mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-22 16:26:39 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-rtsp-server/tests/check/gst/rtspserver.c
Branko Subasic 52c0763042 gst-rtsp-server: Plug a few memory leaks in tests 
Found and fixed a few memory leaks in the gst_rtspserver, gst_onvif and
gst_stream tests by running the tests in valgrind.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1742>
2022-03-07 13:57:27 +00:00

2754 lines
82 KiB
C
Raw Blame History

/* GStreamer unit test for GstRTSPServer
* Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
* @author David Svensson Fors <davidsf at axis dot com>
* Copyright (C) 2015 Centricular Ltd
* @author Tim-Philipp Müller <tim@centricular.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 <gst/sdp/gstsdpmessage.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <stdio.h>
#include <netinet/in.h>
#include "rtsp-server.h"
#define ERRORIGNORE "errorignore ignore-error=false ignore-notlinked=true " \
"ignore-notnegotiated=false convert-to=ok"
#define VIDEO_PIPELINE "videotestsrc ! " \
ERRORIGNORE " ! " \
"video/x-raw,format=I420,width=352,height=288 ! " \
"rtpgstpay name=pay0 pt=96"
#define AUDIO_PIPELINE "audiotestsrc ! " \
ERRORIGNORE " ! " \
"audio/x-raw,rate=8000 ! " \
"rtpgstpay name=pay1 pt=97"
#define TEST_MOUNT_POINT "/test"
#define TEST_PROTO "RTP/AVP"
#define TEST_ENCODING "X-GST"
#define TEST_CLOCK_RATE "90000"
/* tested rtsp server */
static GstRTSPServer *server = NULL;
/* tcp port that the test server listens for rtsp requests on */
static gint test_port = 0;
/* id of the server's source within the GMainContext */
static guint source_id;
/* iterate the default main loop until there are no events to dispatch */
static void
iterate (void)
{
while (g_main_context_iteration (NULL, FALSE)) {
GST_DEBUG ("iteration");
}
}
static void
get_client_ports_full (GstRTSPRange * range, GSocket ** rtp_socket,
GSocket ** rtcp_socket)
{
GSocket *rtp = NULL;
GSocket *rtcp = NULL;
gint rtp_port = 0;
gint rtcp_port;
GInetAddress *anyaddr = g_inet_address_new_any (G_SOCKET_FAMILY_IPV4);
GSocketAddress *sockaddr;
gboolean bound;
for (;;) {
if (rtp_port != 0)
rtp_port += 2;
rtp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
G_SOCKET_PROTOCOL_UDP, NULL);
fail_unless (rtp != NULL);
sockaddr = g_inet_socket_address_new (anyaddr, rtp_port);
fail_unless (sockaddr != NULL);
bound = g_socket_bind (rtp, sockaddr, FALSE, NULL);
g_object_unref (sockaddr);
if (!bound) {
g_object_unref (rtp);
continue;
}
sockaddr = g_socket_get_local_address (rtp, NULL);
fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
rtp_port =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
g_object_unref (sockaddr);
if (rtp_port % 2 != 0) {
rtp_port += 1;
g_object_unref (rtp);
continue;
}
rtcp_port = rtp_port + 1;
rtcp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
G_SOCKET_PROTOCOL_UDP, NULL);
fail_unless (rtcp != NULL);
sockaddr = g_inet_socket_address_new (anyaddr, rtcp_port);
fail_unless (sockaddr != NULL);
bound = g_socket_bind (rtcp, sockaddr, FALSE, NULL);
g_object_unref (sockaddr);
if (!bound) {
g_object_unref (rtp);
g_object_unref (rtcp);
continue;
}
sockaddr = g_socket_get_local_address (rtcp, NULL);
fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
fail_unless (rtcp_port ==
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr)));
g_object_unref (sockaddr);
break;
}
range->min = rtp_port;
range->max = rtcp_port;
if (rtp_socket)
*rtp_socket = rtp;
else
g_object_unref (rtp);
if (rtcp_socket)
*rtcp_socket = rtcp;
else
g_object_unref (rtcp);
GST_DEBUG ("client_port=%d-%d", range->min, range->max);
g_object_unref (anyaddr);
}
/* get a free rtp/rtcp client port pair */
static void
get_client_ports (GstRTSPRange * range)
{
get_client_ports_full (range, NULL, NULL);
}
/* start the tested rtsp server */
static void
start_server (gboolean set_shared_factory)
{
GstRTSPMountPoints *mounts;
gchar *service;
GstRTSPMediaFactory *factory;
GstRTSPAddressPool *pool;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* use an address pool for multicast */
pool = gst_rtsp_address_pool_new ();
gst_rtsp_address_pool_add_range (pool,
"224.3.0.0", "224.3.0.10", 5500, 5510, 16);
gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
GST_RTSP_ADDRESS_POOL_ANY_IPV4, 6000, 6010, 0);
gst_rtsp_media_factory_set_address_pool (factory, pool);
gst_rtsp_media_factory_set_shared (factory, set_shared_factory);
gst_object_unref (pool);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
GST_DEBUG ("rtsp server listening on port %d", test_port);
}
static void
start_tcp_server (gboolean set_shared_factory)
{
GstRTSPMountPoints *mounts;
gchar *service;
GstRTSPMediaFactory *factory;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_protocols (factory, GST_RTSP_LOWER_TRANS_TCP);
gst_rtsp_media_factory_set_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
gst_rtsp_media_factory_set_shared (factory, set_shared_factory);
g_object_unref (mounts);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
GST_DEBUG ("rtsp server listening on port %d", test_port);
}
/* start the testing rtsp server for RECORD mode */
static GstRTSPMediaFactory *
start_record_server (const gchar * launch_line)
{
GstRTSPMediaFactory *factory;
GstRTSPMountPoints *mounts;
gchar *service;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_transport_mode (factory,
GST_RTSP_TRANSPORT_MODE_RECORD);
gst_rtsp_media_factory_set_launch (factory, launch_line);
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
GST_DEBUG ("rtsp server listening on port %d", test_port);
return factory;
}
/* stop the tested rtsp server */
static void
stop_server (void)
{
g_source_remove (source_id);
source_id = 0;
GST_DEBUG ("rtsp server stopped");
}
/* create an rtsp connection to the server on test_port */
static GstRTSPConnection *
connect_to_server (gint port, const gchar * mount_point)
{
GstRTSPConnection *conn = NULL;
gchar *address;
gchar *uri_string;
GstRTSPUrl *url = NULL;
address = gst_rtsp_server_get_address (server);
uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
g_free (address);
fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
g_free (uri_string);
fail_unless (gst_rtsp_connection_create (url, &conn) == GST_RTSP_OK);
gst_rtsp_url_free (url);
fail_unless (gst_rtsp_connection_connect (conn, NULL) == GST_RTSP_OK);
return conn;
}
/* create an rtsp request */
static GstRTSPMessage *
create_request (GstRTSPConnection * conn, GstRTSPMethod method,
const gchar * control)
{
GstRTSPMessage *request = NULL;
gchar *base_uri;
gchar *full_uri;
base_uri = gst_rtsp_url_get_request_uri (gst_rtsp_connection_get_url (conn));
full_uri = g_strdup_printf ("%s/%s", base_uri, control ? control : "");
g_free (base_uri);
if (gst_rtsp_message_new_request (&request, method, full_uri) != GST_RTSP_OK) {
GST_DEBUG ("failed to create request object");
g_free (full_uri);
return NULL;
}
g_free (full_uri);
return request;
}
/* send an rtsp request */
static gboolean
send_request (GstRTSPConnection * conn, GstRTSPMessage * request)
{
if (gst_rtsp_connection_send (conn, request, NULL) != GST_RTSP_OK) {
GST_DEBUG ("failed to send request");
return FALSE;
}
return TRUE;
}
/* read rtsp response. response must be freed by the caller */
static GstRTSPMessage *
read_response (GstRTSPConnection * conn)
{
GstRTSPMessage *response = NULL;
GstRTSPMsgType type;
if (gst_rtsp_message_new (&response) != GST_RTSP_OK) {
