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faad: add unit test
This commit is contained in:
Mark Nauwelaerts
parent
89ac50f14c
commit
4312ec8b40
2 changed files with 232 additions and 0 deletions
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@ -33,6 +33,12 @@ else
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check_faac =
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endif
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if USE_FAAD
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check_faad = elements/faad
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else
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check_faad =
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endif
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if USE_METADATA
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check_metadata = pipelines/metadata
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else
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@ -102,6 +108,7 @@ noinst_PROGRAMS = \
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check_PROGRAMS = \
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generic/states \
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$(check_faac) \
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$(check_faad) \
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$(check_mpeg2enc) \
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$(check_mplex) \
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$(check_ofa) \
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225
tests/check/elements/faad.c
Normal file
225
tests/check/elements/faad.c
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@ -0,0 +1,225 @@
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/* GStreamer
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*
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* unit test for faad
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*
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* Copyright (C) <2009> Mark Nauwelaerts <mnauw@users.sf.net>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include <unistd.h>
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#include <gst/check/gstcheck.h>
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/* For ease of programming we use globals to keep refs for our floating
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* src and sink pads we create; otherwise we always have to do get_pad,
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* get_peer, and then remove references in every test function */
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static GstPad *mysrcpad, *mysinkpad;
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#define AUDIO_CAPS_STRING "audio/x-raw-int, " \
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"rate = (int) 48000, " \
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"channels = (int) 2, " \
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"width = (int) 16, " \
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"depth = (int) 16, " \
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"signed = (boolean) true, " \
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"endianness = (int) BYTE_ORDER "
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#define AAC_CAPS_STRING "audio/mpeg, " \
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"mpegversion = (int) 4, " \
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"rate = (int) 48000, " \
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"channels = (int) 2, " \
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"framed = (boolean) true "
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static 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 (AUDIO_CAPS_STRING));
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static 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 (AAC_CAPS_STRING));
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GstElement *
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setup_faad ()
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{
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GstElement *faad;
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GST_DEBUG ("setup_faad");
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faad = gst_check_setup_element ("faad");
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mysrcpad = gst_check_setup_src_pad (faad, &srctemplate, NULL);
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mysinkpad = gst_check_setup_sink_pad (faad, &sinktemplate, NULL);
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gst_pad_set_active (mysrcpad, TRUE);
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gst_pad_set_active (mysinkpad, TRUE);
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return faad;
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}
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void
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cleanup_faad (GstElement * faad)
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{
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GST_DEBUG ("cleanup_faad");
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gst_element_set_state (faad, GST_STATE_NULL);
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gst_pad_set_active (mysrcpad, FALSE);
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gst_pad_set_active (mysinkpad, FALSE);
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gst_check_teardown_src_pad (faad);
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gst_check_teardown_sink_pad (faad);
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gst_check_teardown_element (faad);
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}
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void
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do_test (GstBuffer * inbuffer)
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{
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GstElement *faad;
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GstBuffer *outbuffer;
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gint i, num_buffers;
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const gint nbuffers = 2;
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faad = setup_faad ();
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fail_unless (gst_element_set_state (faad,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
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"could not set to playing");
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GST_BUFFER_TIMESTAMP (inbuffer) = 0;
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ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
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/* need to push twice to get faad output */
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gst_buffer_ref (inbuffer);
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
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/* send eos to have all flushed if needed */
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fail_unless (gst_pad_push_event (mysrcpad, gst_event_new_eos ()) == TRUE);
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num_buffers = g_list_length (buffers);
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fail_unless (num_buffers >= nbuffers - 1);
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/* clean up buffers */
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for (i = 0; i < num_buffers; ++i) {
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gint size;
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guint8 *data;
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outbuffer = GST_BUFFER (buffers->data);
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fail_if (outbuffer == NULL);
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data = GST_BUFFER_DATA (outbuffer);
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size = GST_BUFFER_SIZE (outbuffer);
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/* 2 16-bit channels */
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fail_unless (size == 1024 * 2 * 2);
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buffers = g_list_remove (buffers, outbuffer);
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ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
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gst_buffer_unref (outbuffer);
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outbuffer = NULL;
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}
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cleanup_faad (faad);
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g_list_free (buffers);
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buffers = NULL;
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}
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static guint8 raw_data_block[] = {
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0x21, 0x1b, 0x80, 0x00, 0x7d, 0xe0, 0x00, 0x3e, 0xf1, 0xe7
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};
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static guint8 adts_header[] = {
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0xff, 0xf8, 0x4c, 0x80, 0x02, 0x7f, 0xfc, 0x04, 0x40
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};
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static guint8 codec_data[] = {
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0x11, 0x90
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};
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GST_START_TEST (test_adts)
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{
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gint size;
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GstBuffer *buf, *header_buf;
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GstCaps *caps;
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size = sizeof (adts_header);
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header_buf = gst_buffer_new_and_alloc (size);
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memcpy (GST_BUFFER_DATA (header_buf), adts_header, size);
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size = sizeof (raw_data_block);
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buf = gst_buffer_new_and_alloc (size);
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memcpy (GST_BUFFER_DATA (buf), raw_data_block, size);
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buf = gst_buffer_join (header_buf, buf);
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caps = gst_caps_from_string (AAC_CAPS_STRING);
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gst_buffer_set_caps (buf, caps);
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gst_caps_unref (caps);
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do_test (buf);
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}
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GST_END_TEST;
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GST_START_TEST (test_raw)
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{
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gint size;
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GstBuffer *buf, *codec_buf;
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GstCaps *caps;
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size = sizeof (codec_data);
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codec_buf = gst_buffer_new_and_alloc (size);
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memcpy (GST_BUFFER_DATA (codec_buf), codec_data, size);
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size = sizeof (raw_data_block);
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buf = gst_buffer_new_and_alloc (size);
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memcpy (GST_BUFFER_DATA (buf), raw_data_block, size);
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caps = gst_caps_from_string (AAC_CAPS_STRING);
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_buf, NULL);
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gst_buffer_unref (codec_buf);
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gst_buffer_set_caps (buf, caps);
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gst_caps_unref (caps);
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do_test (buf);
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}
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GST_END_TEST;
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Suite *
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faad_suite (void)
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{
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Suite *s = suite_create ("faad");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_adts);
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tcase_add_test (tc_chain, test_raw);
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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 = faad_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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