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