mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-06 10:42:22 +00:00
Original commit message from CVS: 2007-03-14 Julien MOUTTE <julien@moutte.net> * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (audioresample_transform_size), (audioresample_do_output), (audioresample_transform), (audioresample_pushthrough): Handle discontinuous streams. * gst/audioresample/gstaudioresample.h: * tests/check/elements/audioresample.c: (test_discont_stream_instance), (GST_START_TEST), (audioresample_suite): Add a test for discontinuous streams. * win32/common/config.h: Updated.
403 lines
12 KiB
C
403 lines
12 KiB
C
/* GStreamer
|
|
*
|
|
* unit test for audioresample
|
|
*
|
|
* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
|
|
* Copyright (C) <2006> Tim-Philipp Müller <tim at centricular net>
|
|
*
|
|
* 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>
|
|
|
|
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 RESAMPLE_CAPS_TEMPLATE_STRING \
|
|
"audio/x-raw-int, " \
|
|
"channels = (int) [ 1, MAX ], " \
|
|
"rate = (int) [ 1, MAX ], " \
|
|
"endianness = (int) BYTE_ORDER, " \
|
|
"width = (int) 16, " \
|
|
"depth = (int) 16, " \
|
|
"signed = (bool) TRUE"
|
|
|
|
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
|
|
);
|
|
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
|
|
GST_PAD_SRC,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS (RESAMPLE_CAPS_TEMPLATE_STRING)
|
|
);
|
|
|
|
GstElement *
|
|
setup_audioresample (int channels, int inrate, int outrate)
|
|
{
|
|
GstElement *audioresample;
|
|
GstCaps *caps;
|
|
GstStructure *structure;
|
|
GstPad *pad;
|
|
|
|
GST_DEBUG ("setup_audioresample");
|
|
audioresample = gst_check_setup_element ("audioresample");
|
|
|
|
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
|
|
"rate", G_TYPE_INT, inrate, NULL);
|
|
fail_unless (gst_caps_is_fixed (caps));
|
|
|
|
fail_unless (gst_element_set_state (audioresample,
|
|
GST_STATE_PAUSED) == GST_STATE_CHANGE_SUCCESS,
|
|
"could not set to paused");
|
|
|
|
mysrcpad = gst_check_setup_src_pad (audioresample, &srctemplate, caps);
|
|
pad = gst_pad_get_peer (mysrcpad);
|
|
gst_pad_set_caps (pad, caps);
|
|
gst_object_unref (GST_OBJECT (pad));
|
|
gst_caps_unref (caps);
|
|
gst_pad_set_active (mysrcpad, TRUE);
|
|
|
|
caps = gst_caps_from_string (RESAMPLE_CAPS_TEMPLATE_STRING);
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
gst_structure_set (structure, "channels", G_TYPE_INT, channels,
|
|
"rate", G_TYPE_INT, outrate, NULL);
|
|
fail_unless (gst_caps_is_fixed (caps));
|
|
|
|
mysinkpad = gst_check_setup_sink_pad (audioresample, &sinktemplate, caps);
|
|
/* this installs a getcaps func that will always return the caps we set
|
|
* later */
|
|
gst_pad_use_fixed_caps (mysinkpad);
|
|
pad = gst_pad_get_peer (mysinkpad);
|
|
gst_pad_set_caps (pad, caps);
|
|
gst_object_unref (GST_OBJECT (pad));
|
|
gst_caps_unref (caps);
|
|
gst_pad_set_active (mysinkpad, TRUE);
|
|
|
|
return audioresample;
|
|
}
|
|
|
|
void
|
|
cleanup_audioresample (GstElement * audioresample)
|
|
{
|
|
GST_DEBUG ("cleanup_audioresample");
|
|
|
|
fail_unless (gst_element_set_state (audioresample,
|
|
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
|
|
|
|
gst_pad_set_active (mysrcpad, FALSE);
|
|
gst_pad_set_active (mysinkpad, FALSE);
|
|
gst_check_teardown_src_pad (audioresample);
|
|
gst_check_teardown_sink_pad (audioresample);
|
|
gst_check_teardown_element (audioresample);
|
|
}
|
|
|
|
static void
|
|
fail_unless_perfect_stream ()
|
|
{
|
|
guint64 timestamp = 0L, duration = 0L;
|
|
guint64 offset = 0L, offset_end = 0L;
|
|
|
|
GList *l;
|
|
GstBuffer *buffer;
|
|
|
|
for (l = buffers; l; l = l->next) {
|
|
buffer = GST_BUFFER (l->data);
|
|
ASSERT_BUFFER_REFCOUNT (buffer, "buffer", 1);
|
|
GST_DEBUG ("buffer timestamp %" G_GUINT64_FORMAT ", duration %"
|
|
G_GUINT64_FORMAT, GST_BUFFER_TIMESTAMP (buffer),
|
|
GST_BUFFER_DURATION (buffer));
|
|
|
|
fail_unless_equals_uint64 (timestamp, GST_BUFFER_TIMESTAMP (buffer));
|
|
fail_unless_equals_uint64 (offset, GST_BUFFER_OFFSET (buffer));
|
|
duration = GST_BUFFER_DURATION (buffer);
|
|
offset_end = GST_BUFFER_OFFSET_END (buffer);
|
|
|
|
timestamp += duration;
