diff options
| author | Gary Scavone <gary@music.mcgill.ca> | 2014-04-01 12:03:38 -0400 |
|---|---|---|
| committer | Gary Scavone <gary@music.mcgill.ca> | 2014-04-01 12:05:51 -0400 |
| commit | 18e6f5126073320450438e99b4833cabc3347dba (patch) | |
| tree | 3b7fb305eec569873cf8fe00ef5ff56c572315ff /RtAudio.cpp | |
| parent | a71027332b636913443e66f6af4566dd86983f24 (diff) | |
Added Wasapi code (though untested); Updates for better data format support in PulseAudio; Commented-out various unused function arguments in tests/ files.
Diffstat (limited to 'RtAudio.cpp')
| -rw-r--r-- | RtAudio.cpp | 1383 |
1 files changed, 1375 insertions, 8 deletions
diff --git a/RtAudio.cpp b/RtAudio.cpp index 7d60e77..d65a7ac 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -53,7 +53,7 @@ const unsigned int RtApi::SAMPLE_RATES[] = { 32000, 44100, 48000, 88200, 96000, 176400, 192000
};
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
#define MUTEX_DESTROY(A) DeleteCriticalSection(A)
#define MUTEX_LOCK(A) EnterCriticalSection(A)
@@ -101,6 +101,9 @@ void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw() #if defined(__WINDOWS_ASIO__)
apis.push_back( WINDOWS_ASIO );
#endif
+#if defined(__WINDOWS_WASAPI__)
+ apis.push_back( WINDOWS_WASAPI );
+#endif
#if defined(__WINDOWS_DS__)
apis.push_back( WINDOWS_DS );
#endif
@@ -138,6 +141,10 @@ void RtAudio :: openRtApi( RtAudio::Api api ) if ( api == WINDOWS_ASIO )
rtapi_ = new RtApiAsio();
#endif
+#if defined(__WINDOWS_WASAPI__)
+ if ( api == WINDOWS_WASAPI )
+ rtapi_ = new RtApiWasapi();
+#endif
#if defined(__WINDOWS_DS__)
if ( api == WINDOWS_DS )
rtapi_ = new RtApiDs();
@@ -3562,10 +3569,1362 @@ static const char* getAsioErrorString( ASIOError result ) return "Unknown error.";
}
+
//******************** End of __WINDOWS_ASIO__ *********************//
#endif
+#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
+
+#include "RtWasapi.inl"
+#include <audioclient.h>
+#include <avrt.h>
+#include <functiondiscoverykeys.h>
+#include <mmdeviceapi.h>
+
+//=============================================================================
+
+#define EXIT_ON_ERROR( hr, errorType, errorText )\
+if( FAILED( hr ) )\
+{\
+ errorText_ = __FUNCTION__ ": " errorText;\
+ error( errorType );\
+ goto Exit;\
+}
+
+#define SAFE_RELEASE( objectPtr )\
+if( objectPtr )\
+{\
+ objectPtr->Release();\
+ objectPtr = NULL;\
+}
+
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+
+//-----------------------------------------------------------------------------
+
+// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
+// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
+// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
+// provide intermediate storage for read / write synchronization.