GST_DEBUG ("failed to create response object");
return NULL;
}
if (gst_rtsp_connection_receive (conn, response, NULL) != GST_RTSP_OK) {
GST_DEBUG ("failed to read response");
gst_rtsp_message_free (response);
return NULL;
}
type = gst_rtsp_message_get_type (response);
fail_unless (type == GST_RTSP_MESSAGE_RESPONSE
|| type == GST_RTSP_MESSAGE_DATA);
return response;
}
/* send an rtsp request and receive response. gchar** parameters are out
* parameters that have to be freed by the caller */
static GstRTSPStatusCode
do_request_full (GstRTSPConnection * conn, GstRTSPMethod method,
const gchar * control, const gchar * session_in, const gchar * transport_in,
const gchar * range_in, const gchar * require_in,
gchar ** content_type, gchar ** content_base, gchar ** body,
gchar ** session_out, gchar ** transport_out, gchar ** range_out,
gchar ** unsupported_out)
{
GstRTSPMessage *request;
GstRTSPMessage *response;
GstRTSPStatusCode code;
gchar *value;
GstRTSPMsgType msg_type;
/* create request */
request = create_request (conn, method, control);
/* add headers */
if (session_in) {
gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session_in);
}
if (transport_in) {
gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT, transport_in);
}
if (range_in) {
gst_rtsp_message_add_header (request, GST_RTSP_HDR_RANGE, range_in);
}
if (require_in) {
gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE, require_in);
}
/* send request */
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
iterate ();
/* read response */
response = read_response (conn);
fail_unless (response != NULL);
msg_type = gst_rtsp_message_get_type (response);
if (msg_type == GST_RTSP_MESSAGE_DATA) {
do {
gst_rtsp_message_free (response);
response = read_response (conn);
msg_type = gst_rtsp_message_get_type (response);
} while (msg_type == GST_RTSP_MESSAGE_DATA);
}
fail_unless (msg_type == GST_RTSP_MESSAGE_RESPONSE);
/* check status line */
gst_rtsp_message_parse_response (response, &code, NULL, NULL);
if (code != GST_RTSP_STS_OK) {
if (unsupported_out != NULL && code == GST_RTSP_STS_OPTION_NOT_SUPPORTED) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED,
&value, 0);
*unsupported_out = g_strdup (value);
}
gst_rtsp_message_free (response);
return code;
}
/* get information from response */
if (content_type) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_TYPE,
&value, 0);
*content_type = g_strdup (value);
}
if (content_base) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
&value, 0);
*content_base = g_strdup (value);
}
if (body) {
*body = g_malloc (response->body_size + 1);
strncpy (*body, (gchar *) response->body, response->body_size);
}
if (session_out) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &value, 0);
value = g_strdup (value);
/* Remove the timeout */
if (value) {
char *pos = strchr (value, ';');
if (pos)
*pos = 0;
}
if (session_in) {
/* check that we got the same session back */
fail_unless (!g_strcmp0 (value, session_in));
}
*session_out = value;
}
if (transport_out) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &value, 0);
*transport_out = g_strdup (value);
}
if (range_out) {
gst_rtsp_message_get_header (response, GST_RTSP_HDR_RANGE, &value, 0);
*range_out = g_strdup (value);
}
gst_rtsp_message_free (response);
return code;
}
/* send an rtsp request and receive response. gchar** parameters are out
* parameters that have to be freed by the caller */
static GstRTSPStatusCode
do_request (GstRTSPConnection * conn, GstRTSPMethod method,
const gchar * control, const gchar * session_in,
const gchar * transport_in, const gchar * range_in,
gchar ** content_type, gchar ** content_base, gchar ** body,
gchar ** session_out, gchar ** transport_out, gchar ** range_out)
{
return do_request_full (conn, method, control, session_in, transport_in,
range_in, NULL, content_type, content_base, body, session_out,
transport_out, range_out, NULL);
}
/* send an rtsp request with a method and a session, and receive response */
static GstRTSPStatusCode
do_simple_request (GstRTSPConnection * conn, GstRTSPMethod method,
const gchar * session)
{
return do_request (conn, method, NULL, session, NULL, NULL, NULL,
NULL, NULL, NULL, NULL, NULL);
}
/* send an rtsp request with a method,session and range in,
* and receive response. range_in is the Range in req header */
static GstRTSPStatusCode
do_simple_request_rangein (GstRTSPConnection * conn, GstRTSPMethod method,
const gchar * session, const gchar * rangein)
{
return do_request (conn, method, NULL, session, NULL, rangein, NULL,
NULL, NULL, NULL, NULL, NULL);
}
/* send a DESCRIBE request and receive response. returns a received
* GstSDPMessage that must be freed by the caller */
static GstSDPMessage *
do_describe (GstRTSPConnection * conn, const gchar * mount_point)
{
GstSDPMessage *sdp_message;
gchar *content_type = NULL;
gchar *content_base = NULL;
gchar *body = NULL;
gchar *address;
gchar *expected_content_base;
/* send DESCRIBE request */
fail_unless (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL, NULL,
&content_type, &content_base, &body, NULL, NULL, NULL) ==
GST_RTSP_STS_OK);
/* check response values */
fail_unless (!g_strcmp0 (content_type, "application/sdp"));
address = gst_rtsp_server_get_address (server);
expected_content_base =
g_strdup_printf ("rtsp://%s:%d%s/", address, test_port, mount_point);
fail_unless (!g_strcmp0 (content_base, expected_content_base));
/* create sdp message */
fail_unless (gst_sdp_message_new (&sdp_message) == GST_SDP_OK);
fail_unless (gst_sdp_message_parse_buffer ((guint8 *) body,
strlen (body), sdp_message) == GST_SDP_OK);
/* clean up */
g_free (content_type);
g_free (content_base);
g_free (body);
g_free (address);
g_free (expected_content_base);
return sdp_message;
}
/* send a SETUP request and receive response. if *session is not NULL,
* it is used in the request. otherwise, *session is set to a returned
* session string that must be freed by the caller. the returned
* transport must be freed by the caller. */
static GstRTSPStatusCode
do_setup_full (GstRTSPConnection * conn, const gchar * control,
GstRTSPLowerTrans lower_transport, const GstRTSPRange * client_ports,
const gchar * require, gchar ** session, GstRTSPTransport ** transport,
gchar ** unsupported)
{
GstRTSPStatusCode code;
gchar *session_in = NULL;
GString *transport_string_in = NULL;
gchar **session_out = NULL;
gchar *transport_string_out = NULL;
/* prepare and send SETUP request */
if (session) {
if (*session) {
session_in = *session;
} else {
session_out = session;
}
}
transport_string_in = g_string_new (TEST_PROTO);
switch (lower_transport) {
case GST_RTSP_LOWER_TRANS_UDP:
transport_string_in =
g_string_append (transport_string_in, "/UDP;unicast");
break;
case GST_RTSP_LOWER_TRANS_UDP_MCAST:
transport_string_in =
g_string_append (transport_string_in, "/UDP;multicast");
break;
case GST_RTSP_LOWER_TRANS_TCP:
transport_string_in =
g_string_append (transport_string_in, "/TCP;unicast");
break;
default:
g_assert_not_reached ();
break;
}
if (client_ports) {
g_string_append_printf (transport_string_in, ";client_port=%d-%d",
client_ports->min, client_ports->max);
}
code =
do_request_full (conn, GST_RTSP_SETUP, control, session_in,
transport_string_in->str, NULL, require, NULL, NULL, NULL, session_out,
&transport_string_out, NULL, unsupported);
g_string_free (transport_string_in, TRUE);
if (transport_string_out) {
/* create transport */
fail_unless (gst_rtsp_transport_new (transport) == GST_RTSP_OK);
