--- /dev/null
+/* 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);
projects / platform / upstream / gstreamer.git / commitdiff