gstreamer/tests/check/elements/videoframe-audiolevel.c

637 lines
17 KiB
C
Raw Normal View History

/* GStreamer unit test for videoframe-audiolevel
*
* Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
*
* 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
/* suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
static gboolean got_eos;
static guint audio_buffer_count, video_buffer_count;
static GstSegment current_audio_segment, current_video_segment;
static guint num_msgs;
static GQueue v_timestamp_q, msg_timestamp_q;
static guint n_abuffers, n_vbuffers;
static guint channels, fill_value;
static gdouble expected_rms;
static gboolean audiodelay, videodelay, per_channel, long_video;
static gboolean early_video, late_video;
static gboolean video_gaps, video_overlaps;
static gboolean audio_nondiscont, audio_drift;
static guint fill_value_per_channel[] = { 0, 1 };
static gdouble expected_rms_per_channel[] = { 0, 0.0078125 };
static void
set_default_params (void)
{
n_abuffers = 40;
n_vbuffers = 15;
channels = 2;
expected_rms = 0.0078125;
fill_value = 1;
audiodelay = FALSE;
videodelay = FALSE;
per_channel = FALSE;
long_video = FALSE;
video_gaps = FALSE;
video_overlaps = FALSE;
audio_nondiscont = FALSE;
audio_drift = FALSE;
early_video = FALSE;
late_video = FALSE;
};
static GstFlowReturn
output_achain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstClockTime timestamp;
guint8 b;
gboolean audio_jitter = audio_nondiscont || audio_drift || early_video;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp,
(audio_buffer_count % n_abuffers) * 1 * GST_SECOND);
timestamp =
gst_segment_to_stream_time (&current_audio_segment, GST_FORMAT_TIME,
timestamp);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp,
(audio_buffer_count % n_abuffers) * 1 * GST_SECOND);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
timestamp =
gst_segment_to_running_time (&current_audio_segment, GST_FORMAT_TIME,
timestamp);
if (!audio_jitter)
fail_unless_equals_int64 (timestamp, audio_buffer_count * 1 * GST_SECOND);
gst_buffer_extract (buffer, 0, &b, 1);
if (per_channel) {
fail_unless_equals_int (b, fill_value_per_channel[0]);
} else {
fail_unless_equals_int (b, fill_value);
}
audio_buffer_count++;
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static gboolean
output_aevent (GstPad * pad, GstObject * parent, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&current_audio_segment, GST_FORMAT_UNDEFINED);
break;
case GST_EVENT_SEGMENT:
gst_event_copy_segment (event, &current_audio_segment);
break;
case GST_EVENT_EOS:
got_eos = TRUE;
break;
default:
break;
}
gst_event_unref (event);
return TRUE;
}
static GstFlowReturn
output_vchain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
GstClockTime timestamp;
guint8 b;
gboolean jitter = video_gaps || video_overlaps || late_video;
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (!jitter)
fail_unless_equals_int64 (timestamp,
(video_buffer_count % n_vbuffers) * 25 * GST_MSECOND);
timestamp =
gst_segment_to_stream_time (&current_video_segment, GST_FORMAT_TIME,
timestamp);
if (!jitter)
fail_unless_equals_int64 (timestamp,
(video_buffer_count % n_vbuffers) * 25 * GST_MSECOND);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
timestamp =
gst_segment_to_running_time (&current_video_segment, GST_FORMAT_TIME,
timestamp);
if (!jitter)
fail_unless_equals_int64 (timestamp, video_buffer_count * 25 * GST_MSECOND);
gst_buffer_extract (buffer, 0, &b, 1);
if (!jitter)
fail_unless_equals_int (b, video_buffer_count % n_vbuffers);
video_buffer_count++;
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static gboolean
output_vevent (GstPad * pad, GstObject * parent, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&current_video_segment, GST_FORMAT_UNDEFINED);
break;
case GST_EVENT_SEGMENT:
gst_event_copy_segment (event, &current_video_segment);
break;
case GST_EVENT_EOS:
got_eos = TRUE;
break;
default:
break;
}
gst_event_unref (event);
return TRUE;
}
static gpointer
push_abuffers (gpointer data)
{
GstSegment segment;
GstPad *pad = data;
gint i, j, k;
GstClockTime timestamp = 0;
GstAudioInfo info;
GstCaps *caps;
guint buf_size = 1000;
if (audiodelay)
g_usleep (2000);
if (early_video)
timestamp = 50 * GST_MSECOND;
gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S8, buf_size, channels,
NULL);
caps = gst_audio_info_to_caps (&info);
gst_pad_send_event (pad, gst_event_new_caps (caps));
gst_caps_unref (caps);
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_send_event (pad, gst_event_new_segment (&segment));
for (i = 0; i < n_abuffers; i++) {
GstBuffer *buf = gst_buffer_new_and_alloc (channels * buf_size);
if (per_channel) {
GstMapInfo map;
guint8 *in_data;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