fail_unless (gst_rtsp_transport_parse (transport_string_out,
*transport) == GST_RTSP_OK);
g_free (transport_string_out);
}
GST_INFO ("code=%d", code);
return code;
}
/* send a SETUP request and receive response. if *session is not NULL,
* it is used in the request. otherwise, *session is set to a returned
* session string that must be freed by the caller. the returned
* transport must be freed by the caller. */
static GstRTSPStatusCode
do_setup (GstRTSPConnection * conn, const gchar * control,
const GstRTSPRange * client_ports, gchar ** session,
GstRTSPTransport ** transport)
{
return do_setup_full (conn, control, GST_RTSP_LOWER_TRANS_UDP, client_ports,
NULL, session, transport, NULL);
}
/* fixture setup function */
static void
setup (void)
{
server = gst_rtsp_server_new ();
}
/* fixture clean-up function */
static void
teardown (void)
{
if (server) {
g_object_unref (server);
server = NULL;
}
test_port = 0;
}
GST_START_TEST (test_connect)
{
GstRTSPConnection *conn;
start_server (FALSE);
/* connect to server */
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
/* clean up */
gst_rtsp_connection_free (conn);
stop_server ();
/* iterate so the clean-up can finish */
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_describe)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
gint32 format;
gchar *expected_rtpmap;
const gchar *rtpmap;
const gchar *control_video;
const gchar *control_audio;
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
/* send DESCRIBE request */
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
/* check video sdp */
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
&format);
expected_rtpmap =
g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
g_free (expected_rtpmap);
control_video = gst_sdp_media_get_attribute_val (sdp_media, "control");
fail_unless (!g_strcmp0 (control_video, "stream=0"));
/* check audio sdp */
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
&format);
expected_rtpmap =
g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
g_free (expected_rtpmap);
control_audio = gst_sdp_media_get_attribute_val (sdp_media, "control");
fail_unless (!g_strcmp0 (control_audio, "stream=1"));
/* clean up and iterate so the clean-up can finish */
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_describe_record_media)
{
GstRTSPConnection *conn;
start_record_server ("( fakesink name=depay0 )");
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
/* send DESCRIBE request */
fail_unless_equals_int (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL,
NULL, NULL, NULL, NULL, NULL, NULL, NULL),
GST_RTSP_STS_METHOD_NOT_ALLOWED);
/* clean up and iterate so the clean-up can finish */
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_describe_non_existing_mount_point)
{
GstRTSPConnection *conn;
start_server (FALSE);
/* send DESCRIBE request for a non-existing mount point
* and check that we get a 404 Not Found */
conn = connect_to_server (test_port, "/non-existing");
fail_unless (do_simple_request (conn, GST_RTSP_DESCRIBE, NULL)
== GST_RTSP_STS_NOT_FOUND);
/* clean up and iterate so the clean-up can finish */
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static void
do_test_setup (GstRTSPLowerTrans lower_transport)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_ports = { 0 };
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_ports);
/* send SETUP request for video */
fail_unless (do_setup_full (conn, video_control, lower_transport,
&client_ports, NULL, &session, &video_transport,
NULL) == GST_RTSP_STS_OK);
GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
/* check response from SETUP */
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
fail_unless (video_transport->lower_transport == lower_transport);
fail_unless (video_transport->mode_play);
gst_rtsp_transport_free (video_transport);
/* send SETUP request for audio */
fail_unless (do_setup_full (conn, audio_control, lower_transport,
&client_ports, NULL, &session, &audio_transport,
NULL) == GST_RTSP_STS_OK);
GST_DEBUG ("set up audio %s with session '%s'", audio_control, session);
/* check response from SETUP */
fail_unless (audio_transport->trans == GST_RTSP_TRANS_RTP);
fail_unless (audio_transport->profile == GST_RTSP_PROFILE_AVP);
fail_unless (audio_transport->lower_transport == lower_transport);
fail_unless (audio_transport->mode_play);
gst_rtsp_transport_free (audio_transport);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_free (session);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_START_TEST (test_setup_udp)
{
do_test_setup (GST_RTSP_LOWER_TRANS_UDP);
}
GST_END_TEST;
GST_START_TEST (test_setup_tcp)
{
do_test_setup (GST_RTSP_LOWER_TRANS_TCP);
}
GST_END_TEST;
GST_START_TEST (test_setup_udp_mcast)
{
do_test_setup (GST_RTSP_LOWER_TRANS_UDP_MCAST);
}
GST_END_TEST;
GST_START_TEST (test_setup_twice)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message;
const GstSDPMedia *sdp_media;
const gchar *video_control;
GstRTSPRange client_ports;
GstRTSPTransport *video_transport = NULL;
gchar *session1 = NULL;
gchar *session2 = NULL;
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
/* we wan't more than one session for this connection */
gst_rtsp_connection_set_remember_session_id (conn, FALSE);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get the control url */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_ports);
/* send SETUP request for one session */
fail_unless (do_setup (conn, video_control, &client_ports, &session1,
&video_transport) == GST_RTSP_STS_OK);
GST_DEBUG ("set up video %s, got session '%s'", video_control, session1);
/* check response from SETUP */
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
fail_unless (video_transport->mode_play);
gst_rtsp_transport_free (video_transport);
/* send SETUP request for another session */
fail_unless (do_setup (conn, video_control, &client_ports, &session2,
&video_transport) == GST_RTSP_STS_OK);
GST_DEBUG ("set up video %s, got session '%s'", video_control, session2);
/* check response from SETUP */
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
fail_unless (video_transport->mode_play);
gst_rtsp_transport_free (video_transport);
/* session can not be the same */
fail_unless (strcmp (session1, session2));
/* send TEARDOWN request for the first session */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session1) == GST_RTSP_STS_OK);
/* send TEARDOWN request for the second session */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session2) == GST_RTSP_STS_OK);
g_free (session1);
g_free (session2);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_setup_with_require_header)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
GstRTSPRange client_ports;
gchar *session = NULL;
gchar *unsupported = NULL;
GstRTSPTransport *video_transport = NULL;
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_ports);
/* send SETUP request for video, with single Require header */
fail_unless_equals_int (do_setup_full (conn, video_control,
GST_RTSP_LOWER_TRANS_UDP, &client_ports, "funky-feature", &session,
&video_transport, &unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
fail_unless_equals_string (unsupported, "funky-feature");
g_free (unsupported);
unsupported = NULL;
/* send SETUP request for video, with multiple Require headers */
fail_unless_equals_int (do_setup_full (conn, video_control,
GST_RTSP_LOWER_TRANS_UDP, &client_ports,
