diff --git a/source/modules/rtaudio/Makefile b/source/modules/rtaudio/Makefile index 5f452984e..20325fb41 100644 --- a/source/modules/rtaudio/Makefile +++ b/source/modules/rtaudio/Makefile @@ -47,7 +47,7 @@ debug: # -------------------------------------------------------------- -RtAudio.cpp.o: RtAudio.cpp RtAudio.h RtError.h +RtAudio.cpp.o: RtAudio.cpp RtAudio.h $(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@ # -------------------------------------------------------------- diff --git a/source/modules/rtaudio/RtAudio.cpp b/source/modules/rtaudio/RtAudio.cpp index 20093c051..96e77d3ba 100644 --- a/source/modules/rtaudio/RtAudio.cpp +++ b/source/modules/rtaudio/RtAudio.cpp @@ -5,12 +5,12 @@ RtAudio provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, Jack, and OSS), Macintosh OS X (CoreAudio and Jack), and Windows - (DirectSound and ASIO) operating systems. + (DirectSound, ASIO and WASAPI) operating systems. RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2013 Gary P. Scavone + Copyright (c) 2001-2014 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -38,7 +38,7 @@ */ /************************************************************************/ -// RtAudio: Version 4.0.12 +// RtAudio: Version 4.1.1 #include "RtAudio.h" #include @@ -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) @@ -75,6 +75,11 @@ const unsigned int RtApi::SAMPLE_RATES[] = { // // *************************************************** // +std::string RtAudio :: getVersion( void ) throw() +{ + return RTAUDIO_VERSION; +} + void RtAudio :: getCompiledApi( std::vector &apis ) throw() { apis.clear(); @@ -96,6 +101,9 @@ void RtAudio :: getCompiledApi( std::vector &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 @@ -133,6 +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(); @@ -147,7 +159,7 @@ void RtAudio :: openRtApi( RtAudio::Api api ) #endif } -RtAudio :: RtAudio( RtAudio::Api api ) throw() +RtAudio :: RtAudio( RtAudio::Api api ) { rtapi_ = 0; @@ -175,13 +187,15 @@ RtAudio :: RtAudio( RtAudio::Api api ) throw() // It should not be possible to get here because the preprocessor // definition __RTAUDIO_DUMMY__ is automatically defined if no // API-specific definitions are passed to the compiler. But just in - // case something weird happens, we'll print out an error message. - std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n"; + // case something weird happens, we'll thow an error. + std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; + throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); } RtAudio :: ~RtAudio() throw() { - delete rtapi_; + if ( rtapi_ ) + delete rtapi_; } void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, @@ -213,6 +227,7 @@ RtApi :: RtApi() stream_.userBuffer[1] = 0; MUTEX_INITIALIZE( &stream_.mutex ); showWarnings_ = true; + firstErrorOccurred_ = false; } RtApi :: ~RtApi() @@ -230,31 +245,34 @@ void RtApi :: openStream( RtAudio::StreamParameters *oParams, { if ( stream_.state != STREAM_CLOSED ) { errorText_ = "RtApi::openStream: a stream is already open!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } + // Clear stream information potentially left from a previously open stream. + clearStreamInfo(); + if ( oParams && oParams->nChannels < 1 ) { errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } if ( iParams && iParams->nChannels < 1 ) { errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } if ( oParams == NULL && iParams == NULL ) { errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } if ( formatBytes(format) == 0 ) { errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } @@ -264,7 +282,7 @@ void RtApi :: openStream( RtAudio::StreamParameters *oParams, oChannels = oParams->nChannels; if ( oParams->deviceId >= nDevices ) { errorText_ = "RtApi::openStream: output device parameter value is invalid."; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } } @@ -274,12 +292,11 @@ void RtApi :: openStream( RtAudio::StreamParameters *oParams, iChannels = iParams->nChannels; if ( iParams->deviceId >= nDevices ) { errorText_ = "RtApi::openStream: input device parameter value is invalid."; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } } - clearStreamInfo(); bool result; if ( oChannels > 0 ) { @@ -287,7 +304,7 @@ void RtApi :: openStream( RtAudio::StreamParameters *oParams, result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, sampleRate, format, bufferFrames, options ); if ( result == false ) { - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } } @@ -298,7 +315,7 @@ void RtApi :: openStream( RtAudio::StreamParameters *oParams, sampleRate, format, bufferFrames, options ); if ( result == false ) { if ( oChannels > 0 ) closeStream(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } } @@ -387,6 +404,14 @@ double RtApi :: getStreamTime( void ) #endif } +void RtApi :: setStreamTime( double time ) +{ + verifyStream(); + + if ( time >= 0.0 ) + stream_.streamTime = time; +} + unsigned int RtApi :: getStreamSampleRate( void ) { verifyStream(); @@ -403,8 +428,6 @@ unsigned int RtApi :: getStreamSampleRate( void ) #if defined(__MACOSX_CORE__) -#include - // The OS X CoreAudio API is designed to use a separate callback // procedure for each of its audio devices. A single RtAudio duplex // stream using two different devices is supported here, though it @@ -453,7 +476,7 @@ RtApiCore:: RtApiCore() OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); if ( result != noErr ) { errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } #endif } @@ -474,7 +497,7 @@ unsigned int RtApiCore :: getDeviceCount( void ) OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); if ( result != noErr ) { errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -492,7 +515,7 @@ unsigned int RtApiCore :: getDefaultInputDevice( void ) OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); if ( result != noErr ) { errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -502,7 +525,7 @@ unsigned int RtApiCore :: getDefaultInputDevice( void ) result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); if ( result != noErr ) { errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -510,7 +533,7 @@ unsigned int RtApiCore :: getDefaultInputDevice( void ) if ( id == deviceList[i] ) return i; errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -525,7 +548,7 @@ unsigned int RtApiCore :: getDefaultOutputDevice( void ) OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); if ( result != noErr ) { errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -535,7 +558,7 @@ unsigned int RtApiCore :: getDefaultOutputDevice( void ) result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); if ( result != noErr ) { errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -543,7 +566,7 @@ unsigned int RtApiCore :: getDefaultOutputDevice( void ) if ( id == deviceList[i] ) return i; errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -556,13 +579,13 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) unsigned int nDevices = getDeviceCount(); if ( nDevices == 0 ) { errorText_ = "RtApiCore::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } if ( device >= nDevices ) { errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } @@ -575,7 +598,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) 0, NULL, &dataSize, (void *) &deviceList ); if ( result != noErr ) { errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -590,14 +613,18 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); int length = CFStringGetLength(cfname); char *mname = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); +#else CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); +#endif info.name.append( (const char *)mname, strlen(mname) ); info.name.append( ": " ); CFRelease( cfname ); @@ -608,14 +635,18 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); length = CFStringGetLength(cfname); char *name = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); +#else CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); +#endif info.name.append( (const char *)name, strlen(name) ); CFRelease( cfname ); free(name); @@ -630,7 +661,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) if ( result != noErr || dataSize == 0 ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -638,7 +669,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) bufferList = (AudioBufferList *) malloc( dataSize ); if ( bufferList == NULL ) { errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -647,7 +678,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) free( bufferList ); errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -663,7 +694,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) if ( result != noErr || dataSize == 0 ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -671,7 +702,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) bufferList = (AudioBufferList *) malloc( dataSize ); if ( bufferList == NULL ) { errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -680,7 +711,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) free( bufferList ); errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -705,7 +736,7 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) if ( result != kAudioHardwareNoError || dataSize == 0 ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -715,26 +746,45 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) if ( result != kAudioHardwareNoError ) { errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } - Float64 minimumRate = 100000000.0, maximumRate = 0.0; + // The sample rate reporting mechanism is a bit of a mystery. It + // seems that it can either return individual rates or a range of + // rates. I assume that if the min / max range values are the same, + // then that represents a single supported rate and if the min / max + // range values are different, the device supports an arbitrary + // range of values (though there might be multiple ranges, so we'll + // use the most conservative range). + Float64 minimumRate = 1.0, maximumRate = 10000000000.0; + bool haveValueRange = false; + info.sampleRates.clear(); for ( UInt32 i=0; i maximumRate ) maximumRate = rangeList[i].mMaximum; + if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) + info.sampleRates.push_back( (unsigned int) rangeList[i].mMinimum ); + else { + haveValueRange = true; + if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; + if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; + } } - info.sampleRates.clear(); - for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); + if ( haveValueRange ) { + for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); + } } + // Sort and remove any redundant values + std::sort( info.sampleRates.begin(), info.sampleRates.end() ); + info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); + if ( info.sampleRates.size() == 0 ) { errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -792,7 +842,6 @@ static OSStatus rateListener( AudioObjectID inDevice, const AudioObjectPropertyAddress /*properties*/[], void* ratePointer ) { - Float64 *rate = (Float64 *) ratePointer; UInt32 dataSize = sizeof( Float64 ); AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, @@ -864,6 +913,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); if (result != noErr || dataSize == 0) { + free( bufferList ); errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; @@ -1004,7 +1054,6 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne dataSize = sizeof( Float64 ); property.mSelector = kAudioDevicePropertyNominalSampleRate; result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); - if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; errorText_ = errorStream_.str(); @@ -1026,8 +1075,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne nominalRate = (Float64) sampleRate; result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); - if ( result != noErr ) { + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; errorText_ = errorStream_.str(); return FAILURE; @@ -1161,7 +1210,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else { errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } } @@ -1293,6 +1342,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Setup the device property listener for over/underload. property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); return SUCCESS; @@ -1324,7 +1374,7 @@ void RtApiCore :: closeStream( void ) { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -1377,7 +1427,7 @@ void RtApiCore :: startStream( void ) verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiCore::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -1410,7 +1460,7 @@ void RtApiCore :: startStream( void ) unlock: if ( result == noErr ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiCore :: stopStream( void ) @@ -1418,7 +1468,7 @@ void RtApiCore :: stopStream( void ) verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -1453,7 +1503,7 @@ void RtApiCore :: stopStream( void ) unlock: if ( result == noErr ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiCore :: abortStream( void ) @@ -1461,7 +1511,7 @@ void RtApiCore :: abortStream( void ) verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -1492,7 +1542,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return FAILURE; } @@ -1640,11 +1690,12 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, } } } + } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; } AudioDeviceID inputDevice; @@ -1823,7 +1874,6 @@ const char* RtApiCore :: getErrorCode( OSStatus code ) #include "jackbridge/JackBridge.hpp" #include #include -#include // A structure to hold various information related to the Jack API // implementation. @@ -1876,7 +1926,7 @@ unsigned int RtApiJack :: getDeviceCount( void ) } } } while ( ports[++nChannels] ); - free( ports ); + jackbridge_free( ports ); } jackbridge_client_close( client ); @@ -1893,7 +1943,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) jack_client_t *client = jackbridge_client_open( "RtApiJackInfo", options, status ); if ( client == 0 ) { errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -1916,13 +1966,13 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) } } } while ( ports[++nPorts] ); - free( ports ); + jackbridge_free( ports ); } if ( device >= nDevices ) { jackbridge_client_close( client ); errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } @@ -1936,7 +1986,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) ports = jackbridge_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; - free( ports ); + jackbridge_free( ports ); info.outputChannels = nChannels; } @@ -1945,14 +1995,14 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) ports = jackbridge_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; - free( ports ); + jackbridge_free( ports ); info.inputChannels = nChannels; } if ( info.outputChannels == 0 && info.inputChannels == 0 ) { jackbridge_client_close(client); errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -1996,10 +2046,10 @@ static void *jackCloseStream( void *ptr ) object->closeStream(); pthread_exit( NULL ); -#ifdef CARLA_OS_WIN + return NULL; -#endif } + static void jackShutdown( void *infoPointer ) { CallbackInfo *info = (CallbackInfo *) infoPointer; @@ -2012,7 +2062,7 @@ static void jackShutdown( void *infoPointer ) // other problem occurred and we should close the stream. if ( object->isStreamRunning() == false ) return; - pthread_t threadId; + ThreadHandle threadId; pthread_create( &threadId, NULL, jackCloseStream, info ); std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; } @@ -2045,7 +2095,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne client = jackbridge_client_open( "RtApiJack", jackoptions, status ); if ( client == 0 ) { errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return FAILURE; } } @@ -2073,7 +2123,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } } } while ( ports[++nPorts] ); - free( ports ); + jackbridge_free( ports ); } if ( device >= nDevices ) { @@ -2089,7 +2139,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne ports = jackbridge_get_ports( client, deviceName.c_str(), NULL, flag ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; - free( ports ); + jackbridge_free( ports ); } // Compare the jack ports for specified client to the requested number of channels. @@ -2122,7 +2172,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.latency[mode] = latrange.min; //stream_.latency[mode] = jackbridge_port_get_latency( jackbridge_port_by_name( client, ports[ firstChannel ] ) ); } - free( ports ); + jackbridge_free( ports ); // The jack server always uses 32-bit floating-point data. stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; @@ -2282,7 +2332,7 @@ void RtApiJack :: closeStream( void ) { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiJack::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -2324,7 +2374,7 @@ void RtApiJack :: startStream( void ) verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiJack::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -2354,12 +2404,12 @@ void RtApiJack :: startStream( void ) if ( ports[ stream_.channelOffset[0] + i ] ) result = jackbridge_connect( handle->client, jackbridge_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); if ( ! result ) { - free( ports ); + jackbridge_free( ports ); errorText_ = "RtApiJack::startStream(): error connecting output ports!"; goto unlock; } } - free(ports); + jackbridge_free(ports); } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { @@ -2376,12 +2426,12 @@ void RtApiJack :: startStream( void ) if ( ports[ stream_.channelOffset[1] + i ] ) result = jackbridge_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jackbridge_port_name( handle->ports[1][i] ) ); if ( ! result ) { - free( ports ); + jackbridge_free( ports ); errorText_ = "RtApiJack::startStream(): error connecting input ports!"; goto unlock; } } - free(ports); + jackbridge_free(ports); } handle->drainCounter = 0; @@ -2390,7 +2440,7 @@ void RtApiJack :: startStream( void ) unlock: if ( result ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiJack :: stopStream( void ) @@ -2398,7 +2448,7 @@ void RtApiJack :: stopStream( void ) verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -2407,11 +2457,7 @@ void RtApiJack :: stopStream( void ) if ( handle->drainCounter == 0 ) { handle->drainCounter = 2; -#ifdef CARLA_OS_WIN // FIXME - Sleep(500); //ms -#else pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled -#endif } } @@ -2424,7 +2470,7 @@ void RtApiJack :: abortStream( void ) verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -2446,9 +2492,8 @@ static void *jackStopStream( void *ptr ) object->stopStream(); pthread_exit( NULL ); -#ifdef CARLA_OS_WIN + return NULL; -#endif } bool RtApiJack :: callbackEvent( unsigned long nframes ) @@ -2456,12 +2501,12 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return FAILURE; } if ( stream_.bufferSize != nframes ) { errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return FAILURE; } @@ -2470,7 +2515,7 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { - pthread_t threadId; + ThreadHandle threadId; stream_.state = STREAM_STOPPING; if ( handle->internalDrain == true ) @@ -2498,7 +2543,7 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) if ( cbReturnValue == 2 ) { stream_.state = STREAM_STOPPING; handle->drainCounter = 2; - pthread_t id; + ThreadHandle id; pthread_create( &id, NULL, jackStopStream, info ); return SUCCESS; } @@ -2535,11 +2580,12 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); } } + } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { @@ -2620,7 +2666,7 @@ RtApiAsio :: RtApiAsio() HRESULT hr = CoInitialize( NULL ); if ( FAILED(hr) ) { errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } coInitialized_ = true; @@ -2651,13 +2697,13 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) unsigned int nDevices = getDeviceCount(); if ( nDevices == 0 ) { errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } if ( device >= nDevices ) { errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } @@ -2665,7 +2711,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) if ( stream_.state != STREAM_CLOSED ) { if ( device >= devices_.size() ) { errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } return devices_[ device ]; @@ -2676,7 +2722,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -2685,7 +2731,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) if ( !drivers.loadDriver( driverName ) ) { errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -2693,7 +2739,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -2704,7 +2750,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -2731,7 +2777,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -2757,7 +2803,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) return info; } -static void bufferSwitch( long index, ASIOBool processNow ) +static void bufferSwitch( long index, ASIOBool /*processNow*/ ) { RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; object->callbackEvent( index ); @@ -3091,7 +3137,7 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; errorText_ = errorStream_.str(); - error( RtError::WARNING); // warn but don't fail + error( RtAudioError::WARNING); // warn but don't fail } else { stream_.latency[0] = outputLatency; @@ -3137,7 +3183,7 @@ void RtApiAsio :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -3180,7 +3226,7 @@ void RtApiAsio :: startStream() verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAsio::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -3202,7 +3248,7 @@ void RtApiAsio :: startStream() stopThreadCalled = false; if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiAsio :: stopStream() @@ -3210,7 +3256,7 @@ void RtApiAsio :: stopStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -3231,7 +3277,7 @@ void RtApiAsio :: stopStream() } if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiAsio :: abortStream() @@ -3239,7 +3285,7 @@ void RtApiAsio :: abortStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -3272,7 +3318,7 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return FAILURE; } @@ -3366,11 +3412,12 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) } } + } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { @@ -3432,7 +3479,7 @@ static void sampleRateChanged( ASIOSampleRate sRate ) try { object->stopStream(); } - catch ( RtError &exception ) { + catch ( RtAudioError &exception ) { std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; return; } @@ -3440,7 +3487,7 @@ static void sampleRateChanged( ASIOSampleRate sRate ) std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; } -static long asioMessages( long selector, long value, void* message, double* opt ) +static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) { long ret = 0; @@ -3534,185 +3581,1707 @@ static const char* getAsioErrorString( ASIOError result ) return "Unknown error."; } + //******************** End of __WINDOWS_ASIO__ *********************// #endif -#if defined(__WINDOWS_DS__) // Windows DirectSound API - -// Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. -// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. -// - Auto-call CoInitialize for DSOUND and ASIO platforms. -// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 -// Changed device query structure for RtAudio 4.0.7, January 2010 +#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API -#include -#include -#include +// Authored by Marcus Tomlinson , April 2014 +// - Introduces support for the Windows WASAPI API +// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required +// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface +// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user -#if defined(__MINGW32__) - // missing from latest mingw winapi -#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ -#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ -#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ -#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#ifndef INITGUID + #define INITGUID #endif +#include +#include +#include +#include + +//============================================================================= + +#define SAFE_RELEASE( objectPtr )\ +if ( objectPtr )\ +{\ + objectPtr->Release();\ + objectPtr = NULL;\ +} -#define MINIMUM_DEVICE_BUFFER_SIZE 32768 +typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); -#ifdef _MSC_VER // if Microsoft Visual C++ -#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. -#endif +//----------------------------------------------------------------------------- -static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +// 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 { - if ( pointer > bufferSize ) pointer -= bufferSize; - if ( laterPointer < earlierPointer ) laterPointer += bufferSize; - if ( pointer < earlierPointer ) pointer += bufferSize; - return pointer >= earlierPointer && pointer < laterPointer; -} +public: + WasapiBuffer() + : buffer_( NULL ), + bufferSize_( 0 ), + inIndex_( 0 ), + outIndex_( 0 ) {} -// A structure to hold various information related to the DirectSound -// API implementation. -struct DsHandle { - unsigned int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - void *id[2]; - void *buffer[2]; - bool xrun[2]; - UINT bufferPointer[2]; - DWORD dsBufferSize[2]; - DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. - HANDLE condition; + ~WasapiBuffer() { + delete buffer_; + } - DsHandle() - :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } -}; + // sets the length of the internal ring buffer + void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { + delete buffer_; -// Declarations for utility functions, callbacks, and structures -// specific to the DirectSound implementation. -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ); + buffer_ = ( char* ) calloc( bufferSize, formatBytes ); -static const char* getErrorString( int code ); + bufferSize_ = bufferSize; + inIndex_ = 0; + outIndex_ = 0; + } -static unsigned __stdcall callbackHandler( void *ptr ); + // 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; + } -struct DsDevice { - LPGUID id[2]; - bool validId[2]; - bool found; - std::string name; + unsigned int relOutIndex = outIndex_; + unsigned int inIndexEnd = inIndex_ + bufferSize; + if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { + relOutIndex += bufferSize_; + } - DsDevice() - : found(false) { validId[0] = false; validId[1] = false; } -}; + // "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 + } -struct DsProbeData { - bool isInput; - std::vector* dsDevices; + // 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_; }; -RtApiDs :: RtApiDs() +//----------------------------------------------------------------------------- + +// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate +// between HW and the user. The convertBufferWasapi function is used to perform this conversion +// 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. +void convertBufferWasapi( char* outBuffer, + const char* inBuffer, + const unsigned int& channelCount, + const unsigned int& inSampleRate, + const unsigned int& outSampleRate, + const unsigned int& inSampleCount, + unsigned int& outSampleCount, + const RtAudioFormat& format ) { - // Dsound will run both-threaded. If CoInitialize fails, then just - // accept whatever the mainline chose for a threading model. - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) coInitialized_ = true; + // calculate the new outSampleCount and relative sampleStep + float sampleRatio = ( float ) outSampleRate / inSampleRate; + float sampleStep = 1.0f / sampleRatio; + float inSampleFraction = 0.0f; + + 