X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.cpp;h=f5b324f22539d713e55310196e747ff37dfd7902;hb=f6708f5edc4719fdc5a54cfcb11159d197dd6dcd;hp=4a51baf75632e8d52038676d3214d0487c54e630;hpb=a3d2ee35944db4dd0a3a342bb7f2df69f229f45d;p=rtaudio.git diff --git a/RtAudio.cpp b/RtAudio.cpp index 4a51baf..f5b324f 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -1,16 +1,16 @@ /************************************************************************/ /*! \class RtAudio - \brief Realtime audio i/o C++ class. + \brief Realtime audio i/o C++ classes. RtAudio provides a common API (Application Programming Interface) - for realtime audio input/output across Linux (native ALSA and - OSS), SGI, Macintosh OS X (CoreAudio), and Windows (DirectSound - and ASIO) operating systems. + for realtime audio input/output across Linux (native ALSA, Jack, + and OSS), SGI, Macintosh OS X (CoreAudio and Jack), and Windows + (DirectSound and ASIO) operating systems. - RtAudio WWW site: http://www-ccrma.stanford.edu/~gary/rtaudio/ + RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ - RtAudio: a realtime audio i/o C++ class - Copyright (c) 2001-2002 Gary P. Scavone + RtAudio: realtime audio i/o C++ classes + Copyright (c) 2001-2007 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -24,8 +24,9 @@ included in all copies or substantial portions of the Software. 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. + 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 @@ -37,318 +38,327 @@ */ /************************************************************************/ +// RtAudio: Version 4.0 #include "RtAudio.h" -#include -#include -#include +#include // Static variable definitions. -const unsigned int RtAudio :: SAMPLE_RATES[] = { +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 }; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT8 = 1; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT16 = 2; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT24 = 4; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT32 = 8; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT32 = 16; -const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT64 = 32; #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) -#else // pthread API +#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 // *************************************************** // // -// Public common (OS-independent) methods. +// RtAudio definitions. // // *************************************************** // -RtAudio :: RtAudio() +void RtAudio :: getCompiledApi( std::vector &apis ) throw() { - initialize(); + 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 +} - if (nDevices <= 0) { - sprintf(message, "RtAudio: no audio devices found!"); - error(RtError::NO_DEVICES_FOUND); - } +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(int *streamId, - int outputDevice, int outputChannels, - int inputDevice, int inputChannels, - RTAUDIO_FORMAT format, int sampleRate, - int *bufferSize, int numberOfBuffers) +RtAudio :: RtAudio( RtAudio::Api api ) throw() { - initialize(); + rtapi_ = 0; - if (nDevices <= 0) { - sprintf(message, "RtAudio: no audio devices found!"); - error(RtError::NO_DEVICES_FOUND); - } + if ( api != UNSPECIFIED ) { + // Attempt to open the specified API. + openRtApi( api ); + if ( rtapi_ ) return; - try { - *streamId = openStream(outputDevice, outputChannels, inputDevice, inputChannels, - format, sampleRate, bufferSize, numberOfBuffers); + // 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; } - catch (RtError &exception) { - // deallocate the RTAUDIO_DEVICE structures - if (devices) free(devices); - throw exception; + + // 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_; } -RtAudio :: ~RtAudio() +void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options ) { - // close any existing streams - while ( streams.size() ) - closeStream( streams.begin()->first ); + return rtapi_->openStream( outputParameters, inputParameters, format, + sampleRate, bufferFrames, callback, + userData, options ); +} + +// *************************************************** // +// +// Public RtApi definitions (see end of file for +// private or protected utility functions). +// +// *************************************************** // - // deallocate the RTAUDIO_DEVICE structures - if (devices) free(devices); +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; } -int RtAudio :: openStream(int outputDevice, int outputChannels, - int inputDevice, int inputChannels, - RTAUDIO_FORMAT format, int sampleRate, - int *bufferSize, int numberOfBuffers) +RtApi :: ~RtApi() { - static int streamKey = 0; // Unique stream identifier ... OK for multiple instances. + MUTEX_DESTROY( &stream_.mutex ); +} - if (outputChannels < 1 && inputChannels < 1) { - sprintf(message,"RtAudio: one or both 'channel' parameters must be greater than zero."); - error(RtError::INVALID_PARAMETER); +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 ( formatBytes(format) == 0 ) { - sprintf(message,"RtAudio: 'format' parameter value is undefined."); - error(RtError::INVALID_PARAMETER); + 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 ( outputChannels > 0 ) { - if (outputDevice > nDevices || outputDevice < 0) { - sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice); - error(RtError::INVALID_PARAMETER); - } + 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 ( inputChannels > 0 ) { - if (inputDevice > nDevices || inputDevice < 0) { - sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice); - error(RtError::INVALID_PARAMETER); - } + if ( oParams == NULL && iParams == NULL ) { + errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; + error( RtError::INVALID_USE ); } - // Allocate a new stream structure. - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) calloc(1, sizeof(RTAUDIO_STREAM)); - if (stream == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); + if ( formatBytes(format) == 0 ) { + errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; + error( RtError::INVALID_USE ); } - stream->mode = UNINITIALIZED; - MUTEX_INITIALIZE(&stream->mutex); - - bool result = FAILURE; - int device, defaultDevice = 0; - STREAM_MODE mode; - int channels; - if ( outputChannels > 0 ) { - - mode = OUTPUT; - channels = outputChannels; - if ( outputDevice == 0 ) { // Try default device first. - defaultDevice = getDefaultOutputDevice(); - device = defaultDevice; + 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 ); } - else - device = outputDevice - 1; + } - for (int i=-1; i= 0 ) { - if ( i == defaultDevice ) continue; - device = i; - } - if (devices[device].probed == false) { - // If the device wasn't successfully probed before, try it - // again now. - clearDeviceInfo(&devices[device]); - probeDeviceInfo(&devices[device]); - } - if ( devices[device].probed ) - result = probeDeviceOpen(device, stream, mode, channels, sampleRate, - format, bufferSize, numberOfBuffers); - if (result == SUCCESS) break; - if ( outputDevice > 0 ) break; + 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 ); } } - if ( inputChannels > 0 && ( result == SUCCESS || outputChannels <= 0 ) ) { + clearStreamInfo(); + bool result; - mode = INPUT; - channels = inputChannels; + if ( oChannels > 0 ) { - if ( inputDevice == 0 ) { // Try default device first. - defaultDevice = getDefaultInputDevice(); - device = defaultDevice; - } - else - device = inputDevice - 1; + result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) error( RtError::SYSTEM_ERROR ); + } - for (int i=-1; i= 0 ) { - if ( i == defaultDevice ) continue; - device = i; - } - if (devices[device].probed == false) { - // If the device wasn't successfully probed before, try it - // again now. - clearDeviceInfo(&devices[device]); - probeDeviceInfo(&devices[device]); - } - if ( devices[device].probed ) - result = probeDeviceOpen(device, stream, mode, channels, sampleRate, - format, bufferSize, numberOfBuffers); - if (result == SUCCESS) break; - if ( outputDevice > 0 ) break; + 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 ); } } - streams[++streamKey] = (void *) stream; - if ( result == SUCCESS ) - return streamKey; - - // If we get here, all attempted probes failed. Close any opened - // devices and delete the allocated stream. - closeStream(streamKey); - if ( ( outputDevice == 0 && outputChannels > 0 ) - || ( inputDevice == 0 && inputChannels > 0 ) ) - sprintf(message,"RtAudio: no devices found for given parameters."); - else - sprintf(message,"RtAudio: unable to open specified device(s) with given stream parameters."); - error(RtError::INVALID_PARAMETER); + stream_.callbackInfo.callback = (void *) callback; + stream_.callbackInfo.userData = userData; - return -1; + if ( options ) options->numberOfBuffers = stream_.nBuffers; + stream_.state = STREAM_STOPPED; } -int RtAudio :: getDeviceCount(void) +unsigned int RtApi :: getDefaultInputDevice( void ) { - return nDevices; + // Should be implemented in subclasses if possible. + return 0; } -void RtAudio :: getDeviceInfo(int device, RTAUDIO_DEVICE *info) +unsigned int RtApi :: getDefaultOutputDevice( void ) { - if (device > nDevices || device < 1) { - sprintf(message, "RtAudio: invalid device specifier (%d)!", device); - error(RtError::INVALID_DEVICE); - } - - int deviceIndex = device - 1; - - // If the device wasn't successfully probed before, try it now (or again). - if (devices[deviceIndex].probed == false) { - clearDeviceInfo(&devices[deviceIndex]); - probeDeviceInfo(&devices[deviceIndex]); - } - - // Clear the info structure. - memset(info, 0, sizeof(RTAUDIO_DEVICE)); - - strncpy(info->name, devices[deviceIndex].name, 128); - info->probed = devices[deviceIndex].probed; - if ( info->probed == true ) { - info->maxOutputChannels = devices[deviceIndex].maxOutputChannels; - info->maxInputChannels = devices[deviceIndex].maxInputChannels; - info->maxDuplexChannels = devices[deviceIndex].maxDuplexChannels; - info->minOutputChannels = devices[deviceIndex].minOutputChannels; - info->minInputChannels = devices[deviceIndex].minInputChannels; - info->minDuplexChannels = devices[deviceIndex].minDuplexChannels; - info->hasDuplexSupport = devices[deviceIndex].hasDuplexSupport; - info->nSampleRates = devices[deviceIndex].nSampleRates; - if (info->nSampleRates == -1) { - info->sampleRates[0] = devices[deviceIndex].sampleRates[0]; - info->sampleRates[1] = devices[deviceIndex].sampleRates[1]; - } - else { - for (int i=0; inSampleRates; i++) - info->sampleRates[i] = devices[deviceIndex].sampleRates[i]; - } - info->nativeFormats = devices[deviceIndex].nativeFormats; - if ( deviceIndex == getDefaultOutputDevice() || - deviceIndex == getDefaultInputDevice() ) - info->isDefault = true; - } + // Should be implemented in subclasses if possible. + return 0; +} +void RtApi :: closeStream( void ) +{ + // MUST be implemented in subclasses! return; } -char * const RtAudio :: getStreamBuffer(int streamId) +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 ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - return stream->userBuffer; + // MUST be implemented in subclasses! + return FAILURE; } -#if defined(__LINUX_ALSA__) || defined(__LINUX_OSS__) || defined(__IRIX_AL__) - -extern "C" void *callbackHandler(void * ptr); - -void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +void RtApi :: tickStreamTime( void ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo; - if ( info->usingCallback ) { - sprintf(message, "RtAudio: A callback is already set for this stream!"); - error(RtError::WARNING); - return; - } + // 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. - info->callback = (void *) callback; - info->userData = userData; - info->usingCallback = true; - info->object = (void *) this; - info->streamId = streamId; + stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); - int err = pthread_create(&info->thread, NULL, callbackHandler, &stream->callbackInfo); - - if (err) { - info->usingCallback = false; - sprintf(message, "RtAudio: error starting callback thread!"); - error(RtError::THREAD_ERROR); - } +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif } -void RtAudio :: cancelStreamCallback(int streamId) +long RtApi :: getStreamLatency( void ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->callbackInfo.usingCallback) { - - if (stream->state == STREAM_RUNNING) - stopStream( streamId ); - - MUTEX_LOCK(&stream->mutex); + verifyStream(); - stream->callbackInfo.usingCallback = false; - pthread_cancel(stream->callbackInfo.thread); - pthread_join(stream->callbackInfo.thread, NULL); - stream->callbackInfo.thread = 0; - stream->callbackInfo.callback = NULL; - stream->callbackInfo.userData = NULL; + 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]; - MUTEX_UNLOCK(&stream->mutex); - } + 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 +} + // *************************************************** // // @@ -362,3236 +372,2724 @@ void RtAudio :: cancelStreamCallback(int streamId) // 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. This same functionality can be -// achieved with better synchrony by opening two separate streams for -// the devices and using RtAudio blocking calls (i.e. tickStream()). -// -// The possibility of having multiple RtAudio streams accessing the -// same CoreAudio device is not currently supported. The problem -// involves the inability to install our callbackHandler function for -// the same device more than once. I experimented with a workaround -// for this, but it requires an additional buffer for mixing output -// data before filling the CoreAudio device buffer. In the end, I -// decided it wasn't worth supporting. +// synchronize these two callbacks. // -// Property listeners are currently not used. The issue is what could +// 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. Some initial listener code is -// commented out. +// be prepared for the result anyway. However, we do provide a flag +// to the client callback function to inform of an over/underrun. +// +// The mechanism for querying and setting system parameters was +// updated (and perhaps simplified) in OS-X version 10.4. However, +// since 10.4 support is not necessarily available to all users, I've +// decided not to update the respective code at this time. Perhaps +// this will happen when Apple makes 10.4 free for everyone. :-) + +// A structure to hold various information related to the CoreAudio API +// implementation. +struct CoreHandle { + AudioDeviceID id[2]; // device ids + UInt32 iStream[2]; // device stream index (first for mono mode) + 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) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; -void RtAudio :: initialize(void) +RtApiCore :: RtApiCore() { - OSStatus err = noErr; - UInt32 dataSize; - AudioDeviceID *deviceList = NULL; - nDevices = 0; - - // Find out how many audio devices there are, if any. - err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &dataSize, NULL); - if (err != noErr) { - sprintf(message, "RtAudio: OSX error getting device info!"); - error(RtError::SYSTEM_ERROR); - } - - nDevices = dataSize / sizeof(AudioDeviceID); - if (nDevices == 0) return; - - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } - - // Make space for the devices we are about to get. - deviceList = (AudioDeviceID *) malloc( dataSize ); - if (deviceList == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } + // Nothing to do here. +} - // Get the array of AudioDeviceIDs. - err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &dataSize, (void *) deviceList); - if (err != noErr) { - free(deviceList); - sprintf(message, "RtAudio: OSX error getting device properties!"); - error(RtError::SYSTEM_ERROR); - } +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(); +} - // Write device identifiers to device structures and then - // probe the device capabilities. - for (int i=0; i= nDevices ) { + errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); } - return false; -} + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; + error( RtError::WARNING ); + return info; + } -void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) -{ - OSStatus err = noErr; + AudioDeviceID id = deviceList[ device ]; - // Get the device manufacturer and name. + // Get the device name. + info.name.erase(); char name[256]; - char fullname[512]; - UInt32 dataSize = 256; - err = AudioDeviceGetProperty( info->id[0], 0, false, - kAudioDevicePropertyDeviceManufacturer, - &dataSize, name ); - if (err != noErr) { - sprintf( message, "RtAudio: OSX error getting device manufacturer." ); - error(RtError::DEBUG_WARNING); - return; - } - strncpy(fullname, name, 256); - strcat(fullname, ": " ); - dataSize = 256; - err = AudioDeviceGetProperty( info->id[0], 0, false, - kAudioDevicePropertyDeviceName, - &dataSize, name ); - if (err != noErr) { - sprintf( message, "RtAudio: OSX error getting device name." ); - error(RtError::DEBUG_WARNING); - return; + result = AudioDeviceGetProperty( id, 0, false, + kAudioDevicePropertyDeviceManufacturer, + &dataSize, name ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - strncat(fullname, name, 254); - strncat(info->name, fullname, 128); + info.name.append( (const char *)name, strlen(name) ); + info.name.append( ": " ); - // Get output channel information. - unsigned int i, minChannels, maxChannels, nStreams = 0; + dataSize = 256; + result = AudioDeviceGetProperty( id, 0, false, + kAudioDevicePropertyDeviceName, + &dataSize, name ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + info.name.append( (const char *)name, strlen(name) ); + + // Get the output stream "configuration". AudioBufferList *bufferList = nil; - err = AudioDeviceGetPropertyInfo( info->id[0], 0, false, - kAudioDevicePropertyStreamConfiguration, - &dataSize, NULL ); - if (err == noErr && dataSize > 0) { - bufferList = (AudioBufferList *) malloc( dataSize ); - if (bufferList == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::DEBUG_WARNING); - return; - } - - err = AudioDeviceGetProperty( info->id[0], 0, false, - kAudioDevicePropertyStreamConfiguration, - &dataSize, bufferList ); - if (err == noErr) { - maxChannels = 0; - minChannels = 1000; - nStreams = bufferList->mNumberBuffers; - for ( i=0; imBuffers[i].mNumberChannels; - if ( bufferList->mBuffers[i].mNumberChannels < minChannels ) - minChannels = bufferList->mBuffers[i].mNumberChannels; - } - } - } - if (err != noErr || dataSize <= 0) { - sprintf( message, "RtAudio: OSX error getting output channels for device (%s).", info->name ); - error(RtError::DEBUG_WARNING); - return; + result = AudioDeviceGetPropertyInfo( id, 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + 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 = AudioDeviceGetProperty( id, 0, false, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if ( result != noErr ) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - free (bufferList); - if ( nStreams ) { - if ( maxChannels > 0 ) - info->maxOutputChannels = maxChannels; - if ( minChannels > 0 ) - info->minOutputChannels = minChannels; + // Get output channel information. + unsigned int i, nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // Get the input stream "configuration". + result = AudioDeviceGetPropertyInfo( id, 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + 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 = AudioDeviceGetProperty( id, 0, true, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if ( result != noErr ) { + 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. - bufferList = nil; - err = AudioDeviceGetPropertyInfo( info->id[0], 0, true, - kAudioDevicePropertyStreamConfiguration, - &dataSize, NULL ); - if (err == noErr && dataSize > 0) { - bufferList = (AudioBufferList *) malloc( dataSize ); - if (bufferList == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::DEBUG_WARNING); - return; - } - err = AudioDeviceGetProperty( info->id[0], 0, true, - kAudioDevicePropertyStreamConfiguration, - &dataSize, bufferList ); - if (err == noErr) { - maxChannels = 0; - minChannels = 1000; - nStreams = bufferList->mNumberBuffers; - for ( i=0; imBuffers[i].mNumberChannels < minChannels ) - minChannels = bufferList->mBuffers[i].mNumberChannels; - maxChannels += bufferList->mBuffers[i].mNumberChannels; - } - } - } - if (err != noErr || dataSize <= 0) { - sprintf( message, "RtAudio: OSX error getting input channels for device (%s).", info->name ); - error(RtError::DEBUG_WARNING); - return; - } - - free (bufferList); - if ( nStreams ) { - if ( maxChannels > 0 ) - info->maxInputChannels = maxChannels; - if ( minChannels > 0 ) - info->minInputChannels = minChannels; - } + 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->maxOutputChannels > 0 && info->maxInputChannels > 0) { - info->hasDuplexSupport = true; - info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? - info->maxInputChannels : info->maxOutputChannels; - info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? - info->minInputChannels : info->minOutputChannels; - } - - // Probe the device sample rate and data format parameters. The - // core audio query mechanism is performed on a "stream" - // description, which can have a variable number of channels and - // apply to input or output only. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - // Create a stream description structure. - AudioStreamBasicDescription description; - dataSize = sizeof( AudioStreamBasicDescription ); - memset(&description, 0, sizeof(AudioStreamBasicDescription)); + // Probe the device sample rates. bool isInput = false; - if ( info->maxOutputChannels == 0 ) isInput = true; - bool isDuplex = false; - if ( info->maxDuplexChannels > 0 ) isDuplex = true; + if ( info.outputChannels == 0 ) isInput = true; // Determine the supported sample rates. - info->nSampleRates = 0; - for (i=0; iid[0], isInput, &description, isDuplex ) ) - info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i]; - } - - if (info->nSampleRates == 0) { - sprintf( message, "RtAudio: No supported sample rates found for OSX device (%s).", info->name ); - error(RtError::DEBUG_WARNING); - return; - } + result = AudioDeviceGetPropertyInfo( id, 0, isInput, + kAudioDevicePropertyAvailableNominalSampleRates, + &dataSize, NULL ); - // Check for continuous sample rate support. - description.mSampleRate = kAudioStreamAnyRate; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) { - info->sampleRates[1] = info->sampleRates[info->nSampleRates-1]; - info->nSampleRates = -1; + if ( result != kAudioHardwareNoError || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - // Determine the supported data formats. - info->nativeFormats = 0; - description.mFormatID = kAudioFormatLinearPCM; - description.mBitsPerChannel = 8; - description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT8; - else { - description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT8; - } + UInt32 nRanges = dataSize / sizeof( AudioValueRange ); + AudioValueRange rangeList[ nRanges ]; + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyAvailableNominalSampleRates, + &dataSize, &rangeList ); - description.mBitsPerChannel = 16; - description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT16; - else { - description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT16; + if ( result != kAudioHardwareNoError ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - description.mBitsPerChannel = 32; - description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT32; - else { - description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT32; + Float64 minimumRate = 100000000.0, maximumRate = 0.0; + for ( UInt32 i=0; i maximumRate ) maximumRate = rangeList[i].mMaximum; } - description.mBitsPerChannel = 24; - description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsAlignedHigh | kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT24; - else { - description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_SINT24; + 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] ); } - description.mBitsPerChannel = 32; - description.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_FLOAT32; - else { - description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_FLOAT32; + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - description.mBitsPerChannel = 64; - description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_FLOAT64; - else { - description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian; - if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) - info->nativeFormats |= RTAUDIO_FLOAT64; - } + // 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; - // Check that we have at least one supported format. - if (info->nativeFormats == 0) { - sprintf(message, "RtAudio: OSX PCM device (%s) data format not supported by RtAudio.", - info->name); - error(RtError::DEBUG_WARNING); - return; - } + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + if ( getDefaultInputDevice() == device ) + info.isDefaultInput = true; - info->probed = 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) +OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* inNow, + const AudioBufferList* inInputData, + const AudioTimeStamp* inInputTime, + AudioBufferList* outOutputData, + const AudioTimeStamp* inOutputTime, + void* infoPointer ) { - CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer; + CallbackInfo *info = (CallbackInfo *) infoPointer; - RtAudio *object = (RtAudio *) info->object; - try { - object->callbackEvent( info->streamId, inDevice, (void *)inInputData, (void *)outOutputData ); - } - catch (RtError &exception) { - fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage()); + RtApiCore *object = (RtApiCore *) info->object; + if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) return kAudioHardwareUnspecifiedError; + else + return kAudioHardwareNoError; +} + +OSStatus deviceListener( AudioDeviceID inDevice, + UInt32 channel, + Boolean isInput, + AudioDevicePropertyID propertyID, + void* handlePointer ) +{ + CoreHandle *handle = (CoreHandle *) handlePointer; + if ( propertyID == kAudioDeviceProcessorOverload ) { + if ( isInput ) + handle->xrun[1] = true; + else + handle->xrun[0] = true; } return kAudioHardwareNoError; } -/* -OSStatus deviceListener(AudioDeviceID inDevice, - UInt32 channel, - Boolean isInput, - AudioDevicePropertyID propertyID, - void* infoPointer) +static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property ) { - CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer; + OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL ); + return result == 0; +} - RtAudio *object = (RtAudio *) info->object; - try { - object->settingChange( info->streamId ); - } - catch (RtError &exception) { - fprintf(stderr, "\nDevice listener error (%s)!\n\n", exception.getMessage()); - return kAudioHardwareUnspecifiedError; +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; } - return kAudioHardwareNoError; -} -*/ + 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; + } -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ - // Check to make sure we don't already have a stream accessing this device. - RTAUDIO_STREAM *streamPtr; - std::map::const_iterator i; - for ( i=streams.begin(); i!