gstreamer/subprojects/gst-plugins-base/tests/check/elements/audiotestsrc.c

313 lines
9.7 KiB
C

/* GStreamer
*
* unit test for audiotestsrc
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>
#include <gst/audio/audio.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 *mysinkpad;
#define CAPS_TEMPLATE_STRING \
"audio/x-raw, " \
"format = (string) "GST_AUDIO_NE(S16)", " \
"channels = (int) 1, " \
"rate = (int) [ 1, MAX ]"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CAPS_TEMPLATE_STRING)
);
static GstElement *
setup_audiotestsrc (void)
{
GstElement *audiotestsrc;
GST_DEBUG ("setup_audiotestsrc");
audiotestsrc = gst_check_setup_element ("audiotestsrc");
mysinkpad = gst_check_setup_sink_pad (audiotestsrc, &sinktemplate);
gst_pad_set_active (mysinkpad, TRUE);
return audiotestsrc;
}
static void
cleanup_audiotestsrc (GstElement * audiotestsrc)
{
GST_DEBUG ("cleanup_audiotestsrc");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_sink_pad (audiotestsrc);
gst_check_teardown_element (audiotestsrc);
}
GST_START_TEST (test_all_waves)
{
GstElement *audiotestsrc;
GObjectClass *oclass;
GParamSpec *property;
GEnumValue *values;
guint j = 0;
audiotestsrc = setup_audiotestsrc ();
oclass = G_OBJECT_GET_CLASS (audiotestsrc);
property = g_object_class_find_property (oclass, "wave");
fail_unless (G_IS_PARAM_SPEC_ENUM (property));
values = G_ENUM_CLASS (g_type_class_ref (property->value_type))->values;
while (values[j].value_name) {
GST_DEBUG_OBJECT (audiotestsrc, "testing wave %s", values[j].value_name);
g_object_set (audiotestsrc, "wave", values[j].value, NULL);
fail_unless (gst_element_set_state (audiotestsrc,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"could not set to playing");
g_mutex_lock (&check_mutex);
while (g_list_length (buffers) < 10)
g_cond_wait (&check_cond, &check_mutex);
g_mutex_unlock (&check_mutex);
gst_element_set_state (audiotestsrc, GST_STATE_READY);
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
++j;
}
/* cleanup */
cleanup_audiotestsrc (audiotestsrc);
}
GST_END_TEST;
#define TEST_LAYOUT_CHANNELS 6
static GstStaticPadTemplate sinktemplate_interleaved =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"channels = (int) " G_STRINGIFY (TEST_LAYOUT_CHANNELS) ", "
"rate = (int) [ 1, MAX ], layout = (string) interleaved")
);
static GstStaticPadTemplate sinktemplate_planar =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"channels = (int) " G_STRINGIFY (TEST_LAYOUT_CHANNELS) ", "
"rate = (int) [ 1, MAX ], layout = (string) non-interleaved")
);
typedef enum
{
GST_AUDIO_TEST_SRC_WAVE_SINE,
GST_AUDIO_TEST_SRC_WAVE_SQUARE,
GST_AUDIO_TEST_SRC_WAVE_SAW,
GST_AUDIO_TEST_SRC_WAVE_TRIANGLE,
GST_AUDIO_TEST_SRC_WAVE_SILENCE,
GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE,
GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE,
GST_AUDIO_TEST_SRC_WAVE_SINE_TAB,
GST_AUDIO_TEST_SRC_WAVE_TICKS,
GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE,
GST_AUDIO_TEST_SRC_WAVE_RED_NOISE,
GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE,
GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE,
_GST_AUDIO_TEST_SRC_WAVE_LAST
} GstAudioTestSrcWave;
GST_START_TEST (test_layout)
{
GstHarness *interleavedsrc, *plannarsrc;
GObjectClass *oclass;
GParamSpec *property;
GEnumValue *values;
guint i, j;
interleavedsrc = gst_harness_new_with_templates ("audiotestsrc", NULL,
&sinktemplate_interleaved);
plannarsrc = gst_harness_new_with_templates ("audiotestsrc", NULL,
&sinktemplate_planar);
gst_harness_use_testclock (interleavedsrc);
gst_harness_use_testclock (plannarsrc);
g_object_set (interleavedsrc->element, "is-live", TRUE, NULL);
g_object_set (plannarsrc->element, "is-live", TRUE, NULL);
oclass = G_OBJECT_GET_CLASS (interleavedsrc->element);
property = g_object_class_find_property (oclass, "wave");
fail_unless (G_IS_PARAM_SPEC_ENUM (property));
