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a66395e72f
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}
61 lines
2.2 KiB
C++
61 lines
2.2 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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#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
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#include <vector>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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#include "modules/audio_processing/aec3/render_delay_buffer.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockRenderDelayBuffer : public RenderDelayBuffer {
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public:
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MockRenderDelayBuffer(int sample_rate_hz, size_t num_channels);
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virtual ~MockRenderDelayBuffer();
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MOCK_METHOD0(Reset, void());
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MOCK_METHOD1(Insert,
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RenderDelayBuffer::BufferingEvent(
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const std::vector<std::vector<std::vector<float>>>& block));
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MOCK_METHOD0(PrepareCaptureProcessing, RenderDelayBuffer::BufferingEvent());
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MOCK_METHOD1(AlignFromDelay, bool(size_t delay));
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MOCK_METHOD0(AlignFromExternalDelay, void());
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MOCK_CONST_METHOD0(Delay, size_t());
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MOCK_CONST_METHOD0(MaxDelay, size_t());
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MOCK_METHOD0(GetRenderBuffer, RenderBuffer*());
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MOCK_CONST_METHOD0(GetDownsampledRenderBuffer,
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const DownsampledRenderBuffer&());
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MOCK_CONST_METHOD1(CausalDelay, bool(size_t delay));
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MOCK_METHOD1(SetAudioBufferDelay, void(size_t delay_ms));
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MOCK_METHOD0(HasReceivedBufferDelay, bool());
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private:
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RenderBuffer* FakeGetRenderBuffer() { return &render_buffer_; }
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const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const {
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return downsampled_render_buffer_;
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}
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MatrixBuffer block_buffer_;
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VectorBuffer spectrum_buffer_;
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FftBuffer fft_buffer_;
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RenderBuffer render_buffer_;
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DownsampledRenderBuffer downsampled_render_buffer_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
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