mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-18 08:07:56 +01:00
d0fa820559
This CL adds support for using any externally reported audio buffer delay to set the initial alignment in AEC3 which is used before the AEC has been able to detect the delay. Bug: chromium:834182,webrtc:9163 Change-Id: Ic71355f69b7c4d5815b78e49987043441e7908fb Reviewed-on: https://webrtc-review.googlesource.com/70580 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22917}
73 lines
2.8 KiB
C++
73 lines
2.8 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.
|
|
*/
|
|
|
|
#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
|
|
#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
|
|
|
|
#include <vector>
|
|
|
|
#include "modules/audio_processing/aec3/aec3_common.h"
|
|
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
|
|
#include "modules/audio_processing/aec3/render_buffer.h"
|
|
#include "modules/audio_processing/aec3/render_delay_buffer.h"
|
|
#include "test/gmock.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
|
|
class MockRenderDelayBuffer : public RenderDelayBuffer {
|
|
public:
|
|
explicit MockRenderDelayBuffer(int sample_rate_hz)
|
|
: block_buffer_(GetRenderDelayBufferSize(4, 4, 12),
|
|
NumBandsForRate(sample_rate_hz),
|
|
kBlockSize),
|
|
spectrum_buffer_(block_buffer_.buffer.size(), kFftLengthBy2Plus1),
|
|
fft_buffer_(block_buffer_.buffer.size()),
|
|
render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_),
|
|
downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) {
|
|
ON_CALL(*this, GetRenderBuffer())
|
|
.WillByDefault(
|
|
testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
|
|
ON_CALL(*this, GetDownsampledRenderBuffer())
|
|
.WillByDefault(testing::Invoke(
|
|
this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
|
|
}
|
|
virtual ~MockRenderDelayBuffer() = default;
|
|
|
|
MOCK_METHOD0(Reset, void());
|
|
MOCK_METHOD1(Insert,
|
|
RenderDelayBuffer::BufferingEvent(
|
|
const std::vector<std::vector<float>>& block));
|
|
MOCK_METHOD0(PrepareCaptureProcessing, RenderDelayBuffer::BufferingEvent());
|
|
MOCK_METHOD1(SetDelay, bool(size_t delay));
|
|
MOCK_CONST_METHOD0(Delay, size_t());
|
|
MOCK_CONST_METHOD0(MaxDelay, size_t());
|
|
MOCK_METHOD0(GetRenderBuffer, RenderBuffer*());
|
|
MOCK_CONST_METHOD0(GetDownsampledRenderBuffer,
|
|
const DownsampledRenderBuffer&());
|
|
MOCK_CONST_METHOD1(CausalDelay, bool(size_t delay));
|
|
MOCK_METHOD1(SetAudioBufferDelay, void(size_t delay_ms));
|
|
|
|
private:
|
|
RenderBuffer* FakeGetRenderBuffer() { return &render_buffer_; }
|
|
const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const {
|
|
return downsampled_render_buffer_;
|
|
}
|
|
MatrixBuffer block_buffer_;
|
|
VectorBuffer spectrum_buffer_;
|
|
FftBuffer fft_buffer_;
|
|
RenderBuffer render_buffer_;
|
|
DownsampledRenderBuffer downsampled_render_buffer_;
|
|
};
|
|
|
|
} // namespace test
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_
|