/* * Copyright (c) 2016 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_BLOCK_PROCESSOR_H_ #define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_ #include #include "modules/audio_processing/aec3/block_processor.h" #include "test/gmock.h" namespace webrtc { namespace test { class MockBlockProcessor : public BlockProcessor { public: MockBlockProcessor(); virtual ~MockBlockProcessor(); MOCK_METHOD3( ProcessCapture, void(bool level_change, bool saturated_microphone_signal, std::vector>>* capture_block)); MOCK_METHOD1(BufferRender, void(const std::vector>>& block)); MOCK_METHOD1(UpdateEchoLeakageStatus, void(bool leakage_detected)); MOCK_CONST_METHOD1(GetMetrics, void(EchoControl::Metrics* metrics)); MOCK_METHOD1(SetAudioBufferDelay, void(size_t delay_ms)); }; } // namespace test } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_