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/* GStreamer
*
* Copyright (C) 2020 Seungha Yang <seungha@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/check/gstcheck.h>
#include <gst/check/gstharness.h>
#include <gst/audio/audio.h>
GST_START_TEST (test_audioenc_drain)
{
GstHarness *h;
GstAudioInfo info;
GstBuffer *in_buf;
gint i = 0;
gint num_output = 0;
GstFlowReturn ret;
GstSegment segment;
GstCaps *caps;
gint samples_per_buffer = 1024;
gint rate = 44100;
gint size;
GstClockTime duration;
h = gst_harness_new ("avenc_aac");
fail_unless (h != NULL);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_F32, rate, 1, NULL);
caps = gst_audio_info_to_caps (&info);
gst_harness_set_src_caps (h, gst_caps_copy (caps));
duration = gst_util_uint64_scale_int (samples_per_buffer, GST_SECOND, rate);
size = samples_per_buffer * GST_AUDIO_INFO_BPF (&info);
for (i = 0; i < 2; i++) {
in_buf = gst_buffer_new_and_alloc (size);
gst_buffer_memset (in_buf, 0, 0, size);
/* small rounding error would be expected, but should be fine */
GST_BUFFER_PTS (in_buf) = i * duration;
GST_BUFFER_DURATION (in_buf) = duration;
ret = gst_harness_push (h, in_buf);
fail_unless (ret == GST_FLOW_OK, "GstFlowReturn was %s",
gst_flow_get_name (ret));
}
gst_segment_init (&segment, GST_FORMAT_TIME);
fail_unless (gst_segment_set_running_time (&segment, GST_FORMAT_TIME,
2 * duration));
/* Push new eos event to drain encoder */
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
/* And start new stream */
fail_unless (gst_harness_push_event (h,
gst_event_new_stream_start ("new-stream-id")));
gst_harness_set_src_caps (h, caps);
fail_unless (gst_harness_push_event (h, gst_event_new_segment (&segment)));
in_buf = gst_buffer_new_and_alloc (size);
GST_BUFFER_PTS (in_buf) = 2 * duration;
GST_BUFFER_DURATION (in_buf) = duration;
ret = gst_harness_push (h, in_buf);
fail_unless (ret == GST_FLOW_OK, "GstFlowReturn was %s",
gst_flow_get_name (ret));
/* Finish encoding and drain again */
fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
do {
GstBuffer *out_buf = NULL;
out_buf = gst_harness_try_pull (h);
if (out_buf) {
num_output++;
gst_buffer_unref (out_buf);
continue;
}
break;
} while (1);
fail_unless (num_output >= 3);
gst_harness_teardown (h);
}
GST_END_TEST;
static Suite *
avaudenc_suite (void)
{
Suite *s = suite_create ("avaudenc");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_audioenc_drain);
return s;
}
GST_CHECK_MAIN (avaudenc)
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