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 * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) ); + break; + } + + // jump to next in sample + inSampleFraction += sampleStep; + } } -RtApiDs :: ~RtApiDs() +//----------------------------------------------------------------------------- + +// A structure to hold various information related to the WASAPI implementation. +struct WasapiHandle { - if ( coInitialized_ ) CoUninitialize(); // balanced call. - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} + 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 ) {} +}; -// The DirectSound default output is always the first device. -unsigned int RtApiDs :: getDefaultOutputDevice( void ) +//============================================================================= + +RtApiWasapi::RtApiWasapi() + : coInitialized_( false ), deviceEnumerator_( NULL ) { - return 0; + // 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 ); + } } -// The DirectSound default input is always the first input device, -// which is the first capture device enumerated. -unsigned int RtApiDs :: getDefaultInputDevice( void ) +//----------------------------------------------------------------------------- + +RtApiWasapi::~RtApiWasapi() { - return 0; + if ( stream_.state != STREAM_CLOSED ) + closeStream(); + + SAFE_RELEASE( deviceEnumerator_ ); + + // If this object previously called CoInitialize() + if ( coInitialized_ ) + CoUninitialize(); } -unsigned int RtApiDs :: getDeviceCount( void ) +//============================================================================= + +unsigned int RtApiWasapi::getDeviceCount( void ) { - // Set query flag for previously found devices to false, so that we - // can check for any devices that have disappeared. - for ( unsigned int i=0; iEnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; + goto Exit; } - // Query DirectSoundCapture devices. - probeInfo.isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; + goto Exit; } - // Clean out any devices that may have disappeared. - std::vector< int > indices; - for ( unsigned int i=0; iEnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; + goto Exit; + } + +Exit: + // release all references + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); - return dsDevices.size(); + if ( errorText_.empty() ) + return captureDeviceCount + renderDeviceCount; + + error( RtAudioError::DRIVER_ERROR ); + return 0; } -RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +//----------------------------------------------------------------------------- + +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; - if ( dsDevices.size() == 0 ) { - // Force a query of all devices - getDeviceCount(); - if ( dsDevices.size() == 0 ) { - errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - return info; - } + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; + goto Exit; } - if ( device >= dsDevices.size() ) { - errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - return info; + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; + goto Exit; } - HRESULT result; - if ( dsDevices[ device ].validId[0] == false ) goto probeInput; + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; + goto Exit; + } - LPDIRECTSOUND output; - DSCAPS outCaps; - result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; + errorType = RtAudioError::INVALID_USE; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; + goto Exit; + } + isCaptureDevice = true; + } + else { + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; + goto Exit; + } + isCaptureDevice = false; + } + + // get default device name + if ( isCaptureDevice ) { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; + goto Exit; + } + } + else { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; + goto Exit; + } + } + + hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; + goto Exit; + } + PropVariantInit( &defaultDeviceNameProp ); + + hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + deviceName = defaultDeviceNameProp.pwszVal; + defaultDeviceName = std::string( deviceName.begin(), deviceName.end() ); + + // name + hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; + goto Exit; + } + + PropVariantInit( &deviceNameProp ); + + hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + 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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; + goto Exit; + } + + hr = audioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + goto Exit; + } + + 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 all sample rates as we have a built-in sample rate converter + for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { + 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 ); + + if ( !errorText_.empty() ) + error( errorType ); + 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 ( WasapiHandle* ) 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 = ( ThreadHandle ) 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 ); + return; + } + } + + // 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 ); + return; + } + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) 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::abortStream: Unable to stop capture stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // stop render client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) 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; + unsigned int bufferBytes; + stream_.state = STREAM_STOPPED; + + // create API Handle if not already created + if ( !stream_.apiHandle ) + stream_.apiHandle = ( void* ) new WasapiHandle(); + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + if ( mode != INPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; + goto Exit; + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + else { + if ( mode != OUTPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device."; + goto Exit; + } + + // retrieve renderAudioClient from devicePtr + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &renderAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; + } + + hr = renderAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; + } + + 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_.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_.nUserChannels != stream_.nDeviceChannels ) + stream_.doConvertBuffer[mode] = true; + else 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 + bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat ); + + stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 ); + if ( !stream_.userBuffer[mode] ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; + goto Exit; + } + + 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(); + + if ( !errorText_.empty() ) + error( errorType ); + 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; + + // 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; + unsigned int convBuffSize = 0; + unsigned int deviceBuffSize = 0; + + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + + // Attempt to assign "Pro Audio" characteristic to thread + HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + 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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; + goto Exit; + } + + hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), + ( void** ) &captureClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; + goto Exit; + } + + // configure captureEvent to trigger on every available capture buffer + captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !captureEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event."; + goto Exit; + } + + hr = captureAudioClient->SetEventHandle( captureEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; + } + + unsigned int inBufferSize = 0; + hr = captureAudioClient->GetBufferSize( &inBufferSize ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; + goto Exit; + } + + // scale outBufferSize according to stream->user sample rate ratio + unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * 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(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; + goto Exit; + } + + // start the capture stream + hr = captureAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream."; + goto Exit; + } + } + + // start render stream if applicable + if ( renderAudioClient ) { + hr = renderAudioClient->GetMixFormat( &renderFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + 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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; + goto Exit; + } + + hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), + ( void** ) &renderClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; + goto Exit; + } + + // configure renderEvent to trigger on every available render buffer + renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !renderEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event."; + goto Exit; + } + + hr = renderAudioClient->SetEventHandle( renderEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; + ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; + } + + unsigned int outBufferSize = 0; + hr = renderAudioClient->GetBufferSize( &outBufferSize ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; + goto Exit; + } + + // scale inBufferSize according to user->stream sample rate ratio + unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * 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(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream."; + goto Exit; + } + + // start the render stream + hr = renderAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream."; + goto Exit; + } + } + + if ( stream_.mode == INPUT ) { + convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + } + else if ( stream_.mode == OUTPUT ) { + convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + } + else if ( stream_.mode == DUPLEX ) { + convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + } + + convBuffer = ( char* ) malloc( convBuffSize ); + stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize ); + if ( !convBuffer || !stream_.deviceBuffer ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; + goto Exit; + } + + // 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 * captureSrRatio ) * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ); + + if ( callbackPulled ) { + // Convert callback buffer to user sample rate + convertBufferWasapi( stream_.deviceBuffer, + convBuffer, + stream_.nDeviceChannels[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 further 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, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; + goto Exit; + } + + callbackStopped = true; + } + else if ( callbackResult == 2 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; + goto Exit; + } + + 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 + convertBufferWasapi( convBuffer, + stream_.deviceBuffer, + 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] ); + } + else { + // if there is no render stream, set callbackPushed flag + callbackPushed = true; + } + + // 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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; + goto Exit; + } + + 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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + + // 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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; + goto Exit; + } + + hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; + goto Exit; + } + + bufferFrameCount -= numFramesPadding; + + if ( bufferFrameCount != 0 ) { + hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; + goto Exit; + } + + // 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 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + + // 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 ); + + free ( convBuffer ); + + CoUninitialize(); + + // update stream state + stream_.state = STREAM_STOPPED; + + if ( errorText_.empty() ) + return; + else + error( errorType ); +} + +//******************** End of __WINDOWS_WASAPI__ *********************// +#endif + + +#if defined(__WINDOWS_DS__) // Windows DirectSound API + +// Modified by Robin Davies, October 2005 +// - Improvements to DirectX pointer chasing. +// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. +// - Auto-call CoInitialize for DSOUND and ASIO platforms. +// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 +// Changed device query structure for RtAudio 4.0.7, January 2010 + +#include +#include +#include + +#if defined(__MINGW32__) + // missing from latest mingw winapi +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ +#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 + +#ifdef _MSC_VER // if Microsoft Visual C++ +#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. +#endif + +static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; +} + +// A structure to hold various information related to the DirectSound +// API implementation. +struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; + + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } +}; + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); + +static const char* getErrorString( int code ); + +static unsigned __stdcall callbackHandler( void *ptr ); + +struct DsDevice { + LPGUID id[2]; + bool validId[2]; + bool found; + std::string name; + + DsDevice() + : found(false) { validId[0] = false; validId[1] = false; } +}; + +struct DsProbeData { + bool isInput; + std::vector* dsDevices; +}; + +RtApiDs :: RtApiDs() +{ + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; +} + +RtApiDs :: ~RtApiDs() +{ + if ( coInitialized_ ) CoUninitialize(); // balanced call. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +// The DirectSound default output is always the first device. +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + return 0; +} + +// The DirectSound default input is always the first input device, +// which is the first capture device enumerated. +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + return 0; +} + +unsigned int RtApiDs :: getDeviceCount( void ) +{ + // Set query flag for previously found devices to false, so that we + // can check for any devices that have disappeared. + for ( unsigned int i=0; i indices; + for ( unsigned int i=0; i(dsDevices.size()); +} + +RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + if ( dsDevices.size() == 0 ) { + // Force a query of all devices + getDeviceCount(); + if ( dsDevices.size() == 0 ) { + errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + } + + if ( device >= dsDevices.size() ) { + errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + HRESULT result; + if ( dsDevices[ device ].validId[0] == false ) goto probeInput; + + LPDIRECTSOUND output; + DSCAPS outCaps; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto probeInput; } @@ -3722,7 +5291,7 @@ RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) output->Release(); errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto probeInput; } @@ -3759,7 +5328,7 @@ RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) if ( FAILED( result ) ) { errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -3770,7 +5339,7 @@ RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) input->Release(); errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -3867,7 +5436,7 @@ bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned return FAILURE; } - unsigned int nDevices = dsDevices.size(); + size_t nDevices = dsDevices.size(); if ( nDevices == 0 ) { // This should not happen because a check is made before this function is called. errorText_ = "RtApiDs::probeDeviceOpen: no devices found!"; @@ -4363,7 +5932,7 @@ void RtApiDs :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -4418,7 +5987,7 @@ void RtApiDs :: startStream() verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiDs::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -4466,7 +6035,7 @@ void RtApiDs :: startStream() stream_.state = STREAM_RUNNING; unlock: - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); } void RtApiDs :: stopStream() @@ -4474,7 +6043,7 @@ void RtApiDs :: stopStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -4490,6 +6059,8 @@ void RtApiDs :: stopStream() stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + // Stop the buffer and clear memory LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; result = buffer->Stop(); @@ -4530,6 +6101,9 @@ void RtApiDs :: stopStream() stream_.state = STREAM_STOPPED; + if ( stream_.mode != DUPLEX ) + MUTEX_LOCK( &stream_.mutex ); + result = buffer->Stop(); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; @@ -4563,7 +6137,9 @@ void RtApiDs :: stopStream() unlock: timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); } void RtApiDs :: abortStream() @@ -4571,7 +6147,7 @@ void RtApiDs :: abortStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -4590,7 +6166,7 @@ void RtApiDs :: callbackEvent() if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -4649,6 +6225,12 @@ void RtApiDs :: callbackEvent() char *buffer; long bufferBytes; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + if ( buffersRolling == false ) { if ( stream_.mode == DUPLEX ) { //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); @@ -4676,14 +6258,14 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } while ( true ) { @@ -4691,14 +6273,14 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; @@ -4719,7 +6301,7 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; @@ -4770,7 +6352,7 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } @@ -4811,7 +6393,7 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } @@ -4824,16 +6406,17 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; handle->bufferPointer[0] = nextWritePointer; + } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; } if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { @@ -4859,7 +6442,7 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } @@ -4920,7 +6503,7 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } @@ -4934,7 +6517,7 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } @@ -4955,7 +6538,7 @@ void RtApiDs :: callbackEvent() if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } handle->bufferPointer[1] = nextReadPointer; @@ -4972,6 +6555,7 @@ void RtApiDs :: callbackEvent() } unlock: + MUTEX_UNLOCK( &stream_.mutex ); RtApi::tickStreamTime(); } @@ -5009,7 +6593,7 @@ static std::string convertTChar( LPCTSTR name ) static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, LPCTSTR description, - LPCTSTR module, + LPCTSTR /*module*/, LPVOID lpContext ) { struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; @@ -5188,7 +6772,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto nextcard; } subdevice = -1; @@ -5197,7 +6781,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); break; } if ( subdevice < 0 ) @@ -5237,7 +6821,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto nextcard; } subdevice = -1; @@ -5246,7 +6830,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); break; } if ( subdevice < 0 ) break; @@ -5272,13 +6856,13 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) if ( nDevices == 0 ) { errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } if ( device >= nDevices ) { errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } @@ -5291,7 +6875,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) snd_ctl_close( chandle ); if ( device >= devices_.size() ) { errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } return devices_[ device ]; @@ -5323,7 +6907,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto captureProbe; } @@ -5333,7 +6917,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto captureProbe; } @@ -5344,7 +6928,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto captureProbe; } info.outputChannels = value; @@ -5364,12 +6948,14 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) goto probeParameters; } } + else + snd_ctl_close( chandle ); result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); if ( info.outputChannels == 0 ) return info; goto probeParameters; } @@ -5380,7 +6966,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); if ( info.outputChannels == 0 ) return info; goto probeParameters; } @@ -5390,7 +6976,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); if ( info.outputChannels == 0 ) return info; goto probeParameters; } @@ -5424,7 +7010,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) if ( result < 0 ) { errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -5434,7 +7020,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -5448,7 +7034,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -5476,17 +7062,20 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) // Check that we have at least one supported format if ( info.nativeFormats == 0 ) { + snd_pcm_close( phandle ); errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } // Get the device name char *cardname; result = snd_card_get_name( card, &cardname ); - if ( result >= 0 ) + if ( result >= 0 ) { sprintf( name, "hw:%s,%d", cardname, subdevice ); + free( cardname ); + } info.name = name; // That's all ... close the device and return @@ -5942,7 +7531,7 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne apiInfo->synchronized = true; else { errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } } else { @@ -6020,7 +7609,7 @@ void RtApiAlsa :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6073,7 +7662,7 @@ void RtApiAlsa :: startStream() verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6096,6 +7685,7 @@ void RtApiAlsa :: startStream() } if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open state = snd_pcm_state( handle[1] ); if ( state != SND_PCM_STATE_PREPARED ) { result = snd_pcm_prepare( handle[1] ); @@ -6115,7 +7705,7 @@ void RtApiAlsa :: startStream() MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiAlsa :: stopStream() @@ -6123,7 +7713,7 @@ void RtApiAlsa :: stopStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6155,10 +7745,11 @@ void RtApiAlsa :: stopStream() } unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiAlsa :: abortStream() @@ -6166,7 +7757,7 @@ void RtApiAlsa :: abortStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6195,10 +7786,11 @@ void RtApiAlsa :: abortStream() } unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiAlsa :: callbackEvent() @@ -6218,7 +7810,7 @@ void RtApiAlsa :: callbackEvent() if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6301,7 +7893,7 @@ void RtApiAlsa :: callbackEvent() errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto tryOutput; } @@ -6371,7 +7963,7 @@ void RtApiAlsa :: callbackEvent() errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto unlock; } @@ -6546,7 +8138,7 @@ void RtApiPulse::callbackEvent( void ) if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " "this shouldn't happen!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6586,7 +8178,7 @@ void RtApiPulse::callbackEvent( void ) errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } } @@ -6602,7 +8194,7 @@ void RtApiPulse::callbackEvent( void ) errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } if ( stream_.doConvertBuffer[INPUT] ) { convertBuffer( stream_.userBuffer[INPUT], @@ -6625,12 +8217,12 @@ void RtApiPulse::startStream( void ) if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::startStream(): the stream is not open!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiPulse::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6649,12 +8241,12 @@ void RtApiPulse::stopStream( void ) if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6668,7 +8260,7 @@ void RtApiPulse::stopStream( void ) pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } } @@ -6683,12 +8275,12 @@ void RtApiPulse::abortStream( void ) if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return; } if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -6702,7 +8294,7 @@ void RtApiPulse::abortStream( void ) pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); return; } } @@ -6750,26 +8342,34 @@ 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 ); @@ -6823,16 +8423,21 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, pah = static_cast( stream_.apiHandle ); int error; + if ( !options->streamName.empty() ) streamName = options->streamName; switch ( mode ) { case INPUT: - pah->s_rec = pa_simple_new( NULL, options ? options->streamName.c_str() : "RtAudio", PA_STREAM_RECORD, NULL, "Record", &ss, NULL, NULL, &error ); + pa_buffer_attr buffer_attr; + buffer_attr.fragsize = bufferBytes; + buffer_attr.maxlength = -1; + + pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error ); if ( !pah->s_rec ) { errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; goto error; } break; case OUTPUT: - pah->s_play = pa_simple_new( NULL, options ? options->streamName.c_str() : "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); if ( !pah->s_play ) { errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; goto error; @@ -6892,7 +8497,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, #include #include #include -#include "soundcard.h" +#include #include #include @@ -6925,7 +8530,7 @@ unsigned int RtApiOss :: getDeviceCount( void ) int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); if ( mixerfd == -1 ) { errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -6933,7 +8538,7 @@ unsigned int RtApiOss :: getDeviceCount( void ) if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -6949,7 +8554,7 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); if ( mixerfd == -1 ) { errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -6958,7 +8563,7 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) if ( result == -1 ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -6966,14 +8571,14 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) if ( nDevices == 0 ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } if ( device >= nDevices ) { close( mixerfd ); errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); return info; } @@ -6984,7 +8589,7 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) if ( result == -1 ) { errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -7013,7 +8618,7 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) if ( info.nativeFormats == 0 ) { errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -7040,7 +8645,7 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) if ( info.sampleRates.size() == 0 ) { errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } else { info.probed = true; @@ -7489,7 +9094,7 @@ void RtApiOss :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7538,7 +9143,7 @@ void RtApiOss :: startStream() verifyStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiOss::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7560,7 +9165,7 @@ void RtApiOss :: stopStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7597,7 +9202,7 @@ void RtApiOss :: stopStream() result = write( handle->id[0], buffer, samples * formatBytes(format) ); if ( result == -1 ) { errorText_ = "RtApiOss::stopStream: audio write error."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); } } @@ -7624,7 +9229,7 @@ void RtApiOss :: stopStream() MUTEX_UNLOCK( &stream_.mutex ); if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiOss :: abortStream() @@ -7632,7 +9237,7 @@ void RtApiOss :: abortStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7670,7 +9275,7 @@ void RtApiOss :: abortStream() MUTEX_UNLOCK( &stream_.mutex ); if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiOss :: callbackEvent() @@ -7688,7 +9293,7 @@ void RtApiOss :: callbackEvent() if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7758,7 +9363,7 @@ void RtApiOss :: callbackEvent() // specific means for determining that. handle->xrun[0] = true; errorText_ = "RtApiOss::callbackEvent: audio write error."