 Chromium Code Reviews
 Chromium Code Reviews Issue 2119093003:
  clang-format on AudioDeviceBuffer  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master
    
  
    Issue 2119093003:
  clang-format on AudioDeviceBuffer  (Closed) 
  Base URL: https://chromium.googlesource.com/external/webrtc.git@master| OLD | NEW | 
|---|---|
| 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_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 11 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 
| 12 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 12 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 
| 13 | 13 | 
| 14 #include "webrtc/modules/audio_device/include/audio_device.h" | 14 #include "webrtc/modules/audio_device/include/audio_device.h" | 
| 15 #include "webrtc/system_wrappers/include/file_wrapper.h" | 15 #include "webrtc/system_wrappers/include/file_wrapper.h" | 
| 16 #include "webrtc/typedefs.h" | 16 #include "webrtc/typedefs.h" | 
| 17 | 17 | 
| 18 namespace webrtc { | 18 namespace webrtc { | 
| 19 class CriticalSectionWrapper; | 19 class CriticalSectionWrapper; | 
| 20 | 20 | 
| 21 const uint32_t kPulsePeriodMs = 1000; | 21 const uint32_t kPulsePeriodMs = 1000; | 
| 22 const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz | 22 const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz | 
| 23 | 23 | 
| 24 class AudioDeviceObserver; | 24 class AudioDeviceObserver; | 
| 25 | 25 | 
| 26 class AudioDeviceBuffer | 26 class AudioDeviceBuffer { | 
| 27 { | 27 public: | 
| 28 public: | 28 AudioDeviceBuffer(); | 
| 29 AudioDeviceBuffer(); | 29 virtual ~AudioDeviceBuffer(); | 
| 30 virtual ~AudioDeviceBuffer(); | |
| 31 | 30 | 
| 32 void SetId(uint32_t id); | 31 void SetId(uint32_t id); | 
| 33 int32_t RegisterAudioCallback(AudioTransport* audioCallback); | 32 int32_t RegisterAudioCallback(AudioTransport* audioCallback); | 
| 34 | 33 | 
| 35 int32_t InitPlayout(); | 34 int32_t InitPlayout(); | 
| 36 int32_t InitRecording(); | 35 int32_t InitRecording(); | 
| 37 | 36 | 
| 38 virtual int32_t SetRecordingSampleRate(uint32_t fsHz); | 37 virtual int32_t SetRecordingSampleRate(uint32_t fsHz); | 
| 39 virtual int32_t SetPlayoutSampleRate(uint32_t fsHz); | 38 virtual int32_t SetPlayoutSampleRate(uint32_t fsHz); | 
| 40 int32_t RecordingSampleRate() const; | 39 int32_t RecordingSampleRate() const; | 
| 41 int32_t PlayoutSampleRate() const; | 40 int32_t PlayoutSampleRate() const; | 
| 42 | 41 | 
| 43 virtual int32_t SetRecordingChannels(size_t channels); | 42 virtual int32_t SetRecordingChannels(size_t channels); | 
| 44 virtual int32_t SetPlayoutChannels(size_t channels); | 43 virtual int32_t SetPlayoutChannels(size_t channels); | 
| 45 size_t RecordingChannels() const; | 44 size_t RecordingChannels() const; | 
| 46 size_t PlayoutChannels() const; | 45 size_t PlayoutChannels() const; | 
| 47 int32_t SetRecordingChannel( | 46 int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel); | 
| 48 const AudioDeviceModule::ChannelType channel); | 47 int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const; | 
| 49 int32_t RecordingChannel( | |
| 50 AudioDeviceModule::ChannelType& channel) const; | |
| 51 | 48 | 
| 52 virtual int32_t SetRecordedBuffer(const void* audioBuffer, | 49 virtual int32_t SetRecordedBuffer(const void* audioBuffer, size_t nSamples); | 
| 53 size_t nSamples); | 50 int32_t SetCurrentMicLevel(uint32_t level); | 
| 54 int32_t SetCurrentMicLevel(uint32_t level); | 51 virtual void SetVQEData(int playDelayMS, int recDelayMS, int clockDrift); | 
| 55 virtual void SetVQEData(int playDelayMS, | 52 virtual int32_t DeliverRecordedData(); | 
| 56 int recDelayMS, | 53 uint32_t NewMicLevel() const; | 
| 57 int clockDrift); | |