=streams.end(); ++i ) { - streamPtr = (RTAUDIO_STREAM *) i->second; - if ( streamPtr->device[0] == device || streamPtr->device[1] == device ) { - sprintf(message, "RtAudio: no current OS X support for multiple streams accessing the same device!"); - error(RtError::WARNING); - return FAILURE; - } + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &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; - AudioDeviceID id = devices[device].id[0]; if ( mode == INPUT ) isInput = true; - // Search for a stream which contains the desired number of channels. - OSStatus err = noErr; - UInt32 dataSize; - unsigned int deviceChannels, nStreams; + // Set or disable "hog" mode. + dataSize = sizeof( UInt32 ); + UInt32 doHog = 0; + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1; + result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the stream "configuration". + AudioBufferList *bufferList; + result = AudioDeviceGetPropertyInfo( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, NULL ); + 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 = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyStreamConfiguration, + &dataSize, bufferList ); + if ( result != noErr ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Search for a stream that contains 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 currently only supports the use of one stream + // of interleaved data or multiple consecutive single-channel + // streams. Thus, our search below is limited to these two + // contexts. + unsigned int streamChannels = 0, nStreams = 0; UInt32 iChannel = 0, iStream = 0; - AudioBufferList *bufferList = nil; - err = AudioDeviceGetPropertyInfo( id, 0, isInput, - kAudioDevicePropertyStreamConfiguration, - &dataSize, NULL ); - - if (err == noErr && dataSize > 0) { - bufferList = (AudioBufferList *) malloc( dataSize ); - if (bufferList == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::DEBUG_WARNING); - return FAILURE; + unsigned int offsetCounter = firstChannel; + stream_.deviceInterleaved[mode] = true; + nStreams = bufferList->mNumberBuffers; + bool foundStream = false; + + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels >= channels + offsetCounter ) { + iChannel += offsetCounter; + foundStream = true; + break; } - err = AudioDeviceGetProperty( id, 0, isInput, - kAudioDevicePropertyStreamConfiguration, - &dataSize, bufferList ); - - if (err == noErr) { - stream->deInterleave[mode] = false; - nStreams = bufferList->mNumberBuffers; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels >= (unsigned int) channels ) break; - iChannel += bufferList->mBuffers[iStream].mNumberChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + iChannel += streamChannels; + } + + // If we didn't find a single stream above, see if we can meet + // the channel specification in mono mode (i.e. using separate + // non-interleaved buffers). This can only work if there are N + // consecutive one-channel streams, where N is the number of + // desired channels (+ channel offset). + if ( foundStream == false ) { + unsigned int counter = 0; + offsetCounter = firstChannel; + iChannel = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( offsetCounter ) { + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; } - // If we didn't find a single stream above, see if we can meet - // the channel specification in mono mode (i.e. using separate - // non-interleaved buffers). This can only work if there are N - // consecutive one-channel streams, where N is the number of - // desired channels. - iChannel = 0; - if ( iStream >= nStreams && nStreams >= (unsigned int) channels ) { - int counter = 0; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels == 1 ) - counter++; - else - counter = 0; - if ( counter == channels ) { - iStream -= channels - 1; - iChannel -= channels - 1; - stream->deInterleave[mode] = true; - break; - } - iChannel += bufferList->mBuffers[iStream].mNumberChannels; - } + else if ( streamChannels == 1 ) + counter++; + else + counter = 0; + if ( counter == channels ) { + iStream -= channels - 1; + iChannel -= channels - 1; + stream_.deviceInterleaved[mode] = false; + foundStream = true; + break; } + iChannel += streamChannels; } } - if (err != noErr || dataSize <= 0) { - if ( bufferList ) free( bufferList ); - sprintf( message, "RtAudio: OSX error getting channels for device (%s).", devices[device].name ); - error(RtError::DEBUG_WARNING); - return FAILURE; - } + free( bufferList ); - if (iStream >= nStreams) { - free (bufferList); - sprintf( message, "RtAudio: unable to find OSX audio stream on device (%s) for requested channels (%d).", - devices[device].name, channels ); - error(RtError::DEBUG_WARNING); + if ( foundStream == false ) { + errorStream_ << "RtApiCore::probeDeviceOpen: unable to find OS-X stream on device (" << device << ") for requested channels."; + errorText_ = errorStream_.str(); return FAILURE; } - // This is ok even for mono mode ... it gets updated later. - deviceChannels = bufferList->mBuffers[iStream].mNumberChannels; - free (bufferList); - // Determine the buffer size. AudioValueRange bufferRange; - dataSize = sizeof(AudioValueRange); - err = AudioDeviceGetProperty( id, 0, isInput, - kAudioDevicePropertyBufferSizeRange, - &dataSize, &bufferRange); - if (err != noErr) { - sprintf( message, "RtAudio: OSX error getting buffer size range for device (%s).", - devices[device].name ); - error(RtError::DEBUG_WARNING); + dataSize = sizeof( AudioValueRange ); + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyBufferFrameSizeRange, + &dataSize, &bufferRange ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; + errorText_ = errorStream_.str(); return FAILURE; } - long bufferBytes = *bufferSize * deviceChannels * formatBytes(RTAUDIO_FLOAT32); - if (bufferRange.mMinimum > bufferBytes) bufferBytes = (int) bufferRange.mMinimum; - else if (bufferRange.mMaximum < bufferBytes) bufferBytes = (int) bufferRange.mMaximum; + 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 mono mode, I'm assuming we only need to - // make this setting for the first channel. - UInt32 theSize = (UInt32) bufferBytes; - dataSize = sizeof( UInt32); - err = AudioDeviceSetProperty(id, NULL, 0, isInput, - kAudioDevicePropertyBufferSize, - dataSize, &theSize); - if (err != noErr) { - sprintf( message, "RtAudio: OSX error setting the buffer size for device (%s).", - devices[device].name ); - error(RtError::DEBUG_WARNING); + // make this setting for the master channel. + UInt32 theSize = (UInt32) *bufferSize; + dataSize = sizeof( UInt32 ); + result = AudioDeviceSetProperty( id, NULL, 0, isInput, + kAudioDevicePropertyBufferFrameSize, + 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 = bufferBytes / ( deviceChannels * formatBytes(RTAUDIO_FLOAT32) ); - if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) { - sprintf( message, "RtAudio: OSX error setting buffer size for duplex stream on device (%s).", - devices[device].name ); - error(RtError::DEBUG_WARNING); + *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; + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + + // Get the stream ID(s) so we can set the stream format. In mono + // mode, we'll have to do this for each stream (channel). + AudioStreamID streamIDs[ nStreams ]; + dataSize = nStreams * sizeof( AudioStreamID ); + result = AudioDeviceGetProperty( id, 0, isInput, + kAudioDevicePropertyStreams, + &dataSize, &streamIDs ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the stream format description. Do for each channel in mono mode. + // Now set the stream format. Also, check the physical format of the + // device and change that if necessary. AudioStreamBasicDescription description; dataSize = sizeof( AudioStreamBasicDescription ); - if ( stream->deInterleave[mode] ) nStreams = channels; - else nStreams = 1; - for ( unsigned int i=0; i 1.0 ) { + description.mSampleRate = (double) sampleRate; + updateFormat = true; + } + + if ( description.mFormatID != kAudioFormatLinearPCM ) { + description.mFormatID = kAudioFormatLinearPCM; + updateFormat = true; + } + + if ( updateFormat ) { + result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, + kAudioStreamPropertyVirtualFormat, + 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. + result = AudioStreamGetProperty( streamIDs[iStream+i], 0, + kAudioStreamPropertyPhysicalFormat, + &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; + errorText_ = errorStream_.str(); return FAILURE; } + + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) { + description.mFormatID = kAudioFormatLinearPCM; + AudioStreamBasicDescription testDescription = description; + unsigned long formatFlags; + + // We'll try higher bit rates first and then work our way down. + testDescription.mBitsPerChannel = 32; + formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 32; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 24; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 16; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result == noErr ) continue; + + testDescription = description; + testDescription.mBitsPerChannel = 8; + testDescription.mFormatFlags = formatFlags; + result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } } - // Check whether we need byte-swapping (assuming OS X host is big-endian). - iChannel -= nStreams; - err = AudioDeviceGetProperty( id, iChannel, isInput, - kAudioDevicePropertyStreamFormat, - &dataSize, &description ); - if (err != noErr) { - sprintf( message, "RtAudio: OSX error getting stream format for device (%s).", devices[device].name ); - error(RtError::DEBUG_WARNING); - return FAILURE; + // Get the stream latency. There can be latency in both the device + // and the stream. First, attempt to get the device latency on the + // master channel or the first open channel. Errors that might + // occur here are not deemed critical. + UInt32 latency, channel = 0; + dataSize = sizeof( UInt32 ); + AudioDevicePropertyID property = kAudioDevicePropertyLatency; + for ( int i=0; i<2; i++ ) { + if ( hasProperty( id, channel, isInput, property ) == true ) break; + channel = iChannel + 1 + i; + } + if ( channel <= iChannel + 1 ) { + result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency ); + if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency; + else { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + } + + // Now try to get the stream latency. For "mono" mode, I assume the + // latency is equal for all single-channel streams. + result = AudioStreamGetProperty( streamIDs[iStream], 0, property, &dataSize, &latency ); + if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency; + else { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); } - stream->doByteSwap[mode] = false; - if ( !description.mFormatFlags & kLinearPCMFormatFlagIsBigEndian ) - stream->doByteSwap[mode] = true; + // Byte-swapping: According to AudioHardware.h, the stream data will + // always be presented in native-endian format, so we should never + // need to byte swap. + stream_.doByteSwap[mode] = false; // From the CoreAudio documentation, PCM data must be supplied as // 32-bit floats. - stream->userFormat = format; - stream->deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + + if ( stream_.deviceInterleaved[mode] ) + stream_.nDeviceChannels[mode] = description.mChannelsPerFrame; + else // mono mode + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = iChannel; // offset within a CoreAudio stream + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // 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 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 ( stream->deInterleave[mode] ) - stream->nDeviceChannels[mode] = channels; + if ( pthread_cond_init( &handle->condition, NULL ) ) { + errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + } else - stream->nDeviceChannels[mode] = description.mChannelsPerFrame; - stream->nUserChannels[mode] = channels; - - // Set handle and flags for buffer conversion. - stream->handle[mode] = iStream; - 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->nUserChannels[mode] > 1 && stream->deInterleave[mode]) - stream->doConvertBuffer[mode] = true; + handle = (CoreHandle *) stream_.apiHandle; + handle->iStream[mode] = iStream; + handle->id[mode] = id; // Allocate necessary internal buffers. - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; + unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; + goto error; } - if ( stream->deInterleave[mode] ) { + // If possible, we will make use of the CoreAudio stream buffers as + // "device buffers". However, we can't do this if the device + // buffers are non-interleaved ("mono" mode). + if ( !stream_.deviceInterleaved[mode] && stream_.doConvertBuffer[mode] ) { - long buffer_bytes; bool makeBuffer = true; - if ( mode == OUTPUT ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == INPUT - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == OUTPUT && stream->deviceBuffer ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes < bytes_out ) makeBuffer = false; + 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 ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; - - // If not de-interleaving, we point stream->deviceBuffer to the - // OS X supplied device buffer before doing any necessary data - // conversions. This presents a problem if we have a duplex - // stream using one device which needs de-interleaving and - // another device which doesn't. So, save a pointer to our own - // device buffer in the CALLBACK_INFO structure. - stream->callbackInfo.buffers = stream->deviceBuffer; - } - } - - stream->sampleRate = sampleRate; - stream->device[mode] = device; - stream->state = STREAM_STOPPED; - stream->callbackInfo.object = (void *) this; - stream->callbackInfo.waitTime = (unsigned long) (200000.0 * stream->bufferSize / stream->sampleRate); - stream->callbackInfo.device[mode] = id; - if ( stream->mode == OUTPUT && mode == INPUT && stream->device[0] == device ) - // Only one callback procedure per device. - stream->mode = DUPLEX; - else { - err = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream->callbackInfo ); - if (err != noErr) { - sprintf( message, "RtAudio: OSX error setting callback for device (%s).", devices[device].name ); - error(RtError::DEBUG_WARNING); - return FAILURE; + 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; + } + + // Save a pointer to our own device buffer in the CoreHandle + // structure because we may need to use the stream_.deviceBuffer + // variable to point to the CoreAudio buffer before buffer + // conversion (if we have a duplex stream with two different + // conversion schemes). + handle->deviceBuffer = stream_.deviceBuffer; } - if ( stream->mode == OUTPUT && mode == INPUT ) - stream->mode = DUPLEX; - else - stream->mode = mode; } - // If we wanted to use property listeners, they would be setup here. + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; - return SUCCESS; + // Setup the buffer conversion information structure. We override + // the channel offset value and perform our own setting for that + // here. + if ( stream_.doConvertBuffer[mode] ) { + setConvertInfo( mode, 0 ); - memory_error: - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; + // Add channel offset for interleaved channels. + if ( firstChannel > 0 && stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; kcallbackInfo.usingCallback) { + // Setup the device property listener for over/underload. + result = AudioDeviceAddPropertyListener( id, 0, isInput, + kAudioDeviceProcessorOverload, + deviceListener, (void *) handle ); - if (stream->state == STREAM_RUNNING) - stopStream( streamId ); + return SUCCESS; - MUTEX_LOCK(&stream->mutex); + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } - stream->callbackInfo.usingCallback = false; - stream->callbackInfo.userData = NULL; - stream->state = STREAM_STOPPED; - stream->callbackInfo.callback = NULL; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - MUTEX_UNLOCK(&stream->mutex); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } + + return FAILURE; } -void RtAudio :: closeStream(int streamId) +void RtApiCore :: closeStream( void ) { - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::closeStream(): no open stream to close!"; + error( RtError::WARNING ); return; } - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; - - AudioDeviceID id; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - id = devices[stream->device[0]].id[0]; - if (stream->state == STREAM_RUNNING) - AudioDeviceStop( id, callbackHandler ); - AudioDeviceRemoveIOProc( id, callbackHandler ); + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[0], callbackHandler ); + AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); } - if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) { - id = devices[stream->device[1]].id[0]; - if (stream->state == STREAM_RUNNING) - AudioDeviceStop( id, callbackHandler ); - AudioDeviceRemoveIOProc( id, callbackHandler ); + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[1], callbackHandler ); + AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); } - pthread_mutex_destroy(&stream->mutex); + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - if (stream->userBuffer) - free(stream->userBuffer); + if ( handle->deviceBuffer ) { + free( handle->deviceBuffer ); + stream_.deviceBuffer = 0; + } - if ( stream->deInterleave[0] || stream->deInterleave[1] ) - free(stream->callbackInfo.buffers); + // Destroy pthread condition variable. + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; - free(stream); - streams.erase(streamId); + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } -void RtAudio :: startStream(int streamId) +void RtApiCore :: startStream( void ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiCore::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - if (stream->state == STREAM_RUNNING) - goto unlock; + MUTEX_LOCK( &stream_.mutex ); - OSStatus err; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - err = AudioDeviceStart(devices[stream->device[0]].id[0], callbackHandler); - if (err != noErr) { - sprintf(message, "RtAudio: OSX error starting callback procedure on device (%s).", - devices[stream->device[0]].name); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + 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]) ) { + if ( stream_.mode == INPUT || + ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - err = AudioDeviceStart(devices[stream->device[1]].id[0], callbackHandler); - if (err != noErr) { - sprintf(message, "RtAudio: OSX error starting input callback procedure on device (%s).", - devices[stream->device[0]].name); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + 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; } } - stream->callbackInfo.streamId = streamId; - stream->state = STREAM_RUNNING; - stream->callbackInfo.blockTick = true; - stream->callbackInfo.stopStream = false; + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; unlock: - MUTEX_UNLOCK(&stream->mutex); + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == noErr ) return; + error( RtError::SYSTEM_ERROR ); } -void RtAudio :: stopStream(int streamId) +void RtApiCore :: stopStream( void ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK( &stream_.mutex ); - if (stream->state == STREAM_STOPPED) - goto unlock; + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - OSStatus err; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } - err = AudioDeviceStop(devices[stream->device[0]].id[0], callbackHandler); - if (err != noErr) { - sprintf(message, "RtAudio: OSX error stopping callback procedure on device (%s).", - devices[stream->device[0]].name); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + result = AudioDeviceStop( handle->id[0], callbackHandler ); + 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]) ) { + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - err = AudioDeviceStop(devices[stream->device[1]].id[0], callbackHandler); - if (err != noErr) { - sprintf(message, "RtAudio: OSX error stopping input callback procedure on device (%s).", - devices[stream->device[0]].name); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + result = AudioDeviceStop( handle->id[1], callbackHandler ); + 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); -} + MUTEX_UNLOCK( &stream_.mutex ); -void RtAudio :: abortStream(int streamId) -{ - stopStream( streamId ); + stream_.state = STREAM_STOPPED; + if ( result == noErr ) return; + error( RtError::SYSTEM_ERROR ); } -// I don't know how this function can be implemented. -int RtAudio :: streamWillBlock(int streamId) +void RtApiCore :: abortStream( void ) { - sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for OS X."); - error(RtError::WARNING); - return 0; -} + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } -void RtAudio :: tickStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + handle->drainCounter = 1; - if (stream->state == STREAM_STOPPED) - return; + stopStream(); +} - if (stream->callbackInfo.usingCallback) { - sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!"); - error(RtError::WARNING); - return; +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; } - // Block waiting here until the user data is processed in callbackEvent(). - while ( stream->callbackInfo.blockTick ) - usleep(stream->callbackInfo.waitTime); + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - MUTEX_LOCK(&stream->mutex); + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + pthread_cond_signal( &handle->condition ); + else + stopStream(); + return SUCCESS; + } - stream->callbackInfo.blockTick = true; + MUTEX_LOCK( &stream_.mutex ); - MUTEX_UNLOCK(&stream->mutex); -} + AudioDeviceID outputDevice = handle->id[0]; -void RtAudio :: callbackEvent( int streamId, DEVICE_ID deviceId, void *inData, void *outData ) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - CALLBACK_INFO *info; - AudioBufferList *inBufferList = (AudioBufferList *) inData; - AudioBufferList *outBufferList = (AudioBufferList *) outData; - - if (stream->state == STREAM_STOPPED) return; - - info = (CALLBACK_INFO *) &stream->callbackInfo; - if ( !info->usingCallback ) { - // Block waiting here until we get new user data in tickStream(). - while ( !info->blockTick ) - usleep(info->waitTime); - } - else if ( info->stopStream ) { - // Check if the stream should be stopped (via the previous user - // callback return value). We stop the stream here, rather than - // after the function call, so that output data can first be - // processed. - this->stopStream(info->streamId); - return; + // 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; } - MUTEX_LOCK(&stream->mutex); + if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { - if ( stream->mode == INPUT || ( stream->mode == DUPLEX && deviceId == info->device[1] ) ) { + if ( handle->drainCounter > 1 ) { // write zeros to the output stream - if (stream->doConvertBuffer[1]) { - - if ( stream->deInterleave[1] ) { - stream->deviceBuffer = (char *) stream->callbackInfo.buffers; - int bufferBytes = inBufferList->mBuffers[stream->handle[1]].mDataByteSize; - for ( int i=0; inDeviceChannels[1]; i++ ) { - memcpy(&stream->deviceBuffer[i*bufferBytes], - inBufferList->mBuffers[stream->handle[1]+i].mData, bufferBytes ); + if ( stream_.deviceInterleaved[0] ) { + memset( outBufferList->mBuffers[handle->iStream[0]].