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

Issue 1413483003: Added option to specify a maximum file size when recording an AEC dump. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added function to avoid breaking Chromium. Created 5 years 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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243 MOCK_CONST_METHOD0(was_stream_delay_set, 243 MOCK_CONST_METHOD0(was_stream_delay_set,
244 bool()); 244 bool());
245 MOCK_METHOD1(set_stream_key_pressed, 245 MOCK_METHOD1(set_stream_key_pressed,
246 void(bool key_pressed)); 246 void(bool key_pressed));
247 MOCK_CONST_METHOD0(stream_key_pressed, 247 MOCK_CONST_METHOD0(stream_key_pressed,
248 bool()); 248 bool());
249 MOCK_METHOD1(set_delay_offset_ms, 249 MOCK_METHOD1(set_delay_offset_ms,
250 void(int offset)); 250 void(int offset));
251 MOCK_CONST_METHOD0(delay_offset_ms, 251 MOCK_CONST_METHOD0(delay_offset_ms,
252 int()); 252 int());
253 MOCK_METHOD1(StartDebugRecording, 253 MOCK_METHOD2(StartDebugRecording,
254 int(const char filename[kMaxFilenameSize])); 254 int(const char filename[kMaxFilenameSize],
255 MOCK_METHOD1(StartDebugRecording, 255 int64_t max_log_size_bytes));
256 int(FILE* handle)); 256 MOCK_METHOD2(StartDebugRecording,
257 int(FILE* handle, int64_t max_log_size_bytes));
257 MOCK_METHOD0(StopDebugRecording, 258 MOCK_METHOD0(StopDebugRecording,
258 int()); 259 int());
259 MOCK_METHOD0(UpdateHistogramsOnCallEnd, void()); 260 MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
260 virtual MockEchoCancellation* echo_cancellation() const { 261 virtual MockEchoCancellation* echo_cancellation() const {
261 return echo_cancellation_.get(); 262 return echo_cancellation_.get();
262 } 263 }
263 virtual MockEchoControlMobile* echo_control_mobile() const { 264 virtual MockEchoControlMobile* echo_control_mobile() const {
264 return echo_control_mobile_.get(); 265 return echo_control_mobile_.get();
265 } 266 }
266 virtual MockGainControl* gain_control() const { 267 virtual MockGainControl* gain_control() const {
(...skipping 18 matching lines...) Expand all
285 rtc::scoped_ptr<MockGainControl> gain_control_; 286 rtc::scoped_ptr<MockGainControl> gain_control_;
286 rtc::scoped_ptr<MockHighPassFilter> high_pass_filter_; 287 rtc::scoped_ptr<MockHighPassFilter> high_pass_filter_;
287 rtc::scoped_ptr<MockLevelEstimator> level_estimator_; 288 rtc::scoped_ptr<MockLevelEstimator> level_estimator_;
288 rtc::scoped_ptr<MockNoiseSuppression> noise_suppression_; 289 rtc::scoped_ptr<MockNoiseSuppression> noise_suppression_;
289 rtc::scoped_ptr<MockVoiceDetection> voice_detection_; 290 rtc::scoped_ptr<MockVoiceDetection> voice_detection_;
290 }; 291 };
291 292
292 } // namespace webrtc 293 } // namespace webrtc
293 294
294 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_ 295 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
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