/* * GStreamer * * unit test for aacparse * * Copyright (C) 2008 Nokia Corporation. All rights reserved. * * Contact: Stefan Kost * * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #include #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) ); const gchar *factory = "aacparse"; /* 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"); 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 */ int main (int argc, char **argv) { int nf; Suite *s = mpegaudioparse_suite (); SRunner *sr = srunner_create (s); gst_check_init (&argc, &argv); /* init test context */ ctx_factory = "mpegaudioparse"; ctx_sink_template = &sinktemplate; ctx_src_template = &srctemplate; srunner_run_all (sr, CK_NORMAL); nf = srunner_ntests_failed (sr); srunner_free (sr); return nf; }