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + else { + for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); } } + } + else if ( stream_.doConvertBuffer[0] ) { + + if ( stream_.deviceInterleaved[0] ) + stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->iStream[0]].mData; else - stream->deviceBuffer = (char *) inBufferList->mBuffers[stream->handle[1]].mData; + stream_.deviceBuffer = handle->deviceBuffer; + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - if ( stream->doByteSwap[1] ) - byteSwapBuffer(stream->deviceBuffer, - stream->bufferSize * stream->nDeviceChannels[1], - stream->deviceFormat[1]); - convertStreamBuffer(stream, INPUT); + if ( !stream_.deviceInterleaved[0] ) { + UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, + &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } + } } else { - memcpy(stream->userBuffer, - inBufferList->mBuffers[stream->handle[1]].mData, - inBufferList->mBuffers[stream->handle[1]].mDataByteSize ); - - if (stream->doByteSwap[1]) - byteSwapBuffer(stream->userBuffer, - stream->bufferSize * stream->nUserChannels[1], - stream->userFormat); + if ( stream_.deviceInterleaved[0] ) { + memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + else { + UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, + &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } + } } - } - // Don't invoke the user callback if duplex mode, the input/output - // devices are different, and this function is called for the output - // device. - if ( info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) ) { - RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback; - info->stopStream = callback(stream->userBuffer, stream->bufferSize, info->userData); + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } } - if ( stream->mode == OUTPUT || ( stream->mode == DUPLEX && deviceId == info->device[0] ) ) { + AudioDeviceID inputDevice = handle->id[1]; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { - if (stream->doConvertBuffer[0]) { + if ( stream_.doConvertBuffer[1] ) { - if ( !stream->deInterleave[0] ) - stream->deviceBuffer = (char *) outBufferList->mBuffers[stream->handle[0]].mData; - else - stream->deviceBuffer = (char *) stream->callbackInfo.buffers; - - convertStreamBuffer(stream, OUTPUT); - if ( stream->doByteSwap[0] ) - byteSwapBuffer(stream->deviceBuffer, - stream->bufferSize * stream->nDeviceChannels[0], - stream->deviceFormat[0]); - - if ( stream->deInterleave[0] ) { - int bufferBytes = outBufferList->mBuffers[stream->handle[0]].mDataByteSize; - for ( int i=0; inDeviceChannels[0]; i++ ) { - memcpy(outBufferList->mBuffers[stream->handle[0]+i].mData, - &stream->deviceBuffer[i*bufferBytes], bufferBytes ); + if ( stream_.deviceInterleaved[1] ) + stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->iStream[1]].mData; + else { + stream_.deviceBuffer = (char *) handle->deviceBuffer; + UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); } } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } else { - if (stream->doByteSwap[0]) - byteSwapBuffer(stream->userBuffer, - stream->bufferSize * stream->nUserChannels[0], - stream->userFormat); - - memcpy(outBufferList->mBuffers[stream->handle[0]].mData, - stream->userBuffer, - outBufferList->mBuffers[stream->handle[0]].mDataByteSize ); + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); } } - if ( !info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) ) - info->blockTick = false; - - MUTEX_UNLOCK(&stream->mutex); + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); + return SUCCESS; } -void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) +const char* RtApiCore :: getErrorCode( OSStatus code ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + switch( code ) { - stream->callbackInfo.callback = (void *) callback; - stream->callbackInfo.userData = userData; - stream->callbackInfo.usingCallback = true; -} + case kAudioHardwareNotRunningError: + return "kAudioHardwareNotRunningError"; -//******************** End of __MACOSX_CORE__ *********************// + case kAudioHardwareUnspecifiedError: + return "kAudioHardwareUnspecifiedError"; -#elif defined(__LINUX_ALSA__) + case kAudioHardwareUnknownPropertyError: + return "kAudioHardwareUnknownPropertyError"; -#define MAX_DEVICES 16 + case kAudioHardwareBadPropertySizeError: + return "kAudioHardwareBadPropertySizeError"; -void RtAudio :: initialize(void) -{ - int card, result, device; - char name[32]; - const char *cardId; - char deviceNames[MAX_DEVICES][32]; - snd_ctl_t *handle; - snd_ctl_card_info_t *info; - snd_ctl_card_info_alloca(&info); + case kAudioHardwareIllegalOperationError: + return "kAudioHardwareIllegalOperationError"; - // Count cards and devices - nDevices = 0; - card = -1; - snd_card_next(&card); - while ( card >= 0 ) { - sprintf(name, "hw:%d", card); - result = snd_ctl_open(&handle, name, 0); - if (result < 0) { - sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(result)); - error(RtError::DEBUG_WARNING); - goto next_card; - } - result = snd_ctl_card_info(handle, info); - if (result < 0) { - sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(result)); - error(RtError::DEBUG_WARNING); - goto next_card; - } - cardId = snd_ctl_card_info_get_id(info); - device = -1; - while (1) { - result = snd_ctl_pcm_next_device(handle, &device); - if (result < 0) { - sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(result)); - error(RtError::DEBUG_WARNING); - break; - } - if (device < 0) - break; - if ( strlen(cardId) ) - sprintf( deviceNames[nDevices++], "hw:%s,%d", cardId, device ); - else - sprintf( deviceNames[nDevices++], "hw:%d,%d", card, device ); - if ( nDevices > MAX_DEVICES ) break; - } - if ( nDevices > MAX_DEVICES ) break; - next_card: - snd_ctl_close(handle); - snd_card_next(&card); - } + case kAudioHardwareBadObjectError: + return "kAudioHardwareBadObjectError"; - if (nDevices == 0) return; + case kAudioHardwareBadDeviceError: + return "kAudioHardwareBadDeviceError"; - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } + case kAudioHardwareBadStreamError: + return "kAudioHardwareBadStreamError"; - // Write device ascii identifiers to device structures and then - // probe the device capabilities. - for (int i=0; i +#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; } +}; + +RtApiJack :: RtApiJack() { - // No ALSA API functions for default devices. - return 0; + // Nothing to do here. } -int RtAudio :: getDefaultOutputDevice(void) +RtApiJack :: ~RtApiJack() { - // No ALSA API functions for default devices. - return 0; + if ( stream_.state != STREAM_CLOSED ) closeStream(); } -void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info) +unsigned int RtApiJack :: getDeviceCount( void ) { - int err; - int open_mode = SND_PCM_ASYNC; - snd_pcm_t *handle; - snd_ctl_t *chandle; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca(&pcminfo); - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca(¶ms); - char name[32]; - char *card; - - // Open the control interface for this card. - strncpy( name, info->name, 32 ); - card = strtok(name, ","); - err = snd_ctl_open(&chandle, card, 0); - if (err < 0) { - sprintf(message, "RtAudio: ALSA control open (%s): %s.", card, snd_strerror(err)); - error(RtError::DEBUG_WARNING); - return; + // See if we can become a jack client. + jack_client_t *client = jack_client_new( "RtApiJackCount" ); + 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 (:). + unsigned int 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 ); } - unsigned int dev = (unsigned int) atoi( strtok(NULL, ",") ); - - // First try for playback - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_device(pcminfo, dev); - snd_pcm_info_set_subdevice(pcminfo, 0); - snd_pcm_info_set_stream(pcminfo, stream); - if ((err = snd_ctl_pcm_info(chandle, pcminfo)) < 0) { - if (err == -ENOENT) { - sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle output!", info->name); - error(RtError::DEBUG_WARNING); - } - else { - sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) output: %s", - info->name, snd_strerror(err)); - error(RtError::DEBUG_WARNING); - } - goto capture_probe; - } + jack_client_close( client ); + return nDevices; +} - err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK ); - if (err < 0) { - if ( err == EBUSY ) - sprintf(message, "RtAudio: ALSA pcm playback device (%s) is busy: %s.", - info->name, snd_strerror(err)); - else - sprintf(message, "RtAudio: ALSA pcm playback open (%s) error: %s.", - info->name, snd_strerror(err)); - error(RtError::DEBUG_WARNING); - goto capture_probe; +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + jack_client_t *client = jack_client_new( "RtApiJackInfo" ); + 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 (:). + unsigned int 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 ); } - // We have an open device ... allocate the parameter structure. - err = snd_pcm_hw_params_any(handle, params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - goto capture_probe; + if ( device >= nDevices ) { + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); } - // Get output channel information. - info->minOutputChannels = snd_pcm_hw_params_get_channels_min(params); - info->maxOutputChannels = snd_pcm_hw_params_get_channels_max(params); - - snd_pcm_close(handle); - - capture_probe: - // Now try for capture - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream(pcminfo, stream); + // Get the current jack server sample rate. + info.sampleRates.clear(); + info.sampleRates.push_back( jack_get_sample_rate( client ) ); - err = snd_ctl_pcm_info(chandle, pcminfo); - snd_ctl_close(chandle); - if ( err < 0 ) { - if (err == -ENOENT) { - sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle input!", info->name); - error(RtError::DEBUG_WARNING); - } - else { - sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) input: %s", - info->name, snd_strerror(err)); - error(RtError::DEBUG_WARNING); - } - if (info->maxOutputChannels == 0) - // didn't open for playback either ... device invalid - return; - goto probe_parameters; + // 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; } - err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK); - if (err < 0) { - if ( err == EBUSY ) - sprintf(message, "RtAudio: ALSA pcm capture device (%s) is busy: %s.", - info->name, snd_strerror(err)); - else - sprintf(message, "RtAudio: ALSA pcm capture open (%s) error: %s.", - info->name, snd_strerror(err)); - error(RtError::DEBUG_WARNING); - if (info->maxOutputChannels == 0) - // didn't open for playback either ... device invalid - return; - goto probe_parameters; - } - - // We have an open capture device ... allocate the parameter structure. - err = snd_pcm_hw_params_any(handle, params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - if (info->maxOutputChannels > 0) - goto probe_parameters; - else - return; + // 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; } - // Get input channel information. - info->minInputChannels = snd_pcm_hw_params_get_channels_min(params); - info->maxInputChannels = snd_pcm_hw_params_get_channels_max(params); - - snd_pcm_close(handle); + 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->maxOutputChannels == 0 || info->maxInputChannels == 0) - goto probe_parameters; - - info->hasDuplexSupport = true; - info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? - info->maxInputChannels : info->maxOutputChannels; - info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? - info->minInputChannels : info->minOutputChannels; - - probe_parameters: - // 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->maxOutputChannels >= info->maxInputChannels) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - err = snd_pcm_open(&handle, info->name, stream, open_mode); - if (err < 0) { - sprintf(message, "RtAudio: ALSA pcm (%s) won't reopen during probe: %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - return; - } + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; - // We have an open device ... allocate the parameter structure. - err = snd_pcm_hw_params_any(handle, params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA hardware reopen probe error (%s): %s.", - info->name, snd_strerror(err)); - error(RtError::WARNING); - return; - } + // 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; - // Test a non-standard sample rate to see if continuous rate is supported. - int dir = 0; - if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) { - // It appears that continuous sample rate support is available. - info->nSampleRates = -1; - info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir); - info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir); - } - else { - // No continuous rate support ... test our discrete set of sample rate values. - info->nSampleRates = 0; - for (int i=0; isampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } - } - if (info->nSampleRates == 0) { - snd_pcm_close(handle); - return; - } - } + jack_client_close(client); + info.probed = true; + return info; +} - // Probe the supported data formats ... we don't care about endian-ness just yet - snd_pcm_format_t format; - info->nativeFormats = 0; - format = SND_PCM_FORMAT_S8; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT8; - format = SND_PCM_FORMAT_S16; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT16; - format = SND_PCM_FORMAT_S24; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT24; - format = SND_PCM_FORMAT_S32; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_SINT32; - format = SND_PCM_FORMAT_FLOAT; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_FLOAT32; - format = SND_PCM_FORMAT_FLOAT64; - if (snd_pcm_hw_params_test_format(handle, params, format) == 0) - info->nativeFormats |= RTAUDIO_FLOAT64; +int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; - // Check that we have at least one supported format - if (info->nativeFormats == 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA PCM device (%s) data format not supported by RtAudio.", - info->name); - error(RtError::WARNING); - return; - } + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; - // That's all ... close the device and return - snd_pcm_close(handle); - info->probed = true; - return; + return 0; } -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) +void jackShutdown( void *infoPointer ) { -#if defined(__RTAUDIO_DEBUG__) - snd_output_t *out; - snd_output_stdio_attach(&out, stderr, 0); -#endif - - // I'm not using the "plug" interface ... too much inconsistent behavior. - const char *name = devices[device].name; + 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; + + object->closeStream(); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; +} - snd_pcm_stream_t alsa_stream; - if (mode == OUTPUT) - alsa_stream = SND_PCM_STREAM_PLAYBACK; - else - alsa_stream = SND_PCM_STREAM_CAPTURE; - - int err; - snd_pcm_t *handle; - int alsa_open_mode = SND_PCM_ASYNC; - err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode); - if (err < 0) { - sprintf(message,"RtAudio: ALSA pcm device (%s) won't open: %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } +int jackXrun( void *infoPointer ) +{ + JackHandle *handle = (JackHandle *) infoPointer; - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca(&hw_params); - err = snd_pcm_hw_params_any(handle, hw_params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error getting parameter handle (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtAudio: ALSA dump hardware params just after device open:\n\n"); - snd_pcm_hw_params_dump(hw_params, out); -#endif + 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; - // Set access ... try interleaved access first, then non-interleaved - if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) ) { - err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); - } - else if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED) ) { - err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED); - stream->deInterleave[mode] = true; + // 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 ) ) { + if ( options && !options->streamName.empty() ) + client = jack_client_new( options->streamName.c_str() ); + else + client = jack_client_new( "RtApiJack" ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtError::WARNING ); + return FAILURE; + } } else { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA device (%s) access not supported by RtAudio.", name); - error(RtError::WARNING); - return FAILURE; + // 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 (:). + unsigned int 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 (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting access ( (%s): %s.", name, snd_strerror(err)); - error(RtError::WARNING); + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; return FAILURE; } - // Determine how to set the device format. - stream->userFormat = format; - snd_pcm_format_t device_format; - - if (format == RTAUDIO_SINT8) - device_format = SND_PCM_FORMAT_S8; - else if (format == RTAUDIO_SINT16) - device_format = SND_PCM_FORMAT_S16; - else if (format == RTAUDIO_SINT24) - device_format = SND_PCM_FORMAT_S24; - else if (format == RTAUDIO_SINT32) - device_format = SND_PCM_FORMAT_S32; - else if (format == RTAUDIO_FLOAT32) - device_format = SND_PCM_FORMAT_FLOAT; - else if (format == RTAUDIO_FLOAT64) - device_format = SND_PCM_FORMAT_FLOAT64; - - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = format; - goto set_format; + // 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 = JackPortIsOutput; + if ( mode == INPUT ) flag = JackPortIsInput; + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); } - // The user requested format is not natively supported by the device. - device_format = SND_PCM_FORMAT_FLOAT64; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_FLOAT64; - goto set_format; + // 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; } - device_format = SND_PCM_FORMAT_FLOAT; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_FLOAT32; - goto set_format; + // 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; - device_format = SND_PCM_FORMAT_S32; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT32; - goto set_format; - } + // 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 ); - device_format = SND_PCM_FORMAT_S24; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT24; - goto set_format; - } + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; - device_format = SND_PCM_FORMAT_S16; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT16; - goto set_format; - } + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; - device_format = SND_PCM_FORMAT_S8; - if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) { - stream->deviceFormat[mode] = RTAUDIO_SINT8; - goto set_format; - } + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; - // If we get here, no supported format was found. - sprintf(message,"RtAudio: ALSA pcm device (%s) data format not supported by RtAudio.", name); - snd_pcm_close(handle); - error(RtError::WARNING); - return FAILURE; + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; - set_format: - err = snd_pcm_hw_params_set_format(handle, hw_params, device_format); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting format (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } + // 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; - // Determine whether byte-swaping is necessary. - stream->doByteSwap[mode] = false; - if (device_format != SND_PCM_FORMAT_S8) { - err = snd_pcm_format_cpu_endian(device_format); - if (err == 0) - stream->doByteSwap[mode] = true; - else if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error getting format endian-ness (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; + 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; } - } - // Set the sample rate. - err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting sample rate (%d) on device (%s): %s.", - sampleRate, name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; + 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; - // 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; - int device_channels = snd_pcm_hw_params_get_channels_max(hw_params); - if (device_channels < channels) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: channels (%d) not supported by device (%s).", - channels, name); - error(RtError::WARNING); - return FAILURE; + // 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; } - device_channels = snd_pcm_hw_params_get_channels_min(hw_params); - if (device_channels < channels) device_channels = channels; - stream->nDeviceChannels[mode] = device_channels; - - // Set the device channels. - err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting channels (%d) on device (%s): %s.", - device_channels, name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } + if ( stream_.doConvertBuffer[mode] ) { - // Set the buffer number, which in ALSA is referred to as the "period". - int dir; - int periods = numberOfBuffers; - // Even though the hardware might allow 1 buffer, it won't work reliably. - if (periods < 2) periods = 2; - err = snd_pcm_hw_params_get_periods_min(hw_params, &dir); - if (err > periods) periods = err; - err = snd_pcm_hw_params_get_periods_max(hw_params, &dir); - if (err < periods) periods = err; - - err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting periods (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; - } + 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; + } + } - // Set the buffer (or period) size. - err = snd_pcm_hw_params_get_period_size_min(hw_params, &dir); - if (err > *bufferSize) *bufferSize = err; - - err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error setting period size (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; + 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; + } + } } - // 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 ) { - sprintf( message, "RtAudio: ALSA error setting buffer size for duplex stream on device (%s).", - name ); - error(RtError::DEBUG_WARNING); - return FAILURE; + // 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->bufferSize = *bufferSize; + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; - // Install the hardware configuration - err = snd_pcm_hw_params(handle, hw_params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error installing hardware configuration (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; + 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 ); } -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtAudio: ALSA dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump(hw_params, out); -#endif - - /* - // Install the software configuration - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca(&sw_params); - snd_pcm_sw_params_current(handle, sw_params); - err = snd_pcm_sw_params(handle, sw_params); - if (err < 0) { - snd_pcm_close(handle); - sprintf(message, "RtAudio: ALSA error installing software configuration (%s): %s.", - name, snd_strerror(err)); - error(RtError::WARNING); - return FAILURE; + // 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 ); + } } - */ - - // Set handle and flags for buffer conversion - stream->handle[mode] = handle; - 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->nUserChannels[mode] > 1 && stream->deInterleave[mode]) - stream->doConvertBuffer[mode] = true; - // Allocate necessary internal buffers - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { + // 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 ); - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; + return SUCCESS; - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; - } + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); - if ( stream->doConvertBuffer[mode] ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); - long buffer_bytes; - bool makeBuffer = true; - if ( mode == OUTPUT ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == INPUT - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == OUTPUT && stream->deviceBuffer ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes < bytes_out ) makeBuffer = false; - } - } + delete handle; + stream_.apiHandle = 0; + } - if ( makeBuffer ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } } - 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 = periods; - stream->sampleRate = sampleRate; - - return SUCCESS; - - memory_error: - if (stream->handle[0]) { - snd_pcm_close(stream->handle[0]); - stream->handle[0] = 0; - } - if (stream->handle[1]) { - snd_pcm_close(stream->handle[1]); - stream->handle[1] = 0; - } - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - sprintf(message, "RtAudio: ALSA error allocating buffer memory (%s).", name); - error(RtError::WARNING); + return FAILURE; } -void RtAudio :: closeStream(int streamId) +void RtApiJack :: closeStream( void ) { - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtError::WARNING ); return; } - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { - if (stream->callbackInfo.usingCallback) { - pthread_cancel(stream->callbackInfo.thread); - pthread_join(stream->callbackInfo.thread, NULL); - } + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); - if (stream->state == STREAM_RUNNING) { - if (stream->mode == OUTPUT || stream->mode == DUPLEX) - snd_pcm_drop(stream->handle[0]); - if (stream->mode == INPUT || stream->mode == DUPLEX) - snd_pcm_drop(stream->handle[1]); + jack_client_close( handle->client ); } - pthread_mutex_destroy(&stream->mutex); - - if (stream->handle[0]) - snd_pcm_close(stream->handle[0]); - - if (stream->handle[1]) - snd_pcm_close(stream->handle[1]); + 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; + } - if (stream->userBuffer) - free(stream->userBuffer); + 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); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - free(stream); - streams.erase(streamId); + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } -void RtAudio :: startStream(int streamId) +void RtApiJack :: startStream( void ) { - // This method calls snd_pcm_prepare if the device isn't already in that state. - - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK(&stream_.mutex); - if (stream->state == STREAM_RUNNING) + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; goto unlock; + } - int err; - snd_pcm_state_t state; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - state = snd_pcm_state(stream->handle[0]); - if (state != SND_PCM_STATE_PREPARED) { - err = snd_pcm_prepare(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } + 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; } - } - if (stream->mode == INPUT || stream->mode == DUPLEX) { - state = snd_pcm_state(stream->handle[1]); - if (state != SND_PCM_STATE_PREPARED) { - err = snd_pcm_prepare(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + // 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); } - stream->state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: stopStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - err = snd_pcm_drain(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + 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; } - } - if (stream->mode == INPUT || stream->mode == DUPLEX) { - err = snd_pcm_drain(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + // 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); } - stream->state = STREAM_STOPPED; + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; unlock: - MUTEX_UNLOCK(&stream->mutex); + MUTEX_UNLOCK(&stream_.mutex); + + if ( result == 0 ) return; + error( RtError::SYSTEM_ERROR ); } -void RtAudio :: abortStream(int streamId) +void RtApiJack :: stopStream( void ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK( &stream_.mutex ); - if (stream->state == STREAM_STOPPED) - goto unlock; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - int err; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - err = snd_pcm_drop(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled } } - if (stream->mode == INPUT || stream->mode == DUPLEX) { - err = snd_pcm_drop(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - stream->state = STREAM_STOPPED; + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; - unlock: - MUTEX_UNLOCK(&stream->mutex); + MUTEX_UNLOCK( &stream_.mutex ); } -int RtAudio :: streamWillBlock(int streamId) +void RtApiJack :: abortStream( void ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - int err = 0, frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; - - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - err = snd_pcm_avail_update(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - - frames = err; - - if (stream->mode == INPUT || stream->mode == DUPLEX) { - err = snd_pcm_avail_update(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - if (frames > err) frames = err; + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - frames = stream->bufferSize - frames; - if (frames < 0) frames = 0; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 1; - unlock: - MUTEX_UNLOCK(&stream->mutex); - return frames; + stopStream(); } -void RtAudio :: tickStream(int streamId) +bool RtApiJack :: callbackEvent( unsigned long nframes ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds - return; + 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; } - else if (stream->callbackInfo.usingCallback) { - RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; - stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtError::WARNING ); + return FAILURE; } - MUTEX_LOCK(&stream->mutex); + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; - // The state might change while waiting on a mutex. - if (stream->state == STREAM_STOPPED) - goto unlock; + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + pthread_cond_signal( &handle->condition ); + else + stopStream(); + return SUCCESS; + } - int err; - char *buffer; - int channels; - RTAUDIO_FORMAT format; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + MUTEX_LOCK( &stream_.mutex ); - // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, OUTPUT); - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[0]; - format = stream->deviceFormat[0]; + // 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; } - else { - buffer = stream->userBuffer; - channels = stream->nUserChannels[0]; - format = stream->userFormat; + 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; + } - // Do byte swapping if necessary. - if (stream->doByteSwap[0]) - byteSwapBuffer(buffer, stream->bufferSize * channels, format); + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - // Write samples to device in interleaved/non-interleaved format. - if (stream->deInterleave[0]) { - void *bufs[channels]; - size_t offset = stream->bufferSize * formatBytes(format); - for (int i=0; ihandle[0], bufs, stream->bufferSize); - } - else - err = snd_pcm_writei(stream->handle[0], buffer, stream->bufferSize); + if ( handle->drainCounter > 0 ) { // write zeros to the output stream - if (err < stream->bufferSize) { - // Either an error or underrun occured. - if (err == -EPIPE) { - snd_pcm_state_t state = snd_pcm_state(stream->handle[0]); - if (state == SND_PCM_STATE_XRUN) { - sprintf(message, "RtAudio: ALSA underrun detected."); - error(RtError::WARNING); - err = snd_pcm_prepare(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing handle after underrun: %s.", - snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - else { - sprintf(message, "RtAudio: ALSA error, current state is %s.", - snd_pcm_state_name(state)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - goto unlock; - } - else { - sprintf(message, "RtAudio: ALSA audio write error for device (%s): %s.", - devices[stream->device[0]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); } + } - } + else if ( stream_.doConvertBuffer[0] ) { - if (stream->mode == INPUT || stream->mode == DUPLEX) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - // Setup parameters. - if (stream->doConvertBuffer[1]) { - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[1]; - format = stream->deviceFormat[1]; + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } } - else { - buffer = stream->userBuffer; - channels = stream->nUserChannels[1]; - format = stream->userFormat; + else { // no buffer conversion + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } } - // Read samples from device in interleaved/non-interleaved format. - if (stream->deInterleave[1]) { - void *bufs[channels]; - size_t offset = stream->bufferSize * formatBytes(format); - for (int i=0; ihandle[1], bufs, stream->bufferSize); + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; } - else - err = snd_pcm_readi(stream->handle[1], buffer, stream->bufferSize); + } - if (err < stream->bufferSize) { - // Either an error or underrun occured. - if (err == -EPIPE) { - snd_pcm_state_t state = snd_pcm_state(stream->handle[1]); - if (state == SND_PCM_STATE_XRUN) { - sprintf(message, "RtAudio: ALSA overrun detected."); - error(RtError::WARNING); - err = snd_pcm_prepare(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: ALSA error preparing handle after overrun: %s.", - snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - } - else { - sprintf(message, "RtAudio: ALSA error, current state is %s.", - snd_pcm_state_name(state)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - 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 ); } - else { - sprintf(message, "RtAudio: ALSA audio read error for device (%s): %s.", - devices[stream->device[1]].name, snd_strerror(err)); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); + 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 ); } } - - // Do byte swapping if necessary. - if (stream->doByteSwap[1]) - byteSwapBuffer(buffer, stream->bufferSize * channels, format); - - // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, INPUT); } unlock: - MUTEX_UNLOCK(&stream->mutex); + MUTEX_UNLOCK(&stream_.mutex); - if (stream->callbackInfo.usingCallback && stopStream) - this->stopStream(streamId); + RtApi::tickStreamTime(); + return SUCCESS; } +//******************** End of __UNIX_JACK__ *********************// +#endif -extern "C" void *callbackHandler(void *ptr) -{ - CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; - RtAudio *object = (RtAudio *) info->object; - int stream = info->streamId; - bool *usingCallback = &info->usingCallback; +#if defined(__WINDOWS_ASIO__) // ASIO API on Windows - while ( *usingCallback ) { - pthread_testcancel(); - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; - } - } +// 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. - return 0; -} +#include "asiosys.h" +#include "asio.h" +#include "iasiothiscallresolver.h" +#include "asiodrivers.h" +#include -//******************** End of __LINUX_ALSA__ *********************// +AsioDrivers drivers; +ASIOCallbacks asioCallbacks; +ASIODriverInfo driverInfo; +CallbackInfo *asioCallbackInfo; +bool asioXRun; -#elif defined(__LINUX_OSS__) +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; -#include -#include -#include -#include -#include -#include -#include -#include + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; -#define DAC_NAME "/dev/dsp" -#define MAX_DEVICES 16 -#define MAX_CHANNELS 16 +// 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 ); -void RtAudio :: initialize(void) +RtApiAsio :: RtApiAsio() { - // Count cards and devices - nDevices = 0; - - // We check /dev/dsp before probing devices. /dev/dsp is supposed to - // be a link to the "default" audio device, of the form /dev/dsp0, - // /dev/dsp1, etc... However, I've seen many cases where /dev/dsp was a - // real device, so we need to check for that. Also, sometimes the - // link is to /dev/dspx and other times just dspx. I'm not sure how - // the latter works, but it does. - char device_name[16]; - struct stat dspstat; - int dsplink = -1; - int i = 0; - if (lstat(DAC_NAME, &dspstat) == 0) { - if (S_ISLNK(dspstat.st_mode)) { - i = readlink(DAC_NAME, device_name, sizeof(device_name)); - if (i > 0) { - device_name[i] = '\0'; - if (i > 8) { // check for "/dev/dspx" - if (!strncmp(DAC_NAME, device_name, 8)) - dsplink = atoi(&device_name[8]); - } - else if (i > 3) { // check for "dspx" - if (!strncmp("dsp", device_name, 3)) - dsplink = atoi(&device_name[3]); - } - } - else { - sprintf(message, "RtAudio: cannot read value of symbolic link %s.", DAC_NAME); - error(RtError::SYSTEM_ERROR); - } - } - } - else { - sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME); - error(RtError::SYSTEM_ERROR); - } - - // The OSS API doesn't provide a routine for determining the number - // of devices. Thus, we'll just pursue a brute force method. The - // idea is to start with /dev/dsp(0) and continue with higher device - // numbers until we reach MAX_DSP_DEVICES. This should tell us how - // many devices we have ... it is not a fullproof scheme, but hopefully - // it will work most of the time. - - int fd = 0; - char names[MAX_DEVICES][16]; - for (i=-1; i= 0) close(fd); - strncpy(names[nDevices], device_name, 16); - nDevices++; - } - - if (nDevices == 0) return; - - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); + // 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; - // Write device ascii identifiers to device control structure and then probe capabilities. - for (i=0; iname, O_WRONLY | O_NONBLOCK); - if (fd == -1) { - // Open device failed ... either busy or doesn't exist - if (errno == EBUSY || errno == EAGAIN) - sprintf(message, "RtAudio: OSS playback device (%s) is busy and cannot be probed.", - info->name); - else - sprintf(message, "RtAudio: OSS playback device (%s) open error.", info->name); - error(RtError::DEBUG_WARNING); - goto capture_probe; - } - - // We have an open device ... see how many channels it can handle - for (i=MAX_CHANNELS; i>0; i--) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { - // This would normally indicate some sort of hardware error, but under ALSA's - // OSS emulation, it sometimes indicates an invalid channel value. Further, - // the returned channel value is not changed. So, we'll ignore the possible - // hardware error. - continue; // try next channel number - } - // Check to see whether the device supports the requested number of channels - if (channels != i ) continue; // try next channel number - // If here, we found the largest working channel value - break; - } - info->maxOutputChannels = i; + RtAudio::DeviceInfo info; + info.probed = false; - // Now find the minimum number of channels it can handle - for (i=1; i<=info->maxOutputChannels; i++) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // try next channel number - // If here, we found the smallest working channel value - break; + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); } - info->minOutputChannels = i; - close(fd); - capture_probe: - // Now try for capture - fd = open(info->name, O_RDONLY | O_NONBLOCK); - if (fd == -1) { - // Open device for capture failed ... either busy or doesn't exist - if (errno == EBUSY || errno == EAGAIN) - sprintf(message, "RtAudio: OSS capture device (%s) is busy and cannot be probed.", - info->name); - else - sprintf(message, "RtAudio: OSS capture device (%s) open error.", info->name); - error(RtError::DEBUG_WARNING); - if (info->maxOutputChannels == 0) - // didn't open for playback either ... device invalid - return; - goto probe_parameters; + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); } - // We have the device open for capture ... see how many channels it can handle - for (i=MAX_CHANNELS; i>0; i--) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { - continue; // as above + // 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; } - // If here, we found a working channel value - break; - } - info->maxInputChannels = i; - - // Now find the minimum number of channels it can handle - for (i=1; i<=info->maxInputChannels; i++) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // try next channel number - // If here, we found the smallest working channel value - break; + return devices_[ device ]; } - info->minInputChannels = i; - close(fd); - if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) { - sprintf(message, "RtAudio: OSS device (%s) reports zero channels for input and output.", - info->name); - error(RtError::DEBUG_WARNING); - return; + 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; } - // If device opens for both playback and capture, we determine the channels. - if (info->maxOutputChannels == 0 || info->maxInputChannels == 0) - goto probe_parameters; - - fd = open(info->name, O_RDWR | O_NONBLOCK); - if (fd == -1) - goto probe_parameters; - - ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0); - ioctl(fd, SNDCTL_DSP_GETCAPS, &mask); - if (mask & DSP_CAP_DUPLEX) { - info->hasDuplexSupport = true; - // We have the device open for duplex ... see how many channels it can handle - for (i=MAX_CHANNELS; i>0; i--) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // as above - // If here, we found a working channel value - break; - } - info->maxDuplexChannels = i; + info.name = driverName; - // Now find the minimum number of channels it can handle - for (i=1; i<=info->maxDuplexChannels; i++) { - channels = i; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) - continue; // try next channel number - // If here, we found the smallest working channel value - break; - } - info->minDuplexChannels = i; + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - close(fd); - - probe_parameters: - // At this point, we need to figure out the supported data formats - // and sample rates. We'll proceed by openning 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->maxOutputChannels >= info->maxInputChannels) { - fd = open(info->name, O_WRONLY | O_NONBLOCK); - channels = info->maxOutputChannels; - } - else { - fd = open(info->name, O_RDONLY | O_NONBLOCK); - channels = info->maxInputChannels; + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - if (fd == -1) { - // We've got some sort of conflict ... abort - sprintf(message, "RtAudio: OSS device (%s) won't reopen during probe.", - info->name); - error(RtError::DEBUG_WARNING); - return; + // 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; } - // We have an open device ... set to maximum channels. - i = channels; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) { - // We've got some sort of conflict ... abort - close(fd); - sprintf(message, "RtAudio: OSS device (%s) won't revert to previous channel setting.", - info->name); - error(RtError::DEBUG_WARNING); - return; - } + info.outputChannels = outputChannels; + info.inputChannels = inputChannels; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", - info->name); - error(RtError::DEBUG_WARNING); - return; + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; inativeFormats = 0; -#if defined (AFMT_S32_BE) - // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h - if (mask & AFMT_S32_BE) { - format = AFMT_S32_BE; - info->nativeFormats |= RTAUDIO_SINT32; - } -#endif -#if defined (AFMT_S32_LE) - /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */ - if (mask & AFMT_S32_LE) { - format = AFMT_S32_LE; - info->nativeFormats |= RTAUDIO_SINT32; - } -#endif - if (mask & AFMT_S8) { - format = AFMT_S8; - info->nativeFormats |= RTAUDIO_SINT8; - } - if (mask & AFMT_S16_BE) { - format = AFMT_S16_BE; - info->nativeFormats |= RTAUDIO_SINT16; - } - if (mask & AFMT_S16_LE) { - format = AFMT_S16_LE; - info->nativeFormats |= RTAUDIO_SINT16; + // Determine supported data types ... just check first channel and assume rest are the same. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + channelInfo.isInput = true; + if ( info.inputChannels <= 0 ) channelInfo.isInput = false; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - // Check that we have at least one supported format - if (info->nativeFormats == 0) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", - info->name); - error(RtError::DEBUG_WARNING); - return; + info.nativeFormats = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) + info.nativeFormats |= RTAUDIO_SINT16; + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) + info.nativeFormats |= RTAUDIO_SINT32; + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) + info.nativeFormats |= RTAUDIO_FLOAT32; + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) + info.nativeFormats |= RTAUDIO_FLOAT64; + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + 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; iname); - error(RtError::DEBUG_WARNING); - return; + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + return FAILURE; } - // Probe the supported sample rates ... first get lower limit - int speed = 1; - if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { - // If we get here, we're probably using an ALSA driver with OSS-emulation, - // which doesn't conform to the OSS specification. In this case, - // we'll probe our predefined list of sample rates for working values. - info->nSampleRates = 0; - for (i=0; isampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } + // The getDeviceInfo() function will not work when a stream is open + // because ASIO does not allow multiple devices to run at the same + // time. Thus, we'll probe the system before opening a stream and + // save the results for use by getDeviceInfo(). + this->saveDeviceInfo(); + + // Only load the driver once for duplex stream. + if ( mode != INPUT || stream_.mode != OUTPUT ) { + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; } - if (info->nSampleRates == 0) { - close(fd); - return; + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; } - goto finished; } - info->sampleRates[0] = speed; - // Now get upper limit - speed = 1000000; - if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) error setting sample rate.", - info->name); - error(RtError::DEBUG_WARNING); - return; + // 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; } - info->sampleRates[1] = speed; - info->nSampleRates = -1; - finished: // That's all ... close the device and return - close(fd); - info->probed = true; - return; -} + 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; -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ - int buffers, buffer_bytes, device_channels, device_format; - int srate, temp, fd; + // 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; + } - const char *name = devices[device].name; + // 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; + } - if (mode == OUTPUT) - fd = open(name, O_WRONLY | O_NONBLOCK); - 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(stream->handle[0]); - stream->handle[0] = 0; - // First check that the number previously set channels is the same. - if (stream->nUserChannels[0] != channels) { - sprintf(message, "RtAudio: input/output channels must be equal for OSS duplex device (%s).", name); - goto error; - } - fd = open(name, O_RDWR | O_NONBLOCK); + // 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; } - else - fd = open(name, O_RDONLY | O_NONBLOCK); } - if (fd == -1) { - if (errno == EBUSY || errno == EAGAIN) - sprintf(message, "RtAudio: OSS device (%s) is busy and cannot be opened.", - name); - else - sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); - goto error; + // 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; } - // Now reopen in blocking mode. - close(fd); - if (mode == OUTPUT) - fd = open(name, O_WRONLY | O_SYNC); - else { // mode == INPUT - if (stream->mode == OUTPUT && stream->device[0] == device) - fd = open(name, O_RDWR | O_SYNC); - else - fd = open(name, O_RDONLY | O_SYNC); + // 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 (fd == -1) { - sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name); - goto error; + if ( stream_.deviceFormat[mode] == 0 ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; } - // Get the sample format mask - int mask; - if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.", - name); - goto error; + // 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; } - // Determine how to set the device format. - stream->userFormat = format; - device_format = -1; - stream->doByteSwap[mode] = false; - if (format == RTAUDIO_SINT8) { - if (mask & AFMT_S8) { - device_format = AFMT_S8; - stream->deviceFormat[mode] = RTAUDIO_SINT8; - } - } - else if (format == RTAUDIO_SINT16) { - if (mask & AFMT_S16_NE) { - device_format = AFMT_S16_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S16_BE) { - device_format = AFMT_S16_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S16_LE) { - device_format = AFMT_S16_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#endif + 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. + double power = std::log10( (double) *bufferSize ) / log10( 2.0 ); + *bufferSize = (int) pow( 2.0, floor(power+0.5) ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else *bufferSize = preferSize; } -#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) - else if (format == RTAUDIO_SINT32) { - if (mask & AFMT_S32_NE) { - device_format = AFMT_S32_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S32_BE) { - device_format = AFMT_S32_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S32_LE) { - device_format = AFMT_S32_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#endif + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; } -#endif - if (device_format == -1) { - // The user requested format is not natively supported by the device. - if (mask & AFMT_S16_NE) { - device_format = AFMT_S16_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S16_BE) { - device_format = AFMT_S16_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S16_LE) { - device_format = AFMT_S16_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->doByteSwap[mode] = true; - } -#endif -#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE) - else if (mask & AFMT_S32_NE) { - device_format = AFMT_S32_NE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - } -#if BYTE_ORDER == LITTLE_ENDIAN - else if (mask & AFMT_S32_BE) { - device_format = AFMT_S32_BE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#else - else if (mask & AFMT_S32_LE) { - device_format = AFMT_S32_LE; - stream->deviceFormat[mode] = RTAUDIO_SINT32; - stream->doByteSwap[mode] = true; - } -#endif -#endif - else if (mask & AFMT_S8) { - device_format = AFMT_S8; - stream->deviceFormat[mode] = RTAUDIO_SINT8; - } + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + return FAILURE; } - if (stream->deviceFormat[mode] == 0) { - // This really shouldn't happen ... - close(fd); - sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.", - name); - goto error; - } + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; - // Determine the number of channels for this device. Note that the - // channel value requested by the user might be < min_X_Channels. - stream->nUserChannels[mode] = channels; - device_channels = channels; - if (mode == OUTPUT) { - if (channels < devices[device].minOutputChannels) - device_channels = devices[device].minOutputChannels; - } - else { // mode == INPUT - if (stream->mode == OUTPUT && stream->device[0] == device) { - // We're doing duplex setup here. - if (channels < devices[device].minDuplexChannels) - device_channels = devices[device].minDuplexChannels; + 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; } - else { - if (channels < devices[device].minInputChannels) - device_channels = devices[device].minInputChannels; + catch ( std::bad_alloc& ) { + //if ( handle == NULL ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; + return FAILURE; } - } - stream->nDeviceChannels[mode] = device_channels; + handle->bufferInfos = 0; - // 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. - buffer_bytes = *bufferSize * formatBytes(stream->deviceFormat[mode]) * device_channels; - if (buffer_bytes < 16) buffer_bytes = 16; - buffers = numberOfBuffers; - if (buffers < 2) buffers = 2; - temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0)); - if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) { - close(fd); - sprintf(message, "RtAudio: OSS error setting fragment size for device (%s).", - name); - 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; } - stream->nBuffers = buffers; - // Set the data format. - temp = device_format; - if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) { - close(fd); - sprintf(message, "RtAudio: OSS error setting data format for device (%s).", - name); - goto error; + // 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 ); } - // Set the number of channels. - temp = device_channels; - if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) { - close(fd); - sprintf(message, "RtAudio: OSS error setting %d channels on device (%s).", - temp, name); + // 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; } - // Set the sample rate. - srate = sampleRate; - temp = srate; - if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS error setting sample rate = %d on device (%s).", - temp, name); - 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; } - - // Verify the sample rate setup worked. - if (abs(srate - temp) > 100) { - close(fd); - sprintf(message, "RtAudio: OSS error ... audio device (%s) doesn't support sample rate of %d.", - name, temp); - goto error; + for ( i=0; iisInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; } - stream->sampleRate = sampleRate; - if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) { - close(fd); - sprintf(message, "RtAudio: OSS error getting buffer size for device (%s).", - name); + // 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; - // Save buffer size (in sample frames). - *bufferSize = buffer_bytes / (formatBytes(stream->deviceFormat[mode]) * device_channels); - stream->bufferSize = *bufferSize; - - 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] = device_channels; - } - - // 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; + // 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 - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) { - close(fd); - sprintf(message, "RtAudio: OSS error allocating user buffer memory (%s).", - name); - goto error; - } + 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] ) { + if ( stream_.doConvertBuffer[mode] ) { - long buffer_bytes; bool makeBuffer = true; - if ( mode == OUTPUT ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == INPUT - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == OUTPUT && stream->deviceBuffer ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes < bytes_out ) makeBuffer = false; + 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 ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) { - close(fd); - free(stream->userBuffer); - sprintf(message, "RtAudio: OSS error allocating device buffer memory (%s).", - name); + 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->device[mode] = device; - stream->handle[mode] = fd; - stream->state = STREAM_STOPPED; - if ( stream->mode == OUTPUT && mode == INPUT ) { - stream->mode = DUPLEX; - if (stream->device[0] == device) - stream->handle[0] = fd; - } + 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; + 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 (stream->handle[0]) { - close(stream->handle[0]); - stream->handle[0] = 0; + 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; + } } - error(RtError::WARNING); + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + return FAILURE; } -void RtAudio :: closeStream(int streamId) +void RtApiAsio :: closeStream() { - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtError::WARNING ); return; } - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; - - if (stream->callbackInfo.usingCallback) { - pthread_cancel(stream->callbackInfo.thread); - pthread_join(stream->callbackInfo.thread, NULL); + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); - if (stream->state == STREAM_RUNNING) { - if (stream->mode == OUTPUT || stream->mode == DUPLEX) - ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); - if (stream->mode == INPUT || stream->mode == DUPLEX) - ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; } - pthread_mutex_destroy(&stream->mutex); - - if (stream->handle[0]) - close(stream->handle[0]); - - if (stream->handle[1]) - close(stream->handle[1]); - - if (stream->userBuffer) - free(stream->userBuffer); + 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); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - free(stream); - streams.erase(streamId); + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } -void RtAudio :: startStream(int streamId) +void RtApiAsio :: startStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK( &stream_.mutex ); - stream->state = STREAM_RUNNING; + 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; + } - // No need to do anything else here ... OSS automatically starts - // when fed samples. + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + asioXRun = false; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - MUTEX_UNLOCK(&stream->mutex); + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); } -void RtAudio :: stopStream(int streamId) +void RtApiAsio :: stopStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - if (stream->state == STREAM_STOPPED) - goto unlock; + MUTEX_LOCK( &stream_.mutex ); - int err; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - err = ioctl(stream->handle[0], SNDCTL_DSP_SYNC, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error stopping device (%s).", - devices[stream->device[0]].name); - error(RtError::DRIVER_ERROR); + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); } } - else { - err = ioctl(stream->handle[1], SNDCTL_DSP_SYNC, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error stopping device (%s).", - devices[stream->device[1]].name); - error(RtError::DRIVER_ERROR); - } + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); } - stream->state = STREAM_STOPPED; - unlock: - MUTEX_UNLOCK(&stream->mutex); + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); } -void RtAudio :: abortStream(int streamId) +void RtApiAsio :: abortStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) - goto unlock; - - int err; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - err = ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error aborting device (%s).", - devices[stream->device[0]].name); - error(RtError::DRIVER_ERROR); - } - } - else { - err = ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0); - if (err < -1) { - sprintf(message, "RtAudio: OSS error aborting device (%s).", - devices[stream->device[1]].name); - error(RtError::DRIVER_ERROR); - } + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - stream->state = STREAM_STOPPED; - unlock: - MUTEX_UNLOCK(&stream->mutex); + // 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 completed + // disposed. So now, calling abort is the same as calling stop. + //AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + //handle->drainCounter = 1; + stopStream(); } -int RtAudio :: streamWillBlock(int streamId) +bool RtApiAsio :: callbackEvent( long bufferIndex ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - int bytes = 0, channels = 0, frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; - - audio_buf_info info; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - ioctl(stream->handle[0], SNDCTL_DSP_GETOSPACE, &info); - bytes = info.bytes; - channels = stream->nDeviceChannels[0]; - } - - if (stream->mode == INPUT || stream->mode == DUPLEX) { - ioctl(stream->handle[1], SNDCTL_DSP_GETISPACE, &info); - if (stream->mode == DUPLEX ) { - bytes = (bytes < info.bytes) ? bytes : info.bytes; - channels = stream->nDeviceChannels[0]; - } - else { - bytes = info.bytes; - channels = stream->nDeviceChannels[1]; - } + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; } - frames = (int) (bytes / (channels * formatBytes(stream->deviceFormat[0]))); - frames -= stream->bufferSize; - if (frames < 0) frames = 0; - - unlock: - MUTEX_UNLOCK(&stream->mutex); - return frames; -} - -void RtAudio :: tickStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds - return; - } - else if (stream->callbackInfo.usingCallback) { - RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; - stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return SUCCESS; } - MUTEX_LOCK(&stream->mutex); + 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; - RTAUDIO_FORMAT format; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + if ( stream_.state == STREAM_STOPPED ) goto unlock; - // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, OUTPUT); - buffer = stream->deviceBuffer; - samples = stream->bufferSize * stream->nDeviceChannels[0]; - format = stream->deviceFormat[0]; + // 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; } - else { - buffer = stream->userBuffer; - samples = stream->bufferSize * stream->nUserChannels[0]; - format = stream->userFormat; + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = false; } - - // Do byte swapping if necessary. - if (stream->doByteSwap[0]) - byteSwapBuffer(buffer, samples, format); - - // Write samples to device. - result = write(stream->handle[0], buffer, samples * formatBytes(format)); - - if (result == -1) { - // This could be an underrun, but the basic OSS API doesn't provide a means for determining that. - sprintf(message, "RtAudio: OSS audio write error for device (%s).", - devices[stream->device[0]].name); - error(RtError::DRIVER_ERROR); + 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 == 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; - samples = stream->bufferSize * stream->nUserChannels[1]; - format = stream->userFormat; - } - - // Read samples from device. - result = read(stream->handle[1], buffer, samples * formatBytes(format)); - - if (result == -1) { - // This could be an overrun, but the basic OSS API doesn't provide a means for determining that. - sprintf(message, "RtAudio: OSS audio read error for device (%s).", - devices[stream->device[1]].name); - error(RtError::DRIVER_ERROR); - } - - // Do byte swapping if necessary. - if (stream->doByteSwap[1]) - byteSwapBuffer(buffer, samples, format); - - // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, INPUT); - } + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - unlock: - MUTEX_UNLOCK(&stream->mutex); + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); - if (stream->callbackInfo.usingCallback && stopStream) - this->stopStream(streamId); -} + if ( handle->drainCounter > 1 ) { // write zeros to the output stream -extern "C" void *callbackHandler(void *ptr) -{ - CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; - RtAudio *object = (RtAudio *) info->object; - int stream = info->streamId; - bool *usingCallback = &info->usingCallback; + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); + } - while ( *usingCallback ) { - pthread_testcancel(); - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; } - } - - return 0; -} - - -//******************** End of __LINUX_OSS__ *********************// - -#elif 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. 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. + else if ( stream_.doConvertBuffer[0] ) { -#include "asio/asiosys.h" -#include "asio/asio.h" -#include "asio/asiodrivers.h" -#include + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); -AsioDrivers drivers; -ASIOCallbacks asioCallbacks; -CALLBACK_INFO *asioCallbackInfo; -ASIODriverInfo driverInfo; + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } -void RtAudio :: initialize(void) -{ - nDevices = drivers.asioGetNumDev(); - if (nDevices <= 0) return; - - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } - - // Write device driver names to device structures and then probe the - // device capabilities. - for (int i=0; i 0 ) { - sprintf(message, "RtAudio: unable to probe ASIO driver while a stream is open."); - error(RtError::DEBUG_WARNING); - return; - } + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); - if ( !drivers.loadDriver( info->name ) ) { - sprintf(message, "RtAudio: ASIO error loading driver (%s).", info->name); - error(RtError::DEBUG_WARNING); - return; - } + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); + } - ASIOError result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - char details[32]; - if ( result == ASE_HWMalfunction ) - sprintf(details, "hardware malfunction"); - else if ( result == ASE_NoMemory ) - sprintf(details, "no memory"); - else if ( result == ASE_NotPresent ) - sprintf(details, "driver/hardware not present"); - else - sprintf(details, "unspecified"); - sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, info->name); - error(RtError::DEBUG_WARNING); - return; - } + } - // Determine the device channel information. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", info->name); - error(RtError::DEBUG_WARNING); - return; + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } } - info->maxOutputChannels = outputChannels; - if ( outputChannels > 0 ) info->minOutputChannels = 1; - - info->maxInputChannels = inputChannels; - if ( inputChannels > 0 ) info->minInputChannels = 1; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - // If device opens for both playback and capture, we determine the channels. - if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) { - info->hasDuplexSupport = true; - info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ? - info->maxInputChannels : info->maxOutputChannels; - info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ? - info->minInputChannels : info->minOutputChannels; - } + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); - // Determine the supported sample rates. - info->nSampleRates = 0; - for (int i=0; isampleRates[info->nSampleRates++] = SAMPLE_RATES[i]; - } + if (stream_.doConvertBuffer[1]) { - if (info->nSampleRates == 0) { - drivers.removeCurrentDriver(); - sprintf( message, "RtAudio: No supported sample rates found for ASIO driver (%s).", info->name ); - error(RtError::DEBUG_WARNING); - return; - } + // 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 ); + } - // Determine supported data types ... just check first channel and assume rest are the same. - ASIOChannelInfo channelInfo; - channelInfo.channel = 0; - channelInfo.isInput = true; - if ( info->maxInputChannels <= 0 ) channelInfo.isInput = false; - result = ASIOGetChannelInfo( &channelInfo ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO error getting driver (%s) channel information.", info->name); - error(RtError::DEBUG_WARNING); - return; - } + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) - info->nativeFormats |= RTAUDIO_SINT16; - else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) - info->nativeFormats |= RTAUDIO_SINT32; - else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) - info->nativeFormats |= RTAUDIO_FLOAT32; - else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) - info->nativeFormats |= RTAUDIO_FLOAT64; + } + else { + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + } - // Check that we have at least one supported format. - if (info->nativeFormats == 0) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.", - info->name); - error(RtError::DEBUG_WARNING); - return; + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); + } } - info->probed = true; - drivers.removeCurrentDriver(); -} + 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(); -void bufferSwitch(long index, ASIOBool processNow) -{ - RtAudio *object = (RtAudio *) asioCallbackInfo->object; - try { - object->callbackEvent( asioCallbackInfo->streamId, index, (void *)NULL, (void *)NULL ); - } - catch (RtError &exception) { - fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage()); - return; - } + MUTEX_UNLOCK( &stream_.mutex ); - return; + RtApi::tickStreamTime(); + return SUCCESS; } -void sampleRateChanged(ASIOSampleRate sRate) +void sampleRateChanged( ASIOSampleRate sRate ) { // The ASIO documentation says that this usually only happens during // external sync. Audio processing is not stopped by the driver, @@ -3599,32 +3097,33 @@ void sampleRateChanged(ASIOSampleRate sRate) // sample rate status of an AES/EBU or S/PDIF digital input at the // audio device. - RtAudio *object = (RtAudio *) asioCallbackInfo->object; + RtApi *object = (RtApi *) asioCallbackInfo->object; try { - object->stopStream( asioCallbackInfo->streamId ); + object->stopStream(); } - catch (RtError &exception) { - fprintf(stderr, "\nRtAudio: sampleRateChanged() error (%s)!\n\n", exception.getMessage()); + catch ( RtError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; return; } - fprintf(stderr, "\nRtAudio: ASIO driver reports sample rate changed to %d ... stream stopped!!!", (int) sRate); + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; } -long asioMessages(long selector, long value, void* message, double* opt) +long asioMessages( long selector, long value, void* message, double* opt ) { long ret = 0; - switch(selector) { + + 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) + 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: @@ -3634,7 +3133,7 @@ long asioMessages(long selector, long value, void* message, double* opt) // done by completely destruct is. I.e. ASIOStop(), // ASIODisposeBuffers(), Destruction Afterwards you initialize the // driver again. - fprintf(stderr, "\nRtAudio: ASIO driver reset requested!!!"); + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; ret = 1L; break; case kAsioResyncRequest: @@ -3645,7 +3144,8 @@ long asioMessages(long selector, long value, void* message, double* opt) // which could lose data because the Mutex was held too long by // another thread. However a driver can issue it in other // situations, too. - fprintf(stderr, "\nRtAudio: ASIO driver resync requested!!!"); + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; ret = 1L; break; case kAsioLatenciesChanged: @@ -3653,7 +3153,7 @@ long asioMessages(long selector, long value, void* message, double* opt) // latencies changed. Beware, it this does not mean that the // buffer sizes have changed! You might need to update internal // delay data. - fprintf(stderr, "\nRtAudio: ASIO driver latency may have changed!!!"); + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; ret = 1L; break; case kAsioEngineVersion: @@ -3670,7 +3170,7 @@ long asioMessages(long selector, long value, void* message, double* opt) ret = 0; break; case kAsioSupportsTimeCode: - // Informs the driver wether application is interested in time + // 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; @@ -3679,3327 +3179,4283 @@ long asioMessages(long selector, long value, void* message, double* opt) return ret; } -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) +static const char* getAsioErrorString( ASIOError result ) { - // Don't attempt to load another driver if a stream is already open. - if ( streams.size() > 0 ) { - sprintf(message, "RtAudio: unable to load ASIO driver while a stream is open."); - error(RtError::WARNING); - return FAILURE; - } + 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 - // For ASIO, a duplex stream MUST use the same driver. - if ( mode == INPUT && stream->mode == OUTPUT && stream->device[0] != device ) { - sprintf(message, "RtAudio: ASIO duplex stream must use the same device for input and output."); - error(RtError::WARNING); - return FAILURE; - } - // Only load the driver once for duplex stream. - ASIOError result; - if ( mode != INPUT || stream->mode != OUTPUT ) { - if ( !drivers.loadDriver( devices[device].name ) ) { - sprintf(message, "RtAudio: ASIO error loading driver (%s).", devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } +#if defined(__WINDOWS_DS__) // Windows DirectSound API - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - char details[32]; - if ( result == ASE_HWMalfunction ) - sprintf(details, "hardware malfunction"); - else if ( result == ASE_NoMemory ) - sprintf(details, "no memory"); - else if ( result == ASE_NotPresent ) - sprintf(details, "driver/hardware not present"); - else - sprintf(details, "unspecified"); - sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } - } +// Modified by Robin Davies, October 2005 +// - Improvements to DirectX pointer chasing. +// - Backdoor RtDsStatistics hook provides DirectX performance information. +// - 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 - // Check the device channel count. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", - devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } +#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 - if ( ( mode == OUTPUT && channels > outputChannels) || - ( mode == INPUT && channels > inputChannels) ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO driver (%s) does not support requested channel count (%d).", - devices[device].name, channels); - error(RtError::DEBUG_WARNING); - return FAILURE; - } - stream->nDeviceChannels[mode] = channels; - stream->nUserChannels[mode] = channels; +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 - // Verify the sample rate is supported. - result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO driver (%s) does not support requested sample rate (%d).", - devices[device].name, sampleRate); - error(RtError::DEBUG_WARNING); - return FAILURE; - } +#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 - // Set the sample rate. - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO driver (%s) error setting sample rate (%d).", - devices[device].name, sampleRate); - error(RtError::DEBUG_WARNING); - return FAILURE; - } +static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if (laterPointer > earlierPointer) + return laterPointer - earlierPointer; + else + return laterPointer - earlierPointer + bufferSize; +} - // 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(); - sprintf(message, "RtAudio: ASIO driver (%s) error getting data format.", - devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } +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; +} - // 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; - } +// 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; } +}; - if ( stream->deviceFormat[mode] == 0 ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.", - devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } +/* +RtApiDs::RtDsStatistics RtApiDs::statistics; - // 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(); - sprintf(message, "RtAudio: ASIO driver (%s) error getting buffer size.", - devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } +// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns. +RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() +{ + RtDsStatistics s = statistics; - if ( *bufferSize < minSize ) *bufferSize = minSize; - else if ( *bufferSize > maxSize ) *bufferSize = maxSize; - else if ( granularity == -1 ) { - // Make sure bufferSize is a power of two. - double power = log10( *bufferSize ) / log10( 2.0 ); - *bufferSize = pow( 2.0, floor(power+0.5) ); - if ( *bufferSize < minSize ) *bufferSize = minSize; - else if ( *bufferSize > maxSize ) *bufferSize = maxSize; - else *bufferSize = preferSize; - } + // update the calculated fields. + if ( s.inputFrameSize != 0 ) + s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate; - if ( mode == INPUT && stream->mode == OUTPUT && stream->bufferSize != *bufferSize ) - cout << "possible input/output buffersize discrepancy" << endl; + if ( s.outputFrameSize != 0 ) + s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate; - stream->bufferSize = *bufferSize; - stream->nBuffers = 2; + return s; +} +*/ - // ASIO always uses deinterleaved channels. - stream->deInterleave[mode] = true; +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); - // 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. - if ( mode == INPUT && stream->mode == OUTPUT ) { - free(stream->callbackInfo.buffers); - result = ASIODisposeBuffers(); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO driver (%s) error disposing previously allocated buffers.", - devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } - } +static char* getErrorString( int code ); - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - int i, nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1]; - stream->callbackInfo.buffers = 0; - ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); - stream->callbackInfo.buffers = (void *) bufferInfos; - ASIOBufferInfo *infos = bufferInfos; - for ( i=0; inDeviceChannels[1]; i++, infos++ ) { - infos->isInput = ASIOTrue; - infos->channelNum = i; - infos->buffers[0] = infos->buffers[1] = 0; - } +extern "C" unsigned __stdcall callbackHandler( void *ptr ); - for ( i=0; inDeviceChannels[0]; i++, infos++ ) { - infos->isInput = ASIOFalse; - infos->channelNum = i; - infos->buffers[0] = infos->buffers[1] = 0; - } +struct EnumInfo { + bool isInput; + bool getDefault; + bool findIndex; + unsigned int counter; + unsigned int index; + LPGUID id; + std::string name; - // 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( bufferInfos, nChannels, stream->bufferSize, &asioCallbacks); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - sprintf(message, "RtAudio: ASIO driver (%s) error creating buffers.", - devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } - - // 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->nUserChannels[mode] > 1 && stream->deInterleave[mode]) - stream->doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; - } - - if ( stream->doConvertBuffer[mode] ) { - - long buffer_bytes; - bool makeBuffer = true; - if ( mode == OUTPUT ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == INPUT - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == OUTPUT && stream->deviceBuffer ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes < bytes_out ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; - } - } - - 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->sampleRate = sampleRate; - asioCallbackInfo = &stream->callbackInfo; - stream->callbackInfo.object = (void *) this; - stream->callbackInfo.waitTime = (unsigned long) (200.0 * stream->bufferSize / stream->sampleRate); - - return SUCCESS; - - memory_error: - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); - - if (stream->callbackInfo.buffers) - free(stream->callbackInfo.buffers); - stream->callbackInfo.buffers = 0; - - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; - } - sprintf(message, "RtAudio: error allocating buffer memory (%s).", - devices[device].name); - error(RtError::WARNING); - return FAILURE; -} - -void RtAudio :: cancelStreamCallback(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->callbackInfo.usingCallback) { - - if (stream->state == STREAM_RUNNING) - stopStream( streamId ); - - MUTEX_LOCK(&stream->mutex); - - stream->callbackInfo.usingCallback = false; - stream->callbackInfo.userData = NULL; - stream->state = STREAM_STOPPED; - stream->callbackInfo.callback = NULL; - - MUTEX_UNLOCK(&stream->mutex); - } -} - -void RtAudio :: closeStream(int streamId) -{ - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); - return; - } - - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; - - if (stream->state == STREAM_RUNNING) - ASIOStop(); - - ASIODisposeBuffers(); - //ASIOExit(); - drivers.removeCurrentDriver(); - - DeleteCriticalSection(&stream->mutex); - - if (stream->callbackInfo.buffers) - free(stream->callbackInfo.buffers); - - if (stream->userBuffer) - free(stream->userBuffer); - - if (stream->deviceBuffer) - free(stream->deviceBuffer); - - free(stream); - streams.erase(streamId); -} - -void RtAudio :: startStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_RUNNING) { - MUTEX_UNLOCK(&stream->mutex); - return; - } - - stream->callbackInfo.blockTick = true; - stream->callbackInfo.stopStream = false; - stream->callbackInfo.streamId = streamId; - ASIOError result = ASIOStart(); - if ( result != ASE_OK ) { - sprintf(message, "RtAudio: ASIO error starting device (%s).", - devices[stream->device[0]].name); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - stream->state = STREAM_RUNNING; - - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: stopStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - if (stream->state == STREAM_STOPPED) { - MUTEX_UNLOCK(&stream->mutex); - return; - } - - ASIOError result = ASIOStop(); - if ( result != ASE_OK ) { - sprintf(message, "RtAudio: ASIO error stopping device (%s).", - devices[stream->device[0]].name); - MUTEX_UNLOCK(&stream->mutex); - error(RtError::DRIVER_ERROR); - } - stream->state = STREAM_STOPPED; - - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: abortStream(int streamId) -{ - stopStream( streamId ); -} + EnumInfo() + : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {} +}; -// I don't know how this function can be implemented. -int RtAudio :: streamWillBlock(int streamId) +RtApiDs :: RtApiDs() { - sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for ASIO."); - error(RtError::WARNING); - return 0; + // 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; } -void RtAudio :: tickStream(int streamId) +RtApiDs :: ~RtApiDs() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->state == STREAM_STOPPED) - return; - - if (stream->callbackInfo.usingCallback) { - sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!"); - error(RtError::WARNING); - return; - } - - // Block waiting here until the user data is processed in callbackEvent(). - while ( stream->callbackInfo.blockTick ) - Sleep(stream->callbackInfo.waitTime); - - MUTEX_LOCK(&stream->mutex); - - stream->callbackInfo.blockTick = true; - - MUTEX_UNLOCK(&stream->mutex); + if ( coInitialized_ ) CoUninitialize(); // balanced call. + if ( stream_.state != STREAM_CLOSED ) closeStream(); } -void RtAudio :: callbackEvent(int streamId, int bufferIndex, void *inData, void *outData) +unsigned int RtApiDs :: getDefaultInputDevice( void ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - CALLBACK_INFO *info = asioCallbackInfo; - if ( !info->usingCallback ) { - // Block waiting here until we get new user data in tickStream(). - while ( !info->blockTick ) - Sleep(info->waitTime); - } - else if ( info->stopStream ) { - // Check if the stream should be stopped (via the previous user - // callback return value). We stop the stream here, rather than - // after the function call, so that output data can first be - // processed. - this->stopStream(asioCallbackInfo->streamId); - return; - } - - MUTEX_LOCK(&stream->mutex); - int nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1]; - int bufferBytes; - ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) info->buffers; - if ( stream->mode == INPUT || stream->mode == DUPLEX ) { - - bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[1]); - if (stream->doConvertBuffer[1]) { - - // Always interleave ASIO input data. - for ( int i=0; inDeviceChannels[1]; i++, bufferInfos++ ) - memcpy(&stream->deviceBuffer[i*bufferBytes], bufferInfos->buffers[bufferIndex], bufferBytes ); - - if ( stream->doByteSwap[1] ) - byteSwapBuffer(stream->deviceBuffer, - stream->bufferSize * stream->nDeviceChannels[1], - stream->deviceFormat[1]); - convertStreamBuffer(stream, INPUT); - - } - else { // single channel only - memcpy(stream->userBuffer, bufferInfos->buffers[bufferIndex], bufferBytes ); - - if (stream->doByteSwap[1]) - byteSwapBuffer(stream->userBuffer, - stream->bufferSize * stream->nUserChannels[1], - stream->userFormat); - } - } - - if ( info->usingCallback ) { - RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback; - if ( callback(stream->userBuffer, stream->bufferSize, info->userData) ) - info->stopStream = true; - } - - if ( stream->mode == OUTPUT || stream->mode == DUPLEX ) { - - bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[0]); - if (stream->doConvertBuffer[0]) { - - convertStreamBuffer(stream, OUTPUT); - if ( stream->doByteSwap[0] ) - byteSwapBuffer(stream->deviceBuffer, - stream->bufferSize * stream->nDeviceChannels[0], - stream->deviceFormat[0]); - - // Always de-interleave ASIO output data. - for ( int i=0; inDeviceChannels[0]; i++, bufferInfos++ ) { - memcpy(bufferInfos->buffers[bufferIndex], - &stream->deviceBuffer[i*bufferBytes], bufferBytes ); - } - } - else { // single channel only - - if (stream->doByteSwap[0]) - byteSwapBuffer(stream->userBuffer, - stream->bufferSize * stream->nUserChannels[0], - stream->userFormat); - - memcpy(bufferInfos->buffers[bufferIndex], stream->userBuffer, bufferBytes ); - } + // Count output devices. + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; } - if ( !info->usingCallback ) - info->blockTick = false; - - MUTEX_UNLOCK(&stream->mutex); -} - -void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - stream->callbackInfo.callback = (void *) callback; - stream->callbackInfo.userData = userData; - stream->callbackInfo.usingCallback = true; -} - -//******************** End of __WINDOWS_ASIO__ *********************// - -#elif defined(__WINDOWS_DS__) // Windows DirectSound API - -#include - -// Declarations for utility functions, callbacks, and structures -// specific to the DirectSound implementation. -static bool CALLBACK deviceCountCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext); - -static bool CALLBACK deviceInfoCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext); - -static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext); - -static bool CALLBACK deviceIdCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext); - -static char* getErrorString(int code); - -struct enum_info { - char name[64]; - LPGUID id; - bool isInput; - bool isValid; -}; - -int RtAudio :: getDefaultInputDevice(void) -{ - enum_info info; - info.name[0] = '\0'; - - // Enumerate through devices to find the default output. - HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Error performing default input device enumeration: %s.", - getErrorString(result)); - error(RtError::WARNING); + // Now enumerate input devices until we find the id = NULL. + info.isInput = true; + info.getDefault = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); return 0; } - for ( int i=0; i 0 ) return info.counter - 1; return 0; } -int RtAudio :: getDefaultOutputDevice(void) +unsigned int RtApiDs :: getDefaultOutputDevice( void ) { - enum_info info; - info.name[0] = '\0'; - - // Enumerate through devices to find the default output. - HRESULT result = DirectSoundEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Error performing default output device enumeration: %s.", - getErrorString(result)); - error(RtError::WARNING); + // Enumerate output devices until we find the id = NULL. + EnumInfo info; + info.getDefault = true; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); return 0; } - for ( int i=0; i 0 ) return info.counter - 1; return 0; } -void RtAudio :: initialize(void) +unsigned int RtApiDs :: getDeviceCount( void ) { - int i, ins = 0, outs = 0, count = 0; - HRESULT result; - nDevices = 0; - // Count DirectSound devices. - result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &outs); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.", - getErrorString(result)); - error(RtError::DRIVER_ERROR); + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); } // Count DirectSoundCapture devices. - result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &ins); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.", - getErrorString(result)); - error(RtError::DRIVER_ERROR); + info.