| /* |
| * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H |
| #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H |
| |
| #include "audio_device_generic.h" |
| #include "critical_section_wrapper.h" |
| #include "audio_mixer_manager_mac.h" |
| |
| #include <CoreAudio/CoreAudio.h> |
| #include <AudioToolbox/AudioConverter.h> |
| #include <mach/semaphore.h> |
| |
| struct PaUtilRingBuffer; |
| |
| namespace webrtc |
| { |
| class EventWrapper; |
| class ThreadWrapper; |
| |
| const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 48000; |
| const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 48000; |
| |
| const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording |
| const WebRtc_UWord32 N_PLAY_CHANNELS = 2; // default is stereo playout |
| const WebRtc_UWord32 N_DEVICE_CHANNELS = 8; |
| |
| const WebRtc_UWord32 ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC |
| / 100); |
| const WebRtc_UWord32 ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC |
| / 100); |
| |
| enum |
| { |
| N_BLOCKS_IO = 2 |
| }; |
| enum |
| { |
| N_BUFFERS_IN = 10 |
| }; |
| enum |
| { |
| N_BUFFERS_OUT = 3 |
| }; // Must be at least N_BLOCKS_IO |
| |
| const WebRtc_UWord32 TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000); |
| |
| const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (ENGINE_REC_BUF_SIZE_IN_SAMPLES |
| * N_DEVICE_CHANNELS * N_BUFFERS_IN); |
| const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = |
| (ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT); |
| |
| class AudioDeviceMac: public AudioDeviceGeneric |
| { |
| public: |
| AudioDeviceMac(const WebRtc_Word32 id); |
| ~AudioDeviceMac(); |
| |
| // Retrieve the currently utilized audio layer |
| virtual WebRtc_Word32 |
| ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const; |
| |
| // Main initializaton and termination |
| virtual WebRtc_Word32 Init(); |
| virtual WebRtc_Word32 Terminate(); |
| virtual bool Initialized() const; |
| |
| // Device enumeration |
| virtual WebRtc_Word16 PlayoutDevices(); |
| virtual WebRtc_Word16 RecordingDevices(); |
| virtual WebRtc_Word32 PlayoutDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]); |
| virtual WebRtc_Word32 RecordingDeviceName( |
| WebRtc_UWord16 index, |
| WebRtc_Word8 name[kAdmMaxDeviceNameSize], |
| WebRtc_Word8 guid[kAdmMaxGuidSize]); |
| |
| // Device selection |
| virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index); |
| virtual WebRtc_Word32 SetPlayoutDevice( |
| AudioDeviceModule::WindowsDeviceType device); |
| virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index); |
| virtual WebRtc_Word32 SetRecordingDevice( |
| AudioDeviceModule::WindowsDeviceType device); |
| |
| // Audio transport initialization |
| virtual WebRtc_Word32 PlayoutIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitPlayout(); |
| virtual bool PlayoutIsInitialized() const; |
| virtual WebRtc_Word32 RecordingIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitRecording(); |
| virtual bool RecordingIsInitialized() const; |
| |
| // Audio transport control |
| virtual WebRtc_Word32 StartPlayout(); |
| virtual WebRtc_Word32 StopPlayout(); |
| virtual bool Playing() const; |
| virtual WebRtc_Word32 StartRecording(); |
| virtual WebRtc_Word32 StopRecording(); |
| virtual bool Recording() const; |
| |
| // Microphone Automatic Gain Control (AGC) |
| virtual WebRtc_Word32 SetAGC(bool enable); |
| virtual bool AGC() const; |
| |
| // Volume control based on the Windows Wave API (Windows only) |
| virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, |
| WebRtc_UWord16 volumeRight); |
| virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, |
| WebRtc_UWord16& volumeRight) const; |
| |
| // Audio mixer initialization |
| virtual WebRtc_Word32 SpeakerIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitSpeaker(); |
| virtual bool SpeakerIsInitialized() const; |
| virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available); |
| virtual WebRtc_Word32 InitMicrophone(); |
| virtual bool MicrophoneIsInitialized() const; |
| |
| // Speaker volume controls |
| virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const; |
| virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const; |
| virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const; |
| virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const; |
| |
| // Microphone volume controls |
| virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume); |
| virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const; |
| virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const; |
| virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const; |
| virtual WebRtc_Word32 |
| MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const; |
| |
| // Microphone mute control |
| virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneMute(bool enable); |
| virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const; |
| |
| // Speaker mute control |
| virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetSpeakerMute(bool enable); |
| virtual WebRtc_Word32 SpeakerMute(bool& enabled) const; |
| |
| // Microphone boost control |
| virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneBoost(bool enable); |
| virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const; |
| |
| // Stereo support |
| virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetStereoPlayout(bool enable); |
| virtual WebRtc_Word32 StereoPlayout(bool& enabled) const; |
| virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetStereoRecording(bool enable); |
| virtual WebRtc_Word32 StereoRecording(bool& enabled) const; |
| |
| // Delay information and control |
| virtual WebRtc_Word32 |
| SetPlayoutBuffer(const AudioDeviceModule::BufferType type, |
| WebRtc_UWord16 sizeMS); |
| virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, |
| WebRtc_UWord16& sizeMS) const; |
| virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const; |
| virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const; |
| |
| // CPU load |
| virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const; |
| |
| public: |
| virtual bool PlayoutWarning() const; |
| virtual bool PlayoutError() const; |
| virtual bool RecordingWarning() const; |
| virtual bool RecordingError() const; |
| virtual void ClearPlayoutWarning(); |
| virtual void ClearPlayoutError(); |
| virtual void ClearRecordingWarning(); |
| virtual void ClearRecordingError(); |
| |
| public: |
| virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); |
| |
| private: |
| void Lock() |
| { |
| _critSect.Enter(); |
| } |
| ; |
| void UnLock() |
| { |
| _critSect.Leave(); |
| } |
| ; |
| WebRtc_Word32 Id() |
| { |
| return _id; |
| } |
| |
| static void AtomicSet32(int32_t* theValue, int32_t newValue); |
| static int32_t AtomicGet32(int32_t* theValue); |
| |
| static void logCAMsg(const TraceLevel level, |
| const TraceModule module, |
| const WebRtc_Word32 id, const char *msg, |
| const char *err); |
| |
| WebRtc_Word32 GetNumberDevices(const AudioObjectPropertyScope scope, |
| AudioDeviceID scopedDeviceIds[], |
| const WebRtc_UWord32 deviceListLength); |
| |
| WebRtc_Word32 GetDeviceName(const AudioObjectPropertyScope scope, |
| const WebRtc_UWord16 index, char* name); |
| |
| WebRtc_Word32 InitDevice(WebRtc_UWord16 userDeviceIndex, |
| AudioDeviceID& deviceId, bool isInput); |
| |
| static OSStatus |
| objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses, |
| const AudioObjectPropertyAddress addresses[], |
| void* clientData); |
| |
| OSStatus |
| implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses, |
| const AudioObjectPropertyAddress addresses[]); |
| |
| WebRtc_Word32 HandleDeviceChange(); |
| |
| WebRtc_Word32 |
| HandleStreamFormatChange(AudioObjectID objectId, |
| AudioObjectPropertyAddress propertyAddress); |
| |
| WebRtc_Word32 |
| HandleDataSourceChange(AudioObjectID objectId, |
| AudioObjectPropertyAddress propertyAddress); |
| |
| WebRtc_Word32 |
| HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress); |
| |
| private: |
| static OSStatus deviceIOProc(AudioDeviceID device, |
| const AudioTimeStamp *now, |
| const AudioBufferList *inputData, |
| const AudioTimeStamp *inputTime, |
