tests/check/elements/videoframe-audiolevel.c - imx-gst-plugins-bad - Git at Google

 /* GStreamer unit test for videoframe-audiolevel
  *
  * Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
  *
  * 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

 /* suppress warnings for deprecated API such as GValueArray
  * with newer GLib versions (>= 2.31.0) */
 #define GLIB_DISABLE_DEPRECATION_WARNINGS

 #include <gst/check/gstcheck.h>
 #include <gst/audio/audio.h>

 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 (&current_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 (&current_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 (&current_audio_segment, GST_FORMAT_UNDEFINED);
       break;
     case GST_EVENT_SEGMENT:
       gst_event_copy_segment (event, &current_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 (&current_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 (&current_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 (&current_video_segment, GST_FORMAT_UNDEFINED);
       break;
     case GST_EVENT_SEGMENT:
       gst_event_copy_segment (event, &current_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 = g_new (GstClockTime, 1);

   if (message->type != GST_MESSAGE_ELEMENT
       || strcmp (name, "videoframe-audiolevel") != 0)
     goto done;

   num_msgs++;
   if (!gst_structure_get_clock_time (s, "running-time", rtime))
     g_warning ("Could not parse running time");
   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);
	/* GStreamer unit test for videoframe-audiolevel
	*
	* Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
	*
	* 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

	/* suppress warnings for deprecated API such as GValueArray
	* with newer GLib versions (>= 2.31.0) */
	#define GLIB_DISABLE_DEPRECATION_WARNINGS

	#include <gst/check/gstcheck.h>
	#include <gst/audio/audio.h>

	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 (&current_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 (&current_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 (&current_audio_segment, GST_FORMAT_UNDEFINED);
	break;
	case GST_EVENT_SEGMENT:
	gst_event_copy_segment (event, &current_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 (&current_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 (&current_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 (&current_video_segment, GST_FORMAT_UNDEFINED);
	break;
	case GST_EVENT_SEGMENT:
	gst_event_copy_segment (event, &current_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 = g_new (GstClockTime, 1);

	if (message->type != GST_MESSAGE_ELEMENT
	\|\| strcmp (name, "videoframe-audiolevel") != 0)
	goto done;

	num_msgs++;
	if (!gst_structure_get_clock_time (s, "running-time", rtime))
	g_warning ("Could not parse running time");
	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);