/* GStreamer * Copyright (C) 2009 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. */ #include #include #include static guint16 _get_first_sample (GstSample * sample) { GstAudioInfo info; GstCaps *caps; GstBuffer *buf; GstMapInfo map; guint16 res; fail_unless (sample != NULL, "NULL sample"); caps = gst_sample_get_caps (sample); fail_unless (caps != NULL, "sample without caps"); buf = gst_sample_get_buffer (sample); GST_DEBUG ("buffer with size=%" G_GSIZE_FORMAT ", caps=%" GST_PTR_FORMAT, gst_buffer_get_size (buf), caps); gst_buffer_map (buf, &map, GST_MAP_READ); /* log buffer details */ GST_MEMDUMP ("buffer data from decoder", map.data, map.size); /* make sure it's the format we expect */ fail_unless (gst_audio_info_from_caps (&info, caps)); fail_unless_equals_int (GST_AUDIO_INFO_WIDTH (&info), 16); fail_unless_equals_int (GST_AUDIO_INFO_DEPTH (&info), 16); fail_unless_equals_int (GST_AUDIO_INFO_RATE (&info), 44100); fail_unless_equals_int (GST_AUDIO_INFO_CHANNELS (&info), 1); if (GST_AUDIO_INFO_IS_LITTLE_ENDIAN (&info)) res = GST_READ_UINT16_LE (map.data); else res = GST_READ_UINT16_BE (map.data); gst_buffer_unmap (buf, &map); return res; } GST_START_TEST (test_decode) { GstElement *pipeline; GstElement *appsink; GstSample *sample = NULL; guint16 first_sample = 0; guint size = 0; gchar *path = g_build_filename (GST_TEST_FILES_PATH, "audiotestsrc.flac", NULL); gchar *pipe_desc = g_strdup_printf ("filesrc location=\"%s\" ! flacparse ! flacdec ! appsink name=sink", path); pipeline = gst_parse_launch (pipe_desc, NULL); fail_unless (pipeline != NULL); g_free (path); g_free (pipe_desc); appsink = gst_bin_get_by_name (GST_BIN (pipeline), "sink"); fail_unless (appsink != NULL); gst_element_set_state (pipeline, GST_STATE_PLAYING); do { g_signal_emit_by_name (appsink, "pull-sample", &sample); if (sample == NULL) break; if (first_sample == 0) first_sample = _get_first_sample (sample); size += gst_buffer_get_size (gst_sample_get_buffer (sample)); gst_sample_unref (sample); sample = NULL; } while (TRUE); /* audiotestsrc with samplesperbuffer 1024 and 10 num-buffers */ fail_unless_equals_int (size, 20480); fail_unless_equals_int (first_sample, 0x066a); gst_element_set_state (pipeline, GST_STATE_NULL); g_object_unref (pipeline); g_object_unref (appsink); } GST_END_TEST; GST_START_TEST (test_decode_seek_full) { GstElement *pipeline; GstElement *appsink; GstEvent *event; GstSample *sample = NULL; guint16 first_sample = 0; guint size = 0; gchar *path = g_build_filename (GST_TEST_FILES_PATH, "audiotestsrc.flac", NULL); gchar *pipe_desc = g_strdup_printf ("filesrc location=\"%s\" ! flacparse ! flacdec ! appsink name=sink", path); pipeline = gst_parse_launch (pipe_desc, NULL); fail_unless (pipeline != NULL); g_free (pipe_desc); g_free (path); appsink = gst_bin_get_by_name (GST_BIN (pipeline), "sink"); fail_unless (appsink != NULL); gst_element_set_state (pipeline, GST_STATE_PAUSED); gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE); /* do a seek that should give us the complete output */ event = gst_event_new_seek (1.0, GST_FORMAT_DEFAULT, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, 0, GST_SEEK_TYPE_SET, 20480); fail_unless (gst_element_send_event (appsink, event)); gst_element_set_state (pipeline, GST_STATE_PLAYING); do { g_signal_emit_by_name (appsink, "pull-sample", &sample); if (sample == NULL) break; if (first_sample == 0) first_sample = _get_first_sample (sample); size += gst_buffer_get_size (gst_sample_get_buffer (sample)); gst_sample_unref (sample); sample = NULL; } while (TRUE); /* file was generated with audiotestsrc * with 1024 samplesperbuffer and 10 num-buffers in 16 bit audio */ fail_unless_equals_int (size, 20480); fail_unless_equals_int (first_sample, 0x066a); gst_element_set_state (pipeline, GST_STATE_NULL); g_object_unref (pipeline); g_object_unref (appsink); } GST_END_TEST; GST_START_TEST (test_decode_seek_partial) { GstElement *pipeline; GstElement *appsink; GstEvent *event; GstSample *sample = NULL; guint size = 0; guint16 first_sample = 0; gchar *path = g_build_filename (GST_TEST_FILES_PATH, "audiotestsrc.flac", NULL); gchar *pipe_desc = g_strdup_printf ("filesrc location=\"%s\" ! flacparse ! flacdec ! appsink name=sink", path); pipeline = gst_parse_launch (pipe_desc, NULL); fail_unless (pipeline != NULL); g_free (path); g_free (pipe_desc); appsink = gst_bin_get_by_name (GST_BIN (pipeline), "sink"); fail_unless (appsink != NULL); gst_element_set_state (pipeline, GST_STATE_PAUSED); gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE); /* do a partial seek to get the first 1024 samples or 2048 bytes */ event = gst_event_new_seek (1.0, GST_FORMAT_DEFAULT, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, 0, GST_SEEK_TYPE_SET, 1024); GST_DEBUG ("seeking"); fail_unless (gst_element_send_event (appsink, event)); GST_DEBUG ("seeked"); gst_element_set_state (pipeline, GST_STATE_PLAYING); do { GST_DEBUG ("pulling sample"); g_signal_emit_by_name (appsink, "pull-sample", &sample); GST_DEBUG ("pulled sample %p", sample); if (sample == NULL) break; if (first_sample == 0) { first_sample = _get_first_sample (sample); } size += gst_buffer_get_size (gst_sample_get_buffer (sample)); gst_sample_unref (sample); sample = NULL; } while (TRUE); /* allow for sample round-up clipping effect */ fail_unless (size == 2048 || size == 2050); fail_unless_equals_int (first_sample, 0x066a); gst_element_set_state (pipeline, GST_STATE_NULL); g_object_unref (pipeline); g_object_unref (appsink); } GST_END_TEST; static Suite * flacdec_suite (void) { Suite *s = suite_create ("flacdec"); TCase *tc_chain = tcase_create ("linear"); /* time out after 60s, not the default 3 */ tcase_set_timeout (tc_chain, 60); suite_add_tcase (s, tc_chain); tcase_add_test (tc_chain, test_decode); tcase_add_test (tc_chain, test_decode_seek_full); tcase_add_test (tc_chain, test_decode_seek_partial); return s; } GST_CHECK_MAIN (flacdec);