mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d524439b35
Makes source code smaller, and ensures we go through common initialization path (like the one that sets up XML unit test output ...)
155 lines
3.9 KiB
C
155 lines
3.9 KiB
C
/*
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* GStreamer
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*
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* unit test for aacparse
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*
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* Copyright (C) 2008 Nokia Corporation. All rights reserved.
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*
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* Contact: Stefan Kost <stefan.kost@nokia.com>
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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 "parser.h"
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#define SRC_CAPS_TMPL "audio/mpeg, parsed=(boolean)false, mpegversion=(int)1"
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#define SINK_CAPS_TMPL "audio/mpeg, parsed=(boolean)true, mpegversion=(int)1"
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GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SINK_CAPS_TMPL)
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);
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GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (SRC_CAPS_TMPL)
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);
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/* some data */
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static guint8 mp3_frame[384] = {
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0xff, 0xfb, 0x94, 0xc4, 0xff, 0x83, 0xc0, 0x00,
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0x01, 0xa4, 0x00, 0x00, 0x00, 0x20, 0x00, 0x00,
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0x34, 0x80, 0x00, 0x00, 0x04, 0x00,
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};
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static guint8 garbage_frame[] = {
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0xff, 0xff, 0xff, 0xff, 0xff
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};
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GST_START_TEST (test_parse_normal)
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{
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gst_parser_test_normal (mp3_frame, sizeof (mp3_frame));
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}
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GST_END_TEST;
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GST_START_TEST (test_parse_drain_single)
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{
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gst_parser_test_drain_single (mp3_frame, sizeof (mp3_frame));
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}
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GST_END_TEST;
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GST_START_TEST (test_parse_drain_garbage)
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{
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gst_parser_test_drain_garbage (mp3_frame, sizeof (mp3_frame),
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garbage_frame, sizeof (garbage_frame));
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}
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GST_END_TEST;
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GST_START_TEST (test_parse_split)
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{
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gst_parser_test_split (mp3_frame, sizeof (mp3_frame));
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}
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GST_END_TEST;
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GST_START_TEST (test_parse_skip_garbage)
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{
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gst_parser_test_skip_garbage (mp3_frame, sizeof (mp3_frame),
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garbage_frame, sizeof (garbage_frame));
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}
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GST_END_TEST;
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#define structure_get_int(s,f) \
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(g_value_get_int(gst_structure_get_value(s,f)))
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#define fail_unless_structure_field_int_equals(s,field,num) \
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fail_unless_equals_int (structure_get_int(s,field), num)
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GST_START_TEST (test_parse_detect_stream)
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{
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GstStructure *s;
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GstCaps *caps;
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caps = gst_parser_test_get_output_caps (mp3_frame, sizeof (mp3_frame), NULL);
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fail_unless (caps != NULL);
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GST_LOG ("mpegaudio output caps: %" GST_PTR_FORMAT, caps);
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s = gst_caps_get_structure (caps, 0);
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fail_unless (gst_structure_has_name (s, "audio/mpeg"));
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fail_unless_structure_field_int_equals (s, "mpegversion", 1);
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fail_unless_structure_field_int_equals (s, "layer", 3);
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fail_unless_structure_field_int_equals (s, "channels", 1);
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fail_unless_structure_field_int_equals (s, "rate", 48000);
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gst_caps_unref (caps);
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}
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GST_END_TEST;
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static Suite *
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mpegaudioparse_suite (void)
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{
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Suite *s = suite_create ("mpegaudioparse");
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TCase *tc_chain = tcase_create ("general");
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/* init test context */
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ctx_factory = "mpegaudioparse";
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ctx_sink_template = &sinktemplate;
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ctx_src_template = &srctemplate;
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_parse_normal);
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tcase_add_test (tc_chain, test_parse_drain_single);
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tcase_add_test (tc_chain, test_parse_drain_garbage);
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tcase_add_test (tc_chain, test_parse_split);
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tcase_add_test (tc_chain, test_parse_skip_garbage);
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tcase_add_test (tc_chain, test_parse_detect_stream);
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return s;
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}
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/*
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* TODO:
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* - Both push- and pull-modes need to be tested
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* * Pull-mode & EOS
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*/
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GST_CHECK_MAIN (mpegaudioparse);
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