/* GStreamer * * unit test for audiotestsrc * * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include /* 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, " \ "format = (string) "GST_AUDIO_NE(S16)", " \ "channels = (int) 1, " \ "rate = (int) [ 1, MAX ]" 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); 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); g_object_set (audiotestsrc, "wave", values[j].value, NULL); 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; #define TEST_LAYOUT_CHANNELS 6 static GstStaticPadTemplate sinktemplate_interleaved = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", " "channels = (int) " G_STRINGIFY (TEST_LAYOUT_CHANNELS) ", " "rate = (int) [ 1, MAX ], layout = (string) interleaved") ); static GstStaticPadTemplate sinktemplate_planar = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", " "channels = (int) " G_STRINGIFY (TEST_LAYOUT_CHANNELS) ", " "rate = (int) [ 1, MAX ], layout = (string) non-interleaved") ); typedef enum { GST_AUDIO_TEST_SRC_WAVE_SINE, GST_AUDIO_TEST_SRC_WAVE_SQUARE, GST_AUDIO_TEST_SRC_WAVE_SAW, GST_AUDIO_TEST_SRC_WAVE_TRIANGLE, GST_AUDIO_TEST_SRC_WAVE_SILENCE, GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE, GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE, GST_AUDIO_TEST_SRC_WAVE_SINE_TAB, GST_AUDIO_TEST_SRC_WAVE_TICKS, GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE, GST_AUDIO_TEST_SRC_WAVE_RED_NOISE, GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE, GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE, _GST_AUDIO_TEST_SRC_WAVE_LAST } GstAudioTestSrcWave; GST_START_TEST (test_layout) { GstHarness *interleavedsrc, *plannarsrc; GObjectClass *oclass; GParamSpec *property; GEnumValue *values; guint i, j; interleavedsrc = gst_harness_new_with_templates ("audiotestsrc", NULL, &sinktemplate_interleaved); plannarsrc = gst_harness_new_with_templates ("audiotestsrc", NULL, &sinktemplate_planar); gst_harness_use_testclock (interleavedsrc); gst_harness_use_testclock (plannarsrc); g_object_set (interleavedsrc->element, "is-live", TRUE, NULL); g_object_set (plannarsrc->element, "is-live", TRUE, NULL); oclass = G_OBJECT_GET_CLASS (interleavedsrc->element); 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; for (j = 0; values[j].value_name; j++) { /* these produce random values by definition, * so we can't compare channels */ switch (j) { case GST_AUDIO_TEST_SRC_WAVE_WHITE_NOISE: case GST_AUDIO_TEST_SRC_WAVE_PINK_NOISE: case GST_AUDIO_TEST_SRC_WAVE_GAUSSIAN_WHITE_NOISE: case GST_AUDIO_TEST_SRC_WAVE_RED_NOISE: case GST_AUDIO_TEST_SRC_WAVE_BLUE_NOISE: case GST_AUDIO_TEST_SRC_WAVE_VIOLET_NOISE: continue; default: break; } GST_DEBUG ("layout test with wave %s", values[j].value_name); g_object_set (interleavedsrc->element, "wave", values[j].value, NULL); g_object_set (plannarsrc->element, "wave", values[j].value, NULL); if (j == 0) { GST_DEBUG ("gst_harness_play"); gst_harness_play (interleavedsrc); gst_harness_play (plannarsrc); } else { GST_DEBUG ("discarding buffers with old wave"); fail_unless (gst_harness_crank_single_clock_wait (interleavedsrc)); fail_unless (gst_harness_crank_single_clock_wait (plannarsrc)); gst_buffer_unref (gst_harness_pull (interleavedsrc)); gst_buffer_unref (gst_harness_pull (plannarsrc)); } for (i = 0; i < 10; i++) { GstBuffer *ibuf, *pbuf; GstMapInfo imap, pmap; GstAudioMeta *meta; GstAudioBuffer pabuf; gint16 *iptr, *pptr; guint isamples, psamples, s, c; GST_DEBUG ("waiting on clock"); fail_unless (gst_harness_crank_single_clock_wait (interleavedsrc)); fail_unless (gst_harness_crank_single_clock_wait (plannarsrc)); ibuf = gst_harness_pull (interleavedsrc); pbuf = gst_harness_pull (plannarsrc); gst_buffer_map (ibuf, &imap, GST_MAP_READ); gst_buffer_map (pbuf, &pmap, GST_MAP_READ); /* buffers should have the same size in bytes and in samples */ fail_unless_equals_int (imap.size, pmap.size); isamples = imap.size / TEST_LAYOUT_CHANNELS; isamples /= 2; /* S16 -> 2 bytes per sample */ fail_unless_equals_int (imap.size % TEST_LAYOUT_CHANNELS, 0); psamples = pmap.size / TEST_LAYOUT_CHANNELS; psamples /= 2; /* S16 -> 2 bytes per sample */ fail_unless_equals_int (pmap.size % TEST_LAYOUT_CHANNELS, 0); fail_unless_equals_int (isamples, psamples); iptr = (gint16 *) imap.data; pptr = (gint16 *) pmap.data; GST_DEBUG ("verifying contents of buffers; samples=%d, channels=%d", isamples, TEST_LAYOUT_CHANNELS); for (s = 0; s < isamples; s++) { for (c = 0; c < TEST_LAYOUT_CHANNELS; c++) { guint iidx = s * TEST_LAYOUT_CHANNELS + c; guint pidx = c * isamples + s; GST_TRACE ("s = %u | c = %u | iidx (s * channels + c) = %u | " "pidx (c * samples + s) = %u", s, c, iidx, pidx); fail_unless (iidx < imap.size / 2); fail_unless (pidx < pmap.size / 2); fail_unless_equals_int (iptr[iidx], pptr[pidx]); } } gst_buffer_unmap (pbuf, &pmap); GST_DEBUG ("verify that mapping through GstAudioBuffer works the same"); meta = gst_buffer_get_audio_meta (pbuf); fail_unless (meta); gst_audio_buffer_map (&pabuf, &meta->info, pbuf, GST_MAP_READ); for (s = 0; s < isamples; s++) { for (c = 0; c < TEST_LAYOUT_CHANNELS; c++) { guint iidx = s * TEST_LAYOUT_CHANNELS + c; fail_unless_equals_int (iptr[iidx], ((gint16 *) pabuf.planes[c])[s]); } } gst_audio_buffer_unmap (&pabuf); gst_buffer_unmap (ibuf, &imap); gst_buffer_unref (ibuf); gst_buffer_unref (pbuf); } /* ensure the audiotestsrcs are not in fill() while we change the wave */ fail_unless (gst_harness_wait_for_clock_id_waits (interleavedsrc, 1, 1)); fail_unless (gst_harness_wait_for_clock_id_waits (plannarsrc, 1, 1)); } /* make sure we ran the test */ fail_unless_equals_int (j, _GST_AUDIO_TEST_SRC_WAVE_LAST); gst_harness_teardown (interleavedsrc); gst_harness_teardown (plannarsrc); } 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); tcase_add_test (tc_chain, test_layout); return s; } GST_CHECK_MAIN (audiotestsrc);