From 9b02537e82866bcfdcb22a5e7fd9c7f5217c2c64 Mon Sep 17 00:00:00 2001 From: Chris Kiefer Date: Tue, 23 Aug 2011 16:59:47 +0100 Subject: maximillian basic examples --- .DS_Store | Bin 0 -> 6148 bytes .gitignore | 4 + README | 28 + RtAudio.cpp | 7987 ++++++++++++++++++++++++++++++ RtAudio.h | 980 ++++ RtError.h | 60 + beat2.wav | Bin 0 -> 211140 bytes main.cpp | 33 + maximilian.cpp | 803 +++ maximilian.h | 316 ++ maximilianTest.xcodeproj/project.pbxproj | 286 ++ maximilian_examples/.DS_Store | Bin 0 -> 6148 bytes maximilian_examples/1.TestTone.cpp | 14 + maximilian_examples/10.Filters.cpp | 33 + maximilian_examples/11.Mixing.cpp | 40 + maximilian_examples/12.SamplePlayer.cpp | 21 + maximilian_examples/2.TwoTones.cpp | 14 + maximilian_examples/3.AM1.cpp | 15 + maximilian_examples/4.AM2.cpp | 16 + maximilian_examples/5.FM1.cpp | 14 + maximilian_examples/6.FM2.cpp | 14 + maximilian_examples/7.Counting.cpp | 15 + maximilian_examples/8.Counting2.cpp | 24 + maximilian_examples/9.Envelopes.cpp | 33 + player.cpp | 153 + player.h | 11 + portaudio.h | 1134 +++++ 27 files changed, 12048 insertions(+) create mode 100644 .DS_Store create mode 100644 .gitignore create mode 100644 RtAudio.cpp create mode 100644 RtAudio.h create mode 100644 RtError.h create mode 100644 beat2.wav create mode 100755 main.cpp create mode 100644 maximilian.cpp create mode 100755 maximilian.h create mode 100755 maximilianTest.xcodeproj/project.pbxproj create mode 100644 maximilian_examples/.DS_Store create mode 100755 maximilian_examples/1.TestTone.cpp create mode 100755 maximilian_examples/10.Filters.cpp create mode 100755 maximilian_examples/11.Mixing.cpp create mode 100755 maximilian_examples/12.SamplePlayer.cpp create mode 100755 maximilian_examples/2.TwoTones.cpp create mode 100755 maximilian_examples/3.AM1.cpp create mode 100755 maximilian_examples/4.AM2.cpp create mode 100755 maximilian_examples/5.FM1.cpp create mode 100755 maximilian_examples/6.FM2.cpp create mode 100755 maximilian_examples/7.Counting.cpp create mode 100755 maximilian_examples/8.Counting2.cpp create mode 100755 maximilian_examples/9.Envelopes.cpp create mode 100644 player.cpp create mode 100644 player.h create mode 100644 portaudio.h diff --git a/.DS_Store b/.DS_Store new file mode 100644 index 0000000..707b116 Binary files /dev/null and b/.DS_Store differ diff --git a/.gitignore b/.gitignore new file mode 100644 index 0000000..493301e --- /dev/null +++ b/.gitignore @@ -0,0 +1,4 @@ +*.pbxuser +*.mode1v3 +*.mode2v3 + diff --git a/README b/README index e69de29..7cbbf4b 100644 --- a/README +++ b/README @@ -0,0 +1,28 @@ + .__ .__.__ .__ + _____ _____ ___ __|__| _____ |__| | |__|____ ____ + / \\__ \ \ \/ / |/ \| | | | \__ \ / \ +| Y Y \/ __ \_> <| | Y Y \ | |_| |/ __ \| | \ +|__|_| (____ /__/\_ \__|__|_| /__|____/__(____ /___| / + \/ \/ \/ \/ \/ \/ + + +:::::::::::::::::::::::::::BASIC EXAMPLES + +You can choose between using RTAudio and PortAudio drivers in player.h by uncommenting the appropriate line. To use PortAudio, you will need to compile the portAudio library from http://http://www.portaudio.com/ and link it with your executable. + +Examples demonstrating different features can be found in the maximilian_examples folder. To try them, replace the contents of main.cpp with the contents of a tutorial file and compile. + + +:::MAC OSX XCODE PROJECT + +You can run the examples using the 'maximilianTest' XCode 3 project provided. + + +:::COMMAND LINE COMPILATION IN MACOSX/LINUX etc + +> g++ -Wall -D__MACOSX_CORE__ -o maximilian main.cpp RtAudio.cpp player.cpp maximilian.cpp -framework CoreAudio -lpthread + +> ./maximilian + + + diff --git a/RtAudio.cpp b/RtAudio.cpp new file mode 100644 index 0000000..4a3966b --- /dev/null +++ b/RtAudio.cpp @@ -0,0 +1,7987 @@ +/************************************************************************/ +/*! \class RtAudio + \brief Realtime audio i/o C++ classes. + + 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. + + RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ + + RtAudio: realtime audio i/o C++ classes + Copyright (c) 2001-2011 Gary P. Scavone + + Permission is hereby granted, free of charge, to any person + obtaining a copy of this software and associated documentation files + (the "Software"), to deal in the Software without restriction, + including without limitation the rights to use, copy, modify, merge, + publish, distribute, sublicense, and/or sell copies of the Software, + and to permit persons to whom the Software is furnished to do so, + subject to the following conditions: + + The above copyright notice and this permission notice shall be + included in all copies or substantial portions of the Software. + + Any person wishing to distribute modifications to the Software is + asked to send the modifications to the original developer so that + they can be incorporated into the canonical version. This is, + however, not a binding provision of this license. + + THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. +*/ +/************************************************************************/ + +// RtAudio: Version 4.0.9 + +#include "RtAudio.h" +#include +#include +#include +#include + +// Static variable definitions. +const unsigned int RtApi::MAX_SAMPLE_RATES = 14; +const unsigned int RtApi::SAMPLE_RATES[] = { + 4000, 5512, 8000, 9600, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) + #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__) + // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#else + #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions + #define MUTEX_DESTROY(A) abs(*A) // dummy definitions +#endif + +// *************************************************** // +// +// RtAudio definitions. +// +// *************************************************** // + +void RtAudio :: getCompiledApi( std::vector &apis ) throw() +{ + apis.clear(); + + // The order here will control the order of RtAudio's API search in + // the constructor. +#if defined(__UNIX_JACK__) + apis.push_back( UNIX_JACK ); +#endif +#if defined(__LINUX_ALSA__) + apis.push_back( LINUX_ALSA ); +#endif +#if defined(__LINUX_OSS__) + apis.push_back( LINUX_OSS ); +#endif +#if defined(__WINDOWS_ASIO__) + apis.push_back( WINDOWS_ASIO ); +#endif +#if defined(__WINDOWS_DS__) + apis.push_back( WINDOWS_DS ); +#endif +#if defined(__MACOSX_CORE__) + apis.push_back( MACOSX_CORE ); +#endif +#if defined(__RTAUDIO_DUMMY__) + apis.push_back( RTAUDIO_DUMMY ); +#endif +} + +void RtAudio :: openRtApi( RtAudio::Api api ) +{ +#if defined(__UNIX_JACK__) + if ( api == UNIX_JACK ) + rtapi_ = new RtApiJack(); +#endif +#if defined(__LINUX_ALSA__) + if ( api == LINUX_ALSA ) + rtapi_ = new RtApiAlsa(); +#endif +#if defined(__LINUX_OSS__) + if ( api == LINUX_OSS ) + rtapi_ = new RtApiOss(); +#endif +#if defined(__WINDOWS_ASIO__) + if ( api == WINDOWS_ASIO ) + rtapi_ = new RtApiAsio(); +#endif +#if defined(__WINDOWS_DS__) + if ( api == WINDOWS_DS ) + rtapi_ = new RtApiDs(); +#endif +#if defined(__MACOSX_CORE__) + if ( api == MACOSX_CORE ) + rtapi_ = new RtApiCore(); +#endif +#if defined(__RTAUDIO_DUMMY__) + if ( api == RTAUDIO_DUMMY ) + rtapi_ = new RtApiDummy(); +#endif +} + +RtAudio :: RtAudio( RtAudio::Api api ) throw() +{ + rtapi_ = 0; + + if ( api != UNSPECIFIED ) { + // Attempt to open the specified API. + openRtApi( api ); + if ( rtapi_ ) return; + + // No compiled support for specified API value. Issue a debug + // warning and continue as if no API was specified. + std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; + } + + // Iterate through the compiled APIs and return as soon as we find + // one with at least one device or we reach the end of the list. + std::vector< RtAudio::Api > apis; + getCompiledApi( apis ); + for ( unsigned int i=0; igetDeviceCount() ) break; + } + + if ( rtapi_ ) return; + + // 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"; +} + +RtAudio :: ~RtAudio() throw() +{ + delete rtapi_; +} + +void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options ) +{ + return rtapi_->openStream( outputParameters, inputParameters, format, + sampleRate, bufferFrames, callback, + userData, options ); +} + +// *************************************************** // +// +// Public RtApi definitions (see end of file for +// private or protected utility functions). +// +// *************************************************** // + +RtApi :: RtApi() +{ + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; + stream_.apiHandle = 0; + stream_.userBuffer[0] = 0; + stream_.userBuffer[1] = 0; + MUTEX_INITIALIZE( &stream_.mutex ); + showWarnings_ = true; +} + +RtApi :: ~RtApi() +{ + MUTEX_DESTROY( &stream_.mutex ); +} + +void RtApi :: openStream( RtAudio::StreamParameters *oParams, + RtAudio::StreamParameters *iParams, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options ) +{ + if ( stream_.state != STREAM_CLOSED ) { + errorText_ = "RtApi::openStream: a stream is already open!"; + error( RtError::INVALID_USE ); + } + + 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 ); + } + + 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 ); + } + + if ( oParams == NULL && iParams == NULL ) { + errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; + error( RtError::INVALID_USE ); + } + + if ( formatBytes(format) == 0 ) { + errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; + error( RtError::INVALID_USE ); + } + + unsigned int nDevices = getDeviceCount(); + unsigned int oChannels = 0; + if ( oParams ) { + oChannels = oParams->nChannels; + if ( oParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: output device parameter value is invalid."; + error( RtError::INVALID_USE ); + } + } + + unsigned int iChannels = 0; + if ( iParams ) { + iChannels = iParams->nChannels; + if ( iParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: input device parameter value is invalid."; + error( RtError::INVALID_USE ); + } + } + + 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 ( iChannels > 0 ) { + + result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + if ( oChannels > 0 ) closeStream(); + error( RtError::SYSTEM_ERROR ); + } + } + + stream_.callbackInfo.callback = (void *) callback; + stream_.callbackInfo.userData = userData; + + if ( options ) options->numberOfBuffers = stream_.nBuffers; + stream_.state = STREAM_STOPPED; +} + +unsigned int RtApi :: getDefaultInputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +unsigned int RtApi :: getDefaultOutputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +void RtApi :: closeStream( void ) +{ + // MUST be implemented in subclasses! + 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 ) +{ + // MUST be implemented in subclasses! + return FAILURE; +} + +void RtApi :: tickStreamTime( void ) +{ + // Subclasses that do not provide their own implementation of + // getStreamTime should call this function once per buffer I/O to + // provide basic stream time support. + + stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); + +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +long RtApi :: getStreamLatency( void ) +{ + verifyStream(); + + long totalLatency = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + totalLatency = stream_.latency[0]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + totalLatency += stream_.latency[1]; + + return totalLatency; +} + +double RtApi :: getStreamTime( void ) +{ + verifyStream(); + +#if defined( HAVE_GETTIMEOFDAY ) + // Return a very accurate estimate of the stream time by + // adding in the elapsed time since the last tick. + struct timeval then; + struct timeval now; + + if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + return stream_.streamTime; + + gettimeofday( &now, NULL ); + then = stream_.lastTickTimestamp; + return stream_.streamTime + + ((now.tv_sec + 0.000001 * now.tv_usec) - + (then.tv_sec + 0.000001 * then.tv_usec)); +#else + return stream_.streamTime; +#endif +} + +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__MACOSX_CORE__) + +// The OS X CoreAudio API is designed to use a separate callback +// procedure for each of its audio devices. A single RtAudio duplex +// stream using two different devices is supported here, though it +// cannot be guaranteed to always behave correctly because we cannot +// synchronize these two callbacks. +// +// A property listener is installed for over/underrun information. +// However, no functionality is currently provided to allow property +// listeners to trigger user handlers because it is unclear what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. However, we do provide a flag +// to the client callback function to inform of an over/underrun. + +// A structure to hold various information related to the CoreAudio API +// implementation. +struct CoreHandle { + AudioDeviceID id[2]; // device ids +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceIOProcID procId[2]; +#endif + UInt32 iStream[2]; // device stream index (or first if using multiple) + UInt32 nStreams[2]; // number of streams to use + bool xrun[2]; + char *deviceBuffer; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + CoreHandle() + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiCore:: RtApiCore() +{ +#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) + // This is a largely undocumented but absolutely necessary + // requirement starting with OS-X 10.6. If not called, queries and + // updates to various audio device properties are not handled + // correctly. + CFRunLoopRef theRunLoop = NULL; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); + if ( result != noErr ) { + errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; + error( RtError::WARNING ); + } +#endif +} + +RtApiCore :: ~RtApiCore() +{ + // The subclass destructor gets called before the base class + // destructor, so close an existing stream before deallocating + // apiDeviceId memory. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiCore :: getDeviceCount( void ) +{ + // Find out how many audio devices there are, if any. + UInt32 dataSize; + AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; + error( RtError::WARNING ); + return 0; + } + + return dataSize / sizeof( AudioDeviceID ); +} + +unsigned int RtApiCore :: getDefaultInputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; + error( RtError::WARNING ); + return 0; + } + + dataSize *= nDevices; + AudioDeviceID deviceList[ nDevices ]; + property.mSelector = kAudioHardwarePropertyDevices; + 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 ); + return 0; + } + + for ( unsigned int i=0; i= nDevices ) { + errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; + error( RtError::WARNING ); + return info; + } + + AudioDeviceID id = deviceList[ device ]; + + // Get the device name. + info.name.erase(); + CFStringRef cfname; + dataSize = sizeof( CFStringRef ); + property.mSelector = kAudioObjectPropertyManufacturer; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + int length = CFStringGetLength(cfname); + char *mname = (char *)malloc(length * 3 + 1); + CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); + info.name.append( (const char *)mname, strlen(mname) ); + info.name.append( ": " ); + CFRelease( cfname ); + free(mname); + + property.mSelector = kAudioObjectPropertyName; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + length = CFStringGetLength(cfname); + char *name = (char *)malloc(length * 3 + 1); + CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); + info.name.append( (const char *)name, strlen(name) ); + CFRelease( cfname ); + free(name); + + // Get the output stream "configuration". + AudioBufferList *bufferList = nil; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + property.mScope = kAudioDevicePropertyScopeOutput; + // property.mElement = kAudioObjectPropertyElementWildcard; + dataSize = 0; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + 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 ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; + error( RtError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if ( result != noErr || dataSize == 0 ) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Get output channel information. + unsigned int i, nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // Get the input stream "configuration". + property.mScope = kAudioDevicePropertyScopeInput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + 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 ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; + error( RtError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Get input channel information. + nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // 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; + + // Probe the device sample rates. + bool isInput = false; + if ( info.outputChannels == 0 ) isInput = true; + + // Determine the supported sample rates. + property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; + if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != kAudioHardwareNoError || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + UInt32 nRanges = dataSize / sizeof( AudioValueRange ); + AudioValueRange rangeList[ nRanges ]; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); + if ( result != kAudioHardwareNoError ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + Float64 minimumRate = 100000000.0, maximumRate = 0.0; + for ( UInt32 i=0; i 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 ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // CoreAudio always uses 32-bit floating point data for PCM streams. + // Thus, any other "physical" formats supported by the device are of + // no interest to the client. + info.nativeFormats = RTAUDIO_FLOAT32; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + return info; +} + +OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* inNow, + const AudioBufferList* inInputData, + const AudioTimeStamp* inInputTime, + AudioBufferList* outOutputData, + const AudioTimeStamp* inOutputTime, + void* infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiCore *object = (RtApiCore *) info->object; + if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) + return kAudioHardwareUnspecifiedError; + else + return kAudioHardwareNoError; +} + +OSStatus xrunListener( AudioObjectID inDevice, + UInt32 nAddresses, + const AudioObjectPropertyAddress properties[], + void* handlePointer ) +{ + CoreHandle *handle = (CoreHandle *) handlePointer; + for ( UInt32 i=0; ixrun[1] = true; + else + handle->xrun[0] = true; + } + } + + return kAudioHardwareNoError; +} + +OSStatus rateListener( AudioObjectID inDevice, + UInt32 nAddresses, + const AudioObjectPropertyAddress properties[], + void* ratePointer ) +{ + + Float64 *rate = (Float64 *) ratePointer; + UInt32 dataSize = sizeof( Float64 ); + AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); + return kAudioHardwareNoError; +} + +bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; + return FAILURE; + } + + AudioDeviceID id = deviceList[ device ]; + + // Setup for stream mode. + bool isInput = false; + if ( mode == INPUT ) { + isInput = true; + property.mScope = kAudioDevicePropertyScopeInput; + } + else + property.mScope = kAudioDevicePropertyScopeOutput; + + // Get the stream "configuration". + AudioBufferList *bufferList = nil; + dataSize = 0; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; + return FAILURE; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Search for one or more streams that contain the desired number of + // channels. CoreAudio devices can have an arbitrary number of + // streams and each stream can have an arbitrary number of channels. + // For each stream, a single buffer of interleaved samples is + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; + bool foundStream = false; + + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels >= channels + offsetCounter ) { + firstStream = iStream; + channelOffset = offsetCounter; + foundStream = true; + break; + } + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. + if ( foundStream == false ) { + monoMode = true; + offsetCounter = firstChannel; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; + } + } + + free( bufferList ); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof( AudioValueRange ); + property.mSelector = kAudioDevicePropertyBufferFrameSizeRange; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; + else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; + + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. + UInt32 theSize = (UInt32) *bufferSize; + dataSize = sizeof( UInt32 ); + property.mSelector = kAudioDevicePropertyBufferFrameSize; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = theSize; + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + + // Try to set "hog" mode ... it's not clear to me this is working. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) { + pid_t hog_pid; + dataSize = sizeof( hog_pid ); + property.mSelector = kAudioDevicePropertyHogMode; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( hog_pid != getpid() ) { + hog_pid = getpid(); + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + } + + // Check and if necessary, change the sample rate for the device. + Float64 nominalRate; + dataSize = sizeof( Float64 ); + property.mSelector = kAudioDevicePropertyNominalSampleRate; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Only change the sample rate if off by more than 1 Hz. + if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) { + + // Set a property listener for the sample rate change + Float64 reportedRate = 0.0; + AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + nominalRate = (Float64) sampleRate; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Now wait until the reported nominal rate is what we just set. + UInt32 microCounter = 0; + while ( reportedRate != nominalRate ) { + microCounter += 5000; + if ( microCounter > 5000000 ) break; + usleep( 5000 ); + } + + // Remove the property listener. + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + + if ( microCounter > 5000000 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now set the stream format for all streams. Also, check the + // physical format of the device and change that if necessary. