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1132 lines
38 KiB
C
1132 lines
38 KiB
C
/* GStreamer unit tests for the audiointerleave element
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* Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
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* Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#ifdef HAVE_VALGRIND
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# include <valgrind/valgrind.h>
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#endif
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#include <gst/check/gstcheck.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/audio-enumtypes.h>
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#include <gst/check/gstharness.h>
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static void
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gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element,
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GstCaps * caps, GstFormat format, const gchar * stream_id)
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{
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GstSegment segment;
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gst_segment_init (&segment, format);
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fail_unless (gst_pad_push_event (srcpad,
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gst_event_new_stream_start (stream_id)));
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if (caps)
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fail_unless (gst_pad_push_event (srcpad, gst_event_new_caps (caps)));
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fail_unless (gst_pad_push_event (srcpad, gst_event_new_segment (&segment)));
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}
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GST_START_TEST (test_create_and_unref)
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{
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GstElement *interleave;
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interleave = gst_element_factory_make ("audiointerleave", NULL);
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fail_unless (interleave != NULL);
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gst_element_set_state (interleave, GST_STATE_NULL);
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gst_object_unref (interleave);
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}
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GST_END_TEST;
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GST_START_TEST (test_request_pads)
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{
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GstElement *interleave;
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GstPad *pad1, *pad2;
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interleave = gst_element_factory_make ("audiointerleave", NULL);
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fail_unless (interleave != NULL);
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pad1 = gst_element_get_request_pad (interleave, "sink_%u");
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fail_unless (pad1 != NULL);
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fail_unless_equals_string (GST_OBJECT_NAME (pad1), "sink_0");
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pad2 = gst_element_get_request_pad (interleave, "sink_%u");
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fail_unless (pad2 != NULL);
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fail_unless_equals_string (GST_OBJECT_NAME (pad2), "sink_1");
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gst_element_release_request_pad (interleave, pad2);
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gst_object_unref (pad2);
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gst_element_release_request_pad (interleave, pad1);
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gst_object_unref (pad1);
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gst_element_set_state (interleave, GST_STATE_NULL);
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gst_object_unref (interleave);
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}
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GST_END_TEST;
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static GstPad **mysrcpads, *mysinkpad;
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static GstBus *bus;
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static GstElement *interleave;
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static GMutex data_mutex;
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static GCond data_cond;
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static gint have_data;
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static gfloat input[2];
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (F32) ", "
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"channels = (int) 2, layout = (string) {interleaved, non-interleaved}, rate = (int) 48000"));
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (F32) ", "
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"channels = (int) 1, layout = (string) interleaved, rate = (int) 48000"));
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#define CAPS_48khz \
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"audio/x-raw, " \
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"format = (string) " GST_AUDIO_NE (F32) ", " \
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"channels = (int) 1, layout = (string) non-interleaved," \
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"rate = (int) 48000"
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static GstFlowReturn
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interleave_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer)
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{
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GstMapInfo map;
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gfloat *outdata;
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gint i;
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fail_unless (GST_IS_BUFFER (buffer));
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fail_unless (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP));
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gst_buffer_map (buffer, &map, GST_MAP_READ);
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outdata = (gfloat *) map.data;
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fail_unless (outdata != NULL);
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#ifdef HAVE_VALGRIND
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if (!(RUNNING_ON_VALGRIND))
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#endif
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for (i = 0; i < map.size / sizeof (float); i += 2) {
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fail_unless_equals_float (outdata[i], input[0]);
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fail_unless_equals_float (outdata[i + 1], input[1]);
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}
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g_mutex_lock (&data_mutex);
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have_data += map.size;
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g_cond_signal (&data_cond);
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g_mutex_unlock (&data_mutex);
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gst_buffer_unmap (buffer, &map);
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gst_buffer_unref (buffer);
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return GST_FLOW_OK;
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}
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GST_START_TEST (test_audiointerleave_2ch)
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{
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GstElement *queue;
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GstPad *sink0, *sink1, *src, *tmp;
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GstCaps *caps;
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gint i;
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GstBuffer *inbuf;
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gfloat *indata;
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GstMapInfo map;
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mysrcpads = g_new0 (GstPad *, 2);
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have_data = 0;
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interleave = gst_element_factory_make ("audiointerleave", NULL);
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fail_unless (interleave != NULL);
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g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
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queue = gst_element_factory_make ("queue", "queue");
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fail_unless (queue != NULL);
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sink0 = gst_element_get_request_pad (interleave, "sink_%u");
