Chromium Code Reviews| Index: webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h |
| diff --git a/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h b/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..934eb1848470d93e6a081bd7de58e6e3f37081f0 |
| --- /dev/null |
| +++ b/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h |
| @@ -0,0 +1,51 @@ |
| +/* |
| + * 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: |
| + // To log an input/output pair, call the AddCapture* methods |
|
peah-webrtc
2017/05/22 06:56:37
Is this comment needed in a mock? Can't one instea
|
| + // followed by a WriteCaptureStreamMessage call. |
| + |
| + MockAecDump(); |
| + virtual ~MockAecDump(); |
| + |
| + 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_METHOD1(WriteInitMessage, |
| + void(const InternalAPMStreamsConfig& streams_config)); |
| + MOCK_METHOD1(WriteRenderStreamMessage, void(const AudioFrame& frame)); |
| + MOCK_METHOD1(WriteRenderStreamMessage, void(const FloatAudioFrame& src)); |
| + MOCK_METHOD0(WriteCaptureStreamMessage, void()); |
| + MOCK_METHOD2(WriteConfig, void(const InternalAPMConfig& config, bool forced)); |
| +}; |
| + |
| +} // namespace test |
| + |
| +} // namespace webrtc |
| + |
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ |