gstreamer/subprojects/gst-plugins-good/tests/check/elements/wavpackdec.c

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/* GStreamer
*
* unit test for wavpackdec
*
* Copyright (c) 2006 Sebastian Dröge <slomo@circular-chaos.org>
*
* 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 <unistd.h>
#include <gst/check/gstcheck.h>
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
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#if G_BYTE_ORDER == G_BIG_ENDIAN
#define AUDIO_FORMAT "S16BE"
#else
#define AUDIO_FORMAT "S16LE"
#endif
guint8 test_frame[] = {
0x77, 0x76, 0x70, 0x6B, /* "wvpk" */
0x2E, 0x00, 0x00, 0x00, /* ckSize */
0x04, 0x04, /* version */
0x00, /* track_no */
0x00, /* index_no */
0x00, 0x64, 0x00, 0x00, /* total_samples */
0x00, 0x00, 0x00, 0x00, /* block_index */
0x00, 0x64, 0x00, 0x00, /* block_samples */
0x05, 0x18, 0x80, 0x04, /* flags */
0xFF, 0xAF, 0x80, 0x60, /* crc */
0x02, 0x00, 0x03, 0x00, /* data */
0x04, 0x00, 0x05, 0x03,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x8A, 0x02,
0x00, 0x00, 0xFF, 0x7F,
0x00, 0xE4,
};
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " AUDIO_FORMAT ", "
"layout = (string) interleaved, "
"channels = (int) 1, " "rate = (int) 44100")
ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept th... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_chain): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.c: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept this and let audioconvert convert to accepted formats instead of doing it in the element for n*8 depths. This also adds support for non-n*8 depths and prevents some useless memory allocations. Fixes #421598 Also add a workaround for bug #421542 in wavpackenc for now... * tests/check/elements/wavpackdec.c: (GST_START_TEST): * tests/check/elements/wavpackenc.c: (GST_START_TEST): * tests/check/elements/wavpackparse.c: (GST_START_TEST): Consider the change above in the unit tests and test if the correct caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in the wavpackparse unit test. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps): Set caps on the src pad as soon as possible. * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.h: Fix indention. gst-indent is now called by cicl.
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);
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#define WAVPACK_CAPS "audio/x-wavpack, " \
"depth = (int) 16, " \
"channels = (int) 1, " "rate = (int) 44100, " "framed = (boolean) true"
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (WAVPACK_CAPS)
);
static GstElement *
setup_wavpackdec (void)
{
GstElement *wavpackdec;
GstCaps *caps;
GST_DEBUG ("setup_wavpackdec");
wavpackdec = gst_check_setup_element ("wavpackdec");
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mysrcpad = gst_check_setup_src_pad (wavpackdec, &srctemplate);
mysinkpad = gst_check_setup_sink_pad (wavpackdec, &sinktemplate);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
caps = gst_caps_from_string (WAVPACK_CAPS);
gst_check_setup_events (mysrcpad, wavpackdec, caps, GST_FORMAT_TIME);
gst_caps_unref (caps);
fail_unless (gst_element_set_state (wavpackdec,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
return wavpackdec;
}
static void
cleanup_wavpackdec (GstElement * wavpackdec)
{
GST_DEBUG ("cleanup_wavpackdec");
gst_element_set_state (wavpackdec, GST_STATE_NULL);
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (wavpackdec);
gst_check_teardown_sink_pad (wavpackdec);
gst_check_teardown_element (wavpackdec);
}
GST_START_TEST (test_decode_frame)
{
GstElement *wavpackdec;
GstBuffer *inbuffer, *outbuffer;
GstBus *bus;
int i;
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GstMapInfo map;
wavpackdec = setup_wavpackdec ();
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame));
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gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame));
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_element_set_bus (wavpackdec, bus);
/* should decode the buffer without problems */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_OK);
outbuffer = GST_BUFFER (buffers->data);
fail_if (outbuffer == NULL);
