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/* GStreamer
* Copyright (C) 2011 Tim-Philipp Müller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/gst.h>
#include <gst/check/gstcheck.h>
#include <gst/app/gstappsink.h>
#include <gst/app/gstappsrc.h>
#include <gst/audio/audio-enumtypes.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiocdsrc.h>
#include <gst/audio/gstaudioclock.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/audio/gstaudiosink.h>
#include <gst/audio/gstaudiosrc.h>
#include <gst/audio/gstaudiobasesink.h>
#include <gst/audio/gstaudiobasesrc.h>
#include <gst/audio/gstaudioringbuffer.h>
#include <gst/audio/streamvolume.h>
#include <gst/fft/gstfftf32.h>
#include <gst/fft/gstfftf64.h>
#include <gst/fft/gstfft.h>
#include <gst/fft/gstffts16.h>
#include <gst/fft/gstffts32.h>
#include <gst/pbutils/codec-utils.h>
#include <gst/pbutils/descriptions.h>
#include <gst/pbutils/encoding-profile.h>
#include <gst/pbutils/encoding-target.h>
#include <gst/pbutils/gstdiscoverer.h>
#include <gst/pbutils/gstpluginsbaseversion.h>
#include <gst/pbutils/install-plugins.h>
#include <gst/pbutils/missing-plugins.h>
#include <gst/pbutils/pbutils-enumtypes.h>
#include <gst/pbutils/pbutils.h>
#include <gst/riff/riff-ids.h>
#include <gst/riff/riff-media.h>
#include <gst/riff/riff-read.h>
#include <gst/rtp/gstrtpbaseaudiopayload.h>
#include <gst/rtp/gstrtpbasedepayload.h>
#include <gst/rtp/gstrtpbasepayload.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtppayloads.h>
#include <gst/rtsp/gstrtspconnection.h>
#include <gst/rtsp/gstrtspdefs.h>
#include <gst/rtsp/gstrtsp-enumtypes.h>
#include <gst/rtsp/gstrtspextension.h>
#include <gst/rtsp/gstrtsp.h>
#include <gst/rtsp/gstrtspmessage.h>
#include <gst/rtsp/gstrtsprange.h>
#include <gst/rtsp/gstrtsptransport.h>
#include <gst/rtsp/gstrtspurl.h>
#include <gst/sdp/gstsdp.h>
#include <gst/sdp/gstsdpmessage.h>
#include <gst/tag/gsttagdemux.h>
#include <gst/tag/tag.h>
#include <gst/video/gstvideofilter.h>
#include <gst/video/gstvideosink.h>
#include <gst/video/video-enumtypes.h>
#include <gst/video/video.h>
#include <gst/video/colorbalancechannel.h>
#include <gst/video/colorbalance.h>
#include <gst/video/videodirection.h>
#include <gst/video/videoorientation.h>
#include <gst/video/videooverlay.h>
#include <gst/video/navigation.h>
/* we mostly just want to make sure that our library headers don't
* contain anything a C++ compiler might not like */
GST_START_TEST (test_nothing)
{
gst_init (NULL, NULL);
}
GST_END_TEST;
GST_START_TEST (test_init_macros)
{
GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
fail_unless_equals_int (rtp.size[0], 0)
fail_unless_equals_int (rtcp.map.size, 0);
}
GST_END_TEST;
static Suite *
libscpp_suite (void)
{
Suite *s = suite_create ("GstLibsCpp");
TCase *tc_chain = tcase_create ("C++ libs header tests");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_nothing);
tcase_add_test (tc_chain, test_init_macros);
return s;
}
GST_CHECK_MAIN (libscpp);