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>

2 files changed, 107 insertions, 0 deletions

diff --git a/subprojects/gst-plugins-bad/tests/check/elements/audiovisualizer.c b/subprojects/gst-plugins-bad/tests/check/elements/audiovisualizer.c
new file mode 100644
index 0000000000..1da3521584
--- /dev/null
+++ b/subprojects/gst-plugins-bad/tests/check/elements/audiovisualizer.c
@@ -0,0 +1,106 @@
+/* GStreamer
+ *
+ * Copyright (C) 2024 Piotr Brzeziński <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);
diff --git a/subprojects/gst-plugins-bad/tests/check/meson.build b/subprojects/gst-plugins-bad/tests/check/meson.build
index 2605a1b82b..cedfed6be7 100644
--- a/subprojects/gst-plugins-bad/tests/check/meson.build
+++ b/subprojects/gst-plugins-bad/tests/check/meson.build
@@ -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()],