/* * 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 #include #include 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; CriticalSectionWrapper& _critSectCb; 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_