audiomixer: remove, moved to -base

https://bugzilla.gnome.org/show_bug.cgi?id=791218

5 files changed, 1 insertions, 3043 deletions

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 0343da7eb..5a7c497b2 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -127,7 +127,7 @@ check_kate=
 endif
 
 if HAVE_ORC
-check_orc = orc/bayer orc/audiomixer orc/compositor
+check_orc = orc/bayer orc/compositor
 else
 check_orc =
 endif
@@ -257,8 +257,6 @@ check_PROGRAMS = \
 	elements/videoframe-audiolevel \
 	elements/autoconvert \
 	elements/autovideoconvert \
-	elements/audiointerleave \
-	elements/audiomixer \
 	elements/asfmux \
 	elements/camerabin \
 	elements/gdppay \
@@ -313,12 +311,6 @@ LDADD = $(GST_CHECK_LIBS)
 generic_states_CFLAGS = $(AM_CFLAGS) $(GLIB_CFLAGS)
 generic_states_LDADD = $(LDADD) $(GLIB_LIBS)
 
-elements_audiomixer_LDADD = $(GST_BASE_LIBS) $(GST_CONTROLLER_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD)
-elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CONTROLLER_CFLAGS) $(AM_CFLAGS)
-
-elements_audiointerleave_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ $(GST_AUDIO_LIBS) $(LDADD)
-elements_audiointerleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
-
 elements_pnm_CFLAGS = \
 	$(GST_PLUGINS_BASE_CFLAGS) \
 	$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)
@@ -542,14 +534,6 @@ orc/bayer.c: $(top_srcdir)/gst/bayer/gstbayerorc.orc
 	$(MKDIR_P) orc
 	$(ORCC) --test -o $@ $<
 
-orc_audiomixer_CFLAGS = $(ORC_CFLAGS)
-orc_audiomixer_LDADD = $(ORC_LIBS) -lorc-test-0.4
-nodist_orc_audiomixer_SOURCES = orc/audiomixer.c
-
-orc/audiomixer.c: $(top_srcdir)/gst/audiomixer/gstaudiomixerorc.orc
-	$(MKDIR_P) orc
-	$(ORCC) --test -o $@ $<
-
 elements_compositor_LDADD = \
 	$(GST_PLUGINS_BASE_LIBS) $(GST_VIDEO_LIBS) $(GST_BASE_LIBS) $(LDADD)
 elements_compositor_CFLAGS = \
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index 741772b5a..d264dae57 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -2,8 +2,6 @@
 aiffparse
 asfmux
 assrender
-audiointerleave
-audiomixer
 autoconvert
 autovideoconvert
 baseaudiovisualizer
diff --git a/tests/check/elements/audiointerleave.c b/tests/check/elements/audiointerleave.c
deleted file mode 100644
index 71348f459..000000000
--- a/tests/check/elements/audiointerleave.c
+++ /dev/null
@@ -1,1128 +0,0 @@
-/* GStreamer unit tests for the audiointerleave element
- * Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
- * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * 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.
- */
-
-/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
- * with newer GLib versions (>= 2.31.0) */
-#define GLIB_DISABLE_DEPRECATION_WARNINGS
-
-#ifdef HAVE_CONFIG_H
-# include "config.h"
-#endif
-
-#ifdef HAVE_VALGRIND
-# include <valgrind/valgrind.h>
-#endif
-
-#include <gst/check/gstcheck.h>
-#include <gst/audio/audio.h>
-#include <gst/audio/audio-enumtypes.h>
-
-#include <gst/check/gstharness.h>
-
-static void
-gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element,
-    GstCaps * caps, GstFormat format, const gchar * stream_id)
-{
-  GstSegment segment;
-
-  gst_segment_init (&segment, format);
-
-  fail_unless (gst_pad_push_event (srcpad,
-          gst_event_new_stream_start (stream_id)));
-  if (caps)
-    fail_unless (gst_pad_push_event (srcpad, gst_event_new_caps (caps)));
-  fail_unless (gst_pad_push_event (srcpad, gst_event_new_segment (&segment)));
-}
-
-GST_START_TEST (test_create_and_unref)
-{
-  GstElement *interleave;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_object_unref (interleave);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_request_pads)
-{
-  GstElement *interleave;
-  GstPad *pad1, *pad2;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  pad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (pad1 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (pad1), "sink_0");
-
-  pad2 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (pad2 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (pad2), "sink_1");
-
-  gst_element_release_request_pad (interleave, pad2);
-  gst_object_unref (pad2);
-  gst_element_release_request_pad (interleave, pad1);
-  gst_object_unref (pad1);
-
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_object_unref (interleave);
-}
-
-GST_END_TEST;
-
-static GstPad **mysrcpads, *mysinkpad;
-static GstBus *bus;
-static GstElement *interleave;
-static GMutex data_mutex;
-static GCond data_cond;
-static gint have_data;
-static gfloat input[2];
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
-    GST_PAD_SINK,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) " GST_AUDIO_NE (F32) ", "
-        "channels = (int) 2, layout = (string) {interleaved, non-interleaved}, rate = (int) 48000"));
-
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
-    GST_PAD_SRC,
-    GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw, "
-        "format = (string) " GST_AUDIO_NE (F32) ", "
-        "channels = (int) 1, layout = (string) interleaved, rate = (int) 48000"));
-
-#define CAPS_48khz \
-        "audio/x-raw, " \
-        "format = (string) " GST_AUDIO_NE (F32) ", " \
-        "channels = (int) 1, layout = (string) non-interleaved," \
-        "rate = (int) 48000"
-
-static GstFlowReturn
-interleave_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer)
-{
-  GstMapInfo map;
-  gfloat *outdata;
-  gint i;
-
-  fail_unless (GST_IS_BUFFER (buffer));
-  fail_unless (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP));
-  gst_buffer_map (buffer, &map, GST_MAP_READ);
-  outdata = (gfloat *) map.data;
-  fail_unless (outdata != NULL);
-
-#ifdef HAVE_VALGRIND
-  if (!(RUNNING_ON_VALGRIND))
-#endif
-    for (i = 0; i < map.size / sizeof (float); i += 2) {
-      fail_unless_equals_float (outdata[i], input[0]);
-      fail_unless_equals_float (outdata[i + 1], input[1]);
-    }
-
-  g_mutex_lock (&data_mutex);
-  have_data += map.size;
-  g_cond_signal (&data_cond);
-  g_mutex_unlock (&data_mutex);
-
-  gst_buffer_unmap (buffer, &map);
-  gst_buffer_unref (buffer);
-
-
-  return GST_FLOW_OK;
-}
-
-GST_START_TEST (test_audiointerleave_2ch)
-{
-  GstElement *queue;
-  GstPad *sink0, *sink1, *src, *tmp;
-  GstCaps *caps;
-  gint i;
-  GstBuffer *inbuf;
-  gfloat *indata;
-  GstMapInfo map;
-
-  mysrcpads = g_new0 (GstPad *, 2);
-
-  have_data = 0;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-
-  sink0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink0 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
-
-  sink1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink1 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
-
-  mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
-  fail_unless (mysrcpads[0] != NULL);
-
-  caps = gst_caps_from_string (CAPS_48khz);
-  gst_pad_set_active (mysrcpads[0], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
-      GST_FORMAT_TIME, "0");
-  gst_pad_use_fixed_caps (mysrcpads[0]);
-
-  mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
-  fail_unless (mysrcpads[1] != NULL);
-
-  gst_pad_set_active (mysrcpads[1], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
-      GST_FORMAT_TIME, "1");
-  gst_pad_use_fixed_caps (mysrcpads[1]);
-
-  tmp = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
-
-  mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
-  fail_unless (mysinkpad != NULL);
-  gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
-  gst_pad_set_active (mysinkpad, TRUE);
-
-  src = gst_element_get_static_pad (interleave, "src");
-  fail_unless (src != NULL);
-  fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
-  gst_object_unref (src);
-
-  bus = gst_bus_new ();
-  gst_element_set_bus (interleave, bus);
-
-  fail_unless (gst_element_set_state (interleave,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-  fail_unless (gst_element_set_state (queue,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
-  input[0] = -1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = -1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
-  input[1] = 1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = GST_SECOND;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = -1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  //GST_BUFFER_PTS (inbuf) = GST_SECOND;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  g_mutex_lock (&data_mutex);
-  while (have_data < 48000 * 2 * 2 * sizeof (float))
-    g_cond_wait (&data_cond, &data_mutex);
-  g_mutex_unlock (&data_mutex);
-
-  gst_bus_set_flushing (bus, TRUE);
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_element_set_state (queue, GST_STATE_NULL);
-
-  gst_object_unref (mysrcpads[0]);
-  gst_object_unref (mysrcpads[1]);
-  gst_object_unref (mysinkpad);
-
-  gst_element_release_request_pad (interleave, sink0);
-  gst_object_unref (sink0);
-  gst_element_release_request_pad (interleave, sink1);
-  gst_object_unref (sink1);
-
-  gst_object_unref (interleave);
-  gst_object_unref (queue);
-  gst_object_unref (bus);
-  gst_caps_unref (caps);
-
-  g_free (mysrcpads);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_1eos)
-{
-  GstElement *queue;
-  GstPad *sink0, *sink1, *src, *tmp;
-  GstCaps *caps;
-  gint i;
-  GstBuffer *inbuf;
-  gfloat *indata;
-  GstMapInfo map;
-
-  mysrcpads = g_new0 (GstPad *, 2);
-
-  have_data = 0;
-
-  interleave = gst_element_factory_make ("audiointerleave", NULL);
-  fail_unless (interleave != NULL);
-
