mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00

This is a reland ofa66395e72f
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 off3a197e553
> > 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}
46 lines
1.4 KiB
C++
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
|