audiointerleave: Add unit tests

Almost a copy of the "interleave" unit tests, improved to support
the thread on the src pad on GstAggregator.

https://bugzilla.gnome.org/show_bug.cgi?id=740236

3 files changed, 876 insertions, 3 deletions

diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
index 91f539e29..4cf2dac85 100644
--- a/tests/check/Makefile.am
+++ b/tests/check/Makefile.am
@@ -238,6 +238,7 @@ check_PROGRAMS = \
 	elements/aiffparse \
 	elements/autoconvert \
 	elements/autovideoconvert \
+	elements/audiointerleave \
 	elements/audiomixer \
 	elements/asfmux \
 	elements/baseaudiovisualizer \
@@ -245,7 +246,7 @@ check_PROGRAMS = \
 	elements/dataurisrc \
 	elements/gdppay \
 	elements/gdpdepay \
- 	elements/compositor \
+	elements/compositor \
 	$(check_jifmux) \
 	elements/jpegparse \
 	elements/h263parse \
@@ -289,6 +290,9 @@ LDADD = $(GST_CHECK_LIBS)
 elements_audiomixer_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD)
 elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
 
+elements_audiointerleave_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ -lgstaudio-@GST_API_VERSION@ $(LDADD)
+elements_audiointerleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS)
+
 # parser unit test convenience lib
 noinst_LTLIBRARIES = libparser.la
 libparser_la_SOURCES = elements/parser.c elements/parser.h
diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore
index d38dcdbb6..1c5d0a924 100644
--- a/tests/check/elements/.gitignore
+++ b/tests/check/elements/.gitignore
@@ -2,6 +2,7 @@
 aiffparse
 asfmux
 assrender
+audiointerleave
 audiomixer
 autoconvert
 autovideoconvert
@@ -14,7 +15,6 @@ curlftpsink
 curlsftpsink
 curlhttpsink
 curlsmtpsink
-deinterleave
 dataurisrc
 faac
 faad
@@ -26,7 +26,6 @@ h264parse
 hlsdemux_m3u8
 id3mux
 imagecapturebin
-interleave
 jifmux
 jpegparse
 kate
diff --git a/tests/check/elements/audiointerleave.c b/tests/check/elements/audiointerleave.c
new file mode 100644
index 000000000..83aaf0f48
--- /dev/null
+++ b/tests/check/elements/audiointerleave.c
@@ -0,0 +1,870 @@
+/* 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>
+
+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_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_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;
+  gsize size;
+  GstCaps *caps;
+  guint64 mask;
+  GstAudioChannelPosition pos;
+
+  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);
+
+  size = 48000 * sizeof (gfloat);
+  data = g_malloc (size);
+  for (i = 0; i < 48000; i++)
+    data[i] = (n % 2 == 0) ? -1.0 : 1.0;
+
+  gst_buffer_append_memory (buffer, gst_memory_new_wrapped (0, data,
+          size, 0, size, data, g_free));
+
+  GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
+  GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
+  GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
+  GST_BUFFER_DURATION (buffer) = GST_SECOND;
+}
+
+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) {
+    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 ();
+  }
+
+  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);
+      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, "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, "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, "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, "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, "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, "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;
+
+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);
+
+  return s;
+}
+
+GST_CHECK_MAIN (audiointerleave);