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Side by Side Diff: webrtc/voice_engine/transmit_mixer.h

Issue 2049683003: FileRecorder + FilePlayer: Let Create functions return unique_ptr (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove3
Patch Set: Created 4 years, 4 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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
11 #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H 11 #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
12 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H 12 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
13 13
14 #include <memory>
15
14 #include "webrtc/base/criticalsection.h" 16 #include "webrtc/base/criticalsection.h"
15 #include "webrtc/common_audio/resampler/include/push_resampler.h" 17 #include "webrtc/common_audio/resampler/include/push_resampler.h"
16 #include "webrtc/common_types.h" 18 #include "webrtc/common_types.h"
17 #include "webrtc/modules/audio_processing/typing_detection.h" 19 #include "webrtc/modules/audio_processing/typing_detection.h"
18 #include "webrtc/modules/include/module_common_types.h" 20 #include "webrtc/modules/include/module_common_types.h"
19 #include "webrtc/modules/utility/include/file_player.h" 21 #include "webrtc/modules/utility/include/file_player.h"
20 #include "webrtc/modules/utility/include/file_recorder.h" 22 #include "webrtc/modules/utility/include/file_recorder.h"
21 #include "webrtc/voice_engine/include/voe_base.h" 23 #include "webrtc/voice_engine/include/voe_base.h"
22 #include "webrtc/voice_engine/level_indicator.h" 24 #include "webrtc/voice_engine/level_indicator.h"
23 #include "webrtc/voice_engine/monitor_module.h" 25 #include "webrtc/voice_engine/monitor_module.h"
(...skipping 165 matching lines...) Expand 10 before | Expand all | Expand 10 after
189 Statistics* _engineStatisticsPtr; 191 Statistics* _engineStatisticsPtr;
190 ChannelManager* _channelManagerPtr; 192 ChannelManager* _channelManagerPtr;
191 AudioProcessing* audioproc_; 193 AudioProcessing* audioproc_;
192 VoiceEngineObserver* _voiceEngineObserverPtr; 194 VoiceEngineObserver* _voiceEngineObserverPtr;
193 ProcessThread* _processThreadPtr; 195 ProcessThread* _processThreadPtr;
194 196
195 // owns 197 // owns
196 MonitorModule _monitorModule; 198 MonitorModule _monitorModule;
197 AudioFrame _audioFrame; 199 AudioFrame _audioFrame;
198 PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate 200 PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate
199 FilePlayer* _filePlayerPtr; 201 std::unique_ptr<FilePlayer> file_player_;
200 FileRecorder* _fileRecorderPtr; 202 std::unique_ptr<FileRecorder> file_recorder_;
201 FileRecorder* _fileCallRecorderPtr; 203 std::unique_ptr<FileRecorder> file_call_recorder_;
202 int _filePlayerId; 204 int _filePlayerId;
203 int _fileRecorderId; 205 int _fileRecorderId;
204 int _fileCallRecorderId; 206 int _fileCallRecorderId;
205 bool _filePlaying; 207 bool _filePlaying;
206 bool _fileRecording; 208 bool _fileRecording;
207 bool _fileCallRecording; 209 bool _fileCallRecording;
208 voe::AudioLevel _audioLevel; 210 voe::AudioLevel _audioLevel;
209 // protect file instances and their variables in MixedParticipants() 211 // protect file instances and their variables in MixedParticipants()
210 rtc::CriticalSection _critSect; 212 rtc::CriticalSection _critSect;
211 rtc::CriticalSection _callbackCritSect; 213 rtc::CriticalSection _callbackCritSect;
(...skipping 13 matching lines...) Expand all
225 bool _mute; 227 bool _mute;
226 bool stereo_codec_; 228 bool stereo_codec_;
227 bool swap_stereo_channels_; 229 bool swap_stereo_channels_;
228 }; 230 };
229 231
230 } // namespace voe 232 } // namespace voe
231 233
232 } // namespace webrtc 234 } // namespace webrtc
233 235
234 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H 236 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
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