"funky-feature, foo-bar, superburst", &session, &video_transport,
&unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
fail_unless_equals_string (unsupported, "funky-feature, foo-bar, superburst");
g_free (unsupported);
unsupported = NULL;
/* ok, just do a normal setup then (make sure that still works) */
fail_unless_equals_int (do_setup (conn, video_control, &client_ports,
&session, &video_transport), GST_RTSP_STS_OK);
GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
/* check response from SETUP */
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
fail_unless (video_transport->mode_play);
gst_rtsp_transport_free (video_transport);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_free (session);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_setup_non_existing_stream)
{
GstRTSPConnection *conn;
GstRTSPRange client_ports;
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
get_client_ports (&client_ports);
/* send SETUP request with a non-existing stream and check that we get a
* 404 Not Found */
fail_unless (do_setup (conn, "stream=7", &client_ports, NULL,
NULL) == GST_RTSP_STS_NOT_FOUND);
/* clean up and iterate so the clean-up can finish */
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static void
receive_rtp (GSocket * socket, GSocketAddress ** addr)
{
GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
for (;;) {
gssize bytes;
GstMapInfo map = GST_MAP_INFO_INIT;
GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
map.maxsize, NULL, NULL);
fail_unless (bytes > 0);
gst_buffer_unmap (buffer, &map);
gst_buffer_set_size (buffer, bytes);
if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer)) {
gst_rtp_buffer_unmap (&rtpbuffer);
break;
}
if (addr)
g_clear_object (addr);
}
gst_buffer_unref (buffer);
}
static void
receive_rtcp (GSocket * socket, GSocketAddress ** addr, GstRTCPType type)
{
GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
for (;;) {
gssize bytes;
GstMapInfo map = GST_MAP_INFO_INIT;
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
map.maxsize, NULL, NULL);
fail_unless (bytes > 0);
gst_buffer_unmap (buffer, &map);
gst_buffer_set_size (buffer, bytes);
if (gst_rtcp_buffer_validate (buffer)) {
GstRTCPBuffer rtcpbuffer = GST_RTCP_BUFFER_INIT;
GstRTCPPacket packet;
if (type) {
fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcpbuffer));
fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuffer, &packet));
do {
if (gst_rtcp_packet_get_type (&packet) == type) {
gst_rtcp_buffer_unmap (&rtcpbuffer);
goto done;
}
} while (gst_rtcp_packet_move_to_next (&packet));
gst_rtcp_buffer_unmap (&rtcpbuffer);
} else {
break;
}
}
if (addr)
g_clear_object (addr);
}
done:
gst_buffer_unref (buffer);
}
static void
do_test_play_tcp_full (const gchar * range)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
gchar *range_out = NULL;
GstRTSPLowerTrans lower_transport = GST_RTSP_LOWER_TRANS_TCP;
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
get_client_ports (&client_port);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
/* do SETUP for video and audio */
fail_unless (do_setup_full (conn, video_control, lower_transport,
&client_port, NULL, &session, &video_transport,
NULL) == GST_RTSP_STS_OK);
fail_unless (do_setup_full (conn, audio_control, lower_transport,
&client_port, NULL, &session, &audio_transport,
NULL) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_request (conn, GST_RTSP_PLAY, NULL, session, NULL, range,
NULL, NULL, NULL, NULL, NULL, &range_out) == GST_RTSP_STS_OK);
if (range)
fail_unless_equals_string (range, range_out);
g_free (range_out);
{
GstRTSPMessage *message;
fail_unless (gst_rtsp_message_new (&message) == GST_RTSP_OK);
fail_unless (gst_rtsp_connection_receive (conn, message,
NULL) == GST_RTSP_OK);
fail_unless (gst_rtsp_message_get_type (message) == GST_RTSP_MESSAGE_DATA);
gst_rtsp_message_free (message);
}
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* FIXME: The rtsp-server always disconnects the transport before
* sending the RTCP BYE
* receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
*/
/* clean up and iterate so the clean-up can finish */
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
}
static void
do_test_play_full (const gchar * range, GstRTSPLowerTrans lower_transport,
GMutex * lock)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
GSocket *rtp_socket, *rtcp_socket;
gchar *range_out = NULL;
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
/* do SETUP for video and audio */
fail_unless (do_setup_full (conn, video_control, lower_transport,
&client_port, NULL, &session, &video_transport,
NULL) == GST_RTSP_STS_OK);
fail_unless (do_setup_full (conn, audio_control, lower_transport,
&client_port, NULL, &session, &audio_transport,
NULL) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_request (conn, GST_RTSP_PLAY, NULL, session, NULL, range,
NULL, NULL, NULL, NULL, NULL, &range_out) == GST_RTSP_STS_OK);
if (range)
fail_unless_equals_string (range, range_out);
g_free (range_out);
for (;;) {
receive_rtp (rtp_socket, NULL);
receive_rtcp (rtcp_socket, NULL, 0);
if (lock != NULL) {
if (g_mutex_trylock (lock) == TRUE) {
g_mutex_unlock (lock);
break;
}
} else {
break;
}
}
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* FIXME: The rtsp-server always disconnects the transport before
* sending the RTCP BYE
* receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
*/
/* clean up and iterate so the clean-up can finish */
g_object_unref (rtp_socket);
g_object_unref (rtcp_socket);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
}
static void
do_test_play (const gchar * range)
{
do_test_play_full (range, GST_RTSP_LOWER_TRANS_UDP, NULL);
}
GST_START_TEST (test_play)
{
start_server (FALSE);
do_test_play (NULL);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_tcp)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_ports = { 0 };
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
start_tcp_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
/* send DESCRIBE request */
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_ports);
/* send SETUP request for the first media */
fail_unless (do_setup_full (conn, video_control, GST_RTSP_LOWER_TRANS_TCP,
&client_ports, NULL, &session, &video_transport,
NULL) == GST_RTSP_STS_OK);
/* check response from SETUP */
fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP);
fail_unless (video_transport->mode_play);
gst_rtsp_transport_free (video_transport);
/* send SETUP request for the second media */
fail_unless (do_setup_full (conn, audio_control, GST_RTSP_LOWER_TRANS_TCP,
&client_ports, NULL, &session, &audio_transport,
NULL) == GST_RTSP_STS_OK);
/* check response from SETUP */
fail_unless (audio_transport->trans == GST_RTSP_TRANS_RTP);
fail_unless (audio_transport->profile == GST_RTSP_PROFILE_AVP);
fail_unless (audio_transport->lower_transport == GST_RTSP_LOWER_TRANS_TCP);
fail_unless (audio_transport->mode_play);
gst_rtsp_transport_free (audio_transport);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_free (session);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_without_session)
{
GstRTSPConnection *conn;
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
/* send PLAY request without a session and check that we get a
* 454 Session Not Found */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
NULL) == GST_RTSP_STS_SESSION_NOT_FOUND);
/* clean up and iterate so the clean-up can finish */
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_bind_already_in_use)
