webrtc/modules/audio_processing/aec3/render_buffer_unittest.cc
Per Åhgren ce202a0f98 Reland "Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3.""
This is a reland of a66395e72f

Original change's description:
> Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
> 
> This is a reland of f3a197e553
> 
> Original change's description:
> > Add core multi-channel pipeline in AEC3
> > This CL adds basic the basic pipeline to support multi-channel
> > processing in AEC3.
> > 
> > Apart from that, it removes the 8 kHz processing support in several
> > places of the AEC3 code.
> > 
> > Bug: webrtc:10913
> > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29017}
> 
> Bug: webrtc:10913
> Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29034}

Bug: webrtc:10913
Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29042}
2019-09-03 06:12:32 +00:00

46 lines
1.4 KiB
C++

/*
* 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(RenderBuffer, NullExternalFftBuffer) {
MatrixBuffer block_buffer(10, 3, 1, kBlockSize);
VectorBuffer spectrum_buffer(10, kFftLengthBy2Plus1);
EXPECT_DEATH(RenderBuffer(&block_buffer, &spectrum_buffer, nullptr), "");
}
// Verifies the check for non-null spectrum buffer.
TEST(RenderBuffer, NullExternalSpectrumBuffer) {
FftBuffer fft_buffer(10);
MatrixBuffer block_buffer(10, 3, 1, kBlockSize);
EXPECT_DEATH(RenderBuffer(&block_buffer, nullptr, &fft_buffer), "");
}
// Verifies the check for non-null block buffer.
TEST(RenderBuffer, NullExternalBlockBuffer) {
FftBuffer fft_buffer(10);
VectorBuffer spectrum_buffer(10, kFftLengthBy2Plus1);
EXPECT_DEATH(RenderBuffer(nullptr, &spectrum_buffer, &fft_buffer), "");
}
#endif
} // namespace webrtc