| /* |
| * 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_ |
| |
| #include <vector> |
| |
| #include "webrtc/modules/audio_processing/aec3/aec3_constants.h" |
| #include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h" |
| #include "webrtc/test/gmock.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class MockRenderDelayBuffer : public RenderDelayBuffer { |
| public: |
| explicit MockRenderDelayBuffer(int sample_rate_hz) |
| : block_(std::vector<std::vector<float>>( |
| NumBandsForRate(sample_rate_hz), |
| std::vector<float>(kBlockSize, 0.f))) { |
| ON_CALL(*this, GetNext()) |
| .WillByDefault( |
| testing::Invoke(this, &MockRenderDelayBuffer::FakeGetNext)); |
| } |
| virtual ~MockRenderDelayBuffer() = default; |
| |
| MOCK_METHOD1(Insert, bool(std::vector<std::vector<float>>* block)); |
| MOCK_METHOD0(GetNext, const std::vector<std::vector<float>>&()); |
| MOCK_METHOD1(SetDelay, void(size_t delay)); |
| MOCK_CONST_METHOD0(Delay, size_t()); |
| MOCK_CONST_METHOD0(MaxDelay, size_t()); |
| MOCK_CONST_METHOD0(IsBlockAvailable, bool()); |
| MOCK_CONST_METHOD0(MaxApiJitter, size_t()); |
| |
| private: |
| const std::vector<std::vector<float>>& FakeGetNext() const { return block_; } |
| std::vector<std::vector<float>> block_; |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_ |