{
GstRTSPServer *serv;
GSocketService *service;
GError *error = NULL;
guint16 port;
gchar *port_str;
serv = gst_rtsp_server_new ();
service = g_socket_service_new ();
/* bind service to port */
port =
g_socket_listener_add_any_inet_port (G_SOCKET_LISTENER (service), NULL,
&error);
g_assert_no_error (error);
port_str = g_strdup_printf ("%d\n", port);
/* try to bind server to the same port */
g_object_set (serv, "service", port_str, NULL);
g_free (port_str);
/* attach to default main context */
fail_unless (gst_rtsp_server_attach (serv, NULL) == 0);
/* cleanup */
g_object_unref (serv);
g_socket_service_stop (service);
g_object_unref (service);
}
GST_END_TEST;
GST_START_TEST (test_play_multithreaded)
{
GstRTSPThreadPool *pool;
pool = gst_rtsp_server_get_thread_pool (server);
gst_rtsp_thread_pool_set_max_threads (pool, 2);
g_object_unref (pool);
start_server (FALSE);
do_test_play (NULL);
stop_server ();
iterate ();
}
GST_END_TEST;
enum
{
BLOCK_ME,
BLOCKED,
UNBLOCK
};
static void
media_constructed_block (GstRTSPMediaFactory * factory,
GstRTSPMedia * media, gpointer user_data)
{
gint *block_state = user_data;
g_mutex_lock (&check_mutex);
*block_state = BLOCKED;
g_cond_broadcast (&check_cond);
while (*block_state != UNBLOCK)
g_cond_wait (&check_cond, &check_mutex);
g_mutex_unlock (&check_mutex);
}
GST_START_TEST (test_play_multithreaded_block_in_describe)
{
GstRTSPConnection *conn;
GstRTSPMountPoints *mounts;
GstRTSPMediaFactory *factory;
gint block_state = BLOCK_ME;
GstRTSPMessage *request;
GstRTSPMessage *response;
GstRTSPStatusCode code;
GstRTSPThreadPool *pool;
pool = gst_rtsp_server_get_thread_pool (server);
gst_rtsp_thread_pool_set_max_threads (pool, 2);
g_object_unref (pool);
mounts = gst_rtsp_server_get_mount_points (server);
fail_unless (mounts != NULL);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
g_signal_connect (factory, "media-constructed",
G_CALLBACK (media_constructed_block), &block_state);
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT "2", factory);
g_object_unref (mounts);
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT "2");
iterate ();
/* do describe, it will not return now as we've blocked it */
request = create_request (conn, GST_RTSP_DESCRIBE, NULL);
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
g_mutex_lock (&check_mutex);
while (block_state != BLOCKED)
g_cond_wait (&check_cond, &check_mutex);
g_mutex_unlock (&check_mutex);
/* Do a second connection while the first one is blocked */
do_test_play (NULL);
/* Now unblock the describe */
g_mutex_lock (&check_mutex);
block_state = UNBLOCK;
g_cond_broadcast (&check_cond);
g_mutex_unlock (&check_mutex);
response = read_response (conn);
gst_rtsp_message_parse_response (response, &code, NULL, NULL);
fail_unless (code == GST_RTSP_STS_OK);
gst_rtsp_message_free (response);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static void
new_session_timeout_one (GstRTSPClient * client,
GstRTSPSession * session, gpointer user_data)
{
gst_rtsp_session_set_timeout (session, 1);
g_signal_handlers_disconnect_by_func (client, new_session_timeout_one,
user_data);
}
static void
session_connected_new_session_cb (GstRTSPServer * server,
GstRTSPClient * client, gpointer user_data)
{
g_signal_connect (client, "new-session", user_data, NULL);
}
GST_START_TEST (test_play_multithreaded_timeout_client)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
GstRTSPSessionPool *pool;
GstRTSPThreadPool *thread_pool;
thread_pool = gst_rtsp_server_get_thread_pool (server);
gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
g_object_unref (thread_pool);
pool = gst_rtsp_server_get_session_pool (server);
g_signal_connect (server, "client-connected",
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_port);
/* do SETUP for video and audio */
fail_unless (do_setup_full (conn, video_control, GST_RTSP_LOWER_TRANS_UDP,
&client_port, NULL, &session, &video_transport,
NULL) == GST_RTSP_STS_OK);
fail_unless (do_setup_full (conn, audio_control, GST_RTSP_LOWER_TRANS_UDP,
&client_port, NULL, &session, &audio_transport,
NULL) == GST_RTSP_STS_OK);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
sleep (7);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 0);
/* clean up and iterate so the clean-up can finish */
g_object_unref (pool);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_multithreaded_timeout_session)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session1 = NULL;
gchar *session2 = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
GstRTSPSessionPool *pool;
GstRTSPThreadPool *thread_pool;
thread_pool = gst_rtsp_server_get_thread_pool (server);
gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
g_object_unref (thread_pool);
pool = gst_rtsp_server_get_session_pool (server);
g_signal_connect (server, "client-connected",
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
gst_rtsp_connection_set_remember_session_id (conn, FALSE);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_port);
/* do SETUP for video and audio */
fail_unless (do_setup (conn, video_control, &client_port, &session1,
&video_transport) == GST_RTSP_STS_OK);
fail_unless (do_setup (conn, audio_control, &client_port, &session2,
&audio_transport) == GST_RTSP_STS_OK);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 2);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session1) == GST_RTSP_STS_OK);
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session2) == GST_RTSP_STS_OK);
sleep (7);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
/* send TEARDOWN request and check that we get 454 Session Not found */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session1) == GST_RTSP_STS_SESSION_NOT_FOUND);
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session2) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_object_unref (pool);
g_free (session1);
g_free (session2);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static void
new_connection_and_session_timeout_one (GstRTSPClient * client,
GstRTSPSession * session, gpointer user_data)
{
gint ps_timeout = 0;
g_object_set (G_OBJECT (client), "post-session-timeout", 1, NULL);
g_object_get (G_OBJECT (client), "post-session-timeout", &ps_timeout, NULL);
fail_unless_equals_int (ps_timeout, 1);
g_object_set (G_OBJECT (session), "extra-timeout", 0, NULL);
gst_rtsp_session_set_timeout (session, 1);
g_signal_handlers_disconnect_by_func (client,
new_connection_and_session_timeout_one, user_data);
}
GST_START_TEST (test_play_timeout_connection)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPSessionPool *pool;
GstRTSPThreadPool *thread_pool;
GstRTSPMessage *request;
GstRTSPMessage *response;
thread_pool = gst_rtsp_server_get_thread_pool (server);
g_object_unref (thread_pool);
pool = gst_rtsp_server_get_session_pool (server);
g_signal_connect (server, "client-connected",
G_CALLBACK (session_connected_new_session_cb),
new_connection_and_session_timeout_one);
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
gst_rtsp_connection_set_remember_session_id (conn, FALSE);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_port);
/* do SETUP for video and audio */
fail_unless (do_setup (conn, video_control, &client_port, &session,
&video_transport) == GST_RTSP_STS_OK);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
sleep (2);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
sleep (3);
request = create_request (conn, GST_RTSP_TEARDOWN, NULL);