-  g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-
-  sink0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink0 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
-
-  sink1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sink1 != NULL);
-  fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
-
-  mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
-  fail_unless (mysrcpads[0] != NULL);
-
-  caps = gst_caps_from_string (CAPS_48khz);
-  gst_pad_set_active (mysrcpads[0], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
-      GST_FORMAT_TIME, "0");
-  gst_pad_use_fixed_caps (mysrcpads[0]);
-
-  mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
-  fail_unless (mysrcpads[1] != NULL);
-
-  gst_pad_set_active (mysrcpads[1], TRUE);
-  gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
-      GST_FORMAT_TIME, "1");
-  gst_pad_use_fixed_caps (mysrcpads[1]);
-
-  tmp = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
-
-  mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
-  fail_unless (mysinkpad != NULL);
-  gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
-  gst_pad_set_active (mysinkpad, TRUE);
-
-  src = gst_element_get_static_pad (interleave, "src");
-  fail_unless (src != NULL);
-  fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
-  gst_object_unref (src);
-
-  bus = gst_bus_new ();
-  gst_element_set_bus (interleave, bus);
-
-  fail_unless (gst_element_set_state (interleave,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-  fail_unless (gst_element_set_state (queue,
-          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
-  input[0] = -1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = -1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
-  input[1] = 1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  GST_BUFFER_PTS (inbuf) = 0;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  g_mutex_lock (&data_mutex);
-  /* 48000 samples per buffer * 2 sources * 2 buffers */
-  while (have_data != 48000 * 2 * sizeof (float))
-    g_cond_wait (&data_cond, &data_mutex);
-  g_mutex_unlock (&data_mutex);
-
-  input[0] = 0.0;
-  gst_pad_push_event (mysrcpads[0], gst_event_new_eos ());
-
-  input[1] = 1.0;
-  inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
-  GST_BUFFER_PTS (inbuf) = GST_SECOND;
-  gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
-  indata = (gfloat *) map.data;
-  for (i = 0; i < 48000; i++)
-    indata[i] = 1.0;
-  gst_buffer_unmap (inbuf, &map);
-  fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
-  g_mutex_lock (&data_mutex);
-  /* 48000 samples per buffer * 2 sources * 2 buffers */
-  while (have_data != 48000 * 2 * 2 * sizeof (float))
-    g_cond_wait (&data_cond, &data_mutex);
-  g_mutex_unlock (&data_mutex);
-
-  gst_bus_set_flushing (bus, TRUE);
-  gst_element_set_state (interleave, GST_STATE_NULL);
-  gst_element_set_state (queue, GST_STATE_NULL);
-
-  gst_object_unref (mysrcpads[0]);
-  gst_object_unref (mysrcpads[1]);
-  gst_object_unref (mysinkpad);
-
-  gst_element_release_request_pad (interleave, sink0);
-  gst_object_unref (sink0);
-  gst_element_release_request_pad (interleave, sink1);
-  gst_object_unref (sink1);
-
-  gst_object_unref (interleave);
-  gst_object_unref (queue);
-  gst_object_unref (bus);
-  gst_caps_unref (caps);
-
-  g_free (mysrcpads);
-}
-
-GST_END_TEST;
-
-static void
-src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
-    gboolean interleaved, gpointer user_data)
-{
-  gint n = GPOINTER_TO_INT (user_data);
-  gfloat *data;
-  gint i, num_samples;
-  GstCaps *caps;
-  guint64 mask;
-  GstAudioChannelPosition pos;
-  GstMapInfo map;
-
-  fail_unless (gst_buffer_is_writable (buffer));
-
-  switch (n) {
-    case 0:
-    case 1:
-    case 2:
-      pos = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT;
-      break;
-    case 3:
-      pos = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT;
-      break;
-    default:
-      pos = GST_AUDIO_CHANNEL_POSITION_INVALID;
-      break;
-  }
-
-  mask = G_GUINT64_CONSTANT (1) << pos;
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-      "channels", G_TYPE_INT, 1,
-      "layout", G_TYPE_STRING, interleaved ? "interleaved" : "non-interleaved",
-      "channel-mask", GST_TYPE_BITMASK, mask, "rate", G_TYPE_INT, 48000, NULL);
-
-  gst_pad_set_caps (pad, caps);
-  gst_caps_unref (caps);
-
-  fail_unless (gst_buffer_map (buffer, &map, GST_MAP_WRITE));
-  fail_unless (map.size % sizeof (gfloat) == 0);
-
-  fail_unless (map.size > 480);
-
-  num_samples = map.size / sizeof (gfloat);
-  data = (gfloat *) map.data;
-
-  for (i = 0; i < num_samples; i++)
-    data[i] = (n % 2 == 0) ? -1.0 : 1.0;
-
-  gst_buffer_unmap (buffer, &map);
-}
-
-static void
-src_handoff_float32_audiointerleaved (GstElement * element, GstBuffer * buffer,
-    GstPad * pad, gpointer user_data)
-{
-  src_handoff_float32 (element, buffer, pad, TRUE, user_data);
-}
-
-static void
-src_handoff_float32_non_audiointerleaved (GstElement * element,
-    GstBuffer * buffer, GstPad * pad, gpointer user_data)
-{
-  src_handoff_float32 (element, buffer, pad, FALSE, user_data);
-}
-
-static void
-sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
-    gpointer user_data)
-{
-  gint i;
-  GstMapInfo map;
-  gfloat *data;
-  GstCaps *caps, *ccaps;
-  gint n = GPOINTER_TO_INT (user_data);
-  guint64 mask;
-
-  fail_unless (GST_IS_BUFFER (buffer));
-  gst_buffer_map (buffer, &map, GST_MAP_READ);
-  data = (gfloat *) map.data;
-
-  /* Give a little leeway for rounding errors */
-  fail_unless (gst_util_uint64_scale (map.size, GST_SECOND,
-          48000 * 2 * sizeof (gfloat)) <= GST_BUFFER_DURATION (buffer) + 1 ||
-      gst_util_uint64_scale (map.size, GST_SECOND,
-          48000 * 2 * sizeof (gfloat)) >= GST_BUFFER_DURATION (buffer) - 1);
-
-  if (n == 0 || n == 3) {
-    GstAudioChannelPosition pos[2] =
-        { GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
-    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
-  } else if (n == 1) {
-    GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
-      GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
-    };
-    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
-  } else if (n == 2) {
-    GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
-      GST_AUDIO_CHANNEL_POSITION_REAR_CENTER
-    };
-    gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask);
-  } else {
-    g_assert_not_reached ();
-  }
-
-  if (pad) {
-    caps = gst_caps_new_simple ("audio/x-raw",
-        "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-        "channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000,
-        "layout", G_TYPE_STRING, "interleaved",
-        "channel-mask", GST_TYPE_BITMASK, mask, NULL);
-
-    ccaps = gst_pad_get_current_caps (pad);
-    fail_unless (gst_caps_is_equal (caps, ccaps));
-    gst_caps_unref (ccaps);
-    gst_caps_unref (caps);
-  }
-#ifdef HAVE_VALGRIND
-  if (!(RUNNING_ON_VALGRIND))
-#endif
-    for (i = 0; i < map.size / sizeof (float); i += 2) {
-      fail_unless_equals_float (data[i], -1.0);
-      if (n != 3)
-        fail_unless_equals_float (data[i + 1], 1.0);
-    }
-  have_data += map.size;
-
-  gst_buffer_unmap (buffer, &map);
-
-}
-
-static void
-test_audiointerleave_2ch_pipeline (gboolean interleaved)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-  void *src_handoff_float32 =
-      interleaved ? &src_handoff_float32_audiointerleaved :
-      &src_handoff_float32_non_audiointerleaved;
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
-      GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
-      GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_audiointerleaved)
-{
-  test_audiointerleave_2ch_pipeline (TRUE);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_non_audiointerleaved)
-{
-  test_audiointerleave_2ch_pipeline (FALSE);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (3));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  g_object_set (interleave, "channel-positions-from-input", TRUE, NULL);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-  GValueArray *arr;
-  GValue val = { 0, };
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  g_object_set (interleave, "channel-positions-from-input", FALSE, NULL);
-  arr = g_value_array_new (2);
-
-  g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
-  g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER);
-  g_value_array_append (arr, &val);
-  g_value_reset (&val);
-  g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER);
-  g_value_array_append (arr, &val);
-  g_value_unset (&val);
-
-  g_object_set (interleave, "channel-positions", arr, NULL);
-  g_value_array_free (arr);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (2));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos)
-{
-  GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-  GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-  GstMessage *msg;
-
-  have_data = 0;
-
-  pipeline = (GstElement *) gst_pipeline_new ("pipeline");
-  fail_unless (pipeline != NULL);
-
-  src1 = gst_element_factory_make ("fakesrc", "src1");
-  fail_unless (src1 != NULL);
-  g_object_set (src1, "num-buffers", 4, NULL);
-  g_object_set (src1, "signal-handoffs", TRUE, NULL);
-  g_object_set (src1, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src1, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src1, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), src1);
