mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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929b8afd16
Original commit message from CVS: * check/pipelines/simple_launch_lines.c: (run_pipeline): Fix for bus API. * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_add_to_queue), (gst_base_rtp_depayload_push), (gst_base_rtp_depayload_set_gst_timestamp), (gst_base_rtp_depayload_queue_release): Some cleanups. * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init), (gst_basertppayload_init), (gst_basertppayload_setcaps), (gst_basertppayload_set_options), (gst_basertppayload_set_outcaps), (gst_basertppayload_push), (gst_basertppayload_get_property), (gst_basertppayload_change_state): Added debugging category.
176 lines
5.5 KiB
C
176 lines
5.5 KiB
C
/* GStreamer
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* Copyright (C) 2005 Andy Wingo <wingo@pobox.com>
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*
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* simple_launch_lines.c: Unit test for simple pipelines
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <gst/check/gstcheck.h>
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static GstElement *
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setup_pipeline (const gchar * pipe_descr)
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{
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GstElement *pipeline;
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pipeline = gst_parse_launch (pipe_descr, NULL);
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g_return_val_if_fail (GST_IS_PIPELINE (pipeline), NULL);
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return pipeline;
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}
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/*
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* run_pipeline:
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* @pipe: the pipeline to run
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* @desc: the description for use in messages
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* @events: is a mask of expected events
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* @tevent: is the expected terminal event.
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*
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* the poll call will time out after half a second.
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*/
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static void
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run_pipeline (GstElement * pipe, const gchar * descr,
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GstMessageType events, GstMessageType tevent)
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{
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GstBus *bus;
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GstMessage *message;
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GstMessageType revent;
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g_assert (pipe);
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bus = gst_element_get_bus (pipe);
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g_assert (bus);
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if (gst_element_set_state (pipe,
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GST_STATE_PLAYING) != GST_STATE_CHANGE_SUCCESS) {
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g_critical ("Couldn't set pipeline to PLAYING");
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goto done;
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}
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while (1) {
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message = gst_bus_poll (bus, GST_MESSAGE_ANY, GST_SECOND / 2);
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/* always have to pop the message before getting back into poll */
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if (message) {
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revent = GST_MESSAGE_TYPE (message);
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gst_message_unref (gst_bus_pop (bus));
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} else {
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revent = GST_MESSAGE_UNKNOWN;
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}
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if (revent == tevent) {
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break;
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} else if (revent == GST_MESSAGE_UNKNOWN) {
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g_critical ("Unexpected timeout in gst_bus_poll, looking for %d: %s",
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tevent, descr);
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break;
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} else if (revent & events) {
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continue;
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}
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g_critical ("Unexpected message received of type %d, looking for %d: %s",
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revent, tevent, descr);
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}
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done:
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gst_element_set_state (pipe, GST_STATE_NULL);
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gst_object_unref (pipe);
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}
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GST_START_TEST (test_element_negotiation)
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{
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gchar *s;
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/* see http://bugzilla.gnome.org/show_bug.cgi?id=315126 */
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s = "fakesrc ! audio/x-raw-int,width=16,depth=16,rate=22050,channels=1 ! audioconvert ! audio/x-raw-int,width=16,depth=16,rate=22050,channels=1 ! fakesink";
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run_pipeline (setup_pipeline (s), s,
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GST_MESSAGE_ANY & ~(GST_MESSAGE_ERROR | GST_MESSAGE_WARNING),
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GST_MESSAGE_UNKNOWN);
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#ifdef HAVE_LIBVISUAL
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s = "sinesrc ! tee name=t ! alsasink t. ! audioconvert ! libvisual_lv_scope ! ffmpegcolorspace ! xvimagesink";
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run_pipeline (setup_pipeline (s), s,
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GST_MESSAGE_ANY & ~(GST_MESSAGE_ERROR | GST_MESSAGE_WARNING),
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GST_MESSAGE_UNKNOWN);
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#endif
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}
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GST_END_TEST
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GST_START_TEST (test_basetransform_based)
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{
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/* Each of these tests is to check whether various basetransform based elements can
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* select output caps when not allowed to do passthrough and going to a generic sink
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* such as fakesink or filesink */
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const gchar *s;
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/* Check that videoscale can pick a height given only a width */
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s = "videotestsrc ! video/x-raw-yuv,format=(fourcc)I420,width=320,height=240 ! " "videoscale ! video/x-raw-yuv,width=640 ! fakesink";
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run_pipeline (setup_pipeline (s), s,
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GST_MESSAGE_ANY & ~(GST_MESSAGE_ERROR | GST_MESSAGE_WARNING),
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GST_MESSAGE_UNKNOWN);
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/* Test that ffmpegcolorspace can pick an output format that isn't passthrough without
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* completely specified output caps */
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s = "videotestsrc ! video/x-raw-yuv,format=(fourcc)I420,width=320,height=240 ! " "ffmpegcolorspace ! video/x-raw-rgb ! fakesink";
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run_pipeline (setup_pipeline (s), s,
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GST_MESSAGE_ANY & ~(GST_MESSAGE_ERROR | GST_MESSAGE_WARNING),
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GST_MESSAGE_UNKNOWN);
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/* Check that audioresample can pick a samplerate to use from a
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* range that doesn't include the input */
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s = "sinesrc ! audio/x-raw-int,width=16,depth=16,rate=8000 ! audioresample ! "
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"audio/x-raw-int,rate=[16000,48000] ! fakesink";
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run_pipeline (setup_pipeline (s), s,
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GST_MESSAGE_ANY & ~(GST_MESSAGE_ERROR | GST_MESSAGE_WARNING),
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GST_MESSAGE_UNKNOWN);
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/* Check that audioconvert can pick a depth to use, given a width */
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s = "sinesrc ! audio/x-raw-int,width=16,depth=16 ! audioconvert ! "
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"audio/x-raw-int,width=32 ! fakesink";
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run_pipeline (setup_pipeline (s), s,
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GST_MESSAGE_ANY & ~(GST_MESSAGE_ERROR | GST_MESSAGE_WARNING),
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GST_MESSAGE_UNKNOWN);
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}
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GST_END_TEST Suite *
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simple_launch_lines_suite (void)
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{
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Suite *s = suite_create ("Pipelines");
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TCase *tc_chain = tcase_create ("linear");
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/* time out after 20s, not the default 3 */
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tcase_set_timeout (tc_chain, 20);
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suite_add_tcase (s, tc_chain);
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// tcase_add_test (tc_chain, test_element_negotiation);
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tcase_add_test (tc_chain, test_basetransform_based);
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return s;
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}
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int
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main (int argc, char **argv)
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{
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int nf;
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Suite *s = simple_launch_lines_suite ();
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SRunner *sr = srunner_create (s);
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gst_check_init (&argc, &argv);
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srunner_run_all (sr, CK_NORMAL);
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nf = srunner_ntests_failed (sr);
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srunner_free (sr);
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return nf;
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}
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