/* add headers */
if (session) {
gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
}
/* send request */
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
iterate ();
/* read response */
response = read_response (conn);
fail_unless (response == NULL);
if (response) {
gst_rtsp_message_free (response);
}
/* clean up and iterate so the clean-up can finish */
g_object_unref (pool);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_no_session_timeout)
{
GstRTSPSession *session;
gint64 now;
gboolean is_expired;
session = gst_rtsp_session_new ("test-session");
gst_rtsp_session_set_timeout (session, 0);
now = g_get_monotonic_time ();
/* add more than the extra 5 seconds that are usually added in
* gst_rtsp_session_next_timeout_usec */
now += 7000000;
is_expired = gst_rtsp_session_is_expired_usec (session, now);
fail_unless (is_expired == FALSE);
g_object_unref (session);
}
GST_END_TEST;
/* media contains two streams: video and audio but only one
* stream is requested */
GST_START_TEST (test_play_one_active_stream)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPSessionPool *pool;
GstRTSPThreadPool *thread_pool;
thread_pool = gst_rtsp_server_get_thread_pool (server);
gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
g_object_unref (thread_pool);
pool = gst_rtsp_server_get_session_pool (server);
g_signal_connect (server, "client-connected",
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
gst_rtsp_connection_set_remember_session_id (conn, FALSE);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_port);
/* do SETUP for video only */
fail_unless (do_setup (conn, video_control, &client_port, &session,
&video_transport) == GST_RTSP_STS_OK);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
/* send TEARDOWN request */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_object_unref (pool);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_disconnect)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
GstRTSPSessionPool *pool;
pool = gst_rtsp_server_get_session_pool (server);
g_signal_connect (server, "client-connected",
G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
start_server (FALSE);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports (&client_port);
/* do SETUP for video and audio */
fail_unless (do_setup (conn, video_control, &client_port, &session,
&video_transport) == GST_RTSP_STS_OK);
fail_unless (do_setup (conn, audio_control, &client_port, &session,
&audio_transport) == GST_RTSP_STS_OK);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
gst_rtsp_connection_free (conn);
sleep (7);
fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
/* clean up and iterate so the clean-up can finish */
g_object_unref (pool);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
stop_server ();
iterate ();
}
GST_END_TEST;
/* Only different with test_play is the specific ports selected */
GST_START_TEST (test_play_specific_server_port)
{
GstRTSPMountPoints *mounts;
gchar *service;
GstRTSPMediaFactory *factory;
GstRTSPAddressPool *pool;
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GSocket *rtp_socket, *rtcp_socket;
GSocketAddress *rtp_address, *rtcp_address;
guint16 rtp_port, rtcp_port;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
/* we have to suspend media after SDP in order to make sure that
* we can reconfigure UDP sink with new UDP ports */
gst_rtsp_media_factory_set_suspend_mode (factory,
GST_RTSP_SUSPEND_MODE_RESET);
pool = gst_rtsp_address_pool_new ();
gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
GST_RTSP_ADDRESS_POOL_ANY_IPV4, 7770, 7780, 0);
gst_rtsp_media_factory_set_address_pool (factory, pool);
g_object_unref (pool);
gst_rtsp_media_factory_set_launch (factory, "( " VIDEO_PIPELINE " )");
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
GST_DEBUG ("rtsp server listening on port %d", test_port);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 1);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
/* do SETUP for video */
fail_unless (do_setup (conn, video_control, &client_port, &session,
&video_transport) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
receive_rtp (rtp_socket, &rtp_address);
receive_rtcp (rtcp_socket, &rtcp_address, 0);
fail_unless (G_IS_INET_SOCKET_ADDRESS (rtp_address));
fail_unless (G_IS_INET_SOCKET_ADDRESS (rtcp_address));
rtp_port =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_address));
rtcp_port =
g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtcp_address));
fail_unless (rtp_port >= 7770 && rtp_port <= 7780 && rtp_port % 2 == 0);
fail_unless (rtcp_port >= 7770 && rtcp_port <= 7780 && rtcp_port % 2 == 1);
fail_unless (rtp_port + 1 == rtcp_port);
g_object_unref (rtp_address);
g_object_unref (rtcp_address);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* FIXME: The rtsp-server always disconnects the transport before
* sending the RTCP BYE
* receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
*/
/* clean up and iterate so the clean-up can finish */
g_object_unref (rtp_socket);
g_object_unref (rtcp_socket);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_smpte_range)
{
start_server (FALSE);
do_test_play ("npt=5-");
do_test_play ("smpte=0:00:00-");
do_test_play ("smpte=1:00:00-");
do_test_play ("smpte=1:00:03-");
do_test_play ("clock=20120321T152256Z-");
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_smpte_range_tcp)
{
start_tcp_server (FALSE);
do_test_play_tcp_full ("npt=5-");
do_test_play_tcp_full ("smpte=0:00:00-");
do_test_play_tcp_full ("smpte=1:00:00-");
do_test_play_tcp_full ("smpte=1:00:03-");
do_test_play_tcp_full ("clock=20120321T152256Z-");
stop_server ();
iterate ();
}
GST_END_TEST;
static gpointer
thread_func_udp (gpointer data)
{
do_test_play_full (NULL, GST_RTSP_LOWER_TRANS_UDP, (GMutex *) data);
return NULL;
}
static gpointer
thread_func_tcp (gpointer data)
{
do_test_play_tcp_full (NULL);
return NULL;
}
static void
test_shared (gpointer (thread_func) (gpointer data))
{
GMutex lock1, lock2, lock3, lock4;
GThread *thread1, *thread2, *thread3, *thread4;
/* Locks for each thread. Each thread will keep reading data as long as the
* thread is locked. */
g_mutex_init (&lock1);
g_mutex_init (&lock2);
g_mutex_init (&lock3);
g_mutex_init (&lock4);
if (thread_func == thread_func_tcp)
start_tcp_server (TRUE);
else
start_server (TRUE);
/* Start the first receiver thread. */
g_mutex_lock (&lock1);
thread1 = g_thread_new ("thread1", thread_func, &lock1);
/* Connect and disconnect another client. */
g_mutex_lock (&lock2);
thread2 = g_thread_new ("thread2", thread_func, &lock2);
g_mutex_unlock (&lock2);
g_mutex_clear (&lock2);
g_thread_join (thread2);
/* Do it again. */
g_mutex_lock (&lock3);
thread3 = g_thread_new ("thread3", thread_func, &lock3);
g_mutex_unlock (&lock3);
g_mutex_clear (&lock3);
g_thread_join (thread3);
/* Disconnect the last client. This will clean up the media. */
g_mutex_unlock (&lock1);
g_mutex_clear (&lock1);
g_thread_join (thread1);
/* Connect and disconnect another client. This will create and clean up the
* media. */
g_mutex_lock (&lock4);
thread4 = g_thread_new ("thread4", thread_func, &lock4);
g_mutex_unlock (&lock4);
g_mutex_clear (&lock4);
g_thread_join (thread4);
stop_server ();
iterate ();
}
/* Test adding and removing clients to a 'Shared' media.
* CASE: unicast UDP */
GST_START_TEST (test_shared_udp)
{
test_shared (thread_func_udp);
}
GST_END_TEST;
/* Test adding and removing clients to a 'Shared' media.
