Commit 543f01eb authored by Tim-Philipp Müller's avatar Tim-Philipp Müller
Browse files

tests/check/: Add some basic unit tests for audiorate. Disabled at the moment...

tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...

Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiorate.c: (probe_cb), (got_buf),
(do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
Add some basic unit tests for audiorate. Disabled at the moment
since it doesn't pass yet (see bug #363119).
parent ad087e01
2006-10-21 Tim-Philipp Müller <tim at centricular dot net>
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiorate.c: (probe_cb), (got_buf),
(do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
Add some basic unit tests for audiorate. Disabled at the moment
since it doesn't pass yet (see bug #363119).
2006-10-20 Tim-Philipp Müller <tim at centricular dot net>
 
* gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
common @ ee0bb43e
Subproject commit efcacf2625da231fbee99b68e0f5db6816cf6fad
Subproject commit ee0bb43e2b66781d04078e2210404da48f6c68f0
......@@ -81,6 +81,7 @@ VALGRIND_TO_FIX = \
# these tests don't even pass
noinst_PROGRAMS = \
elements/audiorate \
elements/ffmpegcolorspace
AM_CFLAGS = $(GST_CFLAGS) $(GST_CHECK_CFLAGS)
......@@ -111,6 +112,9 @@ elements_audioconvert_LDADD = \
$(top_builddir)/gst-libs/gst/audio/libgstaudio-@GST_MAJORMINOR@.la \
$(LDADD)
elements_audiorate_LDADD = $(LDADD)
elements_audiorate_CFLAGS = $(CFLAGS) $(AM_CFLAGS)
elements_gdpdepay_LDADD = $(GST_GDP_LIBS) $(LDADD)
elements_gdppay_LDADD = $(GST_GDP_LIBS) $(LDADD)
......
......@@ -2,6 +2,7 @@
adder
alsa
audioconvert
audiorate
audioresample
audiotestsrc
gdpdepay
......
/* GStreamer unit tests for audiorate
*
* Copyright (C) 2006 Tim-Philipp Müller <tim 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.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <gst/check/gstcheck.h>
static gboolean
probe_cb (GstPad * pad, GstBuffer * buf, gdouble * drop_probability)
{
if (g_random_double () < *drop_probability) {
GST_LOG ("dropping buffer");
return FALSE; /* drop buffer */
}
return TRUE; /* don't drop buffer */
}
static void
got_buf (GstElement * fakesink, GstBuffer * buf, GstPad * pad, GList ** p_bufs)
{
*p_bufs = g_list_append (*p_bufs, gst_buffer_ref (buf));
}
static void
do_perfect_stream_test (guint rate, guint width, gdouble drop_probability)
{
GstElement *pipe, *src, *conv, *filter, *audiorate, *sink;
GstMessage *msg;
GstCaps *caps;
GstPad *srcpad;
GList *l, *bufs = NULL;
GstClockTime next_time = GST_CLOCK_TIME_NONE;
gint64 next_offset = GST_BUFFER_OFFSET_NONE;
caps = gst_caps_new_simple ("audio/x-raw-int", "rate", G_TYPE_INT,
rate, "width", G_TYPE_INT, width, NULL);
GST_INFO ("-------- drop=%.0f%% caps = %" GST_PTR_FORMAT " ---------- ",
drop_probability * 100.0, caps);
g_assert (drop_probability >= 0.0 && drop_probability <= 1.0);
g_assert (width > 0 && (width % 8) == 0);
pipe = gst_pipeline_new ("pipeline");
fail_unless (pipe != NULL);
src = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
fail_unless (src != NULL);
g_object_set (src, "num-buffers", 500, NULL);
conv = gst_element_factory_make ("audioconvert", "audioconvert");
fail_unless (conv != NULL);
filter = gst_element_factory_make ("capsfilter", "capsfilter");
fail_unless (filter != NULL);
g_object_set (filter, "caps", caps, NULL);
srcpad = gst_element_get_pad (filter, "src");
fail_unless (srcpad != NULL);
gst_pad_add_buffer_probe (srcpad, G_CALLBACK (probe_cb), &drop_probability);
