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authorOlivier CrĂȘte <olivier.crete@collabora.com>2013-02-20 23:30:01 (GMT)
committerWim Taymans <wim.taymans@collabora.co.uk>2013-03-11 10:07:20 (GMT)
commitd3c70d4d51cc1b62374c754609d913d024939244 (patch)
treec3daa4c307568e2fa35e57659e960f660b890f0b
parent5a39e259497bf71ff3364a833be4544dd32a0f4e (diff)
downloadgst-rtsp-server-d3c70d4d51cc1b62374c754609d913d024939244.tar.gz
gst-rtsp-server-d3c70d4d51cc1b62374c754609d913d024939244.tar.xz
tests: Make sure packets are actually received
-rw-r--r--tests/check/gst/rtspserver.c194
1 files changed, 162 insertions, 32 deletions
diff --git a/tests/check/gst/rtspserver.c b/tests/check/gst/rtspserver.c
index 9cd07ad..001450f 100644
--- a/tests/check/gst/rtspserver.c
+++ b/tests/check/gst/rtspserver.c
@@ -23,6 +23,8 @@
#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>
@@ -91,50 +93,96 @@ get_unused_port (gint type)
return port;
}
-/* returns TRUE if the given port is not currently bound */
-static gboolean
-port_is_unused (gint port, gint type)
+static void
+get_client_ports_full (GstRTSPRange * range, GSocket ** rtp_socket,
+ GSocket ** rtcp_socket)
{
- int sock;
- struct sockaddr_in addr;
- gboolean is_bound;
+ 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;
+ }
- /* create socket */
- fail_unless ((sock = socket (AF_INET, type, 0)) > 0);
+ 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);
- /* check if the port is already bound by trying to bind to it (again) */
- memset (&addr, 0, sizeof addr);
- addr.sin_family = AF_INET;
- addr.sin_addr.s_addr = INADDR_ANY;
- addr.sin_port = htons (port);
- is_bound = (bind (sock, (struct sockaddr *) &addr, sizeof addr) != 0);
+ if (rtp_port % 2 != 0) {
+ rtp_port += 1;
+ g_object_unref (rtp);
+ continue;
+ }
- /* close the socket, which will unbind if bound by our call to bind */
- close (sock);
+ 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;
+ }
- return !is_bound;
+ 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)
{
- gint rtp_port;
- gint rtcp_port;
-
- /* get a pair of unused ports, where the rtp port is even */
- do {
- rtp_port = get_unused_port (SOCK_DGRAM);
- rtcp_port = rtp_port + 1;
- } while (rtp_port % 2 != 0 || !port_is_unused (rtcp_port, SOCK_DGRAM));
- range->min = rtp_port;
- range->max = rtcp_port;
- GST_DEBUG ("client_port=%d-%d", range->min, range->max);
+ get_client_ports_full (range, NULL, NULL);
}
/* start the tested rtsp server */
static void
-start_server ()
+start_server (void)
{
GstRTSPMountPoints *mounts;
gchar *service;
@@ -146,7 +194,6 @@ start_server ()
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);
@@ -624,6 +671,79 @@ GST_START_TEST (test_setup_non_existing_stream)
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 (void)
{
GstRTSPConnection *conn;
@@ -635,6 +755,7 @@ do_test_play (void)
gchar *session = NULL;
GstRTSPTransport *video_transport = NULL;
GstRTSPTransport *audio_transport = NULL;
+ GSocket *rtp_socket, *rtcp_socket;
conn = connect_to_server (test_port, TEST_MOUNT_POINT);
@@ -647,7 +768,7 @@ do_test_play (void)
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);
+ 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,
@@ -659,11 +780,21 @@ do_test_play (void)
fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
session) == GST_RTSP_STS_OK);
+ 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);
@@ -1016,7 +1147,6 @@ GST_START_TEST (test_play_multithreaded_timeout_session)
GST_END_TEST;
-
GST_START_TEST (test_play_disconnect)
{
GstRTSPConnection *conn;