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/*
* GStreamer
*
* unit test for amrnbenc
*
* 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.
*/
#include <gst/check/gstcheck.h>
#include <gst/audio/audio.h>
#define SRC_CAPS "audio/x-raw, format = (string)" GST_AUDIO_NE (S16) ", " \
"layout = (string) interleaved, channels = (int) 1, rate = (int) 8000"
#define SINK_CAPS "audio/AMR"
GList *buffers;
GList *current_buf = NULL;
GstPad *srcpad, *sinkpad;
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SINK_CAPS)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SRC_CAPS)
);
static void
buffer_unref (void *buffer, void *user_data)
{
gst_buffer_unref (GST_BUFFER (buffer));
}
static GstElement *
setup_amrnbenc (void)
{
GstElement *amrnbenc;
GstCaps *caps;
GstBus *bus;
guint64 granulerate_n, granulerate_d;
GST_DEBUG ("setup_amrnbenc");
amrnbenc = gst_check_setup_element ("amrnbenc");
srcpad = gst_check_setup_src_pad (amrnbenc, &srctemplate);
sinkpad = gst_check_setup_sink_pad (amrnbenc, &sinktemplate);
gst_pad_set_active (srcpad, TRUE);
gst_pad_set_active (sinkpad, TRUE);
bus = gst_bus_new ();
gst_element_set_bus (amrnbenc, bus);
fail_unless (gst_element_set_state (amrnbenc,
GST_STATE_PLAYING) != GST_STATE_CHANGE_FAILURE,
"could not set to playing");
caps = gst_caps_from_string (SRC_CAPS);
gst_check_setup_events (srcpad, amrnbenc, caps, GST_FORMAT_TIME);
gst_caps_unref (caps);
buffers = NULL;
return amrnbenc;
}
static void
cleanup_amrnbenc (GstElement * amrnbenc)
{
GstBus *bus;
/* free encoded buffers */
g_list_foreach (buffers, buffer_unref, NULL);
g_list_free (buffers);
buffers = NULL;
bus = GST_ELEMENT_BUS (amrnbenc);
gst_bus_set_flushing (bus, TRUE);
gst_object_unref (bus);
GST_DEBUG ("cleanup_amrnbenc");
gst_pad_set_active (srcpad, FALSE);
gst_pad_set_active (sinkpad, FALSE);
gst_check_teardown_src_pad (amrnbenc);
gst_check_teardown_sink_pad (amrnbenc);
gst_check_teardown_element (amrnbenc);
}
/* push a random block of audio of the given size */
static void
push_data (gint size, GstFlowReturn expected_return)
{
GstBuffer *buffer;
GstFlowReturn res;
buffer = gst_buffer_new_and_alloc (size);
/* make valgrind happier */
gst_buffer_memset (buffer, 0, 0, size);
res = gst_pad_push (srcpad, buffer);
fail_unless (res == expected_return,
"pushing audio returned %d (%s) not %d (%s)", res,
gst_flow_get_name (res), expected_return,
gst_flow_get_name (expected_return));
}
GST_START_TEST (test_enc)
{
GstElement *amrnbenc;
amrnbenc = setup_amrnbenc ();
push_data (1000, GST_FLOW_OK);
cleanup_amrnbenc (amrnbenc);
}
GST_END_TEST;
static Suite *
amrnbenc_suite ()
{
Suite *s = suite_create ("amrnbenc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_enc);
return s;
}
int
main (int argc, char **argv)
{
int nf;
Suite *s = amrnbenc_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;
}
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