| /* |
| * 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_AEC_DUMP_MOCK_AEC_DUMP_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ |
| |
| #include <memory> |
| |
| #include "webrtc/modules/audio_processing/include/aec_dump.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/test/gmock.h" |
| |
| namespace webrtc { |
| |
| namespace test { |
| |
| class MockAecDump : public AecDump { |
| public: |
| MockAecDump(); |
| virtual ~MockAecDump(); |
| |
| MOCK_METHOD1(WriteInitMessage, |
| void(const InternalAPMStreamsConfig& streams_config)); |
| |
| MOCK_METHOD1(AddCaptureStreamInput, void(const FloatAudioFrame& src)); |
| MOCK_METHOD1(AddCaptureStreamOutput, void(const FloatAudioFrame& src)); |
| MOCK_METHOD1(AddCaptureStreamInput, void(const AudioFrame& frame)); |
| MOCK_METHOD1(AddCaptureStreamOutput, void(const AudioFrame& frame)); |
| MOCK_METHOD1(AddAudioProcessingState, |
| void(const AudioProcessingState& state)); |
| MOCK_METHOD0(WriteCaptureStreamMessage, void()); |
| |
| MOCK_METHOD1(WriteRenderStreamMessage, void(const AudioFrame& frame)); |
| MOCK_METHOD1(WriteRenderStreamMessage, void(const FloatAudioFrame& src)); |
| |
| MOCK_METHOD1(WriteConfig, void(const InternalAPMConfig& config)); |
| }; |
| |
| } // namespace test |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ |