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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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tests: Make sure packets are actually received
This commit is contained in:
Olivier Crête
Wim Taymans
parent
5a39e25949
commit
d3c70d4d51
1 changed files with 162 additions and 32 deletions
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@ -23,6 +23,8 @@
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#include <gst/check/gstcheck.h>
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#include <gst/sdp/gstsdpmessage.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <stdio.h>
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#include <netinet/in.h>
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@ -91,50 +93,96 @@ get_unused_port (gint type)
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return port;
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}
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/* returns TRUE if the given port is not currently bound */
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static gboolean
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port_is_unused (gint port, gint type)
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static void
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get_client_ports_full (GstRTSPRange * range, GSocket ** rtp_socket,
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GSocket ** rtcp_socket)
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{
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int sock;
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struct sockaddr_in addr;
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gboolean is_bound;
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GSocket *rtp = NULL;
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GSocket *rtcp = NULL;
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gint rtp_port = 0;
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gint rtcp_port;
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GInetAddress *anyaddr = g_inet_address_new_any (G_SOCKET_FAMILY_IPV4);
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GSocketAddress *sockaddr;
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gboolean bound;
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/* create socket */
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fail_unless ((sock = socket (AF_INET, type, 0)) > 0);
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for (;;) {
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if (rtp_port != 0)
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rtp_port += 2;
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/* check if the port is already bound by trying to bind to it (again) */
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memset (&addr, 0, sizeof addr);
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addr.sin_family = AF_INET;
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addr.sin_addr.s_addr = INADDR_ANY;
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addr.sin_port = htons (port);
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is_bound = (bind (sock, (struct sockaddr *) &addr, sizeof addr) != 0);
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rtp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
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G_SOCKET_PROTOCOL_UDP, NULL);
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fail_unless (rtp != NULL);
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/* close the socket, which will unbind if bound by our call to bind */
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close (sock);
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sockaddr = g_inet_socket_address_new (anyaddr, rtp_port);
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fail_unless (sockaddr != NULL);
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bound = g_socket_bind (rtp, sockaddr, FALSE, NULL);
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g_object_unref (sockaddr);
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if (!bound) {
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g_object_unref (rtp);
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continue;
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}
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return !is_bound;
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sockaddr = g_socket_get_local_address (rtp, NULL);
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fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
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rtp_port =
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g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
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g_object_unref (sockaddr);
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if (rtp_port % 2 != 0) {
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rtp_port += 1;
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g_object_unref (rtp);
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continue;
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}
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rtcp_port = rtp_port + 1;
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rtcp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
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G_SOCKET_PROTOCOL_UDP, NULL);
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fail_unless (rtcp != NULL);
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sockaddr = g_inet_socket_address_new (anyaddr, rtcp_port);
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fail_unless (sockaddr != NULL);
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bound = g_socket_bind (rtcp, sockaddr, FALSE, NULL);
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g_object_unref (sockaddr);
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if (!bound) {
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g_object_unref (rtp);
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g_object_unref (rtcp);
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continue;
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}
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sockaddr = g_socket_get_local_address (rtcp, NULL);
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fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
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fail_unless (rtcp_port ==
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g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr)));
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g_object_unref (sockaddr);
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break;
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}
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range->min = rtp_port;
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range->max = rtcp_port;
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if (rtp_socket)
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*rtp_socket = rtp;
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else
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g_object_unref (rtp);
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if (rtcp_socket)
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*rtcp_socket = rtcp;
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else
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g_object_unref (rtcp);
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GST_DEBUG ("client_port=%d-%d", range->min, range->max);
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g_object_unref (anyaddr);
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}
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/* get a free rtp/rtcp client port pair */
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static void
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get_client_ports (GstRTSPRange * range)
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{
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gint rtp_port;
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gint rtcp_port;
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/* get a pair of unused ports, where the rtp port is even */
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do {
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rtp_port = get_unused_port (SOCK_DGRAM);
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rtcp_port = rtp_port + 1;
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} while (rtp_port % 2 != 0 || !port_is_unused (rtcp_port, SOCK_DGRAM));
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range->min = rtp_port;
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range->max = rtcp_port;
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GST_DEBUG ("client_port=%d-%d", range->min, range->max);
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get_client_ports_full (range, NULL, NULL);
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}