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + property.mSelector = kAudioStreamPropertyVirtualFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the sample rate and data format id. However, only make the + // change if the sample rate is not within 1.0 of the desired + // rate and the format is not linear pcm. + bool updateFormat = false; + if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { + description.mSampleRate = (Float64) sampleRate; + updateFormat = true; + } + + if ( description.mFormatID != kAudioFormatLinearPCM ) { + description.mFormatID = kAudioFormatLinearPCM; + updateFormat = true; + } + + if ( updateFormat ) { + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now check the physical format. + property.mSelector = kAudioStreamPropertyPhysicalFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + //std::cout << "Current physical stream format:" << std::endl; + //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; + //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; + //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; + //std::cout << " sample rate = " << description.mSampleRate << std::endl; + + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { + description.mFormatID = kAudioFormatLinearPCM; + //description.mSampleRate = (Float64) sampleRate; + AudioStreamBasicDescription testDescription = description; + UInt32 formatFlags; + + // We'll try higher bit rates first and then work our way down. + std::vector< std::pair > physicalFormats; + 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 ) ); + physicalFormats.push_back( std::pair( 24, formatFlags ) ); // 24-bit packed + formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); + physicalFormats.push_back( std::pair( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low + formatFlags |= kAudioFormatFlagIsAlignedHigh; + physicalFormats.push_back( std::pair( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair( 16, formatFlags ) ); + physicalFormats.push_back( std::pair( 8, formatFlags ) ); + + bool setPhysicalFormat = false; + for( unsigned int i=0; iflags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; + + // 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 ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our CoreHandle structure for the stream. + CoreHandle *handle = 0; + if ( stream_.apiHandle == 0 ) { + try { + handle = new CoreHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->condition, NULL ) ) { + errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + } + else + handle = (CoreHandle *) stream_.apiHandle; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; + handle->id[mode] = id; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); + memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + // If possible, we will make use of the CoreAudio stream buffers as + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { + + 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_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) { + if ( streamCount > 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) + // Only one callback procedure per device. + stream_.mode = DUPLEX; + else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() + result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; + errorText_ = errorStream_.str(); + goto error; + } + if ( stream_.mode == OUTPUT && mode == INPUT ) + stream_.mode = DUPLEX; + else + stream_.mode = mode; + } + + // Setup the device property listener for over/underload. + property.mSelector = kAudioDeviceProcessorOverload; + result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &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; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiCore :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + 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 ) + AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); +#endif + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + 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 ) + AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); +#endif + } + + 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; + } + + // Destroy pthread condition variable. + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiCore :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiCore::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + result = AudioDeviceStart( handle->id[0], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || + ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + + result = AudioDeviceStart( handle->id[1], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == noErr ) return; + error( RtError::SYSTEM_ERROR ); +} + +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 ); + return; + } + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + 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(); + goto unlock; + } + } + + 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(); + goto unlock; + } + } + + stream_.state = STREAM_STOPPED; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == noErr ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiCore :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ) +{ + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == true ) + stopStream(); + 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 + // draining stream or duplex mode AND the input/output devices are + // different AND this function is called for the input device. + if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { + 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 SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + if ( handle->nStreams[0] == 1 ) { + memset( outBufferList->mBuffers[handle->iStream[0]].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + else { // fill multiple streams with zeros + for ( unsigned int i=0; inStreams[0]; i++ ) { + memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); + } + } + } + else if ( handle->nStreams[0] == 1 ) { + if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer + convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], stream_.convertInfo[0] ); + } + else { // copy from user buffer + memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + } + else { // fill multiple streams + Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + inBuffer = (Float32 *) stream_.deviceBuffer; + } + + if ( stream_.deviceInterleaved[0] == false ) { // mono mode + UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, + (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); + } + } + else { // fill multiple multi-channel streams with interleaved data + UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; + Float32 *out, *in; + + bool inInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 inChannels = stream_.nUserChannels[0]; + if ( stream_.doConvertBuffer[0] ) { + inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + inChannels = stream_.nDeviceChannels[0]; + } + + if ( inInterleaved ) inOffset = 1; + else inOffset = stream_.bufferSize; + + channelsLeft = inChannels; + for ( unsigned int i=0; inStreams[0]; i++ ) { + in = inBuffer; + out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; + streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; + + outJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[0] > 0 ) { + streamChannels -= stream_.channelOffset[0]; + outJump = stream_.channelOffset[0]; + out += outJump; + } + + // Account for possible unfilled channels at end of the last stream + if ( streamChannels > channelsLeft ) { + outJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine input buffer offsets and skips + if ( inInterleaved ) { + inJump = inChannels; + in += inChannels - channelsLeft; + } + else { + inJump = 1; + in += (inChannels - channelsLeft) * inOffset; + } + + for ( unsigned int i=0; idrainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + AudioDeviceID inputDevice; + inputDevice = handle->id[1]; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { + + if ( handle->nStreams[1] == 1 ) { + if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer + convertBuffer( stream_.userBuffer[1], + (char *) inBufferList->mBuffers[handle->iStream[1]].mData, + stream_.convertInfo[1] ); + } + else { // copy to user buffer + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + } + } + else { // read from multiple streams + Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; + if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; + + if ( stream_.deviceInterleaved[1] == false ) { // mono mode + UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); + } + } + else { // read from multiple multi-channel streams + UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; + Float32 *out, *in; + + bool outInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 outChannels = stream_.nUserChannels[1]; + if ( stream_.doConvertBuffer[1] ) { + outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + outChannels = stream_.nDeviceChannels[1]; + } + + if ( outInterleaved ) outOffset = 1; + else outOffset = stream_.bufferSize; + + channelsLeft = outChannels; + for ( unsigned int i=0; inStreams[1]; i++ ) { + out = outBuffer; + in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; + streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; + + inJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[1] > 0 ) { + streamChannels -= stream_.channelOffset[1]; + inJump = stream_.channelOffset[1]; + in += inJump; + } + + // Account for possible unread channels at end of the last stream + if ( streamChannels > channelsLeft ) { + inJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine output buffer offsets and skips + if ( outInterleaved ) { + outJump = outChannels; + out += outChannels - channelsLeft; + } + else { + outJump = 1; + out += (outChannels - channelsLeft) * outOffset; + } + + for ( unsigned int i=0; i +#include +#include + +// A structure to hold various information related to the Jack API +// implementation. +struct JackHandle { + jack_client_t *client; + jack_port_t **ports[2]; + std::string deviceName[2]; + bool xrun[2]; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + 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 * ) {}; + +RtApiJack :: RtApiJack() +{ + // Nothing to do here. +#if !defined(__RTAUDIO_DEBUG__) + // Turn off Jack's internal error reporting. + jack_set_error_function( &jackSilentError ); +#endif +} + +RtApiJack :: ~RtApiJack() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiJack :: getDeviceCount( void ) +{ + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); + if ( client == 0 ) return 0; + + const char **ports; + std::string port, previousPort; + unsigned int nChannels = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nChannels ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon + 1 ); + if ( port != previousPort ) { + nDevices++; + previousPort = port; + } + } + } while ( ports[++nChannels] ); + free( ports ); + } + + jack_client_close( client ); + return nDevices; +} + +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption + jack_status_t *status = NULL; + 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 ); + return info; + } + + const char **ports; + std::string port, previousPort; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) info.name = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + // Get the current jack server sample rate. + info.sampleRates.clear(); + info.sampleRates.push_back( jack_get_sample_rate( client ) ); + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.outputChannels = nChannels; + } + + // Jack "output ports" equal RtAudio input channels. + nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.inputChannels = nChannels; + } + + if ( info.outputChannels == 0 && info.inputChannels == 0 ) { + jack_client_close(client); + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtError::WARNING ); + return info; + } + + // 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; + + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; + + // Jack 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; + + jack_client_close(client); + info.probed = true; + return info; +} + +int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + + return 0; +} + +// This function will be called by a spawned thread when the Jack +// 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 ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->closeStream(); + + pthread_exit( NULL ); +} +void jackShutdown( void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; + + // Check current stream state. If stopped, then we'll assume this + // was called as a result of a call to RtApiJack::stopStream (the + // deactivation of a client handle causes this function to be called). + // If not, we'll assume the Jack server is shutting down or some + // other problem occurred and we should close the stream. + if ( object->isStreamRunning() == false ) return; + + 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 ) +{ + JackHandle *handle = (JackHandle *) infoPointer; + + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; + + return 0; +} + +bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Look for jack server and try to become a client (only do once per stream). + jack_client_t *client = 0; + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + if ( options && !options->streamName.empty() ) + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); + else + client = jack_client_open( "RtApiJack", jackoptions, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtError::WARNING ); + return FAILURE; + } + } + else { + // The handle must have been created on an earlier pass. + client = handle->client; + } + + const char **ports; + std::string port, previousPort, deviceName; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) deviceName = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + unsigned long flag = JackPortIsInput; + if ( mode == INPUT ) flag = JackPortIsOutput; + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + } + + // Compare the jack ports for specified client to the requested number of channels. + if ( nChannels < (channels + firstChannel) ) { + errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check the jack server sample rate. + unsigned int jackRate = jack_get_sample_rate( client ); + if ( sampleRate != jackRate ) { + jack_client_close( client ); + errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = jackRate; + + // 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 ] ) ); + free( ports ); + + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; + + // Get the buffer size. The buffer size and number of buffers + // (periods) is set when the jack server is started. + stream_.bufferSize = (int) jack_get_buffer_size( client ); + *bufferSize = stream_.bufferSize; + + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our JackHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new JackHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; + goto error; + } + + if ( pthread_cond_init(&handle->condition, NULL) ) { + errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + handle->client = client; + } + handle->deviceName[mode] = deviceName; + + // 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_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + if ( mode == OUTPUT ) + bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + else { // mode == INPUT + bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); + 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_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate memory for the Jack ports (channels) identifiers. + handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); + if ( handle->ports[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; + goto error; + } + + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up the stream for output. + stream_.mode = DUPLEX; + else { + stream_.mode = mode; + jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); + jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); + } + + // Register our ports. + char label[64]; + if ( mode == OUTPUT ) { + for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); + } + } + else { + for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + } + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); + + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + + delete handle; + stream_.apiHandle = 0; + } + + 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; + } + + return FAILURE; +} + +void RtApiJack :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { + + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); + + jack_client_close( handle->client ); + } + + if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + pthread_cond_destroy( &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; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiJack :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK(&stream_.mutex); + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; + goto unlock; + } + + const char **ports; + + // Get the list of available ports. + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; + goto unlock; + } + + // Now make the port connections. Since RtAudio wasn't designed to + // allow the user to select particular channels of a device, we'll + // just open the first "nChannels" ports with offset. + for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting output ports!"; + goto unlock; + } + } + free(ports); + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; + goto unlock; + } + + // Now make the port connections. See note above. + for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting input ports!"; + goto unlock; + } + } + free(ports); + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK(&stream_.mutex); + + if ( result == 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +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 ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + } + + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; + + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiJack :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + 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 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 ) +{ + 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_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, jackStopStream, info ); + else + 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; + } + + // Invoke user callback first, to get fresh output data. + 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 ); + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // no buffer conversion + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } + } + + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + if ( stream_.doConvertBuffer[1] ) { + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); + } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + else { // no buffer conversion + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); + } + } + } + + unlock: + MUTEX_UNLOCK(&stream_.mutex); + + RtApi::tickStreamTime(); + return SUCCESS; +} + //******************** End of __UNIX_JACK__ *********************// +#endif + +#if defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS-X CoreAudio and Linux +// Jack. The primary constraint with ASIO is that it only allows +// access to a single driver at a time. Thus, it is not possible to +// have more than one simultaneous RtAudio stream. +// +// This implementation also requires a number of external ASIO files +// and a few global variables. The ASIO callback scheme does not +// allow for the passing of user data, so we must create a global +// pointer to our callbackInfo structure. +// +// On unix systems, we make use of a pthread condition variable. +// Since there is no equivalent in Windows, I hacked something based +// on information found in +// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. + +#include "asiosys.h" +#include "asio.h" +#include "iasiothiscallresolver.h" +#include "asiodrivers.h" +#include + +AsioDrivers drivers; +ASIOCallbacks asioCallbacks; +ASIODriverInfo driverInfo; +CallbackInfo *asioCallbackInfo; +bool asioXRun; + +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; + + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; + +// 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 ); + +RtApiAsio :: RtApiAsio() +{ + // ASIO cannot run on a multi-threaded appartment. You can call + // CoInitialize beforehand, but it must be for appartment threading + // (in which case, CoInitilialize will return S_FALSE here). + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( FAILED(hr) ) { + errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; + error( RtError::WARNING ); + } + coInitialized_ = true; + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} + +RtApiAsio :: ~RtApiAsio() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); +} + +unsigned int RtApiAsio :: getDeviceCount( void ) +{ + return (unsigned int) drivers.asioGetNumDev(); +} + +RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + // 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 ); + return info; + } + return devices_[ device ]; + } + + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + info.name = driverName; + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + info.outputChannels = outputChannels; + info.inputChannels = inputChannels; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; i 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + drivers.removeCurrentDriver(); + return info; +} + +void bufferSwitch( long index, ASIOBool processNow ) +{ + RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; + object->callbackEvent( index ); +} + +void RtApiAsio :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; isaveDeviceInfo(); + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // 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; + } + + 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; + } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + + // 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; + } + + // 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; + } + + // 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; + } + } + + // Determine the driver data type. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + if ( mode == OUTPUT ) channelInfo.isInput = false; + 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; + } + + // Assuming WINDOWS host is always little-endian. + stream_.doByteSwap[mode] = false; + stream_.userFormat = format; + stream_.deviceFormat[mode] = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + if ( channelInfo.type == ASIOSTFloat64MSB ) 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; + } + + // Set the buffer size. For a duplex stream, this will end up + // setting the buffer size based on the input constraints, which + // should be ok. + 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; + } + + 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; + + 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; + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // ASIO always uses non-interleaved buffers. + 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; + } + handle->bufferInfos = 0; + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + + // 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]; + handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + if ( handle->bufferInfos == NULL ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + ASIOBufferInfo *infos; + infos = handle->bufferInfos; + for ( i=0; iisInput = ASIOFalse; + infos->channelNum = i + stream_.channelOffset[0]; + infos->buffers[0] = infos->buffers[1] = 0; + } + for ( i=0; iisInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; + } + + // 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 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; + errorText_ = errorStream_.str(); + goto error; + } + buffersAllocated = true; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // 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_ = "RtApiAsio::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_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + 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 + 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 + } + else { + stream_.latency[0] = outputLatency; + stream_.latency[1] = inputLatency; + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + return SUCCESS; + + error: + if ( buffersAllocated ) + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } + + 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; + } + + return FAILURE; +} + +void RtApiAsio :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } + + 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_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +bool stopThreadCalled = false; + +void RtApiAsio :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + //MUTEX_LOCK( &stream_.mutex ); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; + errorText_ = errorStream_.str(); + goto unlock; + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + asioXRun = false; + + unlock: + //MUTEX_UNLOCK( &stream_.mutex ); + + stopThreadCalled = false; + + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); +} + +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 ); + 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 ); + } + } + + stream_.state = STREAM_STOPPED; + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } + + // MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAsio :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + // The following lines were commented-out because some behavior was + // noted where the device buffers need to be zeroed to avoid + // continuing sound, even when the device buffers are completely + // disposed. So now, calling abort is the same as calling stop. + // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // handle->drainCounter = 2; + 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 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 ) +{ + 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_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal if finished. + if ( handle->drainCounter > 3 ) { + 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 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && asioXRun == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + asioXRun = false; + } + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = 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(); + unsigned threadId; + stopThreadCalled = true; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } + + } + else { + + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); + } + + } + + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + + if (stream_.doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) + memcpy( &stream_.deviceBuffer[j++*bufferBytes], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + } + else { + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); + } + } + + unlock: + // The following call was suggested by Malte Clasen. While the API + // documentation indicates it should not be required, some device + // drivers apparently do not function correctly without it. + ASIOOutputReady(); + + // MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + return SUCCESS; +} + +void sampleRateChanged( ASIOSampleRate sRate ) +{ + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. + + RtApi *object = (RtApi *) asioCallbackInfo->object; + try { + object->stopStream(); + } + catch ( RtError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } + + 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 ) +{ + long ret = 0; + + switch( selector ) { + case kAsioSelectorSupported: + if ( value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver whether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; +} + +static const char* getAsioErrorString( ASIOError result ) +{ + struct Messages + { + ASIOError value; + const char*message; + }; + + static Messages m[] = + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; + + for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) + if ( m[i].value == result ) return m[i].message; + + 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 + +#include +#include +#include + +#if defined(__MINGW32__) + // missing from latest mingw winapi +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ +#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 + +#ifdef _MSC_VER // if Microsoft Visual C++ +#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. +#endif + +static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; +} + +// A structure to hold various information related to the DirectSound +// API implementation. +struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; + + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } +}; + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); + +static const char* getErrorString( int code ); + +extern "C" unsigned __stdcall callbackHandler( void *ptr ); + +struct DsDevice { + LPGUID id[2]; + bool validId[2]; + bool found; + std::string name; + + DsDevice() + : found(false) { validId[0] = false; validId[1] = false; } +}; + +std::vector< DsDevice > dsDevices; + +RtApiDs :: RtApiDs() +{ + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; +} + +RtApiDs :: ~RtApiDs() +{ + if ( coInitialized_ ) CoUninitialize(); // balanced call. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +// The DirectSound default output is always the first device. +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + return 0; +} + +// The DirectSound default input is always the first input device, +// which is the first capture device enumerated. +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + return 0; +} + +unsigned int RtApiDs :: getDeviceCount( void ) +{ + // Set query flag for previously found devices to false, so that we + // can check for any devices that have disappeared. + for ( unsigned int i=0; i indices; + for ( unsigned int i=0; i= dsDevices.size() ) { + errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + HRESULT result; + if ( dsDevices[ device ].validId[0] == false ) goto probeInput; + + LPDIRECTSOUND output; + DSCAPS outCaps; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto probeInput; + } + + 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; + } + + // 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] ); + } + + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + + output->Release(); + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + + if ( dsDevices[ device ].validId[1] == false ) { + info.name = dsDevices[ device ].name; + info.probed = true; + return info; + } + + probeInput: + + 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; + } + + 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; + } + + // Get input channel information. + info.inputChannels = inCaps.dwChannels; + + // 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 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; + + 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 ( 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 + + input->Release(); + + if ( info.inputChannels == 0 ) return info; + + // 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; + + if ( device == 0 ) info.isDefaultInput = true; + + // Copy name and return. + info.name = dsDevices[ device ].name; + info.probed = true; + return info; +} + +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 ) +{ + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + return FAILURE; + } + + 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 ( 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 ( 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; + } + } + + // 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; + + // Check the lower range of the user-specified buffer size and set + // (arbitrarily) to a lower bound of 32. + if ( *bufferSize < 32 ) *bufferSize = 32; + + // 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; + + // 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; + + void *ohandle = 0, *bhandle = 0; + HRESULT result; + if ( mode == OUTPUT ) { + + 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; + } + + 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; + } + + // 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 { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + 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; + + // 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; + } + + // 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; + + // 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; + } + + // 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; + } + + // 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; + } + } + + // 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; + } + + dsBufferSize = dsbcaps.dwBufferBytes; + + // 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 ); + + // 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; + } + + ohandle = (void *) output; + bhandle = (void *) buffer; + } + + if ( mode == INPUT ) { + + 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; + } + + 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; + } + + // 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; + + // 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; + + // 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; + + // 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; + } + + // 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; + } + + dsBufferSize = dscbcaps.dwBufferBytes; + + // 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. + + // 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; + } + + // 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; + } + + ohandle = (void *) input; + bhandle = (void *) buffer; + } + + // 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; + + // 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; + + // 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; + } + + 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; + } + } + + 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; + } + } + } + + // 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; + } + + // 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; + + 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; + + // 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; + } + + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); + } + return SUCCESS; + + 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; + } + + 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; + } + + return FAILURE; +} + +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + // 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; + } + + 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_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiDs :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + //MUTEX_LOCK( &stream_.mutex ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // 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 ); + + buffersRolling = false; + duplexPrerollBytes = 0; + + 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] ); + } + + 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; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + 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; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + + unlock: + // MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); +} + +void RtApiDs :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + /* + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + */ + + 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 ); + } + + stream_.state = STREAM_STOPPED; + + // 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; + } + + // 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; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; + + stream_.state = STREAM_STOPPED; + + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input 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[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // 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 input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } + + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + // MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); +} + +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +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; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // 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; + } + + /* + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + 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; + } + 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; + } + + 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; + + 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(); + 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 ); + } + + //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(); + error( RtError::SYSTEM_ERROR ); + } + 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(); + error( RtError::SYSTEM_ERROR ); + } + + // 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(); + error( RtError::SYSTEM_ERROR ); + } + + // 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(); + error( RtError::SYSTEM_ERROR ); + } + nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + + 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(); + error( RtError::SYSTEM_ERROR ); + } + + 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 ) { + // 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(); + error( RtError::SYSTEM_ERROR ); + } + + 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(); + error( RtError::SYSTEM_ERROR ); + } + + 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(); + error( RtError::SYSTEM_ERROR ); + } + 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; +} + +#include "tchar.h" + +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 + + return s; +} + +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ) +{ + bool *isInput = (bool *) lpContext; + + HRESULT hr; + bool validDevice = false; + if ( *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 = convertTChar( description ); + if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) + 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; } +}; + +extern "C" 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( RtError::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( RtError::WARNING ); + break; + } + if ( subdevice < 0 ) + break; + nDevices++; + } + nextcard: + snd_ctl_close( handle ); + snd_card_next( &card ); + } + + 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; + 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; + } + 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 ); + } + + 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; + } + 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 + 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 ); + + 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( RtError::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( RtError::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( RtError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); + + captureProbe: + // Now try for capture + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + 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; + } + + 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; + } + + // 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; + } + + 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; + } + 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( RtError::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( RtError::WARNING ); + return info; + } + + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i= 0 ) + sprintf( name, "hw:%s,%d", cardname, subdevice ); + 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 ); + } + + 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. + 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 ) << "."; + 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 ) { + 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; + } + +#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; + + // 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( RtError::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 ) { + 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 ); +#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; + } + + 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; + } + + return FAILURE; +} + +void RtApiAlsa :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + error( RtError::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; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + 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(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + 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; + } + } + } + + 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; + } + } + } + + 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 ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + //if ( stream_.state == STREAM_STOPPED ) { + // MUTEX_UNLOCK( &stream_.mutex ); + // return; + //} + + 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 ( ( 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; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + //if ( stream_.state == STREAM_STOPPED ) { + // MUTEX_UNLOCK( &stream_.mutex ); + // return; + //} + + 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; + } + } + + 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; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtError::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( RtError::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 ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + 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]; + } + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // 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(); + } + 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: + + 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; + } + + // 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 { + 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; + } + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +extern "C" void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_ALSA__ *********************// +#endif + + +#if defined(__LINUX_OSS__) + +#include +#include +#include +#include +#include "soundcard.h" +#include +#include + +extern "C" void *ossCallbackHandler(void * ptr); + +// A structure to hold various information related to the OSS API +// implementation. +struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; + pthread_cond_t runnable; + + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiOss :: RtApiOss() +{ + // Nothing to do here. +} + +RtApiOss :: ~RtApiOss() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +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 ); + return 0; + } + + oss_sysinfo sysinfo; + if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return 0; + } + + close( mixerfd ); + return sysinfo.numaudios; +} + +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return info; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return info; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } + + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Probe channels + if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; + if ( ainfo.caps & PCM_CAP_DUPLEX ) { + if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + } + + // Probe data formats ... do for input + unsigned long mask = ainfo.iformats; + if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) + info.nativeFormats |= RTAUDIO_SINT16; + if ( mask & AFMT_S8 ) + info.nativeFormats |= RTAUDIO_SINT8; + if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) + info.nativeFormats |= RTAUDIO_SINT32; + if ( mask & AFMT_FLOAT ) + info.nativeFormats |= RTAUDIO_FLOAT32; + if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) + info.nativeFormats |= RTAUDIO_SINT24; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Probe the supported sample rates. + info.sampleRates.clear(); + if ( ainfo.nrates ) { + for ( unsigned int i=0; i= (int) SAMPLE_RATES[k] ) + info.sampleRates.push_back( 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 ); + } + else { + info.probed = true; + info.name = ainfo.name; + } + + return info; +} + + +bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; + return FAILURE; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check if device supports input or output + if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || + ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + int flags = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( mode == OUTPUT ) + flags |= O_WRONLY; + else { // mode == INPUT + if (stream_.mode == OUTPUT && stream_.device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close( handle->id[0] ); + handle->id[0] = 0; + if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // Check that the number previously set channels is the same. + if ( stream_.nUserChannels[0] != channels ) { + errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + flags |= O_RDWR; + } + else + flags |= O_RDONLY; + } + + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; + + // Try to open the device. + int fd; + fd = open( ainfo.devnode, flags, 0 ); + if ( fd == -1 ) { + if ( errno == EBUSY ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // For duplex operation, specifically set this mode (this doesn't seem to work). + /* + if ( flags | O_RDWR ) { + result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); + if ( result == -1) { + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + */ + + // Check the device channel support. + stream_.nUserChannels[mode] = channels; + if ( ainfo.max_channels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the number of channels. + int deviceChannels = channels + firstChannel; + result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); + if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = deviceChannels; + + // Get the data format mask + int mask; + result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + int deviceFormat = -1; + stream_.doByteSwap[mode] = false; + if ( format == RTAUDIO_SINT8 ) { + if ( mask & AFMT_S8 ) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT24 ) { + if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + } + + if ( deviceFormat == -1 ) { + // The user requested format is not natively supported by the device. + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S8) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + + if ( stream_.deviceFormat[mode] == 0 ) { + // This really shouldn't happen ... + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the data format. + int temp = deviceFormat; + result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); + if ( result == -1 || deviceFormat != temp ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; + if ( ossBufferBytes < 16 ) ossBufferBytes = 16; + int buffers = 0; + if ( options ) buffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; + if ( buffers < 2 ) buffers = 3; + temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); + result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nBuffers = buffers; + + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; + + // Set the sample rate. + int srate = sampleRate; + result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Verify the sample rate setup worked. + if ( abs( srate - sampleRate ) > 100 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = sampleRate; + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { + // We're doing duplex setup here. + stream_.deviceFormat[0] = stream_.deviceFormat[1]; + stream_.nDeviceChannels[0] = deviceChannels; + } + + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + + // 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 stream handles if necessary and then save. + if ( stream_.apiHandle == 0 ) { + try { + handle = new OssHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + handle->id[mode] = fd; + + // 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_ = "RtApiOss::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_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + 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; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority. The higher priority will only take affect if the + // program is run as root or suid. + 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 ); +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiOss::error creating callback thread!"; + goto error; + } + } + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + 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; + } + + return FAILURE; +} + +void RtApiOss :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + else + ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + stream_.state = STREAM_STOPPED; + } + + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + 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_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiOss :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + stream_.state = STREAM_RUNNING; + + // No need to do anything else here ... OSS automatically starts + // when fed samples. + + MUTEX_UNLOCK( &stream_.mutex ); + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); +} + +void RtApiOss :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + memset( buffer, 0, samples * formatBytes(format) ); + for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); + if ( result == -1 ) { + errorText_ = "RtApiOss::stopStream: audio write error."; + error( RtError::WARNING ); + } + } + + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiOss :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiOss :: callbackEvent() +{ + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } + + // Invoke user callback to get fresh output data. + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.