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3cb6104864
Bug: b/130016532 Change-Id: If41121e9ad95f235200c4b6b146ba8a8749daac4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158220 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29602}
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(int 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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BlockBuffer block_buffer_;
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SpectrumBuffer 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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