blob: c144e95faccfe4c0753e17e84866fa4ed4f81d8a [file] [log] [blame]
/*
* GStreamer
*
* unit test for aacparse
*
* Copyright (C) 2008 Nokia Corporation. All rights reserved.
*
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* 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.
*/
#include <gst/check/gstcheck.h>
#include "parser.h"
#define SRC_CAPS_TMPL "audio/mpeg, parsed=(boolean)false, mpegversion=(int)1"
#define SINK_CAPS_TMPL "audio/mpeg, parsed=(boolean)true, mpegversion=(int)1"
GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SINK_CAPS_TMPL)
);
GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SRC_CAPS_TMPL)
);
/* some data */
static guint8 mp3_frame[384] = {
0xff, 0xfb, 0x94, 0xc4, 0xff, 0x83, 0xc0, 0x00,
0x01, 0xa4, 0x00, 0x00, 0x00, 0x20, 0x00, 0x00,
0x34, 0x80, 0x00, 0x00, 0x04, 0x00,
};
static guint8 garbage_frame[] = {
0xff, 0xff, 0xff, 0xff, 0xff
};
GST_START_TEST (test_parse_normal)
{
gst_parser_test_normal (mp3_frame, sizeof (mp3_frame));
}
GST_END_TEST;
GST_START_TEST (test_parse_drain_single)
{
gst_parser_test_drain_single (mp3_frame, sizeof (mp3_frame));
}
GST_END_TEST;
GST_START_TEST (test_parse_drain_garbage)
{
gst_parser_test_drain_garbage (mp3_frame, sizeof (mp3_frame),
garbage_frame, sizeof (garbage_frame));
}
GST_END_TEST;
GST_START_TEST (test_parse_split)
{
gst_parser_test_split (mp3_frame, sizeof (mp3_frame));
}
GST_END_TEST;
GST_START_TEST (test_parse_skip_garbage)
{
gst_parser_test_skip_garbage (mp3_frame, sizeof (mp3_frame),
garbage_frame, sizeof (garbage_frame));
}
GST_END_TEST;
#define structure_get_int(s,f) \
(g_value_get_int(gst_structure_get_value(s,f)))
#define fail_unless_structure_field_int_equals(s,field,num) \
fail_unless_equals_int (structure_get_int(s,field), num)
GST_START_TEST (test_parse_detect_stream)
{
GstStructure *s;
GstCaps *caps;
caps = gst_parser_test_get_output_caps (mp3_frame, sizeof (mp3_frame), NULL);
fail_unless (caps != NULL);
GST_LOG ("mpegaudio output caps: %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
fail_unless (gst_structure_has_name (s, "audio/mpeg"));
fail_unless_structure_field_int_equals (s, "mpegversion", 1);
fail_unless_structure_field_int_equals (s, "layer", 3);
fail_unless_structure_field_int_equals (s, "channels", 1);
fail_unless_structure_field_int_equals (s, "rate", 48000);
gst_caps_unref (caps);
}
GST_END_TEST;
static Suite *
mpegaudioparse_suite (void)
{
Suite *s = suite_create ("mpegaudioparse");
TCase *tc_chain = tcase_create ("general");
/* init test context */
ctx_factory = "mpegaudioparse";
ctx_sink_template = &sinktemplate;
ctx_src_template = &srctemplate;
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_parse_normal);
tcase_add_test (tc_chain, test_parse_drain_single);
tcase_add_test (tc_chain, test_parse_drain_garbage);
tcase_add_test (tc_chain, test_parse_split);
tcase_add_test (tc_chain, test_parse_skip_garbage);
tcase_add_test (tc_chain, test_parse_detect_stream);
return s;
}
/*
* TODO:
* - Both push- and pull-modes need to be tested
* * Pull-mode & EOS
*/
GST_CHECK_MAIN (mpegaudioparse);