+class WasapiBuffer
+{
+public:
+ WasapiBuffer()
+ : buffer_( NULL ),
+ bufferSize_( 0 ),
+ inIndex_( 0 ),
+ outIndex_( 0 ) {}
+
+ ~WasapiBuffer() {
+ delete buffer_;
+ }
+
+ // sets the length of the internal ring buffer
+ void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+ delete buffer_;
+
+ buffer_ = ( char* ) calloc( bufferSize, formatBytes );
+
+ bufferSize_ = bufferSize;
+ inIndex_ = 0;
+ outIndex_ = 0;
+ }
+
+ // attempt to push a buffer into the ring buffer at the current "in" index
+ bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relOutIndex = outIndex_;
+ unsigned int inIndexEnd = inIndex_ + bufferSize;
+ if( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ )
+ {
+ relOutIndex += bufferSize_;
+ }
+
+ // "in" index can end on the "out" index but cannot begin at it
+ if( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex )
+ {
+ return false; // not enough space between "in" index and "out" index
+ }
+
+ // copy buffer from external to internal
+ int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromInSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
+ memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
+ memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
+ memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
+ memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
+ memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
+ memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "in" index
+ inIndex_ += bufferSize;
+ inIndex_ %= bufferSize_;
+
+ return true;
+ }
+
+ // attempt to pull a buffer from the ring buffer from the current "out" index
+ bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relInIndex = inIndex_;
+ unsigned int outIndexEnd = outIndex_ + bufferSize;
+ if( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ )
+ {
+ relInIndex += bufferSize_;
+ }
+
+ // "out" index can begin at and end on the "in" index
+ if( outIndex_ < relInIndex && outIndexEnd > relInIndex )
+ {
+ return false; // not enough space between "out" index and "in" index
+ }
+
+ // copy buffer from internal to external
+ int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromOutSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+ memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+ memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+ memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+ memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+ memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+ memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "out" index
+ outIndex_ += bufferSize;
+ outIndex_ %= bufferSize_;
+
+ return true;
+ }
+
+private:
+ char* buffer_;
+ unsigned int bufferSize_;
+ unsigned int inIndex_;
+ unsigned int outIndex_;
+};
+
+//-----------------------------------------------------------------------------
+
+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate and
+// channel counts between HW and the user. The convertBufferWasapi function is used to perform
+// these conversions between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
+// This sample rate converter favors speed over quality, and works best with conversions between
+// one rate and its multiple. RtApiWasapi will not populate a device's sample rate list with rates
+// that may cause artifacts via this conversion.
+void convertBufferWasapi( char* outBuffer,
+ const char* inBuffer,
+ const unsigned int& inChannelCount,
+ const unsigned int& outChannelCount,
+ const unsigned int& inSampleRate,
+ const unsigned int& outSampleRate,
+ const unsigned int& inSampleCount,
+ unsigned int& outSampleCount,
+ const RtAudioFormat& format )
+{
+ // calculate the new outSampleCount and relative sampleStep
+ float sampleRatio = ( float ) outSampleRate / inSampleRate;
+ float sampleStep = 1.0f / sampleRatio;
+ float inSampleFraction = 0.0f;
+ unsigned int commonChannelCount = min( inChannelCount, outChannelCount );
+
+ outSampleCount = ( unsigned int ) ( inSampleCount * sampleRatio );
+
+ // frame-by-frame, copy each relative input sample into it's corresponding output sample
+ for( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
+ {
+ unsigned int inSample = ( unsigned int ) inSampleFraction;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) outBuffer )[ outSample * outChannelCount ], &( ( char* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) outBuffer )[ outSample * outChannelCount ], &( ( short* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) outBuffer )[ outSample * outChannelCount ], &( ( S24* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) outBuffer )[ outSample * outChannelCount ], &( ( int* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) outBuffer )[ outSample * outChannelCount ], &( ( float* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) outBuffer )[ outSample * outChannelCount ], &( ( double* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( double ) );
