From 29167f363ed88727e28d3003fc0cd2e56a04a5ac Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Wed, 30 Aug 2017 12:04:14 -0700 Subject: [PATCH] Converted line endings in RtAudio.cpp to unix format. --- RtAudio.cpp | 20654 +++++++++++++++++++++++++------------------------- 1 file changed, 10327 insertions(+), 10327 deletions(-) diff --git a/RtAudio.cpp b/RtAudio.cpp index 47333f6..00d63f0 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -1,10327 +1,10327 @@ -/************************************************************************/ -/*! \class RtAudio - \brief Realtime audio i/o C++ classes. - - RtAudio provides a common API (Application Programming Interface) - for realtime audio input/output across Linux (native ALSA, Jack, - and OSS), Macintosh OS X (CoreAudio and Jack), and Windows - (DirectSound, ASIO and WASAPI) operating systems. - - RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ - - RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2017 Gary P. Scavone - - Permission is hereby granted, free of charge, to any person - obtaining a copy of this software and associated documentation files - (the "Software"), to deal in the Software without restriction, - including without limitation the rights to use, copy, modify, merge, - publish, distribute, sublicense, and/or sell copies of the Software, - and to permit persons to whom the Software is furnished to do so, - subject to the following conditions: - - The above copyright notice and this permission notice shall be - included in all copies or substantial portions of the Software. - - Any person wishing to distribute modifications to the Software is - asked to send the modifications to the original developer so that - they can be incorporated into the canonical version. This is, - however, not a binding provision of this license. - - THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. -*/ -/************************************************************************/ - -// RtAudio: Version 5.0.0 - -#include "RtAudio.h" -#include -#include -#include -#include -#include -#include - -// Static variable definitions. -const unsigned int RtApi::MAX_SAMPLE_RATES = 14; -const unsigned int RtApi::SAMPLE_RATES[] = { - 4000, 5512, 8000, 9600, 11025, 16000, 22050, - 32000, 44100, 48000, 88200, 96000, 176400, 192000 -}; - -#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) - #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) - #define MUTEX_DESTROY(A) DeleteCriticalSection(A) - #define MUTEX_LOCK(A) EnterCriticalSection(A) - #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) - - #include "tchar.h" - - static std::string convertCharPointerToStdString(const char *text) - { - return std::string(text); - } - - static std::string convertCharPointerToStdString(const wchar_t *text) - { - int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); - std::string s( length-1, '\0' ); - WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); - return s; - } - -#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) - // pthread API - #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) - #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) - #define MUTEX_LOCK(A) pthread_mutex_lock(A) - #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) -#else - #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions - #define MUTEX_DESTROY(A) abs(*A) // dummy definitions -#endif - -// *************************************************** // -// -// RtAudio definitions. -// -// *************************************************** // - -std::string RtAudio :: getVersion( void ) -{ - return RTAUDIO_VERSION; -} - -void RtAudio :: getCompiledApi( std::vector &apis ) -{ - 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_PULSE__) - apis.push_back( LINUX_PULSE ); -#endif -#if defined(__LINUX_OSS__) - apis.push_back( LINUX_OSS ); -#endif -#if defined(__WINDOWS_ASIO__) - apis.push_back( WINDOWS_ASIO ); -#endif -#if defined(__WINDOWS_WASAPI__) - apis.push_back( WINDOWS_WASAPI ); -#endif -#if defined(__WINDOWS_DS__) - apis.push_back( WINDOWS_DS ); -#endif -#if defined(__MACOSX_CORE__) - apis.push_back( MACOSX_CORE ); -#endif -#if defined(__RTAUDIO_DUMMY__) - apis.push_back( RTAUDIO_DUMMY ); -#endif -} - -void RtAudio :: openRtApi( RtAudio::Api api ) -{ - if ( rtapi_ ) - delete rtapi_; - rtapi_ = 0; - -#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_PULSE__) - if ( api == LINUX_PULSE ) - rtapi_ = new RtApiPulse(); -#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_WASAPI__) - if ( api == WINDOWS_WASAPI ) - rtapi_ = new RtApiWasapi(); -#endif -#if defined(__WINDOWS_DS__) - if ( api == WINDOWS_DS ) - rtapi_ = new RtApiDs(); -#endif -#if defined(__MACOSX_CORE__) - if ( api == MACOSX_CORE ) - rtapi_ = new RtApiCore(); -#endif -#if defined(__RTAUDIO_DUMMY__) - if ( api == RTAUDIO_DUMMY ) - rtapi_ = new RtApiDummy(); -#endif -} - -RtAudio :: RtAudio( RtAudio::Api api ) -{ - rtapi_ = 0; - - if ( api != UNSPECIFIED ) { - // Attempt to open the specified API. - openRtApi( api ); - if ( rtapi_ ) return; - - // No compiled support for specified API value. Issue a debug - // warning and continue as if no API was specified. - std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; - } - - // Iterate through the compiled APIs and return as soon as we find - // one with at least one device or we reach the end of the list. - std::vector< RtAudio::Api > apis; - getCompiledApi( apis ); - for ( unsigned int i=0; igetDeviceCount() ) break; - } - - if ( rtapi_ ) return; - - // It should not be possible to get here because the preprocessor - // definition __RTAUDIO_DUMMY__ is automatically defined if no - // API-specific definitions are passed to the compiler. But just in - // case something weird happens, we'll thow an error. - std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; - throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); -} - -RtAudio :: ~RtAudio() -{ - if ( rtapi_ ) - delete rtapi_; -} - -void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, - RtAudio::StreamParameters *inputParameters, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, - RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options, - RtAudioErrorCallback errorCallback ) -{ - return rtapi_->openStream( outputParameters, inputParameters, format, - sampleRate, bufferFrames, callback, - userData, options, errorCallback ); -} - -// *************************************************** // -// -// Public RtApi definitions (see end of file for -// private or protected utility functions). -// -// *************************************************** // - -RtApi :: RtApi() -{ - stream_.state = STREAM_CLOSED; - stream_.mode = UNINITIALIZED; - stream_.apiHandle = 0; - stream_.userBuffer[0] = 0; - stream_.userBuffer[1] = 0; - MUTEX_INITIALIZE( &stream_.mutex ); - showWarnings_ = true; - firstErrorOccurred_ = false; -} - -RtApi :: ~RtApi() -{ - MUTEX_DESTROY( &stream_.mutex ); -} - -void RtApi :: openStream( RtAudio::StreamParameters *oParams, - RtAudio::StreamParameters *iParams, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, - RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options, - RtAudioErrorCallback errorCallback ) -{ - if ( stream_.state != STREAM_CLOSED ) { - errorText_ = "RtApi::openStream: a stream is already open!"; - error( RtAudioError::INVALID_USE ); - return; - } - - // Clear stream information potentially left from a previously open stream. - clearStreamInfo(); - - if ( oParams && oParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; - error( RtAudioError::INVALID_USE ); - return; - } - - if ( iParams && iParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; - error( RtAudioError::INVALID_USE ); - return; - } - - if ( oParams == NULL && iParams == NULL ) { - errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; - error( RtAudioError::INVALID_USE ); - return; - } - - if ( formatBytes(format) == 0 ) { - errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; - error( RtAudioError::INVALID_USE ); - return; - } - - 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( RtAudioError::INVALID_USE ); - return; - } - } - - unsigned int iChannels = 0; - if ( iParams ) { - iChannels = iParams->nChannels; - if ( iParams->deviceId >= nDevices ) { - errorText_ = "RtApi::openStream: input device parameter value is invalid."; - error( RtAudioError::INVALID_USE ); - return; - } - } - - bool result; - - if ( oChannels > 0 ) { - - result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) { - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - - if ( iChannels > 0 ) { - - result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) { - if ( oChannels > 0 ) closeStream(); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - - stream_.callbackInfo.callback = (void *) callback; - stream_.callbackInfo.userData = userData; - stream_.callbackInfo.errorCallback = (void *) errorCallback; - - if ( options ) options->numberOfBuffers = stream_.nBuffers; - stream_.state = STREAM_STOPPED; -} - -unsigned int RtApi :: getDefaultInputDevice( void ) -{ - // Should be implemented in subclasses if possible. - return 0; -} - -unsigned int RtApi :: getDefaultOutputDevice( void ) -{ - // Should be implemented in subclasses if possible. - return 0; -} - -void RtApi :: closeStream( void ) -{ - // MUST be implemented in subclasses! - return; -} - -bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, - unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, - RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, - RtAudio::StreamOptions * /*options*/ ) -{ - // MUST be implemented in subclasses! - return FAILURE; -} - -void RtApi :: tickStreamTime( void ) -{ - // Subclasses that do not provide their own implementation of - // getStreamTime should call this function once per buffer I/O to - // provide basic stream time support. - - stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); - -#if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); -#endif -} - -long RtApi :: getStreamLatency( void ) -{ - verifyStream(); - - long totalLatency = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - totalLatency = stream_.latency[0]; - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - totalLatency += stream_.latency[1]; - - return totalLatency; -} - -double RtApi :: getStreamTime( void ) -{ - verifyStream(); - -#if defined( HAVE_GETTIMEOFDAY ) - // Return a very accurate estimate of the stream time by - // adding in the elapsed time since the last tick. - struct timeval then; - struct timeval now; - - if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) - return stream_.streamTime; - - gettimeofday( &now, NULL ); - then = stream_.lastTickTimestamp; - return stream_.streamTime + - ((now.tv_sec + 0.000001 * now.tv_usec) - - (then.tv_sec + 0.000001 * then.tv_usec)); -#else - return stream_.streamTime; -#endif -} - -void RtApi :: setStreamTime( double time ) -{ - verifyStream(); - - if ( time >= 0.0 ) - stream_.streamTime = time; -#if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); -#endif -} - -unsigned int RtApi :: getStreamSampleRate( void ) -{ - verifyStream(); - - return stream_.sampleRate; -} - - -// *************************************************** // -// -// OS/API-specific methods. -// -// *************************************************** // - -#if defined(__MACOSX_CORE__) - -// The OS X CoreAudio API is designed to use a separate callback -// procedure for each of its audio devices. A single RtAudio duplex -// stream using two different devices is supported here, though it -// cannot be guaranteed to always behave correctly because we cannot -// synchronize these two callbacks. -// -// A property listener is installed for over/underrun information. -// However, no functionality is currently provided to allow property -// listeners to trigger user handlers because it is unclear what could -// be done if a critical stream parameter (buffer size, sample rate, -// device disconnect) notification arrived. The listeners entail -// quite a bit of extra code and most likely, a user program wouldn't -// be prepared for the result anyway. However, we do provide a flag -// to the client callback function to inform of an over/underrun. - -// A structure to hold various information related to the CoreAudio API -// implementation. -struct CoreHandle { - AudioDeviceID id[2]; // device ids -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - AudioDeviceIOProcID procId[2]; -#endif - UInt32 iStream[2]; // device stream index (or first if using multiple) - UInt32 nStreams[2]; // number of streams to use - bool xrun[2]; - char *deviceBuffer; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - CoreHandle() - :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -RtApiCore:: RtApiCore() -{ -#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) - // This is a largely undocumented but absolutely necessary - // requirement starting with OS-X 10.6. If not called, queries and - // updates to various audio device properties are not handled - // correctly. - CFRunLoopRef theRunLoop = NULL; - AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); - if ( result != noErr ) { - errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; - error( RtAudioError::WARNING ); - } -#endif -} - -RtApiCore :: ~RtApiCore() -{ - // The subclass destructor gets called before the base class - // destructor, so close an existing stream before deallocating - // apiDeviceId memory. - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiCore :: getDeviceCount( void ) -{ - // Find out how many audio devices there are, if any. - UInt32 dataSize; - AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; - error( RtAudioError::WARNING ); - return 0; - } - - return dataSize / sizeof( AudioDeviceID ); -} - -unsigned int RtApiCore :: getDefaultInputDevice( void ) -{ - unsigned int nDevices = getDeviceCount(); - if ( nDevices <= 1 ) return 0; - - AudioDeviceID id; - UInt32 dataSize = sizeof( AudioDeviceID ); - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; - error( RtAudioError::WARNING ); - return 0; - } - - dataSize *= nDevices; - AudioDeviceID deviceList[ nDevices ]; - property.mSelector = kAudioHardwarePropertyDevices; - result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; - error( RtAudioError::WARNING ); - return 0; - } - - for ( unsigned int i=0; i= nDevices ) { - errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - AudioDeviceID deviceList[ nDevices ]; - UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, - 0, NULL, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; - error( RtAudioError::WARNING ); - return info; - } - - AudioDeviceID id = deviceList[ device ]; - - // Get the device name. - info.name.erase(); - CFStringRef cfname; - dataSize = sizeof( CFStringRef ); - property.mSelector = kAudioObjectPropertyManufacturer; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); - int length = CFStringGetLength(cfname); - char *mname = (char *)malloc(length * 3 + 1); -#if defined( UNICODE ) || defined( _UNICODE ) - CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); -#else - CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); -#endif - info.name.append( (const char *)mname, strlen(mname) ); - info.name.append( ": " ); - CFRelease( cfname ); - free(mname); - - property.mSelector = kAudioObjectPropertyName; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); - length = CFStringGetLength(cfname); - char *name = (char *)malloc(length * 3 + 1); -#if defined( UNICODE ) || defined( _UNICODE ) - CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); -#else - CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); -#endif - info.name.append( (const char *)name, strlen(name) ); - CFRelease( cfname ); - free(name); - - // Get the output stream "configuration". - AudioBufferList *bufferList = nil; - property.mSelector = kAudioDevicePropertyStreamConfiguration; - property.mScope = kAudioDevicePropertyScopeOutput; - // property.mElement = kAudioObjectPropertyElementWildcard; - dataSize = 0; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != noErr || dataSize == 0 ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; - error( RtAudioError::WARNING ); - return info; - } - - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); - if ( result != noErr || dataSize == 0 ) { - free( bufferList ); - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get output channel information. - unsigned int i, nStreams = bufferList->mNumberBuffers; - for ( i=0; imBuffers[i].mNumberChannels; - free( bufferList ); - - // Get the input stream "configuration". - property.mScope = kAudioDevicePropertyScopeInput; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != noErr || dataSize == 0 ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; - error( RtAudioError::WARNING ); - return info; - } - - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); - if (result != noErr || dataSize == 0) { - free( bufferList ); - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get input channel information. - nStreams = bufferList->mNumberBuffers; - for ( i=0; imBuffers[i].mNumberChannels; - free( bufferList ); - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Probe the device sample rates. - bool isInput = false; - if ( info.outputChannels == 0 ) isInput = true; - - // Determine the supported sample rates. - property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; - if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != kAudioHardwareNoError || dataSize == 0 ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - UInt32 nRanges = dataSize / sizeof( AudioValueRange ); - AudioValueRange rangeList[ nRanges ]; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); - if ( result != kAudioHardwareNoError ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // The sample rate reporting mechanism is a bit of a mystery. It - // seems that it can either return individual rates or a range of - // rates. I assume that if the min / max range values are the same, - // then that represents a single supported rate and if the min / max - // range values are different, the device supports an arbitrary - // range of values (though there might be multiple ranges, so we'll - // use the most conservative range). - Float64 minimumRate = 1.0, maximumRate = 10000000000.0; - bool haveValueRange = false; - info.sampleRates.clear(); - for ( UInt32 i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = tmpSr; - - } else { - haveValueRange = true; - if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; - if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; - } - } - - if ( haveValueRange ) { - for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) { - info.sampleRates.push_back( SAMPLE_RATES[k] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - } - } - } - - // Sort and remove any redundant values - std::sort( info.sampleRates.begin(), info.sampleRates.end() ); - info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); - - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // CoreAudio always uses 32-bit floating point data for PCM streams. - // Thus, any other "physical" formats supported by the device are of - // no interest to the client. - info.nativeFormats = RTAUDIO_FLOAT32; - - if ( info.outputChannels > 0 ) - if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; - if ( info.inputChannels > 0 ) - if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; - - info.probed = true; - return info; -} - -static OSStatus callbackHandler( AudioDeviceID inDevice, - const AudioTimeStamp* /*inNow*/, - const AudioBufferList* inInputData, - const AudioTimeStamp* /*inInputTime*/, - AudioBufferList* outOutputData, - const AudioTimeStamp* /*inOutputTime*/, - void* infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - - RtApiCore *object = (RtApiCore *) info->object; - if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) - return kAudioHardwareUnspecifiedError; - else - return kAudioHardwareNoError; -} - -static OSStatus xrunListener( AudioObjectID /*inDevice*/, - UInt32 nAddresses, - const AudioObjectPropertyAddress properties[], - void* handlePointer ) -{ - CoreHandle *handle = (CoreHandle *) handlePointer; - for ( UInt32 i=0; ixrun[1] = true; - else - handle->xrun[0] = true; - } - } - - return kAudioHardwareNoError; -} - -static OSStatus rateListener( AudioObjectID inDevice, - UInt32 /*nAddresses*/, - const AudioObjectPropertyAddress /*properties*/[], - void* ratePointer ) -{ - Float64 *rate = (Float64 *) ratePointer; - UInt32 dataSize = sizeof( Float64 ); - AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); - return kAudioHardwareNoError; -} - -bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - AudioDeviceID deviceList[ nDevices ]; - UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, - 0, NULL, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; - return FAILURE; - } - - AudioDeviceID id = deviceList[ device ]; - - // Setup for stream mode. - bool isInput = false; - if ( mode == INPUT ) { - isInput = true; - property.mScope = kAudioDevicePropertyScopeInput; - } - else - property.mScope = kAudioDevicePropertyScopeOutput; - - // Get the stream "configuration". - AudioBufferList *bufferList = nil; - dataSize = 0; - property.mSelector = kAudioDevicePropertyStreamConfiguration; - result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); - if ( result != noErr || dataSize == 0 ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; - return