-
-  src2 = gst_element_factory_make ("fakesrc", "src2");
-  fail_unless (src2 != NULL);
-  g_object_set (src2, "num-buffers", 4, NULL);
-  g_object_set (src2, "signal-handoffs", TRUE, NULL);
-  g_object_set (src2, "sizetype", 2,
-      "sizemax", (int) 48000 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_object_set (src2, "format", GST_FORMAT_TIME, NULL);
-  g_signal_connect (src2, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1));
-  gst_bin_add (GST_BIN (pipeline), src2);
-
-  queue = gst_element_factory_make ("queue", "queue");
-  fail_unless (queue != NULL);
-  gst_bin_add (GST_BIN (pipeline), queue);
-
-  interleave = gst_element_factory_make ("audiointerleave", "audiointerleave");
-  fail_unless (interleave != NULL);
-  g_object_set (interleave, "channel-positions-from-input", FALSE, NULL);
-  gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
-  sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad0 != NULL);
-  tmp = gst_element_get_static_pad (src1, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u");
-  fail_unless (sinkpad1 != NULL);
-  tmp = gst_element_get_static_pad (src2, "src");
-  tmp2 = gst_element_get_static_pad (queue, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-  tmp = gst_element_get_static_pad (queue, "src");
-  fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  fail_unless (sink != NULL);
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
-      GINT_TO_POINTER (0));
-  gst_bin_add (GST_BIN (pipeline), sink);
-  tmp = gst_element_get_static_pad (interleave, "src");
-  tmp2 = gst_element_get_static_pad (sink, "sink");
-  fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
-  gst_object_unref (tmp);
-  gst_object_unref (tmp2);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
-  gst_message_unref (msg);
-
-  /* 48000 samples per buffer * 2 sources * 4 buffers */
-  fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat));
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_element_release_request_pad (interleave, sinkpad0);
-  gst_object_unref (sinkpad0);
-  gst_element_release_request_pad (interleave, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (interleave);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-forward_check_event (GstHarness * h, GstHarness * hsrc, GstEventType type)
-{
-  GstEvent *e;
-
-  e = gst_harness_pull_event (hsrc);
-  fail_unless (GST_EVENT_TYPE (e) == type);
-  gst_harness_push_event (h, e);
-}
-
-GST_START_TEST (test_audiointerleave_2ch_smallbuf)
-{
-  GstElement *audiointerleave;
-  GstHarness *hsrc;
-  GstHarness *h;
-  GstHarness *h2;
-  GstBuffer *buffer;
-  gint i;
-  GstEvent *ev;
-  GstCaps *ecaps, *caps;
-
-  audiointerleave = gst_element_factory_make ("audiointerleave", NULL);
-
-  g_object_set (audiointerleave, "latency", GST_SECOND / 2,
-      "output-buffer-duration", GST_SECOND / 4, NULL);
-
-  h = gst_harness_new_with_element (audiointerleave, "sink_0", "src");
-  gst_harness_use_testclock (h);
-
-  h2 = gst_harness_new_with_element (audiointerleave, "sink_1", NULL);
-  gst_harness_set_src_caps_str (h2, "audio/x-raw, "
-      "format=" GST_AUDIO_NE (F32) ", channels=(int)1,"
-      " layout=interleaved, rate=48000, channel-mask=(bitmask)8");
-
-  hsrc = gst_harness_new ("fakesrc");
-  gst_harness_use_testclock (hsrc);
-  g_object_set (hsrc->element,
-      "is-live", TRUE,
-      "sync", TRUE,
-      "signal-handoffs", TRUE,
-      "format", GST_FORMAT_TIME,
-      "sizetype", 2,
-      "sizemax", (int) 480 * sizeof (gfloat),
-      "datarate", (int) 48000 * sizeof (gfloat), NULL);
-  g_signal_connect (hsrc->element, "handoff",
-      G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2));
-  gst_harness_play (hsrc);
-
-  gst_harness_crank_single_clock_wait (hsrc);
-  forward_check_event (h, hsrc, GST_EVENT_STREAM_START);
-  forward_check_event (h, hsrc, GST_EVENT_CAPS);
-  forward_check_event (h, hsrc, GST_EVENT_SEGMENT);
-  gst_harness_push (h, gst_harness_pull (hsrc));        /* buffer */
-
-  for (i = 0; i < 24; i++) {
-    gst_harness_crank_single_clock_wait (hsrc);
-    forward_check_event (h, hsrc, GST_EVENT_CAPS);
-    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
-  }
-
-  gst_harness_crank_single_clock_wait (h);
-
-
-  gst_event_unref (gst_harness_pull_event (h)); /* stream-start */
-  ev = gst_harness_pull_event (h);      /* caps */
-  fail_unless_equals_int (GST_EVENT_CAPS, GST_EVENT_TYPE (ev));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-      "channels", G_TYPE_INT, 2,
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 48000, "channel-mask", GST_TYPE_BITMASK,
-      (guint64) 0x9, NULL);
-
-  gst_event_parse_caps (ev, &ecaps);
-  gst_check_caps_equal (ecaps, caps);
-  gst_caps_unref (caps);
-  gst_event_unref (ev);
-
-  /* eat the caps processing */
-  gst_harness_crank_single_clock_wait (h);
-  for (i = 0; i < 23; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 750 * GST_MSECOND);
-
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-  fail_unless_equals_int (gst_harness_buffers_received (h), 1);
-
-  for (i = 0; i < 50; i++) {
-    gst_harness_crank_single_clock_wait (hsrc);
-    forward_check_event (h, hsrc, GST_EVENT_CAPS);
-    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
-  }
-  for (i = 0; i < 25; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 1000 * GST_MSECOND);
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-  fail_unless_equals_int (gst_harness_buffers_received (h), 2);
-
-  for (i = 0; i < 25; i++) {
-    gst_harness_crank_single_clock_wait (hsrc);
-    forward_check_event (h, hsrc, GST_EVENT_CAPS);
-    gst_harness_push (h, gst_harness_pull (hsrc));      /* buffer */
-  }
-  for (i = 0; i < 25; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 1250 * GST_MSECOND);
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-  fail_unless_equals_int (gst_harness_buffers_received (h), 3);
-
-  gst_harness_push_event (h, gst_event_new_eos ());
-
-  for (i = 0; i < 25; i++)
-    gst_harness_crank_single_clock_wait (h);
-  fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
-          (h->element)), 1500 * GST_MSECOND);
-  buffer = gst_harness_pull (h);
-  sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3));
-  gst_buffer_unref (buffer);
-
-  fail_unless_equals_int (gst_harness_buffers_received (h), 4);
-
-  gst_harness_teardown (h2);
-  gst_harness_teardown (h);
-  gst_harness_teardown (hsrc);
-  gst_object_unref (audiointerleave);
-}
-
-GST_END_TEST;
-
-static Suite *
-audiointerleave_suite (void)
-{
-  Suite *s = suite_create ("audiointerleave");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_set_timeout (tc_chain, 180);
-  tcase_add_test (tc_chain, test_create_and_unref);
-  tcase_add_test (tc_chain, test_request_pads);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_1eos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_audiointerleaved);
-  tcase_add_test (tc_chain,
-      test_audiointerleave_2ch_pipeline_non_audiointerleaved);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_input_chanpos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_custom_chanpos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_no_chanpos);
-  tcase_add_test (tc_chain, test_audiointerleave_2ch_smallbuf);
-
-  return s;
-}
-
-GST_CHECK_MAIN (audiointerleave);
diff --git a/tests/check/elements/audiomixer.c b/tests/check/elements/audiomixer.c
deleted file mode 100644
index 4a8a8233b..000000000
--- a/tests/check/elements/audiomixer.c
+++ /dev/null
@@ -1,1894 +0,0 @@
-/* GStreamer
- *
- * unit test for audiomixer
- *
- * Copyright (C) 2005 Thomas Vander Stichele <thomas at apestaart dot org>
- * Copyright (C) 2013 Sebastian Dröge <sebastian@centricular.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
-
-#ifdef HAVE_VALGRIND
-# include <valgrind/valgrind.h>
-#endif
-
-#include <unistd.h>
-
-#include <gst/check/gstcheck.h>
-#include <gst/check/gstconsistencychecker.h>
-#include <gst/audio/audio.h>
-#include <gst/base/gstbasesrc.h>
-#include <gst/controller/gstdirectcontrolbinding.h>
-#include <gst/controller/gstinterpolationcontrolsource.h>
-
-static GMainLoop *main_loop;
-
-/* fixtures */
-
-static void
-test_setup (void)
-{
-  main_loop = g_main_loop_new (NULL, FALSE);
-}
-
-static void
-test_teardown (void)
-{
-  g_main_loop_unref (main_loop);
-  main_loop = NULL;
-}
-
-
-/* some test helpers */
-
-static GstElement *
-setup_pipeline (GstElement * audiomixer, gint num_srcs, GstElement * capsfilter)
-{
-  GstElement *pipeline, *src, *sink;
-  gint i;
-
-  pipeline = gst_pipeline_new ("pipeline");
-  if (!audiomixer) {
-    audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  }
-
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (pipeline), audiomixer, sink, NULL);
-
-  if (capsfilter) {
-    gst_bin_add (GST_BIN (pipeline), capsfilter);
-    gst_element_link_many (audiomixer, capsfilter, sink, NULL);
-  } else {
-    gst_element_link (audiomixer, sink);
-  }
-
-  for (i = 0; i < num_srcs; i++) {
-    src = gst_element_factory_make ("audiotestsrc", NULL);
-    g_object_set (src, "wave", 4, NULL);        /* silence */
-    gst_bin_add (GST_BIN (pipeline), src);
-    gst_element_link (src, audiomixer);
-  }
-  return pipeline;
-}
-
-static GstCaps *
-get_element_sink_pad_caps (GstElement * pipeline, const gchar * element_name)
-{
-  GstElement *sink;
-  GstCaps *caps;
-  GstPad *pad;
