mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-18 16:17:50 +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}
61 lines
2.2 KiB
C++
61 lines
2.2 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:
|
|
MockRenderDelayBuffer(int sample_rate_hz, size_t num_channels);
|
|
virtual ~MockRenderDelayBuffer();
|
|
|
|
MOCK_METHOD0(Reset, void());
|
|
MOCK_METHOD1(Insert,
|
|
RenderDelayBuffer::BufferingEvent(
|
|
const std::vector<std::vector<std::vector<float>>>& block));
|
|
MOCK_METHOD0(PrepareCaptureProcessing, RenderDelayBuffer::BufferingEvent());
|
|
MOCK_METHOD1(AlignFromDelay, bool(size_t delay));
|
|
MOCK_METHOD0(AlignFromExternalDelay, void());
|
|
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));
|
|
MOCK_METHOD0(HasReceivedBufferDelay, bool());
|
|
|
|
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_
|