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gst_buffer_map (outbuffer, &map, GST_MAP_READ);
ext/wavpack/: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept th... Original commit message from CVS: * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset), (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps), (gst_wavpack_dec_clip_outgoing_buffer), (gst_wavpack_dec_post_tags), (gst_wavpack_dec_chain): * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_reset), (gst_wavpack_enc_sink_set_caps), (gst_wavpack_enc_set_wp_config), (gst_wavpack_enc_chain): * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.c: Don't play audioconvert. As wavpack wants/outputs all samples with width==32 and depth=[1,32] accept this and let audioconvert convert to accepted formats instead of doing it in the element for n*8 depths. This also adds support for non-n*8 depths and prevents some useless memory allocations. Fixes #421598 Also add a workaround for bug #421542 in wavpackenc for now... * tests/check/elements/wavpackdec.c: (GST_START_TEST): * tests/check/elements/wavpackenc.c: (GST_START_TEST): * tests/check/elements/wavpackparse.c: (GST_START_TEST): Consider the change above in the unit tests and test if the correct caps are accepted and set. Also check for GST_BUFFER_OFFSET_END in the wavpackparse unit test. * ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_init), (gst_wavpack_dec_sink_set_caps): Set caps on the src pad as soon as possible. * ext/wavpack/gstwavpackdec.h: * ext/wavpack/gstwavpackcommon.h: * ext/wavpack/gstwavpackenc.h: * ext/wavpack/gstwavpackparse.h: Fix indention. gst-indent is now called by cicl.
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/* uncompressed data should be 102400 bytes */
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fail_unless_equals_int (map.size, 51200);
/* and all bytes must be 0, i.e. silence */
for (i = 0; i < 51200; i++)
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fail_unless_equals_int (map.data[i], 0);
gst_buffer_unmap (outbuffer, &map);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
gst_buffer_unref (outbuffer);
outbuffer = NULL;
g_list_free (buffers);
buffers = NULL;
gst_bus_set_flushing (bus, TRUE);
gst_element_set_bus (wavpackdec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_wavpackdec (wavpackdec);
}
GST_END_TEST;
GST_START_TEST (test_decode_frame_with_broken_header)
{
GstElement *wavpackdec;
GstBuffer *inbuffer;
GstBus *bus;
GstMessage *message;
wavpackdec = setup_wavpackdec ();
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame));
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gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame));
/* break header */
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gst_buffer_memset (inbuffer, 2, 'e', 1);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);
gst_message_unref (message);
gst_element_set_bus (wavpackdec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_wavpackdec (wavpackdec);
}
GST_END_TEST;
GST_START_TEST (test_decode_frame_with_incomplete_frame)
{
GstElement *wavpackdec;
GstBuffer *inbuffer;
GstBus *bus;
GstMessage *message;
wavpackdec = setup_wavpackdec ();
bus = gst_bus_new ();
inbuffer = gst_buffer_new_and_alloc (sizeof (test_frame) - 2);
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gst_buffer_fill (inbuffer, 0, test_frame, sizeof (test_frame) - 2);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
gst_element_set_bus (wavpackdec, bus);
/* should fail gracefully */
fail_unless_equals_int (gst_pad_push (mysrcpad, inbuffer), GST_FLOW_ERROR);
fail_if ((message = gst_bus_pop (bus)) == NULL);
fail_unless_message_error (message, STREAM, DECODE);
gst_message_unref (message);
gst_element_set_bus (wavpackdec, NULL);
gst_object_unref (GST_OBJECT (bus));
cleanup_wavpackdec (wavpackdec);
}
GST_END_TEST;
static Suite *
wavpackdec_suite (void)
{
Suite *s = suite_create ("wavpackdec");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_decode_frame);
tcase_add_test (tc_chain, test_decode_frame_with_broken_header);
tcase_add_test (tc_chain, test_decode_frame_with_incomplete_frame);
return s;
}
GST_CHECK_MAIN (wavpackdec);