-
-  sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink");
-  pad = gst_element_get_static_pad (sink, "sink");
-  caps = gst_pad_get_current_caps (pad);
-  gst_object_unref (pad);
-  gst_object_unref (sink);
-
-  return caps;
-}
-
-static void
-set_state_and_wait (GstElement * pipeline, GstState state)
-{
-  GstStateChangeReturn state_res;
-
-  /* prepare paused/playing */
-  state_res = gst_element_set_state (pipeline, state);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* wait for preroll */
-  state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-}
-
-static gboolean
-set_playing (GstElement * element)
-{
-  GstStateChangeReturn state_res;
-
-  state_res = gst_element_set_state (element, GST_STATE_PLAYING);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  return FALSE;
-}
-
-static void
-play_and_wait (GstElement * pipeline)
-{
-  GstStateChangeReturn state_res;
-
-  g_idle_add ((GSourceFunc) set_playing, pipeline);
-
-  GST_INFO ("running main loop");
-  g_main_loop_run (main_loop);
-
-  state_res = gst_element_set_state (pipeline, GST_STATE_NULL);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-}
-
-static void
-message_received (GstBus * bus, GstMessage * message, GstPipeline * bin)
-{
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  switch (message->type) {
-    case GST_MESSAGE_EOS:
-      g_main_loop_quit (main_loop);
-      break;
-    case GST_MESSAGE_WARNING:{
-      GError *gerror;
-      gchar *debug;
-
-      gst_message_parse_warning (message, &gerror, &debug);
-      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
-      g_error_free (gerror);
-      g_free (debug);
-      break;
-    }
-    case GST_MESSAGE_ERROR:{
-      GError *gerror;
-      gchar *debug;
-
-      gst_message_parse_error (message, &gerror, &debug);
-      gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
-      g_error_free (gerror);
-      g_free (debug);
-      g_main_loop_quit (main_loop);
-      break;
-    }
-    default:
-      break;
-  }
-}
-
-static GstBuffer *
-new_buffer (gsize num_bytes, gint data, GstClockTime ts, GstClockTime dur,
-    GstBufferFlags flags)
-{
-  GstMapInfo map;
-  GstBuffer *buffer = gst_buffer_new_and_alloc (num_bytes);
-
-  gst_buffer_map (buffer, &map, GST_MAP_WRITE);
-  memset (map.data, data, map.size);
-  gst_buffer_unmap (buffer, &map);
-  GST_BUFFER_TIMESTAMP (buffer) = ts;
-  GST_BUFFER_DURATION (buffer) = dur;
-  if (flags)
-    GST_BUFFER_FLAG_SET (buffer, flags);
-  GST_DEBUG ("created buffer %p", buffer);
-  return buffer;
-}
-
-/* make sure downstream gets a CAPS event before buffers are sent */
-GST_START_TEST (test_caps)
-{
-  GstElement *pipeline;
-  GstCaps *caps;
-
-  /* build pipeline */
-  pipeline = setup_pipeline (NULL, 1, NULL);
-
-  /* prepare playing */
-  set_state_and_wait (pipeline, GST_STATE_PAUSED);
-
-  /* check caps on fakesink */
-  caps = get_element_sink_pad_caps (pipeline, "sink");
-  fail_unless (caps != NULL);
-  gst_caps_unref (caps);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-/* check that caps set on the property are honoured */
-GST_START_TEST (test_filter_caps)
-{
-  GstElement *pipeline, *audiomixer, *capsfilter;
-  GstCaps *filter_caps, *caps;
-
-  filter_caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (F32),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1,
-      "channel-mask", GST_TYPE_BITMASK, (guint64) 0x04, NULL);
-
-  capsfilter = gst_element_factory_make ("capsfilter", NULL);
-
-  /* build pipeline */
-  audiomixer = gst_element_factory_make ("audiomixer", NULL);
-  g_object_set (capsfilter, "caps", filter_caps, NULL);
-  pipeline = setup_pipeline (audiomixer, 1, capsfilter);
-
-  /* prepare playing */
-  set_state_and_wait (pipeline, GST_STATE_PAUSED);
-
-  /* check caps on fakesink */
-  caps = get_element_sink_pad_caps (pipeline, "sink");
-  fail_unless (caps != NULL);
-  GST_INFO_OBJECT (pipeline, "received caps: %" GST_PTR_FORMAT, caps);
-  fail_unless (gst_caps_is_equal_fixed (caps, filter_caps));
-  gst_caps_unref (caps);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-
-  gst_caps_unref (filter_caps);
-}
-
-GST_END_TEST;
-
-static GstFormat format = GST_FORMAT_UNDEFINED;
-static gint64 position = -1;
-
-static void
-test_event_message_received (GstBus * bus, GstMessage * message,
-    GstPipeline * bin)
-{
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  switch (message->type) {
-    case GST_MESSAGE_SEGMENT_DONE:
-      gst_message_parse_segment_done (message, &format, &position);
-      GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position);
-      g_main_loop_quit (main_loop);
-      break;
-    default:
-      g_assert_not_reached ();
-      break;
-  }
-}
-
-
-GST_START_TEST (test_event)
-{
-  GstElement *bin, *src1, *src2, *audiomixer, *sink;
-  GstBus *bus;
-  GstEvent *seek_event;
-  gboolean res;
-  GstPad *srcpad, *sinkpad;
-  GstStreamConsistency *chk_1, *chk_2, *chk_3;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "wave", 4, NULL); /* silence */
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "wave", 4, NULL); /* silence */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL);
-
-  res = gst_element_link (src1, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (src2, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  chk_3 = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  /* create consistency checkers for the pads */
-  srcpad = gst_element_get_static_pad (src1, "src");
-  chk_1 = gst_consistency_checker_new (srcpad);
-  sinkpad = gst_pad_get_peer (srcpad);
-  gst_consistency_checker_add_pad (chk_3, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  srcpad = gst_element_get_static_pad (src2, "src");
-  chk_2 = gst_consistency_checker_new (srcpad);
-  sinkpad = gst_pad_get_peer (srcpad);
-  gst_consistency_checker_add_pad (chk_3, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  format = GST_FORMAT_UNDEFINED;
-  position = -1;
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) test_event_message_received, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  res = gst_element_send_event (bin, seek_event);
-  fail_unless (res == TRUE, NULL);
-
-  /* run pipeline */
-  play_and_wait (bin);
-
-  ck_assert_int_eq (position, 2 * GST_SECOND);
-
-  /* cleanup */
-  gst_consistency_checker_free (chk_1);
-  gst_consistency_checker_free (chk_2);
-  gst_consistency_checker_free (chk_3);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-static guint play_count = 0;
-static GstEvent *play_seek_event = NULL;
-
-static void
-test_play_twice_message_received (GstBus * bus, GstMessage * message,
-    GstElement * bin)
-{
-  gboolean res;
-  GstStateChangeReturn state_res;
-
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  switch (message->type) {
-    case GST_MESSAGE_SEGMENT_DONE:
-      play_count++;
-      if (play_count == 1) {
-        state_res = gst_element_set_state (bin, GST_STATE_READY);
-        ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-        /* prepare playing again */
-        set_state_and_wait (bin, GST_STATE_PAUSED);
-
-        res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-        fail_unless (res == TRUE, NULL);
-
-        state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
-        ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-      } else {
-        g_main_loop_quit (main_loop);
-      }
-      break;
-    default:
-      g_assert_not_reached ();
-      break;
-  }
-}
-
-
-GST_START_TEST (test_play_twice)
-{
-  GstElement *bin, *audiomixer;
-  GstBus *bus;
-  gboolean res;
-  GstPad *srcpad;
-  GstStreamConsistency *consist;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  bin = setup_pipeline (audiomixer, 2, NULL);
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  consist = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  play_count = 0;
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) test_play_twice_message_received, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-  fail_unless (res == TRUE, NULL);
-
-  GST_INFO ("seeked");
-
-  /* run pipeline */
-  play_and_wait (bin);
-
-  ck_assert_int_eq (play_count, 2);
-
-  /* cleanup */
-  gst_consistency_checker_free (consist);
-  gst_event_unref (play_seek_event);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_play_twice_then_add_and_play_again)
-{
-  GstElement *bin, *src, *audiomixer;
-  GstBus *bus;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  gint i;
-  GstPad *srcpad;
-  GstStreamConsistency *consist;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  bin = setup_pipeline (audiomixer, 2, NULL);
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  consist = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) test_play_twice_message_received, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  /* run it twice */
-  for (i = 0; i < 2; i++) {
-    play_count = 0;
-
-    GST_INFO ("starting test-loop %d", i);
-
-    /* prepare playing */
-    set_state_and_wait (bin, GST_STATE_PAUSED);
-
-    res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-    fail_unless (res == TRUE, NULL);
-
-    GST_INFO ("seeked");
-
-    /* run pipeline */
-    play_and_wait (bin);
-
-    ck_assert_int_eq (play_count, 2);
-
-    /* plug another source */
-    if (i == 0) {
-      src = gst_element_factory_make ("audiotestsrc", NULL);
-      g_object_set (src, "wave", 4, NULL);      /* silence */
