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/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/aec3/render_buffer.h"
#include <algorithm>
#include <functional>
#include <vector>
#include "test/gtest.h"
namespace webrtc {
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies the check for non-null fft buffer.
TEST(RenderBufferDeathTest, NullExternalFftBuffer) {
BlockBuffer block_buffer(10, 3, 1);
SpectrumBuffer spectrum_buffer(10, 1);
EXPECT_DEATH(RenderBuffer(&block_buffer, &spectrum_buffer, nullptr), "");
}
// Verifies the check for non-null spectrum buffer.
TEST(RenderBufferDeathTest, NullExternalSpectrumBuffer) {
FftBuffer fft_buffer(10, 1);
BlockBuffer block_buffer(10, 3, 1);
EXPECT_DEATH(RenderBuffer(&block_buffer, nullptr, &fft_buffer), "");
}
// Verifies the check for non-null block buffer.
TEST(RenderBufferDeathTest, NullExternalBlockBuffer) {
FftBuffer fft_buffer(10, 1);
SpectrumBuffer spectrum_buffer(10, 1);
EXPECT_DEATH(RenderBuffer(nullptr, &spectrum_buffer, &fft_buffer), "");
}
#endif
} // namespace webrtc
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