X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.cpp;h=539cdc273a069c2424c8ea2030575bc4e75d1665;hb=b24942beab530945c8787b8dc27ebd0d1573e95e;hp=8a520c3ca3583738b94d09e3b0e36214287b7971;hpb=325fea749470f53e3d9dcd903d1c9b7c881a37fc;p=rtaudio-cdist.git diff --git a/RtAudio.cpp b/RtAudio.cpp index 8a520c3..539cdc2 100644 --- a/RtAudio.cpp +++ b/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) 2001-2011 Gary P. Scavone + Copyright (c) 2001-2016 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -38,13 +38,15 @@ */ /************************************************************************/ -// RtAudio: Version 4.0.9 +// RtAudio: Version 4.1.2 #include "RtAudio.h" #include #include #include #include +#include +#include // Static variable definitions. const unsigned int RtApi::MAX_SAMPLE_RATES = 14; @@ -53,12 +55,28 @@ 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) #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) -#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + + #include "tchar.h" + + static std::string convertCharPointerToStdString(const char *text) + { + return std::string(text); + } + + static std::string convertCharPointerToStdString(const wchar_t *text) + { + int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); + std::string s( length-1, '\0' ); + WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); + return s; + } + +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) // pthread API #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) @@ -75,6 +93,11 @@ const unsigned int RtApi::SAMPLE_RATES[] = { // // *************************************************** // +std::string RtAudio :: getVersion( void ) throw() +{ + return RTAUDIO_VERSION; +} + void RtAudio :: getCompiledApi( std::vector &apis ) throw() { apis.clear(); @@ -87,12 +110,18 @@ void RtAudio :: getCompiledApi( std::vector &apis ) throw() #if defined(__LINUX_ALSA__) apis.push_back( LINUX_ALSA ); #endif +#if defined(__LINUX_PULSE__) + apis.push_back( LINUX_PULSE ); +#endif #if defined(__LINUX_OSS__) apis.push_back( LINUX_OSS ); #endif #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 @@ -106,6 +135,10 @@ void RtAudio :: getCompiledApi( std::vector &apis ) throw() void RtAudio :: openRtApi( RtAudio::Api api ) { + if ( rtapi_ ) + delete rtapi_; + rtapi_ = 0; + #if defined(__UNIX_JACK__) if ( api == UNIX_JACK ) rtapi_ = new RtApiJack(); @@ -114,6 +147,10 @@ void RtAudio :: openRtApi( RtAudio::Api api ) if ( api == LINUX_ALSA ) rtapi_ = new RtApiAlsa(); #endif +#if defined(__LINUX_PULSE__) + if ( api == LINUX_PULSE ) + rtapi_ = new RtApiPulse(); +#endif #if defined(__LINUX_OSS__) if ( api == LINUX_OSS ) rtapi_ = new RtApiOss(); @@ -122,6 +159,10 @@ 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(); @@ -136,7 +177,7 @@ void RtAudio :: openRtApi( RtAudio::Api api ) #endif } -RtAudio :: RtAudio( RtAudio::Api api ) throw() +RtAudio :: RtAudio( RtAudio::Api api ) { rtapi_ = 0; @@ -156,7 +197,7 @@ RtAudio :: RtAudio( RtAudio::Api api ) throw() getCompiledApi( apis ); for ( unsigned int i=0; igetDeviceCount() ) break; + if ( rtapi_ && rtapi_->getDeviceCount() ) break; } if ( rtapi_ ) return; @@ -164,13 +205,15 @@ 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, @@ -178,11 +221,12 @@ void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, RtAudioFormat format, unsigned int sampleRate, unsigned int *bufferFrames, RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options ) + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) { return rtapi_->openStream( outputParameters, inputParameters, format, sampleRate, bufferFrames, callback, - userData, options ); + userData, options, errorCallback ); } // *************************************************** // @@ -201,6 +245,7 @@ RtApi :: RtApi() stream_.userBuffer[1] = 0; MUTEX_INITIALIZE( &stream_.mutex ); showWarnings_ = true; + firstErrorOccurred_ = false; } RtApi :: ~RtApi() @@ -213,31 +258,40 @@ void RtApi :: openStream( RtAudio::StreamParameters *oParams, RtAudioFormat format, unsigned int sampleRate, unsigned int *bufferFrames, RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options ) + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) { 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; } unsigned int nDevices = getDeviceCount(); @@ -246,7 +300,8 @@ 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; } } @@ -255,18 +310,21 @@ 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 ) { result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, sampleRate, format, bufferFrames, options ); - if ( result == false ) error( RtError::SYSTEM_ERROR ); + if ( result == false ) { + error( RtAudioError::SYSTEM_ERROR ); + return; + } } if ( iChannels > 0 ) { @@ -275,12 +333,14 @@ 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; } } stream_.callbackInfo.callback = (void *) callback; stream_.callbackInfo.userData = userData; + stream_.callbackInfo.errorCallback = (void *) errorCallback; if ( options ) options->numberOfBuffers = stream_.nBuffers; stream_.state = STREAM_STOPPED; @@ -304,10 +364,10 @@ void RtApi :: closeStream( void ) return; } -bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) +bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, + unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, + RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, + RtAudio::StreamOptions * /*options*/ ) { // MUST be implemented in subclasses! return FAILURE; @@ -349,7 +409,9 @@ double RtApi :: getStreamTime( void ) struct timeval then; struct timeval now; - if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + // If lastTickTimestamp is 0 it means we haven't had a "last tick" since + // we started the stream. + if ( stream_.state != STREAM_RUNNING || (stream_.lastTickTimestamp.tv_sec == 0 && stream_.lastTickTimestamp.tv_usec == 0) ) return stream_.streamTime; gettimeofday( &now, NULL ); @@ -362,6 +424,14 @@ double RtApi :: getStreamTime( void ) #endif } +void RtApi :: setStreamTime( double time ) +{ + verifyStream(); + + if ( time >= 0.0 ) + stream_.streamTime = time; +} + unsigned int RtApi :: getStreamSampleRate( void ) { verifyStream(); @@ -369,6 +439,14 @@ unsigned int RtApi :: getStreamSampleRate( void ) return stream_.sampleRate; } +void RtApi :: startStream( void ) +{ +#if defined( HAVE_GETTIMEOFDAY ) + stream_.lastTickTimestamp.tv_sec = 0; + stream_.lastTickTimestamp.tv_usec = 0; +#endif +} + // *************************************************** // // @@ -426,7 +504,7 @@ 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 } @@ -447,7 +525,7 @@ 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; } @@ -465,7 +543,7 @@ 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; } @@ -475,7 +553,7 @@ 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; } @@ -483,7 +561,7 @@ 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; } @@ -498,7 +576,7 @@ 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; } @@ -508,7 +586,7 @@ 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; } @@ -516,7 +594,7 @@ 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; } @@ -529,12 +607,14 @@ 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; } AudioDeviceID deviceList[ nDevices ]; @@ -546,7 +626,7 @@ 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; } @@ -561,14 +641,18 @@ 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 ); @@ -579,14 +663,18 @@ 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); @@ -601,7 +689,7 @@ 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; } @@ -609,7 +697,7 @@ 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; } @@ -618,7 +706,7 @@ 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; } @@ -634,7 +722,7 @@ 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; } @@ -642,7 +730,7 @@ 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; } @@ -651,7 +739,7 @@ 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; } @@ -676,7 +764,7 @@ 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; } @@ -686,26 +774,54 @@ 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 ) { + unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum; + info.sampleRates.push_back( tmpSr ); + + if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) ) + info.preferredSampleRate = tmpSr; + + } 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] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = 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; } @@ -723,13 +839,13 @@ RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) return info; } -OSStatus callbackHandler( AudioDeviceID inDevice, - const AudioTimeStamp* inNow, - const AudioBufferList* inInputData, - const AudioTimeStamp* inInputTime, - AudioBufferList* outOutputData, - const AudioTimeStamp* inOutputTime, - void* infoPointer ) +static OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* /*inNow*/, + const AudioBufferList* inInputData, + const AudioTimeStamp* /*inInputTime*/, + AudioBufferList* outOutputData, + const AudioTimeStamp* /*inOutputTime*/, + void* infoPointer ) { CallbackInfo *info = (CallbackInfo *) infoPointer; @@ -740,10 +856,10 @@ OSStatus callbackHandler( AudioDeviceID inDevice, return kAudioHardwareNoError; } -OSStatus xrunListener( AudioObjectID inDevice, - UInt32 nAddresses, - const AudioObjectPropertyAddress properties[], - void* handlePointer ) +static OSStatus xrunListener( AudioObjectID /*inDevice*/, + UInt32 nAddresses, + const AudioObjectPropertyAddress properties[], + void* handlePointer ) { CoreHandle *handle = (CoreHandle *) handlePointer; for ( UInt32 i=0; i > physicalFormats; - formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; physicalFormats.push_back( std::pair( 32, formatFlags ) ); formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; physicalFormats.push_back( std::pair( 32, formatFlags ) ); @@ -1132,7 +1247,7 @@ 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 ); } } @@ -1264,6 +1379,7 @@ 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; @@ -1287,6 +1403,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.deviceBuffer = 0; } + stream_.state = STREAM_CLOSED; return FAILURE; } @@ -1294,12 +1411,24 @@ void RtApiCore :: closeStream( void ) { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + } if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[0], callbackHandler ); #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) @@ -1311,6 +1440,18 @@ void RtApiCore :: closeStream( void ) } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + } if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[1], callbackHandler ); #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) @@ -1345,14 +1486,13 @@ void RtApiCore :: closeStream( void ) void RtApiCore :: startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiCore::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } - MUTEX_LOCK( &stream_.mutex ); - OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -1381,10 +1521,8 @@ void RtApiCore :: startStream( void ) stream_.state = STREAM_RUNNING; unlock: - MUTEX_UNLOCK( &stream_.mutex ); - if ( result == noErr ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiCore :: stopStream( void ) @@ -1392,14 +1530,7 @@ void RtApiCore :: stopStream( void ) verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::WARNING ); return; } @@ -1412,9 +1543,7 @@ void RtApiCore :: stopStream( void ) pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled } - MUTEX_UNLOCK( &stream_.mutex ); result = AudioDeviceStop( handle->id[0], callbackHandler ); - MUTEX_LOCK( &stream_.mutex ); if ( result != noErr ) { errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); @@ -1424,9 +1553,7 @@ void RtApiCore :: stopStream( void ) if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - MUTEX_UNLOCK( &stream_.mutex ); result = AudioDeviceStop( handle->id[1], callbackHandler ); - MUTEX_LOCK( &stream_.mutex ); if ( result != noErr ) { errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; errorText_ = errorStream_.str(); @@ -1437,10 +1564,8 @@ void RtApiCore :: stopStream( void ) stream_.state = STREAM_STOPPED; unlock: - MUTEX_UNLOCK( &stream_.mutex ); - if ( result == noErr ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiCore :: abortStream( void ) @@ -1448,7 +1573,7 @@ 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; } @@ -1458,14 +1583,28 @@ void RtApiCore :: abortStream( void ) stopStream(); } +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is better to handle it this way because the +// callbackEvent() function probably should return before the AudioDeviceStop() +// function is called. +static void *coreStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiCore *object = (RtApiCore *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, const AudioBufferList *inBufferList, const AudioBufferList *outBufferList ) { - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + 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; } @@ -1474,21 +1613,16 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; if ( handle->internalDrain == true ) - stopStream(); + pthread_create( &threadId, NULL, coreStopStream, info ); else // external call to stopStream() pthread_cond_signal( &handle->condition ); return SUCCESS; } - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return SUCCESS; - } - AudioDeviceID outputDevice = handle->id[0]; // Invoke user callback to get fresh output data UNLESS we are @@ -1507,15 +1641,18 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, handle->xrun[1] = false; } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; abortStream(); return SUCCESS; } - else if ( handle->drainCounter == 1 ) + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; handle->internalDrain = true; + } } if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { @@ -1615,11 +1752,12 @@ 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; @@ -1713,7 +1851,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, } unlock: - MUTEX_UNLOCK( &stream_.mutex ); + //MUTEX_UNLOCK( &stream_.mutex ); RtApi::tickStreamTime(); return SUCCESS; @@ -1814,8 +1952,7 @@ struct JackHandle { :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } }; -ThreadHandle threadId; -void jackSilentError( const char * ) {}; +static void jackSilentError( const char * ) {}; RtApiJack :: RtApiJack() { @@ -1874,7 +2011,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) jack_client_t *client = jack_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; } @@ -1901,13 +2038,17 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) } if ( device >= nDevices ) { + jack_client_close( client ); errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); + error( RtAudioError::INVALID_USE ); + return info; } // Get the current jack server sample rate. info.sampleRates.clear(); - info.sampleRates.push_back( jack_get_sample_rate( client ) ); + + info.preferredSampleRate = jack_get_sample_rate( client ); + info.sampleRates.push_back( info.preferredSampleRate ); // Count the available ports containing the client name as device // channels. Jack "input ports" equal RtAudio output channels. @@ -1931,7 +2072,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) if ( info.outputChannels == 0 && info.inputChannels == 0 ) { jack_client_close(client); errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return info; } @@ -1953,7 +2094,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) return info; } -int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) { CallbackInfo *info = (CallbackInfo *) infoPointer; @@ -1967,7 +2108,7 @@ int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) // server signals that it is shutting down. It is necessary to handle // it this way because the jackShutdown() function must return before // the jack_deactivate() function (in closeStream()) will return. -extern "C" void *jackCloseStream( void *ptr ) +static void *jackCloseStream( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiJack *object = (RtApiJack *) info->object; @@ -1976,7 +2117,7 @@ extern "C" void *jackCloseStream( void *ptr ) pthread_exit( NULL ); } -void jackShutdown( void *infoPointer ) +static void jackShutdown( void *infoPointer ) { CallbackInfo *info = (CallbackInfo *) infoPointer; RtApiJack *object = (RtApiJack *) info->object; @@ -1988,11 +2129,12 @@ void jackShutdown( void *infoPointer ) // other problem occurred and we should close the stream. if ( object->isStreamRunning() == false ) return; + 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; } -int jackXrun( void *infoPointer ) +static int jackXrun( void *infoPointer ) { JackHandle *handle = (JackHandle *) infoPointer; @@ -2020,7 +2162,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne client = jack_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; } } @@ -2086,8 +2228,17 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Get the latency of the JACK port. ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports[ firstChannel ] ) - stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + if ( ports[ firstChannel ] ) { + // Added by Ge Wang + jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); + // the range (usually the min and max are equal) + jack_latency_range_t latrange; latrange.min = latrange.max = 0; + // get the latency range + jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); + // be optimistic, use the min! + stream_.latency[mode] = latrange.min; + //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + } free( ports ); // The jack server always uses 32-bit floating-point data. @@ -2248,7 +2399,7 @@ void RtApiJack :: closeStream( void ) { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiJack::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -2288,14 +2439,13 @@ void RtApiJack :: closeStream( void ) void RtApiJack :: startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiJack::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } - MUTEX_LOCK(&stream_.mutex); - JackHandle *handle = (JackHandle *) stream_.apiHandle; int result = jack_activate( handle->client ); if ( result ) { @@ -2357,10 +2507,8 @@ void RtApiJack :: startStream( void ) stream_.state = STREAM_RUNNING; unlock: - MUTEX_UNLOCK(&stream_.mutex); - if ( result == 0 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiJack :: stopStream( void ) @@ -2368,14 +2516,7 @@ void RtApiJack :: stopStream( void ) verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::WARNING ); return; } @@ -2390,8 +2531,6 @@ void RtApiJack :: stopStream( void ) jack_deactivate( handle->client ); stream_.state = STREAM_STOPPED; - - MUTEX_UNLOCK( &stream_.mutex ); } void RtApiJack :: abortStream( void ) @@ -2399,7 +2538,7 @@ 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; } @@ -2414,27 +2553,26 @@ void RtApiJack :: abortStream( void ) // aborted. It is necessary to handle it this way because the // callbackEvent() function must return before the jack_deactivate() // function will return. -extern "C" void *jackStopStream( void *ptr ) +static void *jackStopStream( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiJack *object = (RtApiJack *) info->object; object->stopStream(); - pthread_exit( NULL ); } bool RtApiJack :: callbackEvent( unsigned long nframes ) { - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + 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; } @@ -2443,6 +2581,9 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; if ( handle->internalDrain == true ) pthread_create( &threadId, NULL, jackStopStream, info ); else @@ -2450,14 +2591,6 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) return SUCCESS; } - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return SUCCESS; - } - // Invoke user callback first, to get fresh output data. if ( handle->drainCounter == 0 ) { RtAudioCallback callback = (RtAudioCallback) info->callback; @@ -2471,16 +2604,19 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) status |= RTAUDIO_INPUT_OVERFLOW; handle->xrun[1] = false; } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; ThreadHandle id; pthread_create( &id, NULL, jackStopStream, info ); return SUCCESS; } - else if ( handle->drainCounter == 1 ) + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; handle->internalDrain = true; + } } jack_default_audio_sample_t *jackbuffer; @@ -2510,11 +2646,12 @@ 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 ) { @@ -2535,8 +2672,6 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) } unlock: - MUTEX_UNLOCK(&stream_.mutex); - RtApi::tickStreamTime(); return SUCCESS; } @@ -2567,11 +2702,11 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) #include "asiodrivers.h" #include -AsioDrivers drivers; -ASIOCallbacks asioCallbacks; -ASIODriverInfo driverInfo; -CallbackInfo *asioCallbackInfo; -bool asioXRun; +static AsioDrivers drivers; +static ASIOCallbacks asioCallbacks; +static ASIODriverInfo driverInfo; +static CallbackInfo *asioCallbackInfo; +static bool asioXRun; struct AsioHandle { int drainCounter; // Tracks callback counts when draining @@ -2585,8 +2720,8 @@ struct AsioHandle { // Function declarations (definitions at end of section) static const char* getAsioErrorString( ASIOError result ); -void sampleRateChanged( ASIOSampleRate sRate ); -long asioMessages( long selector, long value, void* message, double* opt ); +static void sampleRateChanged( ASIOSampleRate sRate ); +static long asioMessages( long selector, long value, void* message, double* opt ); RtApiAsio :: RtApiAsio() { @@ -2597,7 +2732,7 @@ 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; @@ -2628,19 +2763,21 @@ 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; } // If a stream is already open, we cannot probe other devices. Thus, use the saved results. 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 ]; @@ -2651,7 +2788,7 @@ 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; } @@ -2660,7 +2797,7 @@ 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; } @@ -2668,7 +2805,7 @@ 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; } @@ -2679,7 +2816,7 @@ 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; } @@ -2692,8 +2829,12 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) info.sampleRates.clear(); for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } } // Determine supported data types ... just check first channel and assume rest are the same. @@ -2706,7 +2847,7 @@ 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; } @@ -2719,6 +2860,8 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) info.nativeFormats |= RTAUDIO_FLOAT32; else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) info.nativeFormats |= RTAUDIO_FLOAT64; + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) + info.nativeFormats |= RTAUDIO_SINT24; if ( info.outputChannels > 0 ) if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; @@ -2730,7 +2873,7 @@ RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) return info; } -void bufferSwitch( long index, ASIOBool processNow ) +static void bufferSwitch( long index, ASIOBool /*processNow*/ ) { RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; object->callbackEvent( index ); @@ -2750,9 +2893,12 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne unsigned int firstChannel, unsigned int sampleRate, RtAudioFormat format, unsigned int *bufferSize, RtAudio::StreamOptions *options ) -{ +{//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + + bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT; + // For ASIO, a duplex stream MUST use the same driver. - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) { + if ( isDuplexInput && stream_.device[0] != device ) { errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!"; return FAILURE; } @@ -2766,7 +2912,7 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } // Only load the driver once for duplex stream. - if ( mode != INPUT || stream_.mode != OUTPUT ) { + if ( !isDuplexInput ) { // The getDeviceInfo() function will not work when a stream is open // because ASIO does not allow multiple devices to run at the same // time. Thus, we'll probe the system before opening a stream and @@ -2787,22 +2933,26 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } } + // keep them before any "goto error", they are used for error cleanup + goto device boundary checks + bool buffersAllocated = false; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + unsigned int nChannels; + + // Check the device channel count. long inputChannels, outputChannels; result = ASIOGetChannels( &inputChannels, &outputChannels ); if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } stream_.nDeviceChannels[mode] = channels; stream_.nUserChannels[mode] = channels; @@ -2811,30 +2961,27 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Verify the sample rate is supported. result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } // Get the current sample rate ASIOSampleRate currentRate; result = ASIOGetSampleRate( ¤tRate ); if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } // Set the sample rate only if necessary if ( currentRate != sampleRate ) { result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } } @@ -2845,10 +2992,9 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else channelInfo.isInput = true; result = ASIOGetChannelInfo( &channelInfo ); if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } // Assuming WINDOWS host is always little-endian. @@ -2871,12 +3017,15 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; } + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; + } if ( stream_.deviceFormat[mode] == 0 ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } // Set the buffer size. For a duplex stream, this will end up @@ -2885,49 +3034,63 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne long minSize, maxSize, preferSize, granularity; result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; errorText_ = errorStream_.str(); - return FAILURE; + goto error; } - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else if ( granularity == -1 ) { - // Make sure bufferSize is a power of two. - int log2_of_min_size = 0; - int log2_of_max_size = 0; + if ( isDuplexInput ) { + // When this is the duplex input (output was opened before), then we have to use the same + // buffersize as the output, because it might use the preferred buffer size, which most + // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, + // So instead of throwing an error, make them equal. The caller uses the reference + // to the "bufferSize" param as usual to set up processing buffers. - for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { - if ( minSize & ((long)1 << i) ) log2_of_min_size = i; - if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; - } + *bufferSize = stream_.bufferSize; + + } else { + if ( *bufferSize == 0 ) *bufferSize = preferSize; + else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } - long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); - int min_delta_num = log2_of_min_size; + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; - for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { - long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); - if (current_delta < min_delta) { - min_delta = current_delta; - min_delta_num = i; + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } } - } - *bufferSize = ( (unsigned int)1 << min_delta_num ); - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - } - else if ( granularity != 0 ) { - // Set to an even multiple of granularity, rounding up. - *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + *bufferSize = ( (unsigned int)1 << min_delta_num ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + } + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + } } - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { - drivers.removeCurrentDriver(); + /* + // we don't use it anymore, see above! + // Just left it here for the case... + if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; - return FAILURE; + goto error; } + */ stream_.bufferSize = *bufferSize; stream_.nBuffers = 2; @@ -2939,16 +3102,13 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.deviceInterleaved[mode] = false; // Allocate, if necessary, our AsioHandle structure for the stream. - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; if ( handle == 0 ) { try { handle = new AsioHandle; } catch ( std::bad_alloc& ) { - //if ( handle == NULL ) { - drivers.removeCurrentDriver(); errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - return FAILURE; + goto error; } handle->bufferInfos = 0; @@ -2963,15 +3123,14 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Create the ASIO internal buffers. Since RtAudio sets up input // and output separately, we'll have to dispose of previously // created output buffers for a duplex stream. - long inputLatency, outputLatency; if ( mode == INPUT && stream_.mode == OUTPUT ) { ASIODisposeBuffers(); if ( handle->bufferInfos ) free( handle->bufferInfos ); } // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - bool buffersAllocated = false; - unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + unsigned int i; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); if ( handle->bufferInfos == NULL ) { errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; @@ -2992,18 +3151,37 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne infos->buffers[0] = infos->buffers[1] = 0; } + // prepare for callbacks + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.mode = isDuplexInput ? DUPLEX : mode; + + // store this class instance before registering callbacks, that are going to use it + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + // Set up the ASIO callback structure and create the ASIO data buffers. asioCallbacks.bufferSwitch = &bufferSwitch; asioCallbacks.sampleRateDidChange = &sampleRateChanged; asioCallbacks.asioMessage = &asioMessages; asioCallbacks.bufferSwitchTimeInfo = NULL; result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + if ( result != ASE_OK ) { + // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges + // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver + // in that case, let's be naïve and try that instead + *bufferSize = preferSize; + stream_.bufferSize = *bufferSize; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + } + if ( result != ASE_OK ) { errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; errorText_ = errorStream_.str(); goto error; } - buffersAllocated = true; + buffersAllocated = true; + stream_.state = STREAM_STOPPED; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; @@ -3026,11 +3204,9 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } + if ( isDuplexInput && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; } if ( makeBuffer ) { @@ -3044,23 +3220,13 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } } - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - // Determine device latencies + long inputLatency, outputLatency; result = ASIOGetLatencies( &inputLatency, &outputLatency ); 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; @@ -3075,38 +3241,44 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne return SUCCESS; error: - if ( buffersAllocated ) - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); + if ( !isDuplexInput ) { + // the cleanup for error in the duplex input, is done by RtApi::openStream + // So we clean up for single channel only - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } + if ( buffersAllocated ) + ASIODisposeBuffers(); - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + drivers.removeCurrentDriver(); - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + + delete handle; + stream_.apiHandle = 0; + } + + + if ( stream_.userBuffer[mode] ) { + free( stream_.userBuffer[mode] ); + stream_.userBuffer[mode] = 0; + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } } return FAILURE; -} +}//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// void RtApiAsio :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -3147,14 +3319,13 @@ bool stopThreadCalled = false; void RtApiAsio :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAsio::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } - //MUTEX_LOCK( &stream_.mutex ); - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; ASIOError result = ASIOStart(); if ( result != ASE_OK ) { @@ -3170,12 +3341,10 @@ void RtApiAsio :: startStream() asioXRun = false; unlock: - //MUTEX_UNLOCK( &stream_.mutex ); - stopThreadCalled = false; if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiAsio :: stopStream() @@ -3183,27 +3352,15 @@ void RtApiAsio :: stopStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - /* - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::WARNING ); return; } - */ AsioHandle *handle = (AsioHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter == 0 ) { handle->drainCounter = 2; - // MUTEX_UNLOCK( &stream_.mutex ); WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - //ResetEvent( handle->condition ); - // MUTEX_LOCK( &stream_.mutex ); } } @@ -3215,10 +3372,8 @@ void RtApiAsio :: stopStream() errorText_ = errorStream_.str(); } - // MUTEX_UNLOCK( &stream_.mutex ); - if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiAsio :: abortStream() @@ -3226,7 +3381,7 @@ void RtApiAsio :: abortStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -3244,24 +3399,22 @@ void RtApiAsio :: abortStream() // aborted. It is necessary to handle it this way because the // callbackEvent() function must return before the ASIOStop() // function will return. -extern "C" unsigned __stdcall asioStopStream( void *ptr ) +static unsigned __stdcall asioStopStream( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiAsio *object = (RtApiAsio *) info->object; object->stopStream(); - _endthreadex( 0 ); return 0; } bool RtApiAsio :: callbackEvent( long bufferIndex ) { - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stopThreadCalled ) return SUCCESS; + 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; } @@ -3270,22 +3423,18 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) // Check if we were draining the stream and signal if finished. if ( handle->drainCounter > 3 ) { + + stream_.state = STREAM_STOPPING; if ( handle->internalDrain == false ) SetEvent( handle->condition ); else { // spawn a thread to stop the stream unsigned threadId; - stopThreadCalled = true; stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, &stream_.callbackInfo, 0, &threadId ); } return SUCCESS; } - /*MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; */ - // Invoke user callback to get fresh output data UNLESS we are // draining stream. if ( handle->drainCounter == 0 ) { @@ -3300,19 +3449,20 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) status |= RTAUDIO_INPUT_OVERFLOW; asioXRun = false; } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - // MUTEX_UNLOCK( &stream_.mutex ); - // abortStream(); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; unsigned threadId; - stopThreadCalled = true; stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, &stream_.callbackInfo, 0, &threadId ); return SUCCESS; } - else if ( handle->drainCounter == 1 ) + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; handle->internalDrain = true; + } } unsigned int nChannels, bufferBytes, i, j; @@ -3358,11 +3508,12 @@ 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 ) { @@ -3408,13 +3559,11 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) // drivers apparently do not function correctly without it. ASIOOutputReady(); - // MUTEX_UNLOCK( &stream_.mutex ); - RtApi::tickStreamTime(); return SUCCESS; } -void sampleRateChanged( ASIOSampleRate sRate ) +static void sampleRateChanged( ASIOSampleRate sRate ) { // The ASIO documentation says that this usually only happens during // external sync. Audio processing is not stopped by the driver, @@ -3426,7 +3575,7 @@ void sampleRateChanged( ASIOSampleRate sRate ) try { object->stopStream(); } - catch ( RtError &exception ) { + catch ( RtAudioError &exception ) { std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; return; } @@ -3434,7 +3583,7 @@ void sampleRateChanged( ASIOSampleRate sRate ) std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; } -long asioMessages( long selector, long value, void* message, double* opt ) +static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) { long ret = 0; @@ -3512,7 +3661,7 @@ static const char* getAsioErrorString( ASIOError result ) const char*message; }; - static Messages m[] = + static const Messages m[] = { { ASE_NotPresent, "Hardware input or output is not present or available." }, { ASE_HWMalfunction, "Hardware is malfunctioning." }, @@ -3528,2867 +3677,4947 @@ 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() { + free( 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 ) { + free( 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; + } -extern "C" 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 + } -std::vector< DsDevice > dsDevices; + // copy buffer from external to internal + int fromZeroSize = inIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromInSize = bufferSize - fromZeroSize; -RtApiDs :: RtApiDs() + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) ); + memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) ); + memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) ); + memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) ); + memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) ); + memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) ); + memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) ); + break; + } + + // update "in" index + inIndex_ += bufferSize; + inIndex_ %= bufferSize_; + + return true; + } + + // attempt to pull a buffer from the ring buffer from the current "out" index + bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relInIndex = inIndex_; + unsigned int outIndexEnd = outIndex_ + bufferSize; + if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) { + relInIndex += bufferSize_; + } + + // "out" index can begin at and end on the "in" index + if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) { + return false; // not enough space between "out" index and "in" index + } + + // copy buffer from internal to external + int fromZeroSize = outIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromOutSize = bufferSize - fromZeroSize; + + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) ); + memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) ); + memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) ); + memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) ); + memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) ); + memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) ); + memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) ); + break; + } + + // update "out" index + outIndex_ += bufferSize; + outIndex_ %= bufferSize_; + + return true; + } + +private: + char* buffer_; + unsigned int bufferSize_; + unsigned int inIndex_; + unsigned int outIndex_; +}; + +//----------------------------------------------------------------------------- + +// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate +// 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 ) roundf( 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. - isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput ); - 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::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 ); - } + // 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 ); + 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 << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto probeInput; + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; + goto Exit; } - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto probeInput; + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; + errorType = RtAudioError::INVALID_USE; + goto Exit; } - // Get output channel information. - info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + // 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 sample rate information. - info.sampleRates.clear(); - for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && - SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); + // 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; + } } - // Get format information. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; - if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; + goto Exit; + } + PropVariantInit( &defaultDeviceNameProp ); - output->Release(); + hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; + goto Exit; + } - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; + defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); - if ( dsDevices[ device ].validId[1] == false ) { - info.name = dsDevices[ device ].name; - info.probed = true; - return info; + // name + hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; + goto Exit; } - probeInput: + PropVariantInit( &deviceNameProp ); - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; + hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; + goto Exit; } - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; + info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); + + // is default + if ( isCaptureDevice ) { + info.isDefaultInput = info.name == defaultDeviceName; + info.isDefaultOutput = false; + } + else { + info.isDefaultInput = false; + info.isDefaultOutput = info.name == defaultDeviceName; } - // Get input channel information. - info.inputChannels = inCaps.dwChannels; + // 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; + } - // Get sample rate and format information. - std::vector rates; - if ( inCaps.dwChannels >= 2 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + hr = audioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + goto Exit; + } - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); + 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] ); + } + info.preferredSampleRate = deviceFormat->nSamplesPerSec; + + // 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 ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); + else if ( deviceFormat->wBitsPerSample == 64 ) { + info.nativeFormats |= RTAUDIO_FLOAT64; } } - else if ( inCaps.dwChannels == 1 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); + 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 ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); + 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; } } - else info.inputChannels = 0; // technically, this would be an error - input->Release(); + // probed + info.probed = true; - if ( info.inputChannels == 0 ) return info; +Exit: + // release all references + PropVariantClear( &deviceNameProp ); + PropVariantClear( &defaultDeviceNameProp ); - // Copy the supported rates to the info structure but avoid duplication. - bool found; - for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + return 0; +} - if ( device == 0 ) info.isDefaultInput = true; +//----------------------------------------------------------------------------- - // Copy name and return. - info.name = dsDevices[ device ].name; - info.probed = true; - return info; +unsigned int RtApiWasapi::getDefaultInputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultInput ) { + return i; + } + } + + return 0; } -bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) +//----------------------------------------------------------------------------- + +void RtApiWasapi::closeStream( void ) { - if ( channels + firstChannel > 2 ) { - errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; - return FAILURE; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiWasapi::closeStream: No open stream to close."; + error( RtAudioError::WARNING ); + return; } - unsigned int 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!"; - return FAILURE; + 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 ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; - return FAILURE; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - if ( mode == OUTPUT ) { - if ( dsDevices[ device ].validId[0] == false ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // update stream state + stream_.state = STREAM_CLOSED; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::startStream( void ) +{ + verifyStream(); + RtApi::startStream(); + + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiWasapi::startStream: The stream is already running."; + error( RtAudioError::WARNING ); + return; } - else { // mode == INPUT - if ( dsDevices[ device ].validId[1] == false ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + + // 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 ); } +} - // According to a note in PortAudio, using GetDesktopWindow() - // instead of GetForegroundWindow() is supposed to avoid problems - // that occur when the application's window is not the foreground - // window. Also, if the application window closes before the - // DirectSound buffer, DirectSound can crash. In the past, I had - // problems when using GetDesktopWindow() but it seems fine now - // (January 2010). I'll leave it commented here. - // HWND hWnd = GetForegroundWindow(); - HWND hWnd = GetDesktopWindow(); +//----------------------------------------------------------------------------- - // Check the numberOfBuffers parameter and limit the lowest value to - // two. This is a judgement call and a value of two is probably too - // low for capture, but it should work for playback. - int nBuffers = 0; - if ( options ) nBuffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; - if ( nBuffers < 2 ) nBuffers = 3; +void RtApiWasapi::stopStream( void ) +{ + verifyStream(); - // Check the lower range of the user-specified buffer size and set - // (arbitrarily) to a lower bound of 32. - if ( *bufferSize < 32 ) *bufferSize = 32; + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::stopStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } - // Create the wave format structure. The data format setting will - // be determined later. - WAVEFORMATEX waveFormat; - ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); - waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels + firstChannel; - waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; - // Determine the device buffer size. By default, we'll use the value - // defined above (32K), but we will grow it to make allowances for - // very large software buffer sizes. - DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;; - DWORD dsPointerLeadTime = 0; + // wait until stream thread is stopped + while( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } - void *ohandle = 0, *bhandle = 0; - HRESULT result; - if ( mode == OUTPUT ) { + // Wait for the last buffer to play before stopping. + Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); - LPDIRECTSOUND output; - result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + // 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; } + } - DSCAPS outCaps; - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + // 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; } + } - // Check channel information. - if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // 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; + } - // Check format information. Use 16-bit format unless not - // supported or user requests 8-bit. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && - !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - stream_.userFormat = format; + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; +//----------------------------------------------------------------------------- - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > dsBufferSize ) - dsBufferSize *= 2; +void RtApiWasapi::abortStream( void ) +{ + verifyStream(); - // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. - // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); - // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. - result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::abortStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } - // Even though we will write to the secondary buffer, we need to - // access the primary buffer to set the correct output format - // (since the default is 8-bit, 22 kHz!). Setup the DS primary - // buffer description. - DSBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; - // Obtain the primary buffer - LPDIRECTSOUNDBUFFER buffer; - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // wait until stream thread is stopped + while ( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } - // Set the primary DS buffer sound format. - result = buffer->SetFormat( &waveFormat ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + // 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; } + } - // Setup the secondary DS buffer description. - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = dsBufferSize; - bufferDescription.lpwfxFormat = &waveFormat; - - // Try to create the secondary DS buffer. If that doesn't work, - // try to use software mixing. Otherwise, there's a problem. - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // 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; } + } - // Get the buffer size ... might be different from what we specified. - DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof( DSBCAPS ); - result = buffer->GetCaps( &dsbcaps ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // 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; + } - dsBufferSize = dsbcaps.dwBufferBytes; + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} - // Lock the DS buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } +//----------------------------------------------------------------------------- - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); +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; - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // create API Handle if not already created + if ( !stream_.apiHandle ) + stream_.apiHandle = ( void* ) new WasapiHandle(); - ohandle = (void *) output; - bhandle = (void *) buffer; + // 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; } - if ( mode == INPUT ) { + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; + goto Exit; + } - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // 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; + } - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; + goto Exit; + } - // Check channel information. - if ( inCaps.dwChannels < channels + firstChannel ) { - errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; - return FAILURE; - } + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; + goto Exit; + } - // Check format information. Use 16-bit format unless user - // requests 8-bit. - DWORD deviceFormats; - if ( channels + firstChannel == 2 ) { - deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - else { // channel == 1 - deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } + // 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; } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > dsBufferSize ) - dsBufferSize *= 2; + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; - // Setup the secondary DS buffer description. - DSCBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); - bufferDescription.dwFlags = 0; - bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = dsBufferSize; - bufferDescription.lpwfxFormat = &waveFormat; + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; + goto Exit; + } - // Create the capture buffer. - LPDIRECTSOUNDCAPTUREBUFFER buffer; - result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; } - // Get the buffer size ... might be different from what we specified. - DSCBCAPS dscbcaps; - dscbcaps.dwSize = sizeof( DSCBCAPS ); - result = buffer->GetCaps( &dscbcaps ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; } - dsBufferSize = dscbcaps.dwBufferBytes; + 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; + } - // NOTE: We could have a problem here if this is a duplex stream - // and the play and capture hardware buffer sizes are different - // (I'm actually not sure if that is a problem or not). - // Currently, we are not verifying that. + // retrieve renderAudioClient from devicePtr + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; - // Lock the capture buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; } - // Zero the buffer - ZeroMemory( audioPtr, dataLen ); + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &renderAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; + } - // Unlock the buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + hr = renderAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; } - ohandle = (void *) input; - bhandle = (void *) buffer; + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); } - // Set various stream parameters - DsHandle *handle = 0; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.nUserChannels[mode] = channels; + // 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; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - // Set flag for buffer conversion + // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; - if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + if ( stream_.userFormat != stream_.deviceFormat[mode] || + stream_.nUserChannels != stream_.nDeviceChannels ) stream_.doConvertBuffer[mode] = true; - if (stream_.userFormat != stream_.deviceFormat[mode]) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) + 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 - long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; - goto error; + 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 ( stream_.doConvertBuffer[mode] ) { + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) + stream_.callbackInfo.priority = 15; + else + stream_.callbackInfo.priority = 0; - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; - } - } + ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback + ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } + methodResult = SUCCESS; - // Allocate our DsHandle structures for the stream. - if ( stream_.apiHandle == 0 ) { - try { - handle = new DsHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; - } +Exit: + //clean up + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + CoTaskMemFree( deviceFormat ); - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - else - handle = (DsHandle *) stream_.apiHandle; - handle->id[mode] = ohandle; - handle->buffer[mode] = bhandle; - handle->dsBufferSize[mode] = dsBufferSize; - handle->dsPointerLeadTime[mode] = dsPointerLeadTime; + // if method failed, close the stream + if ( methodResult == FAILURE ) + closeStream(); - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - stream_.nBuffers = nBuffers; - stream_.sampleRate = sampleRate; + if ( !errorText_.empty() ) + error( errorType ); + return methodResult; +} - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); +//============================================================================= - // Setup the callback thread. - if ( stream_.callbackInfo.isRunning == false ) { - unsigned threadId; - stream_.callbackInfo.isRunning = true; - stream_.callbackInfo.object = (void *) this; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, - &stream_.callbackInfo, 0, &threadId ); - if ( stream_.callbackInfo.thread == 0 ) { - errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; - goto error; - } +DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); - // Boost DS thread priority - SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); - } - return SUCCESS; + return 0; +} - error: - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) buffer->Release(); - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) buffer->Release(); - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } +DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + return 0; +} - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } +DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); - return FAILURE; + return 0; } -void RtApiDs :: closeStream() +//----------------------------------------------------------------------------- + +void RtApiWasapi::wasapiThread() { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } + // as this is a new thread, we must CoInitialize it + CoInitialize( NULL ); - // Stop the callback thread. - stream_.callbackInfo.isRunning = false; - WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + HRESULT hr; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); + 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; } - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); + + hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), + ( void** ) &captureClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; + goto Exit; } - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + // 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; + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + hr = captureAudioClient->SetEventHandle( captureEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + goto Exit; + } - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} + ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; + } -void RtApiDs :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiDs::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } + unsigned int inBufferSize = 0; + hr = captureAudioClient->GetBufferSize( &inBufferSize ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; + goto Exit; + } - //MUTEX_LOCK( &stream_.mutex ); - - DsHandle *handle = (DsHandle *) stream_.apiHandle; + // scale outBufferSize according to stream->user sample rate ratio + unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT]; + inBufferSize *= stream_.nDeviceChannels[INPUT]; - // Increase scheduler frequency on lesser windows (a side-effect of - // increasing timer accuracy). On greater windows (Win2K or later), - // this is already in effect. - timeBeginPeriod( 1 ); + // set captureBuffer size + captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); - buffersRolling = false; - duplexPrerollBytes = 0; + // reset the capture stream + hr = captureAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; + goto Exit; + } - if ( stream_.mode == DUPLEX ) { - // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. - duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); + // start the capture stream + hr = captureAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream."; + goto Exit; + } } - HRESULT result = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + // 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; } - } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - result = buffer->Start( DSCBSTART_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - } + // initialize render stream according to desire buffer size + float desiredBufferSize = stream_.bufferSize * renderSrRatio; + REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec ); - handle->drainCounter = 0; - handle->internalDrain = false; - ResetEvent( handle->condition ); - stream_.state = STREAM_RUNNING; + 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; + } - unlock: - // MUTEX_UNLOCK( &stream_.mutex ); + hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), + ( void** ) &renderClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; + goto Exit; + } - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); -} + // 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; + } -void RtApiDs :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + hr = renderAudioClient->SetEventHandle( renderEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle."; + goto Exit; + } - /* - MUTEX_LOCK( &stream_.mutex ); + ( ( 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; + } - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - */ + // 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; + } - HRESULT result = 0; - LPVOID audioPtr; - DWORD dataLen; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - // MUTEX_UNLOCK( &stream_.mutex ); - WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - //ResetEvent( handle->condition ); - // MUTEX_LOCK( &stream_.mutex ); + 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; + } + } } - stream_.state = STREAM_STOPPED; + // 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 - // Stop the buffer and clear memory - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + if ( renderAudioClient && callbackPulled ) { + if ( stream_.doConvertBuffer[OUTPUT] ) { + // Convert callback buffer to stream format + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + } - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + // 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; } - // If we start playing again, we must begin at beginning of buffer. - handle->bufferPointer[0] = 0; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - audioPtr = NULL; - dataLen = 0; + // Stream Capture + // ============== + // 1. Get capture buffer from stream + // 2. Push capture buffer into inputBuffer + // 3. If 2. was successful: Release capture buffer - stream_.state = STREAM_STOPPED; + if ( captureAudioClient ) { + // if the callback input buffer was not pulled from captureBuffer, wait for next capture event + if ( !callbackPulled ) { + WaitForSingleObject( captureEvent, INFINITE ); + } - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + // 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; + } - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + 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; + } + } } - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + // 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 - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + if ( renderAudioClient ) { + // if the callback output buffer was not pushed to renderBuffer, wait for next render event + if ( callbackPulled && !callbackPushed ) { + WaitForSingleObject( renderEvent, INFINITE ); + } - // If we start recording again, we must begin at beginning of buffer. - handle->bufferPointer[1] = 0; - } + // Get render buffer from stream + hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; + goto Exit; + } - unlock: - timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - // MUTEX_UNLOCK( &stream_.mutex ); + hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; + goto Exit; + } - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); -} + bufferFrameCount -= numFramesPadding; -void RtApiDs :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + if ( bufferFrameCount != 0 ) { + hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; + goto Exit; + } - DsHandle *handle = (DsHandle *) stream_.apiHandle; - handle->drainCounter = 2; + // 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; + } + } + } - stopStream(); -} + // if the callback buffer was pushed renderBuffer reset callbackPulled flag + if ( callbackPushed ) { + callbackPulled = false; + // tick stream time + RtApi::tickStreamTime(); + } -void RtApiDs :: callbackEvent() -{ - if ( stream_.state == STREAM_STOPPED ) { - Sleep( 50 ); // sleep 50 milliseconds - return; } - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } +Exit: + // clean up + CoTaskMemFree( captureFormat ); + CoTaskMemFree( renderFormat ); - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - DsHandle *handle = (DsHandle *) stream_.apiHandle; + free ( convBuffer ); - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > stream_.nBuffers + 2 ) { - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return; - } + CoUninitialize(); - /* - MUTEX_LOCK( &stream_.mutex ); + // update stream state + stream_.state = STREAM_STOPPED; - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); + if ( errorText_.empty() ) return; - } - */ + else + error( errorType ); +} - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - // MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } +//******************** End of __WINDOWS_WASAPI__ *********************// +#endif - HRESULT result; - DWORD currentWritePointer, safeWritePointer; - DWORD currentReadPointer, safeReadPointer; - UINT nextWritePointer; - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; +#if defined(__WINDOWS_DS__) // Windows DirectSound API - char *buffer; - long bufferBytes; +// 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 ( buffersRolling == false ) { - if ( stream_.mode == DUPLEX ) { - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); +#include +#include +#include - // It takes a while for the devices to get rolling. As a result, - // there's no guarantee that the capture and write device pointers - // will move in lockstep. Wait here for both devices to start - // rolling, and then set our buffer pointers accordingly. - // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 - // bytes later than the write buffer. +#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 - // Stub: a serious risk of having a pre-emptive scheduling round - // take place between the two GetCurrentPosition calls... but I'm - // really not sure how to solve the problem. Temporarily boost to - // Realtime priority, maybe; but I'm not sure what priority the - // DirectSound service threads run at. We *should* be roughly - // within a ms or so of correct. +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; +#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 - DWORD startSafeWritePointer, startSafeReadPointer; +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; +} - result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - 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 ); - } - while ( true ) { - result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - 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 ); - } - if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; - Sleep( 1 ); - } +// 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; - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + 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; } +}; - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - handle->bufferPointer[1] = safeReadPointer; - } - else if ( stream_.mode == OUTPUT ) { +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); - // Set the proper nextWritePosition after initial startup. - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - } +static const char* getErrorString( int code ); - buffersRolling = true; - } +static unsigned __stdcall callbackHandler( void *ptr ); - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; +struct DsDevice { + LPGUID id[2]; + bool validId[2]; + bool found; + std::string name; - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - memset( stream_.userBuffer[0], 0, bufferBytes ); - } + DsDevice() + : found(false) { validId[0] = false; validId[1] = false; } +}; - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; - bufferBytes *= formatBytes( stream_.deviceFormat[0] ); - } - else { - buffer = stream_.userBuffer[0]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - } +struct DsProbeData { + bool isInput; + std::vector* dsDevices; +}; - // No byte swapping necessary in DirectSound implementation. +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; +} - // Ahhh ... windoze. 16-bit data is signed but 8-bit data is - // unsigned. So, we need to convert our signed 8-bit data here to - // unsigned. - if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) - for ( int i=0; idsBufferSize[0]; - nextWritePointer = handle->bufferPointer[0]; +// The DirectSound default output is always the first device. +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + return 0; +} - DWORD endWrite, leadPointer; - while ( true ) { - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } +// The DirectSound default input is always the first input device, +// which is the first capture device enumerated. +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + return 0; +} - // We will copy our output buffer into the region between - // safeWritePointer and leadPointer. If leadPointer is not - // beyond the next endWrite position, wait until it is. - leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; - //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; - if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; - if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset - endWrite = nextWritePointer + bufferBytes; +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= endWrite ) break; + // Query DirectSound devices. + struct DsProbeData probeInfo; + probeInfo.isInput = false; + probeInfo.dsDevices = &dsDevices; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } - // If we are here, then we must wait until the leadPointer advances - // beyond the end of our next write region. We use the - // Sleep() function to suspend operation until that happens. - double millis = ( endWrite - leadPointer ) * 1000.0; - millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - } + // 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( RtAudioError::WARNING ); + } - if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) - || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { - // We've strayed into the forbidden zone ... resync the read pointer. - handle->xrun[0] = true; - nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; - if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; - handle->bufferPointer[0] = nextWritePointer; - endWrite = nextWritePointer + bufferBytes; - } + // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut). + for ( unsigned int i=0; iLock( nextWritePointer, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + return static_cast(dsDevices.size()); +} - // Copy our buffer into the DS buffer - CopyMemory( buffer1, buffer, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); +RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + 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; } - nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; - handle->bufferPointer[0] = nextWritePointer; + } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } + if ( device >= dsDevices.size() ) { + errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + HRESULT result; + if ( dsDevices[ device ].validId[0] == false ) goto probeInput; - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; - bufferBytes *= formatBytes( stream_.deviceFormat[1] ); - } - else { - buffer = stream_.userBuffer[1]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; - bufferBytes *= formatBytes( stream_.userFormat ); - } + 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( RtAudioError::WARNING ); + goto probeInput; + } - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - long nextReadPointer = handle->bufferPointer[1]; - DWORD dsBufferSize = handle->dsBufferSize[1]; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto probeInput; + } - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; } + } - if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPointer + bufferBytes; + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; - // Handling depends on whether we are INPUT or DUPLEX. - // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, - // then a wait here will drag the write pointers into the forbidden zone. - // - // In DUPLEX mode, rather than wait, we will back off the read pointer until - // it's in a safe position. This causes dropouts, but it seems to be the only - // practical way to sync up the read and write pointers reliably, given the - // the very complex relationship between phase and increment of the read and write - // pointers. - // - // In order to minimize audible dropouts in DUPLEX mode, we will - // provide a pre-roll period of 0.5 seconds in which we return - // zeros from the read buffer while the pointers sync up. + output->Release(); - if ( stream_.mode == DUPLEX ) { - if ( safeReadPointer < endRead ) { - if ( duplexPrerollBytes <= 0 ) { - // Pre-roll time over. Be more agressive. - int adjustment = endRead-safeReadPointer; + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; - handle->xrun[1] = true; - // Two cases: - // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, - // and perform fine adjustments later. - // - small adjustments: back off by twice as much. - if ( adjustment >= 2*bufferBytes ) - nextReadPointer = safeReadPointer-2*bufferBytes; - else - nextReadPointer = safeReadPointer-bufferBytes-adjustment; + if ( dsDevices[ device ].validId[1] == false ) { + info.name = dsDevices[ device ].name; + info.probed = true; + return info; + } - if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + probeInput: - } - else { - // In pre=roll time. Just do it. - nextReadPointer = safeReadPointer - bufferBytes; - while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; - } - endRead = nextReadPointer + bufferBytes; - } - } - else { // mode == INPUT - while ( safeReadPointer < endRead ) { - // See comments for playback. - double millis = (endRead - safeReadPointer) * 1000.0; - millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } - // Wake up and find out where we are now. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset - } - } + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } - // Lock free space in the buffer - result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + // Get input channel information. + info.inputChannels = inCaps.dwChannels; - if ( duplexPrerollBytes <= 0 ) { - // Copy our buffer into the DS buffer - CopyMemory( buffer, buffer1, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + // Get sample rate and format information. + std::vector rates; + if ( inCaps.dwChannels >= 2 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); } - else { - memset( buffer, 0, bufferSize1 ); - if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); - duplexPrerollBytes -= bufferSize1 + bufferSize2; + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); } + } + else if ( inCaps.dwChannels == 1 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; - // Update our buffer offset and unlock sound buffer - nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); } - handle->bufferPointer[1] = nextReadPointer; + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); + } + } + else info.inputChannels = 0; // technically, this would be an error - // No byte swapping necessary in DirectSound implementation. + input->Release(); - // If necessary, convert 8-bit data from unsigned to signed. - if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) - for ( int j=0; j 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - RtApi::tickStreamTime(); -} + if ( device == 0 ) info.isDefaultInput = true; -// Definitions for utility functions and callbacks -// specific to the DirectSound implementation. + // Copy name and return. + info.name = dsDevices[ device ].name; + info.probed = true; + return info; +} -extern "C" unsigned __stdcall callbackHandler( void *ptr ) +bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) { - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiDs *object = (RtApiDs *) info->object; - bool* isRunning = &info->isRunning; + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + return FAILURE; + } - while ( *isRunning == true ) { - object->callbackEvent(); + 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!"; + return FAILURE; } - _endthreadex( 0 ); - return 0; -} + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } -#include "tchar.h" + if ( mode == OUTPUT ) { + if ( dsDevices[ device ].validId[0] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + else { // mode == INPUT + if ( dsDevices[ device ].validId[1] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } -std::string convertTChar( LPCTSTR name ) -{ -#if defined( UNICODE ) || defined( _UNICODE ) - int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL); - std::string s( length, 0 ); - length = WideCharToMultiByte(CP_UTF8, 0, name, wcslen(name), &s[0], length, NULL, NULL); -#else - std::string s( name ); -#endif + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // window. Also, if the application window closes before the + // DirectSound buffer, DirectSound can crash. In the past, I had + // problems when using GetDesktopWindow() but it seems fine now + // (January 2010). I'll leave it commented here. + // HWND hWnd = GetForegroundWindow(); + HWND hWnd = GetDesktopWindow(); - return s; -} + // Check the numberOfBuffers parameter and limit the lowest value to + // two. This is a judgement call and a value of two is probably too + // low for capture, but it should work for playback. + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ) -{ - bool *isInput = (bool *) lpContext; + // Check the lower range of the user-specified buffer size and set + // (arbitrarily) to a lower bound of 32. + if ( *bufferSize < 32 ) *bufferSize = 32; - HRESULT hr; - bool validDevice = false; - if ( *isInput == true ) { - DSCCAPS caps; - LPDIRECTSOUNDCAPTURE object; + // Create the wave format structure. The data format setting will + // be determined later. + WAVEFORMATEX waveFormat; + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels + firstChannel; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; + // Determine the device buffer size. By default, we'll use the value + // defined above (32K), but we will grow it to make allowances for + // very large software buffer sizes. + DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE; + DWORD dsPointerLeadTime = 0; - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) - validDevice = true; - } - object->Release(); - } - else { - DSCAPS caps; - LPDIRECTSOUND object; - hr = DirectSoundCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; + void *ohandle = 0, *bhandle = 0; + HRESULT result; + if ( mode == OUTPUT ) { - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) - validDevice = true; + LPDIRECTSOUND output; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - object->Release(); - } - // If good device, then save its name and guid. - std::string name = convertTChar( description ); - if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) - name = "Default Device"; - if ( validDevice ) { - for ( unsigned int i=0; iGetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - DsDevice device; - device.name = name; - device.found = true; - if ( *isInput ) { - device.id[1] = lpguid; - device.validId[1] = true; + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check format information. Use 16-bit format unless not + // supported or user requests 8-bit. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && + !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; } else { - device.id[0] = lpguid; - device.validId[0] = true; + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; } - dsDevices.push_back( device ); - } - - return TRUE; -} + stream_.userFormat = format; -static const char* getErrorString( int code ) -{ - switch ( code ) { + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - case DSERR_ALLOCATED: - return "Already allocated"; + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; - case DSERR_CONTROLUNAVAIL: - return "Control unavailable"; + // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. + // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. + result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - case DSERR_INVALIDPARAM: - return "Invalid parameter"; + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format + // (since the default is 8-bit, 22 kHz!). Setup the DS primary + // buffer description. + DSBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; - case DSERR_INVALIDCALL: - return "Invalid call"; + // Obtain the primary buffer + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - case DSERR_GENERIC: - return "Generic error"; + // Set the primary DS buffer sound format. + result = buffer->SetFormat( &waveFormat ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - case DSERR_PRIOLEVELNEEDED: - return "Priority level needed"; + // Setup the secondary DS buffer description. + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; - case DSERR_OUTOFMEMORY: - return "Out of memory"; + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } - case DSERR_BADFORMAT: - return "The sample rate or the channel format is not supported"; + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - case DSERR_UNSUPPORTED: - return "Not supported"; + dsBufferSize = dsbcaps.dwBufferBytes; - case DSERR_NODRIVER: - return "No driver"; + // Lock the DS buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - case DSERR_ALREADYINITIALIZED: - return "Already initialized"; + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); - case DSERR_NOAGGREGATION: - return "No aggregation"; + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - case DSERR_BUFFERLOST: - return "Buffer lost"; + ohandle = (void *) output; + bhandle = (void *) buffer; + } - case DSERR_OTHERAPPHASPRIO: - return "Another application already has priority"; + if ( mode == INPUT ) { - case DSERR_UNINITIALIZED: - return "Uninitialized"; + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - default: - return "DirectSound unknown error"; - } -} -//******************** End of __WINDOWS_DS__ *********************// -#endif + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; + return FAILURE; + } -#if defined(__LINUX_ALSA__) + // Check format information. Use 16-bit format unless user + // requests 8-bit. + DWORD deviceFormats; + if ( channels + firstChannel == 2 ) { + deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + else { // channel == 1 + deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + stream_.userFormat = format; -#include -#include + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - // A structure to hold various information related to the ALSA API - // implementation. -struct AlsaHandle { - snd_pcm_t *handles[2]; - bool synchronized; - bool xrun[2]; - pthread_cond_t runnable_cv; - bool runnable; + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; - AlsaHandle() - :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } -}; + // Setup the secondary DS buffer description. + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; -extern "C" void *alsaCallbackHandler( void * ptr ); + // Create the capture buffer. + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } -RtApiAlsa :: RtApiAlsa() -{ - // Nothing to do here. -} + // Get the buffer size ... might be different from what we specified. + DSCBCAPS dscbcaps; + dscbcaps.dwSize = sizeof( DSCBCAPS ); + result = buffer->GetCaps( &dscbcaps ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } -RtApiAlsa :: ~RtApiAlsa() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} + dsBufferSize = dscbcaps.dwBufferBytes; -unsigned int RtApiAlsa :: getDeviceCount( void ) -{ - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *handle; + // NOTE: We could have a problem here if this is a duplex stream + // and the play and capture hardware buffer sizes are different + // (I'm actually not sure if that is a problem or not). + // Currently, we are not verifying that. - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &handle, name, 0 ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; + // Lock the capture buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto nextcard; + return FAILURE; } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( handle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - break; - } - if ( subdevice < 0 ) - break; - nDevices++; + + // Zero the buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - nextcard: - snd_ctl_close( handle ); - snd_card_next( &card ); + + ohandle = (void *) input; + bhandle = (void *) buffer; } - return nDevices; -} + // Set various stream parameters + DsHandle *handle = 0; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.nUserChannels[mode] = channels; + stream_.bufferSize = *bufferSize; + stream_.channelOffset[mode] = firstChannel; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; -RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; + // Set flag for buffer conversion + stream_.doConvertBuffer[mode] = false; + if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + stream_.doConvertBuffer[mode] = true; + if (stream_.userFormat != stream_.deviceFormat[mode]) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *chandle; + // Allocate necessary internal buffers + long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - break; - } - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - goto foundDevice; + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; } - nDevices++; } - nextcard: - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - if ( nDevices == 0 ) { - errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } } - foundDevice: - - // If a stream is already open, we cannot probe the stream devices. - // Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED && - ( stream_.device[0] == device || stream_.device[1] == device ) ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); - return info; + // Allocate our DsHandle structures for the stream. + if ( stream_.apiHandle == 0 ) { + try { + handle = new DsHandle; } - return devices_[ device ]; + catch ( std::bad_alloc& ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; } + else + handle = (DsHandle *) stream_.apiHandle; + handle->id[mode] = ohandle; + handle->buffer[mode] = bhandle; + handle->dsBufferSize[mode] = dsBufferSize; + handle->dsPointerLeadTime[mode] = dsPointerLeadTime; - int openMode = SND_PCM_ASYNC; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); - snd_pcm_t *phandle; - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca( ¶ms ); + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + stream_.nBuffers = nBuffers; + stream_.sampleRate = sampleRate; - // First try for playback - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_device( pcminfo, subdevice ); - snd_pcm_info_set_subdevice( pcminfo, 0 ); - snd_pcm_info_set_stream( pcminfo, stream ); + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - result = snd_ctl_pcm_info( chandle, pcminfo ); - if ( result < 0 ) { - // Device probably doesn't support playback. - goto captureProbe; - } + // Setup the callback thread. + if ( stream_.callbackInfo.isRunning == false ) { + unsigned threadId; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, + &stream_.callbackInfo, 0, &threadId ); + if ( stream_.callbackInfo.thread == 0 ) { + errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; + goto error; + } - 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 ); - goto captureProbe; + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); } + return SUCCESS; - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; + error: + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) buffer->Release(); + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) buffer->Release(); + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; } - // Get output channel information. - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } } - info.outputChannels = value; - snd_pcm_close( phandle ); - captureProbe: - // Now try for capture - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - result = snd_ctl_pcm_info( chandle, pcminfo ); - snd_ctl_close( chandle ); - if ( result < 0 ) { - // Device probably doesn't support capture. - if ( info.outputChannels == 0 ) return info; - goto probeParameters; + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; } - 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 ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; } - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } } - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - info.inputChannels = value; - snd_pcm_close( phandle ); - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} - // ALSA doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; +void RtApiDs :: startStream() +{ + verifyStream(); + RtApi::startStream(); + + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } - probeParameters: - // At this point, we just need to figure out the supported data - // formats and sample rates. We'll proceed by opening the device in - // the direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. + DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( info.outputChannels >= info.inputChannels ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); + // Increase scheduler frequency on lesser windows (a side-effect of + // increasing timer accuracy). On greater windows (Win2K or later), + // this is already in effect. + timeBeginPeriod( 1 ); - 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 ); - return info; - } + buffersRolling = false; + duplexPrerollBytes = 0; - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; + if ( stream_.mode == DUPLEX ) { + // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. + duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); } - // Test our discrete set of sample rate values. - info.sampleRates.clear(); - for ( unsigned int i=0; ibuffer[0]; + result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } } - // Probe the supported data formats ... we don't care about endian-ness just yet - snd_pcm_format_t format; - info.nativeFormats = 0; - format = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT8; - format = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT16; - format = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT24; - format = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT32; - format = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT32; - format = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT64; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + result = buffer->Start( DSCBSTART_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } } - // Get the device name - char *cardname; - result = snd_card_get_name( card, &cardname ); - if ( result >= 0 ) - sprintf( name, "hw:%s,%d", cardname, subdevice ); - info.name = name; + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; - // That's all ... close the device and return - snd_pcm_close( phandle ); - info.probed = true; - return info; + unlock: + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); } -void RtApiAlsa :: saveDeviceInfo( void ) +void RtApiDs :: stopStream() { - devices_.clear(); - - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; idrainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } - // I'm not using the "plug" interface ... too much inconsistent behavior. + stream_.state = STREAM_STOPPED; - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *chandle; + MUTEX_LOCK( &stream_.mutex ); - if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT ) - snprintf(name, sizeof(name), "%s", "default"); - else { - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) break; - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - snd_ctl_close( chandle ); - goto foundDevice; - } - nDevices++; - } - snd_ctl_close( chandle ); - snd_card_next( &card ); + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; } - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; - return FAILURE; + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; } - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; - return FAILURE; + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; } + + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; } - foundDevice: + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; - // The getDeviceInfo() function will not work for a device that is - // already open. Thus, we'll probe the system before opening a - // stream and save the results for use by getDeviceInfo(). - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once - this->saveDeviceInfo(); + stream_.state = STREAM_STOPPED; - snd_pcm_stream_t stream; - if ( mode == OUTPUT ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; + if ( stream_.mode != DUPLEX ) + MUTEX_LOCK( &stream_.mutex ); - snd_pcm_t *phandle; - int openMode = SND_PCM_ASYNC; - result = snd_pcm_open( &phandle, name, stream, openMode ); - if ( result < 0 ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; - else - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; - errorText_ = errorStream_.str(); - return FAILURE; - } + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca( &hw_params ); - result = snd_pcm_hw_params_any( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } -#if defined(__RTAUDIO_DEBUG__) - fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); - snd_pcm_hw_params_dump( hw_params, out ); -#endif + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); - // Set access ... check user preference. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { - stream_.userInterleaved = false; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - stream_.deviceInterleaved[mode] = true; - } - else - stream_.deviceInterleaved[mode] = false; - } - else { - stream_.userInterleaved = true; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - stream_.deviceInterleaved[mode] = false; + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; } - else - stream_.deviceInterleaved[mode] = true; - } - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; } - // Determine how to set the device format. - stream_.userFormat = format; - snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + MUTEX_UNLOCK( &stream_.mutex ); - if ( format == RTAUDIO_SINT8 ) - deviceFormat = SND_PCM_FORMAT_S8; - else if ( format == RTAUDIO_SINT16 ) - deviceFormat = SND_PCM_FORMAT_S16; - else if ( format == RTAUDIO_SINT24 ) - deviceFormat = SND_PCM_FORMAT_S24; - else if ( format == RTAUDIO_SINT32 ) - deviceFormat = SND_PCM_FORMAT_S32; - else if ( format == RTAUDIO_FLOAT32 ) - deviceFormat = SND_PCM_FORMAT_FLOAT; - else if ( format == RTAUDIO_FLOAT64 ) - deviceFormat = SND_PCM_FORMAT_FLOAT64; + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { - stream_.deviceFormat[mode] = format; - goto setFormat; +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; } - // The user requested format is not natively supported by the device. - deviceFormat = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - goto setFormat; - } + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 2; - deviceFormat = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - goto setFormat; - } + stopStream(); +} - deviceFormat = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - goto setFormat; +void RtApiDs :: callbackEvent() +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { + Sleep( 50 ); // sleep 50 milliseconds + return; } - deviceFormat = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - goto setFormat; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; } - deviceFormat = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - goto setFormat; - } + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; - deviceFormat = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - goto setFormat; - } + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > stream_.nBuffers + 2 ) { - // If we get here, no supported format was found. - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + HRESULT result; + DWORD currentWritePointer, safeWritePointer; + DWORD currentReadPointer, safeReadPointer; + UINT nextWritePointer; + + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + + 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] ); + + // It takes a while for the devices to get rolling. As a result, + // there's no guarantee that the capture and write device pointers + // will move in lockstep. Wait here for both devices to start + // rolling, and then set our buffer pointers accordingly. + // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 + // bytes later than the write buffer. + + // Stub: a serious risk of having a pre-emptive scheduling round + // take place between the two GetCurrentPosition calls... but I'm + // really not sure how to solve the problem. Temporarily boost to + // Realtime priority, maybe; but I'm not sure what priority the + // DirectSound service threads run at. We *should* be roughly + // within a ms or so of correct. + + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + + DWORD startSafeWritePointer, startSafeReadPointer; + + result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + 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(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; + Sleep( 1 ); + } + + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + handle->bufferPointer[1] = safeReadPointer; + } + else if ( stream_.mode == OUTPUT ) { + + // Set the proper nextWritePosition after initial startup. + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + } + + buffersRolling = true; + } + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + memset( stream_.userBuffer[0], 0, bufferBytes ); + } + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; + bufferBytes *= formatBytes( stream_.deviceFormat[0] ); + } + else { + buffer = stream_.userBuffer[0]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + // No byte swapping necessary in DirectSound implementation. + + // Ahhh ... windoze. 16-bit data is signed but 8-bit data is + // unsigned. So, we need to convert our signed 8-bit data here to + // unsigned. + if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) + for ( int i=0; idsBufferSize[0]; + nextWritePointer = handle->bufferPointer[0]; + + DWORD endWrite, leadPointer; + while ( true ) { + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // We will copy our output buffer into the region between + // safeWritePointer and leadPointer. If leadPointer is not + // beyond the next endWrite position, wait until it is. + leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; + //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; + if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; + if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset + endWrite = nextWritePointer + bufferBytes; + + // Check whether the entire write region is behind the play pointer. + if ( leadPointer >= endWrite ) break; + + // If we are here, then we must wait until the leadPointer advances + // beyond the end of our next write region. We use the + // Sleep() function to suspend operation until that happens. + double millis = ( endWrite - leadPointer ) * 1000.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + } + + if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + handle->xrun[0] = true; + nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; + if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + endWrite = nextWritePointer + bufferBytes; + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; + bufferBytes *= formatBytes( stream_.deviceFormat[1] ); + } + else { + buffer = stream_.userBuffer[1]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPointer = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPointer + bufferBytes; + + // Handling depends on whether we are INPUT or DUPLEX. + // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, + // then a wait here will drag the write pointers into the forbidden zone. + // + // In DUPLEX mode, rather than wait, we will back off the read pointer until + // it's in a safe position. This causes dropouts, but it seems to be the only + // practical way to sync up the read and write pointers reliably, given the + // the very complex relationship between phase and increment of the read and write + // pointers. + // + // In order to minimize audible dropouts in DUPLEX mode, we will + // provide a pre-roll period of 0.5 seconds in which we return + // zeros from the read buffer while the pointers sync up. + + if ( stream_.mode == DUPLEX ) { + if ( safeReadPointer < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPointer; + + handle->xrun[1] = true; + // Two cases: + // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, + // and perform fine adjustments later. + // - small adjustments: back off by twice as much. + if ( adjustment >= 2*bufferBytes ) + nextReadPointer = safeReadPointer-2*bufferBytes; + else + nextReadPointer = safeReadPointer-bufferBytes-adjustment; + + if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + + } + else { + // In pre=roll time. Just do it. + nextReadPointer = safeReadPointer - bufferBytes; + while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + } + endRead = nextReadPointer + bufferBytes; + } + } + else { // mode == INPUT + while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { + // See comments for playback. + double millis = (endRead - safeReadPointer) * 1000.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up and find out where we are now. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + } + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( duplexPrerollBytes <= 0 ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer, buffer1, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + } + else { + memset( buffer, 0, bufferSize1 ); + if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); + duplexPrerollBytes -= bufferSize1 + bufferSize2; + } + + // Update our buffer offset and unlock sound buffer + nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[1] = nextReadPointer; + + // No byte swapping necessary in DirectSound implementation. + + // If necessary, convert 8-bit data from unsigned to signed. + if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) + for ( int j=0; jobject; + bool* isRunning = &info->isRunning; + + while ( *isRunning == true ) { + object->callbackEvent(); + } + + _endthreadex( 0 ); + return 0; +} + +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR /*module*/, + LPVOID lpContext ) +{ + struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; + std::vector& dsDevices = *probeInfo.dsDevices; + + HRESULT hr; + bool validDevice = false; + if ( probeInfo.isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; + + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) + validDevice = true; + } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + validDevice = true; + } + object->Release(); + } + + // If good device, then save its name and guid. + std::string name = convertCharPointerToStdString( description ); + //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) + if ( lpguid == NULL ) + name = "Default Device"; + if ( validDevice ) { + for ( unsigned int i=0; i +#include + + // A structure to hold various information related to the ALSA API + // implementation. +struct AlsaHandle { + snd_pcm_t *handles[2]; + bool synchronized; + bool xrun[2]; + pthread_cond_t runnable_cv; + bool runnable; + + AlsaHandle() + :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } +}; + +static void *alsaCallbackHandler( void * ptr ); + +RtApiAlsa :: RtApiAlsa() +{ + // Nothing to do here. +} + +RtApiAlsa :: ~RtApiAlsa() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiAlsa :: getDeviceCount( void ) +{ + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *handle; + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &handle, name, 0 ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( handle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) + break; + nDevices++; + } + nextcard: + snd_ctl_close( handle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &handle, "default", 0 ); + if (result == 0) { + nDevices++; + snd_ctl_close( handle ); + } + + return nDevices; +} + +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *chandle; + + // Count cards and devices + card = -1; + subdevice = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + goto foundDevice; + } + nDevices++; + } + nextcard: + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + nDevices++; + } + + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + foundDevice: + + // If a stream is already open, we cannot probe the stream devices. + // Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED && + ( stream_.device[0] == device || stream_.device[1] == device ) ) { + snd_ctl_close( chandle ); + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + int openMode = SND_PCM_ASYNC; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); + snd_pcm_t *phandle; + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca( ¶ms ); + + // First try for playback unless default device (which has subdev -1) + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_stream( pcminfo, stream ); + if ( subdevice != -1 ) { + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + + result = snd_ctl_pcm_info( chandle, pcminfo ); + if ( result < 0 ) { + // Device probably doesn't support playback. + goto captureProbe; + } + } + + 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( RtAudioError::WARNING ); + goto captureProbe; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + + // Get output channel information. + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); + + captureProbe: + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + // Now try for capture unless default device (with subdev = -1) + if ( subdevice != -1 ) { + result = snd_ctl_pcm_info( chandle, pcminfo ); + snd_ctl_close( chandle ); + if ( result < 0 ) { + // Device probably doesn't support capture. + if ( info.outputChannels == 0 ) return info; + 