gstreamer/subprojects/gst-libav/tests/check/elements/avaudenc.c
Piotr Brzeziński 2ec7f9f9b3 avaudenc: Add simple 16 channel encoding test
Used to be crashing because of a double-free introduced years ago and never really noticed, so let's add a test to
make sure it doesn't happen again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6373>
2024-03-15 12:30:04 +00:00

162 lines
4.5 KiB
C

/* 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;
GST_START_TEST (test_audioenc_16_channels)
{
/* avaudenc used to have a bug for >8ch where a double-free attempt would occur,
* crashing the whole process. Since >8ch encoding is quite rarely used, this test
* is meant to detect any crashes that would indicate somebody broke that again */
GstHarness *h;
GstAudioInfo info;
GstBuffer *in_buf;
GstCaps *caps;
gint size;
GstAudioChannelPosition position[16];
/* 16ch hexadecagonal layout */
guint64 channel_mask = 0x3137D37;
h = gst_harness_new ("avenc_aac");
fail_unless (h != NULL);
gst_audio_channel_positions_from_mask (16, channel_mask, position);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_F32, 44100, 16, position);
caps = gst_audio_info_to_caps (&info);
gst_harness_set_src_caps (h, caps);
size = 1024 * GST_AUDIO_INFO_BPF (&info);
in_buf = gst_buffer_new_and_alloc (size);
gst_buffer_memset (in_buf, 0, 0, size);
GstFlowReturn ret = gst_harness_push (h, in_buf);
fail_if (ret != GST_FLOW_OK);
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);
tcase_add_test (tc_chain, test_audioenc_16_channels);
return s;
}
GST_CHECK_MAIN (avaudenc)