mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-21 07:46:38 +00:00
b1950b6409
This reverts commit adc9694ed7
.
No need to restrict the conversion, we can handle interlace correctly. We
basically unpack each field, then convert each field to the target colorspace
and pack and interleave each field to the target format. We also disable any
fast path that can't deal with interlaced formats.
240 lines
8.1 KiB
C
240 lines
8.1 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio-format.h>
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#ifndef GST_DISABLE_PARSE
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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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GST_LOG ("pipeline: %s", pipe_descr);
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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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GstStateChangeReturn ret;
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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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fail_if (gst_element_set_state (pipe, GST_STATE_PLAYING) ==
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GST_STATE_CHANGE_FAILURE, "Could not set pipeline %s to playing", descr);
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ret = gst_element_get_state (pipe, NULL, NULL, 10 * GST_SECOND);
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if (ret == GST_STATE_CHANGE_ASYNC) {
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g_critical ("Pipeline '%s' failed to go to PLAYING fast enough", descr);
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goto done;
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} else if (ret != GST_STATE_CHANGE_SUCCESS) {
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g_critical ("Pipeline '%s' failed to go into PLAYING state", descr);
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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 (message);
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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
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("Unexpected message received of type %d, '%s', looking for %d: %s",
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revent, gst_message_type_get_name (revent), tevent, descr);
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}
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done:
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fail_if (gst_element_set_state (pipe, GST_STATE_NULL) ==
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GST_STATE_CHANGE_FAILURE, "Could not set pipeline %s to NULL", descr);
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gst_element_get_state (pipe, NULL, NULL, GST_CLOCK_TIME_NONE);
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gst_object_unref (pipe);
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gst_bus_set_flushing (bus, TRUE);
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gst_object_unref (bus);
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}
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GST_START_TEST (test_element_negotiation)
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{
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const gchar *s;
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/* Ensures that filtering buffers with unknown caps down to fixed-caps
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* will apply those caps to the buffers.
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* see http://bugzilla.gnome.org/show_bug.cgi?id=315126 */
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s = "fakesrc num-buffers=2 ! "
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"audio/x-raw,format=" GST_AUDIO_NE (S16) ",rate=22050,channels=1 "
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"! audioconvert "
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"! audio/x-raw,format=" GST_AUDIO_NE (S16) ",rate=22050,channels=1 "
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"! 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 = "audiotestsrc num-buffers=30 ! tee name=t ! alsasink t. ! audioconvert ! "
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"libvisual_lv_scope ! videoconvert ! 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
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* elements can select output caps when not allowed to do passthrough
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* and going to a generic sink 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 num-buffers=2 ! "
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"video/x-raw,format=(string)I420,width=320,height=240 ! "
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"videoscale ! video/x-raw,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 videoconvert can pick an output format that isn't
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* passthrough without completely specified output caps */
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s = "videotestsrc num-buffers=2 ! "
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"video/x-raw,format=(string)I420,width=320,height=240 ! "
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"videoconvert ! video/x-raw,format=(string)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 = "audiotestsrc num-buffers=2 ! "
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"audio/x-raw,format=" GST_AUDIO_NE (S16) ",rate=8000 ! "
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"audioresample ! audio/x-raw,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 = "audiotestsrc num-buffers=30 ! audio/x-raw,format=" GST_AUDIO_NE (S16)
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" ! audioconvert ! " "audio/x-raw,format=" GST_AUDIO_NE (S32)
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" ! 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 videoscale doesn't claim to be able to transform input in
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* formats it can't handle for a given scaling method; videoconvert
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* should then make sure a format that can be handled is chosen (4-tap
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* scaling is not implemented for RGB and packed yuv currently) */
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s = "videotestsrc num-buffers=2 ! video/x-raw,format=(string)ARGB64 ! "
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"videoconvert ! videoscale method=4-tap ! videoconvert ! "
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"video/x-raw,format=(string)RGB, width=32,height=32,framerate=(fraction)30/1,"
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"pixel-aspect-ratio=(fraction)1/1 ! 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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s = "videotestsrc num-buffers=2 ! video/x-raw,format=(string)AYUV,"
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"width=64,height=64 ! videoconvert ! videoscale method=4-tap ! "
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"videoconvert ! video/x-raw,format=(string)AYUV,width=32,"
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"height=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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/* make sure nothing funny happens in passthrough mode (we don't check that
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* passthrough mode is chosen though) */
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s = "videotestsrc num-buffers=2 ! video/x-raw,format=(string)I420,"
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"width=64,height=64 ! videoconvert ! videoscale method=4-tap ! "
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"videoconvert ! video/x-raw,format=(string)I420,width=32,"
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"height=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;
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#endif /* #ifndef GST_DISABLE_PARSE */
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static 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 60s, not the default 3 */
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tcase_set_timeout (tc_chain, 60);
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suite_add_tcase (s, tc_chain);
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#ifndef GST_DISABLE_PARSE
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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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#endif
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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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