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/* start the tested rtsp server */
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static void
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start_server ()
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start_server (void)
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{
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GstRTSPMountPoints *mounts;
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gchar *service;
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@ -146,7 +194,6 @@ start_server ()
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gst_rtsp_media_factory_set_launch (factory,
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"( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
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gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
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g_object_unref (mounts);
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@ -623,6 +670,79 @@ GST_START_TEST (test_setup_non_existing_stream)
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GST_END_TEST;
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static void
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receive_rtp (GSocket * socket, GSocketAddress ** addr)
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{
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GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
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for (;;) {
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gssize bytes;
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GstMapInfo map = GST_MAP_INFO_INIT;
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GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
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map.maxsize, NULL, NULL);
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fail_unless (bytes > 0);
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gst_buffer_unmap (buffer, &map);
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gst_buffer_set_size (buffer, bytes);
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if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer)) {
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gst_rtp_buffer_unmap (&rtpbuffer);
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break;
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}
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if (addr)
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g_clear_object (addr);
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}
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gst_buffer_unref (buffer);
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}
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static void
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receive_rtcp (GSocket * socket, GSocketAddress ** addr, GstRTCPType type)
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{
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GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
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for (;;) {
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gssize bytes;
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GstMapInfo map = GST_MAP_INFO_INIT;
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gst_buffer_map (buffer, &map, GST_MAP_WRITE);
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bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
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map.maxsize, NULL, NULL);
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fail_unless (bytes > 0);
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gst_buffer_unmap (buffer, &map);
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gst_buffer_set_size (buffer, bytes);
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if (gst_rtcp_buffer_validate (buffer)) {
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GstRTCPBuffer rtcpbuffer = GST_RTCP_BUFFER_INIT;
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GstRTCPPacket packet;
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if (type) {
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fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcpbuffer));
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fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuffer, &packet));
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do {
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if (gst_rtcp_packet_get_type (&packet) == type) {
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gst_rtcp_buffer_unmap (&rtcpbuffer);
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goto done;
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}
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} while (gst_rtcp_packet_move_to_next (&packet));
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gst_rtcp_buffer_unmap (&rtcpbuffer);
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} else {
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break;
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}
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}
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if (addr)
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g_clear_object (addr);
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}
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done:
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gst_buffer_unref (buffer);
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}
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static void
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do_test_play (void)
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{
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@ -635,6 +755,7 @@ do_test_play (void)
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gchar *session = NULL;
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GstRTSPTransport *video_transport = NULL;
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GstRTSPTransport *audio_transport = NULL;
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GSocket *rtp_socket, *rtcp_socket;
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conn = connect_to_server (test_port, TEST_MOUNT_POINT);
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@ -647,7 +768,7 @@ do_test_play (void)
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sdp_media = gst_sdp_message_get_media (sdp_message, 1);
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audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
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get_client_ports (&client_port);
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get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
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/* do SETUP for video and audio */
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fail_unless (do_setup (conn, video_control, &client_port, &session,
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@ -659,11 +780,21 @@ do_test_play (void)
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fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
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session) == GST_RTSP_STS_OK);
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receive_rtp (rtp_socket, NULL);
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receive_rtcp (rtcp_socket, NULL, 0);
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/* send TEARDOWN request and check that we get 200 OK */
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fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
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session) == GST_RTSP_STS_OK);
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/* FIXME: The rtsp-server always disconnects the transport before
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* sending the RTCP BYE
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* receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
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*/
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/* clean up and iterate so the clean-up can finish */
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g_object_unref (rtp_socket);
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g_object_unref (rtcp_socket);
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g_free (session);
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gst_rtsp_transport_free (video_transport);
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gst_rtsp_transport_free (audio_transport);
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@ -1016,7 +1147,6 @@ GST_START_TEST (test_play_multithreaded_timeout_session)
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GST_END_TEST;
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GST_START_TEST (test_play_disconnect)
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{
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GstRTSPConnection *conn;
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