+ break;
+ }
+
+ // jump to next in sample
+ inSampleFraction += sampleStep;
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
+{
+ IAudioClient* captureAudioClient;
+ IAudioClient* renderAudioClient;
+ IAudioCaptureClient* captureClient;
+ IAudioRenderClient* renderClient;
+ HANDLE captureEvent;
+ HANDLE renderEvent;
+
+ WasapiHandle()
+ : captureAudioClient( NULL ),
+ renderAudioClient( NULL ),
+ captureClient( NULL ),
+ renderClient( NULL ),
+ captureEvent( NULL ),
+ renderEvent( NULL ) {}
+};
+
+//=============================================================================
+
+RtApiWasapi::RtApiWasapi()
+ : coInitialized_( false ), deviceEnumerator_( NULL )
+{
+ // WASAPI can run either apartment or multi-threaded
+ HRESULT hr = CoInitialize( NULL );
+
+ if( !FAILED( hr ) )
+ coInitialized_ = true;
+
+ // instantiate device enumerator
+ hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+ CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+ ( void** ) &deviceEnumerator_ );
+
+ if( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
+ error( RtAudioError::DRIVER_ERROR );
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::~RtApiWasapi()
+{
+ // if this object previously called CoInitialize()
+ if( coInitialized_ )
+ {
+ CoUninitialize();
+ }
+
+ if( stream_.state != STREAM_CLOSED )
+ {
+ closeStream();
+ }
+
+ SAFE_RELEASE( deviceEnumerator_ );
+}
+
+//=============================================================================
+
+unsigned int RtApiWasapi::getDeviceCount( void )
+{
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+
+ // count capture devices
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device collection" );
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device count" );
+
+ // count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device collection" );
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device count" );
+
+Exit:
+ // release all references
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+
+ return captureDeviceCount + renderDeviceCount;
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+ std::wstring deviceName;
+ std::string defaultDeviceName;
+ bool isCaptureDevice = false;
+
+ PROPVARIANT deviceNameProp;
+ PROPVARIANT defaultDeviceNameProp;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ IMMDevice* defaultDevicePtr = NULL;
+ IAudioClient* audioClient = NULL;
+ IPropertyStore* devicePropStore = NULL;
+ IPropertyStore* defaultDevicePropStore = NULL;
+
+ WAVEFORMATEX* deviceFormat = NULL;
+ WAVEFORMATEX* closestMatchFormat = NULL;
+
+ // probed
+ info.probed = false;
+
+ // count capture devices
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device collection" );
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device count" );
+
+ // count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device collection" );
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device count" );
+
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount )
+ EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Invalid device index" );
+
+ // determine whether index falls within capture or render devices
+ if ( device < captureDeviceCount ) {
+ hr = captureDevices->Item( device, &devicePtr );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device handle" );
+
+ isCaptureDevice = true;
+ }
+ else {
+ hr = renderDevices->Item( device - captureDeviceCount, &devicePtr );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device handle" );
+
+ isCaptureDevice = false;
+ }
+
+ // get default device name
+ if ( isCaptureDevice ) {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve default render device handle" );
+ }
+ else {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve default capture device handle" );
+ }
+
+ hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to open default device property store" );
+
+ PropVariantInit( &defaultDeviceNameProp );
+
+ hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve default device property: PKEY_Device_FriendlyName" );
+
+ deviceName = defaultDeviceNameProp.pwszVal;
+ defaultDeviceName = std::string( deviceName.begin(), deviceName.end() );
+
+ // name
+ hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to open device property store" );
+
+ PropVariantInit( &deviceNameProp );
+
+ hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device property: PKEY_Device_FriendlyName" );
+
+ deviceName = deviceNameProp.pwszVal;
+ info.name = std::string( deviceName.begin(), deviceName.end() );
+
+ // is default
+ if ( isCaptureDevice ) {
+ info.isDefaultInput = info.name == defaultDeviceName;
+ info.isDefaultOutput = false;