* CASE: unicast TCP */
GST_START_TEST (test_shared_tcp)
{
test_shared (thread_func_tcp);
}
GST_END_TEST;
GST_START_TEST (test_announce_without_sdp)
{
GstRTSPConnection *conn;
GstRTSPStatusCode status;
GstRTSPMessage *request;
GstRTSPMessage *response;
start_record_server ("( fakesink name=depay0 )");
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
/* create and send ANNOUNCE request */
request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
fail_unless (send_request (conn, request));
iterate ();
response = read_response (conn);
/* check response */
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
gst_rtsp_message_free (response);
/* try again, this type with content-type, but still no SDP */
gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
fail_unless (send_request (conn, request));
iterate ();
response = read_response (conn);
/* check response */
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
gst_rtsp_message_free (response);
/* try again, this type with an unknown content-type */
gst_rtsp_message_remove_header (request, GST_RTSP_HDR_CONTENT_TYPE, -1);
gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
"application/x-something");
fail_unless (send_request (conn, request));
iterate ();
response = read_response (conn);
/* check response */
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
fail_unless_equals_int (status, GST_RTSP_STS_BAD_REQUEST);
gst_rtsp_message_free (response);
/* clean up and iterate so the clean-up can finish */
gst_rtsp_message_free (request);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static GstRTSPStatusCode
do_announce (GstRTSPConnection * conn, GstSDPMessage * sdp)
{
GstRTSPMessage *request;
GstRTSPMessage *response;
GstRTSPStatusCode code;
gchar *str;
/* create request */
request = create_request (conn, GST_RTSP_ANNOUNCE, NULL);
gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
"application/sdp");
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (request, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
/* send request */
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
iterate ();
/* read response */
response = read_response (conn);
/* check status line */
gst_rtsp_message_parse_response (response, &code, NULL, NULL);
gst_rtsp_message_free (response);
return code;
}
static void
media_constructed_cb (GstRTSPMediaFactory * mfactory, GstRTSPMedia * media,
gpointer user_data)
{
GstElement **p_sink = user_data;
GstElement *bin;
bin = gst_rtsp_media_get_element (media);
*p_sink = gst_bin_get_by_name (GST_BIN (bin), "sink");
GST_INFO ("media constructed!: %" GST_PTR_FORMAT, *p_sink);
gst_object_unref (bin);
}
#define RECORD_N_BUFS 10
GST_START_TEST (test_record_tcp)
{
GstRTSPMediaFactory *mfactory;
GstRTSPConnection *conn;
GstRTSPStatusCode status;
GstRTSPMessage *response;
GstRTSPMessage *request;
GstSDPMessage *sdp;
GstRTSPResult rres;
GSocketAddress *sa;
GInetAddress *ia;
GstElement *server_sink = NULL;
GSocket *conn_socket;
const gchar *proto;
gchar *client_ip, *sess_id, *session = NULL;
gint i;
mfactory =
start_record_server
("( rtppcmadepay name=depay0 ! appsink name=sink async=false )");
g_signal_connect (mfactory, "media-constructed",
G_CALLBACK (media_constructed_cb), &server_sink);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
conn_socket = gst_rtsp_connection_get_read_socket (conn);
sa = g_socket_get_local_address (conn_socket, NULL);
ia = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (sa));
client_ip = g_inet_address_to_string (ia);
if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV6)
proto = "IP6";
else if (g_socket_address_get_family (sa) == G_SOCKET_FAMILY_IPV4)
proto = "IP4";
else
g_assert_not_reached ();
g_object_unref (sa);
gst_sdp_message_new (&sdp);
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
/* session ID doesn't have to be super-unique in this case */
sess_id = g_strdup_printf ("%u", g_random_int ());
gst_sdp_message_set_origin (sdp, "-", sess_id, "1", "IN", proto, client_ip);
g_free (sess_id);
g_free (client_ip);
gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
gst_sdp_message_set_information (sdp, "rtsp-server-test");
gst_sdp_message_add_time (sdp, "0", "0", NULL);
gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
/* add stream 0 */
{
GstSDPMedia *smedia;
gst_sdp_media_new (&smedia);
gst_sdp_media_set_media (smedia, "audio");
gst_sdp_media_add_format (smedia, "8"); /* pcma/alaw */
gst_sdp_media_set_port_info (smedia, 0, 1);
gst_sdp_media_set_proto (smedia, "RTP/AVP");
gst_sdp_media_add_attribute (smedia, "rtpmap", "8 PCMA/8000");
gst_sdp_message_add_media (sdp, smedia);
gst_sdp_media_free (smedia);
}
/* send ANNOUNCE request */
status = do_announce (conn, sdp);
fail_unless_equals_int (status, GST_RTSP_STS_OK);
/* create and send SETUP request */
request = create_request (conn, GST_RTSP_SETUP, NULL);
gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT,
"RTP/AVP/TCP;interleaved=0;mode=record");
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
iterate ();
response = read_response (conn);
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
fail_unless_equals_int (status, GST_RTSP_STS_OK);
rres =
gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &session, 0);
session = g_strdup (session);
fail_unless_equals_int (rres, GST_RTSP_OK);
gst_rtsp_message_free (response);
/* send RECORD */
request = create_request (conn, GST_RTSP_RECORD, NULL);
gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
iterate ();
response = read_response (conn);
gst_rtsp_message_parse_response (response, &status, NULL, NULL);
fail_unless_equals_int (status, GST_RTSP_STS_OK);
gst_rtsp_message_free (response);
/* send some data */
{
GstElement *pipeline, *src, *enc, *pay, *sink;
pipeline = gst_pipeline_new ("send-pipeline");
src = gst_element_factory_make ("audiotestsrc", NULL);
g_object_set (src, "num-buffers", RECORD_N_BUFS,
"samplesperbuffer", 1000, NULL);
enc = gst_element_factory_make ("alawenc", NULL);
pay = gst_element_factory_make ("rtppcmapay", NULL);
sink = gst_element_factory_make ("appsink", NULL);
fail_unless (pipeline && src && enc && pay && sink);
gst_bin_add_many (GST_BIN (pipeline), src, enc, pay, sink, NULL);
gst_element_link_many (src, enc, pay, sink, NULL);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
do {
GstRTSPMessage *data_msg;
GstMapInfo map = GST_MAP_INFO_INIT;
GstRTSPResult rres;
GstSample *sample = NULL;
GstBuffer *buf;
g_signal_emit_by_name (G_OBJECT (sink), "pull-sample", &sample);
if (sample == NULL)
break;
buf = gst_sample_get_buffer (sample);
rres = gst_rtsp_message_new_data (&data_msg, 0);
fail_unless_equals_int (rres, GST_RTSP_OK);
gst_buffer_map (buf, &map, GST_MAP_READ);
GST_INFO ("sending %u bytes of data on channel 0", (guint) map.size);
GST_MEMDUMP ("data on channel 0", map.data, map.size);
rres = gst_rtsp_message_set_body (data_msg, map.data, map.size);
fail_unless_equals_int (rres, GST_RTSP_OK);
gst_buffer_unmap (buf, &map);
rres = gst_rtsp_connection_send (conn, data_msg, NULL);
fail_unless_equals_int (rres, GST_RTSP_OK);
gst_rtsp_message_free (data_msg);
gst_sample_unref (sample);
} while (TRUE);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
}
/* check received data (we assume every buffer created by audiotestsrc and
* subsequently encoded by mulawenc results in exactly one RTP packet) */
for (i = 0; i < RECORD_N_BUFS; ++i) {
GstSample *sample = NULL;
g_signal_emit_by_name (G_OBJECT (server_sink), "pull-sample", &sample);
GST_INFO ("%2d recv sample: %p", i, sample);
gst_sample_unref (sample);
}
fail_unless_equals_int (GST_STATE (server_sink), GST_STATE_PLAYING);
/* clean up and iterate so the clean-up can finish */
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
g_free (session);
/* release the reference to server_sink, obtained in media_constructed_cb */
gst_object_unref (server_sink);
}
GST_END_TEST;
static void
do_test_multiple_transports (GstRTSPLowerTrans trans1, GstRTSPLowerTrans trans2)
{
GstRTSPConnection *conn1;
GstRTSPConnection *conn2;
GstSDPMessage *sdp_message1 = NULL;
GstSDPMessage *sdp_message2 = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port1, client_port2;
gchar *session1 = NULL;
gchar *session2 = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
GSocket *rtp_socket, *rtcp_socket;
conn1 = connect_to_server (test_port, TEST_MOUNT_POINT);