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fail_unless (sink0 != NULL);
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fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
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sink1 = gst_element_get_request_pad (interleave, "sink_%u");
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fail_unless (sink1 != NULL);
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fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
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mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
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fail_unless (mysrcpads[0] != NULL);
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caps = gst_caps_from_string (CAPS_48khz);
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gst_pad_set_active (mysrcpads[0], TRUE);
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gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
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GST_FORMAT_TIME, "0");
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gst_pad_use_fixed_caps (mysrcpads[0]);
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mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
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fail_unless (mysrcpads[1] != NULL);
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gst_pad_set_active (mysrcpads[1], TRUE);
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gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
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GST_FORMAT_TIME, "1");
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gst_pad_use_fixed_caps (mysrcpads[1]);
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tmp = gst_element_get_static_pad (queue, "sink");
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fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
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gst_object_unref (tmp);
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tmp = gst_element_get_static_pad (queue, "src");
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fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
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gst_object_unref (tmp);
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fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
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mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
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fail_unless (mysinkpad != NULL);
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gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
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gst_pad_set_active (mysinkpad, TRUE);
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src = gst_element_get_static_pad (interleave, "src");
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fail_unless (src != NULL);
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fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
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gst_object_unref (src);
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bus = gst_bus_new ();
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gst_element_set_bus (interleave, bus);
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fail_unless (gst_element_set_state (interleave,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
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fail_unless (gst_element_set_state (queue,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
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input[0] = -1.0;
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inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
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//GST_BUFFER_PTS (inbuf) = 0;
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gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
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indata = (gfloat *) map.data;
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for (i = 0; i < 48000; i++)
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indata[i] = -1.0;
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gst_buffer_unmap (inbuf, &map);
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fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
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input[1] = 1.0;
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inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
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//GST_BUFFER_PTS (inbuf) = 0;
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gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
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indata = (gfloat *) map.data;
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for (i = 0; i < 48000; i++)
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indata[i] = 1.0;
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gst_buffer_unmap (inbuf, &map);
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fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
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inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
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//GST_BUFFER_PTS (inbuf) = GST_SECOND;
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gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
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indata = (gfloat *) map.data;
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for (i = 0; i < 48000; i++)
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indata[i] = -1.0;
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gst_buffer_unmap (inbuf, &map);
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fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
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inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
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//GST_BUFFER_PTS (inbuf) = GST_SECOND;
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gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
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indata = (gfloat *) map.data;
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for (i = 0; i < 48000; i++)
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indata[i] = 1.0;
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gst_buffer_unmap (inbuf, &map);
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fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
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g_mutex_lock (&data_mutex);
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while (have_data < 48000 * 2 * 2 * sizeof (float))
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g_cond_wait (&data_cond, &data_mutex);
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g_mutex_unlock (&data_mutex);
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gst_bus_set_flushing (bus, TRUE);
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gst_element_set_state (interleave, GST_STATE_NULL);
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gst_element_set_state (queue, GST_STATE_NULL);
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gst_object_unref (mysrcpads[0]);
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gst_object_unref (mysrcpads[1]);
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gst_object_unref (mysinkpad);
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gst_element_release_request_pad (interleave, sink0);
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gst_object_unref (sink0);
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gst_element_release_request_pad (interleave, sink1);
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gst_object_unref (sink1);
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gst_object_unref (interleave);
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gst_object_unref (queue);
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gst_object_unref (bus);
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gst_caps_unref (caps);
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g_free (mysrcpads);
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}
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GST_END_TEST;
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GST_START_TEST (test_audiointerleave_2ch_1eos)
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{
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GstElement *queue;
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GstPad *sink0, *sink1, *src, *tmp;
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GstCaps *caps;
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gint i;
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GstBuffer *inbuf;
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gfloat *indata;
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GstMapInfo map;
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mysrcpads = g_new0 (GstPad *, 2);
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have_data = 0;
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interleave = gst_element_factory_make ("audiointerleave", NULL);
