1 /* 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H 12 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H 13 14 #include "webrtc/modules/audio_device/include/audio_device.h" 15 #include "webrtc/system_wrappers/interface/file_wrapper.h" 16 #include "webrtc/typedefs.h" 17 18 namespace webrtc { 19 class CriticalSectionWrapper; 20 21 const uint32_t kPulsePeriodMs = 1000; 22 const uint32_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz 23 24 class AudioDeviceObserver; 25 class MediaFile; 26 27 class AudioDeviceBuffer 28 { 29 public: 30 AudioDeviceBuffer(); 31 virtual ~AudioDeviceBuffer(); 32 33 void SetId(uint32_t id); 34 int32_t RegisterAudioCallback(AudioTransport* audioCallback); 35 36 int32_t InitPlayout(); 37 int32_t InitRecording(); 38 39 virtual int32_t SetRecordingSampleRate(uint32_t fsHz); 40 virtual int32_t SetPlayoutSampleRate(uint32_t fsHz); 41 int32_t RecordingSampleRate() const; 42 int32_t PlayoutSampleRate() const; 43 44 virtual int32_t SetRecordingChannels(uint8_t channels); 45 virtual int32_t SetPlayoutChannels(uint8_t channels); 46 uint8_t RecordingChannels() const; 47 uint8_t PlayoutChannels() const; 48 int32_t SetRecordingChannel( 49 const AudioDeviceModule::ChannelType channel); 50 int32_t RecordingChannel( 51 AudioDeviceModule::ChannelType& channel) const; 52 53 virtual int32_t SetRecordedBuffer(const void* audioBuffer, 54 uint32_t nSamples); 55 int32_t SetCurrentMicLevel(uint32_t level); 56 virtual void SetVQEData(int playDelayMS, 57 int recDelayMS, 58 int clockDrift); 59 virtual int32_t DeliverRecordedData(); 60 uint32_t NewMicLevel() const; 61 62 virtual int32_t RequestPlayoutData(uint32_t nSamples); 63 virtual int32_t GetPlayoutData(void* audioBuffer); 64 65 int32_t StartInputFileRecording( 66 const char fileName[kAdmMaxFileNameSize]); 67 int32_t StopInputFileRecording(); 68 int32_t StartOutputFileRecording( 69 const char fileName[kAdmMaxFileNameSize]); 70 int32_t StopOutputFileRecording(); 71 72 int32_t SetTypingStatus(bool typingStatus); 73 74 private: 75 int32_t _id; 76 CriticalSectionWrapper& _critSect; 77 CriticalSectionWrapper& _critSectCb; 78 79 AudioTransport* _ptrCbAudioTransport; 80 81 uint32_t _recSampleRate; 82 uint32_t _playSampleRate; 83 84 uint8_t _recChannels; 85 uint8_t _playChannels; 86 87 // selected recording channel (left/right/both) 88 AudioDeviceModule::ChannelType _recChannel; 89 90 // 2 or 4 depending on mono or stereo 91 uint8_t _recBytesPerSample; 92 uint8_t _playBytesPerSample; 93 94 // 10ms in stereo @ 96kHz 95 int8_t _recBuffer[kMaxBufferSizeBytes]; 96 97 // one sample <=> 2 or 4 bytes 98 uint32_t _recSamples; 99 uint32_t _recSize; // in bytes 100 101 // 10ms in stereo @ 96kHz 102 int8_t _playBuffer[kMaxBufferSizeBytes]; 103 104 // one sample <=> 2 or 4 bytes 105 uint32_t _playSamples; 106 uint32_t _playSize; // in bytes 107 108 FileWrapper& _recFile; 109 FileWrapper& _playFile; 110 111 uint32_t _currentMicLevel; 112 uint32_t _newMicLevel; 113 114 bool _typingStatus; 115 116 int _playDelayMS; 117 int _recDelayMS; 118 int _clockDrift; 119 int high_delay_counter_; 120 }; 121 122 } // namespace webrtc 123 124 #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H 125
