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