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); } - count = ins + outs; - if (count == 0) return; + return info.counter; +} - std::vector info(count); - for (i=0; iGetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - // Get capture device info and check capabilities. - result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.", - getErrorString(result)); - error(RtError::DRIVER_ERROR); + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); } - // Parse the devices and check validity. Devices are considered - // invalid if they cannot be opened, they report < 1 supported - // channels, or they report no supported data (capture only). - for (i=0; iRelease(); - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); - } + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; - // Copy the names to our devices structures. - int index = 0; - for (i=0; iname, 64 ); - dsinfo.isValid = false; - - // Enumerate through input devices to find the id (if it exists). - HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Error performing input device id enumeration: %s.", - getErrorString(result)); - error(RtError::WARNING); - return; + dsinfo.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); } - // Do capture probe first. - if ( dsinfo.isValid == false ) - goto playback_probe; + if ( dsinfo.name.empty() ) return info; - LPDIRECTSOUNDCAPTURE input; + LPDIRECTSOUNDCAPTURE input; result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto playback_probe; + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } - DSCCAPS in_caps; - in_caps.dwSize = sizeof(in_caps); - result = input->GetCaps( &in_caps ); - if ( FAILED(result) ) { + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { input->Release(); - sprintf(message, "RtAudio: Could not get DirectSound capture capabilities (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto playback_probe; + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; } // Get input channel information. - info->minInputChannels = 1; - info->maxInputChannels = in_caps.dwChannels; + info.inputChannels = inCaps.dwChannels; // Get sample rate and format information. - if( in_caps.dwChannels == 2 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->nativeFormats |= RTAUDIO_SINT8; - - if ( info->nativeFormats & RTAUDIO_SINT16 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates[info->nSampleRates++] = 44100; - } - else if ( info->nativeFormats & RTAUDIO_SINT8 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates[info->nSampleRates++] = 44100; - } - } - else if ( in_caps.dwChannels == 1 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->nativeFormats |= RTAUDIO_SINT16; - if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->nativeFormats |= RTAUDIO_SINT8; - if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->nativeFormats |= RTAUDIO_SINT8; - - if ( info->nativeFormats & RTAUDIO_SINT16 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates[info->nSampleRates++] = 44100; - } - else if ( info->nativeFormats & RTAUDIO_SINT8 ) { - if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates[info->nSampleRates++] = 11025; - if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates[info->nSampleRates++] = 22050; - if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates[info->nSampleRates++] = 44100; - } - } - else info->minInputChannels = 0; // technically, this would be an error + 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 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 ); + } + } + 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 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 ); + } + } + else info.inputChannels = 0; // technically, this would be an error input->Release(); - playback_probe: + if ( info.inputChannels == 0 ) return info; - dsinfo.isValid = false; - - // Enumerate through output devices to find the id (if it exists). - result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Error performing output device id enumeration: %s.", - getErrorString(result)); - error(RtError::WARNING); - return; - } + if ( getDefaultInputDevice() == device ) + info.isDefaultInput = true; - // Now do playback probe. - if ( dsinfo.isValid == false ) - goto check_parameters; + // Copy name and return. + info.name = dsinfo.name; + info.probed = true; + return info; +} - LPDIRECTSOUND output; - DSCAPS out_caps; - result = DirectSoundCreate( dsinfo.id, &output, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto check_parameters; +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; } - out_caps.dwSize = sizeof(out_caps); - result = output->GetCaps( &out_caps ); - if ( FAILED(result) ) { - output->Release(); - sprintf(message, "RtAudio: Could not get DirectSound playback capabilities (%s): %s.", - info->name, getErrorString(result)); - error(RtError::WARNING); - goto check_parameters; + // Enumerate through devices to find the id (if it exists). Note + // that we have to do the output enumeration first, even if this is + // an input device, in order for the device counter to be correct. + EnumInfo dsinfo; + dsinfo.findIndex = true; + dsinfo.index = device; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + return FAILURE; } - // Get output channel information. - info->minOutputChannels = 1; - info->maxOutputChannels = ( out_caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - - // Get sample rate information. Use capture device rate information - // if it exists. - if ( info->nSampleRates == 0 ) { - info->sampleRates[0] = (int) out_caps.dwMinSecondarySampleRate; - info->sampleRates[1] = (int) out_caps.dwMaxSecondarySampleRate; - if ( out_caps.dwFlags & DSCAPS_CONTINUOUSRATE ) - info->nSampleRates = -1; - else if ( out_caps.dwMinSecondarySampleRate == out_caps.dwMaxSecondarySampleRate ) { - if ( out_caps.dwMinSecondarySampleRate == 0 ) { - // This is a bogus driver report ... fake the range and cross - // your fingers. - info->sampleRates[0] = 11025; - info->sampleRates[1] = 48000; - info->nSampleRates = -1; /* continuous range */ - sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using defaults (%s).", - info->name); - error(RtError::DEBUG_WARNING); - } - else { - info->nSampleRates = 1; - } - } - else if ( (out_caps.dwMinSecondarySampleRate < 1000.0) && - (out_caps.dwMaxSecondarySampleRate > 50000.0) ) { - // This is a bogus driver report ... support for only two - // distant rates. We'll assume this is a range. - info->nSampleRates = -1; - sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using range (%s).", - info->name); - error(RtError::WARNING); + if ( mode == OUTPUT ) { + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; } - else info->nSampleRates = 2; } - else { - // Check input rates against output rate range - for ( int i=info->nSampleRates-1; i>=0; i-- ) { - if ( info->sampleRates[i] <= out_caps.dwMaxSecondarySampleRate ) - break; - info->nSampleRates--; + else { // mode == INPUT + dsinfo.isInput = true; + HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + return FAILURE; } - while ( info->sampleRates[0] < out_caps.dwMinSecondarySampleRate ) { - info->nSampleRates--; - for ( int i=0; inSampleRates; i++) - info->sampleRates[i] = info->sampleRates[i+1]; - if ( info->nSampleRates <= 0 ) break; + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; } } - // Get format information. - if ( out_caps.dwFlags & DSCAPS_PRIMARY16BIT ) info->nativeFormats |= RTAUDIO_SINT16; - if ( out_caps.dwFlags & DSCAPS_PRIMARY8BIT ) info->nativeFormats |= RTAUDIO_SINT8; - - output->Release(); - - check_parameters: - if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) - return; - if ( info->nSampleRates == 0 || info->nativeFormats == 0 ) - return; - - // Determine duplex status. - if (info->maxInputChannels < info->maxOutputChannels) - info->maxDuplexChannels = info->maxInputChannels; - else - info->maxDuplexChannels = info->maxOutputChannels; - if (info->minInputChannels < info->minOutputChannels) - info->minDuplexChannels = info->minInputChannels; - else - info->minDuplexChannels = info->minOutputChannels; - - if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true; - else info->hasDuplexSupport = false; - - info->probed = true; - - return; -} - -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ - HRESULT result; - HWND hWnd = GetForegroundWindow(); // 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. However, for console // applications, no sound was produced when using GetDesktopWindow(). - long buffer_size; - LPVOID audioPtr; - DWORD dataLen; - int nBuffers; + HWND hWnd = GetForegroundWindow(); // 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. - if (numberOfBuffers < 2) - nBuffers = 2; - else - nBuffers = numberOfBuffers; + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; - // Define the wave format structure (16-bit PCM, srate, channels) + // Create the wave format structure. The data format setting will + // be determined later. WAVEFORMATEX waveFormat; - ZeroMemory(&waveFormat, sizeof(WAVEFORMATEX)); + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels; + waveFormat.nChannels = channels + firstChannel; waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - // Determine the data format. - if ( devices[device].nativeFormats ) { // 8-bit and/or 16-bit support - if ( format == RTAUDIO_SINT8 ) { - if ( devices[device].nativeFormats & RTAUDIO_SINT8 ) - waveFormat.wBitsPerSample = 8; - else - waveFormat.wBitsPerSample = 16; - } - else { - if ( devices[device].nativeFormats & RTAUDIO_SINT16 ) - waveFormat.wBitsPerSample = 16; - else - waveFormat.wBitsPerSample = 8; - } - } - else { - sprintf(message, "RtAudio: no reported data formats for DirectSound device (%s).", - devices[device].name); - error(RtError::DEBUG_WARNING); - return FAILURE; - } - - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + // Determine the device buffer size. By default, 32k, but we will + // grow it to make allowances for very large software buffer sizes. + DWORD dsBufferSize = 0; + DWORD dsPointerLeadTime = 0; + long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this - enum_info dsinfo; - strncpy( dsinfo.name, devices[device].name, 64 ); - dsinfo.isValid = false; + void *ohandle = 0, *bhandle = 0; if ( mode == OUTPUT ) { - if ( devices[device].maxOutputChannels < channels ) + LPDIRECTSOUND output; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; + } - // Enumerate through output devices to find the id (if it exists). - result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Error performing output device id enumeration: %s.", - getErrorString(result)); - error(RtError::DEBUG_WARNING); + DSCAPS outCaps; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } - if ( dsinfo.isValid == false ) { - sprintf(message, "RtAudio: DS output device (%s) id not found!", devices[device].name); - error(RtError::DEBUG_WARNING); + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); return FAILURE; } - LPGUID id = dsinfo.id; - LPDIRECTSOUND object; - LPDIRECTSOUNDBUFFER buffer; - DSBUFFERDESC bufferDescription; - - result = DirectSoundCreate( id, &object, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::DEBUG_WARNING); - 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 > (DWORD) bufferBytes ) + bufferBytes *= 2; // Set cooperative level to DSSCL_EXCLUSIVE - result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to set DirectSound cooperative level (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.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. - // The default is 8-bit, 22 kHz! - // Setup the DS primary buffer description. - ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); - bufferDescription.dwSize = sizeof(DSBUFFERDESC); + // 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 - result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to access DS primary buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } // Set the primary DS buffer sound format. - result = buffer->SetFormat(&waveFormat); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to set DS primary buffer format (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + result = buffer->SetFormat( &waveFormat ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } // Setup the secondary DS buffer description. - buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; - ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC)); - bufferDescription.dwSize = sizeof(DSBUFFERDESC); + dsBufferSize = (DWORD) bufferBytes; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = buffer_size; + bufferDescription.dwBufferBytes = bufferBytes; 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 = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to create secondary DS buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } } // Get the buffer size ... might be different from what we specified. DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof(DSBCAPS); - buffer->GetCaps(&dsbcaps); - buffer_size = dsbcaps.dwBufferBytes; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + bufferBytes = dsbcaps.dwBufferBytes; // Lock the DS buffer - result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } // Zero the DS buffer - ZeroMemory(audioPtr, dataLen); + ZeroMemory( audioPtr, dataLen ); // Unlock the DS buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to unlock DS buffer(%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } - stream->handle[0].object = (void *) object; - stream->handle[0].buffer = (void *) buffer; - stream->nDeviceChannels[0] = channels; + dsBufferSize = bufferBytes; + ohandle = (void *) output; + bhandle = (void *) buffer; } if ( mode == INPUT ) { - if ( devices[device].maxInputChannels < channels ) - return FAILURE; - - // Enumerate through input devices to find the id (if it exists). - result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Error performing input device id enumeration: %s.", - getErrorString(result)); - error(RtError::DEBUG_WARNING); + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } - if ( dsinfo.isValid == false ) { - sprintf(message, "RtAudio: DS input device (%s) id not found!", devices[device].name); - error(RtError::DEBUG_WARNING); + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } - LPGUID id = dsinfo.id; - LPDIRECTSOUNDCAPTURE object; - LPDIRECTSOUNDCAPTUREBUFFER buffer; - DSCBUFFERDESC bufferDescription; - - result = DirectSoundCaptureCreate( id, &object, NULL ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; return FAILURE; } - // Setup the secondary DS buffer description. - buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8; - ZeroMemory(&bufferDescription, sizeof(DSCBUFFERDESC)); - bufferDescription.dwSize = sizeof(DSCBUFFERDESC); - bufferDescription.dwFlags = 0; + // 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; + + // Setup the secondary DS buffer description. + dsBufferSize = bufferBytes; + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = buffer_size; + bufferDescription.dwBufferBytes = bufferBytes; bufferDescription.lpwfxFormat = &waveFormat; // Create the capture buffer. - result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to create DS capture buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } // Lock the capture buffer - result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } // Zero the buffer - ZeroMemory(audioPtr, dataLen); + ZeroMemory( audioPtr, dataLen ); // Unlock the buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - object->Release(); - sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", - devices[device].name, getErrorString(result)); - error(RtError::WARNING); + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } - stream->handle[1].object = (void *) object; - stream->handle[1].buffer = (void *) buffer; - stream->nDeviceChannels[1] = channels; + dsBufferSize = bufferBytes; + ohandle = (void *) input; + bhandle = (void *) buffer; } - stream->userFormat = format; - if ( waveFormat.wBitsPerSample == 8 ) - stream->deviceFormat[mode] = RTAUDIO_SINT8; - else - stream->deviceFormat[mode] = RTAUDIO_SINT16; - stream->nUserChannels[mode] = channels; - *bufferSize = buffer_size / (channels * nBuffers * waveFormat.wBitsPerSample / 8); - stream->bufferSize = *bufferSize; + // 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 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; + // 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 - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; - - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; + 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] ) { + if ( stream_.doConvertBuffer[mode] ) { - long buffer_bytes; bool makeBuffer = true; - if ( mode == OUTPUT ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == INPUT - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == OUTPUT && stream->deviceBuffer ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes < bytes_out ) makeBuffer = false; + 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 ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; + 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; + } } } - stream->device[mode] = device; - stream->state = STREAM_STOPPED; - if ( stream->mode == OUTPUT && mode == INPUT ) + // 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; + stream_.mode = DUPLEX; else - stream->mode = mode; - stream->nBuffers = nBuffers; - stream->sampleRate = sampleRate; + 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. + unsigned threadId; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.isRunning = true; + 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; + } return SUCCESS; - memory_error: - if (stream->handle[0].object) { - LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - if (buffer) { - buffer->Release(); - stream->handle[0].buffer = NULL; + 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(); } - object->Release(); - stream->handle[0].object = NULL; + 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; } - if (stream->handle[1].object) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - if (buffer) { - buffer->Release(); - stream->handle[1].buffer = NULL; + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } - object->Release(); - stream->handle[1].object = NULL; } - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - sprintf(message, "RtAudio: error allocating buffer memory (%s).", - devices[device].name); - error(RtError::WARNING); + return FAILURE; } -void RtAudio :: cancelStreamCallback(int streamId) +void RtApiDs :: closeStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->callbackInfo.usingCallback) { + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - if (stream->state == STREAM_RUNNING) - stopStream( streamId ); + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); - MUTEX_LOCK(&stream->mutex); + 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; + } - stream->callbackInfo.usingCallback = false; - WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE)stream->callbackInfo.thread ); - stream->callbackInfo.thread = 0; - stream->callbackInfo.callback = NULL; - stream->callbackInfo.userData = NULL; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - MUTEX_UNLOCK(&stream->mutex); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } -void RtAudio :: closeStream(int streamId) +void RtApiDs :: startStream() { - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtError::WARNING ); return; } - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + // Increase scheduler frequency on lesser windows (a side-effect of + // increasing timer accuracy). On greater windows (Win2K or later), + // this is already in effect. - if (stream->callbackInfo.usingCallback) { - stream->callbackInfo.usingCallback = false; - WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE)stream->callbackInfo.thread ); + MUTEX_LOCK( &stream_.mutex ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + timeBeginPeriod( 1 ); + + /* + memset( &statistics, 0, sizeof( statistics ) ); + statistics.sampleRate = stream_.sampleRate; + statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; + */ + + 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] ); } - DeleteCriticalSection(&stream->mutex); + HRESULT result = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0]; - if (stream->handle[0].object) { - LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - if (buffer) { - buffer->Stop(); - buffer->Release(); + 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; } - object->Release(); } - if (stream->handle[1].object) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - if (buffer) { - buffer->Stop(); - buffer->Release(); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1]; + + 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; } - object->Release(); } - if (stream->userBuffer) - free(stream->userBuffer); + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; - if (stream->deviceBuffer) - free(stream->deviceBuffer); + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - free(stream); - streams.erase(streamId); + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); } -void RtAudio :: startStream(int streamId) +void RtApiDs :: stopStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK( &stream_.mutex ); - if (stream->state == STREAM_RUNNING) - goto unlock; + 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 = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); + } + + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - HRESULT result; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - result = buffer->Play(0, 0, DSBPLAY_LOOPING ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to start DS buffer (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + // 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::abortStream: 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::abortStream: 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) stream->handle[1].buffer; - result = buffer->Start(DSCBSTART_LOOPING ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to start DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; + + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: 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::abortStream: 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::abortStream: 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; } - stream->state = STREAM_RUNNING; unlock: - MUTEX_UNLOCK(&stream->mutex); + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + stream_.state = STREAM_STOPPED; + 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 = 1; + + stopStream(); } -void RtAudio :: stopStream(int streamId) +void RtApiDs :: callbackEvent() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + 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; + } - MUTEX_LOCK(&stream->mutex); + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; - if (stream->state == STREAM_STOPPED) { - MUTEX_UNLOCK(&stream->mutex); + // 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; } - // There is no specific DirectSound API call to "drain" a buffer - // before stopping. We can hack this for playback by writing zeroes - // for another bufferSize * nBuffers frames. For capture, the - // concept is less clear so we'll repeat what we do in the - // abortStream() case. + MUTEX_LOCK( &stream_.mutex ); + + // 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 dsBufferSize; + DWORD currentWritePos, safeWritePos; + DWORD currentReadPos, safeReadPos; + DWORD leadPos; + UINT nextWritePos; + +#ifdef GENERATE_DEBUG_LOG + DWORD writeTime, readTime; +#endif + LPVOID buffer1 = NULL; LPVOID buffer2 = NULL; DWORD bufferSize1 = 0; DWORD bufferSize2 = 0; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - DWORD currentPos, safePos; - long buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; - buffer_bytes *= formatBytes(stream->deviceFormat[0]); + char *buffer; + long bufferBytes; + + if ( stream_.mode == DUPLEX && !buffersRolling ) { + 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 initialWritePos, initialSafeWritePos; + DWORD initialReadPos, initialSafeReadPos; + + result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos ); + 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( ¤tWritePos, &safeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break; + Sleep( 1 ); + } + + assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + buffersRolling = true; + handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); + handle->bufferPointer[1] = safeReadPos; + } + + 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. - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - UINT nextWritePos = stream->handle[0].bufferPointer; - dsBufferSize = buffer_bytes * stream->nBuffers; + // 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; inBuffers; i++) { + DWORD dsBufferSize = handle->dsBufferSize[0]; + nextWritePos = handle->bufferPointer[0]; + DWORD endWrite; + while ( true ) { // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); } - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset - DWORD endWrite = nextWritePos + buffer_bytes; + leadPos = safeWritePos + handle->dsPointerLeadTime[0]; + if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; + if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset + endWrite = nextWritePos + bufferBytes; // Check whether the entire write region is behind the play pointer. - while ( currentPos < endWrite ) { - float millis = (endWrite - currentPos) * 900.0; - millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); + if ( leadPos >= endWrite ) break; - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + // If we are here, then we must wait until the play pointer gets + // beyond the write region. The approach here is to use the + // Sleep() function to suspend operation until safePos catches + // up. Calculate number of milliseconds to wait as: + // time = distance * (milliseconds/second) * fudgefactor / + // ((bytes/sample) * (samples/second)) + // A "fudgefactor" less than 1 is used because it was found + // that sleeping too long was MUCH worse than sleeping for + // several shorter periods. + double millis = ( endWrite - leadPos ) * 900.