Index: webrtc/modules/audio_device/audio_device_buffer.h |
diff --git a/webrtc/modules/audio_device/audio_device_buffer.h b/webrtc/modules/audio_device/audio_device_buffer.h |
index 1095971040d37e41afc846f079c28977bcf39cd3..11b49c1100b50881f93e63272849cb76d4a101fa 100644 |
--- a/webrtc/modules/audio_device/audio_device_buffer.h |
+++ b/webrtc/modules/audio_device/audio_device_buffer.h |
@@ -19,103 +19,95 @@ namespace webrtc { |
class CriticalSectionWrapper; |
const uint32_t kPulsePeriodMs = 1000; |
-const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz |
+const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz |
class AudioDeviceObserver; |
-class AudioDeviceBuffer |
-{ |
-public: |
- AudioDeviceBuffer(); |
- virtual ~AudioDeviceBuffer(); |
+class AudioDeviceBuffer { |
+ public: |
+ AudioDeviceBuffer(); |
+ virtual ~AudioDeviceBuffer(); |
- void SetId(uint32_t id); |
- int32_t RegisterAudioCallback(AudioTransport* audioCallback); |
+ void SetId(uint32_t id); |
+ int32_t RegisterAudioCallback(AudioTransport* audioCallback); |
- int32_t InitPlayout(); |
- int32_t InitRecording(); |
+ int32_t InitPlayout(); |
+ int32_t InitRecording(); |
- virtual int32_t SetRecordingSampleRate(uint32_t fsHz); |
- virtual int32_t SetPlayoutSampleRate(uint32_t fsHz); |
- int32_t RecordingSampleRate() const; |
- int32_t PlayoutSampleRate() const; |
+ virtual int32_t SetRecordingSampleRate(uint32_t fsHz); |
+ virtual int32_t SetPlayoutSampleRate(uint32_t fsHz); |
+ int32_t RecordingSampleRate() const; |
+ int32_t PlayoutSampleRate() const; |
- virtual int32_t SetRecordingChannels(size_t channels); |
- virtual int32_t SetPlayoutChannels(size_t channels); |
- size_t RecordingChannels() const; |
- size_t PlayoutChannels() const; |
- int32_t SetRecordingChannel( |
- const AudioDeviceModule::ChannelType channel); |
- int32_t RecordingChannel( |
- AudioDeviceModule::ChannelType& channel) const; |
+ virtual int32_t SetRecordingChannels(size_t channels); |
+ virtual int32_t SetPlayoutChannels(size_t channels); |
+ size_t RecordingChannels() const; |
+ size_t PlayoutChannels() const; |
+ int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel); |
+ int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const; |
- virtual int32_t SetRecordedBuffer(const void* audioBuffer, |
- size_t nSamples); |
- int32_t SetCurrentMicLevel(uint32_t level); |
- virtual void SetVQEData(int playDelayMS, |
- int recDelayMS, |
- int clockDrift); |
- virtual int32_t DeliverRecordedData(); |
- uint32_t NewMicLevel() const; |
+ virtual int32_t SetRecordedBuffer(const void* audioBuffer, size_t nSamples); |
+ int32_t SetCurrentMicLevel(uint32_t level); |
+ virtual void SetVQEData(int playDelayMS, int recDelayMS, int clockDrift); |
+ virtual int32_t DeliverRecordedData(); |
+ uint32_t NewMicLevel() const; |
- virtual int32_t RequestPlayoutData(size_t nSamples); |
- virtual int32_t GetPlayoutData(void* audioBuffer); |
+ virtual int32_t RequestPlayoutData(size_t nSamples); |
+ virtual int32_t GetPlayoutData(void* audioBuffer); |
- int32_t StartInputFileRecording( |
- const char fileName[kAdmMaxFileNameSize]); |
- int32_t StopInputFileRecording(); |
- int32_t StartOutputFileRecording( |
- const char fileName[kAdmMaxFileNameSize]); |
- int32_t StopOutputFileRecording(); |
+ int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
+ int32_t StopInputFileRecording(); |
+ int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
+ int32_t StopOutputFileRecording(); |
- int32_t SetTypingStatus(bool typingStatus); |
+ int32_t SetTypingStatus(bool typingStatus); |
-private: |
- int32_t _id; |
- CriticalSectionWrapper& _critSect; |
- CriticalSectionWrapper& _critSectCb; |
+ private: |
+ int32_t _id; |
+ CriticalSectionWrapper& _critSect; |
+ CriticalSectionWrapper& _critSectCb; |
- AudioTransport* _ptrCbAudioTransport; |
+ AudioTransport* _ptrCbAudioTransport; |
- uint32_t _recSampleRate; |
- uint32_t _playSampleRate; |
+ uint32_t _recSampleRate; |
magjed_webrtc
2016/07/04 10:04:25
Will you fix variable names to follow the style gu
|
+ uint32_t _playSampleRate; |
- size_t _recChannels; |
- size_t _playChannels; |
+ size_t _recChannels; |
+ size_t _playChannels; |
- // selected recording channel (left/right/both) |
- AudioDeviceModule::ChannelType _recChannel; |
+ // selected recording channel (left/right/both) |
+ AudioDeviceModule::ChannelType _recChannel; |
- // 2 or 4 depending on mono or stereo |
- size_t _recBytesPerSample; |
- size_t _playBytesPerSample; |
+ // 2 or 4 depending on mono or stereo |
+ size_t _recBytesPerSample; |
+ size_t _playBytesPerSample; |
- // 10ms in stereo @ 96kHz |
- int8_t _recBuffer[kMaxBufferSizeBytes]; |
+ // 10ms in stereo @ 96kHz |
+ int8_t _recBuffer[kMaxBufferSizeBytes]; |
- // one sample <=> 2 or 4 bytes |
- size_t _recSamples; |
- size_t _recSize; // in bytes |
+ // one sample <=> 2 or 4 bytes |
+ size_t _recSamples; |
+ size_t _recSize; // in bytes |
- // 10ms in stereo @ 96kHz |
- int8_t _playBuffer[kMaxBufferSizeBytes]; |
+ // 10ms in stereo @ 96kHz |
+ int8_t _playBuffer[kMaxBufferSizeBytes]; |
- // one sample <=> 2 or 4 bytes |
- size_t _playSamples; |
- size_t _playSize; // in bytes |
+ // one sample <=> 2 or 4 bytes |
+ size_t _playSamples; |
+ size_t _playSize; // in bytes |
- FileWrapper& _recFile; |
- FileWrapper& _playFile; |
+ FileWrapper& _recFile; |
+ FileWrapper& _playFile; |
- uint32_t _currentMicLevel; |
- uint32_t _newMicLevel; |
+ uint32_t _currentMicLevel; |
+ uint32_t _newMicLevel; |
- bool _typingStatus; |
+ bool _typingStatus; |
- int _playDelayMS; |
- int _recDelayMS; |
- int _clockDrift; |
- int high_delay_counter_; |
+ int _playDelayMS; |
+ int _recDelayMS; |
+ int _clockDrift; |
+ int high_delay_counter_; |
}; |
} // namespace webrtc |