/* GStreamer

 *

 * unit test for audiotestsrc

 *

 * Copyright (C) <2005> 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., 59 Temple Place - Suite 330,

 * Boston, MA 02111-1307, USA.

 */

#include <unistd.h>

#include <gst/check/gstcheck.h>

/* For ease of programming we use globals to keep refs for our floating

 * src and sink pads we create; otherwise we always have to do get_pad,

 * get_peer, and then remove references in every test function */

static GstPad *mysinkpad;

#define CAPS_TEMPLATE_STRING            \

    "audio/x-raw-int, "                 \

    "channels = (int) 1, "              \

    "rate = (int) [ 1,  MAX ], "        \

    "endianness = (int) BYTE_ORDER, "   \

    "width = (int) 16, "                \

    "depth = (int) 16, "                \

    "signed = (bool) TRUE"

static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS (CAPS_TEMPLATE_STRING)

    );

static GstElement *

setup_audiotestsrc (void)

{

  GstElement *audiotestsrc;

  GST_DEBUG ("setup_audiotestsrc");

  audiotestsrc = gst_check_setup_element ("audiotestsrc");

  mysinkpad = gst_check_setup_sink_pad (audiotestsrc, &sinktemplate, NULL);

  gst_pad_set_active (mysinkpad, TRUE);

  return audiotestsrc;

}

static void

cleanup_audiotestsrc (GstElement * audiotestsrc)

{

  GST_DEBUG ("cleanup_audiotestsrc");

  g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);

  g_list_free (buffers);

  buffers = NULL;

  gst_pad_set_active (mysinkpad, FALSE);

  gst_check_teardown_sink_pad (audiotestsrc);

  gst_check_teardown_element (audiotestsrc);

}

GST_START_TEST (test_all_waves)

{

  GstElement *audiotestsrc;

  GObjectClass *oclass;

  GParamSpec *property;

  GEnumValue *values;

  guint j = 0;

  audiotestsrc = setup_audiotestsrc ();

  oclass = G_OBJECT_GET_CLASS (audiotestsrc);

  property = g_object_class_find_property (oclass, "wave");

  fail_unless (G_IS_PARAM_SPEC_ENUM (property));

  values = G_ENUM_CLASS (g_type_class_ref (property->value_type))->values;

  while (values[j].value_name) {

    GST_DEBUG_OBJECT (audiotestsrc, "testing wave %s", values[j].value_name);

    fail_unless (gst_element_set_state (audiotestsrc,

            GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,

        "could not set to playing");

    g_mutex_lock (check_mutex);

    while (g_list_length (buffers) < 10)

      g_cond_wait (check_cond, check_mutex);

    g_mutex_unlock (check_mutex);

    gst_element_set_state (audiotestsrc, GST_STATE_READY);

    g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);

    g_list_free (buffers);

    buffers = NULL;

    ++j;

  }

  /* cleanup */

  cleanup_audiotestsrc (audiotestsrc);

}

GST_END_TEST;

static Suite *

audiotestsrc_suite (void)

{

  Suite *s = suite_create ("audiotestsrc");

  TCase *tc_chain = tcase_create ("general");

  suite_add_tcase (s, tc_chain);

  tcase_add_test (tc_chain, test_all_waves);

  return s;

}

int

main (int argc, char **argv)

{

  int nf;

  Suite *s = audiotestsrc_suite ();

  SRunner *sr = srunner_create (s);

  gst_check_init (&argc, &argv);

  srunner_run_all (sr, CK_NORMAL);

  nf = srunner_ntests_failed (sr);

  srunner_free (sr);

  return nf;

}