/* GStreamer * * unit test for audioconvert * * Copyright (C) <2005> 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 #include GList *buffers = NULL; gboolean have_eos = FALSE; /* 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 */ GstPad *mysrcpad, *mysinkpad; #define CONVERT_CAPS_TEMPLATE_STRING \ "audio/x-raw-float, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) BYTE_ORDER, " \ "width = (int) 32, " \ "buffer-frames = (int) [ 0, MAX ];" \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 32, " \ "depth = (int) [ 1, 32 ], " \ "signed = (boolean) { true, false }; " \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 24, " \ "depth = (int) [ 1, 24 ], " \ "signed = (boolean) { true, false }; " \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 16, " \ "depth = (int) [ 1, 16 ], " \ "signed = (boolean) { true, false }; " \ "audio/x-raw-int, " \ "rate = (int) [ 1, MAX ], " \ "channels = (int) [ 1, 8 ], " \ "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " \ "width = (int) 8, " \ "depth = (int) [ 1, 8 ], " \ "signed = (boolean) { true, false } " static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING) ); static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (CONVERT_CAPS_TEMPLATE_STRING) ); /* takes over reference for outcaps */ GstElement * setup_audioconvert (GstCaps * outcaps) { GstElement *audioconvert; GST_DEBUG ("setup_audioconvert"); audioconvert = gst_check_setup_element ("audioconvert"); mysrcpad = gst_check_setup_src_pad (audioconvert, &srctemplate, NULL); mysinkpad = gst_check_setup_sink_pad (audioconvert, &sinktemplate, NULL); /* this installs a getcaps func that will always return the caps we set * later */ gst_pad_use_fixed_caps (mysinkpad); gst_pad_set_caps (mysinkpad, outcaps); gst_caps_unref (outcaps); outcaps = gst_pad_get_negotiated_caps (mysinkpad); fail_unless (gst_caps_is_fixed (outcaps)); gst_caps_unref (outcaps); return audioconvert; } void cleanup_audioconvert (GstElement * audioconvert) { GST_DEBUG ("cleanup_audioconvert"); gst_check_teardown_src_pad (audioconvert); gst_check_teardown_sink_pad (audioconvert); gst_check_teardown_element (audioconvert); } GstCaps * get_int_caps (guint rate, guint channels, gchar * endianness, guint width, guint depth, gboolean signedness) { GstCaps *caps; gchar *string; string = g_strdup_printf ("audio/x-raw-int, " "rate = (int) %d, " "channels = (int) %d, " "endianness = (int) %s, " "width = (int) %d, " "depth = (int) %d, " "signed = (boolean) %s ", rate, channels, endianness, width, depth, signedness ? "true" : "false"); GST_DEBUG ("creating caps from %s", string); caps = gst_caps_from_string (string); fail_unless (caps != NULL); g_free (string); return caps; } static void verify_convert (GstElement * audioconvert, void *in, int inlength, void *out, int outlength, GstCaps * incaps) { GstBuffer *inbuffer, *outbuffer; fail_unless (gst_element_set_state (audioconvert, GST_STATE_PLAYING) == GST_STATE_SUCCESS, "could not set to playing"); GST_DEBUG ("Creating buffer of %d bytes", inlength); inbuffer = gst_buffer_new_and_alloc (inlength); memcpy (GST_BUFFER_DATA (inbuffer), in, inlength); gst_buffer_set_caps (inbuffer, incaps); ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1); /* pushing gives away my reference ... */ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK); /* ... and puts a new buffer on the global list */ fail_unless (g_list_length (buffers) == 1); fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL); ASSERT_BUFFER_REFCOUNT (outbuffer, "outbuffer", 1); fail_unless_equals_int (GST_BUFFER_SIZE (outbuffer), outlength); fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), out, outlength) == 0); } GST_START_TEST (test_unity) { GstElement *audioconvert; GstCaps *incaps, *outcaps; gint16 in[] = { 16384, -256 }; gint16 out[] = { 8064 }; outcaps = get_int_caps (44100, 1, "LITTLE_ENDIAN", 16, 16, TRUE); audioconvert = setup_audioconvert (outcaps); incaps = get_int_caps (44100, 2, "LITTLE_ENDIAN", 16, 16, TRUE); verify_convert (audioconvert, in, sizeof (in), out, sizeof (out), incaps); /* cleanup */ cleanup_audioconvert (audioconvert); } GST_END_TEST; Suite * audioconvert_suite (void) { Suite *s = suite_create ("audioconvert"); TCase *tc_chain = tcase_create ("general"); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_unity); return s; } int main (int argc, char **argv) { int nf; Suite *s = audioconvert_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; }