| 58 virtual int32_t DeliverRecordedData(); | |
| 59 uint32_t NewMicLevel() const; | |
| 60 | 54 | 
| 61 virtual int32_t RequestPlayoutData(size_t nSamples); | 55 virtual int32_t RequestPlayoutData(size_t nSamples); | 
| 62 virtual int32_t GetPlayoutData(void* audioBuffer); | 56 virtual int32_t GetPlayoutData(void* audioBuffer); | 
| 63 | 57 | 
| 64 int32_t StartInputFileRecording( | 58 int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); | 
| 65 const char fileName[kAdmMaxFileNameSize]); | 59 int32_t StopInputFileRecording(); | 
| 66 int32_t StopInputFileRecording(); | 60 int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); | 
| 67 int32_t StartOutputFileRecording( | 61 int32_t StopOutputFileRecording(); | 
| 68 const char fileName[kAdmMaxFileNameSize]); | |
| 69 int32_t StopOutputFileRecording(); | |
| 70 | 62 | 
| 71 int32_t SetTypingStatus(bool typingStatus); | 63 int32_t SetTypingStatus(bool typingStatus); | 
| 72 | 64 | 
| 73 private: | 65 private: | 
| 74 int32_t _id; | 66 int32_t _id; | 
| 75 CriticalSectionWrapper& _critSect; | 67 CriticalSectionWrapper& _critSect; | 
| 76 CriticalSectionWrapper& _critSectCb; | 68 CriticalSectionWrapper& _critSectCb; | 
| 77 | 69 | 
| 78 AudioTransport* _ptrCbAudioTransport; | 70 AudioTransport* _ptrCbAudioTransport; | 
| 79 | 71 | 
| 80 uint32_t _recSampleRate; | 72 uint32_t _recSampleRate; | 
| 
magjed_webrtc
2016/07/04 10:04:25
Will you fix variable names to follow the style gu
 | |
| 81 uint32_t _playSampleRate; | 73 uint32_t _playSampleRate; | 
| 82 | 74 | 
| 83 size_t _recChannels; | 75 size_t _recChannels; | 
| 84 size_t _playChannels; | 76 size_t _playChannels; | 
| 85 | 77 | 
| 86 // selected recording channel (left/right/both) | 78 // selected recording channel (left/right/both) | 
| 87 AudioDeviceModule::ChannelType _recChannel; | 79 AudioDeviceModule::ChannelType _recChannel; | 
| 88 | 80 | 
| 89 // 2 or 4 depending on mono or stereo | 81 // 2 or 4 depending on mono or stereo | 
| 90 size_t _recBytesPerSample; | 82 size_t _recBytesPerSample; | 
| 91 size_t _playBytesPerSample; | 83 size_t _playBytesPerSample; | 
| 92 | 84 | 
| 93 // 10ms in stereo @ 96kHz | 85 // 10ms in stereo @ 96kHz | 
| 94 int8_t _recBuffer[kMaxBufferSizeBytes]; | 86 int8_t _recBuffer[kMaxBufferSizeBytes]; | 
| 95 | 87 | 
| 96 // one sample <=> 2 or 4 bytes | 88 // one sample <=> 2 or 4 bytes | 
| 97 size_t _recSamples; | 89 size_t _recSamples; | 
| 98 size_t _recSize; // in bytes | 90 size_t _recSize; // in bytes | 
| 99 | 91 | 
| 100 // 10ms in stereo @ 96kHz | 92 // 10ms in stereo @ 96kHz | 
| 101 int8_t _playBuffer[kMaxBufferSizeBytes]; | 93 int8_t _playBuffer[kMaxBufferSizeBytes]; | 
| 102 | 94 | 
| 103 // one sample <=> 2 or 4 bytes | 95 // one sample <=> 2 or 4 bytes | 
| 104 size_t _playSamples; | 96 size_t _playSamples; | 
| 105 size_t _playSize; // in bytes | 97 size_t _playSize; // in bytes | 
| 106 | 98 | 
| 107 FileWrapper& _recFile; | 99 FileWrapper& _recFile; | 
| 108 FileWrapper& _playFile; | 100 FileWrapper& _playFile; | 
| 109 | 101 | 
| 110 uint32_t _currentMicLevel; | 102 uint32_t _currentMicLevel; | 
| 111 uint32_t _newMicLevel; | 103 uint32_t _newMicLevel; | 
| 112 | 104 | 
| 113 bool _typingStatus; | 105 bool _typingStatus; | 
| 114 | 106 | 
| 115 int _playDelayMS; | 107 int _playDelayMS; | 
| 116 int _recDelayMS; | 108 int _recDelayMS; | 
| 117 int _clockDrift; | 109 int _clockDrift; | 
| 118 int high_delay_counter_; | 110 int high_delay_counter_; | 
| 119 }; | 111 }; | 
| 120 | 112 | 
| 121 } // namespace webrtc | 113 } // namespace webrtc | 
| 122 | 114 | 
| 123 #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 115 #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H | 
| OLD | NEW |