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); // Continue on to input section. } } @@ -7785,7 +9390,7 @@ void RtApiOss :: callbackEvent() // specific means for determining that. handle->xrun[1] = true; errorText_ = "RtApiOss::callbackEvent: audio read error."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); goto unlock; } @@ -7831,42 +9436,41 @@ static void *ossCallbackHandler( void *ptr ) // This method can be modified to control the behavior of error // message printing. -void RtApi :: error( RtError::Type type ) +void RtApi :: error( RtAudioError::Type type ) { errorStream_.str(""); // clear the ostringstream RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; if ( errorCallback ) { // abortStream() can generate new error messages. Ignore them. Just keep original one. - static bool firstErrorOccured = false; - if ( firstErrorOccured ) + if ( firstErrorOccurred_ ) return; - firstErrorOccured = true; + firstErrorOccurred_ = true; const std::string errorMessage = errorText_; - if ( type != RtError::WARNING && stream_.state != STREAM_STOPPED) { + if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { stream_.callbackInfo.isRunning = false; // exit from the thread abortStream(); } errorCallback( type, errorMessage ); - firstErrorOccured = false; + firstErrorOccurred_ = false; return; } - if ( type == RtError::WARNING && showWarnings_ == true ) + if ( type == RtAudioError::WARNING && showWarnings_ == true ) std::cerr << '\n' << errorText_ << "\n\n"; - else if ( type != RtError::WARNING ) - throw( RtError( errorText_, type ) ); + else if ( type != RtAudioError::WARNING ) + throw( RtAudioError( errorText_, type ) ); } void RtApi :: verifyStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApi:: a stream is not open!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); } } @@ -7921,7 +9525,7 @@ unsigned int RtApi :: formatBytes( RtAudioFormat format ) return 1; errorText_ = "RtApi::formatBytes: undefined format."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } diff --git a/source/modules/rtaudio/RtAudio.h b/source/modules/rtaudio/RtAudio.h index 4d1fbc3a0..57d2c8dcd 100644 --- a/source/modules/rtaudio/RtAudio.h +++ b/source/modules/rtaudio/RtAudio.h @@ -5,12 +5,12 @@ RtAudio provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, Jack, and OSS), Macintosh OS X (CoreAudio and Jack), and Windows - (DirectSound and ASIO) operating systems. + (DirectSound, ASIO and WASAPI) operating systems. RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2013 Gary P. Scavone + Copyright (c) 2001-2014 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -45,12 +45,12 @@ #ifndef __RTAUDIO_H #define __RTAUDIO_H +#define RTAUDIO_VERSION "4.1.1" + #include #include -#include "RtError.h" - -// RtAudio version -static const std::string VERSION( "4.0.12" ); +#include +#include /*! \typedef typedef unsigned long RtAudioFormat; \brief RtAudio data format type. @@ -185,12 +185,63 @@ typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, RtAudioStreamStatus status, void *userData ); +/************************************************************************/ +/*! \class RtAudioError + \brief Exception handling class for RtAudio. + + The RtAudioError class is quite simple but it does allow errors to be + "caught" by RtAudioError::Type. See the RtAudio documentation to know + which methods can throw an RtAudioError. +*/ +/************************************************************************/ + +class RtAudioError : public std::exception +{ + public: + //! Defined RtAudioError types. + enum Type { + WARNING, /*!< A non-critical error. */ + DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ + UNSPECIFIED, /*!< The default, unspecified error type. */ + NO_DEVICES_FOUND, /*!< No devices found on system. */ + INVALID_DEVICE, /*!< An invalid device ID was specified. */ + MEMORY_ERROR, /*!< An error occured during memory allocation. */ + INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ + INVALID_USE, /*!< The function was called incorrectly. */ + DRIVER_ERROR, /*!< A system driver error occured. */ + SYSTEM_ERROR, /*!< A system error occured. */ + THREAD_ERROR /*!< A thread error occured. */ + }; + + //! The constructor. + RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) throw() : message_(message), type_(type) {} + + //! The destructor. + virtual ~RtAudioError( void ) throw() {} + + //! Prints thrown error message to stderr. + virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; } + + //! Returns the thrown error message type. + virtual const Type& getType(void) const throw() { return type_; } + + //! Returns the thrown error message string. + virtual const std::string& getMessage(void) const throw() { return message_; } + + //! Returns the thrown error message as a c-style string. + virtual const char* what( void ) const throw() { return message_.c_str(); } + + protected: + std::string message_; + Type type_; +}; + //! RtAudio error callback function prototype. /*! \param type Type of error. \param errorText Error description. */ -typedef void (*RtAudioErrorCallback)( RtError::Type type, const std::string &errorText ); +typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText ); // **************************************************************** // // @@ -219,6 +270,7 @@ class RtAudio LINUX_OSS, /*!< The Linux Open Sound System API. */ UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ + WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */ WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ WINDOWS_DS, /*!< The Microsoft Direct Sound API. */ RTAUDIO_DUMMY /*!< A compilable but non-functional API. */ @@ -322,7 +374,7 @@ class RtAudio }; //! A static function to determine the current RtAudio version. - static std::string getVersion( void ) { return VERSION; } + static std::string getVersion( void ) throw(); //! A static function to determine the available compiled audio APIs. /*! @@ -334,14 +386,14 @@ class RtAudio //! The class constructor. /*! - The constructor performs minor initialization tasks. No exceptions - can be thrown. + The constructor performs minor initialization tasks. An exception + can be thrown if no API support is compiled. If no API argument is specified and multiple API support has been compiled, the default order of use is JACK, ALSA, OSS (Linux systems) and ASIO, DS (Windows systems). */ - RtAudio( RtAudio::Api api=UNSPECIFIED ) throw(); + RtAudio( RtAudio::Api api=UNSPECIFIED ); //! The destructor. /*! @@ -351,7 +403,7 @@ class RtAudio ~RtAudio() throw(); //! Returns the audio API specifier for the current instance of RtAudio. - RtAudio::Api getCurrentApi( void ) const throw(); + RtAudio::Api getCurrentApi( void ) throw(); //! A public function that queries for the number of audio devices available. /*! @@ -365,7 +417,7 @@ class RtAudio /*! Any device integer between 0 and getDeviceCount() - 1 is valid. - If an invalid argument is provided, an RtError (type = INVALID_USE) + If an invalid argument is provided, an RtAudioError (type = INVALID_USE) will be thrown. If a device is busy or otherwise unavailable, the structure member "probed" will have a value of "false" and all other members are undefined. If the specified device is the @@ -396,9 +448,9 @@ class RtAudio //! A public function for opening a stream with the specified parameters. /*! - An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be + An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be opened with the specified parameters or an error occurs during - processing. An RtError (type = INVALID_USE) is thrown if any + processing. An RtAudioError (type = INVALID_USE) is thrown if any invalid device ID or channel number parameters are specified. \param outputParameters Specifies output stream parameters to use @@ -449,8 +501,8 @@ class RtAudio //! A function that starts a stream. /*! - An RtError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtError (type = INVALID_USE) is thrown if a + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a stream is not open. A warning is issued if the stream is already running. */ @@ -458,8 +510,8 @@ class RtAudio //! Stop a stream, allowing any samples remaining in the output queue to be played. /*! - An RtError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtError (type = INVALID_USE) is thrown if a + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a stream is not open. A warning is issued if the stream is already stopped. */ @@ -467,8 +519,8 @@ class RtAudio //! Stop a stream, discarding any samples remaining in the input/output queue. /*! - An RtError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtError (type = INVALID_USE) is thrown if a + An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtAudioError (type = INVALID_USE) is thrown if a stream is not open. A warning is issued if the stream is already stopped. */ @@ -482,17 +534,23 @@ class RtAudio //! Returns the number of elapsed