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gstreamer/tests/check/gst/rtspserver.c
Wim Taymans cd8a81b3b1 tests: fix test 
Actually do what the comment says. Also keep the old code around, not sure what
should happen when you get a 454 from a TEARDOWN, does it close the connection?
it currently doesn't.
2013-06-20 17:30:49 +02:00

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/* 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>
*
* 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 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"
#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");
}
}
/* returns an unused port that can be used by the test */
static int
get_unused_port (gint type)
{
int sock;
struct sockaddr_in addr;
socklen_t addr_len;
gint port;
/* create socket */
fail_unless ((sock = socket (AF_INET, type, 0)) > 0);
/* pass port 0 to bind, which will bind to any free port */
memset (&addr, 0, sizeof addr);
addr.sin_family = AF_INET;
addr.sin_addr.s_addr = INADDR_ANY;
addr.sin_port = htons (0);
fail_unless (bind (sock, (struct sockaddr *) &addr, sizeof addr) == 0);
/* ask what port was bound using getsockname */
addr_len = sizeof addr;
memset (&addr, 0, addr_len);
fail_unless (getsockname (sock, (struct sockaddr *) &addr, &addr_len) == 0);
port = ntohs (addr.sin_port);
/* close the socket so the port gets unbound again (and can be used by the
* test) */
close (sock);
return port;
}
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 (void)
{
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_launch (factory,
"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
g_object_unref (mounts);
/* set port */
test_port = get_unused_port (SOCK_STREAM);
service = g_strdup_printf ("%d", test_port);
gst_rtsp_server_set_service (server, service);
g_free (service);
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
GST_DEBUG ("rtsp server listening on port %d", test_port);
}
/* stop the tested rtsp server */
static void
stop_server ()
{
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);
gst_rtsp_url_parse (uri_string, &url);
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;
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;
}
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
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 (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)
{
GstRTSPMessage *request;
GstRTSPMessage *response;
GstRTSPStatusCode code;
gchar *value;
/* 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);
}
/* 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);
if (code != GST_RTSP_STS_OK) {
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 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 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;
gchar *content_base;
gchar *body;
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 (GstRTSPConnection * conn, const gchar * control,
const GstRTSPRange * client_ports, gchar ** session,
GstRTSPTransport ** transport)
{
GstRTSPStatusCode code;
gchar *session_in = NULL;
gchar *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_strdup_printf (TEST_PROTO ";unicast;client_port=%d-%d",
client_ports->min, client_ports->max);
code =
do_request (conn, GST_RTSP_SETUP, control, session_in,
transport_string_in, NULL, NULL, NULL, NULL, session_out,
&transport_string_out, NULL);
g_free (transport_string_in);
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);
}
return code;
}
/* 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 ();
/* 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 ();
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_non_existing_mount_point)
{
GstRTSPConnection *conn;
start_server ();
/* 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;
GST_START_TEST (test_setup)
{
GstRTSPConnection *conn;
GstSDPMessage *sdp_message = NULL;
const GstSDPMedia *sdp_media;
const gchar *video_control;
const gchar *audio_control;
GstRTSPRange client_ports;
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
start_server ();
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 (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 SETUP request for audio */
fail_unless (do_setup (conn, audio_control, &client_ports, &session,
&audio_transport) == 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 == GST_RTSP_LOWER_TRANS_UDP);
fail_unless (audio_transport->mode_play);
gst_rtsp_transport_free (audio_transport);
/* 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 ();
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 (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;
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 (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);
/* 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);
receive_rtp (rtp_socket, NULL);
receive_rtcp (rtcp_socket, NULL, 0);
/* 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);
}
GST_START_TEST (test_play)
{
start_server ();
do_test_play (NULL);
stop_server ();
iterate ();
}
GST_END_TEST;
GST_START_TEST (test_play_without_session)
{
GstRTSPConnection *conn;
start_server ();
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)
{
gst_rtsp_server_set_max_threads (server, 2);
start_server ();
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;
gst_rtsp_server_set_max_threads (server, 2);
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 ();
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;
GstRTSPMessage *request;
GstRTSPMessage *response;
GstRTSPStatusCode code;
gst_rtsp_server_set_max_threads (server, 2);
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 ();
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);
sleep (7);
fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
/* send TEARDOWN request and check that we get 454 Session Not found */
request = create_request (conn, GST_RTSP_TEARDOWN, NULL);
gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session);
fail_unless (send_request (conn, request));
gst_rtsp_message_free (request);
fail_unless (gst_rtsp_message_new (&response) == GST_RTSP_OK);
fail_unless (gst_rtsp_connection_receive (conn, response, NULL) ==
GST_RTSP_OK);
gst_rtsp_message_parse_response (response, &code, NULL, NULL);
fail_unless (code == GST_RTSP_STS_SESSION_NOT_FOUND);
gst_rtsp_message_free (response);
#if 0
fail_unless (gst_rtsp_message_new (&response) == GST_RTSP_OK);
fail_unless (gst_rtsp_connection_receive (conn, response, NULL) ==
GST_RTSP_ESYS);
fail_unless (errno == ECONNRESET);
gst_rtsp_message_free (response);
#endif
/* 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;
gst_rtsp_server_set_max_threads (server, 2);
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 ();
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;
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 ();
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 ();
pool = gst_rtsp_address_pool_new ();
gst_rtsp_address_pool_add_range_unicast (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
GST_RTSP_ADDRESS_POOL_ANY_IPV4, 7770, 7780);
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 */
test_port = get_unused_port (SOCK_STREAM);
service = g_strdup_printf ("%d", test_port);
gst_rtsp_server_set_service (server, service);
g_free (service);
/* attach to default main context */
source_id = gst_rtsp_server_attach (server, NULL);
fail_if (source_id == 0);
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 ();
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;
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, 20);
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_setup);
tcase_add_test (tc, test_setup_non_existing_stream);
tcase_add_test (tc, test_play);
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_disconnect);
tcase_add_test (tc, test_play_specific_server_port);
tcase_add_test (tc, test_play_smpte_range);
return s;
}
GST_CHECK_MAIN (rtspserver);
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