#include <rtsp-client.h>
+#define VIDEO_PIPELINE "videotestsrc ! " \
+ "video/x-raw,width=352,height=288 ! " \
+ "rtpgstpay name=pay0 pt=96"
+#define AUDIO_PIPELINE "audiotestsrc ! " \
+ "audio/x-raw,rate=8000 ! " \
+ "rtpgstpay name=pay1 pt=97"
+
static gchar *session_id;
static gint cseq;
static guint expected_session_timeout = 60;
return TRUE;
}
+static void
+create_connection (GstRTSPConnection ** conn)
+{
+ GSocket *sock;
+ GError *error = NULL;
+
+ sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
+ G_SOCKET_PROTOCOL_TCP, &error);
+ g_assert_no_error (error);
+ fail_unless (gst_rtsp_connection_create_from_socket (sock, "127.0.0.1", 444,
+ NULL, conn) == GST_RTSP_OK);
+ g_object_unref (sock);
+}
+
static GstRTSPClient *
setup_client (const gchar * launch_line)
{
factory = gst_rtsp_media_factory_new ();
if (launch_line == NULL)
gst_rtsp_media_factory_set_launch (factory,
- "videotestsrc ! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96");
+ "( " VIDEO_PIPELINE " " AUDIO_PIPELINE " )");
else
gst_rtsp_media_factory_set_launch (factory, launch_line);
GstRTSPMessage request = { 0, };
gchar *str;
GstRTSPConnection *conn;
- GSocket *sock;
- GError *error = NULL;
client = gst_rtsp_client_new ();
/* OPTIONS with an absolute path instead of an absolute url */
/* set host information */
- sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
- G_SOCKET_PROTOCOL_TCP, &error);
- g_assert_no_error (error);
- gst_rtsp_connection_create_from_socket (sock, "localhost", 444, NULL, &conn);
+ create_connection (&conn);
fail_unless (gst_rtsp_client_set_connection (client, conn));
- g_object_unref (sock);
-
fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
"/test") == GST_RTSP_OK);
str = g_strdup_printf ("%d", cseq);
static const gchar *expected_transport = NULL;
static gboolean
-test_setup_response_200_multicast (GstRTSPClient * client,
- GstRTSPMessage * response, gboolean close, gpointer user_data)
+test_setup_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+ gboolean close, gpointer user_data)
{
GstRTSPStatusCode code;
const gchar *reason;
GstRTSPVersion version;
gchar *str;
+ gchar *pattern;
GstRTSPSessionPool *session_pool;
GstRTSPSession *session;
gchar **session_hdr_params;
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT,
&str, 0) == GST_RTSP_OK);
- fail_unless_equals_string (str, expected_transport);
+ pattern = g_strdup_printf ("^%s$", expected_transport);
+ fail_unless (g_regex_match_simple (pattern, str, 0, 0),
+ "Transport '%s' doesn't match pattern '%s'", str, pattern);
+ g_free (pattern);
fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
&str, 0) == GST_RTSP_OK);
g_strfreev (session_hdr_params);
/* remember session id to be able to send teardown */
+ if (session_id)
+ g_free (session_id);
session_id = g_strdup (gst_rtsp_session_get_sessionid (session));
fail_unless (session_id != NULL);
session_id = NULL;
}
+GST_START_TEST (test_setup_tcp)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast");
+
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client);
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_tcp_two_streams_same_channels)
+{
+ GstRTSPClient *client;
+ GstRTSPConnection *conn;
+ GstRTSPMessage request = { 0, };
+ gchar *str;
+
+ client = setup_client (NULL);
+ create_connection (&conn);
+ fail_unless (gst_rtsp_client_set_connection (client, conn));
+
+ /* test SETUP of a video stream with 0-1 as interleaved channels */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast;interleaved=0-1");
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ /* test SETUP of an audio stream with *the same* interleaved channels.
+ * we expect the server to allocate new channel numbers */
+ fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+ "rtsp://localhost/test/stream=1") == GST_RTSP_OK);
+ str = g_strdup_printf ("%d", cseq);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+ g_free (str);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+ "RTP/AVP/TCP;unicast;interleaved=0-1");
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, session_id);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
+ expected_transport =
+ "RTP/AVP/TCP;unicast;interleaved=2-3;ssrc=.*;mode=\"PLAY\"";
+ fail_unless (gst_rtsp_client_handle_message (client,
+ &request) == GST_RTSP_OK);
+ gst_rtsp_message_unset (&request);
+
+ send_teardown (client);
+ teardown_client (client);
+}
+
+GST_END_TEST;
+
static GstRTSPClient *
setup_multicast_client (guint max_ttl)
{
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
"ttl=1;port=5000-5001;mode=\"PLAY\"";
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
expected_transport);
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
- gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client, test_setup_response_200, NULL, NULL);
session_pool = gst_rtsp_client_get_session_pool (client);
fail_unless (session_pool != NULL);
fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport1);
- gst_rtsp_client_set_send_func (client1, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client1, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client1,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
g_free (client_addr);
gst_rtsp_context_pop_current (&ctx);
- session_id1 = session_id;
+ session_id1 = g_strdup (session_id);
/* second multicast client with transport specific request */
cseq = 0;
gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT, transport2);
- gst_rtsp_client_set_send_func (client2, test_setup_response_200_multicast,
- NULL, NULL);
+ gst_rtsp_client_set_send_func (client2, test_setup_response_200, NULL, NULL);
fail_unless (gst_rtsp_client_handle_message (client2,
&request) == GST_RTSP_OK);
gst_rtsp_message_unset (&request);
tcase_add_test (tc, test_request);
tcase_add_test (tc, test_options);
tcase_add_test (tc, test_describe);
+ tcase_add_test (tc, test_setup_tcp);
+ tcase_add_test (tc, test_setup_tcp_two_streams_same_channels);
tcase_add_test (tc, test_client_multicast_transport_404);
tcase_add_test (tc, test_client_multicast_transport);
tcase_add_test (tc, test_client_multicast_ignore_transport_specific);
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