-      gst_bin_add (GST_BIN (bin), src);
-
-      res = gst_element_link (src, audiomixer);
-      fail_unless (res == TRUE, NULL);
-    }
-
-    gst_consistency_checker_reset (consist);
-  }
-
-  state_res = gst_element_set_state (bin, GST_STATE_NULL);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* cleanup */
-  gst_event_unref (play_seek_event);
-  gst_consistency_checker_free (consist);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-
-static GstElement *
-test_live_seeking_try_audiosrc (const gchar * factory_name)
-{
-  GstElement *src;
-  GstStateChangeReturn state_res;
-
-  if (!(src = gst_element_factory_make (factory_name, NULL))) {
-    GST_INFO ("can't make '%s', skipping", factory_name);
-    return NULL;
-  }
-
-  /* Test that the audio source can get to ready, else skip */
-  state_res = gst_element_set_state (src, GST_STATE_READY);
-  gst_element_set_state (src, GST_STATE_NULL);
-
-  if (state_res == GST_STATE_CHANGE_FAILURE) {
-    GST_INFO_OBJECT (src, "can't go to ready, skipping");
-    gst_object_unref (src);
-    return NULL;
-  }
-
-  return src;
-}
-
-/* test failing seeks on live-sources */
-GST_START_TEST (test_live_seeking)
-{
-  GstElement *bin, *src1 = NULL, *cf, *src2, *audiomixer, *sink;
-  GstCaps *caps;
-  GstBus *bus;
-  gboolean res;
-  GstPad *srcpad;
-  GstPad *sinkpad;
-  gint i;
-  GstStreamConsistency *consist;
-  /* don't use autoaudiosrc, as then we can't set anything here */
-  const gchar *audio_src_factories[] = {
-    "alsasrc",
-    "pulseaudiosrc"
-  };
-
-  GST_INFO ("preparing test");
-  play_seek_event = NULL;
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  for (i = 0; (i < G_N_ELEMENTS (audio_src_factories) && src1 == NULL); i++) {
-    src1 = test_live_seeking_try_audiosrc (audio_src_factories[i]);
-  }
-  if (!src1) {
-    /* normal audiosources behave differently than audiotestsrc */
-    GST_WARNING ("no real audiosrc found, using audiotestsrc is-live");
-    src1 = gst_element_factory_make ("audiotestsrc", "src1");
-    g_object_set (src1, "wave", 4, "is-live", TRUE, NULL);      /* silence */
-  } else {
-    /* live sources ignore seeks, force eos after 2 sec (4 buffers half second
-     * each)
-     */
-    g_object_set (src1, "num-buffers", 4, "blocksize", 44100, NULL);
-  }
-
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  cf = gst_element_factory_make ("capsfilter", "capsfilter");
-  sink = gst_element_factory_make ("fakesink", "sink");
-
-  gst_bin_add_many (GST_BIN (bin), src1, cf, audiomixer, sink, NULL);
-  res = gst_element_link_many (src1, cf, audiomixer, sink, NULL);
-  fail_unless (res == TRUE, NULL);
-
-  /* get the caps for the livesrc, we'll reuse this for the non-live source */
-  set_state_and_wait (bin, GST_STATE_PLAYING);
-
-  sinkpad = gst_element_get_static_pad (sink, "sink");
-  fail_unless (sinkpad != NULL);
-  caps = gst_pad_get_current_caps (sinkpad);
-  fail_unless (caps != NULL);
-  gst_object_unref (sinkpad);
-
-  gst_element_set_state (bin, GST_STATE_NULL);
-
-  g_object_set (cf, "caps", caps, NULL);
-
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "wave", 4, NULL); /* silence */
-  gst_bin_add (GST_BIN (bin), src2);
-
-  res = gst_element_link_filtered (src2, audiomixer, caps);
-  fail_unless (res == TRUE, NULL);
-
-  gst_caps_unref (caps);
-
-  play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  consist = gst_consistency_checker_new (srcpad);
-  gst_object_unref (srcpad);
-
-  GST_INFO ("starting test");
-
-  /* run it twice */
-  for (i = 0; i < 2; i++) {
-
-    GST_INFO ("starting test-loop %d", i);
-
-    /* prepare playing */
-    set_state_and_wait (bin, GST_STATE_PAUSED);
-
-    res = gst_element_send_event (bin, gst_event_ref (play_seek_event));
-    fail_unless (res == TRUE, NULL);
-
-    GST_INFO ("seeked");
-
-    /* run pipeline */
-    play_and_wait (bin);
-
-    gst_consistency_checker_reset (consist);
-  }
-
-  /* cleanup */
-  GST_INFO ("cleaning up");
-  gst_consistency_checker_free (consist);
-  if (play_seek_event)
-    gst_event_unref (play_seek_event);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-/* check if adding pads work as expected */
-GST_START_TEST (test_add_pad)
-{
-  GstElement *bin, *src1, *src2, *audiomixer, *sink;
-  GstBus *bus;
-  GstPad *srcpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "num-buffers", 4, "wave", /* silence */ 4, NULL);
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  /* one buffer less, we connect with 1 buffer of delay */
-  g_object_set (src2, "num-buffers", 3, "wave", /* silence */ 4, NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src1, audiomixer, sink, NULL);
-
-  res = gst_element_link (src1, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  gst_object_unref (srcpad);
-
-  g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
-      bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  /* add other element */
-  gst_bin_add_many (GST_BIN (bin), src2, NULL);
-
-  /* now link the second element */
-  res = gst_element_link (src2, audiomixer);
-  fail_unless (res == TRUE, NULL);
-
-  /* set to PAUSED as well */
-  state_res = gst_element_set_state (src2, GST_STATE_PAUSED);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* now play all */
-  play_and_wait (bin);
-
-  /* cleanup */
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-/* check if removing pads work as expected */
-GST_START_TEST (test_remove_pad)
-{
-  GstElement *bin, *src, *audiomixer, *sink;
-  GstBus *bus;
-  GstPad *pad, *srcpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  src = gst_element_factory_make ("audiotestsrc", "src");
-  g_object_set (src, "num-buffers", 4, "wave", 4, NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src, audiomixer, sink, NULL);
-
-  res = gst_element_link (src, audiomixer);
-  fail_unless (res == TRUE, NULL);
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  /* create an unconnected sinkpad in audiomixer */
-  pad = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (pad == NULL, NULL);
-
-  srcpad = gst_element_get_static_pad (audiomixer, "src");
-  gst_object_unref (srcpad);
-
-  g_signal_connect (bus, "message::segment-done", (GCallback) message_received,
-      bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing, this will not preroll as audiomixer is waiting
-   * on the unconnected sinkpad. */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* wait for completion for one second, will return ASYNC */
-  state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND);
-  ck_assert_int_eq (state_res, GST_STATE_CHANGE_ASYNC);
-
-  /* get rid of the pad now, audiomixer should stop waiting on it and
-   * continue the preroll */
-  gst_element_release_request_pad (audiomixer, pad);
-  gst_object_unref (pad);
-
-  /* wait for completion, should work now */
-  state_res =
-      gst_element_get_state (GST_ELEMENT (bin), NULL, NULL,
-      GST_CLOCK_TIME_NONE);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* now play all */
-  play_and_wait (bin);
-
-  /* cleanup */
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (G_OBJECT (bus));
-  gst_object_unref (G_OBJECT (bin));
-}
-
-GST_END_TEST;
-
-
-static GstBuffer *handoff_buffer = NULL;
-
-static void
-handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
-    gpointer user_data)
-{
-  GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_FORMAT
-      " -- %p PTS is %" GST_TIME_FORMAT " END is %" GST_TIME_FORMAT,
-      gst_buffer_get_size (buffer), buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-
-  gst_buffer_replace (&handoff_buffer, buffer);
-}
-
-/* check if clipping works as expected */
-GST_START_TEST (test_clip)
-{
-  GstSegment segment;
-  GstElement *bin, *audiomixer, *sink;
-  GstBus *bus;
-  GstPad *sinkpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  GstFlowReturn ret;
-  GstEvent *event;
-  GstBuffer *buffer;
-  GstCaps *caps;
-  GstQuery *drain = gst_query_new_drain ();
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  /* just an audiomixer and a fakesink */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  g_object_set (audiomixer, "output-buffer-duration", 50 * GST_MSECOND, NULL);
-  sink = gst_element_factory_make ("fakesink", "sink");
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL);
-  gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL);
-
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  /* set to playing */
-  state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* create an unconnected sinkpad in audiomixer, should also automatically activate
-   * the pad */
-  sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad == NULL, NULL);
-
-  gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL);
-
-  gst_pad_set_caps (sinkpad, caps);
-  gst_caps_unref (caps);
-
-  /* send segment to audiomixer */
-  gst_segment_init (&segment, GST_FORMAT_TIME);
-  segment.start = GST_SECOND;
-  segment.stop = 2 * GST_SECOND;
-  segment.time = 0;
-  event = gst_event_new_segment (&segment);
-  gst_pad_send_event (sinkpad, event);
-
-  /* should be clipped and ok */
-  buffer = new_buffer (44100, 0, 0, 250 * GST_MSECOND, 0);
-  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
-      buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  /* The aggregation is done in a dedicated thread, so we can't
-   * know when it is actually going to happen, so we use a DRAIN query
-   * to wait for it to complete.
