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Needed for https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/930 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3064>
102 lines
3.2 KiB
C
102 lines
3.2 KiB
C
/* GStreamer
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*
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* Copyright (C) 2006 Thomas Vander Stichele <thomas at apestaart dot org>
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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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/* compile with :
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* gcc -Wall $(pkg-config --cflags --libs gstreamer-0.10 gstreamer-controller-0.10) pitch.c -o pitch
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*/
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#include <string.h>
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#ifndef _MSC_VER
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#include <unistd.h>
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#endif
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#include <gst/gst.h>
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#include <gst/controller/gsttimedvaluecontrolsource.h>
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#include <gst/controller/gstinterpolationcontrolsource.h>
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#include <gst/controller/gstdirectcontrolbinding.h>
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int
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main (int argc, char **argv)
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{
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GMainLoop *loop;
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gint i;
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GstElement *audiotestsrc;
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GstElement *audioconvert1, *audioconvert2;
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GstElement *pitch;
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GstElement *sink;
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GstElement *pipeline;
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GstControlSource *cs;
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GstTimedValueControlSource *tvcs;
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if (argc != 2) {
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g_printerr ("Usage: %s <audiosink>\n", argv[0]);
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return 1;
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}
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/* initialize GStreamer */
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gst_init (&argc, &argv);
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loop = g_main_loop_new (NULL, FALSE);
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pipeline = gst_pipeline_new ("audio-player");
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audiotestsrc = gst_element_factory_make ("audiotestsrc", "audiotestsrc");
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g_assert (audiotestsrc != NULL);
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audioconvert1 = gst_element_factory_make ("audioconvert", "audioconvert1");
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g_assert (audioconvert1 != NULL);
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audioconvert2 = gst_element_factory_make ("audioconvert", "audioconvert2");
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g_assert (audioconvert2 != NULL);
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pitch = gst_element_factory_make ("pitch", "pitch");
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g_assert (pitch != NULL);
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sink = gst_element_factory_make (argv[1], "sink");
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g_assert (sink != NULL);
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gst_bin_add_many (GST_BIN (pipeline),
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audiotestsrc, audioconvert1, pitch, audioconvert2, sink, NULL);
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gst_element_link_many (audiotestsrc, audioconvert1, pitch, audioconvert2,
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sink, NULL);
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/* set up a controller */
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cs = gst_interpolation_control_source_new ();
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g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
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gst_object_add_control_binding (GST_OBJECT (pitch),
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gst_direct_control_binding_new (GST_OBJECT (pitch), "pitch", cs));
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tvcs = (GstTimedValueControlSource *) cs;
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for (i = 0; i < 100; ++i) {
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if (i % 2)
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gst_timed_value_control_source_set (tvcs, i * GST_SECOND, 0.5);
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else
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gst_timed_value_control_source_set (tvcs, i * GST_SECOND, 1.5);
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}
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gst_element_set_state (pipeline, GST_STATE_PLAYING);
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g_print ("Running\n");
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g_main_loop_run (loop);
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/* clean up nicely */
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gst_object_unref (cs);
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g_print ("Returned, stopping playback\n");
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gst_element_set_state (pipeline, GST_STATE_NULL);
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g_print ("Deleting pipeline\n");
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gst_object_unref (GST_OBJECT (pipeline));
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return 0;
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}
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