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( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; + snd_pcm_close( phandle ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // ALSA doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + probeParameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if ( info.outputChannels >= info.inputChannels ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + 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( RtAudioError::WARNING ); + return info; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } + } + if ( info.sampleRates.size() == 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe the supported data formats ... we don't care about endian-ness just yet + snd_pcm_format_t format; + info.nativeFormats = 0; + format = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT8; + format = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT16; + format = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT24; + format = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT32; + format = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT32; + format = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT64; + + // 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( RtAudioError::WARNING ); + return info; + } + + // Get the device name + char *cardname; + result = snd_card_get_name( card, &cardname ); + if ( result >= 0 ) { + sprintf( name, "hw:%s,%d", cardname, subdevice ); + free( cardname ); + } + info.name = name; + + // That's all ... close the device and return + snd_pcm_close( phandle ); + info.probed = true; + return info; +} + +void RtApiAlsa :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; iflags & RTAUDIO_ALSA_USE_DEFAULT ) + snprintf(name, sizeof(name), "%s", "default"); + else { + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) break; + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + snd_ctl_close( chandle ); + goto foundDevice; + } + nDevices++; + } + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + nDevices++; + } + + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + } + + foundDevice: + + // The getDeviceInfo() function will not work for a device that is + // already open. Thus, we'll probe the system before opening a + // stream and save the results for use by getDeviceInfo(). + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once + this->saveDeviceInfo(); + + snd_pcm_stream_t stream; + if ( mode == OUTPUT ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + + snd_pcm_t *phandle; + int openMode = SND_PCM_ASYNC; + result = snd_pcm_open( &phandle, name, stream, openMode ); + if ( result < 0 ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; + else + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca( &hw_params ); + result = snd_pcm_hw_params_any( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set access ... check user preference. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { + stream_.userInterleaved = false; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + stream_.deviceInterleaved[mode] = true; + } + else + stream_.deviceInterleaved[mode] = false; + } + else { + stream_.userInterleaved = true; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + stream_.deviceInterleaved[mode] = false; + } + else + stream_.deviceInterleaved[mode] = true; + } + + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + + if ( format == RTAUDIO_SINT8 ) + deviceFormat = SND_PCM_FORMAT_S8; + else if ( format == RTAUDIO_SINT16 ) + deviceFormat = SND_PCM_FORMAT_S16; + else if ( format == RTAUDIO_SINT24 ) + deviceFormat = SND_PCM_FORMAT_S24; + else if ( format == RTAUDIO_SINT32 ) + deviceFormat = SND_PCM_FORMAT_S32; + else if ( format == RTAUDIO_FLOAT32 ) + deviceFormat = SND_PCM_FORMAT_FLOAT; + else if ( format == RTAUDIO_FLOAT64 ) + deviceFormat = SND_PCM_FORMAT_FLOAT64; + + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { + stream_.deviceFormat[mode] = format; + goto setFormat; + } + + // The user requested format is not natively supported by the device. + deviceFormat = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + goto setFormat; + } + + // If we get here, no supported format was found. + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; setFormat: result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); if ( result < 0 ) { snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine whether byte-swaping is necessary. + stream_.doByteSwap[mode] = false; + if ( deviceFormat != SND_PCM_FORMAT_S8 ) { + result = snd_pcm_format_cpu_endian( deviceFormat ); + if ( result == 0 ) + stream_.doByteSwap[mode] = true; + else if (result < 0) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Set the sample rate. + result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine the number of channels for this device. We support a possible + // minimum device channel number > than the value requested by the user. + stream_.nUserChannels[mode] = channels; + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); + unsigned int deviceChannels = value; + if ( result < 0 || deviceChannels < channels + firstChannel ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + deviceChannels = value; + if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; + stream_.nDeviceChannels[mode] = deviceChannels; + + // Set the device channels. + result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the buffer (or period) size. + int dir = 0; + snd_pcm_uframes_t periodSize = *bufferSize; + result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + *bufferSize = periodSize; + + // Set the buffer number, which in ALSA is referred to as the "period". + unsigned int periods = 0; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; + if ( periods < 2 ) periods = 4; // a fairly safe default value + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + + // Install the hardware configuration + result = snd_pcm_hw_params( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca( &sw_params ); + snd_pcm_sw_params_current( phandle, sw_params ); + snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); + snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + + result = snd_pcm_sw_params( phandle, sw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } - // Determine whether byte-swaping is necessary. - stream_.doByteSwap[mode] = false; - if ( deviceFormat != SND_PCM_FORMAT_S8 ) { - result = snd_pcm_format_cpu_endian( deviceFormat ); - if ( result == 0 ) - stream_.doByteSwap[mode] = true; - else if (result < 0) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); + snd_pcm_sw_params_dump( sw_params, out ); +#endif + + // 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; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the ApiHandle if necessary and then save. + AlsaHandle *apiInfo = 0; + if ( stream_.apiHandle == 0 ) { + try { + apiInfo = (AlsaHandle *) new AlsaHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; + goto error; + } + + if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) apiInfo; + apiInfo->handles[0] = 0; + apiInfo->handles[1] = 0; + } + else { + apiInfo = (AlsaHandle *) stream_.apiHandle; + } + apiInfo->handles[mode] = phandle; + phandle = 0; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.nBuffers = periods; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + // Link the streams if possible. + apiInfo->synchronized = false; + if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) + apiInfo->synchronized = true; + else { + errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; + error( RtAudioError::WARNING ); + } + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); + +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + // We previously attempted to increase the audio callback priority + // to SCHED_RR here via the attributes. However, while no errors + // were reported in doing so, it did not work. So, now this is + // done in the alsaCallbackHandler function. + stream_.callbackInfo.doRealtime = true; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + stream_.callbackInfo.priority = priority; + } +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiAlsa::error creating callback thread!"; + goto error; + } + } + + return SUCCESS; + + error: + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + if ( phandle) snd_pcm_close( phandle ); + + 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; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiAlsa :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[0] ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[1] ); + } + + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } } - // Set the sample rate. - result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - // Determine the number of channels for this device. We support a possible - // minimum device channel number > than the value requested by the user. - stream_.nUserChannels[mode] = channels; - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); - unsigned int deviceChannels = value; - if ( result < 0 || deviceChannels < channels + firstChannel ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiAlsa :: startStream() +{ + // This method calls snd_pcm_prepare if the device isn't already in that state. + + verifyStream(); + RtApi::startStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; } - result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + snd_pcm_state_t state; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + state = snd_pcm_state( handle[0] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } } - deviceChannels = value; - if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; - stream_.nDeviceChannels[mode] = deviceChannels; - // Set the device channels. - result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + 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] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } } - // Set the buffer (or period) size. - int dir = 0; - snd_pcm_uframes_t periodSize = *bufferSize; - result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + stream_.state = STREAM_RUNNING; + + unlock: + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; } - *bufferSize = periodSize; - // Set the buffer number, which in ALSA is referred to as the "period". - unsigned int periods = 0; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; - if ( periods < 2 ) periods = 4; // a fairly safe default value - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( apiInfo->synchronized ) + result = snd_pcm_drop( handle[0] ); + else + result = snd_pcm_drain( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } } - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } } - stream_.bufferSize = *bufferSize; + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); - // Install the hardware configuration - result = snd_pcm_hw_params( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; } -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump( hw_params, out ); -#endif + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); - // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca( &sw_params ); - snd_pcm_sw_params_current( phandle, sw_params ); - snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); - snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = snd_pcm_drop( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } - // The following two settings were suggested by Theo Veenker - //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); - //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } - // here are two options for a fix - //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); - snd_pcm_uframes_t val; - snd_pcm_sw_params_get_boundary( sw_params, &val ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: callbackEvent() +{ + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !apiInfo->runnable ) + pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + apiInfo->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + apiInfo->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - result = snd_pcm_sw_params( phandle, sw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + if ( doStopStream == 2 ) { + abortStream(); + return; } -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); - snd_pcm_sw_params_dump( sw_params, out ); -#endif + MUTEX_LOCK( &stream_.mutex ); - // 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; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; - // Allocate the ApiHandle if necessary and then save. - AlsaHandle *apiInfo = 0; - if ( stream_.apiHandle == 0 ) { - try { - apiInfo = (AlsaHandle *) new AlsaHandle; + int result; + char *buffer; + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + channels = stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; - goto error; + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; } - if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; - goto error; + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ihandles[0] = 0; - apiInfo->handles[1] = 0; - } - else { - apiInfo = (AlsaHandle *) stream_.apiHandle; - } - apiInfo->handles[mode] = phandle; + if ( result < (int) stream_.bufferSize ) { + // Either an error or overrun occured. + if ( result == -EPIPE ) { + snd_pcm_state_t state = snd_pcm_state( handle[1] ); + if ( state == SND_PCM_STATE_XRUN ) { + apiInfo->xrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto tryOutput; + } - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; - goto error; + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; } - if ( stream_.doConvertBuffer[mode] ) { + tryOutput: - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; - goto error; + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + + // Write samples to device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[0] ) + result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + else + errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); } + error( RtAudioError::WARNING ); + goto unlock; } + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; } - stream_.sampleRate = sampleRate; - stream_.nBuffers = periods; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - // Link the streams if possible. - apiInfo->synchronized = false; - if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) - apiInfo->synchronized = true; - else { - errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; - error( RtError::WARNING ); - } +static void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( info->doRealtime ) { + pthread_t tID = pthread_self(); // ID of this thread + sched_param prio = { info->priority }; // scheduling priority of thread + pthread_setschedparam( tID, SCHED_RR, &prio ); } - else { - stream_.mode = mode; +#endif - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } - // Set the thread attributes for joinable and realtime scheduling - // priority (optional). The higher priority will only take affect - // if the program is run as root or suid. Note, under Linux - // processes with CAP_SYS_NICE privilege, a user can change - // scheduling policy and priority (thus need not be root). See - // POSIX "capabilities". - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); + pthread_exit( NULL ); +} + +//******************** End of __LINUX_ALSA__ *********************// #endif - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiAlsa::error creating callback thread!"; - goto error; - } - } +#if defined(__LINUX_PULSE__) - return SUCCESS; +// Code written by Peter Meerwald, pmeerw@pmeerw.net +// and Tristan Matthews. - error: - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable_cv ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } +#include +#include +#include - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } +static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, + 44100, 48000, 96000, 0}; - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } +struct rtaudio_pa_format_mapping_t { + RtAudioFormat rtaudio_format; + pa_sample_format_t pa_format; +}; - return FAILURE; +static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { + {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, + {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, + {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, + {0, PA_SAMPLE_INVALID}}; + +struct PulseAudioHandle { + pa_simple *s_play; + pa_simple *s_rec; + pthread_t thread; + pthread_cond_t runnable_cv; + bool runnable; + PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } +}; + +RtApiPulse::~RtApiPulse() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); } -void RtApiAlsa :: closeStream() +unsigned int RtApiPulse::getDeviceCount( void ) { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; + return 1; +} + +RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) +{ + RtAudio::DeviceInfo info; + info.probed = true; + info.name = "PulseAudio"; + info.outputChannels = 2; + info.inputChannels = 2; + info.duplexChannels = 2; + info.isDefaultOutput = true; + info.isDefaultInput = true; + + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + info.sampleRates.push_back( *sr ); + + info.preferredSampleRate = 48000; + info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; + + return info; +} + +static void *pulseaudio_callback( void * user ) +{ + CallbackInfo *cbi = static_cast( user ); + RtApiPulse *context = static_cast( cbi->object ); + volatile bool *isRunning = &cbi->isRunning; + + while ( *isRunning ) { + pthread_testcancel(); + context->callbackEvent(); } - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + pthread_exit( NULL ); +} + +void RtApiPulse::closeStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) { - apiInfo->runnable = true; - pthread_cond_signal( &apiInfo->runnable_cv ); - } - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); + if ( pah ) { + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[0] ); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[1] ); - } + pthread_join( pah->thread, 0 ); + if ( pah->s_play ) { + pa_simple_flush( pah->s_play, NULL ); + pa_simple_free( pah->s_play ); + } + if ( pah->s_rec ) + pa_simple_free( pah->s_rec ); - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable_cv ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; stream_.apiHandle = 0; } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } + if ( stream_.userBuffer[0] ) { + free( stream_.userBuffer[0] ); + stream_.userBuffer[0] = 0; } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + if ( stream_.userBuffer[1] ) { + free( stream_.userBuffer[1] ); + stream_.userBuffer[1] = 0; } - stream_.mode = UNINITIALIZED; stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; } -void RtApiAlsa :: startStream() +void RtApiPulse::callbackEvent( void ) { - // This method calls snd_pcm_prepare if the device isn't already in that state. - - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !pah->runnable ) + pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); - int result = 0; - snd_pcm_state_t state; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - state = snd_pcm_state( handle[0] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; } + MUTEX_UNLOCK( &stream_.mutex ); } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - state = snd_pcm_state( handle[1] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " + "this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; } - stream_.state = STREAM_RUNNING; - - unlock: - apiInfo->runnable = true; - pthread_cond_signal( &apiInfo->runnable_cv ); - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], + stream_.bufferSize, streamTime, status, + stream_.callbackInfo.userData ); -void RtApiAlsa :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + if ( doStopStream == 2 ) { + abortStream(); return; } - stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); + void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT]; + void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT]; - //if ( stream_.state == STREAM_STOPPED ) { - // MUTEX_UNLOCK( &stream_.mutex ); - // return; - //} + if ( stream_.state != STREAM_RUNNING ) + goto unlock; - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) - result = snd_pcm_drop( handle[0] ); - else - result = snd_pcm_drain( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + int pa_error; + size_t bytes; + if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.doConvertBuffer[OUTPUT] ) { + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[OUTPUT] ); + } else + bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << + pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); - goto unlock; + error( RtAudioError::WARNING ); } } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX) { + if ( stream_.doConvertBuffer[INPUT] ) + bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[INPUT] ); + else + bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << + pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); - goto unlock; + error( RtAudioError::WARNING ); + } + if ( stream_.doConvertBuffer[INPUT] ) { + convertBuffer( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.convertInfo[INPUT] ); } } unlock: - stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); + if (pah->s_play) { + int e = 0; + pa_usec_t const lat = pa_simple_get_latency(pah->s_play, &e); + if (e == 0) { + stream_.latency[0] = lat * stream_.sampleRate / 1000000; + } + } + + if ( doStopStream == 1 ) + stopStream(); } -void RtApiAlsa :: abortStream() +void RtApiPulse::startStream( void ) { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + RtApi::startStream(); + + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::startStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiPulse::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); return; } - stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); - //if ( stream_.state == STREAM_STOPPED ) { - // MUTEX_UNLOCK( &stream_.mutex ); - // return; - //} + stream_.state = STREAM_RUNNING; + + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); +} - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = snd_pcm_drop( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } +void RtApiPulse::stopStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + stream_.state = STREAM_STOPPED; + pah->runnable = false; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::stopStream: error draining output device, " << + pa_strerror( pa_error ) << "."; errorText_ = errorStream_.str(); - goto unlock; + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; } } - unlock: stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); } -void RtApiAlsa :: callbackEvent() +void RtApiPulse::abortStream( void ) { - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - while ( !apiInfo->runnable ) - pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); + errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } - if ( stream_.state != STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + pah->runnable = false; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); return; } - MUTEX_UNLOCK( &stream_.mutex ); } - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - apiInfo->xrun[0] = false; +bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, + unsigned int channels, unsigned int firstChannel, + unsigned int sampleRate, RtAudioFormat format, + unsigned int *bufferSize, RtAudio::StreamOptions *options ) +{ + PulseAudioHandle *pah = 0; + unsigned long bufferBytes = 0; + pa_sample_spec ss; + + if ( device != 0 ) return false; + if ( mode != INPUT && mode != OUTPUT ) return false; + if ( channels != 1 && channels != 2 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels."; + return false; + } + ss.channels = channels; + + if ( firstChannel != 0 ) return false; + + bool sr_found = false; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { + if ( sampleRate == *sr ) { + sr_found = true; + stream_.sampleRate = sampleRate; + ss.rate = sampleRate; + break; + } } - if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - apiInfo->xrun[1] = false; + if ( !sr_found ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; + return false; } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - if ( doStopStream == 2 ) { - abortStream(); - return; + bool sf_found = 0; + for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; + sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { + 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 ) { // Use internal data format conversion. + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + ss.format = PA_SAMPLE_FLOAT32LE; } - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; + // 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_.nUserChannels[mode] = channels; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.channelOffset[mode] = 0; + std::string streamName = "RtAudio"; - int result; - char *buffer; - int channels; - snd_pcm_t **handle; - snd_pcm_sframes_t frames; - RtAudioFormat format; - handle = (snd_pcm_t **) apiInfo->handles; + // 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; - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + // Allocate necessary internal buffers. + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + stream_.bufferSize = *bufferSize; - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - channels = stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - channels = stream_.nUserChannels[1]; - format = stream_.userFormat; - } + if ( stream_.doConvertBuffer[mode] ) { - // Read samples from device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[1] ) - result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ixrun[1] = true; - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory."; + goto error; } - error( RtError::WARNING ); - goto tryOutput; } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - // Check stream latency - result = snd_pcm_delay( handle[1], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; } - tryOutput: + stream_.device[mode] = device; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - channels = stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; + if ( !stream_.apiHandle ) { + PulseAudioHandle *pah = new PulseAudioHandle; + if ( !pah ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; + goto error; } - else { - buffer = stream_.userBuffer[0]; - channels = stream_.nUserChannels[0]; - format = stream_.userFormat; + + stream_.apiHandle = pah; + if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; + goto error; } + } + pah = static_cast( stream_.apiHandle ); - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + int error; + if ( options && !options->streamName.empty() ) streamName = options->streamName; + switch ( mode ) { + case INPUT: + pa_buffer_attr buffer_attr; + buffer_attr.fragsize = bufferBytes; + buffer_attr.maxlength = -1; - // Write samples to device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[0] ) - result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; is_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; } - - if ( result < (int) stream_.bufferSize ) { - // Either an error or underrun occured. - if ( result == -EPIPE ) { - snd_pcm_state_t state = snd_pcm_state( handle[0] ); - if ( state == SND_PCM_STATE_XRUN ) { - apiInfo->xrun[0] = true; - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtError::WARNING ); - goto unlock; + break; + case OUTPUT: + pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + if ( !pah->s_play ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; + goto error; } - - // Check stream latency - result = snd_pcm_delay( handle[0], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + break; + default: + goto error; } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + if ( stream_.mode == UNINITIALIZED ) + stream_.mode = mode; + else if ( stream_.mode == mode ) + goto error; + else + stream_.mode = DUPLEX; - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} + if ( !stream_.callbackInfo.isRunning ) { + stream_.callbackInfo.object = this; + stream_.callbackInfo.isRunning = true; + if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; + goto error; + } + } -extern "C" void *alsaCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAlsa *object = (RtApiAlsa *) info->object; - bool *isRunning = &info->isRunning; + stream_.state = STREAM_STOPPED; + return true; + + error: + if ( pah && stream_.callbackInfo.isRunning ) { + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + stream_.apiHandle = 0; + } - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } } - pthread_exit( NULL ); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; } -//******************** End of __LINUX_ALSA__ *********************// +//******************** End of __LINUX_PULSE__ *********************// #endif - #if defined(__LINUX_OSS__) #include #include #include #include -#include "soundcard.h" +#include #include #include -extern "C" void *ossCallbackHandler(void * ptr); +static void *ossCallbackHandler(void * ptr); // A structure to hold various information related to the OSS API // implementation. @@ -6417,7 +8646,7 @@ 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; } @@ -6425,7 +8654,7 @@ 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; } @@ -6441,7 +8670,7 @@ 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; } @@ -6450,7 +8679,7 @@ 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; } @@ -6458,13 +8687,15 @@ 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; } oss_audioinfo ainfo; @@ -6474,7 +8705,7 @@ 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; } @@ -6503,7 +8734,7 @@ 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; } @@ -6514,6 +8745,10 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) for ( unsigned int k=0; k info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + break; } } @@ -6522,15 +8757,19 @@ RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) else { // Check min and max rate values; for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) + if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) { info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } } } 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; @@ -6979,7 +9218,7 @@ void RtApiOss :: closeStream() { if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7026,9 +9265,10 @@ void RtApiOss :: closeStream() void RtApiOss :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiOss::startStream(): the stream is already running!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7050,7 +9290,7 @@ void RtApiOss :: stopStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7087,7 +9327,7 @@ 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 ); } } @@ -7114,7 +9354,7 @@ void RtApiOss :: stopStream() MUTEX_UNLOCK( &stream_.mutex ); if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiOss :: abortStream() @@ -7122,7 +9362,7 @@ void RtApiOss :: abortStream() verifyStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return; } @@ -7160,7 +9400,7 @@ void RtApiOss :: abortStream() MUTEX_UNLOCK( &stream_.mutex ); if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); + error( RtAudioError::SYSTEM_ERROR ); } void RtApiOss :: callbackEvent() @@ -7178,7 +9418,7 @@ 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; } @@ -7248,7 +9488,7 @@ 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. } } @@ -7275,7 +9515,7 @@ 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; } @@ -7295,7 +9535,7 @@ void RtApiOss :: callbackEvent() if ( doStopStream == 1 ) this->stopStream(); } -extern "C" void *ossCallbackHandler( void *ptr ) +static void *ossCallbackHandler( void *ptr ) { CallbackInfo *info = (CallbackInfo *) ptr; RtApiOss *object = (RtApiOss *) info->object; @@ -7321,20 +9561,41 @@ extern "C" 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 - if ( type == RtError::WARNING && showWarnings_ == true ) + + RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; + if ( errorCallback ) { + // abortStream() can generate new error messages. Ignore them. Just keep original one. + + if ( firstErrorOccurred_ ) + return; + + firstErrorOccurred_ = true; + const std::string errorMessage = errorText_; + + if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { + stream_.callbackInfo.isRunning = false; // exit from the thread + abortStream(); + } + + errorCallback( type, errorMessage ); + firstErrorOccurred_ = false; + return; + } + + 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 ); } } @@ -7353,6 +9614,7 @@ void RtApi :: clearStreamInfo() stream_.callbackInfo.callback = 0; stream_.callbackInfo.userData = 0; stream_.callbackInfo.isRunning = false; + stream_.callbackInfo.errorCallback = 0; for ( int i=0; i<2; i++ ) { stream_.device[i] = 11111; stream_.doConvertBuffer[i] = false; @@ -7378,16 +9640,17 @@ unsigned int RtApi :: formatBytes( RtAudioFormat format ) { if ( format == RTAUDIO_SINT16 ) return 2; - else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || - format == RTAUDIO_FLOAT32 ) + else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 ) return 4; else if ( format == RTAUDIO_FLOAT64 ) return 8; + else if ( format == RTAUDIO_SINT24 ) + return 3; else if ( format == RTAUDIO_SINT8 ) return 1; errorText_ = "RtApi::formatBytes: undefined format."; - error( RtError::WARNING ); + error( RtAudioError::WARNING ); return 0; } @@ -7515,11 +9778,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } } else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; + Int24 *in = (Int24 *)inBuffer; scale = 1.0 / 8388607.5; for (unsigned int i=0; i>= 8; + out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8); + //out[info.outOffset[j]] >>= 8; } in += info.inJump; out += info.outJump; @@ -7799,10 +10062,10 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } } else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; + Int24 *in = (Int24 *)inBuffer; for (unsigned int i=0; i> 8) & 0x0000ffff); + out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8); } in += info.inJump; out += info.outJump; @@ -7863,10 +10126,10 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } } else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; + Int24 *in = (Int24 *)inBuffer; for (unsigned int i=0; i> 16) & 0x000000ff); + out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16); } in += info.inJump; out += info.outJump; @@ -7905,14 +10168,14 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info } } - //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); } - //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } - //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } +//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); } +//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } +//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) { - register char val; - register char *ptr; + char val; + char *ptr; ptr = buffer; if ( format == RTAUDIO_SINT16 ) { @@ -7926,8 +10189,7 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat ptr += 2; } } - else if ( format == RTAUDIO_SINT24 || - format == RTAUDIO_SINT32 || + else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 ) { for ( unsigned int i=0; i