/* * GStreamer * * unit test for amrnbenc * * Copyright (C) 2006 Thomas Vander Stichele * * 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 #define SRC_CAPS "audio/x-raw-int,width=16,depth=16,channels=1,rate=8000,signed=true,endianness=BYTE_ORDER" #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) ); /* takes a copy of the passed buffer data */ GstBuffer * buffer_new (const gchar * buffer_data, guint size) { GstBuffer *buffer; GstCaps *caps; buffer = gst_buffer_new_and_alloc (size); memcpy (GST_BUFFER_DATA (buffer), buffer_data, size); caps = gst_caps_from_string (SRC_CAPS); gst_buffer_set_caps (buffer, caps); gst_caps_unref (caps); return buffer; } static void buffer_unref (void *buffer, void *user_data) { gst_buffer_unref (GST_BUFFER (buffer)); } GstElement * setup_amrnbenc () { GstElement *amrnbenc; 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, NULL); sinkpad = gst_check_setup_sink_pad (amrnbenc, &sinktemplate, NULL); 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"); 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; gchar *data = g_malloc0 (size); buffer = buffer_new (data, size); g_free (data); res = gst_pad_push (srcpad, buffer); fail_unless (res == expected_return, "pushing audio returned %d not %d", res, 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; }