conn2 = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message1 = do_describe (conn1, TEST_MOUNT_POINT);
get_client_ports_full (&client_port1, &rtp_socket, &rtcp_socket);
/* get control strings from DESCRIBE response */
sdp_media = gst_sdp_message_get_media (sdp_message1, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message1, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
/* do SETUP for video and audio */
fail_unless (do_setup_full (conn1, video_control, trans1,
&client_port1, NULL, &session1, &video_transport,
NULL) == GST_RTSP_STS_OK);
fail_unless (do_setup_full (conn1, audio_control, trans1,
&client_port1, NULL, &session1, &audio_transport,
NULL) == GST_RTSP_STS_OK);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
sdp_message2 = do_describe (conn2, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
sdp_media = gst_sdp_message_get_media (sdp_message2, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message2, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports_full (&client_port2, NULL, NULL);
/* do SETUP for video and audio */
fail_unless (do_setup_full (conn2, video_control, trans2,
&client_port2, NULL, &session2, &video_transport,
NULL) == GST_RTSP_STS_OK);
fail_unless (do_setup_full (conn2, audio_control, trans2,
&client_port2, NULL, &session2, &audio_transport,
NULL) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_request (conn1, GST_RTSP_PLAY, NULL, session1, NULL, NULL,
NULL, NULL, NULL, NULL, NULL, NULL) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_request (conn2, GST_RTSP_PLAY, NULL, session2, NULL, NULL,
NULL, NULL, NULL, NULL, NULL, NULL) == GST_RTSP_STS_OK);
/* receive UDP data */
receive_rtp (rtp_socket, NULL);
receive_rtcp (rtcp_socket, NULL, 0);
/* receive TCP data */
{
GstRTSPMessage *message;
fail_unless (gst_rtsp_message_new (&message) == GST_RTSP_OK);
fail_unless (gst_rtsp_connection_receive (conn2, message,
NULL) == GST_RTSP_OK);
fail_unless (gst_rtsp_message_get_type (message) == GST_RTSP_MESSAGE_DATA);
gst_rtsp_message_free (message);
}
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn1, GST_RTSP_TEARDOWN,
session1) == GST_RTSP_STS_OK);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn2, GST_RTSP_TEARDOWN,
session2) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_object_unref (rtp_socket);
g_object_unref (rtcp_socket);
g_free (session1);
g_free (session2);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message1);
gst_sdp_message_free (sdp_message2);
gst_rtsp_connection_free (conn1);
gst_rtsp_connection_free (conn2);
}
GST_START_TEST (test_multiple_transports)
{
start_server (TRUE);
do_test_multiple_transports (GST_RTSP_LOWER_TRANS_UDP,
GST_RTSP_LOWER_TRANS_TCP);
stop_server ();
}
GST_END_TEST;
GST_START_TEST (test_suspend_mode_reset_only_audio)
{
GstRTSPMountPoints *mounts;
gchar *service;
GstRTSPMediaFactory *factory;
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *audio_transport = NULL;
GSocket *rtp_socket, *rtcp_socket;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_suspend_mode (factory,
GST_RTSP_SUSPEND_MODE_RESET);
gst_rtsp_media_factory_set_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
/* do SETUP for audio */
fail_unless (do_setup (conn, audio_control, &client_port, &session,
&audio_transport) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_free (session);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
}
GST_END_TEST;
static GstRTSPStatusCode
adjust_play_mode (GstRTSPClient * client, GstRTSPContext * ctx,
GstRTSPTimeRange ** range, GstSeekFlags * flags, gdouble * rate,
GstClockTime * trickmode_interval, gboolean * enable_rate_control)
{
GstRTSPState rtspstate;
rtspstate = gst_rtsp_session_media_get_rtsp_state (ctx->sessmedia);
if (rtspstate == GST_RTSP_STATE_PLAYING) {
if (!gst_rtsp_session_media_set_state (ctx->sessmedia, GST_STATE_PAUSED))
return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
if (!gst_rtsp_media_unsuspend (ctx->media))
return GST_RTSP_STS_INTERNAL_SERVER_ERROR;
}
return GST_RTSP_STS_OK;
}
GST_START_TEST (test_double_play)
{
GstRTSPMountPoints *mounts;
gchar *service;
GstRTSPMediaFactory *factory;
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_port;
gchar *session = NULL;
GstRTSPTransport *audio_transport = NULL;
GstRTSPTransport *video_transport = NULL;
GSocket *rtp_socket, *rtcp_socket;
GstRTSPClient *client;
GstRTSPClientClass *klass;
client = gst_rtsp_client_new ();
klass = GST_RTSP_CLIENT_GET_CLASS (client);
klass->adjust_play_mode = adjust_play_mode;
mounts = gst_rtsp_server_get_mount_points (server);
factory = gst_rtsp_media_factory_new ();
gst_rtsp_media_factory_set_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* set port to any */
gst_rtsp_server_set_service (server, "0");
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
/* get port */
service = gst_rtsp_server_get_service (server);
test_port = atoi (service);
fail_unless (test_port != 0);
g_free (service);
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
sdp_message = do_describe (conn, TEST_MOUNT_POINT);
/* get control strings from DESCRIBE response */
fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
sdp_media = gst_sdp_message_get_media (sdp_message, 0);
video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
sdp_media = gst_sdp_message_get_media (sdp_message, 1);
audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
/* do SETUP for video */
fail_unless (do_setup (conn, video_control, &client_port, &session,
&video_transport) == GST_RTSP_STS_OK);
/* do SETUP for audio */
fail_unless (do_setup (conn, audio_control, &client_port, &session,
&audio_transport) == GST_RTSP_STS_OK);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
session, "npt=0-") == GST_RTSP_STS_OK);
/* let it play for a while, so it needs to seek
* for next play (npt=0-) */
g_usleep (30000);
/* send PLAY request and check that we get 200 OK */
fail_unless (do_simple_request_rangein (conn, GST_RTSP_PLAY,
session, "npt=0-") == GST_RTSP_STS_OK);
/* send TEARDOWN request and check that we get 200 OK */
fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
session) == GST_RTSP_STS_OK);
/* clean up and iterate so the clean-up can finish */
g_object_unref (rtp_socket);
g_object_unref (rtcp_socket);
g_free (session);
gst_rtsp_transport_free (video_transport);
gst_rtsp_transport_free (audio_transport);
gst_sdp_message_free (sdp_message);
gst_rtsp_connection_free (conn);
stop_server ();
iterate ();
g_object_unref (client);
}
GST_END_TEST;
static Suite *
rtspserver_suite (void)
{
Suite *s = suite_create ("rtspserver");
TCase *tc = tcase_create ("general");
suite_add_tcase (s, tc);
tcase_add_checked_fixture (tc, setup, teardown);
tcase_set_timeout (tc, 120);
tcase_add_test (tc, test_connect);
tcase_add_test (tc, test_describe);
tcase_add_test (tc, test_describe_non_existing_mount_point);
tcase_add_test (tc, test_describe_record_media);
tcase_add_test (tc, test_setup_udp);
tcase_add_test (tc, test_setup_tcp);
tcase_add_test (tc, test_setup_udp_mcast);
tcase_add_test (tc, test_setup_twice);
tcase_add_test (tc, test_setup_with_require_header);
tcase_add_test (tc, test_setup_non_existing_stream);
tcase_add_test (tc, test_play);
tcase_add_test (tc, test_play_tcp);
tcase_add_test (tc, test_play_without_session);
tcase_add_test (tc, test_bind_already_in_use);
tcase_add_test (tc, test_play_multithreaded);
tcase_add_test (tc, test_play_multithreaded_block_in_describe);
tcase_add_test (tc, test_play_multithreaded_timeout_client);
tcase_add_test (tc, test_play_multithreaded_timeout_session);
tcase_add_test (tc, test_play_timeout_connection);
tcase_add_test (tc, test_no_session_timeout);
tcase_add_test (tc, test_play_one_active_stream);
tcase_add_test (tc, test_play_disconnect);
tcase_add_test (tc, test_play_specific_server_port);
tcase_add_test (tc, test_play_smpte_range);
tcase_add_test (tc, test_play_smpte_range_tcp);
tcase_add_test (tc, test_shared_udp);
tcase_add_test (tc, test_shared_tcp);
tcase_add_test (tc, test_announce_without_sdp);
tcase_add_test (tc, test_record_tcp);
tcase_add_test (tc, test_multiple_transports);
tcase_add_test (tc, test_suspend_mode_reset_only_audio);
tcase_add_test (tc, test_double_play);
return s;
}
GST_CHECK_MAIN (rtspserver);
Reference in a new issue View git blame Copy permalink