+ }
+ else {
+ info.isDefaultInput = false;
+ info.isDefaultOutput = info.name == defaultDeviceName;
+ }
+
+ // channel count
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device audio client" );
+
+ hr = audioClient->GetMixFormat( &deviceFormat );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );
+
+ if ( isCaptureDevice ) {
+ info.inputChannels = deviceFormat->nChannels;
+ info.outputChannels = 0;
+ info.duplexChannels = 0;
+ }
+ else {
+ info.inputChannels = 0;
+ info.outputChannels = deviceFormat->nChannels;
+ info.duplexChannels = 0;
+ }
+
+ // sample rates
+ info.sampleRates.clear();
+
+ // allow support for sample rates that are multiples of the base rate
+ for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
+ if ( SAMPLE_RATES[i] < deviceFormat->nSamplesPerSec ) {
+ if ( deviceFormat->nSamplesPerSec % SAMPLE_RATES[i] == 0 ) {
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+ }
+ else {
+ if ( SAMPLE_RATES[i] % deviceFormat->nSamplesPerSec == 0 ) {
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+ }
+ }
+
+ // native format
+ info.nativeFormats = 0;
+
+ if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ }
+ else if ( deviceFormat->wBitsPerSample == 64 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+ }
+ }
+ else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 8 ) {
+ info.nativeFormats |= RTAUDIO_SINT8;
+ }
+ else if ( deviceFormat->wBitsPerSample == 16 ) {
+ info.nativeFormats |= RTAUDIO_SINT16;
+ }
+ else if ( deviceFormat->wBitsPerSample == 24 ) {
+ info.nativeFormats |= RTAUDIO_SINT24;
+ }
+ else if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_SINT32;
+ }
+ }
+
+ // probed
+ info.probed = true;
+
+Exit:
+ // release all references
+ PropVariantClear( &deviceNameProp );
+ PropVariantClear( &defaultDeviceNameProp );
+
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ SAFE_RELEASE( defaultDevicePtr );
+ SAFE_RELEASE( audioClient );
+ SAFE_RELEASE( devicePropStore );
+ SAFE_RELEASE( defaultDevicePropStore );
+
+ CoTaskMemFree( deviceFormat );
+ CoTaskMemFree( closestMatchFormat );
+
+ return info;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultOutput ) {
+ return i;
+ }
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultInput ) {
+ return i;
+ }
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiWasapi::closeStream: No open stream to close";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ if ( stream_.state != STREAM_STOPPED )
+ stopStream();
+
+ // clean up stream memory
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
+
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
+
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
+
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
+
+ delete stream_.apiHandle;
+ stream_.apiHandle = NULL;
+
+ for ( int i = 0; i < 2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ // update stream state
+ stream_.state = STREAM_CLOSED;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::startStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiWasapi::startStream: The stream is already running";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // update stream state
+ stream_.state = STREAM_RUNNING;
+
+ // create WASAPI stream thread
+ stream_.callbackInfo.thread = ( unsigned int ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
+
+ if ( !stream_.callbackInfo.thread ) {
+ errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread";
+ error( RtAudioError::THREAD_ERROR );
+ }
+ else {
+ SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
+ ResumeThread( ( void* ) stream_.callbackInfo.thread );
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::stopStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::stopStream: The stream is already stopped";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ // wait until stream thread is stopped
+ while( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
+
+ // Wait for the last buffer to play before stopping.
+ Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
+
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream";
+ error( RtAudioError::DRIVER_ERROR );
+ }
+ }
+
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream";
+ error( RtAudioError::DRIVER_ERROR );
+ }
+ }
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread";
+ error( RtAudioError::THREAD_ERROR );
+ }
+
+ stream_.callbackInfo.thread = NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::abortStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::abortStream: The stream is already stopped";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ // wait until stream thread is stopped
+ while ( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
+
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream";
+ error( RtAudioError::DRIVER_ERROR );