seconds since the stream was started. /*! - If a stream is not open, an RtError (type = INVALID_USE) will be thrown. + If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. */ double getStreamTime( void ); + //! Set the stream time to a time in seconds greater than or equal to 0.0. + /*! + If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. + */ + void setStreamTime( double time ); + //! Returns the internal stream latency in sample frames. /*! The stream latency refers to delay in audio input and/or output caused by internal buffering by the audio system and/or hardware. For duplex streams, the returned value will represent the sum of the input and output latencies. If a stream is not open, an - RtError (type = INVALID_USE) will be thrown. If the API does not + RtAudioError (type = INVALID_USE) will be thrown. If the API does not report latency, the return value will be zero. */ long getStreamLatency( void ); @@ -501,7 +559,7 @@ class RtAudio /*! On some systems, the sample rate used may be slightly different than that specified in the stream parameters. If a stream is not - open, an RtError (type = INVALID_USE) will be thrown. + open, an RtAudioError (type = INVALID_USE) will be thrown. */ unsigned int getStreamSampleRate( void ); @@ -515,12 +573,15 @@ class RtAudio }; // Operating system dependent thread functionality. -#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) - #include +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) + + #ifndef NOMINMAX + #define NOMINMAX + #endif #include #include - typedef unsigned long ThreadHandle; + typedef uintptr_t ThreadHandle; typedef CRITICAL_SECTION StreamMutex; #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) @@ -613,7 +674,7 @@ public: RtApi(); virtual ~RtApi(); - virtual RtAudio::Api getCurrentApi( void ) const = 0; + virtual RtAudio::Api getCurrentApi( void ) = 0; virtual unsigned int getDeviceCount( void ) = 0; virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; virtual unsigned int getDefaultInputDevice( void ); @@ -631,6 +692,7 @@ public: long getStreamLatency( void ); unsigned int getStreamSampleRate( void ); virtual double getStreamTime( void ); + virtual void setStreamTime( double time ); bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; } bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; } void showWarnings( bool value ) { showWarnings_ = value; } @@ -710,6 +772,7 @@ protected: std::string errorText_; bool showWarnings_; RtApiStream stream_; + bool firstErrorOccurred_; /*! Protected, api-specific method that attempts to open a device @@ -730,13 +793,13 @@ protected: void clearStreamInfo(); /*! - Protected common method that throws an RtError (type = + Protected common method that throws an RtAudioError (type = INVALID_USE) if a stream is not open. */ void verifyStream( void ); //! Protected common error method to allow global control over error handling. - void error( RtError::Type type ); + void error( RtAudioError::Type type ); /*! Protected method used to perform format, channel number, and/or interleaving @@ -760,7 +823,7 @@ protected: // // **************************************************************** // -inline RtAudio::Api RtAudio :: getCurrentApi( void ) const throw() { return rtapi_->getCurrentApi(); } +inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); } inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); } inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); } @@ -774,6 +837,7 @@ inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->is inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } +inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); } inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); } // RtApi Subclass prototypes. @@ -788,7 +852,7 @@ public: RtApiCore(); ~RtApiCore(); - RtAudio::Api getCurrentApi( void ) const { return RtAudio::MACOSX_CORE; } + RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); unsigned int getDefaultOutputDevice( void ); @@ -826,7 +890,7 @@ public: RtApiJack(); ~RtApiJack(); - RtAudio::Api getCurrentApi( void ) const { return RtAudio::UNIX_JACK; } + RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); @@ -859,7 +923,7 @@ public: RtApiAsio(); ~RtApiAsio(); - RtAudio::Api getCurrentApi( void ) const { return RtAudio::WINDOWS_ASIO; } + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); @@ -895,7 +959,7 @@ public: RtApiDs(); ~RtApiDs(); - RtAudio::Api getCurrentApi( void ) const { return RtAudio::WINDOWS_DS; } + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; } unsigned int getDeviceCount( void ); unsigned int getDefaultOutputDevice( void ); unsigned int getDefaultInputDevice( void ); @@ -926,6 +990,43 @@ public: #endif +#if defined(__WINDOWS_WASAPI__) + +struct IMMDeviceEnumerator; + +class RtApiWasapi : public RtApi +{ +public: + RtApiWasapi(); + ~RtApiWasapi(); + + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; } + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + +private: + bool coInitialized_; + IMMDeviceEnumerator* deviceEnumerator_; + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int* bufferSize, + RtAudio::StreamOptions* options ); + + static DWORD WINAPI runWasapiThread( void* wasapiPtr ); + static DWORD WINAPI stopWasapiThread( void* wasapiPtr ); + static DWORD WINAPI abortWasapiThread( void* wasapiPtr ); + void wasapiThread(); +}; + +#endif + #if defined(__LINUX_ALSA__) class RtApiAlsa: public RtApi @@ -934,7 +1035,7 @@ public: RtApiAlsa(); ~RtApiAlsa(); - RtAudio::Api getCurrentApi() const { return RtAudio::LINUX_ALSA; } + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); @@ -966,7 +1067,7 @@ class RtApiPulse: public RtApi { public: ~RtApiPulse(); - RtAudio::Api getCurrentApi() const { return RtAudio::LINUX_PULSE; } + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); @@ -1000,7 +1101,7 @@ public: RtApiOss(); ~RtApiOss(); - RtAudio::Api getCurrentApi() const { return RtAudio::LINUX_OSS; } + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; } unsigned int getDeviceCount( void ); RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); void closeStream( void ); @@ -1030,8 +1131,8 @@ class RtApiDummy: public RtApi { public: - RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); } - RtAudio::Api getCurrentApi( void ) const { return RtAudio::RTAUDIO_DUMMY; } + RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); } + RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; } unsigned int getDeviceCount( void ) { return 0; } RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; } void closeStream( void ) {} diff --git a/source/modules/rtaudio/RtError.h b/source/modules/rtaudio/RtError.h deleted file mode 100644 index a64f43430..000000000 --- a/source/modules/rtaudio/RtError.h +++ /dev/null @@ -1,60 +0,0 @@ -/************************************************************************/ -/*! \class RtError - \brief Exception handling class for RtAudio & RtMidi. - - The RtError class is quite simple but it does allow errors to be - "caught" by RtError::Type. See the RtAudio and RtMidi - documentation to know which methods can throw an RtError. - -*/ -/************************************************************************/ - -#ifndef RTERROR_H -#define RTERROR_H - -#include -#include -#include - -class RtError : public std::exception -{ - public: - //! Defined RtError types. - enum Type { - WARNING, /*!< A non-critical error. */ - DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ - UNSPECIFIED, /*!< The default, unspecified error type. */ - NO_DEVICES_FOUND, /*!< No devices found on system. */ - INVALID_DEVICE, /*!< An invalid device ID was specified. */ - MEMORY_ERROR, /*!< An error occured during memory allocation. */ - INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ - INVALID_USE, /*!< The function was called incorrectly. */ - DRIVER_ERROR, /*!< A system driver error occured. */ - SYSTEM_ERROR, /*!< A system error occured. */ - THREAD_ERROR /*!< A thread error occured. */ - }; - - //! The constructor. - RtError( const std::string& message, Type type = RtError::UNSPECIFIED ) throw() : message_(message), type_(type) {} - - //! The destructor. - virtual ~RtError( void ) throw() {} - - //! Prints thrown error message to stderr. - virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; } - - //! Returns the thrown error message type. - virtual const Type& getType(void) const throw() { return type_; } - - //! Returns the thrown error message string. - virtual const std::string& getMessage(void) const throw() { return message_; } - - //! Returns the thrown error message as a c-style string. - virtual const char* what( void ) const throw() { return message_.c_str(); } - - protected: - std::string message_; - Type type_; -}; - -#endif diff --git a/source/utils/CarlaBase64Utils.hpp b/source/utils/CarlaBase64Utils.hpp index f4e05be8e..9f36c7ea6 100644 --- a/source/utils/CarlaBase64Utils.hpp +++ b/source/utils/CarlaBase64Utils.hpp @@ -37,7 +37,7 @@ static const char* const kBase64Chars = static inline uint8_t findBase64CharIndex(const char c) { - static const uint8_t kBase64CharsLen(std::strlen(kBase64Chars)); + static const uint8_t kBase64CharsLen(static_cast(std::strlen(kBase64Chars))); for (uint8_t i=0; i