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + if ( millis > 50.0 ) { + static int nOverruns = 0; + ++nOverruns; } + Sleep( (DWORD) millis ); + } - // Lock free space in the buffer - result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } + //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) { + // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ); + //} - // Zero the free space - ZeroMemory(buffer1, bufferSize1); - if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2); + if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + //++statistics.numberOfWriteUnderruns; + handle->xrun[0] = true; + nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize; + while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize; + handle->bufferPointer[0] = nextWritePos; + endWrite = nextWritePos + bufferBytes; + } - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; - stream->handle[0].bufferPointer = nextWritePos; + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePos, 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 ); } + nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePos; - // If we play again, start at the beginning of the buffer. - stream->handle[0].bufferPointer = 0; + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } } - if (stream->mode == INPUT || stream->mode == DUPLEX) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - buffer1 = NULL; - bufferSize1 = 0; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - result = buffer->Stop(); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + // 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 ); } - dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; - dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPos = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; - // 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, dsBufferSize, &buffer1, &bufferSize1, NULL, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPos + 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 ( safeReadPos < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPos; + + handle->xrun[1] = true; + //++statistics.numberOfReadOverruns; + // 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 ) + nextReadPos = safeReadPos-2*bufferBytes; + else + nextReadPos = safeReadPos-bufferBytes-adjustment; + + //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos; + //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize; + if ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + + } + else { + // In pre=roll time. Just do it. + nextReadPos = safeReadPos-bufferBytes; + while ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + } + endRead = nextReadPos + bufferBytes; + } + } + else { // mode == INPUT + while ( safeReadPos < endRead ) { + // See comments for playback. + double millis = (endRead - safeReadPos) * 900.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + } } - // Zero the DS buffer - ZeroMemory(buffer1, bufferSize1); + //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) ) + // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ); - // Unlock the DS buffer - result = buffer->Unlock(buffer1, bufferSize1, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + // Lock free space in the buffer + result = dsBuffer->Lock( nextReadPos, 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 we start recording again, we must begin at beginning of buffer. - stream->handle[1].bufferPointer = 0; + 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 + nextReadPos = ( nextReadPos + 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] = nextReadPos; + + // 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; jstate = STREAM_STOPPED; +#endif + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - MUTEX_UNLOCK(&stream->mutex); + RtApi::tickStreamTime(); } -void RtAudio :: abortStream(int streamId) +// Definitions for utility functions and callbacks +// specific to the DirectSound implementation. + +extern "C" unsigned __stdcall callbackHandler( void *ptr ) { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiDs *object = (RtApiDs *) info->object; + bool* isRunning = &info->isRunning; - MUTEX_LOCK(&stream->mutex); + while ( *isRunning == true ) { + object->callbackEvent(); + } - if (stream->state == STREAM_STOPPED) - goto unlock; + _endthreadex( 0 ); + return 0; +} - HRESULT result; - long dsBufferSize; - LPVOID audioPtr; - DWORD dataLen; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - result = buffer->Stop(); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to stop DS buffer (%s): %s", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } +#include "tchar.h" + +std::string convertTChar( LPCTSTR name ) +{ + std::string s; + +#if defined( UNICODE ) || defined( _UNICODE ) + // Yes, this conversion doesn't make sense for two-byte characters + // but RtAudio is currently written to return an std::string of + // one-byte chars for the device name. + for ( unsigned int i=0; ibufferSize * stream->nDeviceChannels[0]; - dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; + return s; +} - // 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, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ) +{ + EnumInfo *info = (EnumInfo *) lpContext; - // Zero the DS buffer - ZeroMemory(audioPtr, dataLen); + HRESULT hr; + if ( info->isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; - // Unlock the DS buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + 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 ) + info->counter++; } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; - // If we start playing again, we must begin at beginning of buffer. - stream->handle[0].bufferPointer = 0; + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + info->counter++; + } + object->Release(); } - if (stream->mode == INPUT || stream->mode == DUPLEX) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - audioPtr = NULL; - dataLen = 0; + if ( info->getDefault && lpguid == NULL ) return FALSE; - result = buffer->Stop(); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + if ( info->findIndex && info->counter > info->index ) { + info->id = lpguid; + info->name = convertTChar( description ); + return FALSE; + } + + return TRUE; +} + +static char* getErrorString( int code ) +{ + switch ( code ) { + + case DSERR_ALLOCATED: + return "Already allocated"; + + case DSERR_CONTROLUNAVAIL: + return "Control unavailable"; + + case DSERR_INVALIDPARAM: + return "Invalid parameter"; + + case DSERR_INVALIDCALL: + return "Invalid call"; + + case DSERR_GENERIC: + return "Generic error"; + + case DSERR_PRIOLEVELNEEDED: + return "Priority level needed"; + + case DSERR_OUTOFMEMORY: + return "Out of memory"; + + case DSERR_BADFORMAT: + return "The sample rate or the channel format is not supported"; + + case DSERR_UNSUPPORTED: + return "Not supported"; + + case DSERR_NODRIVER: + return "No driver"; + + case DSERR_ALREADYINITIALIZED: + return "Already initialized"; + + case DSERR_NOAGGREGATION: + return "No aggregation"; + + case DSERR_BUFFERLOST: + return "Buffer lost"; + + case DSERR_OTHERAPPHASPRIO: + return "Another application already has priority"; + + case DSERR_UNINITIALIZED: + return "Uninitialized"; + + default: + return "DirectSound unknown error"; + } +} +//******************** End of __WINDOWS_DS__ *********************// +#endif + + +#if defined(__LINUX_ALSA__) + +#include +#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]; + + AlsaHandle() + :synchronized(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; +} - dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1]; - dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; - // 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, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + 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 ); + } - // Zero the DS buffer - ZeroMemory(audioPtr, dataLen); + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } - // Unlock the DS buffer - result = buffer->Unlock(audioPtr, dataLen, NULL, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } + + foundDevice: + + 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; +} + +bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) + +{ +#if defined(__RTAUDIO_DEBUG__) + snd_output_t *out; + snd_output_stdio_attach(&out, stderr, 0); +#endif + + // I'm not using the "plug" interface ... too much inconsistent behavior. + + 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::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 ); + 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: + + 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 number, which in ALSA is referred to as the "period". + int dir; + unsigned int periods = 0; + if ( options ) periods = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if ( periods < 2 ) periods = 2; + 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; + } + + // Set the buffer (or period) size. + 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; + + // 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, 0x7fffffff ); + snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, INT_MAX ); + 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; + } + 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. 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) + pthread_attr_setschedpolicy( &attr, SCHED_RR ); +#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 ) { + 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; + } + + stream_.callbackInfo.isRunning = false; + pthread_join( stream_.callbackInfo.thread, NULL ); + + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + 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 ) { + 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 we start recording again, we must begin at beginning of buffer. - stream->handle[1].bufferPointer = 0; + 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_STOPPED; + + stream_.state = STREAM_RUNNING; unlock: - MUTEX_UNLOCK(&stream->mutex); + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); } -int RtAudio :: streamWillBlock(int streamId) +void RtApiAlsa :: stopStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); - - int channels; - int frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; - - HRESULT result; - DWORD currentPos, safePos; - channels = 1; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - UINT nextWritePos = stream->handle[0].bufferPointer; - channels = stream->nDeviceChannels[0]; - DWORD dsBufferSize = stream->bufferSize * channels; - dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers; + // Change the state before the lock to improve shutdown response + // when using a callback. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + 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 ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset - frames = currentPos - nextWritePos; - frames /= channels * formatBytes(stream->deviceFormat[0]); + 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; + } } - if (stream->mode == INPUT || stream->mode == DUPLEX) { + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - UINT nextReadPos = stream->handle[1].bufferPointer; - channels = stream->nDeviceChannels[1]; - DWORD dsBufferSize = stream->bufferSize * channels; - dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers; + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset + // Change the state before the lock to improve shutdown response + // when using a callback. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); - if (stream->mode == DUPLEX ) { - // Take largest value of the two. - int temp = safePos - nextReadPos; - temp /= channels * formatBytes(stream->deviceFormat[1]); - frames = ( temp > frames ) ? temp : frames; - } - else { - frames = safePos - nextReadPos; - frames /= channels * formatBytes(stream->deviceFormat[1]); + 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; } } - frames = stream->bufferSize - frames; - if (frames < 0) frames = 0; + 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: - MUTEX_UNLOCK(&stream->mutex); - return frames; + MUTEX_UNLOCK( &stream_.mutex ); + + stream_.state = STREAM_STOPPED; + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); } -void RtAudio :: tickStream(int streamId) +void RtApiAlsa :: callbackEvent() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + if ( stream_.state == STREAM_STOPPED ) { + if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds + return; + } - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->callbackInfo.usingCallback) Sleep(50); // sleep 50 milliseconds + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); return; } - else if (stream->callbackInfo.usingCallback) { - RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; - stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + + int doStopStream = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + 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 ); - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK( &stream_.mutex ); // The state might change while waiting on a mutex. - if (stream->state == STREAM_STOPPED) { - MUTEX_UNLOCK(&stream->mutex); - return; - } + if ( stream_.state == STREAM_STOPPED ) goto unlock; - HRESULT result; - DWORD currentPos, safePos; - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; + int result; char *buffer; - long buffer_bytes; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; - // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, OUTPUT); - buffer = stream->deviceBuffer; - buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0]; - buffer_bytes *= formatBytes(stream->deviceFormat[0]); + 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; - buffer_bytes = stream->bufferSize * stream->nUserChannels[0]; - buffer_bytes *= formatBytes(stream->userFormat); + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; } - // No byte swapping necessary in DirectSound implementation. - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer; - UINT nextWritePos = stream->handle[0].bufferPointer; - DWORD dsBufferSize = buffer_bytes * stream->nBuffers; - - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ideviceFormat[0]) * stream->sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + if ( result < (int) stream_.bufferSize ) { + // Either an error or underrun occured. + if ( result == -EPIPE ) { + snd_pcm_state_t state = snd_pcm_state( handle[1] ); + if ( state == SND_PCM_STATE_XRUN ) { + apiInfo->xrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); } - if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset + error( RtError::WARNING ); + goto unlock; } - // Lock free space in the buffer - result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - // Copy our buffer into the DS buffer - CopyMemory(buffer1, buffer, bufferSize1); - if (buffer2 != NULL) CopyMemory(buffer2, buffer+bufferSize1, bufferSize2); + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.", - devices[stream->device[0]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize; - stream->handle[0].bufferPointer = nextWritePos; + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; } - if (stream->mode == INPUT || stream->mode == DUPLEX) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - // Setup parameters. - if (stream->doConvertBuffer[1]) { - buffer = stream->deviceBuffer; - buffer_bytes = stream->bufferSize * stream->nDeviceChannels[1]; - buffer_bytes *= formatBytes(stream->deviceFormat[1]); + // 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; - buffer_bytes = stream->bufferSize * stream->nUserChannels[1]; - buffer_bytes *= formatBytes(stream->userFormat); + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; } - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer; - UINT nextReadPos = stream->handle[1].bufferPointer; - DWORD dsBufferSize = buffer_bytes * stream->nBuffers; + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition(¤tPos, &safePos); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + // 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; ideviceFormat[1]) * stream->sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos ); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + if ( result < (int) stream_.bufferSize ) { + // Either an error or underrun occured. + if ( result == -EPIPE ) { + snd_pcm_state_t state = snd_pcm_state( handle[0] ); + if ( state == SND_PCM_STATE_XRUN ) { + apiInfo->xrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } } - - if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset - } - - // Lock free space in the buffer - result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to lock DS buffer during capture (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); - } - - // Copy our buffer into the DS buffer - CopyMemory(buffer, buffer1, bufferSize1); - if (buffer2 != NULL) CopyMemory(buffer+bufferSize1, buffer2, bufferSize2); - - // Update our buffer offset and unlock sound buffer - nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize; - dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2); - if ( FAILED(result) ) { - sprintf(message, "RtAudio: Unable to unlock DS buffer during capture (%s): %s.", - devices[stream->device[1]].name, getErrorString(result)); - error(RtError::DRIVER_ERROR); + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtError::WARNING ); + goto unlock; } - stream->handle[1].bufferPointer = nextReadPos; - - // No byte swapping necessary in DirectSound implementation. - // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, INPUT); + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; } - MUTEX_UNLOCK(&stream->mutex); + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - if (stream->callbackInfo.usingCallback && stopStream) - this->stopStream(streamId); + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); + else if ( doStopStream == 2 ) this->abortStream(); } -// Definitions for utility functions and callbacks -// specific to the DirectSound implementation. - -extern "C" unsigned __stdcall callbackHandler(void *ptr) +extern "C" void *alsaCallbackHandler( void *ptr ) { - CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; - RtAudio *object = (RtAudio *) info->object; - int stream = info->streamId; - bool *usingCallback = &info->usingCallback; + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR + // Set a higher scheduler priority (P.J. Leonard) + struct sched_param param; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + param.sched_priority = min + ( max - min ) / 2; // Is this the best number? + sched_setscheduler( 0, SCHED_RR, ¶m ); +#endif - while ( *usingCallback ) { - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; - } + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); } - _endthreadex( 0 ); - return 0; + pthread_exit( NULL ); } -void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); +//******************** End of __LINUX_ALSA__ *********************// +#endif - CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo; - if ( info->usingCallback ) { - sprintf(message, "RtAudio: A callback is already set for this stream!"); - error(RtError::WARNING); - return; - } - info->callback = (void *) callback; - info->userData = userData; - info->usingCallback = true; - info->object = (void *) this; - info->streamId = streamId; +#if defined(__LINUX_OSS__) - unsigned thread_id; - info->thread = _beginthreadex(NULL, 0, &callbackHandler, - &stream->callbackInfo, 0, &thread_id); - if (info->thread == 0) { - info->usingCallback = false; - sprintf(message, "RtAudio: error starting callback thread!"); - error(RtError::THREAD_ERROR); - } +#include +#include +#include +#include +#include "oss/soundcard.h" +#include +#include - // When spawning multiple threads in quick succession, it appears to be - // necessary to wait a bit for each to initialize ... another windoism! - Sleep(1); -} +extern "C" void *ossCallbackHandler(void * ptr); -static bool CALLBACK deviceCountCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext) -{ - int *pointer = ((int *) lpContext); - (*pointer)++; +// A structure to hold various information related to the OSS API +// implementation. +struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; - return true; -} + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; -static bool CALLBACK deviceInfoCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext) +RtApiOss :: RtApiOss() { - enum_info *info = ((enum_info *) lpContext); - while (strlen(info->name) > 0) info++; - - strncpy(info->name, lpcstrDescription, 64); - info->id = lpguid; - - HRESULT hr; - info->isValid = false; - if (info->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) - info->isValid = 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 ) - info->isValid = true; - } - object->Release(); - } - - return true; + // Nothing to do here. } -static bool CALLBACK defaultDeviceCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext) +RtApiOss :: ~RtApiOss() { - enum_info *info = ((enum_info *) lpContext); - - if ( lpguid == NULL ) { - strncpy(info->name, lpcstrDescription, 64); - return false; - } - - return true; + if ( stream_.state != STREAM_CLOSED ) closeStream(); } -static bool CALLBACK deviceIdCallback(LPGUID lpguid, - LPCSTR lpcstrDescription, - LPCSTR lpcstrModule, - LPVOID lpContext) +unsigned int RtApiOss :: getDeviceCount( void ) { - enum_info *info = ((enum_info *) lpContext); + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return 0; + } - if ( strncmp( info->name, lpcstrDescription, 64 ) == 0 ) { - info->id = lpguid; - info->isValid = true; - return false; + 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; } - return true; + return sysinfo.numaudios; } -static char* getErrorString(int code) +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) { - switch (code) { - - case DSERR_ALLOCATED: - return "Direct Sound already allocated"; - - case DSERR_CONTROLUNAVAIL: - return "Direct Sound control unavailable"; + RtAudio::DeviceInfo info; + info.probed = false; - case DSERR_INVALIDPARAM: - return "Direct Sound invalid parameter"; - - case DSERR_INVALIDCALL: - return "Direct Sound invalid call"; - - case DSERR_GENERIC: - return "Direct Sound generic error"; - - case DSERR_PRIOLEVELNEEDED: - return "Direct Sound Priority level needed"; - - case DSERR_OUTOFMEMORY: - return "Direct Sound out of memory"; - - case DSERR_BADFORMAT: - return "Direct Sound bad format"; - - case DSERR_UNSUPPORTED: - return "Direct Sound unsupported error"; + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return info; + } - case DSERR_NODRIVER: - return "Direct Sound no driver error"; + 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; + } - case DSERR_ALREADYINITIALIZED: - return "Direct Sound already initialized"; + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } - case DSERR_NOAGGREGATION: - return "Direct Sound no aggregation"; + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - case DSERR_BUFFERLOST: - return "Direct Sound buffer lost"; + 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; - case DSERR_OTHERAPPHASPRIO: - return "Direct Sound other app has priority"; + // 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] ); + } + } - case DSERR_UNINITIALIZED: - return "Direct Sound uninitialized"; + 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; + } - default: - return "Direct Sound unknown error"; - } + return info; } -//******************** End of __WINDOWS_DS__ *********************// - -#elif defined(__IRIX_AL__) // SGI's AL API for IRIX -#include -#include - -void RtAudio :: initialize(void) +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 ) { - // Count cards and devices - nDevices = 0; - - // Determine the total number of input and output devices. - nDevices = alQueryValues(AL_SYSTEM, AL_DEVICES, 0, 0, 0, 0); - if (nDevices < 0) { - sprintf(message, "RtAudio: AL error counting devices: %s.", - alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; } - if (nDevices <= 0) return; - - ALvalue *vls = (ALvalue *) new ALvalue[nDevices]; - - // Allocate the RTAUDIO_DEVICE structures. - devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE)); - if (devices == NULL) { - sprintf(message, "RtAudio: memory allocation error!"); - error(RtError::MEMORY_ERROR); + 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; } - // Write device ascii identifiers and resource ids to device info - // structure. - char name[32]; - int outs, ins, i; - ALpv pvs[1]; - pvs[0].param = AL_NAME; - pvs[0].value.ptr = name; - pvs[0].sizeIn = 32; - - outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices, 0, 0); - if (outs < 0) { - sprintf(message, "RtAudio: AL error getting output devices: %s.", - alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); + 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; } - for (i=0; i= 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; } - ins = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &vls[outs], nDevices-outs, 0, 0); - if (ins < 0) { - sprintf(message, "RtAudio: AL error getting input devices: %s.", - alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); + 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; } - for (i=outs; iid[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; } - return 0; -} + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; -int RtAudio :: getDefaultOutputDevice(void) -{ - ALvalue value; - int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, &value, 1, 0, 0); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting default output device id: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); + // 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; } - else { - for ( int i=0; iid[0]; - if (resource > 0) { + // 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; + } - // Probe output device parameters. - result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); + // 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 { - info->maxOutputChannels = value.i; - info->minOutputChannels = 1; + } + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; } - - result = alGetParamInfo(resource, AL_RATE, &pinfo); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; } - else { - info->nSampleRates = 0; - for (i=0; i= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { - info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } - } + } + 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; } - - // The AL library supports all our formats, except 24-bit and 32-bit ints. - info->nativeFormats = (RTAUDIO_FORMAT) 51; } - - // Now get input resource ID if it exists. - resource = info->id[1]; - if (resource > 0) { - - // Probe input device parameters. - result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; } - else { - info->maxInputChannels = value.i; - info->minInputChannels = 1; + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; } + } - result = alGetParamInfo(resource, AL_RATE, &pinfo); - if (result < 0) { - sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.", - info->name, alGetErrorString(oserror())); - error(RtError::WARNING); + 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 { - // In the case of the default device, these values will - // overwrite the rates determined for the output device. Since - // the input device is most likely to be more limited than the - // output device, this is ok. - info->nSampleRates = 0; - for (i=0; i= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) { - info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i]; - info->nSampleRates++; - } - } + 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; } - - // The AL library supports all our formats, except 24-bit and 32-bit ints. - info->nativeFormats = (RTAUDIO_FORMAT) 51; } - if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) - return; - if ( info->nSampleRates == 0 ) - return; - - // Determine duplex status. - if (info->maxInputChannels < info->maxOutputChannels) - info->maxDuplexChannels = info->maxInputChannels; - else - info->maxDuplexChannels = info->maxOutputChannels; - if (info->minInputChannels < info->minOutputChannels) - info->minDuplexChannels = info->minInputChannels; - else - info->minDuplexChannels = info->minOutputChannels; - - if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true; - else info->hasDuplexSupport = false; - - info->probed = true; - - return; -} - -bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream, - STREAM_MODE mode, int channels, - int sampleRate, RTAUDIO_FORMAT format, - int *bufferSize, int numberOfBuffers) -{ - int result, resource, nBuffers; - ALconfig al_config; - ALport port; - ALpv pvs[2]; - - // Get a new ALconfig structure. - al_config = alNewConfig(); - if ( !al_config ) { - sprintf(message,"RtAudio: can't get AL config: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); + 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 channels. - result = alSetChannels(al_config, channels); - if ( result < 0 ) { - sprintf(message,"RtAudio: can't set %d channels in AL config: %s.", - channels, alGetErrorString(oserror())); - error(RtError::WARNING); + // 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 queue size. The al API doesn't provide a - // means for querying the minimum/maximum buffer size of a device, - // so if the specified size doesn't work, take whatever the - // al_config structure returns. - if ( numberOfBuffers < 1 ) - nBuffers = 1; - else - nBuffers = numberOfBuffers; - long buffer_size = *bufferSize * nBuffers; - result = alSetQueueSize(al_config, buffer_size); // in sample frames - if ( result < 0 ) { - // Get the buffer size specified by the al_config and try that. - buffer_size = alGetQueueSize(al_config); - result = alSetQueueSize(al_config, buffer_size); - if ( result < 0 ) { - sprintf(message,"RtAudio: can't set buffer size (%ld) in AL config: %s.", - buffer_size, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - *bufferSize = buffer_size / nBuffers; + // 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; - // Set the data format. - stream->userFormat = format; - stream->deviceFormat[mode] = format; - if (format == RTAUDIO_SINT8) { - result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); - result = alSetWidth(al_config, AL_SAMPLE_8); - } - else if (format == RTAUDIO_SINT16) { - result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP); - result = alSetWidth(al_config, AL_SAMPLE_16); - } - else if (format == RTAUDIO_SINT24) { - // Our 24-bit format assumes the upper 3 bytes of a 4 byte word. - // The AL library uses the lower 3 bytes, so we'll need to do our - // own conversion. - result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); - stream->deviceFormat[mode] = RTAUDIO_FLOAT32; - } - else if (format == RTAUDIO_SINT32) { - // The AL library doesn't seem to support the 32-bit integer - // format, so we'll need to do our own conversion. - result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); - stream->deviceFormat[mode] = RTAUDIO_FLOAT32; - } - else if (format == RTAUDIO_FLOAT32) - result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT); - else if (format == RTAUDIO_FLOAT64) - result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE); + // 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 ) { - sprintf(message,"RtAudio: AL error setting sample format in AL config: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); return FAILURE; } - if (mode == OUTPUT) { - - // Set our device. - if (device == 0) - resource = AL_DEFAULT_OUTPUT; - else - resource = devices[device].id[0]; - result = alSetDevice(al_config, resource); - if ( result == -1 ) { - sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", - devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } - - // Open the port. - port = alOpenPort("RtAudio Output Port", "w", al_config); - if( !port ) { - sprintf(message,"RtAudio: AL error opening output port: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - 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; - // Set the sample rate - pvs[0].param = AL_MASTER_CLOCK; - pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; - pvs[1].param = AL_RATE; - pvs[1].value.ll = alDoubleToFixed((double)sampleRate); - result = alSetParams(resource, pvs, 2); - if ( result < 0 ) { - alClosePort(port); - sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", - sampleRate, devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } + 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; } - else { // mode == INPUT - // Set our device. - if (device == 0) - resource = AL_DEFAULT_INPUT; - else - resource = devices[device].id[1]; - result = alSetDevice(al_config, resource); - if ( result == -1 ) { - sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.", - devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; - } + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; - // Open the port. - port = alOpenPort("RtAudio Output Port", "r", al_config); - if( !port ) { - sprintf(message,"RtAudio: AL error opening input port: %s.", - alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; + // 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; } - - // Set the sample rate - pvs[0].param = AL_MASTER_CLOCK; - pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE; - pvs[1].param = AL_RATE; - pvs[1].value.ll = alDoubleToFixed((double)sampleRate); - result = alSetParams(resource, pvs, 2); - if ( result < 0 ) { - alClosePort(port); - sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.", - sampleRate, devices[device].name, alGetErrorString(oserror())); - error(RtError::WARNING); - return FAILURE; + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; } - } - - alFreeConfig(al_config); - - stream->nUserChannels[mode] = channels; - stream->nDeviceChannels[mode] = channels; - // Set handle and flags for buffer conversion - stream->handle[mode] = port; - stream->doConvertBuffer[mode] = false; - if (stream->userFormat != stream->deviceFormat[mode]) - stream->doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) { - - long buffer_bytes; - if (stream->nUserChannels[0] >= stream->nUserChannels[1]) - buffer_bytes = stream->nUserChannels[0]; - else - buffer_bytes = stream->nUserChannels[1]; + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + handle->id[mode] = fd; - buffer_bytes *= *bufferSize * formatBytes(stream->userFormat); - if (stream->userBuffer) free(stream->userBuffer); - stream->userBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->userBuffer == NULL) - goto memory_error; + // 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] ) { + if ( stream_.doConvertBuffer[mode] ) { - long buffer_bytes; bool makeBuffer = true; - if ( mode == OUTPUT ) - buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - else { // mode == INPUT - buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]); - if ( stream->mode == OUTPUT && stream->deviceBuffer ) { - long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]); - if ( buffer_bytes < bytes_out ) makeBuffer = false; + 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 ) { - buffer_bytes *= *bufferSize; - if (stream->deviceBuffer) free(stream->deviceBuffer); - stream->deviceBuffer = (char *) calloc(buffer_bytes, 1); - if (stream->deviceBuffer == NULL) - goto memory_error; + 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; - if ( stream->mode == OUTPUT && mode == INPUT ) + 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; - else - stream->mode = mode; - stream->nBuffers = nBuffers; - stream->bufferSize = *bufferSize; - stream->sampleRate = sampleRate; + 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) + pthread_attr_setschedpolicy( &attr, SCHED_RR ); +#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; - memory_error: - if (stream->handle[0]) { - alClosePort(stream->handle[0]); - stream->handle[0] = 0; + error: + if ( handle ) { + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; } - if (stream->handle[1]) { - alClosePort(stream->handle[1]); - stream->handle[1] = 0; + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } } - if (stream->userBuffer) { - free(stream->userBuffer); - stream->userBuffer = 0; + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - sprintf(message, "RtAudio: ALSA error allocating buffer memory for device (%s).", - devices[device].name); - error(RtError::WARNING); + return FAILURE; } -void RtAudio :: closeStream(int streamId) +void RtApiOss :: closeStream() { - // We don't want an exception to be thrown here because this - // function is called by our class destructor. So, do our own - // streamId check. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::WARNING); + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtError::WARNING ); return; } - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId]; + stream_.callbackInfo.isRunning = false; + pthread_join( stream_.callbackInfo.thread, NULL ); - if (stream->callbackInfo.usingCallback) { - pthread_cancel(stream->callbackInfo.thread); - pthread_join(stream->callbackInfo.thread, NULL); + OssHandle *handle = (OssHandle *) stream_.apiHandle; + 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; } - pthread_mutex_destroy(&stream->mutex); - - if (stream->handle[0]) - alClosePort(stream->handle[0]); - - if (stream->handle[1]) - alClosePort(stream->handle[1]); + if ( handle ) { + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } - if (stream->userBuffer) - free(stream->userBuffer); + 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); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - free(stream); - streams.erase(streamId); + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } -void RtAudio :: startStream(int streamId) +void RtApiOss :: startStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - if (stream->state == STREAM_RUNNING) + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtError::WARNING ); return; + } - // The AL port is ready as soon as it is opened. - stream->state = STREAM_RUNNING; -} - -void RtAudio :: stopStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK( &stream_.mutex ); - if (stream->state == STREAM_STOPPED) - goto unlock; + stream_.state = STREAM_RUNNING; - int result; - int buffer_size = stream->bufferSize * stream->nBuffers; + // No need to do anything else here ... OSS automatically starts + // when fed samples. - if (stream->mode == OUTPUT || stream->mode == DUPLEX) - alZeroFrames(stream->handle[0], buffer_size); + MUTEX_UNLOCK( &stream_.mutex ); +} - if (stream->mode == INPUT || stream->mode == DUPLEX) { - result = alDiscardFrames(stream->handle[1], buffer_size); - if (result == -1) { - sprintf(message, "RtAudio: AL error draining stream device (%s): %s.", - devices[stream->device[1]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); - } +void RtApiOss :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - stream->state = STREAM_STOPPED; - unlock: - MUTEX_UNLOCK(&stream->mutex); -} + // Change the state before the lock to improve shutdown response + // when using a callback. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); -void RtAudio :: abortStream(int streamId) -{ - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - MUTEX_LOCK(&stream->mutex); + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; - if (stream->state == STREAM_STOPPED) - goto unlock; + 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; + } - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + 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 ); + } + } - int buffer_size = stream->bufferSize * stream->nBuffers; - int result = alDiscardFrames(stream->handle[0], buffer_size); - if (result == -1) { - sprintf(message, "RtAudio: AL error aborting stream device (%s): %s.", - devices[stream->device[0]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); + 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; } - // There is no clear action to take on the input stream, since the - // port will continue to run in any event. - stream->state = STREAM_STOPPED; + 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: - MUTEX_UNLOCK(&stream->mutex); + MUTEX_UNLOCK( &stream_.mutex ); + + stream_.state = STREAM_STOPPED; + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); } -int RtAudio :: streamWillBlock(int streamId) +void RtApiOss :: abortStream() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); - - MUTEX_LOCK(&stream->mutex); + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - int frames = 0; - if (stream->state == STREAM_STOPPED) - goto unlock; + // Change the state before the lock to improve shutdown response + // when using a callback. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); - int err = 0; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { - err = alGetFillable(stream->handle[0]); - if (err < 0) { - sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", - devices[stream->device[0]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); + 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; } - frames = err; - - if (stream->mode == INPUT || stream->mode == DUPLEX) { - err = alGetFilled(stream->handle[1]); - if (err < 0) { - sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.", - devices[stream->device[1]].name, alGetErrorString(oserror())); - error(RtError::DRIVER_ERROR); + 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; } - if (frames > err) frames = err; } - frames = stream->bufferSize - frames; - if (frames < 0) frames = 0; - unlock: - MUTEX_UNLOCK(&stream->mutex); - return frames; + MUTEX_UNLOCK( &stream_.mutex ); + + stream_.state = STREAM_STOPPED; + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); } -void RtAudio :: tickStream(int streamId) +void RtApiOss :: callbackEvent() { - RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId); + if ( stream_.state == STREAM_STOPPED ) { + if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds + return; + } - int stopStream = 0; - if (stream->state == STREAM_STOPPED) { - if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); return; } - else if (stream->callbackInfo.usingCallback) { - RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback; - stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData); + + // Invoke user callback to get fresh output data. + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + 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 ); - MUTEX_LOCK(&stream->mutex); + MUTEX_LOCK( &stream_.mutex ); // The state might change while waiting on a mutex. - if (stream->state == STREAM_STOPPED) - goto unlock; + if ( stream_.state == STREAM_STOPPED ) goto unlock; + int result; char *buffer; - int channels; - RTAUDIO_FORMAT format; - if (stream->mode == OUTPUT || stream->mode == DUPLEX) { + int samples; + RtAudioFormat format; + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { // Setup parameters and do buffer conversion if necessary. - if (stream->doConvertBuffer[0]) { - convertStreamBuffer(stream, OUTPUT); - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[0]; - format = stream->deviceFormat[0]; + 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; - channels = stream->nUserChannels[0]; - format = stream->userFormat; + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; } // Do byte swapping if necessary. - if (stream->doByteSwap[0]) - byteSwapBuffer(buffer, stream->bufferSize * channels, format); + 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) ); - // Write interleaved samples to device. - alWriteFrames(stream->handle[0], buffer, stream->bufferSize); + 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 ); + goto unlock; + } } - if (stream->mode == INPUT || stream->mode == DUPLEX) { + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { // Setup parameters. - if (stream->doConvertBuffer[1]) { - buffer = stream->deviceBuffer; - channels = stream->nDeviceChannels[1]; - format = stream->deviceFormat[1]; + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; } else { - buffer = stream->userBuffer; - channels = stream->nUserChannels[1]; - format = stream->userFormat; + buffer = stream_.userBuffer[1]; + samples = stream_.bufferSize * stream_.nUserChannels[1]; + format = stream_.userFormat; } - // Read interleaved samples from device. - alReadFrames(stream->handle[1], buffer, stream->bufferSize); + // 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, stream->bufferSize * channels, format); + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); // Do buffer conversion if necessary. - if (stream->doConvertBuffer[1]) - convertStreamBuffer(stream, INPUT); + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); } unlock: - MUTEX_UNLOCK(&stream->mutex); + MUTEX_UNLOCK( &stream_.mutex ); - if (stream->callbackInfo.usingCallback && stopStream) - this->stopStream(streamId); + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); + else if ( doStopStream == 2 ) this->abortStream(); } -extern "C" void *callbackHandler(void *ptr) +extern "C" void *ossCallbackHandler( void *ptr ) { - CALLBACK_INFO *info = (CALLBACK_INFO *) ptr; - RtAudio *object = (RtAudio *) info->object; - int stream = info->streamId; - bool *usingCallback = &info->usingCallback; + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR + // Set a higher scheduler priority (P.J. Leonard) + struct sched_param param; + param.sched_priority = 39; // Is this the best number? + sched_setscheduler( 0, SCHED_RR, ¶m ); +#endif - while ( *usingCallback ) { + while ( *isRunning == true ) { pthread_testcancel(); - try { - object->tickStream(stream); - } - catch (RtError &exception) { - fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n", - exception.getMessage()); - break; - } + object->callbackEvent(); } - return 0; + pthread_exit( NULL ); } -//******************** End of __IRIX_AL__ *********************// - +//******************** End of __LINUX_OSS__ *********************// #endif // *************************************************** // // -// Private common (OS-independent) RtAudio methods. +// Protected common (OS-independent) RtAudio methods. // // *************************************************** // // This method can be modified to control the behavior of error -// message reporting and throwing. -void RtAudio :: error(RtError::TYPE type) +// message printing. +void RtApi :: error( RtError::Type type ) { - if (type == RtError::WARNING) { - fprintf(stderr, "\n%s\n\n", message); - } - else if (type == RtError::DEBUG_WARNING) { -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\n%s\n\n", message); -#endif - } - else { - fprintf(stderr, "\n%s\n\n", message); - throw RtError(message, type); - } + if ( type == RtError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else + throw( RtError( errorText_, type ) ); + errorStream_.str(""); // clear the ostringstream } -void *RtAudio :: verifyStream(int streamId) +void RtApi :: verifyStream() { - // Verify the stream key. - if ( streams.find( streamId ) == streams.end() ) { - sprintf(message, "RtAudio: invalid stream identifier!"); - error(RtError::INVALID_STREAM); + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtError::INVALID_USE ); } - - return streams[streamId]; } -void RtAudio :: clearDeviceInfo(RTAUDIO_DEVICE *info) +void RtApi :: clearStreamInfo() { - // Don't clear the name or DEVICE_ID fields here ... they are - // typically set prior to a call of this function. - info->probed = false; - info->maxOutputChannels = 0; - info->maxInputChannels = 0; - info->maxDuplexChannels = 0; - info->minOutputChannels = 0; - info->minInputChannels = 0; - info->minDuplexChannels = 0; - info->hasDuplexSupport = false; - info->nSampleRates = 0; - for (int i=0; isampleRates[i] = 0; - info->nativeFormats = 0; + 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] = 0; + 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(); + } } -int RtAudio :: formatBytes(RTAUDIO_FORMAT format) +unsigned int RtApi :: formatBytes( RtAudioFormat format ) { - if (format == RTAUDIO_SINT16) + if ( format == RTAUDIO_SINT16 ) return 2; - else if (format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || - format == RTAUDIO_FLOAT32) + else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || + format == RTAUDIO_FLOAT32 ) return 4; - else if (format == RTAUDIO_FLOAT64) + else if ( format == RTAUDIO_FLOAT64 ) return 8; - else if (format == RTAUDIO_SINT8) + else if ( format == RTAUDIO_SINT8 ) return 1; - sprintf(message,"RtAudio: undefined format in formatBytes()."); - error(RtError::WARNING); + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtError::WARNING ); return 0; } -void RtAudio :: convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode) +void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) { - // This method does format conversion, input/output channel compensation, and - // data interleaving/deinterleaving. 24-bit integers are assumed to occupy - // the upper three bytes of a 32-bit integer. - - int j, jump_in, jump_out, channels; - RTAUDIO_FORMAT format_in, format_out; - char *input, *output; - - if (mode == INPUT) { // convert device to user buffer - input = stream->deviceBuffer; - output = stream->userBuffer; - jump_in = stream->nDeviceChannels[1]; - jump_out = stream->nUserChannels[1]; - format_in = stream->deviceFormat[1]; - format_out = stream->userFormat; + 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 - input = stream->userBuffer; - output = stream->deviceBuffer; - jump_in = stream->nUserChannels[0]; - jump_out = stream->nDeviceChannels[0]; - format_in = stream->userFormat; - format_out = stream->deviceFormat[0]; - - // clear our device buffer when in/out duplex device channels are different - if ( stream->mode == DUPLEX && - stream->nDeviceChannels[0] != stream->nDeviceChannels[1] ) - memset(output, 0, stream->bufferSize * jump_out * formatBytes(format_out)); + 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]; } - channels = (jump_in < jump_out) ? jump_in : jump_out; - - // Set up the interleave/deinterleave offsets - std::vector offset_in(channels); - std::vector offset_out(channels); - if (mode == INPUT && stream->deInterleave[1]) { - for (int k=0; kbufferSize; - offset_out[k] = k; - jump_in = 1; + 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; kdeInterleave[0]) { - for (int k=0; kbufferSize; - jump_out = 1; + else { // no (de)interleaving + if ( stream_.userInterleaved ) { + for ( int k=0; k 0 ) { + if ( stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; kbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; j>= 8; } - in += jump_in; - out += jump_out; + in += info.inJump; + out += info.outJump; } } - else if (format_in == RTAUDIO_FLOAT32) { - FLOAT32 *in = (FLOAT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; j> 16) & 0x0000ffff); + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x0000ffff); } - in += jump_in; - out += jump_out; + in += info.inJump; + out += info.outJump; } } - else if (format_in == RTAUDIO_SINT32) { - INT32 *in = (INT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; j> 16) & 0x0000ffff); + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x0000ffff); } - in += jump_in; - out += jump_out; + in += info.inJump; + out += info.outJump; } } - else if (format_in == RTAUDIO_FLOAT32) { - FLOAT32 *in = (FLOAT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; j> 8) & 0x00ff); + if (info.inFormat == RTAUDIO_SINT16) { + Int16 *in = (Int16 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x00ff); } - in += jump_in; - out += jump_out; + in += info.inJump; + out += info.outJump; } } - else if (format_in == RTAUDIO_SINT24) { - INT32 *in = (INT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; j> 24) & 0x000000ff); + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x000000ff); } - in += jump_in; - out += jump_out; + in += info.inJump; + out += info.outJump; } } - else if (format_in == RTAUDIO_SINT32) { - INT32 *in = (INT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; j> 24) & 0x000000ff); + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 24) & 0x000000ff); } - in += jump_in; - out += jump_out; + in += info.inJump; + out += info.outJump; } } - else if (format_in == RTAUDIO_FLOAT32) { - FLOAT32 *in = (FLOAT32 *)input; - for (int i=0; ibufferSize; i++) { - for (j=0; jbufferSize; i++) { - for (j=0; j