+ }
+ }
+
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream";
+ error( RtAudioError::DRIVER_ERROR );
+ }
+ }
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread";
+ error( RtAudioError::THREAD_ERROR );
+ }
+
+ stream_.callbackInfo.thread = NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int* bufferSize,
+ RtAudio::StreamOptions* options )
+{
+ bool methodResult = FAILURE;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ WAVEFORMATEX* deviceFormat = NULL;
+
+ // create API Handle if not already created
+ if ( !stream_.apiHandle )
+ stream_.apiHandle = ( void* ) new WasapiHandle();
+
+ // count capture devices
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device collection" );
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device count" );
+
+ // count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device collection" );
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device count" );
+
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount )
+ EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Invalid device index" );
+
+ // determine whether index falls within capture or render devices
+ if ( device < captureDeviceCount ) {
+ if ( mode != INPUT )
+ EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Capture device selected as output device" );
+
+ // retrieve captureAudioClient from devicePtr
+ IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+
+ hr = captureDevices->Item( device, &devicePtr );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device handle" );
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &captureAudioClient );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device audio client" );
+
+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+ else {
+ if ( mode != OUTPUT )
+ EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Render device selected as input device" );
+
+ // retrieve renderAudioClient from devicePtr
+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+
+ hr = renderDevices->Item( device - captureDeviceCount, &devicePtr );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device handle" );
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &renderAudioClient );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device audio client" );
+
+ hr = renderAudioClient->GetMixFormat( &deviceFormat );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+
+ // fill stream data
+ if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
+ ( stream_.mode == INPUT && mode == OUTPUT ) ) {
+ stream_.mode = DUPLEX;
+ }
+ else {
+ stream_.mode = mode;
+ }
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ stream_.doByteSwap[mode] = false;
+ stream_.sampleRate = sampleRate;
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+ else
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ if ( stream_.doConvertBuffer[mode] )
+ setConvertInfo( mode, 0 );
+
+ // Allocate necessary internal buffers
+ unsigned int bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
+
+ stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
+ if ( !stream_.userBuffer[mode] )
+ EXIT_ON_ERROR( -1, RtAudioError::MEMORY_ERROR, "Error allocating user buffer memory" );
+
+ if ( stream_.doConvertBuffer[mode] && !stream_.deviceBuffer ) {
+ unsigned int deviceBufferSize = max( stream_.nUserChannels[INPUT] * stream_.bufferSize * formatBytes( stream_.userFormat ),
+ stream_.nUserChannels[OUTPUT] * stream_.bufferSize * formatBytes( stream_.userFormat ) );
+
+ stream_.deviceBuffer = ( char* ) calloc( deviceBufferSize, 1 );
+ if ( !stream_.deviceBuffer )
+ EXIT_ON_ERROR( -1, RtAudioError::MEMORY_ERROR, "Error allocating device buffer memory" );
+ }
+
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
+ stream_.callbackInfo.priority = 15;
+ else
+ stream_.callbackInfo.priority = 0;
+
+ ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
+ ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
+
+ methodResult = SUCCESS;
+
+Exit:
+ //clean up
+
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+
+ CoTaskMemFree( deviceFormat );
+
+ // if method failed, close the stream
+ if ( methodResult == FAILURE )
+ closeStream();
+
+ return methodResult;
+}
+
+//=============================================================================
+
+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::wasapiThread()
+{
+ // as this is a new thread, we must CoInitialize it
+ CoInitialize( NULL );
+
+ HRESULT hr;
+
+ IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+ IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+ IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
+ IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
+ HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
+ HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
+
