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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>
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/* GStreamer
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*
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* Copyright (C) 2024 Piotr Brzeziński <piotr@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/check/gstcheck.h>
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static void
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eos_cb (GstBus * bus, GstMessage * message, GMainLoop * loop)
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{
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GST_DEBUG ("Received EOS");
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g_main_loop_quit (loop);
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}
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static void
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error_cb (GstBus * bus, GstMessage * message, GMainLoop * loop)
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{
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GError *err = NULL;
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gchar *dbg = NULL;
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gst_message_parse_error (message, &err, &dbg);
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g_error ("ERROR: %s\n%s\n", err->message, dbg);
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}
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static void
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test_element (const gchar * element)
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{
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GstElement *pipeline;
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GstBus *bus;
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GError *error = NULL;
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gchar *pipe_str;
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GMainLoop *loop;
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pipe_str =
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g_strdup_printf
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("audiotestsrc num-buffers=20 ! audio/x-raw,format=S16LE,channels=2 ! %s ! fakesink",
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element);
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pipeline = gst_parse_launch (pipe_str, &error);
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fail_unless (pipeline != NULL, "Could not create pipeline: %s",
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error->message);
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g_free (pipe_str);
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loop = g_main_loop_new (NULL, FALSE);
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bus = gst_element_get_bus (pipeline);
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fail_if (bus == NULL);
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gst_bus_add_signal_watch (bus);
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g_signal_connect (bus, "message::eos", (GCallback) eos_cb, loop);
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g_signal_connect (bus, "message::error", (GCallback) error_cb, loop);
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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g_main_loop_run (loop);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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g_main_loop_unref (loop);
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gst_object_unref (bus);
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gst_object_unref (pipeline);
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}
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GST_START_TEST (test_simple_pipelines)
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{
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/* Simple pipeline tests to see if these elements run at all.
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* Will help catch breakages like https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6800. */
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test_element ("wavescope");
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test_element ("spacescope");
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test_element ("spectrascope");
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test_element ("synaescope");
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}
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GST_END_TEST;
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static Suite *
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audiovisualizer_suite (void)
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{
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Suite *s = suite_create ("audiovisualizer");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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#ifndef GST_DISABLE_PARSE
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tcase_add_test (tc_chain, test_simple_pipelines);
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#endif
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return s;
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}
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GST_CHECK_MAIN (audiovisualizer);
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@ -25,6 +25,7 @@ base_tests = [
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[['elements/aesdec.c'], not aes_dep.found(), [aes_dep]],
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[['elements/aiffparse.c'], get_option('aiff').disabled()],
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[['elements/asfmux.c'], get_option('asfmux').disabled()],
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[['elements/audiovisualizer.c'], get_option('audiovisualizers').disabled()],
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[['elements/autoconvert.c'], get_option('autoconvert').disabled()],
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[['elements/autovideoconvert.c'], get_option('autoconvert').disabled()],
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[['elements/avwait.c'], get_option('timecode').disabled()],
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