values = G_ENUM_CLASS (g_type_class_ref (property->value_type))->values;
for (j = 0; values[j].value_name; j++) {
/* these produce random values by definition,
* so we can't compare channels */
switch (j) {
case GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE:
case GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE:
case GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE:
case GST_AUDIO_TEST_SRC_WAVE_RED_NOISE:
case GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE:
case GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE:
continue;
default:
break;
}
GST_DEBUG ("layout test with wave %s", values[j].value_name);
g_object_set (interleavedsrc->element, "wave", values[j].value, NULL);
g_object_set (plannarsrc->element, "wave", values[j].value, NULL);
if (j == 0) {
GST_DEBUG ("gst_harness_play");
gst_harness_play (interleavedsrc);
gst_harness_play (plannarsrc);
} else {
GST_DEBUG ("discarding buffers with old wave");
fail_unless (gst_harness_crank_single_clock_wait (interleavedsrc));
fail_unless (gst_harness_crank_single_clock_wait (plannarsrc));
gst_buffer_unref (gst_harness_pull (interleavedsrc));
gst_buffer_unref (gst_harness_pull (plannarsrc));
}
for (i = 0; i < 10; i++) {
GstBuffer *ibuf, *pbuf;
GstMapInfo imap, pmap;
GstAudioMeta *meta;
GstAudioBuffer pabuf;
gint16 *iptr, *pptr;
guint isamples, psamples, s, c;
GST_DEBUG ("waiting on clock");
fail_unless (gst_harness_crank_single_clock_wait (interleavedsrc));
fail_unless (gst_harness_crank_single_clock_wait (plannarsrc));
ibuf = gst_harness_pull (interleavedsrc);
pbuf = gst_harness_pull (plannarsrc);
gst_buffer_map (ibuf, &imap, GST_MAP_READ);
gst_buffer_map (pbuf, &pmap, GST_MAP_READ);
/* buffers should have the same size in bytes and in samples */
fail_unless_equals_int (imap.size, pmap.size);
isamples = imap.size / TEST_LAYOUT_CHANNELS;
isamples /= 2; /* S16 -> 2 bytes per sample */
fail_unless_equals_int (imap.size % TEST_LAYOUT_CHANNELS, 0);
psamples = pmap.size / TEST_LAYOUT_CHANNELS;
psamples /= 2; /* S16 -> 2 bytes per sample */
fail_unless_equals_int (pmap.size % TEST_LAYOUT_CHANNELS, 0);
fail_unless_equals_int (isamples, psamples);
iptr = (gint16 *) imap.data;
pptr = (gint16 *) pmap.data;
GST_DEBUG ("verifying contents of buffers; samples=%d, channels=%d",
isamples, TEST_LAYOUT_CHANNELS);
for (s = 0; s < isamples; s++) {
for (c = 0; c < TEST_LAYOUT_CHANNELS; c++) {
guint iidx = s * TEST_LAYOUT_CHANNELS + c;
guint pidx = c * isamples + s;
GST_TRACE ("s = %u | c = %u | iidx (s * channels + c) = %u | "
"pidx (c * samples + s) = %u", s, c, iidx, pidx);
fail_unless (iidx < imap.size / 2);
fail_unless (pidx < pmap.size / 2);
fail_unless_equals_int (iptr[iidx], pptr[pidx]);
}
}
gst_buffer_unmap (pbuf, &pmap);
GST_DEBUG ("verify that mapping through GstAudioBuffer works the same");
meta = gst_buffer_get_audio_meta (pbuf);
fail_unless (meta);
gst_audio_buffer_map (&pabuf, &meta->info, pbuf, GST_MAP_READ);
for (s = 0; s < isamples; s++) {
for (c = 0; c < TEST_LAYOUT_CHANNELS; c++) {
guint iidx = s * TEST_LAYOUT_CHANNELS + c;
fail_unless_equals_int (iptr[iidx], ((gint16 *) pabuf.planes[c])[s]);
}
}
gst_audio_buffer_unmap (&pabuf);
gst_buffer_unmap (ibuf, &imap);
gst_buffer_unref (ibuf);
gst_buffer_unref (pbuf);
}
/* ensure the audiotestsrcs are not in fill() while we change the wave */
fail_unless (gst_harness_wait_for_clock_id_waits (interleavedsrc, 1, 1));
fail_unless (gst_harness_wait_for_clock_id_waits (plannarsrc, 1, 1));
}
/* make sure we ran the test */
fail_unless_equals_int (j, _GST_AUDIO_TEST_SRC_WAVE_LAST);
gst_harness_teardown (interleavedsrc);
gst_harness_teardown (plannarsrc);
}
GST_END_TEST;
static Suite *
audiotestsrc_suite (void)
{
Suite *s = suite_create ("audiotestsrc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_all_waves);
tcase_add_test (tc_chain, test_layout);
return s;
}
GST_CHECK_MAIN (audiotestsrc);