|
|
offset = offset_end;
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
g_list_free (buffers);
|
|
buffers = NULL;
|
|
}
|
|
|
|
/* this tests that the output is a perfect stream if the input is */
|
|
static void
|
|
test_perfect_stream_instance (int inrate, int outrate, int samples,
|
|
int numbuffers)
|
|
{
|
|
GstElement *audioresample;
|
|
GstBuffer *inbuffer, *outbuffer;
|
|
GstCaps *caps;
|
|
|
|
int i, j;
|
|
gint16 *p;
|
|
|
|
audioresample = setup_audioresample (2, inrate, outrate);
|
|
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
|
fail_unless (gst_caps_is_fixed (caps));
|
|
|
|
fail_unless (gst_element_set_state (audioresample,
|
|
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
|
"could not set to playing");
|
|
|
|
for (j = 1; j <= numbuffers; ++j) {
|
|
|
|
inbuffer = gst_buffer_new_and_alloc (samples * 4);
|
|
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
|
|
GST_BUFFER_TIMESTAMP (inbuffer) = GST_BUFFER_DURATION (inbuffer) * (j - 1);
|
|
GST_BUFFER_OFFSET (inbuffer) = 0;
|
|
GST_BUFFER_OFFSET_END (inbuffer) = samples;
|
|
|
|
gst_buffer_set_caps (inbuffer, caps);
|
|
|
|
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
|
|
|
|
/* create a 16 bit signed ramp */
|
|
for (i = 0; i < samples; ++i) {
|
|
*p = -32767 + i * (65535 / samples);
|
|
++p;
|
|
*p = -32767 + i * (65535 / samples);
|
|
++p;
|
|
}
|
|
|
|
/* pushing gives away my reference ... */
|
|
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
|
/* ... but it ends up being collected on the global buffer list */
|
|
fail_unless_equals_int (g_list_length (buffers), j);
|
|
}
|
|
|
|
/* FIXME: we should make audioresample handle eos by flushing out the last
|
|
* samples, which will give us one more, small, buffer */
|
|
fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
|
|
ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1);
|
|
|
|
fail_unless_perfect_stream ();
|
|
|
|
/* cleanup */
|
|
gst_caps_unref (caps);
|
|
cleanup_audioresample (audioresample);
|
|
}
|
|
|
|
|
|
/* make sure that outgoing buffers are contiguous in timestamp/duration and
|
|
* offset/offsetend
|
|
*/
|
|
GST_START_TEST (test_perfect_stream)
|
|
{
|
|
/* integral scalings */
|
|
test_perfect_stream_instance (48000, 24000, 500, 20);
|
|
test_perfect_stream_instance (48000, 12000, 500, 20);
|
|
test_perfect_stream_instance (12000, 24000, 500, 20);
|
|
test_perfect_stream_instance (12000, 48000, 500, 20);
|
|
|
|
/* non-integral scalings */
|
|
test_perfect_stream_instance (44100, 8000, 500, 20);
|
|
test_perfect_stream_instance (8000, 44100, 500, 20);
|
|
|
|
/* wacky scalings */
|
|
test_perfect_stream_instance (12345, 54321, 500, 20);
|
|
test_perfect_stream_instance (101, 99, 500, 20);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
/* this tests that the output is a correct discontinuous stream
|
|
* if the input is; ie input drops in time come out the same way */
|
|
static void
|
|
test_discont_stream_instance (int inrate, int outrate, int samples,
|
|
int numbuffers)
|
|
{
|
|
GstElement *audioresample;
|
|
GstBuffer *inbuffer, *outbuffer;
|
|
GstCaps *caps;
|
|
GstClockTime ints;
|
|
|
|
int i, j;
|
|
gint16 *p;
|
|
|
|
audioresample = setup_audioresample (2, inrate, outrate);
|
|
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
|
fail_unless (gst_caps_is_fixed (caps));
|
|
|
|
fail_unless (gst_element_set_state (audioresample,
|
|
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
|
"could not set to playing");
|
|
|
|
for (j = 1; j <= numbuffers; ++j) {
|
|
|
|
inbuffer = gst_buffer_new_and_alloc (samples * 4);
|
|
GST_BUFFER_DURATION (inbuffer) = samples * GST_SECOND / inrate;
|
|
/* "drop" half the buffers */
|
|
ints = GST_BUFFER_DURATION (inbuffer) * 2 * (j - 1);
|
|
GST_BUFFER_TIMESTAMP (inbuffer) = ints;
|
|
GST_BUFFER_OFFSET (inbuffer) = (j - 1) * 2 * samples;
|
|
GST_BUFFER_OFFSET_END (inbuffer) = j * 2 * samples + samples;
|
|
|
|
gst_buffer_set_caps (inbuffer, caps);
|
|
|
|
p = (gint16 *) GST_BUFFER_DATA (inbuffer);
|
|
|
|
/* create a 16 bit signed ramp */
|
|
for (i = 0; i < samples; ++i) {
|
|
*p = -32767 + i * (65535 / samples);
|
|
++p;
|
|
*p = -32767 + i * (65535 / samples);