gst_object_unref (srcpad);
audiorate = gst_element_factory_make ("audiorate", "audiorate");
fail_unless (audiorate != NULL);
sink = gst_element_factory_make ("fakesink", "fakesink");
fail_unless (sink != NULL);
g_object_set (sink, "signal-handoffs", TRUE, NULL);
g_signal_connect (sink, "handoff", G_CALLBACK (got_buf), &bufs);
gst_bin_add_many (GST_BIN (pipe), src, conv, filter, audiorate, sink, NULL);
gst_element_link_many (src, conv, filter, audiorate, sink, NULL);
fail_unless_equals_int (gst_element_set_state (pipe, GST_STATE_PLAYING),
GST_STATE_CHANGE_ASYNC);
fail_unless_equals_int (gst_element_get_state (pipe, NULL, NULL, -1),
GST_STATE_CHANGE_SUCCESS);
msg = gst_bus_poll (GST_ELEMENT_BUS (pipe),
GST_MESSAGE_EOS | GST_MESSAGE_ERROR, -1);
fail_unless_equals_string (GST_MESSAGE_TYPE_NAME (msg), "eos");
for (l = bufs; l != NULL; l = l->next) {
GstBuffer *buf = GST_BUFFER (l->data);
guint num_samples;
fail_unless (GST_BUFFER_TIMESTAMP_IS_VALID (buf));
fail_unless (GST_BUFFER_DURATION_IS_VALID (buf));
fail_unless (GST_BUFFER_OFFSET_IS_VALID (buf));
fail_unless (GST_BUFFER_OFFSET_END_IS_VALID (buf));
GST_LOG ("buffer: ts=%" GST_TIME_FORMAT ", end_ts=%" GST_TIME_FORMAT
" off=%" G_GINT64_FORMAT ", end_off=%" G_GINT64_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf)),
GST_BUFFER_OFFSET (buf), GST_BUFFER_OFFSET_END (buf));
if (GST_CLOCK_TIME_IS_VALID (next_time)) {
fail_unless_equals_uint64 (next_time, GST_BUFFER_TIMESTAMP (buf));
}
if (next_offset != GST_BUFFER_OFFSET_NONE) {
fail_unless_equals_uint64 (next_offset, GST_BUFFER_OFFSET (buf));
}
/* check buffer size for sanity */
fail_unless_equals_int (GST_BUFFER_SIZE (buf) % (width / 8), 0);
/* check there is actually as much data as there should be */
num_samples = GST_BUFFER_OFFSET_END (buf) - GST_BUFFER_OFFSET (buf);
fail_unless_equals_int (GST_BUFFER_SIZE (buf), num_samples * (width / 8));
next_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
next_offset = GST_BUFFER_OFFSET_END (buf);
}
gst_message_unref (msg);
gst_element_set_state (pipe, GST_STATE_NULL);
gst_object_unref (pipe);
g_list_foreach (bufs, (GFunc) gst_mini_object_unref, NULL);
g_list_free (bufs);
gst_caps_unref (caps);
}
static const guint rates[] = { 8000, 11025, 16000, 22050, 32000, 44100,
48000, 3333, 33333, 66666, 9999
};
GST_START_TEST (test_perfect_stream_drop0)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
do_perfect_stream_test (rates[i], 8, 0.0);
do_perfect_stream_test (rates[i], 16, 0.0);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_drop10)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
do_perfect_stream_test (rates[i], 8, 0.10);
do_perfect_stream_test (rates[i], 16, 0.10);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_drop50)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
do_perfect_stream_test (rates[i], 8, 0.50);
do_perfect_stream_test (rates[i], 16, 0.50);
}
}
GST_END_TEST;
GST_START_TEST (test_perfect_stream_drop90)
{
guint i;
for (i = 0; i < G_N_ELEMENTS (rates); ++i) {
do_perfect_stream_test (rates[i], 8, 0.90);
do_perfect_stream_test (rates[i], 16, 0.90);
}
}
GST_END_TEST;
static Suite *
audiorate_suite (void)
{
Suite *s = suite_create ("audiorate");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_perfect_stream_drop0);
tcase_add_test (tc_chain, test_perfect_stream_drop10);
tcase_add_test (tc_chain, test_perfect_stream_drop50);
tcase_add_test (tc_chain, test_perfect_stream_drop90);
return s;
}
GST_CHECK_MAIN (audiorate);
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