-   */
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer == NULL);
-
-  /* should be partially clipped */
-  buffer = new_buffer (44100, 0, 900 * GST_MSECOND, 250 * GST_MSECOND,
-      GST_BUFFER_FLAG_DISCONT);
-  GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %"
-      GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
-      GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-
-  fail_unless (handoff_buffer != NULL);
-  ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
-      GST_BUFFER_DURATION (handoff_buffer), 150 * GST_MSECOND);
-  gst_buffer_replace (&handoff_buffer, NULL);
-
-  /* should not be clipped */
-  buffer = new_buffer (44100, 0, 1150 * GST_MSECOND, 250 * GST_MSECOND, 0);
-  GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT,
-      buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer != NULL);
-  ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) +
-      GST_BUFFER_DURATION (handoff_buffer), 400 * GST_MSECOND);
-  gst_buffer_replace (&handoff_buffer, NULL);
-  fail_unless (handoff_buffer == NULL);
-
-  /* should be clipped and ok */
-  buffer = new_buffer (44100, 0, 2 * GST_SECOND, 250 * GST_MSECOND,
-      GST_BUFFER_FLAG_DISCONT);
-  GST_DEBUG ("pushing buffer %p PTS is %" GST_TIME_FORMAT
-      " END is %" GST_TIME_FORMAT,
-      buffer,
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer)),
-      GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer)));
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer == NULL);
-
-  gst_element_release_request_pad (audiomixer, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-  gst_query_unref (drain);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_duration_is_max)
-{
-  GstElement *bin, *src[3], *audiomixer, *sink;
-  GstStateChangeReturn state_res;
-  GstFormat format = GST_FORMAT_TIME;
-  gboolean res;
-  gint64 duration;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-
-  /* 3 sources, an audiomixer and a fakesink */
-  src[0] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[1] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[2] = gst_element_factory_make ("audiotestsrc", NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
-      NULL);
-
-  gst_element_link (src[0], audiomixer);
-  gst_element_link (src[1], audiomixer);
-  gst_element_link (src[2], audiomixer);
-  gst_element_link (audiomixer, sink);
-
-  /* irks, duration is reset on basesrc */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
-
-  /* set durations on src */
-  GST_BASE_SRC (src[0])->segment.duration = 1000;
-  GST_BASE_SRC (src[1])->segment.duration = 3000;
-  GST_BASE_SRC (src[2])->segment.duration = 2000;
-
-  /* set to playing */
-  set_state_and_wait (bin, GST_STATE_PLAYING);
-
-  res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
-  fail_unless (res, NULL);
-
-  ck_assert_int_eq (duration, 3000);
-
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_duration_unknown_overrides)
-{
-  GstElement *bin, *src[3], *audiomixer, *sink;
-  GstStateChangeReturn state_res;
-  GstFormat format = GST_FORMAT_TIME;
-  gboolean res;
-  gint64 duration;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-
-  /* 3 sources, an audiomixer and a fakesink */
-  src[0] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[1] = gst_element_factory_make ("audiotestsrc", NULL);
-  src[2] = gst_element_factory_make ("audiotestsrc", NULL);
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink,
-      NULL);
-
-  gst_element_link (src[0], audiomixer);
-  gst_element_link (src[1], audiomixer);
-  gst_element_link (src[2], audiomixer);
-  gst_element_link (audiomixer, sink);
-
-  /* irks, duration is reset on basesrc */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL);
-
-  /* set durations on src */
-  GST_BASE_SRC (src[0])->segment.duration = GST_CLOCK_TIME_NONE;
-  GST_BASE_SRC (src[1])->segment.duration = 3000;
-  GST_BASE_SRC (src[2])->segment.duration = 2000;
-
-  /* set to playing */
-  set_state_and_wait (bin, GST_STATE_PLAYING);
-
-  res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration);
-  fail_unless (res, NULL);
-
-  ck_assert_int_eq (duration, GST_CLOCK_TIME_NONE);
-
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-
-static gboolean looped = FALSE;
-
-static void
-loop_segment_done (GstBus * bus, GstMessage * message, GstElement * bin)
-{
-  GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT,
-      GST_MESSAGE_SRC (message), message);
-
-  if (looped) {
-    g_main_loop_quit (main_loop);
-  } else {
-    GstEvent *seek_event;
-    gboolean res;
-
-    seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-        GST_SEEK_FLAG_SEGMENT,
-        GST_SEEK_TYPE_SET, (GstClockTime) 0,
-        GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
-
-    res = gst_element_send_event (bin, seek_event);
-    fail_unless (res == TRUE, NULL);
-    looped = TRUE;
-  }
-}
-
-GST_START_TEST (test_loop)
-{
-  GstElement *bin;
-  GstBus *bus;
-  GstEvent *seek_event;
-  gboolean res;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = setup_pipeline (NULL, 2, NULL);
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
-      GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH,
-      GST_SEEK_TYPE_SET, (GstClockTime) 0,
-      GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND);
-
-  g_signal_connect (bus, "message::segment-done",
-      (GCallback) loop_segment_done, bin);
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  GST_INFO ("starting test");
-
-  /* prepare playing */
-  set_state_and_wait (bin, GST_STATE_PAUSED);
-
-  res = gst_element_send_event (bin, seek_event);
-  fail_unless (res == TRUE, NULL);
-
-  /* run pipeline */
-  play_and_wait (bin);
-
-  fail_unless (looped);
-
-  /* cleanup */
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_flush_start_flush_stop)
-{
-  GstPadTemplate *sink_template;
-  GstPad *tmppad, *srcpad1, *sinkpad1, *sinkpad2, *audiomixer_src;
-  GstElement *pipeline, *src1, *src2, *audiomixer, *sink;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  pipeline = gst_pipeline_new ("pipeline");
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "wave", 4, NULL); /* silence */
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "wave", 4, NULL); /* silence */
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  gst_bin_add_many (GST_BIN (pipeline), src1, src2, audiomixer, sink, NULL);
-
-  sink_template =
-      gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (audiomixer),
-      "sink_%u");
-  fail_unless (GST_IS_PAD_TEMPLATE (sink_template));
-  sinkpad1 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
-  srcpad1 = gst_element_get_static_pad (src1, "src");
-  gst_pad_link (srcpad1, sinkpad1);
-
-  sinkpad2 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL);
-  tmppad = gst_element_get_static_pad (src2, "src");
-  gst_pad_link (tmppad, sinkpad2);
-  gst_object_unref (tmppad);
-
-  gst_element_link (audiomixer, sink);
-
-  /* prepare playing */
-  set_state_and_wait (pipeline, GST_STATE_PLAYING);
-
-  audiomixer_src = gst_element_get_static_pad (audiomixer, "src");
-  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
-  gst_pad_send_event (sinkpad1, gst_event_new_flush_start ());
-  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
-  fail_unless (GST_PAD_IS_FLUSHING (sinkpad1));
-  /* Hold the streamlock to make sure the flush stop is not between
-     the attempted push of a segment event and of the following buffer. */
-  GST_PAD_STREAM_LOCK (srcpad1);
-  gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE));
-  GST_PAD_STREAM_UNLOCK (srcpad1);
-  fail_if (GST_PAD_IS_FLUSHING (audiomixer_src));
-  fail_if (GST_PAD_IS_FLUSHING (sinkpad1));
-  gst_object_unref (audiomixer_src);
-
-  gst_element_release_request_pad (audiomixer, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_element_release_request_pad (audiomixer, sinkpad2);
-  gst_object_unref (sinkpad2);
-  gst_object_unref (srcpad1);
-
-  /* cleanup */
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer,
-    GstPad * pad, gpointer user_data)
-{
-  GList **received_buffers = user_data;
-
-  GST_DEBUG ("got buffer %p", buffer);
-  *received_buffers =
-      g_list_append (*received_buffers, gst_buffer_ref (buffer));
-}
-
-typedef void (*SendBuffersFunction) (GstPad * pad1, GstPad * pad2);
-typedef void (*CheckBuffersFunction) (GList * buffers);
-
-static void
-run_sync_test (SendBuffersFunction send_buffers,
-    CheckBuffersFunction check_buffers)
-{
-  GstSegment segment;
-  GstElement *bin, *audiomixer, *queue1, *queue2, *sink;
-  GstBus *bus;
-  GstPad *sinkpad1, *sinkpad2;
-  GstPad *queue1_sinkpad, *queue2_sinkpad;
-  GstPad *pad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  GstEvent *event;
-  GstCaps *caps;
-  GList *received_buffers = NULL;
-
-  GST_INFO ("preparing test");
-
-  /* build pipeline */
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  /* just an audiomixer and a fakesink */
-  queue1 = gst_element_factory_make ("queue", "queue1");
-  queue2 = gst_element_factory_make ("queue", "queue2");
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  g_object_set (audiomixer, "output-buffer-duration", 500 * GST_MSECOND, NULL);
-  sink = gst_element_factory_make ("fakesink", "sink");
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb,
-      &received_buffers);
-  gst_bin_add_many (GST_BIN (bin), queue1, queue2, audiomixer, sink, NULL);
-
-  res = gst_element_link (audiomixer, sink);
-  fail_unless (res == TRUE, NULL);
-
-  /* set to paused */
-  state_res = gst_element_set_state (bin, GST_STATE_PAUSED);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  /* create an unconnected sinkpad in audiomixer, should also automatically activate
-   * the pad */
-  sinkpad1 = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad1 == NULL, NULL);
-
-  queue1_sinkpad = gst_element_get_static_pad (queue1, "sink");
-  pad = gst_element_get_static_pad (queue1, "src");
-  fail_unless (gst_pad_link (pad, sinkpad1) == GST_PAD_LINK_OK);
-  gst_object_unref (pad);
-
-  sinkpad2 = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad2 == NULL, NULL);
-
-  queue2_sinkpad = gst_element_get_static_pad (queue2, "sink");
-  pad = gst_element_get_static_pad (queue2, "src");
-  fail_unless (gst_pad_link (pad, sinkpad2) == GST_PAD_LINK_OK);
-  gst_object_unref (pad);
-
-  gst_pad_send_event (queue1_sinkpad, gst_event_new_stream_start ("test"));
-  gst_pad_send_event (queue2_sinkpad, gst_event_new_stream_start ("test"));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL);
