OLD | NEW |
---|---|
(Empty) | |
1 /* | |
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ | |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ | |
13 | |
14 #include <memory> | |
15 | |
16 #include "webrtc/modules/audio_processing/include/aec_dump.h" | |
17 #include "webrtc/modules/include/module_common_types.h" | |
18 #include "webrtc/test/gmock.h" | |
19 | |
20 namespace webrtc { | |
21 | |
22 namespace test { | |
23 | |
24 class MockAecDump : public AecDump { | |
25 public: | |
26 // 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
| |
27 // followed by a WriteCaptureStreamMessage call. | |
28 | |
29 MockAecDump(); | |
30 virtual ~MockAecDump(); | |
31 | |
32 MOCK_METHOD1(AddCaptureStreamInput, void(const FloatAudioFrame& src)); | |
33 | |
34 MOCK_METHOD1(AddCaptureStreamOutput, void(const FloatAudioFrame& src)); | |
35 MOCK_METHOD1(AddCaptureStreamInput, void(const AudioFrame& frame)); | |
36 MOCK_METHOD1(AddCaptureStreamOutput, void(const AudioFrame& frame)); | |
37 MOCK_METHOD1(AddAudioProcessingState, | |
38 void(const AudioProcessingState& state)); | |
39 MOCK_METHOD1(WriteInitMessage, | |
40 void(const InternalAPMStreamsConfig& streams_config)); | |
41 MOCK_METHOD1(WriteRenderStreamMessage, void(const AudioFrame& frame)); | |
42 MOCK_METHOD1(WriteRenderStreamMessage, void(const FloatAudioFrame& src)); | |
43 MOCK_METHOD0(WriteCaptureStreamMessage, void()); | |
44 MOCK_METHOD2(WriteConfig, void(const InternalAPMConfig& config, bool forced)); | |
45 }; | |
46 | |
47 } // namespace test | |
48 | |
49 } // namespace webrtc | |
50 | |
51 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ | |
OLD | NEW |