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fail_unless (interleave != NULL);
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g_object_set (interleave, "latency", GST_SECOND / 4, NULL);
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queue = gst_element_factory_make ("queue", "queue");
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fail_unless (queue != NULL);
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sink0 = gst_element_get_request_pad (interleave, "sink_%u");
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fail_unless (sink0 != NULL);
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fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0");
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sink1 = gst_element_get_request_pad (interleave, "sink_%u");
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fail_unless (sink1 != NULL);
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fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1");
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mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
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fail_unless (mysrcpads[0] != NULL);
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caps = gst_caps_from_string (CAPS_48khz);
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gst_pad_set_active (mysrcpads[0], TRUE);
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gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps,
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GST_FORMAT_TIME, "0");
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gst_pad_use_fixed_caps (mysrcpads[0]);
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mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
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fail_unless (mysrcpads[1] != NULL);
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gst_pad_set_active (mysrcpads[1], TRUE);
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gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps,
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GST_FORMAT_TIME, "1");
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gst_pad_use_fixed_caps (mysrcpads[1]);
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tmp = gst_element_get_static_pad (queue, "sink");
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fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
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gst_object_unref (tmp);
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tmp = gst_element_get_static_pad (queue, "src");
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fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
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gst_object_unref (tmp);
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fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
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mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
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fail_unless (mysinkpad != NULL);
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gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
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gst_pad_set_active (mysinkpad, TRUE);
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src = gst_element_get_static_pad (interleave, "src");
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fail_unless (src != NULL);
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fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
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gst_object_unref (src);
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bus = gst_bus_new ();
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gst_element_set_bus (interleave, bus);
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fail_unless (gst_element_set_state (interleave,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
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fail_unless (gst_element_set_state (queue,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
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input[0] = -1.0;
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inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
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GST_BUFFER_PTS (inbuf) = 0;
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gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
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indata = (gfloat *) map.data;
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for (i = 0; i < 48000; i++)
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indata[i] = -1.0;
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gst_buffer_unmap (inbuf, &map);
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fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
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input[1] = 1.0;
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inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
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GST_BUFFER_PTS (inbuf) = 0;
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gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
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indata = (gfloat *) map.data;
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for (i = 0; i < 48000; i++)
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indata[i] = 1.0;
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gst_buffer_unmap (inbuf, &map);
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fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
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g_mutex_lock (&data_mutex);
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/* 48000 samples per buffer * 2 sources * 2 buffers */
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while (have_data != 48000 * 2 * sizeof (float))
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g_cond_wait (&data_cond, &data_mutex);
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g_mutex_unlock (&data_mutex);
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input[0] = 0.0;
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gst_pad_push_event (mysrcpads[0], gst_event_new_eos ());
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input[1] = 1.0;
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inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
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GST_BUFFER_PTS (inbuf) = GST_SECOND;
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gst_buffer_map (inbuf, &map, GST_MAP_WRITE);
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indata = (gfloat *) map.data;
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for (i = 0; i < 48000; i++)
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indata[i] = 1.0;
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gst_buffer_unmap (inbuf, &map);
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fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
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g_mutex_lock (&data_mutex);
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/* 48000 samples per buffer * 2 sources * 2 buffers */
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while (have_data != 48000 * 2 * 2 * sizeof (float))
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g_cond_wait (&data_cond, &data_mutex);
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g_mutex_unlock (&data_mutex);
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gst_bus_set_flushing (bus, TRUE);
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gst_element_set_state (interleave, GST_STATE_NULL);
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gst_element_set_state (queue, GST_STATE_NULL);
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gst_object_unref (mysrcpads[0]);
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gst_object_unref (mysrcpads[1]);
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gst_object_unref (mysinkpad);
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gst_element_release_request_pad (interleave, sink0);
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gst_object_unref (sink0);
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gst_element_release_request_pad (interleave, sink1);
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gst_object_unref (sink1);
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gst_object_unref (interleave);
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gst_object_unref (queue);
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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;
|
|
GstQuery *q;
|
|
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);
|
|
|
|
for (i = 0; i < 24; i++)
|
|
gst_harness_crank_single_clock_wait (h);
|
|
fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK
|
|
(h->element)), 750 * GST_MSECOND);
|
|
|
|
/* Check that the queue is really empty */
|
|
q = gst_query_new_drain ();
|
|
gst_pad_peer_query (h->srcpad, q);
|
|
gst_query_unref (q);
|
|
|
|
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);
|