|
|
++p;
|
|
}
|
|
|
|
/* pushing gives away my reference ... */
|
|
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
|
|
|
/* check if the timestamp of the pushed buffer matches the incoming one */
|
|
outbuffer = g_list_nth_data (buffers, g_list_length (buffers) - 1);
|
|
fail_if (outbuffer == NULL);
|
|
fail_unless_equals_uint64 (ints, GST_BUFFER_TIMESTAMP (outbuffer));
|
|
if (j > 1) {
|
|
fail_unless (GST_BUFFER_IS_DISCONT (outbuffer),
|
|
"expected discont buffer");
|
|
}
|
|
}
|
|
|
|
/* cleanup */
|
|
gst_caps_unref (caps);
|
|
cleanup_audioresample (audioresample);
|
|
}
|
|
|
|
GST_START_TEST (test_discont_stream)
|
|
{
|
|
/* integral scalings */
|
|
test_discont_stream_instance (48000, 24000, 500, 20);
|
|
test_discont_stream_instance (48000, 12000, 500, 20);
|
|
test_discont_stream_instance (12000, 24000, 500, 20);
|
|
test_discont_stream_instance (12000, 48000, 500, 20);
|
|
|
|
/* non-integral scalings */
|
|
test_discont_stream_instance (44100, 8000, 500, 20);
|
|
test_discont_stream_instance (8000, 44100, 500, 20);
|
|
|
|
/* wacky scalings */
|
|
test_discont_stream_instance (12345, 54321, 500, 20);
|
|
test_discont_stream_instance (101, 99, 500, 20);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
|
|
|
|
GST_START_TEST (test_reuse)
|
|
{
|
|
GstElement *audioresample;
|
|
GstEvent *newseg;
|
|
GstBuffer *inbuffer;
|
|
GstCaps *caps;
|
|
|
|
audioresample = setup_audioresample (1, 9343, 48000);
|
|
caps = gst_pad_get_negotiated_caps (mysrcpad);
|
|
fail_unless (gst_caps_is_fixed (caps));
|
|
|
|
fail_unless (gst_element_set_state (audioresample,
|
|
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
|
"could not set to playing");
|
|
|
|
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
|
|
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
|
|
|
|
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
|
|
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
|
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
|
|
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
|
|
GST_BUFFER_OFFSET (inbuffer) = 0;
|
|
gst_buffer_set_caps (inbuffer, caps);
|
|
|
|
/* pushing gives away my reference ... */
|
|
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
|
|
|
/* ... but it ends up being collected on the global buffer list */
|
|
fail_unless_equals_int (g_list_length (buffers), 1);
|
|
|
|
/* now reset and try again ... */
|
|
fail_unless (gst_element_set_state (audioresample,
|
|
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS, "could not set to NULL");
|
|
|
|
fail_unless (gst_element_set_state (audioresample,
|
|
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
|
|
"could not set to playing");
|
|
|
|
newseg = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
|
|
fail_unless (gst_pad_push_event (mysrcpad, newseg) != FALSE);
|
|
|
|
inbuffer = gst_buffer_new_and_alloc (9343 * 4);
|
|
memset (GST_BUFFER_DATA (inbuffer), 0, GST_BUFFER_SIZE (inbuffer));
|
|
GST_BUFFER_DURATION (inbuffer) = GST_SECOND;
|
|
GST_BUFFER_TIMESTAMP (inbuffer) = 0;
|
|
GST_BUFFER_OFFSET (inbuffer) = 0;
|
|
gst_buffer_set_caps (inbuffer, caps);
|
|
|
|
fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
|
|
|
|
/* ... it also ends up being collected on the global buffer list. If we
|
|
* now have more than 2 buffers, then audioresample probably didn't clean
|
|
* up its internal buffer properly and tried to push the remaining samples
|
|
* when it got the second NEWSEGMENT event */
|
|
fail_unless_equals_int (g_list_length (buffers), 2);
|
|
|
|
cleanup_audioresample (audioresample);
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
GST_END_TEST;
|
|
|
|
Suite *
|
|
audioresample_suite (void)
|
|
{
|
|
Suite *s = suite_create ("audioresample");
|
|
TCase *tc_chain = tcase_create ("general");
|
|
|
|
suite_add_tcase (s, tc_chain);
|
|
tcase_add_test (tc_chain, test_perfect_stream);
|
|
tcase_add_test (tc_chain, test_discont_stream);
|
|
tcase_add_test (tc_chain, test_reuse);
|
|
|
|
return s;
|
|
}
|
|
|
|
int
|
|
main (int argc, char **argv)
|
|
{
|
|
int nf;
|
|
|
|
Suite *s = audioresample_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;
|
|
}
|