gstreamer/subprojects/gst-plugins-bad/tests/icles/pitch-test.c

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/* GStreamer
*
* Copyright (C) 2006 Thomas Vander Stichele <thomas at apestaart dot org>
*
* 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.
*/
/* compile with :
* gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10 gstreamer-controller-0.10) pitch.c -o pitch
*/
#include <string.h>
#include <unistd.h>
#include <gst/gst.h>
#include <gst/controller/gsttimedvaluecontrolsource.h>
#include <gst/controller/gstinterpolationcontrolsource.h>
#include <gst/controller/gstdirectcontrolbinding.h>
int
main (int argc, char **argv)
{
GMainLoop *loop;
gint i;
GstElement *audiotestsrc;
GstElement *audioconvert1, *audioconvert2;
GstElement *pitch;
GstElement *sink;
GstElement *pipeline;
GstControlSource *cs;
GstTimedValueControlSource *tvcs;
if (argc != 2) {
g_printerr ("Usage: %s <audiosink>\n", argv[0]);
return 1;
}
/* initialize GStreamer */
gst_init (&argc, &argv);
loop = g_main_loop_new (NULL, FALSE);
pipeline = gst_pipeline_new ("audio-player");
audiotestsrc = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
g_assert (audiotestsrc != NULL);
audioconvert1 = gst_element_factory_make ("audioconvert", "audioconvert1");
g_assert (audioconvert1 != NULL);
audioconvert2 = gst_element_factory_make ("audioconvert", "audioconvert2");
g_assert (audioconvert2 != NULL);
pitch = gst_element_factory_make ("pitch", "pitch");
g_assert (pitch != NULL);
sink = gst_element_factory_make (argv[1], "sink");
g_assert (sink != NULL);
gst_bin_add_many (GST_BIN (pipeline),
audiotestsrc, audioconvert1, pitch, audioconvert2, sink, NULL);
gst_element_link_many (audiotestsrc, audioconvert1, pitch, audioconvert2,
sink, NULL);
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/* set up a controller */
cs = gst_interpolation_control_source_new ();
g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
gst_object_add_control_binding (GST_OBJECT (pitch),
gst_direct_control_binding_new (GST_OBJECT (pitch), "pitch", cs));
tvcs = (GstTimedValueControlSource *) cs;
for (i = 0; i < 100; ++i) {
if (i % 2)
gst_timed_value_control_source_set (tvcs, i * GST_SECOND, 0.5);
else
gst_timed_value_control_source_set (tvcs, i * GST_SECOND, 1.5);
}
gst_element_set_state (pipeline, GST_STATE_PLAYING);
g_print ("Running\n");
g_main_loop_run (loop);
/* clean up nicely */
gst_object_unref (cs);
g_print ("Returned, stopping playback\n");
gst_element_set_state (pipeline, GST_STATE_NULL);
g_print ("Deleting pipeline\n");
gst_object_unref (GST_OBJECT (pipeline));
return 0;
}