audiovisualizer: Add simple pipeline unit test

Creates pipelines with each of our visualizer elements and runs them with 20 buffers from audiotestsrc.
Added after a completely broken (segfaulting) synaescope went unnoticed for a while.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6800>
This commit is contained in:
Piotr Brzeziński 2024-05-07 11:18:10 +02:00 committed by GStreamer Marge Bot
parent 5d7d3c6c0f
commit a9378c048e
2 changed files with 107 additions and 0 deletions

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@ -0,0 +1,106 @@
/* GStreamer
*
* Copyright (C) 2024 Piotr BrzeziÅski <piotr@centricular.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
#include <gst/check/gstcheck.h>
static void
eos_cb (GstBus * bus, GstMessage * message, GMainLoop * loop)
{
GST_DEBUG ("Received EOS");
g_main_loop_quit (loop);
}
static void
error_cb (GstBus * bus, GstMessage * message, GMainLoop * loop)
{
GError *err = NULL;
gchar *dbg = NULL;
gst_message_parse_error (message, &err, &dbg);
g_error ("ERROR: %s\n%s\n", err->message, dbg);
}
static void
test_element (const gchar * element)
{
GstElement *pipeline;
GstBus *bus;
GError *error = NULL;
gchar *pipe_str;
GMainLoop *loop;
pipe_str =
g_strdup_printf
("audiotestsrc num-buffers=20 ! audio/x-raw,format=S16LE,channels=2 ! %s ! fakesink",
element);
pipeline = gst_parse_launch (pipe_str, &error);
fail_unless (pipeline != NULL, "Could not create pipeline: %s",
error->message);
g_free (pipe_str);
loop = g_main_loop_new (NULL, FALSE);
bus = gst_element_get_bus (pipeline);
fail_if (bus == NULL);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message::eos", (GCallback) eos_cb, loop);
g_signal_connect (bus, "message::error", (GCallback) error_cb, loop);
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_main_loop_run (loop);
gst_element_set_state (pipeline, GST_STATE_NULL);
g_main_loop_unref (loop);
gst_object_unref (bus);
gst_object_unref (pipeline);
}
GST_START_TEST (test_simple_pipelines)
{
/* Simple pipeline tests to see if these elements run at all.
* Will help catch breakages like https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6800. */
test_element ("wavescope");
test_element ("spacescope");
test_element ("spectrascope");
test_element ("synaescope");
}
GST_END_TEST;
static Suite *
audiovisualizer_suite (void)
{
Suite *s = suite_create ("audiovisualizer");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
#ifndef GST_DISABLE_PARSE
tcase_add_test (tc_chain, test_simple_pipelines);
#endif
return s;
}
GST_CHECK_MAIN (audiovisualizer);

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@ -25,6 +25,7 @@ base_tests = [
[['elements/aesdec.c'], not aes_dep.found(), [aes_dep]],
[['elements/aiffparse.c'], get_option('aiff').disabled()],
[['elements/asfmux.c'], get_option('asfmux').disabled()],
[['elements/audiovisualizer.c'], get_option('audiovisualizers').disabled()],
[['elements/autoconvert.c'], get_option('autoconvert').disabled()],
[['elements/autovideoconvert.c'], get_option('autoconvert').disabled()],
[['elements/avwait.c'], get_option('timecode').disabled()],