| AudioBufferList *outputData, |
| const AudioTimeStamp* outputTime, |
| void *clientData); |
| |
| static OSStatus |
| outConverterProc(AudioConverterRef audioConverter, |
| UInt32 *numberDataPackets, AudioBufferList *data, |
| AudioStreamPacketDescription **dataPacketDescription, |
| void *userData); |
| |
| static OSStatus inDeviceIOProc(AudioDeviceID device, |
| const AudioTimeStamp *now, |
| const AudioBufferList *inputData, |
| const AudioTimeStamp *inputTime, |
| AudioBufferList *outputData, |
| const AudioTimeStamp *outputTime, |
| void *clientData); |
| |
| static OSStatus |
| inConverterProc(AudioConverterRef audioConverter, |
| UInt32 *numberDataPackets, AudioBufferList *data, |
| AudioStreamPacketDescription **dataPacketDescription, |
| void *inUserData); |
| |
| OSStatus implDeviceIOProc(const AudioBufferList *inputData, |
| const AudioTimeStamp *inputTime, |
| AudioBufferList *outputData, |
| const AudioTimeStamp *outputTime); |
| |
| OSStatus implOutConverterProc(UInt32 *numberDataPackets, |
| AudioBufferList *data); |
| |
| OSStatus implInDeviceIOProc(const AudioBufferList *inputData, |
| const AudioTimeStamp *inputTime); |
| |
| OSStatus implInConverterProc(UInt32 *numberDataPackets, |
| AudioBufferList *data); |
| |
| static bool RunCapture(void*); |
| static bool RunRender(void*); |
| bool CaptureWorkerThread(); |
| bool RenderWorkerThread(); |
| |
| private: |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| |
| CriticalSectionWrapper& _critSect; |
| |
| EventWrapper& _stopEventRec; |
| EventWrapper& _stopEvent; |
| |
| ThreadWrapper* _captureWorkerThread; |
| ThreadWrapper* _renderWorkerThread; |
| WebRtc_UWord32 _captureWorkerThreadId; |
| WebRtc_UWord32 _renderWorkerThreadId; |
| |
| WebRtc_Word32 _id; |
| |
| AudioMixerManagerMac _mixerManager; |
| |
| WebRtc_UWord16 _inputDeviceIndex; |
| WebRtc_UWord16 _outputDeviceIndex; |
| AudioDeviceID _inputDeviceID; |
| AudioDeviceID _outputDeviceID; |
| #if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050 |
| AudioDeviceIOProcID _inDeviceIOProcID; |
| AudioDeviceIOProcID _deviceIOProcID; |
| #endif |
| bool _inputDeviceIsSpecified; |
| bool _outputDeviceIsSpecified; |
| |
| WebRtc_UWord8 _recChannels; |
| WebRtc_UWord8 _playChannels; |
| |
| Float32* _captureBufData; |
| SInt16* _renderBufData; |
| |
| SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES]; |
| |
| AudioDeviceModule::BufferType _playBufType; |
| |
| private: |
| bool _initialized; |
| bool _isShutDown; |
| bool _recording; |
| bool _playing; |
| bool _recIsInitialized; |
| bool _playIsInitialized; |
| bool _startRec; |
| bool _stopRec; |
| bool _stopPlay; |
| bool _AGC; |
| |
| // Atomically set varaibles |
| int32_t _renderDeviceIsAlive; |
| int32_t _captureDeviceIsAlive; |
| |
| bool _twoDevices; |
| bool _doStop; // For play if not shared device or play+rec if shared device |
| bool _doStopRec; // For rec if not shared device |
| bool _macBookPro; |
| bool _macBookProPanRight; |
| bool _stereoRender; |
| bool _stereoRenderRequested; |
| |
| AudioConverterRef _captureConverter; |
| AudioConverterRef _renderConverter; |
| |
| AudioStreamBasicDescription _outStreamFormat; |
| AudioStreamBasicDescription _outDesiredFormat; |
| AudioStreamBasicDescription _inStreamFormat; |
| AudioStreamBasicDescription _inDesiredFormat; |
| |
| WebRtc_UWord32 _captureLatencyUs; |
| WebRtc_UWord32 _renderLatencyUs; |
| |
| // Atomically set variables |
| mutable int32_t _captureDelayUs; |
| mutable int32_t _renderDelayUs; |
| |
| WebRtc_Word32 _renderDelayOffsetSamples; |
| |
| private: |
| WebRtc_UWord16 _playBufDelay; // playback delay |
| WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay |
| |
| WebRtc_UWord16 _playWarning; |
| WebRtc_UWord16 _playError; |
| WebRtc_UWord16 _recWarning; |
| WebRtc_UWord16 _recError; |
| |
| PaUtilRingBuffer* _paCaptureBuffer; |
| PaUtilRingBuffer* _paRenderBuffer; |
| |
| semaphore_t _renderSemaphore; |
| semaphore_t _captureSemaphore; |
| |
| WebRtc_UWord32 _captureBufSizeSamples; |
| WebRtc_UWord32 _renderBufSizeSamples; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_ |