+ WAVEFORMATEX* captureFormat = NULL;
+ WAVEFORMATEX* renderFormat = NULL;
+ float captureSrRatio = 0.0f;
+ float renderSrRatio = 0.0f;
+ WasapiBuffer captureBuffer;
+ WasapiBuffer renderBuffer;
+
+ // Attempt to assign "Pro Audio" characteristic to thread
+ HMODULE AvrtDll = LoadLibrary( "AVRT.dll" );
+ if ( AvrtDll ) {
+ DWORD taskIndex = 0;
+ TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+ AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+ FreeLibrary( AvrtDll );
+ }
+
+ // start capture stream if applicable
+ if ( captureAudioClient ) {
+ hr = captureAudioClient->GetMixFormat( &captureFormat );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );
+
+ captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
+
+ // initialize capture stream according to desire buffer size
+ float desiredBufferSize = stream_.bufferSize * captureSrRatio;
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
+
+ if ( !captureClient ) {
+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ captureFormat,
+ NULL );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to initialize capture audio client" );
+
+ hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+ ( void** ) &captureClient );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture client handle" );
+
+ // configure captureEvent to trigger on every available capture buffer
+ captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !captureEvent )
+ EXIT_ON_ERROR( -1, RtAudioError::SYSTEM_ERROR, "Unable to create capture event" );
+
+ hr = captureAudioClient->SetEventHandle( captureEvent );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to set capture event handle" );
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
+ }
+
+ unsigned int inBufferSize = 0;
+ hr = captureAudioClient->GetBufferSize( &inBufferSize );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to get capture buffer size" );
+
+ // scale outBufferSize according to stream->user sample rate ratio
+ // (outBufferSize must be a multiple of the input channel count)
+ unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio );
+ if ( outBufferSize % stream_.nDeviceChannels[INPUT] )
+ outBufferSize += stream_.nDeviceChannels[INPUT] - ( outBufferSize % stream_.nDeviceChannels[INPUT] );
+
+ inBufferSize *= stream_.nDeviceChannels[INPUT];
+
+ // set captureBuffer size
+ captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
+
+ // reset the capture stream
+ hr = captureAudioClient->Reset();
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to reset capture stream" );
+
+ // start the capture stream
+ hr = captureAudioClient->Start();
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to start capture stream" );
+ }
+
+ // start render stream if applicable
+ if ( renderAudioClient ) {
+ hr = renderAudioClient->GetMixFormat( &renderFormat );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );
+
+ renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
+
+ // initialize render stream according to desire buffer size
+ float desiredBufferSize = stream_.bufferSize * renderSrRatio;
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
+
+ if ( !renderClient ) {
+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ renderFormat,
+ NULL );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to initialize render audio client" );
+
+ hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+ ( void** ) &renderClient );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render client handle" );
+
+ // configure renderEvent to trigger on every available render buffer
+ renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !renderEvent )
+ EXIT_ON_ERROR( -1, RtAudioError::SYSTEM_ERROR, "Unable to create render event" );
+
+ hr = renderAudioClient->SetEventHandle( renderEvent );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to set render event handle" );
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
+ }
+
+ unsigned int outBufferSize = 0;
+ hr = renderAudioClient->GetBufferSize( &outBufferSize );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to get render buffer size" );
+
+ // scale inBufferSize according to user->stream sample rate ratio
+ // (inBufferSize must be a multiple of the output channel count)
+ unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * renderSrRatio );
+ if ( inBufferSize % stream_.nDeviceChannels[OUTPUT] ) {
+ inBufferSize += stream_.nDeviceChannels[OUTPUT] - ( inBufferSize % stream_.nDeviceChannels[OUTPUT] );
+ }
+
+ outBufferSize *= stream_.nDeviceChannels[OUTPUT];
+
+ // set renderBuffer size
+ renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
+
+ // reset the render stream
+ hr = renderAudioClient->Reset();
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to reset render stream" );
+
+ // start the render stream
