| /* |
| * Copyright (c) 2012 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_CORE_WIN_H |
| #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H |
| |
| #if (_MSC_VER >= 1400) // only include for VS 2005 and higher |
| |
| #include "audio_device_generic.h" |
| |
| #pragma once |
| #include <wmcodecdsp.h> // CLSID_CWMAudioAEC |
| // (must be before audioclient.h) |
| #include <Audioclient.h> // WASAPI |
| #include <Audiopolicy.h> |
| #include <avrt.h> // Avrt |
| #include <endpointvolume.h> |
| #include <mediaobj.h> // IMediaObject |
| #include <Mmdeviceapi.h> // MMDevice |
| |
| #include "atomic32_wrapper.h" |
| #include "critical_section_wrapper.h" |
| #include "scoped_refptr.h" |
| |
| // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority |
| #pragma comment( lib, "avrt.lib" ) |
| // AVRT function pointers |
| typedef BOOL (WINAPI *PAvRevertMmThreadCharacteristics)(HANDLE); |
| typedef HANDLE (WINAPI *PAvSetMmThreadCharacteristicsA)(LPCSTR, LPDWORD); |
| typedef BOOL (WINAPI *PAvSetMmThreadPriority)(HANDLE, AVRT_PRIORITY); |
| |
| namespace webrtc { |
| |
| const float MAX_CORE_SPEAKER_VOLUME = 255.0f; |
| const float MIN_CORE_SPEAKER_VOLUME = 0.0f; |
| const float MAX_CORE_MICROPHONE_VOLUME = 255.0f; |
| const float MIN_CORE_MICROPHONE_VOLUME = 0.0f; |
| const WebRtc_UWord16 CORE_SPEAKER_VOLUME_STEP_SIZE = 1; |
| const WebRtc_UWord16 CORE_MICROPHONE_VOLUME_STEP_SIZE = 1; |
| |
| // Utility class which initializes COM in the constructor (STA or MTA), |
| // and uninitializes COM in the destructor. |
| class ScopedCOMInitializer { |
| public: |
| // Enum value provided to initialize the thread as an MTA instead of STA. |
| enum SelectMTA { kMTA }; |
| |
| // Constructor for STA initialization. |
| ScopedCOMInitializer() { |
| Initialize(COINIT_APARTMENTTHREADED); |
| } |
| |
| // Constructor for MTA initialization. |
| explicit ScopedCOMInitializer(SelectMTA mta) { |
| Initialize(COINIT_MULTITHREADED); |
| } |
| |
| ScopedCOMInitializer::~ScopedCOMInitializer() { |
| if (SUCCEEDED(hr_)) |
| CoUninitialize(); |
| } |
| |
| bool succeeded() const { return SUCCEEDED(hr_); } |
| |
| private: |
| void Initialize(COINIT init) { |
| hr_ = CoInitializeEx(NULL, init); |
| } |
| |
| HRESULT hr_; |
| |
| ScopedCOMInitializer(const ScopedCOMInitializer&); |
| void operator=(const ScopedCOMInitializer&); |
| }; |
| |
| |
| class AudioDeviceWindowsCore : public AudioDeviceGeneric |
| { |
| public: |
| AudioDeviceWindowsCore(const WebRtc_Word32 id); |
| ~AudioDeviceWindowsCore(); |
| |
| static bool CoreAudioIsSupported(); |
| |
| // 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; |
| |
| // Speaker mute control |
| virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetSpeakerMute(bool enable); |
| virtual WebRtc_Word32 SpeakerMute(bool& enabled) const; |
| |
| // Microphone mute control |
| virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available); |
| virtual WebRtc_Word32 SetMicrophoneMute(bool enable); |
| virtual WebRtc_Word32 MicrophoneMute(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; |
| |
| virtual int32_t EnableBuiltInAEC(bool enable); |
| virtual bool BuiltInAECIsEnabled() 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: // avrt function pointers |
| PAvRevertMmThreadCharacteristics _PAvRevertMmThreadCharacteristics; |
| PAvSetMmThreadCharacteristicsA _PAvSetMmThreadCharacteristicsA; |
| PAvSetMmThreadPriority _PAvSetMmThreadPriority; |
| HMODULE _avrtLibrary; |
| bool _winSupportAvrt; |
| |
| private: // thread functions |
| DWORD InitCaptureThreadPriority(); |
| void RevertCaptureThreadPriority(); |
| static DWORD WINAPI WSAPICaptureThread(LPVOID context); |
| DWORD DoCaptureThread(); |
| |
| static DWORD WINAPI WSAPICaptureThreadPollDMO(LPVOID context); |
| DWORD DoCaptureThreadPollDMO(); |
| |
| static DWORD WINAPI WSAPIRenderThread(LPVOID context); |
| DWORD DoRenderThread(); |
| |
| static DWORD WINAPI GetCaptureVolumeThread(LPVOID context); |
| DWORD DoGetCaptureVolumeThread(); |
| |
| static DWORD WINAPI SetCaptureVolumeThread(LPVOID context); |
| DWORD DoSetCaptureVolumeThread(); |
| |
| void _SetThreadName(DWORD dwThreadID, LPCSTR szThreadName); |
| void _Lock() { _critSect.Enter(); }; |
| void _UnLock() { _critSect.Leave(); }; |
| |
| private: |
| WebRtc_Word32 Id() {return _id;} |
| |
| private: |
| int SetDMOProperties(); |
| |
| int SetBoolProperty(IPropertyStore* ptrPS, |
| REFPROPERTYKEY key, |
| VARIANT_BOOL value); |
| |
| int SetVtI4Property(IPropertyStore* ptrPS, |
| REFPROPERTYKEY key, |
| LONG value); |
| |
| WebRtc_Word32 _EnumerateEndpointDevicesAll(EDataFlow dataFlow) const; |
| void _TraceCOMError(HRESULT hr) const; |
| |
| WebRtc_Word32 _RefreshDeviceList(EDataFlow dir); |
| WebRtc_Word16 _DeviceListCount(EDataFlow dir); |
| WebRtc_Word32 _GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen); |
| WebRtc_Word32 _GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen); |
| WebRtc_Word32 _GetDeviceName(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen); |
| WebRtc_Word32 _GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen); |
| WebRtc_Word32 _GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen); |
| WebRtc_Word32 _GetDefaultDeviceIndex(EDataFlow dir, ERole role, int* index); |
| WebRtc_Word32 _GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen); |
| WebRtc_Word32 _GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice); |
| WebRtc_Word32 _GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice); |
| |
| void _Get44kHzDrift(); |
| |
| // Converts from wide-char to UTF-8 if UNICODE is defined. |
| // Does nothing if UNICODE is undefined. |
| char* WideToUTF8(const TCHAR* src) const; |
| |
| WebRtc_Word32 InitRecordingDMO(); |
| |
| private: |
| ScopedCOMInitializer _comInit; |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| CriticalSectionWrapper& _critSect; |
| CriticalSectionWrapper& _volumeMutex; |
| WebRtc_Word32 _id; |
| |
| private: // MMDevice |
| IMMDeviceEnumerator* _ptrEnumerator; |
| IMMDeviceCollection* _ptrRenderCollection; |
| IMMDeviceCollection* _ptrCaptureCollection; |
| IMMDevice* _ptrDeviceOut; |
| IMMDevice* _ptrDeviceIn; |
| |
| private: // WASAPI |
| IAudioClient* _ptrClientOut; |
| IAudioClient* _ptrClientIn; |
| IAudioRenderClient* _ptrRenderClient; |
| IAudioCaptureClient* _ptrCaptureClient; |
| IAudioEndpointVolume* _ptrCaptureVolume; |
| ISimpleAudioVolume* _ptrRenderSimpleVolume; |
| |
| // DirectX Media Object (DMO) for the built-in AEC. |
| scoped_refptr<IMediaObject> _dmo; |
| scoped_refptr<IMediaBuffer> _mediaBuffer; |
| bool _builtInAecEnabled; |
| |
| HANDLE _hRenderSamplesReadyEvent; |
| HANDLE _hPlayThread; |
| HANDLE _hRenderStartedEvent; |
| HANDLE _hShutdownRenderEvent; |
| |
| HANDLE _hCaptureSamplesReadyEvent; |
| HANDLE _hRecThread; |
| HANDLE _hCaptureStartedEvent; |
| HANDLE _hShutdownCaptureEvent; |
| |
| HANDLE _hGetCaptureVolumeThread; |
| HANDLE _hSetCaptureVolumeThread; |
| HANDLE _hSetCaptureVolumeEvent; |
| |
| HANDLE _hMmTask; |
| |
| UINT _playAudioFrameSize; |
| WebRtc_UWord32 _playSampleRate; |
| WebRtc_UWord32 _devicePlaySampleRate; |
| WebRtc_UWord32 _playBlockSize; |
| WebRtc_UWord32 _devicePlayBlockSize; |
| WebRtc_UWord32 _playChannels; |
| Atomic32Wrapper _sndCardPlayDelay; |
| UINT64 _writtenSamples; |
| LONGLONG _playAcc; |
| |
| UINT _recAudioFrameSize; |
| WebRtc_UWord32 _recSampleRate; |
| WebRtc_UWord32 _recBlockSize; |
| WebRtc_UWord32 _recChannels; |
| UINT64 _readSamples; |
| Atomic32Wrapper _sndCardRecDelay; |
| |
| float _sampleDriftAt48kHz; |
| float _driftAccumulator; |
| |
| WebRtc_UWord16 _recChannelsPrioList[2]; |
| WebRtc_UWord16 _playChannelsPrioList[2]; |
| |
| LARGE_INTEGER _perfCounterFreq; |
| double _perfCounterFactor; |
| float _avgCPULoad; |
| |
| private: |
| bool _initialized; |
| bool _recording; |
| bool _playing; |
| bool _recIsInitialized; |
| bool _playIsInitialized; |
| bool _speakerIsInitialized; |
| bool _microphoneIsInitialized; |
| |
| bool _usingInputDeviceIndex; |
| bool _usingOutputDeviceIndex; |
| AudioDeviceModule::WindowsDeviceType _inputDevice; |
| AudioDeviceModule::WindowsDeviceType _outputDevice; |
| WebRtc_UWord16 _inputDeviceIndex; |
| WebRtc_UWord16 _outputDeviceIndex; |
| |
| bool _AGC; |
| |
| WebRtc_UWord16 _playWarning; |
| WebRtc_UWord16 _playError; |
| WebRtc_UWord16 _recWarning; |
| WebRtc_UWord16 _recError; |
| |
| AudioDeviceModule::BufferType _playBufType; |
| WebRtc_UWord16 _playBufDelay; |
| WebRtc_UWord16 _playBufDelayFixed; |
| |
| WebRtc_UWord16 _newMicLevel; |
| |
| mutable char _str[512]; |
| }; |
| |
| #endif // #if (_MSC_VER >= 1400) |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H |
| |