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; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int samples; + RtAudioFormat format; + + 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] ); + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( buffer, samples, format ); + + if ( stream_.mode == DUPLEX && handle->triggered == false ) { + int trig = 0; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + handle->triggered = true; + } + else + // Write samples to device. + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an underrun, though there isn't a + // specific means for determining that. + handle->xrun[0] = true; + errorText_ = "RtApiOss::callbackEvent: audio write error."; + error( RtError::WARNING ); + // Continue on to input section. + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + samples = stream_.bufferSize * stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device. + result = read( handle->id[1], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an overrun, though there isn't a + // specific means for determining that. + handle->xrun[1] = true; + errorText_ = "RtApiOss::callbackEvent: audio read error."; + error( RtError::WARNING ); + goto unlock; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +extern "C" void *ossCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_OSS__ *********************// +#endif + + +// *************************************************** // +// +// Protected common (OS-independent) RtAudio methods. +// +// *************************************************** // + +// This method can be modified to control the behavior of error +// message printing. +void RtApi :: error( RtError::Type type ) +{ + errorStream_.str(""); // clear the ostringstream + if ( type == RtError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else if ( type != RtError::WARNING ) + throw( RtError( errorText_, type ) ); +} + +void RtApi :: verifyStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtError::INVALID_USE ); + } +} + +void RtApi :: clearStreamInfo() +{ + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; + stream_.sampleRate = 0; + stream_.bufferSize = 0; + stream_.nBuffers = 0; + stream_.userFormat = 0; + stream_.userInterleaved = true; + stream_.streamTime = 0.0; + stream_.apiHandle = 0; + stream_.deviceBuffer = 0; + stream_.callbackInfo.callback = 0; + stream_.callbackInfo.userData = 0; + stream_.callbackInfo.isRunning = false; + for ( int i=0; i<2; i++ ) { + stream_.device[i] = 11111; + stream_.doConvertBuffer[i] = false; + stream_.deviceInterleaved[i] = true; + stream_.doByteSwap[i] = false; + stream_.nUserChannels[i] = 0; + stream_.nDeviceChannels[i] = 0; + stream_.channelOffset[i] = 0; + stream_.deviceFormat[i] = 0; + stream_.latency[i] = 0; + stream_.userBuffer[i] = 0; + stream_.convertInfo[i].channels = 0; + stream_.convertInfo[i].inJump = 0; + stream_.convertInfo[i].outJump = 0; + stream_.convertInfo[i].inFormat = 0; + stream_.convertInfo[i].outFormat = 0; + stream_.convertInfo[i].inOffset.clear(); + stream_.convertInfo[i].outOffset.clear(); + } +} + +unsigned int RtApi :: formatBytes( RtAudioFormat format ) +{ + if ( format == RTAUDIO_SINT16 ) + return 2; + else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || + format == RTAUDIO_FLOAT32 ) + return 4; + else if ( format == RTAUDIO_FLOAT64 ) + return 8; + else if ( format == RTAUDIO_SINT8 ) + return 1; + + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtError::WARNING ); + + return 0; +} + +void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) +{ + if ( mode == INPUT ) { // convert device to user buffer + stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; + stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; + stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; + stream_.convertInfo[mode].outFormat = stream_.userFormat; + } + else { // convert user to device buffer + stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; + stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; + stream_.convertInfo[mode].inFormat = stream_.userFormat; + stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; + } + + if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; + else + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; + + // Set up the interleave/deinterleave offsets. + if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { + if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || + ( mode == INPUT && stream_.userInterleaved ) ) { + for ( int k=0; k 0 ) { + if ( stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; k>= 8; + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x00ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 24) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i>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; + + ptr = buffer; + if ( format == RTAUDIO_SINT16 ) { + for ( unsigned int i=0; i +#include +#include "RtError.h" + +/*! \typedef typedef unsigned long RtAudioFormat; + \brief RtAudio data format type. + + Support for signed integers and floats. Audio data fed to/from an + RtAudio stream is assumed to ALWAYS be in host byte order. The + internal routines will automatically take care of any necessary + byte-swapping between the host format and the soundcard. Thus, + endian-ness is not a concern in the following format definitions. + Note that 24-bit data is expected to be encapsulated in a 32-bit + format. + + - \e RTAUDIO_SINT8: 8-bit signed integer. + - \e RTAUDIO_SINT16: 16-bit signed integer. + - \e RTAUDIO_SINT24: Lower 3 bytes of 32-bit signed integer. + - \e RTAUDIO_SINT32: 32-bit signed integer. + - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. + - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0. +*/ +typedef unsigned long RtAudioFormat; +static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // Lower 3 bytes of 32-bit signed integer. +static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. +static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. +static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0. + +/*! \typedef typedef unsigned long RtAudioStreamFlags; + \brief RtAudio stream option flags. + + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). + - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. + - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). + + By default, RtAudio streams pass and receive audio data from the + client in an interleaved format. By passing the + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows Direct Sound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. + + If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt + to select realtime scheduling (round-robin) for the callback thread. + + If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to + open the "default" PCM device when using the ALSA API. Note that this + will override any specified input or output device id. +*/ +typedef unsigned int RtAudioStreamFlags; +static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). +static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. +static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. +static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. +static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only). + +/*! \typedef typedef unsigned long RtAudioStreamStatus; + \brief RtAudio stream status (over- or underflow) flags. + + Notification of a stream over- or underflow is indicated by a + non-zero stream \c status argument in the RtAudioCallback function. + The stream status can be one of the following two options, + depending on whether the stream is open for output and/or input: + + - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. + - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. +*/ +typedef unsigned int RtAudioStreamStatus; +static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. +static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. + +//! RtAudio callback function prototype. +/*! + All RtAudio clients must create a function of type RtAudioCallback + to read and/or write data from/to the audio stream. When the + underlying audio system is ready for new input or output data, this + function will be invoked. + + \param outputBuffer For output (or duplex) streams, the client + should write \c nFrames of audio sample frames into this + buffer. This argument should be recast to the datatype + specified when the stream was opened. For input-only + streams, this argument will be NULL. + + \param inputBuffer For input (or duplex) streams, this buffer will + hold \c nFrames of input audio sample frames. This + argument should be recast to the datatype specified when the + stream was opened. For output-only streams, this argument + will be NULL. + + \param nFrames The number of sample frames of input or output + data in the buffers. The actual buffer size in bytes is + dependent on the data type and number of channels in use. + + \param streamTime The number of seconds that have elapsed since the + stream was started. + + \param status If non-zero, this argument indicates a data overflow + or underflow condition for the stream. The particular + condition can be determined by comparison with the + RtAudioStreamStatus flags. + + \param userData A pointer to optional data provided by the client + when opening the stream (default = NULL). + + To continue normal stream operation, the RtAudioCallback function + should return a value of zero. To stop the stream and drain the + output buffer, the function should return a value of one. To abort + the stream immediately, the client should return a value of two. + */ +typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, + unsigned int nFrames, + double streamTime, + RtAudioStreamStatus status, + void *userData ); + + +// **************************************************************** // +// +// RtAudio class declaration. +// +// RtAudio is a "controller" used to select an available audio i/o +// interface. It presents a common API for the user to call but all +// functionality is implemented by the class RtApi and its +// subclasses. RtAudio creates an instance of an RtApi subclass +// based on the user's API choice. If no choice is made, RtAudio +// attempts to make a "logical" API selection. +// +// **************************************************************** // + +class RtApi; + +class RtAudio +{ + public: + + //! Audio API specifier arguments. + enum Api { + UNSPECIFIED, /*!< Search for a working compiled API. */ + LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ + LINUX_OSS, /*!< The Linux Open Sound System API. */ + UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ + MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ + WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ + WINDOWS_DS, /*!< The Microsoft Direct Sound API. */ + RTAUDIO_DUMMY /*!< A compilable but non-functional API. */ + }; + + //! The public device information structure for returning queried values. + struct DeviceInfo { + bool probed; /*!< true if the device capabilities were successfully probed. */ + std::string name; /*!< Character string device identifier. */ + unsigned int outputChannels; /*!< Maximum output channels supported by device. */ + unsigned int inputChannels; /*!< Maximum input channels supported by device. */ + unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */ + bool isDefaultOutput; /*!< true if this is the default output device. */ + bool isDefaultInput; /*!< true if this is the default input device. */ + std::vector sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ + RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */ + + // Default constructor. + DeviceInfo() + :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0), + isDefaultOutput(false), isDefaultInput(false), nativeFormats(0) {} + }; + + //! The structure for specifying input or ouput stream parameters. + struct StreamParameters { + unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */ + unsigned int nChannels; /*!< Number of channels. */ + unsigned int firstChannel; /*!< First channel index on device (default = 0). */ + + // Default constructor. + StreamParameters() + : deviceId(0), nChannels(0), firstChannel(0) {} + }; + + //! The structure for specifying stream options. + /*! + The following flags can be OR'ed together to allow a client to + make changes to the default stream behavior: + + - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). + - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. + - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread. + - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). + + By default, RtAudio streams pass and receive audio data from the + client in an interleaved format. By passing the + RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio + data will instead be presented in non-interleaved buffers. In + this case, each buffer argument in the RtAudioCallback function + will point to a single array of data, with \c nFrames samples for + each channel concatenated back-to-back. For example, the first + sample of data for the second channel would be located at index \c + nFrames (assuming the \c buffer pointer was recast to the correct + data type for the stream). + + Certain audio APIs offer a number of parameters that influence the + I/O latency of a stream. By default, RtAudio will attempt to set + these parameters internally for robust (glitch-free) performance + (though some APIs, like Windows Direct Sound, make this difficult). + By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() + function, internal stream settings will be influenced in an attempt + to minimize stream latency, though possibly at the expense of stream + performance. + + If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to + open the input and/or output stream device(s) for exclusive use. + Note that this is not possible with all supported audio APIs. + + If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt + to select realtime scheduling (round-robin) for the callback thread. + The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME + flag is set. It defines the thread's realtime priority. + + If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to + open the "default" PCM device when using the ALSA API. Note that this + will override any specified input or output device id. + + The \c numberOfBuffers parameter can be used to control stream + latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs + only. A value of two is usually the smallest allowed. Larger + numbers can potentially result in more robust stream performance, + though likely at the cost of stream latency. The value set by the + user is replaced during execution of the RtAudio::openStream() + function by the value actually used by the system. + + The \c streamName parameter can be used to set the client name + when using the Jack API. By default, the client name is set to + RtApiJack. However, if you wish to create multiple instances of + RtAudio with Jack, each instance must have a unique client name. + */ + struct StreamOptions { + RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */ + unsigned int numberOfBuffers; /*!< Number of stream buffers. */ + std::string streamName; /*!< A stream name (currently used only in Jack). */ + int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */ + + // Default constructor. + StreamOptions() + : flags(0), numberOfBuffers(0), priority(0) {} + }; + + //! A static function to determine the available compiled audio APIs. + /*! + The values returned in the std::vector can be compared against + the enumerated list values. Note that there can be more than one + API compiled for certain operating systems. + */ + static void getCompiledApi( std::vector &apis ) throw(); + + //! The class constructor. + /*! + The constructor performs minor initialization tasks. No exceptions + can be thrown. + + If no API argument is specified and multiple API support has been + compiled, the default order of use is JACK, ALSA, OSS (Linux + systems) and ASIO, DS (Windows systems). + */ + RtAudio( RtAudio::Api api=UNSPECIFIED ) throw(); + + //! The destructor. + /*! + If a stream is running or open, it will be stopped and closed + automatically. + */ + ~RtAudio() throw(); + + //! Returns the audio API specifier for the current instance of RtAudio. + RtAudio::Api getCurrentApi( void ) throw(); + + //! A public function that queries for the number of audio devices available. + /*! + This function performs a system query of available devices each time it + is called, thus supporting devices connected \e after instantiation. If + a system error occurs during processing, a warning will be issued. + */ + unsigned int getDeviceCount( void ) throw(); + + //! Return an RtAudio::DeviceInfo structure for a specified device number. + /*! + + Any device integer between 0 and getDeviceCount() - 1 is valid. + If an invalid argument is provided, an RtError (type = INVALID_USE) + will be thrown. If a device is busy or otherwise unavailable, the + structure member "probed" will have a value of "false" and all + other members are undefined. If the specified device is the + current default input or output device, the corresponding + "isDefault" member will have a value of "true". + */ + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + + //! A function that returns the index of the default output device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultOutputDevice( void ) throw(); + + //! A function that returns the index of the default input device. + /*! + If the underlying audio API does not provide a "default + device", or if no devices are available, the return value will be + 0. Note that this is a valid device identifier and it is the + client's responsibility to verify that a device is available + before attempting to open a stream. + */ + unsigned int getDefaultInputDevice( void ) throw(); + + //! A public function for opening a stream with the specified parameters. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if a stream cannot be + opened with the specified parameters or an error occurs during + processing. An RtError (type = INVALID_USE) is thrown if any + invalid device ID or channel number parameters are specified. + + \param outputParameters Specifies output stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For input-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param inputParameters Specifies input stream parameters to use + when opening a stream, including a device ID, number of channels, + and starting channel number. For output-only streams, this + argument should be NULL. The device ID is an index value between + 0 and getDeviceCount() - 1. + \param format An RtAudioFormat specifying the desired sample data format. + \param sampleRate The desired sample rate (sample frames per second). + \param *bufferFrames A pointer to a value indicating the desired + internal buffer size in sample frames. The actual value + used by the device is returned via the same pointer. A + value of zero can be specified, in which case the lowest + allowable value is determined. + \param callback A client-defined function that will be invoked + when input data is available and/or output data is needed. + \param userData An optional pointer to data that can be accessed + from within the callback function. + \param options An optional pointer to a structure containing various + global stream options, including a list of OR'ed RtAudioStreamFlags + and a suggested number of stream buffers that can be used to + control stream latency. More buffers typically result in more + robust performance, though at a cost of greater latency. If a + value of zero is specified, a system-specific median value is + chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the + lowest allowable value is used. The actual value used is + returned via the structure argument. The parameter is API dependent. + */ + void openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, RtAudioCallback callback, + void *userData = NULL, RtAudio::StreamOptions *options = NULL ); + + //! A function that closes a stream and frees any associated stream memory. + /*! + If a stream is not open, this function issues a warning and + returns (no exception is thrown). + */ + void closeStream( void ) throw(); + + //! A function that starts a stream. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + running. + */ + void startStream( void ); + + //! Stop a stream, allowing any samples remaining in the output queue to be played. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void stopStream( void ); + + //! Stop a stream, discarding any samples remaining in the input/output queue. + /*! + An RtError (type = SYSTEM_ERROR) is thrown if an error occurs + during processing. An RtError (type = INVALID_USE) is thrown if a + stream is not open. A warning is issued if the stream is already + stopped. + */ + void abortStream( void ); + + //! Returns true if a stream is open and false if not. + bool isStreamOpen( void ) const throw(); + + //! Returns true if the stream is running and false if it is stopped or not open. + bool isStreamRunning( void ) const throw(); + + //! Returns the number of elapsed seconds since the stream was started. + /*! + If a stream is not open, an RtError (type = INVALID_USE) will be thrown. + */ + double getStreamTime( void ); + + //! Returns the internal stream latency in sample frames. + /*! + The stream latency refers to delay in audio input and/or output + caused by internal buffering by the audio system and/or hardware. + For duplex streams, the returned value will represent the sum of + the input and output latencies. If a stream is not open, an + RtError (type = INVALID_USE) will be thrown. If the API does not + report latency, the return value will be zero. + */ + long getStreamLatency( void ); + + //! Returns actual sample rate in use by the stream. + /*! + On some systems, the sample rate used may be slightly different + than that specified in the stream parameters. If a stream is not + open, an RtError (type = INVALID_USE) will be thrown. + */ + unsigned int getStreamSampleRate( void ); + + //! Specify whether warning messages should be printed to stderr. + void showWarnings( bool value = true ) throw(); + + protected: + + void openRtApi( RtAudio::Api api ); + RtApi *rtapi_; +}; + +// Operating system dependent thread functionality. +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) + #include + #include + + typedef unsigned long ThreadHandle; + typedef CRITICAL_SECTION StreamMutex; + +#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // Using pthread library for various flavors of unix. + #include + + typedef pthread_t ThreadHandle; + typedef pthread_mutex_t StreamMutex; + +#else // Setup for "dummy" behavior + + #define __RTAUDIO_DUMMY__ + typedef int ThreadHandle; + typedef int StreamMutex; + +#endif + +// This global structure type is used to pass callback information +// between the private RtAudio stream structure and global callback +// handling functions. +struct CallbackInfo { + void *object; // Used as a "this" pointer. + ThreadHandle thread; + void *callback; + void *userData; + void *apiInfo; // void pointer for API specific callback information + bool isRunning; + + // Default constructor. + CallbackInfo() + :object(0), callback(0), userData(0), apiInfo(0), isRunning(false) {} +}; + +// **************************************************************** // +// +// RtApi class declaration. +// +// Subclasses of RtApi contain all API- and OS-specific code necessary +// to fully implement the RtAudio API. +// +// Note that RtApi is an abstract base class and cannot be +// explicitly instantiated. The class RtAudio will create an +// instance of an RtApi subclass (RtApiOss, RtApiAlsa, +// RtApiJack, RtApiCore, RtApiAl, RtApiDs, or RtApiAsio). +// +// **************************************************************** // + +#if defined( HAVE_GETTIMEOFDAY ) + #include +#endif + +#include + +class RtApi +{ +public: + + RtApi(); + virtual ~RtApi(); + virtual RtAudio::Api getCurrentApi( void ) = 0; + virtual unsigned int getDeviceCount( void ) = 0; + virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; + virtual unsigned int getDefaultInputDevice( void ); + virtual unsigned int getDefaultOutputDevice( void ); + void openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, RtAudioCallback callback, + void *userData, RtAudio::StreamOptions *options ); + virtual void closeStream( void ); + virtual void startStream( void ) = 0; + virtual void stopStream( void ) = 0; + virtual void abortStream( void ) = 0; + long getStreamLatency( void ); + unsigned int getStreamSampleRate( void ); + virtual double getStreamTime( void ); + bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }; + bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }; + void showWarnings( bool value ) { showWarnings_ = value; }; + + +protected: + + static const unsigned int MAX_SAMPLE_RATES; + static const unsigned int SAMPLE_RATES[]; + + enum { FAILURE, SUCCESS }; + + enum StreamState { + STREAM_STOPPED, + STREAM_RUNNING, + STREAM_CLOSED = -50 + }; + + enum StreamMode { + OUTPUT, + INPUT, + DUPLEX, + UNINITIALIZED = -75 + }; + + // A protected structure used for buffer conversion. + struct ConvertInfo { + int channels; + int inJump, outJump; + RtAudioFormat inFormat, outFormat; + std::vector inOffset; + std::vector outOffset; + }; + + // A protected structure for audio streams. + struct RtApiStream { + unsigned int device[2]; // Playback and record, respectively. + void *apiHandle; // void pointer for API specific stream handle information + StreamMode mode; // OUTPUT, INPUT, or DUPLEX. + StreamState state; // STOPPED, RUNNING, or CLOSED + char *userBuffer[2]; // Playback and record, respectively. + char *deviceBuffer; + bool doConvertBuffer[2]; // Playback and record, respectively. + bool userInterleaved; + bool deviceInterleaved[2]; // Playback and record, respectively. + bool doByteSwap[2]; // Playback and record, respectively. + unsigned int sampleRate; + unsigned int bufferSize; + unsigned int nBuffers; + unsigned int nUserChannels[2]; // Playback and record, respectively. + unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. + unsigned int channelOffset[2]; // Playback and record, respectively. + unsigned long latency[2]; // Playback and record, respectively. + RtAudioFormat userFormat; + RtAudioFormat deviceFormat[2]; // Playback and record, respectively. + StreamMutex mutex; + CallbackInfo callbackInfo; + ConvertInfo convertInfo[2]; + double streamTime; // Number of elapsed seconds since the stream started. + +#if defined(HAVE_GETTIMEOFDAY) + struct timeval lastTickTimestamp; +#endif + + RtApiStream() + :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } + }; + + typedef signed short Int16; + typedef signed int Int32; + typedef float Float32; + typedef double Float64; + + std::ostringstream errorStream_; + std::string errorText_; + bool showWarnings_; + RtApiStream stream_; + + /*! + Protected, api-specific method that attempts to open a device + with the given parameters. This function MUST be implemented by + all subclasses. If an error is encountered during the probe, a + "warning" message is reported and FAILURE is returned. A + successful probe is indicated by a return value of SUCCESS. + */ + virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); + + //! A protected function used to increment the stream time. + void tickStreamTime( void ); + + //! Protected common method to clear an RtApiStream structure. + void clearStreamInfo(); + + /*! + Protected common method that throws an RtError (type = + INVALID_USE) if a stream is not open. + */ + void verifyStream( void ); + + //! Protected common error method to allow global control over error handling. + void error( RtError::Type type ); + + /*! + Protected method used to perform format, channel number, and/or interleaving + conversions between the user and device buffers. + */ + void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); + + //! Protected common method used to perform byte-swapping on buffers. + void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); + + //! Protected common method that returns the number of bytes for a given format. + unsigned int formatBytes( RtAudioFormat format ); + + //! Protected common method that sets up the parameters for buffer conversion. + void setConvertInfo( StreamMode mode, unsigned int firstChannel ); +}; + +// **************************************************************** // +// +// Inline RtAudio definitions. +// +// **************************************************************** // + +inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); } +inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); } +inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } +inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); } +inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); } +inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); } +inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } +inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } +inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } +inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); } +inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); } +inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } +inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }; +inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } +inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); } + +// RtApi Subclass prototypes. + +#if defined(__MACOSX_CORE__) + +#include + +class RtApiCore: public RtApi +{ +public: + + RtApiCore(); + ~RtApiCore(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); + static const char* getErrorCode( OSStatus code ); +}; + +#endif + +#if defined(__UNIX_JACK__) + +class RtApiJack: public RtApi +{ +public: + + RtApiJack(); + ~RtApiJack(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( unsigned long nframes ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__WINDOWS_ASIO__) + +class RtApiAsio: public RtApi +{ +public: + + RtApiAsio(); + ~RtApiAsio(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + bool callbackEvent( long bufferIndex ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool coInitialized_; + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__WINDOWS_DS__) + +class RtApiDs: public RtApi +{ +public: + + RtApiDs(); + ~RtApiDs(); + RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; }; + unsigned int getDeviceCount( void ); + unsigned int getDefaultOutputDevice( void ); + unsigned int getDefaultInputDevice( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + long getStreamLatency( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + bool coInitialized_; + bool buffersRolling; + long duplexPrerollBytes; + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__LINUX_ALSA__) + +class RtApiAlsa: public RtApi +{ +public: + + RtApiAlsa(); + ~RtApiAlsa(); + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__LINUX_OSS__) + +class RtApiOss: public RtApi +{ +public: + + RtApiOss(); + ~RtApiOss(); + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + +#if defined(__RTAUDIO_DUMMY__) + +class RtApiDummy: public RtApi +{ +public: + + RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtError::WARNING ); }; + RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; }; + unsigned int getDeviceCount( void ) { return 0; }; + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) { RtAudio::DeviceInfo info; return info; }; + void closeStream( void ) {}; + void startStream( void ) {}; + void stopStream( void ) {}; + void abortStream( void ) {}; + + private: + + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) { return false; }; +}; + +#endif + +#endif + +// Indentation settings for Vim and Emacs +// +// Local Variables: +// c-basic-offset: 2 +// indent-tabs-mode: nil +// End: +// +// vim: et sts=2 sw=2 diff --git a/RtError.h b/RtError.h new file mode 100644 index 0000000..a64f434 --- /dev/null +++ b/RtError.h @@ -0,0 +1,60 @@ +/************************************************************************/ +/*! \class RtError + \brief Exception handling class for RtAudio & RtMidi. + + The RtError class is quite simple but it does allow errors to be + "caught" by RtError::Type. See the RtAudio and RtMidi + documentation to know which methods can throw an RtError. + +*/ +/************************************************************************/ + +#ifndef RTERROR_H +#define RTERROR_H + +#include +#include +#include + +class RtError : public std::exception +{ + public: + //! Defined RtError types. + enum Type { + WARNING, /*!< A non-critical error. */ + DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ + UNSPECIFIED, /*!< The default, unspecified error type. */ + NO_DEVICES_FOUND, /*!< No devices found on system. */ + INVALID_DEVICE, /*!< An invalid device ID was specified. */ + MEMORY_ERROR, /*!< An error occured during memory allocation. */ + INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ + INVALID_USE, /*!< The function was called incorrectly. */ + DRIVER_ERROR, /*!< A system driver error occured. */ + SYSTEM_ERROR, /*!< A system error occured. */ + THREAD_ERROR /*!< A thread error occured. */ + }; + + //! The constructor. + RtError( const std::string& message, Type type = RtError::UNSPECIFIED ) throw() : message_(message), type_(type) {} + + //! The destructor. + virtual ~RtError( void ) throw() {} + + //! Prints thrown error message to stderr. + virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; } + + //! Returns the thrown error message type. + virtual const Type& getType(void) const throw() { return type_; } + + //! Returns the thrown error message string. + virtual const std::string& getMessage(void) const throw() { return message_; } + + //! Returns the thrown error message as a c-style string. + virtual const char* what( void ) const throw() { return message_.c_str(); } + + protected: + std::string message_; + Type type_; +}; + +#endif diff --git a/beat2.wav b/beat2.wav new file mode 100644 index 0000000..c71cee6 Binary files /dev/null and b/beat2.wav differ diff --git a/main.cpp b/main.cpp new file mode 100755 index 0000000..b80b512 --- /dev/null +++ b/main.cpp @@ -0,0 +1,33 @@ +#include "maximilian.h" + +maxiOsc myCounter,mySwitchableOsc;// +int CurrentCount;// +double myOscOutput,myFilteredOutput;// +double myEnvelopeData[6] = {500,0,1000,500,0,500};//this data will be used to make an envelope. Value and time to value in ms. +maxiEnvelope myEnvelope; +maxiFilter myFilter; + +void setup() {//some inits + myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope +} + +void play(double *output) { + + CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value. + + if (CurrentCount<5)//simple if statement + + myOscOutput=mySwitchableOsc.square(CurrentCount*100); + + else if (CurrentCount>=5)//and the 'else' bit. + + myOscOutput=mySwitchableOsc.saw(CurrentCount*50);//one osc object can produce whichever waveform you want. + + if (CurrentCount==1) + + myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope + + myFilteredOutput=myFilter.lores(myOscOutput,(myEnvelope.line(6, myEnvelopeData)),10);//lores takes an audio input, a frequency and a resonance factor (1-100) + + *output=myFilteredOutput;//point me at your speakers and fire. +} diff --git a/maximilian.cpp b/maximilian.cpp new file mode 100644 index 0000000..ac2ccba --- /dev/null +++ b/maximilian.cpp @@ -0,0 +1,803 @@ + +/* + * maximilian.cpp + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#include "maximilian.h" +#include "math.h" + + +//int channels=2; +//int samplerate=44100; +//int buffersize=1024; +float chandiv= 1; + +int maxiSettings::sampleRate = 44100; +int maxiSettings::channels = 2; +int maxiSettings::bufferSize = 1024; + +double sineBuffer[514]={0,0.012268,0.024536,0.036804,0.049042,0.06131,0.073547,0.085785,0.097992,0.1102,0.12241,0.13455,0.1467,0.15884,0.17093,0.18301,0.19507,0.20709,0.21909,0.23105,0.24295,0.25485,0.26669,0.2785,0.29025,0.30197,0.31366,0.32529,0.33685,0.34839,0.35986,0.37128,0.38266,0.39395,0.40521,0.41641,0.42752,0.4386,0.44958,0.46051,0.47137,0.48215,0.49286,0.50351,0.51407,0.52457,0.53497,0.54529,0.55554,0.5657,0.57578,0.58575,0.59567,0.60547,0.6152,0.62482,0.63437,0.6438,0.65314,0.66238,0.67151,0.68057,0.68951,0.69833,0.70706,0.7157,0.72421,0.7326,0.74091,0.74908,0.75717,0.76514,0.77298,0.7807,0.7883,0.79581,0.80316,0.81042,0.81754,0.82455,0.83142,0.8382,0.84482,0.85132,0.8577,0.86392,0.87006,0.87604,0.88187,0.8876,0.89319,0.89862,0.90396,0.90912,0.91415,0.91907,0.92383,0.92847,0.93295,0.93729,0.9415,0.94556,0.94949,0.95325,0.95691,0.96039,0.96375,0.96692,0.97,0.9729,0.97565,0.97827,0.98074,0.98306,0.98523,0.98724,0.98914,0.99084,0.99243,0.99387,0.99515,0.99628,0.99725,0.99808,0.99875,0.99927,0.99966,0.99988,0.99997,0.99988,0.99966,0.99927,0.99875,0.99808,0.99725,0.99628,0.99515,0.99387,0.99243,0.99084,0.98914,0.98724,0.98523,0.98306,0.98074,0.97827,0.97565,0.9729,0.97,0.96692,0.96375,0.96039,0.95691,0.95325,0.94949,0.94556,0.9415,0.93729,0.93295,0.92847,0.92383,0.91907,0.91415,0.90912,0.90396,0.89862,0.89319,0.8876,0.88187,0.87604,0.87006,0.86392,0.8577,0.85132,0.84482,0.8382,0.83142,0.82455,0.81754,0.81042,0.80316,0.79581,0.7883,0.7807,0.77298,0.76514,0.75717,0.74908,0.74091,0.7326,0.72421,0.7157,0.70706,0.69833,0.68951,0.68057,0.67151,0.66238,0.65314,0.6438,0.63437,0.62482,0.6152,0.60547,0.59567,0.58575,0.57578,0.5657,0.55554,0.54529,0.53497,0.52457,0.51407,0.50351,0.49286,0.48215,0.47137,0.46051,0.44958,0.4386,0.42752,0.41641,0.40521,0.39395,0.38266,0.37128,0.35986,0.34839,0.33685,0.32529,0.31366,0.30197,0.29025,0.2785,0.26669,0.25485,0.24295,0.23105,0.21909,0.20709,0.19507,0.18301,0.17093,0.15884,0.1467,0.13455,0.12241,0.1102,0.097992,0.085785,0.073547,0.06131,0.049042,0.036804,0.024536,0.012268,0,-0.012268,-0.024536,-0.036804,-0.049042,-0.06131,-0.073547,-0.085785,-0.097992,-0.1102,-0.12241,-0.13455,-0.1467,-0.15884,-0.17093,-0.18301,-0.19507,-0.20709,-0.21909,-0.23105,-0.24295,-0.25485,-0.26669,-0.2785,-0.29025,-0.30197,-0.31366,-0.32529,-0.33685,-0.34839,-0.35986,-0.37128,-0.38266,-0.39395,-0.40521,-0.41641,-0.42752,-0.4386,-0.44958,-0.46051,-0.47137,-0.48215,-0.49286,-0.50351,-0.51407,-0.52457,-0.53497,-0.54529,-0.55554,-0.5657,-0.57578,-0.58575,-0.59567,-0.60547,-0.6152,-0.62482,-0.63437,-0.6438,-0.65314,-0.66238,-0.67151,-0.68057,-0.68951,-0.69833,-0.70706,-0.7157,-0.72421,-0.7326,-0.74091,-0.74908,-0.75717,-0.76514,-0.77298,-0.7807,-0.7883,-0.79581,-0.80316,-0.81042,-0.81754,-0.82455,-0.83142,-0.8382,-0.84482,-0.85132,-0.8577,-0.86392,-0.87006,-0.87604,-0.88187,-0.8876,-0.89319,-0.89862,-0.90396,-0.90912,-0.91415,-0.91907,-0.92383,-0.92847,-0.93295,-0.93729,-0.9415,-0.94556,-0.94949,-0.95325,-0.95691,-0.96039,-0.96375,-0.96692,-0.97,-0.9729,-0.97565,-0.97827,-0.98074,-0.98306,-0.98523,-0.98724,-0.98914,-0.99084,-0.99243,-0.99387,-0.99515,-0.99628,-0.99725,-0.99808,-0.99875,-0.99927,-0.99966,-0.99988,-0.99997,-0.99988,-0.99966,-0.99927,-0.99875,-0.99808,-0.99725,-0.99628,-0.99515,-0.99387,-0.99243,-0.99084,-0.98914,-0.98724,-0.98523,-0.98306,-0.98074,-0.97827,-0.97565,-0.9729,-0.97,-0.96692,-0.96375,-0.96039,-0.95691,-0.95325,-0.94949,-0.94556,-0.9415,-0.93729,-0.93295,-0.92847,-0.92383,-0.91907,-0.91415,-0.90912,-0.90396,-0.89862,-0.89319,-0.8876,-0.88187,-0.87604,-0.87006,-0.86392,-0.8577,-0.85132,-0.84482,-0.8382,-0.83142,-0.82455,-0.81754,-0.81042,-0.80316,-0.79581,-0.7883,-0.7807,-0.77298,-0.76514,-0.75717,-0.74908,-0.74091,-0.7326,-0.72421,-0.7157,-0.70706,-0.69833,-0.68951,-0.68057,-0.67151,-0.66238,-0.65314,-0.6438,-0.63437,-0.62482,-0.6152,-0.60547,-0.59567,-0.58575,-0.57578,-0.5657,-0.55554,-0.54529,-0.53497,-0.52457,-0.51407,-0.50351,-0.49286,-0.48215,-0.47137,-0.46051,-0.44958,-0.4386,-0.42752,-0.41641,-0.40521,-0.39395,-0.38266,-0.37128,-0.35986,-0.34839,-0.33685,-0.32529,-0.31366,-0.30197,-0.29025,-0.2785,-0.26669,-0.25485,-0.24295,-0.23105,-0.21909,-0.20709,-0.19507,-0.18301,-0.17093,-0.15884,-0.1467,-0.13455,-0.12241,-0.1102,-0.097992,-0.085785,-0.073547,-0.06131,-0.049042,-0.036804,-0.024536,-0.012268,0,0.012268 +}; + +void setup();//use this to do any initialisation if you want. + +void play(double *channels);//run dac! + +maxiOsc::maxiOsc(){ + phase = 0.0; +// memset(phases,0,500); +// memset(freqs,0,500); +} + +double maxiOsc::noise() { + //always the same unless you seed it. + float r = rand()/(float)RAND_MAX; + output=r*2-1; + return(output); +} + +double maxiOsc::sinewave(double frequency) { + output=sin (phase*(TWOPI)); + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(maxiSettings::sampleRate/(frequency))); + return(output); + +} + +double maxiOsc::sinebuf4(double frequency) { + double remainder; + double a,b,c,d,a1,a2,a3; + phase += 512./(maxiSettings::sampleRate/(frequency)); + if ( phase >= 511 ) phase -=512; + remainder = phase - floor(phase); + + if (phase==0) { + a=sineBuffer[(long) 512]; + b=sineBuffer[(long) phase]; + c=sineBuffer[(long) phase+1]; + d=sineBuffer[(long) phase+2]; + + } else { + a=sineBuffer[(long) phase-1]; + b=sineBuffer[(long) phase]; + c=sineBuffer[(long) phase+1]; + d=sineBuffer[(long) phase+2]; + + } + + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = double (((a3 * remainder + a2) * remainder + a1) * remainder + b); + return(output); +} + +double maxiOsc::sinebuf(double frequency) { + double remainder; + phase += 512./(maxiSettings::sampleRate/(frequency*chandiv)); + if ( phase >= 511 ) phase -=512; + remainder = phase - floor(phase); + output = (double) ((1-remainder) * sineBuffer[1+ (long) phase] + remainder * sineBuffer[2+(long) phase]); + return(output); +} + +double maxiOsc::coswave(double frequency) { + output=cos (phase*(TWOPI)); + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(maxiSettings::sampleRate/(frequency))); + return(output); + +} + +double maxiOsc::phasor(double frequency) { + output=phase; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(maxiSettings::sampleRate/(frequency))); + return(output); +} + +double maxiOsc::square(double frequency) { + if (phase<0.5) output=-1; + if (phase>0.5) output=1; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(maxiSettings::sampleRate/(frequency))); + return(output); +} + +double maxiOsc::pulse(double frequency, double duty) { + if (duty<0.) duty=0; + if (duty>1.) duty=1; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(maxiSettings::sampleRate/(frequency))); + if (phaseduty) output=1.; + return(output); +} + +double maxiOsc::phasor(double frequency, double startphase, double endphase) { + output=phase; + if (phase= endphase ) phase = startphase; + phase += ((endphase-startphase)/(maxiSettings::sampleRate/(frequency))); + return(output); +} + + +double maxiOsc::saw(double frequency) { + + output=phase; + if ( phase >= 1.0 ) phase -= 2.0; + phase += (1./(maxiSettings::sampleRate/(frequency))); + return(output); + +} + +double maxiOsc::triangle(double frequency, double phase) { + output=tri*2; + if ( phase >= 1.0 ) phase -= 1.0; + phase += (1./(maxiSettings::sampleRate/(frequency*chandiv))); + if (phase <= 0.5 ) { + tri = phase; + } else { + tri =(1-phase); + } + return(output); + +} + +//I like this. +double maxiEnvelope::line(int numberofsegments,double segments[1000]) { + period=2./(segments[valindex+1]*0.004); + nextval=segments[valindex+2]; + currentval=segments[valindex]; + if (currentval-amplitude > 0.0000001 && valindex < numberofsegments) { + amplitude += ((currentval-startval)/(maxiSettings::sampleRate/period)); + } else if (currentval-amplitude < -0.0000001 && valindex < numberofsegments) { + amplitude -= (((currentval-startval)*(-1))/(maxiSettings::sampleRate/period)); + } else if (valindex >numberofsegments-1) { + valindex=numberofsegments-2; + } else { + valindex=valindex+2; + startval=currentval; + } + output=amplitude; + return(output); +} + +//and this +void maxiEnvelope::trigger(int index, double amp) { + valindex=index; + amplitude=amp; + +} + +//and this +double maxiDelayline::dl(double input, int size, double feedback) { + if ( phase >=size ) phase = 0; + memory[phase]=(memory[phase]*feedback)+(input*feedback)*0.5; + phase+=1; + output=memory[phase+1]; + return(output); + +} + +double maxiDelayline::dl(double input, int size, double feedback, int position) { + if ( phase >=size ) phase = 0; + if ( position >=size ) position = 0; + memory[phase]=(memory[phase]*feedback)+(input*feedback)*chandiv; + phase+=1; + output=memory[position]; + return(output); + +} + +//I particularly like these. cutoff between 0 and 1 +double maxiFilter::lopass(double input, double cutoff) { + output=outputs[0] + cutoff*(input-outputs[0]); + outputs[0]=output; + return(output); +} +//as above +double maxiFilter::hipass(double input, double cutoff) { + output=input-(outputs[0] + cutoff*(input-outputs[0])); + outputs[0]=output; + return(output); +} +//awesome. cuttof is freq in hz. res is between 1 and whatever. Watch out! +double maxiFilter::lores(double input,double cutoff1, double resonance) { + cutoff=cutoff1*0.5; + if (cutoff<10) cutoff=10; + if (cutoff>(maxiSettings::sampleRate*0.25)) cutoff=(maxiSettings::sampleRate*0.25); + if (resonance<1.) resonance = 1.; + z=cos(TWOPI*cutoff/maxiSettings::sampleRate); + c=2-2*z; + double r=(sqrt(2.0)*sqrt(-pow((z-1.0),3.0))+resonance*(z-1))/(resonance*(z-1)); + x=x+(input-y)*c; + y=y+x; + x=x*r; + output=y; + return(output); +} + +//working hires filter +double maxiFilter::hires(double input,double cutoff1, double resonance) { + cutoff=cutoff1*0.5; + if (cutoff<10) cutoff=10; + if (cutoff>(maxiSettings::sampleRate*0.25)) cutoff=(maxiSettings::sampleRate*0.25); + if (resonance<1.) resonance = 1.; + z=cos(TWOPI*cutoff/maxiSettings::sampleRate); + c=2-2*z; + double r=(sqrt(2.0)*sqrt(-pow((z-1.0),3.0))+resonance*(z-1))/(resonance*(z-1)); + x=x+(input-y)*c; + y=y+x; + x=x*r; + output=input-y; + return(output); +} + +//This works a bit. Needs attention. +double maxiFilter::bandpass(double input,double cutoff1, double resonance) { + cutoff=cutoff1; + if (cutoff>(maxiSettings::sampleRate*0.5)) cutoff=(maxiSettings::sampleRate*0.5); + if (resonance>=1.) resonance=0.999999; + z=cos(TWOPI*cutoff/maxiSettings::sampleRate); + inputs[0] = (1-resonance)*(sqrt(resonance*(resonance-4.0*pow(z,2.0)+2.0)+1)); + inputs[1] = 2*z*resonance; + inputs[2] = pow((resonance*-1),2); + + output=inputs[0]*input+inputs[1]*outputs[1]+inputs[2]*outputs[2]; + outputs[2]=outputs[1]; + outputs[1]=output; + return(output); +} + +//stereo bus +double *maxiMix::stereo(double input,double two[2],double x) { + if (x>1) x=1; + if (x<0) x=0; + two[0]=input*sqrt(1.0-x); + two[1]=input*sqrt(x); + return(two); +} + +//quad bus +double *maxiMix::quad(double input,double four[4],double x,double y) { + if (x>1) x=1; + if (x<0) x=0; + if (y>1) y=1; + if (y<0) y=0; + four[0]=input*sqrt((1.0-x)*y); + four[1]=input*sqrt((1.0-x)*(1.0-y)); + four[2]=input*sqrt(x*y); + four[3]=input*sqrt(x*(1.0-y)); + return(four); +} + +//ambisonic bus +double *maxiMix::ambisonic(double input,double eight[8],double x,double y,double z) { + if (x>1) x=1; + if (x<0) x=0; + if (y>1) y=1; + if (y<0) y=0; + if (z>1) y=1; + if (z<0) y=0; + eight[0]=input*(sqrt((1.0-x)*y)*1.0-z); + eight[1]=input*(sqrt((1.0-x)*(1.0-y))*1.0-z); + eight[2]=input*(sqrt(x*y)*1.0-z); + eight[3]=input*(sqrt(x*(1.0-y))*1.0-z); + eight[4]=input*(sqrt((1.0-x)*y)*z); + eight[5]=input*(sqrt((1.0-x)*(1.0-y))*z); + eight[6]=input*sqrt((x*y)*z); + eight[7]=input*sqrt((x*(1.0-y))*z); + return(eight); +} + + +bool maxiSample::load(string fileName) { + myPath = fileName; + return read(); +} + +void maxiSample::trigger() { + position = 0; +} + +bool maxiSample::read() +{ + bool result; + ifstream inFile( myPath.c_str(), ios::in | ios::binary); + result = inFile; + if (inFile) { + bool datafound = false; + inFile.seekg(4, ios::beg); + inFile.read( (char*) &myChunkSize, 4 ); // read the ChunkSize + + inFile.seekg(16, ios::beg); + inFile.read( (char*) &mySubChunk1Size, 4 ); // read the SubChunk1Size + + //inFile.seekg(20, ios::beg); + inFile.read( (char*) &myFormat, sizeof(short) ); // read the file format. This should be 1 for PCM + + //inFile.seekg(22, ios::beg); + inFile.read( (char*) &myChannels, sizeof(short) ); // read the # of channels (1 or 2) + + //inFile.seekg(24, ios::beg); + inFile.read( (char*) &mySampleRate, sizeof(int) ); // read the samplerate + + //inFile.seekg(28, ios::beg); + inFile.read( (char*) &myByteRate, sizeof(int) ); // read the byterate + + //inFile.seekg(32, ios::beg); + inFile.read( (char*) &myBlockAlign, sizeof(short) ); // read the blockalign + + //inFile.seekg(34, ios::beg); + inFile.read( (char*) &myBitsPerSample, sizeof(short) ); // read the bitspersample + + //ignore any extra chunks + char chunkID[5]=""; + chunkID[4] = 0; + int filePos = 36; + while(!datafound && !inFile.eof()) { + inFile.seekg(filePos, ios::beg); + inFile.read((char*) &chunkID, sizeof(char) * 4); + inFile.seekg(filePos + 4, ios::beg); + inFile.read( (char*) &myDataSize, sizeof(int) ); // read the size of the data + filePos += 8; + if (strcmp(chunkID,"data") == 0) { + datafound = true; + }else{ + filePos += myDataSize; + } + } + + // read the data chunk + myData = (char*) malloc(myDataSize * sizeof(char)); + inFile.seekg(filePos, ios::beg); + inFile.read(myData, myDataSize); + length=myDataSize*(0.5/myChannels); + inFile.close(); // close the input file + + }else { + cout << "ERROR: Could not load sample: " <length) position=0; + output = (double) ((1-remainder) * buffer[1+ (long) position] + remainder * buffer[2+(long) position])/32767;//linear interpolation + return(output); +} + +double maxiSample::playOnce() { +// long length=myDataSize*(0.5/myChannels); + double remainder; + short* buffer = (short *)myData; + position=(position+1); + remainder = position - (long) position; + if ((long) position=0) { + + if ((long) position>=length-1) position=1; + remainder = position - floor(position); + if (position+1=0) { + a=position-1; + + } + else { + a=0; + } + if (position-2>=0) { + b=position-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation + } + return(output); +} + +double maxiSample::play(double frequency, double start, double end) { + return play(frequency, start, end, position); +} + +double maxiSample::play(double frequency, double start, double end, double &pos) { + double remainder; + // long length=myDataSize; + + if (end>=length) end=length-1; + long a,b; + short* buffer = (short *)myData; + if (frequency >0.) { + if (pos= end ) pos = start; + pos += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv))); + remainder = pos - floor(pos); + long posl = floor(pos); + if (posl+1=0) { + a=posl-1; + } + else { + a=0; + } + if (posl-2>=0) { + b=posl-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + + remainder * buffer[b])/32767;//linear interpolation + + } + + return(output); +} + + +//better cubic inerpolation. Cobbled together from various (pd externals, yehar, other places). +double maxiSample::play4(double frequency, double start, double end) { + double remainder; + double a,b,c,d,a1,a2,a3; + short* buffer = (short*)myData; + if (frequency >0.) { + if (position= end ) position = start; + position += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv))); + remainder = position - floor(position); + if (position>0) { + a=buffer[(int)(floor(position))-1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (positionstart && position < end-1) { + a=buffer[(long) position+1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (position>start) { + c=buffer[(long) position-1]; + + } else { + c=buffer[0]; + + } + if (position>start+1) { + d=buffer[(long) position-2]; + + } else { + d=buffer[0]; + } + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = (double) (((a3 * remainder + a2) * -remainder + a1) * -remainder + b) / 32767; + + } + + return(output); +} + +double maxiSample::bufferPlay(unsigned char &bufferin,long length) { + double remainder; + short* buffer = (short *)bufferin; + position=(position+1); + remainder = position - (long) position; + if ((long) position>length) position=0; + output = (double) ((1-remainder) * buffer[1+ (long) position] + remainder * buffer[2+(long) position])/32767;//linear interpolation + return(output); +} + +double maxiSample::bufferPlay(unsigned char &bufferin,double speed,long length) { + double remainder; + long a,b; + short* buffer = (short *)bufferin; + position=position+((speed*chandiv*myChannels)/(maxiSettings::sampleRate/mySampleRate)); + if (speed >=0) { + + if ((long) position>=length-1) position=1; + remainder = position - floor(position); + if (position+1=0) { + a=position-1; + + } + else { + a=0; + } + if (position-2>=0) { + b=position-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + remainder * buffer[b])/32767;//linear interpolation + } + return(output); +} + +double maxiSample::bufferPlay(unsigned char &bufferin,double frequency, double start, double end) { + double remainder; + length=end; + long a,b; + short* buffer = (short *)bufferin; + if (frequency >0.) { + if (position= end ) position = start; + position += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv))); + remainder = position - floor(position); + long pos = floor(position); + if (pos+1=0) { + a=pos-1; + } + else { + a=0; + } + if (pos-2>=0) { + b=pos-2; + } + else { + b=0; + } + output = (double) ((-1-remainder) * buffer[a] + + remainder * buffer[b])/32767;//linear interpolation + + } + + return(output); +} + +//better cubic inerpolation. Cobbled together from various (pd externals, yehar, other places). +double maxiSample::bufferPlay4(unsigned char &bufferin,double frequency, double start, double end) { + double remainder; + double a,b,c,d,a1,a2,a3; + short* buffer = (short*)bufferin; + if (frequency >0.) { + if (position= end ) position = start; + position += ((end-start)/(maxiSettings::sampleRate/(frequency*chandiv))); + remainder = position - floor(position); + if (position>0) { + a=buffer[(int)(floor(position))-1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (positionstart && position < end-1) { + a=buffer[(long) position+1]; + + } else { + a=buffer[0]; + + } + + b=buffer[(long) position]; + if (position>start) { + c=buffer[(long) position-1]; + + } else { + c=buffer[0]; + + } + if (position>start+1) { + d=buffer[(long) position-2]; + + } else { + d=buffer[0]; + } + a1 = 0.5f * (c - a); + a2 = a - 2.5 * b + 2.f * c - 0.5f * d; + a3 = 0.5f * (d - a) + 1.5f * (b - c); + output = (double) (((a3 * remainder + a2) * -remainder + a1) * -remainder + b) / 32767; + + } + + return(output); +} + + +void maxiSample::getLength() { + length=myDataSize*0.5; +} + + diff --git a/maximilian.h b/maximilian.h new file mode 100755 index 0000000..3304bac --- /dev/null +++ b/maximilian.h @@ -0,0 +1,316 @@ +/* + * maximilian.h + * platform independent synthesis library using portaudio or rtaudio + * + * Created by Mick Grierson on 29/12/2009. + * Copyright 2009 Mick Grierson & Strangeloop Limited. All rights reserved. + * Thanks to the Goldsmiths Creative Computing Team. + * Special thanks to Arturo Castro for the PortAudio implementation. + * + * Permission is hereby granted, free of charge, to any person + * obtaining a copy of this software and associated documentation + * files (the "Software"), to deal in the Software without + * restriction, including without limitation the rights to use, + * copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following + * conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES + * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND + * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT + * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, + * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING + * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + */ + +#ifndef MAXIMILIAN_H +#define MAXIMILIAN_H + +//#define MAXIMILIAN_PORTAUDIO +#define MAXIMILIAN_RT_AUDIO + +//#include "ofMain.h" +//#include "ofUtils.h" + + +#include +#include +#include +#include +#include "math.h" + +using namespace std; +#define TWOPI (3.14159*2) + +class maxiSettings { +public: + static int sampleRate; + static int channels; + static int bufferSize; + static void setup(int initSampleRate, int initChannels, int initBufferSize) { + maxiSettings::sampleRate = initSampleRate; + maxiSettings::channels = initChannels; + maxiSettings::bufferSize = initBufferSize; + } +}; + + +class maxiOsc { + + double frequency; + double phase; + double startphase; + double endphase; + double output; + double tri; + + +public: + maxiOsc(); + double sinewave(double frequency); + double coswave(double frequency); + double phasor(double frequency); + double phasor(double frequency, double startphase, double endphase); + double saw(double frequency); + double triangle(double frequency,double phase); + double square(double frequency); + double pulse(double frequency, double duty); + double noise(); + double sinebuf(double frequency); + double sinebuf4(double frequency); + +}; + + +class maxiEnvelope { + + double period; + double output; + double startval; + double currentval; + double nextval; + +public: + double line(int numberofsegments,double segments[100]); + void trigger(int index,double amp); + int valindex; + double amplitude; + +}; + + +class maxiDelayline { + double frequency; + int phase; + double startphase; + double endphase; + double output; + double memory[88200]; + +public: + double dl(double input, int size, double feedback); + double dl(double input, int size, double feedback, int position); + + +}; + + +class maxiFilter { + double gain; + double input; + double output; + double inputs[10]; + double outputs[10]; + double cutoff1; + double x;//speed + double y;//pos + double z;//pole + double c;//filter coefficient +public: + double cutoff; + double resonance; + double lores(double input,double cutoff1, double resonance); + double hires(double input,double cutoff1, double resonance); + double bandpass(double input,double cutoff1, double resonance); + double lopass(double input,double cutoff); + double hipass(double input,double cutoff); + +}; + +class maxiMix { + double input; + double two[2]; + double four[4]; + double eight[8]; +public: + double x; + double y; + double z; + double *stereo(double input,double two[2],double x); + double *quad(double input,double four[4], double x,double y); + double *ambisonic(double input,double eight[8],double x,double y, double z); + +}; + + +class maxiSample { + +private: + string myPath; + int myChunkSize; + int mySubChunk1Size; + short myFormat; + int myByteRate; + short myBlockAlign; + short myBitsPerSample; + int myDataSize; + double position; + double speed; + double output; + +public: + short myChannels; + int mySampleRate; + long length; + void getLength(); + + char* myData; + + // get/set for the Path property + + ~maxiSample() + { + delete myData; + myChunkSize = NULL; + mySubChunk1Size = NULL; + myFormat = NULL; + myChannels = NULL; + mySampleRate = NULL; + myByteRate = NULL; + myBlockAlign = NULL; + myBitsPerSample = NULL; + myDataSize = NULL; + } + +// maxiSample(); + + bool load(string fileName); + + void trigger(); + + // read a wav file into this class + bool read(); + + double play(); + + double playOnce(); + + double play(double speed); + + double play(double frequency, double start, double end, double &pos); + + double play(double frequency, double start, double end); + + double play4(double frequency, double start, double end); + + double bufferPlay(unsigned char &bufferin,long length); + + double bufferPlay(unsigned char &bufferin,double speed,long length); + + double bufferPlay(unsigned char &bufferin,double frequency, double start, double end); + + double bufferPlay4(unsigned char &bufferin,double frequency, double start, double end); + + //int open(char *filename) { + // // open the wav file + // char *path = new char[50]; + // strcpy(path, filename); + // sample *myWav = new sample(path); + // return 0; + // } + // // print a summary of the wav file + // int info() { + // char *summary = myWav->getSummary(); + // printf("Summary:\n%s", summary); + // return 0; + // } + // int savefile() { + // // write the summary back out + // strcpy(path, "testout.wav"); + // myWav->setPath(path); + // myWav->save(); + // return 0: + // } + // int close() { + // // collect the garbage + // delete summary; + // delete path; + // delete myWav; + // + // return 0; + // } + // write out the wav file + bool save() + { + fstream myFile (myPath.c_str(), ios::out | ios::binary); + + // write the wav file per the wav file format + myFile.seekp (0, ios::beg); + myFile.write ("RIFF", 4); + myFile.write ((char*) &myChunkSize, 4); + myFile.write ("WAVE", 4); + myFile.write ("fmt ", 4); + myFile.write ((char*) &mySubChunk1Size, 4); + myFile.write ((char*) &myFormat, 2); + myFile.write ((char*) &myChannels, 2); + myFile.write ((char*) &mySampleRate, 4); + myFile.write ((char*) &myByteRate, 4); + myFile.write ((char*) &myBlockAlign, 2); + myFile.write ((char*) &myBitsPerSample, 2); + myFile.write ("data", 4); + myFile.write ((char*) &myDataSize, 4); + myFile.write (myData, myDataSize); + + return true; + } + + // return a printable summary of the wav file + char *getSummary() + { + char *summary = new char[250]; + sprintf(summary, " Format: %d\n Channels: %d\n SampleRate: %d\n ByteRate: %d\n BlockAlign: %d\n BitsPerSample: %d\n DataSize: %d\n", myFormat, myChannels, mySampleRate, myByteRate, myBlockAlign, myBitsPerSample, myDataSize); + std::cout << myDataSize; + return summary; + } +}; + + +class maxiMap { +public: + static double inline linlin(double val, double inMin, double inMax, double outMin, double outMax) { + return ((val - inMin) / (inMax - inMin) * (outMax - outMin)) + outMin; + } + + static double inline linexp(double val, double inMin, double inMax, double outMin, double outMax) { + //clipping + val = max(min(val, inMax), inMin); + return pow((outMax / outMin), (val - inMin) / (inMax - inMin)) * outMin; + } + + static double inline explin(double val, double inMin, double inMax, double outMin, double outMax) { + //clipping + val = max(min(val, inMax), inMin); + return (log(val/inMin) / log(inMax/inMin) * (outMax - outMin)) + outMin; + } + +}; + + +#endif diff --git a/maximilianTest.xcodeproj/project.pbxproj b/maximilianTest.xcodeproj/project.pbxproj new file mode 100755 index 0000000..7aa34fd --- /dev/null +++ b/maximilianTest.xcodeproj/project.pbxproj @@ -0,0 +1,286 @@ +// !$*UTF8*$! +{ + archiveVersion = 1; + classes = { + }; + objectVersion = 45; + objects = { + +/* Begin PBXBuildFile section */ + 00B5A18610F54EF50011B459 /* maximilian.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 00B5A18510F54EF50011B459 /* maximilian.cpp */; }; + 00FD793810CF318A00EA9294 /* CoreAudio.framework in Frameworks */ = {isa = PBXBuildFile; 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Very slow magic. + + *output=mySine.sinewave(440);//simple as that! + +} + diff --git a/maximilian_examples/10.Filters.cpp b/maximilian_examples/10.Filters.cpp new file mode 100755 index 0000000..b80b512 --- /dev/null +++ b/maximilian_examples/10.Filters.cpp @@ -0,0 +1,33 @@ +#include "maximilian.h" + +maxiOsc myCounter,mySwitchableOsc;// +int CurrentCount;// +double myOscOutput,myFilteredOutput;// +double myEnvelopeData[6] = {500,0,1000,500,0,500};//this data will be used to make an envelope. Value and time to value in ms. +maxiEnvelope myEnvelope; +maxiFilter myFilter; + +void setup() {//some inits + myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope +} + +void play(double *output) { + + CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value. + + if (CurrentCount<5)//simple if statement + + myOscOutput=mySwitchableOsc.square(CurrentCount*100); + + else if (CurrentCount>=5)//and the 'else' bit. + + myOscOutput=mySwitchableOsc.saw(CurrentCount*50);//one osc object can produce whichever waveform you want. + + if (CurrentCount==1) + + myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope + + myFilteredOutput=myFilter.lores(myOscOutput,(myEnvelope.line(6, myEnvelopeData)),10);//lores takes an audio input, a frequency and a resonance factor (1-100) + + *output=myFilteredOutput;//point me at your speakers and fire. +} diff --git a/maximilian_examples/11.Mixing.cpp b/maximilian_examples/11.Mixing.cpp new file mode 100755 index 0000000..6fd7a34 --- /dev/null +++ b/maximilian_examples/11.Mixing.cpp @@ -0,0 +1,40 @@ +#include "maximilian.h" + +maxiOsc myCounter,mySwitchableOsc,myAutoPanner;// +int CurrentCount;// +double myOscOutput,myFilteredOutput,myPanPosition;// +double myStereoOutput[2];// we need an output for each channel. +double myEnvelopeData[6] = {500,0,1000,500,0,500};//this data will be used to make an envelope. Value and time to value in ms. +maxiEnvelope myEnvelope; +maxiFilter myFilter; +maxiMix myOutputs;//this is the stereo mixer channel. + +void setup() {//some inits + myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope +} + +void play(double *output) { + + CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value. + + if (CurrentCount<5)//simple if statement + + myOscOutput=mySwitchableOsc.square(CurrentCount*100); + + else if (CurrentCount>=5)//and the 'else' bit. + + myOscOutput=mySwitchableOsc.saw(CurrentCount*50);//one osc object can produce whichever waveform you want. + + if (CurrentCount==1) + + myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope + + myFilteredOutput=myFilter.lores(myOscOutput,(myEnvelope.line(6, myEnvelopeData)),10);//lores takes an audio input, a frequency and a resonance factor (1-100) + + myPanPosition=myAutoPanner.sinewave(1); + + myOutputs.stereo(myFilteredOutput,myStereoOutput,myPanPosition);//Stereo, Quad or 8 Channel. Specify the input to be mixed, the output[numberofchannels], and the pan (0-1,equal power). + output[0]=myStereoOutput[0];//When working with mixing, you need to specify the outputs explicityly + output[1]=myStereoOutput[1];// + +} diff --git a/maximilian_examples/12.SamplePlayer.cpp b/maximilian_examples/12.SamplePlayer.cpp new file mode 100755 index 0000000..2706f0b --- /dev/null +++ b/maximilian_examples/12.SamplePlayer.cpp @@ -0,0 +1,21 @@ +#include "maximilian.h" + +maxiSample beats; //We give our sample a name. It's called beats this time. We could have loads of them, but they have to have different names. + +void setup() {//some inits + + beats.load("/Users/username/somewhere/Maximilian/beat2.wav");//load in your samples. Provide the full path to a wav file. + printf("Summary:\n%s", beats.getSummary());//get info on samples if you like. + +} + +void play(double *output) {//this is where the magic happens. Very slow magic. + + *output=beats.play();//just play the file. Looping is default for all play functions. + // *output=beats.play(0.69);//play the file with a speed setting. 1. is normal speed. + // *output=beats.play(0.5,0,44100);//linear interpolationplay with a frequency input, start point and end point. Useful for syncing. + // *output=beats.play4(0.5,0,44100);//cubic interpolation play with a frequency input, start point and end point. Useful for syncing. + + +} + diff --git a/maximilian_examples/2.TwoTones.cpp b/maximilian_examples/2.TwoTones.cpp new file mode 100755 index 0000000..4e22994 --- /dev/null +++ b/maximilian_examples/2.TwoTones.cpp @@ -0,0 +1,14 @@ +#include "maximilian.h" + +maxiOsc mySine,myOtherSine;//Two oscillators with names. + +void setup() {//some inits + //nothing to go here this time +} + +void play(double *output) {//this is where the magic happens. Very slow magic. + + *output=mySine.sinewave(440)+myOtherSine.sinewave(441);//these two sines will beat together. They're now a bit too loud though.. + +} + diff --git a/maximilian_examples/3.AM1.cpp b/maximilian_examples/3.AM1.cpp new file mode 100755 index 0000000..ec96c7f --- /dev/null +++ b/maximilian_examples/3.AM1.cpp @@ -0,0 +1,15 @@ +#include "maximilian.h" + +//This shows how to use maximilian to do basic amplitude modulation + +maxiOsc mySine,myOtherSine;//Two oscillators. They can be called anything. They can be any of the available waveforms. These ones will be sinewaves + +void setup() {//some inits + //nothing to go here this time +} + +void play(double *output) { + + *output=mySine.sinewave(440)*myOtherSine.sinewave(10); + +} diff --git a/maximilian_examples/4.AM2.cpp b/maximilian_examples/4.AM2.cpp new file mode 100755 index 0000000..4e783d3 --- /dev/null +++ b/maximilian_examples/4.AM2.cpp @@ -0,0 +1,16 @@ +#include "maximilian.h" + +//This shows how to use maximilian to do basic amplitude modulation + +maxiOsc mySine,myOtherSine,myPhasor;//Three oscillators. They can be called anything. They can be any of the available waveforms. + + +void setup() {//some inits + //nothing to go here this time +} + +void play(double *output) { + + *output=mySine.sinewave(440)*myOtherSine.sinewave(myPhasor.phasor(0.1,0,440)); + +} diff --git a/maximilian_examples/5.FM1.cpp b/maximilian_examples/5.FM1.cpp new file mode 100755 index 0000000..c734dcf --- /dev/null +++ b/maximilian_examples/5.FM1.cpp @@ -0,0 +1,14 @@ +#include "maximilian.h" + +maxiOsc mySine,myOtherSine;//Two oscillators + + +void setup() {//some inits + //nothing to go here this time +} + +void play(double *output) { + + *output=mySine.sinewave(myOtherSine.sinewave(1)*440); + +} diff --git a/maximilian_examples/6.FM2.cpp b/maximilian_examples/6.FM2.cpp new file mode 100755 index 0000000..d9acd72 --- /dev/null +++ b/maximilian_examples/6.FM2.cpp @@ -0,0 +1,14 @@ +#include "maximilian.h" + +maxiOsc mySine,myOtherSine,myLastSine,myPhasor;//Three oscillators + + +void setup() {//some inits + //nothing to go here this time +} + +void play(double *output) { + + *output=mySine.sinewave(myOtherSine.sinewave(myLastSine.sinewave(0.1)*30)*440);//awesome bassline + +} diff --git a/maximilian_examples/7.Counting.cpp b/maximilian_examples/7.Counting.cpp new file mode 100755 index 0000000..7de2f0e --- /dev/null +++ b/maximilian_examples/7.Counting.cpp @@ -0,0 +1,15 @@ +#include "maximilian.h" + +maxiOsc myCounter,mySquare;//these oscillators will help us count and play sound +int CurrentCount;//we're going to put the current count in this variable so that we can use it more easily. + + +void setup() {//some inits + //nothing to go here this time +} + +void play(double *output) { + + CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value. + *output=mySquare.square(CurrentCount*100); +} diff --git a/maximilian_examples/8.Counting2.cpp b/maximilian_examples/8.Counting2.cpp new file mode 100755 index 0000000..ce016d0 --- /dev/null +++ b/maximilian_examples/8.Counting2.cpp @@ -0,0 +1,24 @@ +#include "maximilian.h" + +maxiOsc myCounter,mySwitchableOsc;//these oscillators will help us count and make sound. +int CurrentCount;//we're going to put the current count in this variable so that we can use it more easily. +double myOscOutput;//we're going to stick the output here to make it easier to mess with stuff. + +void setup() {//some inits + //nothing to go here this time +} + +void play(double *output) { + + CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value. + + if (CurrentCount<5)//simple if statement + + myOscOutput=mySwitchableOsc.square(CurrentCount*100); + + else if (CurrentCount>=5)//and the 'else' bit. + + myOscOutput=mySwitchableOsc.sinewave(CurrentCount*50);//one osc object can produce whichever waveform you want. + + *output=myOscOutput;//point me at your speakers and fire. +} diff --git a/maximilian_examples/9.Envelopes.cpp b/maximilian_examples/9.Envelopes.cpp new file mode 100755 index 0000000..806e14f --- /dev/null +++ b/maximilian_examples/9.Envelopes.cpp @@ -0,0 +1,33 @@ +#include "maximilian.h" + +maxiOsc myCounter,mySwitchableOsc;// +int CurrentCount;// +double myOscOutput,myCurrentVolume;// +double myEnvelopeData[4] = {1,0,0,500};//this data will be used to make an envelope. Value and time to value in ms. +maxiEnvelope myEnvelope; + + +void setup() {//some inits + myEnvelope.amplitude=myEnvelopeData[0]; //initialise the envelope +} + +void play(double *output) { + + myCurrentVolume=myEnvelope.line(4,myEnvelopeData); + + CurrentCount=myCounter.phasor(1, 1, 9);//phasor can take three arguments; frequency, start value and end value. + + if (CurrentCount<5)//simple if statement + + myOscOutput=mySwitchableOsc.square(CurrentCount*100); + + else if (CurrentCount>=5)//and the 'else' bit. + + myOscOutput=mySwitchableOsc.sinewave(CurrentCount*50);//one osc object can produce whichever waveform you want. + + if (CurrentCount==1) + + myEnvelope.trigger(0,myEnvelopeData[0]); //trigger the envelope + + *output=myOscOutput*myCurrentVolume;//point me at your speakers and fire. +} diff --git a/player.cpp b/player.cpp new file mode 100644 index 0000000..37edf80 --- /dev/null +++ b/player.cpp @@ -0,0 +1,153 @@ +/* + * player.cpp + * rtaudiotest + * + * Created by Chris on 23/08/2011. + * Copyright 2011 Goldsmiths Creative Computing. All rights reserved. + * + */ + +#include "player.h" +#include "maximilian.h" +#include + + +#ifdef MAXIMILIAN_PORTAUDIO +#include "portaudio.h" +//#include "pa_mac_core.h" +#elif defined(MAXIMILIAN_RT_AUDIO) +#include "RtAudio.h" +#endif + + +void setup();//use this to do any initialisation if you want. + +void play(double *channels);//run dac! Very very often. Too often in fact. er... + + + +#ifdef MAXIMILIAN_PORTAUDIO +int routing(const void *inputBuffer, + void *outputBuffer, + unsigned long nBufferFrames, + const PaStreamCallbackTimeInfo* timeInfo, + PaStreamCallbackFlags status, + void *userData ){ +#elif defined(MAXIMILIAN_RT_AUDIO) +int routing (void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, + double streamTime, RtAudioStreamStatus status, void *userData ) { +#endif + + unsigned int i, j; + +#ifdef MAXIMILIAN_PORTAUDIO + float *buffer = (float *) outputBuffer; +#elif defined(MAXIMILIAN_RT_AUDIO) + double *buffer = (double *) outputBuffer; +#endif + double *lastValues = (double *) userData; + // double currentTime = (double) streamTime; Might come in handy for control + if ( status ) + std::cout << "Stream underflow detected!" << std::endl; + for ( i=0; i to quit.