+ hr = renderAudioClient->Start();
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to start render stream" );
+ }
+
+ // declare local stream variables
+ RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+
+ BYTE* streamBuffer = NULL;
+ unsigned long captureFlags = 0;
+
+ unsigned int bufferFrameCount = 0;
+ unsigned int numFramesPadding = 0;
+ unsigned int convBufferSize = 0;
+
+ bool callbackPushed = false;
+ bool callbackPulled = false;
+ bool callbackStopped = false;
+
+ int callbackResult = 0;
+
+ // convBuffer is used to store converted buffers between WASAPI and the user
+ char* convBuffer = NULL;
+
+ if ( stream_.mode == INPUT ) {
+ convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio * formatBytes( stream_.deviceFormat[INPUT] ) ) );
+ }
+ else if ( stream_.mode == OUTPUT ) {
+ convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * renderSrRatio * formatBytes( stream_.deviceFormat[OUTPUT] ) ) );
+ }
+ else if ( stream_.mode == DUPLEX ) {
+ convBuffer = ( char* ) malloc( max( ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio * formatBytes( stream_.deviceFormat[INPUT] ) ),
+ ( size_t ) ( stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * renderSrRatio * formatBytes( stream_.deviceFormat[OUTPUT] ) ) ) );
+ }
+
+ // stream process loop
+ while ( stream_.state != STREAM_STOPPING ) {
+ if ( !callbackPulled ) {
+ // Callback Input
+ // ==============
+ // 1. Pull callback buffer from inputBuffer
+ // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
+ // Convert callback buffer to user format
+
+ if ( captureAudioClient ) {
+ // Pull callback buffer from inputBuffer
+ callbackPulled = captureBuffer.pullBuffer( convBuffer,
+ ( unsigned int ) ( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * captureSrRatio ),
+ stream_.deviceFormat[INPUT] );
+
+ if ( callbackPulled ) {
+ // Convert callback buffer to user sample rate and channel count
+ convertBufferWasapi( stream_.deviceBuffer,
+ convBuffer,
+ stream_.nDeviceChannels[INPUT],
+ stream_.nUserChannels[INPUT],
+ captureFormat->nSamplesPerSec,
+ stream_.sampleRate,
+ ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
+ convBufferSize,
+ stream_.deviceFormat[INPUT] );
+
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ // Convert callback buffer to user format
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
+ }
+ else {
+ // no conversion, simple copy deviceBuffer to userBuffer
+ memcpy( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
+ }
+ }
+ }
+ else {
+ // if there is no capture stream, set callbackPulled flag
+ callbackPulled = true;
+ }
+
+ // Execute Callback
+ // ================
+ // 1. Execute user callback method
+ // 2. Handle return value from callback
+
+ // if callback has not requested the stream to stop
+ if ( callbackPulled && !callbackStopped ) {
+ // Execute user callback method
+ callbackResult = callback( stream_.userBuffer[OUTPUT],
+ stream_.userBuffer[INPUT],
+ stream_.bufferSize,
+ getStreamTime(),
+ captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+ stream_.callbackInfo.userData );
+
+ // Handle return value from callback
+ if ( callbackResult == 1 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, NULL, NULL );
+
+ if ( !threadHandle ) {
+ EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to instantiate stream stop thread" );
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to close stream stop thread handle" );
+ }
+
+ callbackStopped = true;
+ }
+ else if ( callbackResult == 2 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, NULL, NULL );
+
+ if ( !threadHandle ) {
+ EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to instantiate stream abort thread" );
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to close stream abort thread handle" );
+ }
+
+ callbackStopped = true;
+ }
+ }
+ }
+
+ // Callback Output
+ // ===============
+ // 1. Convert callback buffer to stream format
+ // 2. Convert callback buffer to stream sample rate and channel count
+ // 3. Push callback buffer into outputBuffer
+
+ if ( renderAudioClient && callbackPulled ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ // Convert callback buffer to stream format
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+
+ // Convert callback buffer to stream sample rate and channel count
+ convertBufferWasapi( convBuffer,
+ stream_.deviceBuffer,
+ stream_.nUserChannels[OUTPUT],
+ stream_.nDeviceChannels[OUTPUT],
+ stream_.sampleRate,
+ renderFormat->nSamplesPerSec,
+ stream_.bufferSize,
+ convBufferSize,
+ stream_.deviceFormat[OUTPUT] );
+ }
+ else {
+ // Convert callback buffer to stream sample rate and channel count
+ convertBufferWasapi( convBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.nUserChannels[OUTPUT],
+ stream_.nDeviceChannels[OUTPUT],
+ stream_.sampleRate,
