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126 lines
3.4 KiB
C
126 lines
3.4 KiB
C
/* GStreamer
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*
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* Copyright (C) 2020 Seungha Yang <seungha@centricular.com>
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/check/gstcheck.h>
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#include <gst/check/gstharness.h>
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#include <gst/audio/audio.h>
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GST_START_TEST (test_audioenc_drain)
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{
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GstHarness *h;
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GstAudioInfo info;
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GstBuffer *in_buf;
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gint i = 0;
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gint num_output = 0;
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GstFlowReturn ret;
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GstSegment segment;
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GstCaps *caps;
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gint samples_per_buffer = 1024;
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gint rate = 44100;
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gint size;
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GstClockTime duration;
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h = gst_harness_new ("avenc_aac");
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fail_unless (h != NULL);
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_F32, rate, 1, NULL);
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caps = gst_audio_info_to_caps (&info);
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gst_harness_set_src_caps (h, gst_caps_copy (caps));
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duration = gst_util_uint64_scale_int (samples_per_buffer, GST_SECOND, rate);
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size = samples_per_buffer * GST_AUDIO_INFO_BPF (&info);
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for (i = 0; i < 2; i++) {
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in_buf = gst_buffer_new_and_alloc (size);
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gst_buffer_memset (in_buf, 0, 0, size);
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/* small rounding error would be expected, but should be fine */
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GST_BUFFER_PTS (in_buf) = i * duration;
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GST_BUFFER_DURATION (in_buf) = duration;
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ret = gst_harness_push (h, in_buf);
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fail_unless (ret == GST_FLOW_OK, "GstFlowReturn was %s",
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gst_flow_get_name (ret));
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}
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gst_segment_init (&segment, GST_FORMAT_TIME);
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fail_unless (gst_segment_set_running_time (&segment, GST_FORMAT_TIME,
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2 * duration));
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/* Push new eos event to drain encoder */
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fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
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/* And start new stream */
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fail_unless (gst_harness_push_event (h,
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gst_event_new_stream_start ("new-stream-id")));
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gst_harness_set_src_caps (h, caps);
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fail_unless (gst_harness_push_event (h, gst_event_new_segment (&segment)));
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in_buf = gst_buffer_new_and_alloc (size);
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GST_BUFFER_PTS (in_buf) = 2 * duration;
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GST_BUFFER_DURATION (in_buf) = duration;
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ret = gst_harness_push (h, in_buf);
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fail_unless (ret == GST_FLOW_OK, "GstFlowReturn was %s",
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gst_flow_get_name (ret));
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/* Finish encoding and drain again */
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fail_unless (gst_harness_push_event (h, gst_event_new_eos ()));
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do {
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GstBuffer *out_buf = NULL;
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out_buf = gst_harness_try_pull (h);
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if (out_buf) {
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num_output++;
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gst_buffer_unref (out_buf);
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continue;
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}
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break;
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} while (1);
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fail_unless (num_output >= 3);
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gst_harness_teardown (h);
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}
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GST_END_TEST;
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static Suite *
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avaudenc_suite (void)
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{
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Suite *s = suite_create ("avaudenc");
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TCase *tc_chain = tcase_create ("general");
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suite_add_tcase (s, tc_chain);
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tcase_add_test (tc_chain, test_audioenc_drain);
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return s;
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}
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GST_CHECK_MAIN (avaudenc)
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