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Side by Side Diff: webrtc/modules/audio_processing/include/mock_audio_processing.h

Issue 2406193002: Made the AudioProcessing class a pure interface. (Closed)
Patch Set: Corrected mock of the AudioProcessing interface Created 4 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 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 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 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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252 bool()); 252 bool());
253 MOCK_METHOD1(set_delay_offset_ms, 253 MOCK_METHOD1(set_delay_offset_ms,
254 void(int offset)); 254 void(int offset));
255 MOCK_CONST_METHOD0(delay_offset_ms, 255 MOCK_CONST_METHOD0(delay_offset_ms,
256 int()); 256 int());
257 MOCK_METHOD2(StartDebugRecording, 257 MOCK_METHOD2(StartDebugRecording,
258 int(const char filename[kMaxFilenameSize], 258 int(const char filename[kMaxFilenameSize],
259 int64_t max_log_size_bytes)); 259 int64_t max_log_size_bytes));
260 MOCK_METHOD2(StartDebugRecording, 260 MOCK_METHOD2(StartDebugRecording,
261 int(FILE* handle, int64_t max_log_size_bytes)); 261 int(FILE* handle, int64_t max_log_size_bytes));
262 MOCK_METHOD1(StartDebugRecording,
263 int (FILE* handle));
264 MOCK_METHOD1(StartDebugRecording,
265 int(rtc::PlatformFile handle));
262 MOCK_METHOD0(StopDebugRecording, 266 MOCK_METHOD0(StopDebugRecording,
263 int()); 267 int());
264 MOCK_METHOD0(UpdateHistogramsOnCallEnd, void()); 268 MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
265 virtual MockEchoCancellation* echo_cancellation() const { 269 virtual MockEchoCancellation* echo_cancellation() const {
266 return echo_cancellation_.get(); 270 return echo_cancellation_.get();
267 } 271 }
268 virtual MockEchoControlMobile* echo_control_mobile() const { 272 virtual MockEchoControlMobile* echo_control_mobile() const {
269 return echo_control_mobile_.get(); 273 return echo_control_mobile_.get();
270 } 274 }
271 virtual MockGainControl* gain_control() const { 275 virtual MockGainControl* gain_control() const {
(...skipping 18 matching lines...) Expand all
290 std::unique_ptr<MockGainControl> gain_control_; 294 std::unique_ptr<MockGainControl> gain_control_;
291 std::unique_ptr<MockHighPassFilter> high_pass_filter_; 295 std::unique_ptr<MockHighPassFilter> high_pass_filter_;
292 std::unique_ptr<MockLevelEstimator> level_estimator_; 296 std::unique_ptr<MockLevelEstimator> level_estimator_;
293 std::unique_ptr<MockNoiseSuppression> noise_suppression_; 297 std::unique_ptr<MockNoiseSuppression> noise_suppression_;
294 std::unique_ptr<MockVoiceDetection> voice_detection_; 298 std::unique_ptr<MockVoiceDetection> voice_detection_;
295 }; 299 };
296 300
297 } // namespace webrtc 301 } // namespace webrtc
298 302
299 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_ 303 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
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