| /* |
| * 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_WAVE_WIN_H |
| #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H |
| |
| #include "audio_device_generic.h" |
| #include "audio_mixer_manager_win.h" |
| |
| #pragma comment( lib, "winmm.lib" ) |
| |
| namespace webrtc { |
| class EventWrapper; |
| class ThreadWrapper; |
| |
| const WebRtc_UWord32 TIMER_PERIOD_MS = 2; |
| const WebRtc_UWord32 REC_CHECK_TIME_PERIOD_MS = 4; |
| const WebRtc_UWord16 REC_PUT_BACK_DELAY = 4; |
| |
| 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 |
| |
| // NOTE - CPU load will not be correct for other sizes than 10ms |
| const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC/100); |
| const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC/100); |
| |
| enum { N_BUFFERS_IN = 200 }; |
| enum { N_BUFFERS_OUT = 200 }; |
| |
| class AudioDeviceWindowsWave : public AudioDeviceGeneric |
| { |
| public: |
| AudioDeviceWindowsWave(const WebRtc_Word32 id); |
| ~AudioDeviceWindowsWave(); |
| |
| // 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; |
| |
| 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;} |
| bool IsUsingOutputDeviceIndex() const {return _usingOutputDeviceIndex;} |
| AudioDeviceModule::WindowsDeviceType OutputDevice() const {return _outputDevice;} |
| WebRtc_UWord16 OutputDeviceIndex() const {return _outputDeviceIndex;} |
| bool IsUsingInputDeviceIndex() const {return _usingInputDeviceIndex;} |
| AudioDeviceModule::WindowsDeviceType InputDevice() const {return _inputDevice;} |
| WebRtc_UWord16 InputDeviceIndex() const {return _inputDeviceIndex;} |
| |
| private: |
| inline WebRtc_Word32 InputSanityCheckAfterUnlockedPeriod() const; |
| inline WebRtc_Word32 OutputSanityCheckAfterUnlockedPeriod() const; |
| |
| private: |
| WebRtc_Word32 EnumeratePlayoutDevices(); |
| WebRtc_Word32 EnumerateRecordingDevices(); |
| void TraceSupportFlags(DWORD dwSupport) const; |
| void TraceWaveInError(MMRESULT error) const; |
| void TraceWaveOutError(MMRESULT error) const; |
| WebRtc_Word32 PrepareStartRecording(); |
| WebRtc_Word32 PrepareStartPlayout(); |
| |
| WebRtc_Word32 RecProc(LONGLONG& consumedTime); |
| int PlayProc(LONGLONG& consumedTime); |
| |
| WebRtc_Word32 GetPlayoutBufferDelay(WebRtc_UWord32& writtenSamples, WebRtc_UWord32& playedSamples); |
| WebRtc_Word32 GetRecordingBufferDelay(WebRtc_UWord32& readSamples, WebRtc_UWord32& recSamples); |
| WebRtc_Word32 Write(WebRtc_Word8* data, WebRtc_UWord16 nSamples); |
| WebRtc_Word32 GetClockDrift(const WebRtc_UWord32 plSamp, const WebRtc_UWord32 rcSamp); |
| WebRtc_Word32 MonitorRecording(const WebRtc_UWord32 time); |
| WebRtc_Word32 RestartTimerIfNeeded(const WebRtc_UWord32 time); |
| |
| private: |
| static bool ThreadFunc(void*); |
| bool ThreadProcess(); |
| |
| static DWORD WINAPI GetCaptureVolumeThread(LPVOID context); |
| DWORD DoGetCaptureVolumeThread(); |
| |
| static DWORD WINAPI SetCaptureVolumeThread(LPVOID context); |
| DWORD DoSetCaptureVolumeThread(); |
| |
| private: |
| AudioDeviceBuffer* _ptrAudioBuffer; |
| |
| CriticalSectionWrapper& _critSect; |
| EventWrapper& _timeEvent; |
| EventWrapper& _recStartEvent; |
| EventWrapper& _playStartEvent; |
| |
| HANDLE _hGetCaptureVolumeThread; |
| HANDLE _hShutdownGetVolumeEvent; |
| HANDLE _hSetCaptureVolumeThread; |
| HANDLE _hShutdownSetVolumeEvent; |
| HANDLE _hSetCaptureVolumeEvent; |
| |
| ThreadWrapper* _ptrThread; |
| WebRtc_UWord32 _threadID; |
