rtsp-stream: Fixed TCP transport case

Make sure that the appsink element is actually added to
the bin before trying to link it with the elements in it.

https://bugzilla.gnome.org/show_bug.cgi?id=776343
This commit is contained in:
Patricia Muscalu 2016-12-21 13:41:50 +01:00 committed by Sebastian Dröge
parent 143063a68f
commit 42f270e7f2
2 changed files with 123 additions and 2 deletions

View file

@ -2462,6 +2462,7 @@ create_sender_part (GstRTSPStream * stream, GstBin * bin, GstState state)
}
} else if (is_tcp) {
/* only appsink needed, link it to the session */
gst_bin_add (bin, priv->appsink[i]);
pad = gst_element_get_static_pad (priv->appsink[i], "sink");
gst_pad_link (priv->send_src[i], pad);
gst_object_unref (pad);

View file

@ -191,6 +191,40 @@ start_server (gboolean set_shared_factory)
GST_DEBUG ("rtsp server listening on port %d", test_port);
}
static void
start_tcp_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_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);
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)
@ -296,6 +330,7 @@ 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");
@ -306,8 +341,8 @@ read_response (GstRTSPConnection * conn)
gst_rtsp_message_free (response);
return NULL;
}
fail_unless (gst_rtsp_message_get_type (response) ==
GST_RTSP_MESSAGE_RESPONSE);
type = gst_rtsp_message_get_type (response);
fail_unless (type == GST_RTSP_MESSAGE_RESPONSE || type == GST_RTSP_MESSAGE_DATA);
return response;
}
@ -325,6 +360,7 @@ do_request_full (GstRTSPConnection * conn, GstRTSPMethod method,
GstRTSPMessage *response;
GstRTSPStatusCode code;
gchar *value;
GstRTSPMsgType msg_type;
/* create request */
request = create_request (conn, method, control);
@ -351,6 +387,19 @@ do_request_full (GstRTSPConnection * conn, GstRTSPMethod method,
/* 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);
@ -1105,6 +1154,76 @@ GST_START_TEST (test_play)
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 ();
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;
@ -1992,6 +2111,7 @@ rtspserver_suite (void)
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);