\n"; + std::cin.get( input ); + + + + err = Pa_Terminate(); + if( err != paNoError ) + std::cout << "PortAudio error: "<< Pa_GetErrorText( err ) << std::endl; + +#elif defined(MAXIMILIAN_RT_AUDIO) + RtAudio dac; + if ( dac.getDeviceCount() < 1 ) { + std::cout << "\nNo audio devices found!\n"; + exit( 0 ); + } + + RtAudio::StreamParameters parameters; + parameters.deviceId = dac.getDefaultOutputDevice(); + parameters.nChannels = maxiSettings::channels; + parameters.firstChannel = 0; + unsigned int sampleRate = maxiSettings::sampleRate; + unsigned int bufferFrames = maxiSettings::bufferSize; + double data[parameters.nChannels]; + + try { + dac.openStream( ¶meters, NULL, RTAUDIO_FLOAT64, + sampleRate, &bufferFrames, &routing, (void *)&data); + + dac.startStream(); + } + catch ( RtError& e ) { + e.printMessage(); + exit( 0 ); + } + + char input; + std::cout << "\nMaximilian is playing ... press to quit.\n"; + std::cin.get( input ); + + try { + // Stop the stream + dac.stopStream(); + } + catch (RtError& e) { + e.printMessage(); + } + + if ( dac.isStreamOpen() ) dac.closeStream(); +#endif + + return 0; +} diff --git a/player.h b/player.h new file mode 100644 index 0000000..3fea524 --- /dev/null +++ b/player.h @@ -0,0 +1,11 @@ +/* + * player.h + * rtaudiotest + * + * Created by Chris on 23/08/2011. + * Copyright 2011 Goldsmiths Creative Computing. All rights reserved. + * + */ + +//#define MAXIMILIAN_PORTAUDIO +#define MAXIMILIAN_RT_AUDIO diff --git a/portaudio.h b/portaudio.h new file mode 100644 index 0000000..89964ab --- /dev/null +++ b/portaudio.h @@ -0,0 +1,1134 @@ +#ifndef PORTAUDIO_H +#define PORTAUDIO_H +/* + * $Id: portaudio.h 1337 2008-02-15 07:32:09Z rossb $ + * PortAudio Portable Real-Time Audio Library + * PortAudio API Header File + * Latest version available at: http://www.portaudio.com/ + * + * Copyright (c) 1999-2002 Ross Bencina and Phil Burk + * + * Permission is hereby granted, free of charge, to any person obtaining + * a copy of this software and associated documentation files + * (the "Software"), to deal in the Software without restriction, + * including without limitation the rights to use, copy, modify, merge, + * publish, distribute, sublicense, and/or sell copies of the Software, + * and to permit persons to whom the Software is furnished to do so, + * subject to the following conditions: + * + * The above copyright notice and this permission notice shall be + * included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, + * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF + * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. + * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR + * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF + * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION + * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + */ + +/* + * The text above constitutes the entire PortAudio license; however, + * the PortAudio community also makes the following non-binding requests: + * + * Any person wishing to distribute modifications to the Software is + * requested to send the modifications to the original developer so that + * they can be incorporated into the canonical version. It is also + * requested that these non-binding requests be included along with the + * license above. + */ + +/** @file + @ingroup public_header + @brief The portable PortAudio API. +*/ + + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + + +/** Retrieve the release number of the currently running PortAudio build, + eg 1900. +*/ +int Pa_GetVersion( void ); + + +/** Retrieve a textual description of the current PortAudio build, + eg "PortAudio V19-devel 13 October 2002". +*/ +const char* Pa_GetVersionText( void ); + + +/** Error codes returned by PortAudio functions. + Note that with the exception of paNoError, all PaErrorCodes are negative. +*/ + +typedef int PaError; +typedef enum PaErrorCode +{ + paNoError = 0, + + paNotInitialized = -10000, + paUnanticipatedHostError, + paInvalidChannelCount, + paInvalidSampleRate, + paInvalidDevice, + paInvalidFlag, + paSampleFormatNotSupported, + paBadIODeviceCombination, + paInsufficientMemory, + paBufferTooBig, + paBufferTooSmall, + paNullCallback, + paBadStreamPtr, + paTimedOut, + paInternalError, + paDeviceUnavailable, + paIncompatibleHostApiSpecificStreamInfo, + paStreamIsStopped, + paStreamIsNotStopped, + paInputOverflowed, + paOutputUnderflowed, + paHostApiNotFound, + paInvalidHostApi, + paCanNotReadFromACallbackStream, /**< @todo review error code name */ + paCanNotWriteToACallbackStream, /**< @todo review error code name */ + paCanNotReadFromAnOutputOnlyStream, /**< @todo review error code name */ + paCanNotWriteToAnInputOnlyStream, /**< @todo review error code name */ + paIncompatibleStreamHostApi, + paBadBufferPtr +} PaErrorCode; + + +/** Translate the supplied PortAudio error code into a human readable + message. +*/ +const char *Pa_GetErrorText( PaError errorCode ); + + +/** Library initialization function - call this before using PortAudio. + This function initialises internal data structures and prepares underlying + host APIs for use. With the exception of Pa_GetVersion(), Pa_GetVersionText(), + and Pa_GetErrorText(), this function MUST be called before using any other + PortAudio API functions. + + If Pa_Initialize() is called multiple times, each successful + call must be matched with a corresponding call to Pa_Terminate(). + Pairs of calls to Pa_Initialize()/Pa_Terminate() may overlap, and are not + required to be fully nested. + + Note that if Pa_Initialize() returns an error code, Pa_Terminate() should + NOT be called. + + @return paNoError if successful, otherwise an error code indicating the cause + of failure. + + @see Pa_Terminate +*/ +PaError Pa_Initialize( void ); + + +/** Library termination function - call this when finished using PortAudio. + This function deallocates all resources allocated by PortAudio since it was + initializied by a call to Pa_Initialize(). In cases where Pa_Initialise() has + been called multiple times, each call must be matched with a corresponding call + to Pa_Terminate(). The final matching call to Pa_Terminate() will automatically + close any PortAudio streams that are still open. + + Pa_Terminate() MUST be called before exiting a program which uses PortAudio. + Failure to do so may result in serious resource leaks, such as audio devices + not being available until the next reboot. + + @return paNoError if successful, otherwise an error code indicating the cause + of failure. + + @see Pa_Initialize +*/ +PaError Pa_Terminate( void ); + + + +/** The type used to refer to audio devices. Values of this type usually + range from 0 to (Pa_GetDeviceCount()-1), and may also take on the PaNoDevice + and paUseHostApiSpecificDeviceSpecification values. + + @see Pa_GetDeviceCount, paNoDevice, paUseHostApiSpecificDeviceSpecification +*/ +typedef int PaDeviceIndex; + + +/** A special PaDeviceIndex value indicating that no device is available, + or should be used. + + @see PaDeviceIndex +*/ +#define paNoDevice ((PaDeviceIndex)-1) + + +/** A special PaDeviceIndex value indicating that the device(s) to be used + are specified in the host api specific stream info structure. + + @see PaDeviceIndex +*/ +#define paUseHostApiSpecificDeviceSpecification ((PaDeviceIndex)-2) + + +/* Host API enumeration mechanism */ + +/** The type used to enumerate to host APIs at runtime. Values of this type + range from 0 to (Pa_GetHostApiCount()-1). + + @see Pa_GetHostApiCount +*/ +typedef int PaHostApiIndex; + + +/** Retrieve the number of available host APIs. Even if a host API is + available it may have no devices available. + + @return A non-negative value indicating the number of available host APIs + or, a PaErrorCode (which are always negative) if PortAudio is not initialized + or an error is encountered. + + @see PaHostApiIndex +*/ +PaHostApiIndex Pa_GetHostApiCount( void ); + + +/** Retrieve the index of the default host API. The default host API will be + the lowest common denominator host API on the current platform and is + unlikely to provide the best performance. + + @return A non-negative value ranging from 0 to (Pa_GetHostApiCount()-1) + indicating the default host API index or, a PaErrorCode (which are always + negative) if PortAudio is not initialized or an error is encountered. +*/ +PaHostApiIndex Pa_GetDefaultHostApi( void ); + + +/** Unchanging unique identifiers for each supported host API. This type + is used in the PaHostApiInfo structure. The values are guaranteed to be + unique and to never change, thus allowing code to be written that + conditionally uses host API specific extensions. + + New type ids will be allocated when support for a host API reaches + "public alpha" status, prior to that developers should use the + paInDevelopment type id. + + @see PaHostApiInfo +*/ +typedef enum PaHostApiTypeId +{ + paInDevelopment=0, /* use while developing support for a new host API */ + paDirectSound=1, + paMME=2, + paASIO=3, + paSoundManager=4, + paCoreAudio=5, + paOSS=7, + paALSA=8, + paAL=9, + paBeOS=10, + paWDMKS=11, + paJACK=12, + paWASAPI=13, + paAudioScienceHPI=14 +} PaHostApiTypeId; + + +/** A structure containing information about a particular host API. */ + +typedef struct PaHostApiInfo +{ + /** this is struct version 1 */ + int structVersion; + /** The well known unique identifier of this host API @see PaHostApiTypeId */ + PaHostApiTypeId type; + /** A textual description of the host API for display on user interfaces. */ + const char *name; + + /** The number of devices belonging to this host API. This field may be + used in conjunction with Pa_HostApiDeviceIndexToDeviceIndex() to enumerate + all devices for this host API. + @see Pa_HostApiDeviceIndexToDeviceIndex + */ + int deviceCount; + + /** The default input device for this host API. The value will be a + device index ranging from 0 to (Pa_GetDeviceCount()-1), or paNoDevice + if no default input device is available. + */ + PaDeviceIndex defaultInputDevice; + + /** The default output device for this host API. The value will be a + device index ranging from 0 to (Pa_GetDeviceCount()-1), or paNoDevice + if no default output device is available. + */ + PaDeviceIndex defaultOutputDevice; + +} PaHostApiInfo; + + +/** Retrieve a pointer to a structure containing information about a specific + host Api. + + @param hostApi A valid host API index ranging from 0 to (Pa_GetHostApiCount()-1) + + @return A pointer to an immutable PaHostApiInfo structure describing + a specific host API. If the hostApi parameter is out of range or an error + is encountered, the function returns NULL. + + The returned structure is owned by the PortAudio implementation and must not + be manipulated or freed. The pointer is only guaranteed to be valid between + calls to Pa_Initialize() and Pa_Terminate(). +*/ +const PaHostApiInfo * Pa_GetHostApiInfo( PaHostApiIndex hostApi ); + + +/** Convert a static host API unique identifier, into a runtime + host API index. + + @param type A unique host API identifier belonging to the PaHostApiTypeId + enumeration. + + @return A valid PaHostApiIndex ranging from 0 to (Pa_GetHostApiCount()-1) or, + a PaErrorCode (which are always negative) if PortAudio is not initialized + or an error is encountered. + + The paHostApiNotFound error code indicates that the host API specified by the + type parameter is not available. + + @see PaHostApiTypeId +*/ +PaHostApiIndex Pa_HostApiTypeIdToHostApiIndex( PaHostApiTypeId type ); + + +/** Convert a host-API-specific device index to standard PortAudio device index. + This function may be used in conjunction with the deviceCount field of + PaHostApiInfo to enumerate all devices for the specified host API. + + @param hostApi A valid host API index ranging from 0 to (Pa_GetHostApiCount()-1) + + @param hostApiDeviceIndex A valid per-host device index in the range + 0 to (Pa_GetHostApiInfo(hostApi)->deviceCount-1) + + @return A non-negative PaDeviceIndex ranging from 0 to (Pa_GetDeviceCount()-1) + or, a PaErrorCode (which are always negative) if PortAudio is not initialized + or an error is encountered. + + A paInvalidHostApi error code indicates that the host API index specified by + the hostApi parameter is out of range. + + A paInvalidDevice error code indicates that the hostApiDeviceIndex parameter + is out of range. + + @see PaHostApiInfo +*/ +PaDeviceIndex Pa_HostApiDeviceIndexToDeviceIndex( PaHostApiIndex hostApi, + int hostApiDeviceIndex ); + + + +/** Structure used to return information about a host error condition. +*/ +typedef struct PaHostErrorInfo{ + PaHostApiTypeId hostApiType; /**< the host API which returned the error code */ + long errorCode; /**< the error code returned */ + const char *errorText; /**< a textual description of the error if available, otherwise a zero-length string */ +}PaHostErrorInfo; + + +/** Return information about the last host error encountered. The error + information returned by Pa_GetLastHostErrorInfo() will never be modified + asyncronously by errors occurring in other PortAudio owned threads + (such as the thread that manages the stream callback.) + + This function is provided as a last resort, primarily to enhance debugging + by providing clients with access to all available error information. + + @return A pointer to an immutable structure constaining information about + the host error. The values in this structure will only be valid if a + PortAudio function has previously returned the paUnanticipatedHostError + error code. +*/ +const PaHostErrorInfo* Pa_GetLastHostErrorInfo( void ); + + + +/* Device enumeration and capabilities */ + +/** Retrieve the number of available devices. The number of available devices + may be zero. + + @return A non-negative value indicating the number of available devices or, + a PaErrorCode (which are always negative) if PortAudio is not initialized + or an error is encountered. +*/ +PaDeviceIndex Pa_GetDeviceCount( void ); + + +/** Retrieve the index of the default input device. The result can be + used in the inputDevice parameter to Pa_OpenStream(). + + @return The default input device index for the default host API, or paNoDevice + if no default input device is available or an error was encountered. +*/ +PaDeviceIndex Pa_GetDefaultInputDevice( void ); + + +/** Retrieve the index of the default output device. The result can be + used in the outputDevice parameter to Pa_OpenStream(). + + @return The default output device index for the defualt host API, or paNoDevice + if no default output device is available or an error was encountered. + + @note + On the PC, the user can specify a default device by + setting an environment variable. For example, to use device #1. +
+ set PA_RECOMMENDED_OUTPUT_DEVICE=1
+
+ The user should first determine the available device ids by using + the supplied application "pa_devs". +*/ +PaDeviceIndex Pa_GetDefaultOutputDevice( void ); + + +/** The type used to represent monotonic time in seconds that can be used + for syncronisation. The type is used for the outTime argument to the + PaStreamCallback and as the result of Pa_GetStreamTime(). + + @see PaStreamCallback, Pa_GetStreamTime +*/ +typedef double PaTime; + + +/** A type used to specify one or more sample formats. Each value indicates + a possible format for sound data passed to and from the stream callback, + Pa_ReadStream and Pa_WriteStream. + + The standard formats paFloat32, paInt16, paInt32, paInt24, paInt8 + and aUInt8 are usually implemented by all implementations. + + The floating point representation (paFloat32) uses +1.0 and -1.0 as the + maximum and minimum respectively. + + paUInt8 is an unsigned 8 bit format where 128 is considered "ground" + + The paNonInterleaved flag indicates that a multichannel buffer is passed + as a set of non-interleaved pointers. + + @see Pa_OpenStream, Pa_OpenDefaultStream, PaDeviceInfo + @see paFloat32, paInt16, paInt32, paInt24, paInt8 + @see paUInt8, paCustomFormat, paNonInterleaved +*/ +typedef unsigned long PaSampleFormat; + + +#define paFloat32 ((PaSampleFormat) 0x00000001) /**< @see PaSampleFormat */ +#define paInt32 ((PaSampleFormat) 0x00000002) /**< @see PaSampleFormat */ +#define paInt24 ((PaSampleFormat) 0x00000004) /**< Packed 24 bit format. @see PaSampleFormat */ +#define paInt16 ((PaSampleFormat) 0x00000008) /**< @see PaSampleFormat */ +#define paInt8 ((PaSampleFormat) 0x00000010) /**< @see PaSampleFormat */ +#define paUInt8 ((PaSampleFormat) 0x00000020) /**< @see PaSampleFormat */ +#define paCustomFormat ((PaSampleFormat) 0x00010000)/**< @see PaSampleFormat */ + +#define paNonInterleaved ((PaSampleFormat) 0x80000000) + +/** A structure providing information and capabilities of PortAudio devices. + Devices may support input, output or both input and output. +*/ +typedef struct PaDeviceInfo +{ + int structVersion; /* this is struct version 2 */ + const char *name; + PaHostApiIndex hostApi; /* note this is a host API index, not a type id*/ + + int maxInputChannels; + int maxOutputChannels; + + /* Default latency values for interactive performance. */ + PaTime defaultLowInputLatency; + PaTime defaultLowOutputLatency; + /* Default latency values for robust non-interactive applications (eg. playing sound files). */ + PaTime defaultHighInputLatency; + PaTime defaultHighOutputLatency; + + double defaultSampleRate; +} PaDeviceInfo; + + +/** Retrieve a pointer to a PaDeviceInfo structure containing information + about the specified device. + @return A pointer to an immutable PaDeviceInfo structure. If the device + parameter is out of range the function returns NULL. + + @param device A valid device index in the range 0 to (Pa_GetDeviceCount()-1) + + @note PortAudio manages the memory referenced by the returned pointer, + the client must not manipulate or free the memory. The pointer is only + guaranteed to be valid between calls to Pa_Initialize() and Pa_Terminate(). + + @see PaDeviceInfo, PaDeviceIndex +*/ +const PaDeviceInfo* Pa_GetDeviceInfo( PaDeviceIndex device ); + + +/** Parameters for one direction (input or output) of a stream. +*/ +typedef struct PaStreamParameters +{ + /** A valid device index in the range 0 to (Pa_GetDeviceCount()-1) + specifying the device to be used or the special constant + paUseHostApiSpecificDeviceSpecification which indicates that the actual + device(s) to use are specified in hostApiSpecificStreamInfo. + This field must not be set to paNoDevice. + */ + PaDeviceIndex device; + + /** The number of channels of sound to be delivered to the + stream callback or accessed by Pa_ReadStream() or Pa_WriteStream(). + It can range from 1 to the value of maxInputChannels in the + PaDeviceInfo record for the device specified by the device parameter. + */ + int channelCount; + + /** The sample format of the buffer provided to the stream callback, + a_ReadStream() or Pa_WriteStream(). It may be any of the formats described + by the PaSampleFormat enumeration. + */ + PaSampleFormat sampleFormat; + + /** The desired latency in seconds. Where practical, implementations should + configure their latency based on these parameters, otherwise they may + choose the closest viable latency instead. Unless the suggested latency + is greater than the absolute upper limit for the device implementations + should round the suggestedLatency up to the next practial value - ie to + provide an equal or higher latency than suggestedLatency wherever possibe. + Actual latency values for an open stream may be retrieved using the + inputLatency and outputLatency fields of the PaStreamInfo structure + returned by Pa_GetStreamInfo(). + @see default*Latency in PaDeviceInfo, *Latency in PaStreamInfo + */ + PaTime suggestedLatency; + + /** An optional pointer to a host api specific data structure + containing additional information for device setup and/or stream processing. + hostApiSpecificStreamInfo is never required for correct operation, + if not used it should be set to NULL. + */ + void *hostApiSpecificStreamInfo; + +} PaStreamParameters; + + +/** Return code for Pa_IsFormatSupported indicating success. */ +#define paFormatIsSupported (0) + +/** Determine whether it would be possible to open a stream with the specified + parameters. + + @param inputParameters A structure that describes the input parameters used to + open a stream. The suggestedLatency field is ignored. See PaStreamParameters + for a description of these parameters. inputParameters must be NULL for + output-only