-
-  gst_pad_set_caps (queue1_sinkpad, caps);
-  gst_pad_set_caps (queue2_sinkpad, caps);
-  gst_caps_unref (caps);
-
-  /* send segment to audiomixer */
-  gst_segment_init (&segment, GST_FORMAT_TIME);
-  event = gst_event_new_segment (&segment);
-  gst_pad_send_event (queue1_sinkpad, gst_event_ref (event));
-  gst_pad_send_event (queue2_sinkpad, event);
-
-  /* Push buffers */
-  send_buffers (queue1_sinkpad, queue2_sinkpad);
-
-  /* Set PLAYING */
-  g_idle_add ((GSourceFunc) set_playing, bin);
-
-  /* Collect buffers and messages */
-  g_main_loop_run (main_loop);
-
-  /* Here we get once we got EOS, for errors we failed */
-
-  check_buffers (received_buffers);
-
-  g_list_free_full (received_buffers, (GDestroyNotify) gst_buffer_unref);
-
-  gst_element_release_request_pad (audiomixer, sinkpad1);
-  gst_object_unref (sinkpad1);
-  gst_object_unref (queue1_sinkpad);
-  gst_element_release_request_pad (audiomixer, sinkpad2);
-  gst_object_unref (sinkpad2);
-  gst_object_unref (queue2_sinkpad);
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-}
-
-static void
-send_buffers_sync (GstPad * pad1, GstPad * pad2)
-{
-  GstBuffer *buffer;
-  GstFlowReturn ret;
-
-  buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 1, 2 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad1, gst_event_new_eos ());
-
-  buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad2, gst_event_new_eos ());
-}
-
-static void
-check_buffers_sync (GList * received_buffers)
-{
-  GstBuffer *buffer;
-  GList *l;
-  gint i;
-  GstMapInfo map;
-
-  /* Should have 8 * 0.5s buffers */
-  fail_unless_equals_int (g_list_length (received_buffers), 8);
-  for (i = 0, l = received_buffers; l; l = l->next, i++) {
-    buffer = l->data;
-
-    gst_buffer_map (buffer, &map, GST_MAP_READ);
-
-    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else {
-      g_assert_not_reached ();
-    }
-
-    gst_buffer_unmap (buffer, &map);
-
-  }
-}
-
-GST_START_TEST (test_sync)
-{
-  run_sync_test (send_buffers_sync, check_buffers_sync);
-}
-
-GST_END_TEST;
-
-static void
-send_buffers_sync_discont (GstPad * pad1, GstPad * pad2)
-{
-  GstBuffer *buffer;
-  GstFlowReturn ret;
-
-  buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 1, 3 * GST_SECOND, 1 * GST_SECOND,
-      GST_BUFFER_FLAG_DISCONT);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad1, gst_event_new_eos ());
-
-  buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad2, gst_event_new_eos ());
-}
-
-static void
-check_buffers_sync_discont (GList * received_buffers)
-{
-  GstBuffer *buffer;
-  GList *l;
-  gint i;
-  GstMapInfo map;
-
-  /* Should have 8 * 0.5s buffers */
-  fail_unless_equals_int (g_list_length (received_buffers), 8);
-  for (i = 0, l = received_buffers; l; l = l->next, i++) {
-    buffer = l->data;
-
-    gst_buffer_map (buffer, &map, GST_MAP_READ);
-
-    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else {
-      g_assert_not_reached ();
-    }
-
-    gst_buffer_unmap (buffer, &map);
-
-  }
-}
-
-GST_START_TEST (test_sync_discont)
-{
-  run_sync_test (send_buffers_sync_discont, check_buffers_sync_discont);
-}
-
-GST_END_TEST;
-
-static void
-send_buffers_sync_unaligned (GstPad * pad1, GstPad * pad2)
-{
-  GstBuffer *buffer;
-  GstFlowReturn ret;
-
-  buffer = new_buffer (2000, 1, 750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 1, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad1, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad1, gst_event_new_eos ());
-
-  buffer = new_buffer (2000, 2, 1750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  buffer = new_buffer (2000, 2, 2750 * GST_MSECOND, 1 * GST_SECOND, 0);
-  ret = gst_pad_chain (pad2, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-
-  gst_pad_send_event (pad2, gst_event_new_eos ());
-}
-
-static void
-check_buffers_sync_unaligned (GList * received_buffers)
-{
-  GstBuffer *buffer;
-  GList *l;
-  gint i;
-  GstMapInfo map;
-
-  /* Should have 8 * 0.5s buffers */
-  fail_unless_equals_int (g_list_length (received_buffers), 8);
-  for (i = 0, l = received_buffers; l; l = l->next, i++) {
-    buffer = l->data;
-
-    gst_buffer_map (buffer, &map, GST_MAP_READ);
-
-    if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[map.size - 1] == 0);
-    } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 0);
-      fail_unless (map.data[499] == 0);
-      fail_unless (map.data[500] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[map.size - 1] == 1);
-    } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 1);
-      fail_unless (map.data[499] == 1);
-      fail_unless (map.data[500] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[499] == 3);
-      fail_unless (map.data[500] == 3);
-      fail_unless (map.data[map.size - 1] == 3);
-    } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) {
-      fail_unless (map.data[0] == 3);
-      fail_unless (map.data[499] == 3);
-      fail_unless (map.data[500] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) {
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[499] == 2);
-      fail_unless (map.data[500] == 2);
-      fail_unless (map.data[map.size - 1] == 2);
-    } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) {
-      fail_unless (map.size == 500);
-      fail_unless (GST_BUFFER_DURATION (buffer) == 250 * GST_MSECOND);
-      fail_unless (map.data[0] == 2);
-      fail_unless (map.data[499] == 2);
-    } else {
-      g_assert_not_reached ();
-    }
-
-    gst_buffer_unmap (buffer, &map);
-
-  }
-}
-
-GST_START_TEST (test_sync_unaligned)
-{
-  run_sync_test (send_buffers_sync_unaligned, check_buffers_sync_unaligned);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_segment_base_handling)
-{
-  GstElement *pipeline, *sink, *mix, *src1, *src2;
-  GstPad *srcpad, *sinkpad;
-  GstClockTime end_time;
-  GstSample *last_sample = NULL;
-  GstSample *sample;
-  GstBuffer *buf;
-  GstCaps *caps;
-
-  caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100,
-      "channels", G_TYPE_INT, 2, NULL);
-
-  pipeline = gst_pipeline_new ("pipeline");
-  mix = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("appsink", "sink");
-  g_object_set (sink, "caps", caps, "sync", FALSE, NULL);
-  gst_caps_unref (caps);
-  /* 50 buffers of 1/10 sec = 5 sec */
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
-  src2 = gst_element_factory_make ("audiotestsrc", "src2");
-  g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL);
-  gst_bin_add_many (GST_BIN (pipeline), src1, src2, mix, sink, NULL);
-  fail_unless (gst_element_link (mix, sink));
-
-  srcpad = gst_element_get_static_pad (src1, "src");
-  sinkpad = gst_element_get_request_pad (mix, "sink_1");
-  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  srcpad = gst_element_get_static_pad (src2, "src");
-  sinkpad = gst_element_get_request_pad (mix, "sink_2");
-  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
-  /* set a pad offset of another 5 seconds */
-  gst_pad_set_offset (sinkpad, 5 * GST_SECOND);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  do {
-    g_signal_emit_by_name (sink, "pull-sample", &sample);
-    if (sample == NULL)
-      break;
-    if (last_sample)
-      gst_sample_unref (last_sample);
-    last_sample = sample;
-  } while (TRUE);
-
-  buf = gst_sample_get_buffer (last_sample);
-  end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
-  fail_unless_equals_int64 (end_time, 10 * GST_SECOND);
-  gst_sample_unref (last_sample);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value,
-    GstClockTime end, gdouble end_value)
-{
-  GstControlSource *cs;
-  GstTimedValueControlSource *tvcs;
-
-  cs = gst_interpolation_control_source_new ();
-  fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad),
-          gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad),
-              "volume", cs)));
-
-  /* set volume interpolation mode */
-  g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
-
-  tvcs = (GstTimedValueControlSource *) cs;
-  fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value));
-  fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value));
-  gst_object_unref (cs);
-}
-
-GST_START_TEST (test_sinkpad_property_controller)
-{
-  GstBus *bus;
-  GstMessage *msg;
-  GstElement *pipeline, *sink, *mix, *src1;
-  GstPad *srcpad, *sinkpad;
-  GError *error = NULL;
-  gchar *debug;
-
-  pipeline = gst_pipeline_new ("pipeline");
-  mix = gst_element_factory_make ("audiomixer", "audiomixer");
-  sink = gst_element_factory_make ("fakesink", "sink");
-  src1 = gst_element_factory_make ("audiotestsrc", "src1");
-  g_object_set (src1, "num-buffers", 100, NULL);
-  gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL);
-  fail_unless (gst_element_link (mix, sink));
-
-  srcpad = gst_element_get_static_pad (src1, "src");
-  sinkpad = gst_element_get_request_pad (mix, "sink_0");
-  fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK);
-  set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0);
-  gst_object_unref (sinkpad);
-  gst_object_unref (srcpad);
-
-  gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
-  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
-  msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,
-      GST_MESSAGE_EOS | GST_MESSAGE_ERROR);
-  switch (GST_MESSAGE_TYPE (msg)) {
-    case GST_MESSAGE_ERROR:
-      gst_message_parse_error (msg, &error, &debug);
-      g_printerr ("ERROR from element %s: %s\n",
-          GST_OBJECT_NAME (msg->src), error->message);
-      g_printerr ("Debug info: %s\n", debug);
-      g_error_free (error);
-      g_free (debug);
-      break;
-    case GST_MESSAGE_EOS:
-      break;
-    default:
-      g_assert_not_reached ();
-  }
-  gst_message_unref (msg);
-  g_object_unref (bus);
-
-  gst_element_set_state (pipeline, GST_STATE_NULL);
-  gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static void
-change_src_caps (GstElement * fakesink, GstBuffer * buffer, GstPad * pad,
-    GstElement * capsfilter)
-{
-  GstCaps *caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, GST_AUDIO_NE (S32),
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
-
-  g_object_set (capsfilter, "caps", caps, NULL);
-  g_signal_connect (fakesink, "handoff", (GCallback) handoff_buffer_cb, NULL);
-  g_signal_handlers_disconnect_by_func (fakesink, change_src_caps, capsfilter);
-}
-
-/* In this test, we create an input buffer with a duration of 2 seconds,
- * and require the audiomixer to output 1 second long buffers.
