcase_skip_broken_test (tc,
test_client_multicast_transport_specific_two_clients_same_ports);
#endif
tcase_add_test (tc,
test_client_multicast_transport_specific_two_clients_same_destination);
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_shared_media_teardown_play);
tcase_add_test (tc,
test_client_multicast_two_clients_not_shared_media_teardown_play);
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);
tcase_add_test (tc, test_scale_and_speed);
tcase_add_test (tc, test_client_play);
tcase_add_test (tc, test_client_play_root_mount_point);
return s;
}
GST_CHECK_MAIN (rtspclient);
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