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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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206 lines
5.9 KiB
C
206 lines
5.9 KiB
C
/* GStreamer
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*
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* unit test for wavenc
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*
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* Copyright (C) <2010> Stefan Kost <ensonic@users.sf.net>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/audio-enumtypes.h>
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static gboolean
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bus_handler (GstBus * bus, GstMessage * message, gpointer data)
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{
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GMainLoop *loop = (GMainLoop *) data;
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switch (message->type) {
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case GST_MESSAGE_EOS:
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g_main_loop_quit (loop);
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break;
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case GST_MESSAGE_WARNING:
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case GST_MESSAGE_ERROR:{
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GError *gerror;
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gchar *debug;
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gst_message_parse_error (message, &gerror, &debug);
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gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
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gst_message_unref (message);
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g_error_free (gerror);
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g_free (debug);
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g_main_loop_quit (loop);
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break;
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}
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default:
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break;
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}
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return TRUE;
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}
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/*
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* gst-launch \
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* audiotestsrc freq=440 num-buffers=100 ! interleave name=i ! audioconvert ! wavenc ! filesink location=/tmp/mc.wav \
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* audiotestsrc freq=880 num-buffers=100 ! i.
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* ...
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*
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* http://www.microsoft.com/whdc/device/audio/multichaud.mspx
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*/
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static void
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make_n_channel_wav (const gint channels, const GValueArray * arr)
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{
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GstElement *pipeline;
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GstElement **audiotestsrc, *interleave, *wavenc, *conv, *fakesink;
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GstBus *bus;
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GMainLoop *loop;
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guint i;
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guint bus_watch = 0;
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audiotestsrc = g_new0 (GstElement *, channels);
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pipeline = gst_pipeline_new ("pipeline");
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fail_unless (pipeline != NULL);
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interleave = gst_element_factory_make ("interleave", NULL);
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fail_unless (interleave != NULL);
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g_object_set (interleave, "channel-positions", arr, NULL);
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gst_bin_add (GST_BIN (pipeline), interleave);
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if (G_BYTE_ORDER == G_BIG_ENDIAN) {
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/* we're not here to test orc; audioconvert misbehaves on ppc32 */
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g_setenv ("ORC_CODE", "backup", 1);
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conv = gst_element_factory_make ("audioconvert", NULL);
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} else {
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conv = gst_element_factory_make ("identity", NULL);
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}
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fail_unless (conv != NULL);
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gst_bin_add (GST_BIN (pipeline), conv);
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fail_unless (gst_element_link (interleave, conv));
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wavenc = gst_element_factory_make ("wavenc", NULL);
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fail_unless (wavenc != NULL);
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gst_bin_add (GST_BIN (pipeline), wavenc);
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fail_unless (gst_element_link (conv, wavenc));
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fakesink = gst_element_factory_make ("fakesink", NULL);
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fail_unless (fakesink != NULL);
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gst_bin_add (GST_BIN (pipeline), fakesink);
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fail_unless (gst_element_link (wavenc, fakesink));
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for (i = 0; i < channels; i++) {
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audiotestsrc[i] = gst_element_factory_make ("audiotestsrc", NULL);
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fail_unless (audiotestsrc[i] != NULL);
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g_object_set (G_OBJECT (audiotestsrc[i]), "wave", 0, "freq", 440.0 * i,
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"num-buffers", 100, NULL);
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gst_bin_add (GST_BIN (pipeline), audiotestsrc[i]);
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fail_unless (gst_element_link (audiotestsrc[i], interleave));
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}
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loop = g_main_loop_new (NULL, TRUE);
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fail_unless (loop != NULL);
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bus = gst_element_get_bus (pipeline);
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fail_unless (bus != NULL);
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bus_watch = gst_bus_add_watch (bus, bus_handler, loop);
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gst_object_unref (bus);
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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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gst_object_unref (pipeline);
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g_free (audiotestsrc);
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g_main_loop_unref (loop);
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g_source_remove (bus_watch);
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}
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GST_START_TEST (test_encode_stereo)
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{
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GValueArray *arr;
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GValue val = { 0, };
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arr = g_value_array_new (2);
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g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT);
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g_value_array_append (arr, &val);
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g_value_reset (&val);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT);
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g_value_array_append (arr, &val);
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g_value_unset (&val);
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make_n_channel_wav (2, arr);
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g_value_array_free (arr);
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}
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GST_END_TEST;
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#if 0
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GST_START_TEST (test_encode_multichannel)
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{
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GValueArray *arr;
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GValue val = { 0, };
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arr = g_value_array_new (6);
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g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT);
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g_value_array_append (arr, &val);
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g_value_reset (&val);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT);
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g_value_array_append (arr, &val);
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g_value_reset (&val);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER);
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g_value_array_append (arr, &val);
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g_value_reset (&val);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_LFE);
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g_value_array_append (arr, &val);
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g_value_reset (&val);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_LEFT);
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g_value_array_append (arr, &val);
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g_value_reset (&val);
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g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT);
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g_value_array_append (arr, &val);
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g_value_unset (&val);
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make_n_channel_wav (6);
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}
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GST_END_TEST;
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#endif
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static Suite *
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wavenc_suite (void)
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{
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Suite *s = suite_create ("wavenc");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_encode_stereo);
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/* FIXME: improve wavenc
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tcase_add_test (tc_chain, test_encode_multichannel);
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*/
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return s;
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}
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GST_CHECK_MAIN (wavenc);
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