- * The input buffer will thus be mixed twice, and the audiomixer will
- * output two buffers.
- *
- * After audiomixer has output a first buffer, we change its output format
- * from S8 to S32. As our sample rate stays the same at 10 fps, and we use
- * mono, the first buffer should be 10 bytes long, and the second 40.
- *
- * The input buffer is made up of 15 0-valued bytes, and 5 1-valued bytes.
- * We verify that the second buffer contains 5 0-valued integers, and
- * 5 1 << 24 valued integers.
- */
-GST_START_TEST (test_change_output_caps)
-{
-  GstSegment segment;
-  GstElement *bin, *audiomixer, *capsfilter, *sink;
-  GstBus *bus;
-  GstPad *sinkpad;
-  gboolean res;
-  GstStateChangeReturn state_res;
-  GstFlowReturn ret;
-  GstEvent *event;
-  GstBuffer *buffer;
-  GstCaps *caps;
-  GstQuery *drain = gst_query_new_drain ();
-  GstMapInfo inmap;
-  GstMapInfo outmap;
-  gsize i;
-
-  bin = gst_pipeline_new ("pipeline");
-  bus = gst_element_get_bus (bin);
-  gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH);
-
-  g_signal_connect (bus, "message::error", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::warning", (GCallback) message_received, bin);
-  g_signal_connect (bus, "message::eos", (GCallback) message_received, bin);
-
-  audiomixer = gst_element_factory_make ("audiomixer", "audiomixer");
-  g_object_set (audiomixer, "output-buffer-duration", GST_SECOND, NULL);
-  capsfilter = gst_element_factory_make ("capsfilter", NULL);
-  sink = gst_element_factory_make ("fakesink", "sink");
-  g_object_set (sink, "signal-handoffs", TRUE, NULL);
-  g_signal_connect (sink, "handoff", (GCallback) change_src_caps, capsfilter);
-  gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL);
-
-  res = gst_element_link_many (audiomixer, capsfilter, sink, NULL);
-  fail_unless (res == TRUE, NULL);
-
-  state_res = gst_element_set_state (bin, GST_STATE_PLAYING);
-  ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE);
-
-  sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u");
-  fail_if (sinkpad == NULL, NULL);
-
-  gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test"));
-
-  caps = gst_caps_new_simple ("audio/x-raw",
-      "format", G_TYPE_STRING, "S8",
-      "layout", G_TYPE_STRING, "interleaved",
-      "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL);
-
-  gst_pad_set_caps (sinkpad, caps);
-  g_object_set (capsfilter, "caps", caps, NULL);
-  gst_caps_unref (caps);
-
-  gst_segment_init (&segment, GST_FORMAT_TIME);
-  segment.start = 0;
-  segment.stop = 2 * GST_SECOND;
-  segment.time = 0;
-  event = gst_event_new_segment (&segment);
-  gst_pad_send_event (sinkpad, event);
-
-  gst_buffer_replace (&handoff_buffer, NULL);
-
-  buffer = new_buffer (20, 0, 0, 2 * GST_SECOND, 0);
-  gst_buffer_map (buffer, &inmap, GST_MAP_WRITE);
-  memset (inmap.data + 15, 1, 5);
-  gst_buffer_unmap (buffer, &inmap);
-  ret = gst_pad_chain (sinkpad, buffer);
-  ck_assert_int_eq (ret, GST_FLOW_OK);
-  gst_pad_query (sinkpad, drain);
-  fail_unless (handoff_buffer != NULL);
-  fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 40);
-
-  gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ);
-  for (i = 0; i < 10; i++) {
-    guint32 sample;
-
-#if G_BYTE_ORDER == G_LITTLE_ENDIAN
-    sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]);
-#else
-    sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]);
-#endif
-
-    if (i < 5) {
-      fail_unless_equals_int (sample, 0);
-    } else {
-      fail_unless_equals_int (sample, 1 << 24);
-    }
-  }
-  gst_buffer_unmap (handoff_buffer, &outmap);
-
-  gst_element_release_request_pad (audiomixer, sinkpad);
-  gst_object_unref (sinkpad);
-  gst_element_set_state (bin, GST_STATE_NULL);
-  gst_bus_remove_signal_watch (bus);
-  gst_object_unref (bus);
-  gst_object_unref (bin);
-  gst_query_unref (drain);
-}
-
-GST_END_TEST;
-
-static Suite *
-audiomixer_suite (void)
-{
-  Suite *s = suite_create ("audiomixer");
-  TCase *tc_chain = tcase_create ("general");
-
-  suite_add_tcase (s, tc_chain);
-  tcase_add_test (tc_chain, test_caps);
-  tcase_add_test (tc_chain, test_filter_caps);
-  tcase_add_test (tc_chain, test_event);
-  tcase_add_test (tc_chain, test_play_twice);
-  tcase_add_test (tc_chain, test_play_twice_then_add_and_play_again);
-  tcase_add_test (tc_chain, test_live_seeking);
-  tcase_add_test (tc_chain, test_add_pad);
-  tcase_add_test (tc_chain, test_remove_pad);
-  tcase_add_test (tc_chain, test_clip);
-  tcase_add_test (tc_chain, test_duration_is_max);
-  tcase_add_test (tc_chain, test_duration_unknown_overrides);
-  tcase_add_test (tc_chain, test_loop);
-  tcase_add_test (tc_chain, test_flush_start_flush_stop);
-  tcase_add_test (tc_chain, test_sync);
-  tcase_add_test (tc_chain, test_sync_discont);
-  tcase_add_test (tc_chain, test_sync_unaligned);
-  tcase_add_test (tc_chain, test_segment_base_handling);
-  tcase_add_test (tc_chain, test_sinkpad_property_controller);
-  tcase_add_checked_fixture (tc_chain, test_setup, test_teardown);
-  tcase_add_test (tc_chain, test_change_output_caps);
-
-  /* Use a longer timeout */
-#ifdef HAVE_VALGRIND
-  if (RUNNING_ON_VALGRIND) {
-    tcase_set_timeout (tc_chain, 5 * 60);
-  } else
-#endif
-  {
-    /* this is shorter than the default 60 seconds?! (tpm) */
-    /* tcase_set_timeout (tc_chain, 6); */
-  }
-
-  return s;
-}
-
-GST_CHECK_MAIN (audiomixer);
diff --git a/tests/check/meson.build b/tests/check/meson.build
index 55f1513e8..1cb817164 100644
--- a/tests/check/meson.build
+++ b/tests/check/meson.build
@@ -18,8 +18,6 @@ base_tests = [
   [['elements/aiffparse.c']],
   [['elements/asfmux.c']],
   [['elements/assrender.c'], not ass_dep.found(), [ass_dep]],
-  [['elements/audiointerleave.c']],
-  [['elements/audiomixer.c']],
   [['elements/autoconvert.c']],
   [['elements/autovideoconvert.c']],
   [['elements/camerabin.c']],