| /* |
| * 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_METHOD(void, Reset, (), (override)); |
| MOCK_METHOD(RenderDelayBuffer::BufferingEvent, |
| Insert, |
| (const Block& block), |
| (override)); |
| MOCK_METHOD(void, HandleSkippedCaptureProcessing, (), (override)); |
| MOCK_METHOD(RenderDelayBuffer::BufferingEvent, |
| PrepareCaptureProcessing, |
| (), |
| (override)); |
| MOCK_METHOD(bool, AlignFromDelay, (size_t delay), (override)); |
| MOCK_METHOD(void, AlignFromExternalDelay, (), (override)); |
| MOCK_METHOD(size_t, Delay, (), (const, override)); |
| MOCK_METHOD(size_t, MaxDelay, (), (const, override)); |
| MOCK_METHOD(RenderBuffer*, GetRenderBuffer, (), (override)); |
| MOCK_METHOD(const DownsampledRenderBuffer&, |
| GetDownsampledRenderBuffer, |
| (), |
| (const, override)); |
| MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override)); |
| MOCK_METHOD(bool, HasReceivedBufferDelay, (), (override)); |
| |
| private: |
| RenderBuffer* FakeGetRenderBuffer() { return &render_buffer_; } |
| const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const { |
| return downsampled_render_buffer_; |
| } |
| BlockBuffer block_buffer_; |
| SpectrumBuffer 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_ |