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https://github.com/mollyim/webrtc.git
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5c532d3774
This CL robustifies the echo removal behavior when headsets are used. In particular it: -Introduces a secondary, more refined alignment when no alignment can be found using the delay estimator. -Changes decision logic for when to use the linear filter output. -Changes the decision logic for when to be transparent. -Changes the way that the transparent mode works. -Makes the nonlinear mode less aggressive. -Removes the detector for non-audible echoes. -Makes the attenuation when there are signals with strong narrowband characteristics more mild in scenarios with low render. Furthermore the CL: -Removes the input of external echo leakage information. Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919 Change-Id: Ied1fe0c0a35d3c31b47606ed2db319a73644d406 Reviewed-on: https://webrtc-review.googlesource.com/60866 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22548}
72 lines
2.8 KiB
C++
72 lines
2.8 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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explicit MockRenderDelayBuffer(int sample_rate_hz)
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: block_buffer_(GetRenderDelayBufferSize(4, 4, 12),
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NumBandsForRate(sample_rate_hz),
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kBlockSize),
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spectrum_buffer_(block_buffer_.buffer.size(), kFftLengthBy2Plus1),
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fft_buffer_(block_buffer_.buffer.size()),
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render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_),
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downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) {
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ON_CALL(*this, GetRenderBuffer())
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.WillByDefault(
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testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
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ON_CALL(*this, GetDownsampledRenderBuffer())
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.WillByDefault(testing::Invoke(
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this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
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}
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virtual ~MockRenderDelayBuffer() = default;
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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<float>>& block));
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MOCK_METHOD0(PrepareCaptureProcessing, RenderDelayBuffer::BufferingEvent());
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MOCK_METHOD1(SetDelay, bool(size_t delay));
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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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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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