+ renderFormat->nSamplesPerSec,
+ stream_.bufferSize,
+ convBufferSize,
+ stream_.deviceFormat[OUTPUT] );
+ }
+
+ // Push callback buffer into outputBuffer
+ callbackPushed = renderBuffer.pushBuffer( convBuffer,
+ convBufferSize * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] );
+ }
+
+ // Stream Capture
+ // ==============
+ // 1. Get capture buffer from stream
+ // 2. Push capture buffer into inputBuffer
+ // 3. If 2. was successful: Release capture buffer
+
+ if ( captureAudioClient ) {
+ // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+ if ( !callbackPulled ) {
+ WaitForSingleObject( captureEvent, INFINITE );
+ }
+
+ // Get capture buffer from stream
+ hr = captureClient->GetBuffer( &streamBuffer,
+ &bufferFrameCount,
+ &captureFlags, NULL, NULL );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture buffer" );
+
+ if ( bufferFrameCount != 0 ) {
+ // Push capture buffer into inputBuffer
+ if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] ) )
+ {
+ // Release capture buffer
+ hr = captureClient->ReleaseBuffer( bufferFrameCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release capture buffer" );
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release capture buffer" );
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release capture buffer" );
+ }
+ }
+
+ // Stream Render
+ // =============
+ // 1. Get render buffer from stream
+ // 2. Pull next buffer from outputBuffer
+ // 3. If 2. was successful: Fill render buffer with next buffer
+ // Release render buffer
+
+ if ( renderAudioClient ) {
+ // if the callback output buffer was not pushed to renderBuffer, wait for next render event
+ if ( callbackPulled && !callbackPushed ) {
+ WaitForSingleObject( renderEvent, INFINITE );
+ }
+
+ // Get render buffer from stream
+ hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render buffer size" );
+
+ hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render buffer padding" );
+
+ bufferFrameCount -= numFramesPadding;
+
+ if ( bufferFrameCount != 0 ) {
+ hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render buffer" );
+
+ // Pull next buffer from outputBuffer
+ // Fill render buffer with next buffer
+ if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] ) )
+ {
+ // Release render buffer
+ hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release render buffer" );
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release render buffer" );
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release render buffer" );
+ }
+ }
+
+ // if the callback buffer was pushed renderBuffer reset callbackPulled flag
+ if ( callbackPushed ) {
+ callbackPulled = false;
+ }
+
+ // tick stream time
+ RtApi::tickStreamTime();
+ }
+
+Exit:
+ // clean up
+ CoTaskMemFree( captureFormat );
+ CoTaskMemFree( renderFormat );
+
+ delete convBuffer;
+
+ CoUninitialize();
+
+ // update stream state
+ stream_.state = STREAM_STOPPED;
+}
+
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
+
+
#if defined(__WINDOWS_DS__) // Windows DirectSound API
// Modified by Robin Davies, October 2005
@@ -6144,6 +7503,7 @@ void RtApiAlsa :: startStream() stream_.state = STREAM_RUNNING;
unlock:
+ apiInfo->runnable = true;
pthread_cond_signal( &apiInfo->runnable_cv );
MUTEX_UNLOCK( &stream_.mutex );
@@ -6490,7 +7850,7 @@ unsigned int RtApiPulse::getDeviceCount( void ) return 1;
}
-RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device )
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
{
RtAudio::DeviceInfo info;
info.probed = true;
@@ -6785,28 +8145,35 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, if ( format == sf->rtaudio_format ) {
sf_found = true;
stream_.userFormat = sf->rtaudio_format;
+ stream_.deviceFormat[mode] = stream_.userFormat;
ss.format = sf->pa_format;
break;
}
}
- if ( !sf_found ) {
- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample format.";
- return false;
+ if ( !sf_found ) { // Use internal data format conversion.
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ ss.format = PA_SAMPLE_FLOAT32LE;
}
- // Set interleaving parameters.
+ // Set other stream parameters.
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
else stream_.userInterleaved = true;
stream_.deviceInterleaved[mode] = true;
stream_.nBuffers = 1;
stream_.doByteSwap[mode] = false;
- stream_.doConvertBuffer[mode] = channels > 1 && !stream_.userInterleaved;
- stream_.deviceFormat[mode] = stream_.userFormat;
stream_.nUserChannels[mode] = channels;
stream_.nDeviceChannels[mode] = channels + firstChannel;
stream_.channelOffset[mode] = 0;
std::string streamName = "RtAudio";
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+
// Allocate necessary internal buffers.
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
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