streams. + + @param outputParameters A structure that describes the output parameters used + to open a stream. The suggestedLatency field is ignored. See PaStreamParameters + for a description of these parameters. outputParameters must be NULL for + input-only streams. + + @param sampleRate The required sampleRate. For full-duplex streams it is the + sample rate for both input and output + + @return Returns 0 if the format is supported, and an error code indicating why + the format is not supported otherwise. The constant paFormatIsSupported is + provided to compare with the return value for success. + + @see paFormatIsSupported, PaStreamParameters +*/ +PaError Pa_IsFormatSupported( const PaStreamParameters *inputParameters, + const PaStreamParameters *outputParameters, + double sampleRate ); + + + +/* Streaming types and functions */ + + +/** + A single PaStream can provide multiple channels of real-time + streaming audio input and output to a client application. A stream + provides access to audio hardware represented by one or more + PaDevices. Depending on the underlying Host API, it may be possible + to open multiple streams using the same device, however this behavior + is implementation defined. Portable applications should assume that + a PaDevice may be simultaneously used by at most one PaStream. + + Pointers to PaStream objects are passed between PortAudio functions that + operate on streams. + + @see Pa_OpenStream, Pa_OpenDefaultStream, Pa_OpenDefaultStream, Pa_CloseStream, + Pa_StartStream, Pa_StopStream, Pa_AbortStream, Pa_IsStreamActive, + Pa_GetStreamTime, Pa_GetStreamCpuLoad + +*/ +typedef void PaStream; + + +/** Can be passed as the framesPerBuffer parameter to Pa_OpenStream() + or Pa_OpenDefaultStream() to indicate that the stream callback will + accept buffers of any size. +*/ +#define paFramesPerBufferUnspecified (0) + + +/** Flags used to control the behavior of a stream. They are passed as + parameters to Pa_OpenStream or Pa_OpenDefaultStream. Multiple flags may be + ORed together. + + @see Pa_OpenStream, Pa_OpenDefaultStream + @see paNoFlag, paClipOff, paDitherOff, paNeverDropInput, + paPrimeOutputBuffersUsingStreamCallback, paPlatformSpecificFlags +*/ +typedef unsigned long PaStreamFlags; + +/** @see PaStreamFlags */ +#define paNoFlag ((PaStreamFlags) 0) + +/** Disable default clipping of out of range samples. + @see PaStreamFlags +*/ +#define paClipOff ((PaStreamFlags) 0x00000001) + +/** Disable default dithering. + @see PaStreamFlags +*/ +#define paDitherOff ((PaStreamFlags) 0x00000002) + +/** Flag requests that where possible a full duplex stream will not discard + overflowed input samples without calling the stream callback. This flag is + only valid for full duplex callback streams and only when used in combination + with the paFramesPerBufferUnspecified (0) framesPerBuffer parameter. Using + this flag incorrectly results in a paInvalidFlag error being returned from + Pa_OpenStream and Pa_OpenDefaultStream. + + @see PaStreamFlags, paFramesPerBufferUnspecified +*/ +#define paNeverDropInput ((PaStreamFlags) 0x00000004) + +/** Call the stream callback to fill initial output buffers, rather than the + default behavior of priming the buffers with zeros (silence). This flag has + no effect for input-only and blocking read/write streams. + + @see PaStreamFlags +*/ +#define paPrimeOutputBuffersUsingStreamCallback ((PaStreamFlags) 0x00000008) + +/** A mask specifying the platform specific bits. + @see PaStreamFlags +*/ +#define paPlatformSpecificFlags ((PaStreamFlags)0xFFFF0000) + +/** + Timing information for the buffers passed to the stream callback. +*/ +typedef struct PaStreamCallbackTimeInfo{ + PaTime inputBufferAdcTime; + PaTime currentTime; + PaTime outputBufferDacTime; +} PaStreamCallbackTimeInfo; + + +/** + Flag bit constants for the statusFlags to PaStreamCallback. + + @see paInputUnderflow, paInputOverflow, paOutputUnderflow, paOutputOverflow, + paPrimingOutput +*/ +typedef unsigned long PaStreamCallbackFlags; + +/** In a stream opened with paFramesPerBufferUnspecified, indicates that + input data is all silence (zeros) because no real data is available. In a + stream opened without paFramesPerBufferUnspecified, it indicates that one or + more zero samples have been inserted into the input buffer to compensate + for an input underflow. + @see PaStreamCallbackFlags +*/ +#define paInputUnderflow ((PaStreamCallbackFlags) 0x00000001) + +/** In a stream opened with paFramesPerBufferUnspecified, indicates that data + prior to the first sample of the input buffer was discarded due to an + overflow, possibly because the stream callback is using too much CPU time. + Otherwise indicates that data prior to one or more samples in the + input buffer was discarded. + @see PaStreamCallbackFlags +*/ +#define paInputOverflow ((PaStreamCallbackFlags) 0x00000002) + +/** Indicates that output data (or a gap) was inserted, possibly because the + stream callback is using too much CPU time. + @see PaStreamCallbackFlags +*/ +#define paOutputUnderflow ((PaStreamCallbackFlags) 0x00000004) + +/** Indicates that output data will be discarded because no room is available. + @see PaStreamCallbackFlags +*/ +#define paOutputOverflow ((PaStreamCallbackFlags) 0x00000008) + +/** Some of all of the output data will be used to prime the stream, input + data may be zero. + @see PaStreamCallbackFlags +*/ +#define paPrimingOutput ((PaStreamCallbackFlags) 0x00000010) + +/** + Allowable return values for the PaStreamCallback. + @see PaStreamCallback +*/ +typedef enum PaStreamCallbackResult +{ + paContinue=0, + paComplete=1, + paAbort=2 +} PaStreamCallbackResult; + + +/** + Functions of type PaStreamCallback are implemented by PortAudio clients. + They consume, process or generate audio in response to requests from an + active PortAudio stream. + + @param input and @param output are arrays of interleaved samples, + the format, packing and number of channels used by the buffers are + determined by parameters to Pa_OpenStream(). + + @param frameCount The number of sample frames to be processed by + the stream callback. + + @param timeInfo The time in seconds when the first sample of the input + buffer was received at the audio input, the time in seconds when the first + sample of the output buffer will begin being played at the audio output, and + the time in seconds when the stream callback was called. + See also Pa_GetStreamTime() + + @param statusFlags Flags indicating whether input and/or output buffers + have been inserted or will be dropped to overcome underflow or overflow + conditions. + + @param userData The value of a user supplied pointer passed to + Pa_OpenStream() intended for storing synthesis data etc. + + @return + The stream callback should return one of the values in the + PaStreamCallbackResult enumeration. To ensure that the callback continues + to be called, it should return paContinue (0). Either paComplete or paAbort + can be returned to finish stream processing, after either of these values is + returned the callback will not be called again. If paAbort is returned the + stream will finish as soon as possible. If paComplete is returned, the stream + will continue until all buffers generated by the callback have been played. + This may be useful in applications such as soundfile players where a specific + duration of output is required. However, it is not necessary to utilise this + mechanism as Pa_StopStream(), Pa_AbortStream() or Pa_CloseStream() can also + be used to stop the stream. The callback must always fill the entire output + buffer irrespective of its return value. + + @see Pa_OpenStream, Pa_OpenDefaultStream + + @note With the exception of Pa_GetStreamCpuLoad() it is not permissable to call + PortAudio API functions from within the stream callback. +*/ +typedef int PaStreamCallback( + const void *input, void *output, + unsigned long frameCount, + const PaStreamCallbackTimeInfo* timeInfo, + PaStreamCallbackFlags statusFlags, + void *userData ); + + +/** Opens a stream for either input, output or both. + + @param stream The address of a PaStream pointer which will receive + a pointer to the newly opened stream. + + @param inputParameters A structure that describes the input parameters used by + the opened stream. See PaStreamParameters for a description of these parameters. + inputParameters must be NULL for output-only streams. + + @param outputParameters A structure that describes the output parameters used by + the opened stream. See PaStreamParameters for a description of these parameters. + outputParameters must be NULL for input-only streams. + + @param sampleRate The desired sampleRate. For full-duplex streams it is the + sample rate for both input and output + + @param framesPerBuffer The number of frames passed to the stream callback + function, or the preferred block granularity for a blocking read/write stream. + The special value paFramesPerBufferUnspecified (0) may be used to request that + the stream callback will recieve an optimal (and possibly varying) number of + frames based on host requirements and the requested latency settings. + Note: With some host APIs, the use of non-zero framesPerBuffer for a callback + stream may introduce an additional layer of buffering which could introduce + additional latency. PortAudio guarantees that the additional latency + will be kept to the theoretical minimum however, it is strongly recommended + that a non-zero framesPerBuffer value only be used when your algorithm + requires a fixed number of frames per stream callback. + + @param streamFlags Flags which modify the behaviour of the streaming process. + This parameter may contain a combination of flags ORed together. Some flags may + only be relevant to certain buffer formats. + + @param streamCallback A pointer to a client supplied function that is responsible + for processing and filling input and output buffers. If this parameter is NULL + the stream will be opened in 'blocking read/write' mode. In blocking mode, + the client can receive sample data using Pa_ReadStream and write sample data + using Pa_WriteStream, the number of samples that may be read or written + without blocking is returned by Pa_GetStreamReadAvailable and + Pa_GetStreamWriteAvailable respectively. + + @param userData A client supplied pointer which is passed to the stream callback + function. It could for example, contain a pointer to instance data necessary + for processing the audio buffers. This parameter is ignored if streamCallback + is NULL. + + @return + Upon success Pa_OpenStream() returns paNoError and places a pointer to a + valid PaStream in the stream argument. The stream is inactive (stopped). + If a call to Pa_OpenStream() fails, a non-zero error code is returned (see + PaError for possible error codes) and the value of stream is invalid. + + @see PaStreamParameters, PaStreamCallback, Pa_ReadStream, Pa_WriteStream, + Pa_GetStreamReadAvailable, Pa_GetStreamWriteAvailable +*/ +PaError Pa_OpenStream( PaStream** stream, + const PaStreamParameters *inputParameters, + const PaStreamParameters *outputParameters, + double sampleRate, + unsigned long framesPerBuffer, + PaStreamFlags streamFlags, + PaStreamCallback *streamCallback, + void *userData ); + + +/** A simplified version of Pa_OpenStream() that opens the default input + and/or output devices. + + @param stream The address of a PaStream pointer which will receive + a pointer to the newly opened stream. + + @param numInputChannels The number of channels of sound that will be supplied + to the stream callback or returned by Pa_ReadStream. It can range from 1 to + the value of maxInputChannels in the PaDeviceInfo record for the default input + device. If 0 the stream is opened as an output-only stream. + + @param numOutputChannels The number of channels of sound to be delivered to the + stream callback or passed to Pa_WriteStream. It can range from 1 to the value + of maxOutputChannels in the PaDeviceInfo record for the default output dvice. + If 0 the stream is opened as an output-only stream. + + @param sampleFormat The sample format of both the input and output buffers + provided to the callback or passed to and from Pa_ReadStream and Pa_WriteStream. + sampleFormat may be any of the formats described by the PaSampleFormat + enumeration. + + @param sampleRate Same as Pa_OpenStream parameter of the same name. + @param framesPerBuffer Same as Pa_OpenStream parameter of the same name. + @param streamCallback Same as Pa_OpenStream parameter of the same name. + @param userData Same as Pa_OpenStream parameter of the same name. + + @return As for Pa_OpenStream + + @see Pa_OpenStream, PaStreamCallback +*/ +PaError Pa_OpenDefaultStream( PaStream** stream, + int numInputChannels, + int numOutputChannels, + PaSampleFormat sampleFormat, + double sampleRate, + unsigned long framesPerBuffer, + PaStreamCallback *streamCallback, + void *userData ); + + +/** Closes an audio stream. If the audio stream is active it + discards any pending buffers as if Pa_AbortStream() had been called. +*/ +PaError Pa_CloseStream( PaStream *stream ); + + +/** Functions of type PaStreamFinishedCallback are implemented by PortAudio + clients. They can be registered with a stream using the Pa_SetStreamFinishedCallback + function. Once registered they are called when the stream becomes inactive + (ie once a call to Pa_StopStream() will not block). + A stream will become inactive after the stream callback returns non-zero, + or when Pa_StopStream or Pa_AbortStream is called. For a stream providing audio + output, if the stream callback returns paComplete, or Pa_StopStream is called, + the stream finished callback will not be called until all generated sample data + has been played. + + @param userData The userData parameter supplied to Pa_OpenStream() + + @see Pa_SetStreamFinishedCallback +*/ +typedef void PaStreamFinishedCallback( void *userData ); + + +/** Register a stream finished callback function which will be called when the + stream becomes inactive. See the description of PaStreamFinishedCallback for + further details about when the callback will be called. + + @param stream a pointer to a PaStream that is in the stopped state - if the + stream is not stopped, the stream's finished callback will remain unchanged + and an error code will be returned. + + @param streamFinishedCallback a pointer to a function with the same signature + as PaStreamFinishedCallback, that will be called when the stream becomes + inactive. Passing NULL for this parameter will un-register a previously + registered stream finished callback function. + + @return on success returns paNoError, otherwise an error code indicating the cause + of the error. + + @see PaStreamFinishedCallback +*/ +PaError Pa_SetStreamFinishedCallback( PaStream *stream, PaStreamFinishedCallback* streamFinishedCallback ); + + +/** Commences audio processing. +*/ +PaError Pa_StartStream( PaStream *stream ); + + +/** Terminates audio processing. It waits until all pending + audio buffers have been played before it returns. +*/ +PaError Pa_StopStream( PaStream *stream ); + + +/** Terminates audio processing immediately without waiting for pending + buffers to complete. +*/ +PaError Pa_AbortStream( PaStream *stream ); + + +/** Determine whether the stream is stopped. + A stream is considered to be stopped prior to a successful call to + Pa_StartStream and after a successful call to Pa_StopStream or Pa_AbortStream. + If a stream callback returns a value other than paContinue the stream is NOT + considered to be stopped. + + @return Returns one (1) when the stream is stopped, zero (0) when + the stream is running or, a PaErrorCode (which are always negative) if + PortAudio is not initialized or an error is encountered. + + @see Pa_StopStream, Pa_AbortStream, Pa_IsStreamActive +*/ +PaError Pa_IsStreamStopped( PaStream *stream ); + + +/** Determine whether the stream is active. + A stream is active after a successful call to Pa_StartStream(), until it + becomes inactive either as a result of a call to Pa_StopStream() or + Pa_AbortStream(), or as a result of a return value other than paContinue from + the stream callback. In the latter case, the stream is considered inactive + after the last buffer has finished playing. + + @return Returns one (1) when the stream is active (ie playing or recording + audio), zero (0) when not playing or, a PaErrorCode (which are always negative) + if PortAudio is not initialized or an error is encountered. + + @see Pa_StopStream, Pa_AbortStream, Pa_IsStreamStopped +*/ +PaError Pa_IsStreamActive( PaStream *stream ); + + + +/** A structure containing unchanging information about an open stream. + @see Pa_GetStreamInfo +*/ + +typedef struct PaStreamInfo +{ + /** this is struct version 1 */ + int structVersion; + + /** The input latency of the stream in seconds. This value provides the most + accurate estimate of input latency available to the implementation. It may + differ significantly from the suggestedLatency value passed to Pa_OpenStream(). + The value of this field will be zero (0.) for output-only streams. + @see PaTime + */ + PaTime inputLatency; + + /** The output latency of the stream in seconds. This value provides the most + accurate estimate of output latency available to the implementation. It may + differ significantly from the suggestedLatency value passed to Pa_OpenStream(). + The value of this field will be zero (0.) for input-only streams. + @see PaTime + */ + PaTime outputLatency; + + /** The sample rate of the stream in Hertz (samples per second). In cases + where the hardware sample rate is inaccurate and PortAudio is aware of it, + the value of this field may be different from the sampleRate parameter + passed to Pa_OpenStream(). If information about the actual hardware sample + rate is not available, this field will have the same value as the sampleRate + parameter passed to Pa_OpenStream(). + */ + double sampleRate; + +} PaStreamInfo; + + +/** Retrieve a pointer to a PaStreamInfo structure containing information + about the specified stream. + @return A pointer to an immutable PaStreamInfo structure. If the stream + parameter invalid, or an error is encountered, the function returns NULL. + + @param stream A pointer to an open stream previously created with Pa_OpenStream. + + @note PortAudio manages the memory referenced by the returned pointer, + the client must not manipulate or free the memory. The pointer is only + guaranteed to be valid until the specified stream is closed. + + @see PaStreamInfo +*/ +const PaStreamInfo* Pa_GetStreamInfo( PaStream *stream ); + + +/** Determine the current time for the stream according to the same clock used + to generate buffer timestamps. This time may be used for syncronising other + events to the audio stream, for example synchronizing audio to MIDI. + + @return The stream's current time in seconds, or 0 if an error occurred. + + @see PaTime, PaStreamCallback +*/ +PaTime Pa_GetStreamTime( PaStream *stream ); + + +/** Retrieve CPU usage information for the specified stream. + The "CPU Load" is a fraction of total CPU time consumed by a callback stream's + audio processing routines including, but not limited to the client supplied + stream callback. This function does not work with blocking read/write streams. + + This function may be called from the stream callback function or the + application. + + @return + A floating point value, typically between 0.0 and 1.0, where 1.0 indicates + that the stream callback is consuming the maximum number of CPU cycles possible + to maintain real-time operation. A value of 0.5 would imply that PortAudio and + the stream callback was consuming roughly 50% of the available CPU time. The + return value may exceed 1.0. A value of 0.0 will always be returned for a + blocking read/write stream, or if an error occurrs. +*/ +double Pa_GetStreamCpuLoad( PaStream* stream ); + + +/** Read samples from an input stream. The function doesn't return until + the entire buffer has been filled - this may involve waiting for the operating + system to supply the data. + + @param stream A pointer to an open stream previously created with Pa_OpenStream. + + @param buffer A pointer to a buffer of sample frames. The buffer contains + samples in the format specified by the inputParameters->sampleFormat field + used to open the stream, and the number of channels specified by + inputParameters->numChannels. If non-interleaved samples were requested, + buffer is a pointer to the first element of an array of non-interleaved + buffer pointers, one for each channel. + + @param frames The number of frames to be read into buffer. This parameter + is not constrained to a specific range, however high performance applications + will want to match this parameter to the framesPerBuffer parameter used + when opening the stream. + + @return On success PaNoError will be returned, or PaInputOverflowed if input + data was discarded by PortAudio after the previous call and before this call. +*/ +PaError Pa_ReadStream( PaStream* stream, + void *buffer, + unsigned long frames ); + + +/** Write samples to an output stream. This function doesn't return until the + entire buffer has been consumed - this may involve waiting for the operating + system to consume the data. + + @param stream A pointer to an open stream previously created with Pa_OpenStream. + + @param buffer A pointer to a buffer of sample frames. The buffer contains + samples in the format specified by the outputParameters->sampleFormat field + used to open the stream, and the number of channels specified by + outputParameters->numChannels. If non-interleaved samples were requested, + buffer is a pointer to the first element of an array of non-interleaved + buffer pointers, one for each channel. + + @param frames The number of frames to be written from buffer. This parameter + is not constrained to a specific range, however high performance applications + will want to match this parameter to the framesPerBuffer parameter used + when opening the stream. + + @return On success PaNoError will be returned, or paOutputUnderflowed if + additional output data was inserted after the previous call and before this + call. +*/ +PaError Pa_WriteStream( PaStream* stream, + const void *buffer, + unsigned long frames ); + + +/** Retrieve the number of frames that can be read from the stream without + waiting. + + @return Returns a non-negative value representing the maximum number of frames + that can be read from the stream without blocking or busy waiting or, a + PaErrorCode (which are always negative) if PortAudio is not initialized or an + error is encountered. +*/ +signed long Pa_GetStreamReadAvailable( PaStream* stream ); + + +/** Retrieve the number of frames that can be written to the stream without + waiting. + + @return Returns a non-negative value representing the maximum number of frames + that can be written to the stream without blocking or busy waiting or, a + PaErrorCode (which are always negative) if PortAudio is not initialized or an + error is encountered. +*/ +signed long Pa_GetStreamWriteAvailable( PaStream* stream ); + + +/* Miscellaneous utilities */ + + +/** Retrieve the size of a given sample format in bytes. + + @return The size in bytes of a single sample in the specified format, + or paSampleFormatNotSupported if the format is not supported. +*/ +PaError Pa_GetSampleSize( PaSampleFormat format ); + + +/** Put the caller to sleep for at least 'msec' milliseconds. This function is + provided only as a convenience for authors of portable code (such as the tests + and examples in the PortAudio distribution.) + + The function may sleep longer than requested so don't rely on this for accurate + musical timing. +*/ +void Pa_Sleep( long msec ); + + + +#ifdef __cplusplus +} +#endif /* __cplusplus */ +#endif /* PORTAUDIO_H */ -- cgit v1.3