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check/: add a test for audioconvert

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Original commit message from CVS:

* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
This commit is contained in:
Thomas Vander Stichele 2005-08-25 17:20:02 +00:00
parent 8f3a11d6f2
commit 6dff9c2cbd
7 changed files with 445 additions and 6 deletions
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13
ChangeLog
View file

@ -1,3 +1,16 @@
2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
2005-08-25 Thomas Vander Stichele <thomas at apestaart dot org>
* check/Makefile.am:

3
check/Makefile.am
View file

@ -29,13 +29,14 @@ check_vorbis =
endif
check_PROGRAMS = \
elements/audioconvert \
elements/audioresample \
elements/volume \
$(check_vorbis)
# these tests don't even pass
# generic/states: elements need state fixin' before this can be added
noinst_PROGRAMS = \
elements/audioresample \
generic/states
AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS)

209
check/elements/audioconvert.c Normal file
View file

@ -0,0 +1,209 @@
/* GStreamer
*
* unit test for audioconvert
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
GList *buffers = NULL;
gboolean have_eos = FALSE;
/* 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 */
GstPad *mysrcpad, *mysinkpad;
#define CONVERT_CAPS_TEMPLATE_STRING \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"buffer-frames = (int) [ 0, MAX ];" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 24, " \
"depth = (int) [ 1, 24 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 16, " \
"depth = (int) [ 1, 16 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 8, " \
"depth = (int) [ 1, 8 ], " \
"signed = (boolean) { true, false } "
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
);
/* takes over reference for outcaps */
GstElement *
setup_audioconvert (GstCaps * outcaps)
{
GstElement *audioconvert;
GST_DEBUG ("setup_audioconvert");
audioconvert = gst_check_setup_element ("audioconvert");
mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
/* this installs a getcaps func that will always return the caps we set
* later */
gst_pad_use_fixed_caps (mysinkpad);
gst_pad_set_caps (mysinkpad, outcaps);
gst_caps_unref (outcaps);
outcaps = gst_pad_get_negotiated_caps (mysinkpad);
fail_unless (gst_caps_is_fixed (outcaps));
gst_caps_unref (outcaps);
return audioconvert;
}
void
cleanup_audioconvert (GstElement * audioconvert)
{
GST_DEBUG ("cleanup_audioconvert");
gst_check_teardown_src_pad (audioconvert);
gst_check_teardown_sink_pad (audioconvert);
gst_check_teardown_element (audioconvert);
}
GstCaps *
get_int_caps (guint rate, guint channels, gchar * endianness, guint width,
guint depth, gboolean signedness)
{
GstCaps *caps;
gchar *string;
string = g_strdup_printf ("audio/x-raw-int, "
"rate = (int) %d, "
"channels = (int) %d, "
"endianness = (int) %s, "
"width = (int) %d, "
"depth = (int) %d, "
"signed = (boolean) %s ",
rate, channels, endianness, width, depth, signedness ? "true" : "false");
GST_DEBUG ("creating caps from %s", string);
caps = gst_caps_from_string (string);
fail_unless (caps != NULL);
g_free (string);
return caps;
}
static void
verify_convert (GstElement * audioconvert, void *in, int inlength, void *out,
int outlength, GstCaps * incaps)
{
GstBuffer *inbuffer, *outbuffer;
fail_unless (gst_element_set_state (audioconvert,
GST_STATE_PLAYING) == GST_STATE_SUCCESS, "could not set to playing");
GST_DEBUG ("Creating buffer of %d bytes", inlength);
inbuffer = gst_buffer_new_and_alloc (inlength);
memcpy (GST_BUFFER_DATA (inbuffer), in, inlength);
gst_buffer_set_caps (inbuffer, incaps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) == 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), outlength);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0);
}
GST_START_TEST (test_unity)
{
GstElement *audioconvert;
GstCaps *incaps, *outcaps;
gint16 in[] = { 16384, -256 };
gint16 out[] = { 8064 };
outcaps = get_int_caps (44100, 1, "LITTLE_ENDIAN", 16, 16, TRUE);
audioconvert = setup_audioconvert (outcaps);
incaps = get_int_caps (44100, 2, "LITTLE_ENDIAN", 16, 16, TRUE);
verify_convert (audioconvert, in, sizeof (in), out, sizeof (out), incaps);
/* cleanup */
cleanup_audioconvert (audioconvert);
}
GST_END_TEST;
Suite *
audioconvert_suite (void)
{
Suite *s = suite_create ("audioconvert");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_unity);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audioconvert_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

12
gst/audioresample/gstaudioresample.c
View file

@ -395,14 +395,20 @@ static GstFlowReturn
GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
outsize, outsamples);
GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
GST_BUFFER_DURATION (outbuf) =
outsamples * GST_SECOND / audioresample->o_rate;
GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
audioresample->offset += outsamples;
GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
/* we calculate DURATION as the difference between "next" timestamp
* and current timestamp so we ensure a contiguous stream, instead of
* having rounding errors. */
GST_BUFFER_DURATION (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate -
GST_BUFFER_TIMESTAMP (outbuf);
/* check for possible mem corruption */
if (outsize > GST_BUFFER_SIZE (outbuf)) {
/* this is an error that when it happens, would need fixing in the