| |
| CriticalSectionWrapper& _critSectCb; |
| |
| WebRtc_Word32 _id; |
| |
| AudioMixerManager _mixerManager; |
| |
| bool _usingInputDeviceIndex; |
| bool _usingOutputDeviceIndex; |
| AudioDeviceModule::WindowsDeviceType _inputDevice; |
| AudioDeviceModule::WindowsDeviceType _outputDevice; |
| WebRtc_UWord16 _inputDeviceIndex; |
| WebRtc_UWord16 _outputDeviceIndex; |
| bool _inputDeviceIsSpecified; |
| bool _outputDeviceIsSpecified; |
| |
| WAVEFORMATEX _waveFormatIn; |
| WAVEFORMATEX _waveFormatOut; |
| |
| HWAVEIN _hWaveIn; |
| HWAVEOUT _hWaveOut; |
| |
| WAVEHDR _waveHeaderIn[N_BUFFERS_IN]; |
| WAVEHDR _waveHeaderOut[N_BUFFERS_OUT]; |
| |
| WebRtc_UWord8 _recChannels; |
| WebRtc_UWord8 _playChannels; |
| WebRtc_UWord16 _recBufCount; |
| WebRtc_UWord16 _recDelayCount; |
| WebRtc_UWord16 _recPutBackDelay; |
| |
| WebRtc_Word8 _recBuffer[N_BUFFERS_IN][4*REC_BUF_SIZE_IN_SAMPLES]; |
| WebRtc_Word8 _playBuffer[N_BUFFERS_OUT][4*PLAY_BUF_SIZE_IN_SAMPLES]; |
| |
| AudioDeviceModule::BufferType _playBufType; |
| |
| private: |
| bool _initialized; |
| bool _recording; |
| bool _playing; |
| bool _recIsInitialized; |
| bool _playIsInitialized; |
| bool _startRec; |
| bool _stopRec; |
| bool _startPlay; |
| bool _stopPlay; |
| bool _AGC; |
| |
| private: |
| WebRtc_UWord32 _prevPlayTime; |
| WebRtc_UWord32 _prevRecTime; |
| WebRtc_UWord32 _prevTimerCheckTime; |
| |
| WebRtc_UWord16 _playBufCount; // playout buffer index |
| WebRtc_UWord16 _dTcheckPlayBufDelay; // dT for check of play buffer, {2,5,10} [ms] |
| WebRtc_UWord16 _playBufDelay; // playback delay |
| WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay |
| WebRtc_UWord16 _minPlayBufDelay; // minimum playback delay |
| WebRtc_UWord16 _MAX_minBuffer; // level of (adaptive) min threshold must be < _MAX_minBuffer |
| |
| WebRtc_Word32 _erZeroCounter; // counts "buffer-is-empty" events |
| WebRtc_Word32 _intro; |
| WebRtc_Word32 _waitCounter; |
| |
| WebRtc_UWord32 _writtenSamples; |
| WebRtc_UWord32 _writtenSamplesOld; |
| WebRtc_UWord32 _playedSamplesOld; |
| |
| WebRtc_UWord32 _sndCardPlayDelay; |
| WebRtc_UWord32 _sndCardRecDelay; |
| |
| WebRtc_UWord32 _plSampOld; |
| WebRtc_UWord32 _rcSampOld; |
| |
| WebRtc_UWord32 _read_samples; |
| WebRtc_UWord32 _read_samples_old; |
| WebRtc_UWord32 _rec_samples_old; |
| |
| // State that detects driver problems: |
| WebRtc_Word32 _dc_diff_mean; |
| WebRtc_Word32 _dc_y_prev; |
| WebRtc_Word32 _dc_penalty_counter; |
| WebRtc_Word32 _dc_prevtime; |
| WebRtc_UWord32 _dc_prevplay; |
| |
| WebRtc_UWord32 _recordedBytes; // accumulated #recorded bytes (reset periodically) |
| WebRtc_UWord32 _prevRecByteCheckTime; // time when we last checked the recording process |
| |
| // CPU load measurements |
| LARGE_INTEGER _perfFreq; |
| LONGLONG _playAcc; // accumulated time for playout callback |
| float _avgCPULoad; // average total (rec+play) CPU load |
| |
| WebRtc_Word32 _wrapCounter; |
| |
| WebRtc_Word32 _useHeader; |
| WebRtc_Word16 _timesdwBytes; |
| WebRtc_Word32 _no_of_msecleft_warnings; |
| WebRtc_Word32 _writeErrors; |
| WebRtc_Word32 _timerFaults; |
| WebRtc_Word32 _timerRestartAttempts; |
| |
| WebRtc_UWord16 _playWarning; |
| WebRtc_UWord16 _playError; |
| WebRtc_UWord16 _recWarning; |
| WebRtc_UWord16 _recError; |
| |
| WebRtc_UWord32 _newMicLevel; |
| WebRtc_UWord32 _minMicVolume; |
| WebRtc_UWord32 _maxMicVolume; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H |
