/* GStreamer unit test for videoframe-audiolevel * * Copyright (C) 2015 Vivia Nikolaidou * * 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 #endif /* suppress warnings for deprecated API such as GValueArray * with newer GLib versions (>= 2.31.0) */ #define GLIB_DISABLE_DEPRECATION_WARNINGS #include #include static gboolean got_eos; static guint audio_buffer_count, video_buffer_count; static GstSegment current_audio_segment, current_video_segment; static guint num_msgs; static GQueue v_timestamp_q, msg_timestamp_q; static guint n_abuffers, n_vbuffers; static guint channels, fill_value; static gdouble expected_rms; static gboolean audiodelay, videodelay, per_channel, long_video; static gboolean early_video, late_video; static gboolean video_gaps, video_overlaps; static gboolean audio_nondiscont, audio_drift; static guint fill_value_per_channel[] = { 0, 1 }; static gdouble expected_rms_per_channel[] = { 0, 0.0078125 }; static void set_default_params (void) { n_abuffers = 40; n_vbuffers = 15; channels = 2; expected_rms = 0.0078125; fill_value = 1; audiodelay = FALSE; videodelay = FALSE; per_channel = FALSE; long_video = FALSE; video_gaps = FALSE; video_overlaps = FALSE; audio_nondiscont = FALSE; audio_drift = FALSE; early_video = FALSE; late_video = FALSE; }; static GstFlowReturn output_achain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstClockTime timestamp; guint8 b; gboolean audio_jitter = audio_nondiscont || audio_drift || early_video; timestamp = GST_BUFFER_TIMESTAMP (buffer); if (!audio_jitter) fail_unless_equals_int64 (timestamp, (audio_buffer_count % n_abuffers) * 1 * GST_SECOND); timestamp = gst_segment_to_stream_time (¤t_audio_segment, GST_FORMAT_TIME, timestamp); if (!audio_jitter) fail_unless_equals_int64 (timestamp, (audio_buffer_count % n_abuffers) * 1 * GST_SECOND); timestamp = GST_BUFFER_TIMESTAMP (buffer); timestamp = gst_segment_to_running_time (¤t_audio_segment, GST_FORMAT_TIME, timestamp); if (!audio_jitter) fail_unless_equals_int64 (timestamp, audio_buffer_count * 1 * GST_SECOND); gst_buffer_extract (buffer, 0, &b, 1); if (per_channel) { fail_unless_equals_int (b, fill_value_per_channel[0]); } else { fail_unless_equals_int (b, fill_value); } audio_buffer_count++; gst_buffer_unref (buffer); return GST_FLOW_OK; } static gboolean output_aevent (GstPad * pad, GstObject * parent, GstEvent * event) { switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: gst_segment_init (¤t_audio_segment, GST_FORMAT_UNDEFINED); break; case GST_EVENT_SEGMENT: gst_event_copy_segment (event, ¤t_audio_segment); break; case GST_EVENT_EOS: got_eos = TRUE; break; default: break; } gst_event_unref (event); return TRUE; } static GstFlowReturn output_vchain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstClockTime timestamp; guint8 b; gboolean jitter = video_gaps || video_overlaps || late_video; timestamp = GST_BUFFER_TIMESTAMP (buffer); if (!jitter) fail_unless_equals_int64 (timestamp, (video_buffer_count % n_vbuffers) * 25 * GST_MSECOND); timestamp = gst_segment_to_stream_time (¤t_video_segment, GST_FORMAT_TIME, timestamp); if (!jitter) fail_unless_equals_int64 (timestamp, (video_buffer_count % n_vbuffers) * 25 * GST_MSECOND); timestamp = GST_BUFFER_TIMESTAMP (buffer); timestamp = gst_segment_to_running_time (¤t_video_segment, GST_FORMAT_TIME, timestamp); if (!jitter) fail_unless_equals_int64 (timestamp, video_buffer_count * 25 * GST_MSECOND); gst_buffer_extract (buffer, 0, &b, 1); if (!jitter) fail_unless_equals_int (b, video_buffer_count % n_vbuffers); video_buffer_count++; gst_buffer_unref (buffer); return GST_FLOW_OK; } static