2
gst/audioresample/gstaudioresample.h
View file

@ -52,7 +52,7 @@ struct _GstAudioresample {
gboolean passthru;
gint64 offset;
guint64 offset;
int channels;
int i_rate;

3
tests/check/Makefile.am
View file

@ -29,13 +29,14 @@ check_vorbis =
endif
check_PROGRAMS = \
elements/audioconvert \
elements/audioresample \
elements/volume \
$(check_vorbis)
# these tests don't even pass
# generic/states: elements need state fixin' before this can be added
noinst_PROGRAMS = \
elements/audioresample \
generic/states
AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS)

209
tests/check/elements/audioconvert.c Normal file
View file

@ -0,0 +1,209 @@
/* GStreamer
*
* unit test for audioconvert
*
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <unistd.h>
#include <gst/check/gstcheck.h>
GList *buffers = NULL;
gboolean have_eos = FALSE;
/* 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 */
GstPad *mysrcpad, *mysinkpad;
#define CONVERT_CAPS_TEMPLATE_STRING \
"audio/x-raw-float, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 32, " \
"buffer-frames = (int) [ 0, MAX ];" \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 32, " \
"depth = (int) [ 1, 32 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 24, " \
"depth = (int) [ 1, 24 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 16, " \
"depth = (int) [ 1, 16 ], " \
"signed = (boolean) { true, false }; " \
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 8 ], " \
"endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \
"width = (int) 8, " \
"depth = (int) [ 1, 8 ], " \
"signed = (boolean) { true, false } "
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING)
);
/* takes over reference for outcaps */
GstElement *
setup_audioconvert (GstCaps * outcaps)
{
GstElement *audioconvert;
GST_DEBUG ("setup_audioconvert");
audioconvert = gst_check_setup_element ("audioconvert");
mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL);
/* this installs a getcaps func that will always return the caps we set
* later */
gst_pad_use_fixed_caps (mysinkpad);
gst_pad_set_caps (mysinkpad, outcaps);
gst_caps_unref (outcaps);
outcaps = gst_pad_get_negotiated_caps (mysinkpad);
fail_unless (gst_caps_is_fixed (outcaps));
gst_caps_unref (outcaps);
return audioconvert;
}
void
cleanup_audioconvert (GstElement * audioconvert)
{
GST_DEBUG ("cleanup_audioconvert");
gst_check_teardown_src_pad (audioconvert);
gst_check_teardown_sink_pad (audioconvert);
gst_check_teardown_element (audioconvert);
}
GstCaps *
get_int_caps (guint rate, guint channels, gchar * endianness, guint width,
guint depth, gboolean signedness)
{
GstCaps *caps;
gchar *string;
string = g_strdup_printf ("audio/x-raw-int, "
"rate = (int) %d, "
"channels = (int) %d, "
"endianness = (int) %s, "
"width = (int) %d, "
"depth = (int) %d, "
"signed = (boolean) %s ",
rate, channels, endianness, width, depth, signedness ? "true" : "false");
GST_DEBUG ("creating caps from %s", string);
caps = gst_caps_from_string (string);
fail_unless (caps != NULL);
g_free (string);
return caps;
}
static void
verify_convert (GstElement * audioconvert, void *in, int inlength, void *out,
int outlength, GstCaps * incaps)
{
GstBuffer *inbuffer, *outbuffer;
fail_unless (gst_element_set_state (audioconvert,
GST_STATE_PLAYING) == GST_STATE_SUCCESS, "could not set to playing");
GST_DEBUG ("Creating buffer of %d bytes", inlength);
inbuffer = gst_buffer_new_and_alloc (inlength);
memcpy (GST_BUFFER_DATA (inbuffer), in, inlength);
gst_buffer_set_caps (inbuffer, incaps);
ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
/* pushing gives away my reference ... */
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
/* ... and puts a new buffer on the global list */
fail_unless (g_list_length (buffers) == 1);
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), outlength);
fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0);
}
GST_START_TEST (test_unity)
{
GstElement *audioconvert;
GstCaps *incaps, *outcaps;
gint16 in[] = { 16384, -256 };
gint16 out[] = { 8064 };
outcaps = get_int_caps (44100, 1, "LITTLE_ENDIAN", 16, 16, TRUE);
audioconvert = setup_audioconvert (outcaps);
incaps = get_int_caps (44100, 2, "LITTLE_ENDIAN", 16, 16, TRUE);
verify_convert (audioconvert, in, sizeof (in), out, sizeof (out), incaps);
/* cleanup */
cleanup_audioconvert (audioconvert);
}
GST_END_TEST;
Suite *
audioconvert_suite (void)
{
Suite *s = suite_create ("audioconvert");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_unity);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = audioconvert_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_NORMAL);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}