in_data = map.data;
for (j = 0; j < buf_size; j++) {
for (k = 0; k < channels; k++) {
in_data[j * channels + k] = fill_value_per_channel[k];
}
}
gst_buffer_unmap (buf, &map);
} else {
gst_buffer_memset (buf, 0, fill_value, channels * buf_size);
}
GST_BUFFER_TIMESTAMP (buf) = timestamp;
timestamp += 1 * GST_SECOND;
if (audio_drift)
timestamp += 50 * GST_MSECOND;
else if (i == 4 && audio_nondiscont)
timestamp += 30 * GST_MSECOND;
GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
}
gst_pad_send_event (pad, gst_event_new_eos ());
return NULL;
}
static gpointer
push_vbuffers (gpointer data)
{
GstSegment segment;
GstPad *pad = data;
gint i;
GstClockTime timestamp = 0;
if (videodelay)
g_usleep (2000);
if (late_video)
timestamp = 50 * GST_MSECOND;
gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
gst_segment_init (&segment, GST_FORMAT_TIME);
gst_pad_send_event (pad, gst_event_new_segment (&segment));
for (i = 0; i < n_vbuffers; i++) {
GstBuffer *buf = gst_buffer_new_and_alloc (1000);
GstClockTime *rtime = g_new (GstClockTime, 1);
gst_buffer_memset (buf, 0, i, 1);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
timestamp += 25 * GST_MSECOND;
GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
*rtime = gst_segment_to_running_time (&segment, GST_FORMAT_TIME, timestamp);
g_queue_push_tail (&v_timestamp_q, rtime);
if (i == 4) {
if (video_gaps)
timestamp += 10 * GST_MSECOND;
else if (video_overlaps)
timestamp -= 10 * GST_MSECOND;
}
fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
}
gst_pad_send_event (pad, gst_event_new_eos ());
return NULL;
}
static GstBusSyncReply
on_message (GstBus * bus, GstMessage * message, gpointer user_data)
{
const GstStructure *s = gst_message_get_structure (message);
const gchar *name = gst_structure_get_name (s);
GValueArray *rms_arr;
const GValue *array_val;
const GValue *value;
gdouble rms;
gint channels2;
guint i;
GstClockTime *rtime = g_new (GstClockTime, 1);
if (message->type != GST_MESSAGE_ELEMENT
|| strcmp (name, "videoframe-audiolevel") != 0)
goto done;
num_msgs++;
if (!gst_structure_get_clock_time (s, "running-time", rtime))
g_warning ("Could not parse running time");
else
g_queue_push_tail (&msg_timestamp_q, rtime);
/* the values are packed into GValueArrays with the value per channel */
array_val = gst_structure_get_value (s, "rms");
rms_arr = (GValueArray *) g_value_get_boxed (array_val);
channels2 = rms_arr->n_values;
fail_unless_equals_int (channels2, channels);
for (i = 0; i < channels; ++i) {
value = g_value_array_get_nth (rms_arr, i);
rms = g_value_get_double (value);
if (per_channel) {
fail_unless_equals_float (rms, expected_rms_per_channel[i]);
} else if (early_video && *rtime <= 50 * GST_MSECOND) {
fail_unless_equals_float (rms, 0);
} else {
fail_unless_equals_float (rms, expected_rms);
}
}
done:
return GST_BUS_PASS;
}
static void
test_videoframe_audiolevel_generic (void)
{
GstElement *alevel;
GstPad *asink, *vsink, *asrc, *vsrc, *aoutput_sink, *voutput_sink;
GThread *athread, *vthread;
GstBus *bus;
guint i;
got_eos = FALSE;
audio_buffer_count = 0;
video_buffer_count = 0;
num_msgs = 0;
g_queue_init (&v_timestamp_q);
g_queue_init (&msg_timestamp_q);
alevel = gst_element_factory_make ("videoframe-audiolevel", NULL);
fail_unless (alevel != NULL);
bus = gst_bus_new ();
gst_element_set_bus (alevel, bus);
gst_bus_set_sync_handler (bus, on_message, NULL, NULL);
asink = gst_element_get_static_pad (alevel, "asink");
fail_unless (asink != NULL);
vsink = gst_element_get_static_pad (alevel, "vsink");
fail_unless (vsink != NULL);
asrc = gst_element_get_static_pad (alevel, "asrc");
aoutput_sink = gst_pad_new ("sink", GST_PAD_SINK);
fail_unless (aoutput_sink != NULL);
fail_unless (gst_pad_link (asrc, aoutput_sink) == GST_PAD_LINK_OK);
vsrc = gst_element_get_static_pad (alevel, "vsrc");
voutput_sink = gst_pad_new ("sink", GST_PAD_SINK);
fail_unless (voutput_sink != NULL);
fail_unless (gst_pad_link (vsrc, voutput_sink) == GST_PAD_LINK_OK);
gst_pad_set_chain_function (aoutput_sink, output_achain);
gst_pad_set_event_function (aoutput_sink, output_aevent);
gst_pad_set_chain_function (voutput_sink, output_vchain);
gst_pad_set_event_function (voutput_sink, output_vevent);
gst_pad_set_active (aoutput_sink, TRUE);
gst_pad_set_active (voutput_sink, TRUE);
fail_unless (gst_element_set_state (alevel,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
athread = g_thread_new ("athread", (GThreadFunc) push_abuffers, asink);
vthread = g_thread_new ("vthread", (GThreadFunc) push_vbuffers, vsink);
g_thread_join (vthread);
g_thread_join (athread);
fail_unless (got_eos);
fail_unless_equals_int (audio_buffer_count, n_abuffers);
fail_unless_equals_int (video_buffer_count, n_vbuffers);