gboolean output_vevent (GstPad * pad, GstObject * parent, GstEvent * event) { switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: gst_segment_init (¤t_video_segment, GST_FORMAT_UNDEFINED); break; case GST_EVENT_SEGMENT: gst_event_copy_segment (event, ¤t_video_segment); break; case GST_EVENT_EOS: got_eos = TRUE; break; default: break; } gst_event_unref (event); return TRUE; } static gpointer push_abuffers (gpointer data) { GstSegment segment; GstPad *pad = data; gint i, j, k; GstClockTime timestamp = 0; GstAudioInfo info; GstCaps *caps; guint buf_size = 1000; if (audiodelay) g_usleep (2000); if (early_video) timestamp = 50 * GST_MSECOND; gst_pad_send_event (pad, gst_event_new_stream_start ("test")); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S8, buf_size, channels, NULL); caps = gst_audio_info_to_caps (&info); gst_pad_send_event (pad, gst_event_new_caps (caps)); gst_caps_unref (caps); gst_segment_init (&segment, GST_FORMAT_TIME); gst_pad_send_event (pad, gst_event_new_segment (&segment)); for (i = 0; i < n_abuffers; i++) { GstBuffer *buf = gst_buffer_new_and_alloc (channels * buf_size); if (per_channel) { GstMapInfo map; guint8 *in_data; gst_buffer_map (buf, &map, GST_MAP_WRITE); in_data = map.data; for (j = 0; j < buf_size; j++) { for (k = 0; k < channels; k++) { in_data[j * channels + k] = fill_value_per_channel[k]; } } gst_buffer_unmap (buf, &map); } else { gst_buffer_memset (buf, 0, fill_value, channels * buf_size); } GST_BUFFER_TIMESTAMP (buf) = timestamp; timestamp += 1 * GST_SECOND; if (audio_drift) timestamp += 50 * GST_MSECOND; else if (i == 4 && audio_nondiscont) timestamp += 30 * GST_MSECOND; GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf); fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK); } gst_pad_send_event (pad, gst_event_new_eos ()); return NULL; } static gpointer push_vbuffers (gpointer data) { GstSegment segment; GstPad *pad = data; gint i; GstClockTime timestamp = 0; if (videodelay) g_usleep (2000); if (late_video) timestamp = 50 * GST_MSECOND; gst_pad_send_event (pad, gst_event_new_stream_start ("test")); gst_segment_init (&segment, GST_FORMAT_TIME); gst_pad_send_event (pad, gst_event_new_segment (&segment)); for (i = 0; i < n_vbuffers; i++) { GstBuffer *buf = gst_buffer_new_and_alloc (1000); GstClockTime *rtime = g_new (GstClockTime, 1); gst_buffer_memset (buf, 0, i, 1); GST_BUFFER_TIMESTAMP (buf) = timestamp; timestamp += 25 * GST_MSECOND; GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf); *rtime = gst_segment_to_running_time (&segment, GST_FORMAT_TIME, timestamp); g_queue_push_tail (&v_timestamp_q, rtime); if (i == 4) { if (video_gaps) timestamp += 10 * GST_MSECOND; else if (video_overlaps) timestamp -= 10 * GST_MSECOND; } fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK); } gst_pad_send_event (pad, gst_event_new_eos ()); return NULL; } static GstBusSyncReply on_message (GstBus * bus, GstMessage * message, gpointer user_data) { const GstStructure *s = gst_message_get_structure (message); const gchar *name = gst_structure_get_name (s); GValueArray *rms_arr; const GValue *array_val; const GValue *value; gdouble rms; gint channels2; guint i; GstClockTime *rtime; if (message->type != GST_MESSAGE_ELEMENT || strcmp (name, "videoframe-audiolevel") != 0) goto done; num_msgs++; rtime = g_new (GstClockTime, 1); if (!gst_structure_get_clock_time (s, "running-time", rtime)) { g_warning ("Could not parse running time"); g_free (rtime); } else { g_queue_push_tail (&msg_timestamp_q, rtime); } /* the values are packed into GValueArrays with the value per channel */ array_val = gst_structure_get_value (s, "rms"); rms_arr = (GValueArray *) g_value_get_boxed (array_val); channels2 = rms_arr->n_values; fail_unless_equals_int (channels2, channels); for (i = 0; i < channels; ++i) { value = g_value_array_get_nth (rms_arr, i); rms = g_value_get_double (value); if (per_channel) { fail_unless_equals_float (rms, expected_rms_per_channel[i]); } else if (early_video && *rtime <= 50 * GST_MSECOND) { fail_unless_equals_float (rms, 0); } else { fail_unless_equals_float (rms, expected_rms); } } done: return GST_BUS_PASS; } static void test_videoframe_audiolevel_generic (void) { GstElement *alevel; GstPad *asink, *vsink, *asrc, *vsrc, *aoutput_sink, *voutput_sink; GThread *athread, *vthread; GstBus *bus; guint i; got_eos = FALSE; audio_buffer_count = 0; video_buffer_count = 0; num_msgs = 0; g_queue_init (&v_timestamp_q); g_queue_init (&msg_timestamp_q); alevel = gst_element_factory_make ("videoframe-audiolevel", NULL); fail_unless (alevel != NULL); bus = gst_bus_new (); gst_element_set_bus (alevel, bus); gst_bus_set_sync_handler (bus, on_message, NULL, NULL); asink = gst_element_get_static_pad (alevel, "asink"); fail_unless (asink != NULL); vsink = gst_element_get_static_pad (alevel, "vsink"); fail_unless (vsink != NULL); asrc = gst_element_get_static_pad (alevel, "asrc"); aoutput_sink = gst_pad_new ("sink", GST_PAD_SINK); fail_unless (aoutput_sink != NULL); fail_unless (gst_pad_link (asrc, aoutput_sink) == GST_PAD_LINK_OK); vsrc = gst_element_get_static_pad (alevel, "vsrc"); voutput_sink = gst_pad_new ("sink", GST_PAD_SINK); fail_unless (voutput_sink != NULL); fail_unless (gst_pad_link (vsrc, voutput_sink) == GST_PAD_LINK_OK); gst_pad_set_chain_function (aoutput_sink, output_achain); gst_pad_set_event_function (aoutput_sink, output_aevent); gst_pad_set_chain_function (voutput_sink, output_vchain); gst_pad_set_event_function (voutput_sink, output_vevent); gst_pad_set_active (aoutput_sink, TRUE); gst_pad_set_active (voutput_sink, TRUE); fail_unless (gst_element_set_state (alevel, GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); athread = g_thread_new ("athread", (GThreadFunc) push_abuffers, asink); vthread = g_thread_new ("vthread", (GThreadFunc) push_vbuffers, vsink); g_thread_join (vthread); g_thread_join (athread); fail_unless (got_eos); fail_unless_equals_int (audio_buffer_count, n_abuffers); fail_unless_equals_int (video_buffer_count, n_vbuffers); if (!long_video) fail_unless_equals_int (num_msgs, n_vbuffers); fail_unless_equals_int (g_queue_get_length (&v_timestamp_q), n_vbuffers); /* num_msgs is equal to n_vbuffers except in the case of long_video */ fail_unless_equals_int (g_queue_get_length (&msg_timestamp_q), num_msgs); for (i = 0; i < g_queue_get_length (&msg_timestamp_q); i++) { GstClockTime *vt = g_queue_pop_head (&v_timestamp_q); GstClockTime *mt = g_queue_pop_head (&msg_timestamp_q); fail_unless (vt != NULL); fail_unless (mt != NULL); if (!video_gaps && !video_overlaps && !early_video) fail_unless_equals_uint64 (*vt, *mt); g_free (vt); g_free (mt); } /* teardown */ gst_element_set_state (alevel, GST_STATE_NULL); gst_bus_set_flushing (bus, TRUE); gst_object_unref (bus); g_queue_foreach (&v_timestamp_q, (GFunc) g_free, NULL); g_queue_foreach (&msg_timestamp_q, (GFunc) g_free, NULL); g_queue_clear (&v_timestamp_q); g_queue_clear (&msg_timestamp_q); gst_pad_unlink (asrc, aoutput_sink); gst_object_unref (asrc); gst_pad_unlink (vsrc, voutput_sink); gst_object_unref (vsrc); gst_object_unref (asink); gst_object_unref (vsink); gst_pad_set_active (aoutput_sink, FALSE); gst_object_unref (aoutput_sink); gst_pad_set_active (voutput_sink, FALSE); gst_object_unref (voutput_sink); gst_object_unref (alevel); } GST_START_TEST (test_videoframe_audiolevel_16chan_1) { set_default_params (); channels = 16; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_8chan_1) { set_default_params (); channels = 8; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_2chan_1) { set_default_params (); test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_1chan_1) { set_default_params (); channels = 1; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_16chan_0) { set_default_params (); channels = 16; expected_rms = 0; fill_value = 0; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_8chan_0) { set_default_params (); channels = 8; expected_rms = 0; fill_value = 0; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_2chan_0) { set_default_params (); channels = 2; expected_rms = 0; fill_value = 0; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_1chan_0) { set_default_params (); channels = 1; expected_rms = 0; fill_value = 0; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_adelay) { set_default_params (); audiodelay = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_vdelay) { set_default_params (); videodelay = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_per_channel) { set_default_params (); per_channel = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_long_video) { set_default_params (); n_abuffers = 6; n_vbuffers = 255; long_video = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_video_gaps) { set_default_params (); video_gaps = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_video_overlaps) { set_default_params (); video_overlaps = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_audio_nondiscont) { set_default_params (); audio_nondiscont = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_audio_drift) { set_default_params (); audio_drift = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_early_video) { set_default_params (); early_video = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; GST_START_TEST (test_videoframe_audiolevel_late_video) { set_default_params (); late_video = TRUE; test_videoframe_audiolevel_generic (); } GST_END_TEST; static Suite * videoframe_audiolevel_suite (void) { Suite *s = suite_create ("videoframe-audiolevel"); TCase *tc_chain; tc_chain = tcase_create ("videoframe-audiolevel"); tcase_add_test (tc_chain, test_videoframe_audiolevel_16chan_1); tcase_add_test (tc_chain, test_videoframe_audiolevel_8chan_1); tcase_add_test (tc_chain, test_videoframe_audiolevel_2chan_1); tcase_add_test (tc_chain, test_videoframe_audiolevel_1chan_1); tcase_add_test (tc_chain, test_videoframe_audiolevel_16chan_0); tcase_add_test (tc_chain, test_videoframe_audiolevel_8chan_0); tcase_add_test (tc_chain, test_videoframe_audiolevel_2chan_0); tcase_add_test (tc_chain, test_videoframe_audiolevel_1chan_0); tcase_add_test (tc_chain, test_videoframe_audiolevel_adelay); tcase_add_test (tc_chain, test_videoframe_audiolevel_vdelay); tcase_add_test (tc_chain, test_videoframe_audiolevel_per_channel); tcase_add_test (tc_chain, test_videoframe_audiolevel_long_video); tcase_add_test (tc_chain, test_videoframe_audiolevel_video_gaps); tcase_add_test (tc_chain, test_videoframe_audiolevel_video_overlaps); tcase_add_test (tc_chain, test_videoframe_audiolevel_audio_nondiscont); tcase_add_test (tc_chain, test_videoframe_audiolevel_audio_drift); tcase_add_test (tc_chain, test_videoframe_audiolevel_early_video); tcase_add_test (tc_chain, test_videoframe_audiolevel_late_video); suite_add_tcase (s, tc_chain); return s; } GST_CHECK_MAIN (videoframe_audiolevel);