if (!long_video)
fail_unless_equals_int (num_msgs, n_vbuffers);
fail_unless_equals_int (g_queue_get_length (&v_timestamp_q), n_vbuffers);
/* num_msgs is equal to n_vbuffers except in the case of long_video */
fail_unless_equals_int (g_queue_get_length (&msg_timestamp_q), num_msgs);
for (i = 0; i < g_queue_get_length (&msg_timestamp_q); i++) {
GstClockTime *vt = g_queue_pop_head (&v_timestamp_q);
GstClockTime *mt = g_queue_pop_head (&msg_timestamp_q);
fail_unless (vt != NULL);
fail_unless (mt != NULL);
if (!video_gaps && !video_overlaps && !early_video)
fail_unless_equals_uint64 (*vt, *mt);
g_free (vt);
g_free (mt);
}
/* teardown */
gst_element_set_state (alevel, GST_STATE_NULL);
gst_bus_set_flushing (bus, TRUE);
gst_object_unref (bus);
g_queue_foreach (&v_timestamp_q, (GFunc) g_free, NULL);
g_queue_foreach (&msg_timestamp_q, (GFunc) g_free, NULL);
g_queue_clear (&v_timestamp_q);
g_queue_clear (&msg_timestamp_q);
gst_pad_unlink (asrc, aoutput_sink);
gst_object_unref (asrc);
gst_pad_unlink (vsrc, voutput_sink);
gst_object_unref (vsrc);
gst_object_unref (asink);
gst_object_unref (vsink);
gst_pad_set_active (aoutput_sink, FALSE);
gst_object_unref (aoutput_sink);
gst_pad_set_active (voutput_sink, FALSE);
gst_object_unref (voutput_sink);
gst_object_unref (alevel);
}
GST_START_TEST (test_videoframe_audiolevel_16chan_1)
{
set_default_params ();
channels = 16;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_8chan_1)
{
set_default_params ();
channels = 8;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_2chan_1)
{
set_default_params ();
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_1chan_1)
{
set_default_params ();
channels = 1;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_16chan_0)
{
set_default_params ();
channels = 16;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_8chan_0)
{
set_default_params ();
channels = 8;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_2chan_0)
{
set_default_params ();
channels = 2;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_1chan_0)
{
set_default_params ();
channels = 1;
expected_rms = 0;
fill_value = 0;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_adelay)
{
set_default_params ();
audiodelay = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_vdelay)
{
set_default_params ();
videodelay = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_per_channel)
{
set_default_params ();
per_channel = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_long_video)
{
set_default_params ();
n_abuffers = 6;
n_vbuffers = 255;
long_video = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_video_gaps)
{
set_default_params ();
video_gaps = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_video_overlaps)
{
set_default_params ();
video_overlaps = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_audio_nondiscont)
{
set_default_params ();
audio_nondiscont = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_audio_drift)
{
set_default_params ();
audio_drift = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_early_video)
{
set_default_params ();
early_video = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
GST_START_TEST (test_videoframe_audiolevel_late_video)
{
set_default_params ();
late_video = TRUE;
test_videoframe_audiolevel_generic ();
}
GST_END_TEST;
static Suite *
videoframe_audiolevel_suite (void)
{
Suite *s = suite_create ("videoframe-audiolevel");
TCase *tc_chain;
tc_chain = tcase_create ("videoframe-audiolevel");
tcase_add_test (tc_chain, test_videoframe_audiolevel_16chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_8chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_2chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_1chan_1);
tcase_add_test (tc_chain, test_videoframe_audiolevel_16chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_8chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_2chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_1chan_0);
tcase_add_test (tc_chain, test_videoframe_audiolevel_adelay);
tcase_add_test (tc_chain, test_videoframe_audiolevel_vdelay);
tcase_add_test (tc_chain, test_videoframe_audiolevel_per_channel);
tcase_add_test (tc_chain, test_videoframe_audiolevel_long_video);
tcase_add_test (tc_chain, test_videoframe_audiolevel_video_gaps);
tcase_add_test (tc_chain, test_videoframe_audiolevel_video_overlaps);
tcase_add_test (tc_chain, test_videoframe_audiolevel_audio_nondiscont);
tcase_add_test (tc_chain, test_videoframe_audiolevel_audio_drift);
tcase_add_test (tc_chain, test_videoframe_audiolevel_early_video);
tcase_add_test (tc_chain, test_videoframe_audiolevel_late_video);
suite_add_tcase (s, tc_chain);
return s;
}
GST_CHECK_MAIN (videoframe_audiolevel);