FAILURE; - } - - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); - if (result != noErr || dataSize == 0) { - free( bufferList ); - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Search for one or more streams that contain the desired number of - // channels. CoreAudio devices can have an arbitrary number of - // streams and each stream can have an arbitrary number of channels. - // For each stream, a single buffer of interleaved samples is - // provided. RtAudio prefers the use of one stream of interleaved - // data or multiple consecutive single-channel streams. However, we - // now support multiple consecutive multi-channel streams of - // interleaved data as well. - UInt32 iStream, offsetCounter = firstChannel; - UInt32 nStreams = bufferList->mNumberBuffers; - bool monoMode = false; - bool foundStream = false; - - // First check that the device supports the requested number of - // channels. - UInt32 deviceChannels = 0; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; - - if ( deviceChannels < ( channels + firstChannel ) ) { - free( bufferList ); - errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Look for a single stream meeting our needs. - UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; - if ( streamChannels >= channels + offsetCounter ) { - firstStream = iStream; - channelOffset = offsetCounter; - foundStream = true; - break; - } - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - - // If we didn't find a single stream above, then we should be able - // to meet the channel specification with multiple streams. - if ( foundStream == false ) { - monoMode = true; - offsetCounter = firstChannel; - for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - - firstStream = iStream; - channelOffset = offsetCounter; - Int32 channelCounter = channels + offsetCounter - streamChannels; - - if ( streamChannels > 1 ) monoMode = false; - while ( channelCounter > 0 ) { - streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; - if ( streamChannels > 1 ) monoMode = false; - channelCounter -= streamChannels; - streamCount++; - } - } - - free( bufferList ); - - // Determine the buffer size. - AudioValueRange bufferRange; - dataSize = sizeof( AudioValueRange ); - property.mSelector = kAudioDevicePropertyBufferFrameSizeRange; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange ); - - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; - else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; - - // Set the buffer size. For multiple streams, I'm assuming we only - // need to make this setting for the master channel. - UInt32 theSize = (UInt32) *bufferSize; - dataSize = sizeof( UInt32 ); - property.mSelector = kAudioDevicePropertyBufferFrameSize; - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize ); - - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - *bufferSize = theSize; - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 1; - - // Try to set "hog" mode ... it's not clear to me this is working. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) { - pid_t hog_pid; - dataSize = sizeof( hog_pid ); - property.mSelector = kAudioDevicePropertyHogMode; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( hog_pid != getpid() ) { - hog_pid = getpid(); - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - } - - // Check and if necessary, change the sample rate for the device. - Float64 nominalRate; - dataSize = sizeof( Float64 ); - property.mSelector = kAudioDevicePropertyNominalSampleRate; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Only change the sample rate if off by more than 1 Hz. - if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) { - - // Set a property listener for the sample rate change - Float64 reportedRate = 0.0; - AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; - result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - nominalRate = (Float64) sampleRate; - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); - if ( result != noErr ) { - AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Now wait until the reported nominal rate is what we just set. - UInt32 microCounter = 0; - while ( reportedRate != nominalRate ) { - microCounter += 5000; - if ( microCounter > 5000000 ) break; - usleep( 5000 ); - } - - // Remove the property listener. - AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); - - if ( microCounter > 5000000 ) { - errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Now set the stream format for all streams. Also, check the - // physical format of the device and change that if necessary. - AudioStreamBasicDescription description; - dataSize = sizeof( AudioStreamBasicDescription ); - property.mSelector = kAudioStreamPropertyVirtualFormat; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the sample rate and data format id. However, only make the - // change if the sample rate is not within 1.0 of the desired - // rate and the format is not linear pcm. - bool updateFormat = false; - if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { - description.mSampleRate = (Float64) sampleRate; - updateFormat = true; - } - - if ( description.mFormatID != kAudioFormatLinearPCM ) { - description.mFormatID = kAudioFormatLinearPCM; - updateFormat = true; - } - - if ( updateFormat ) { - result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Now check the physical format. - property.mSelector = kAudioStreamPropertyPhysicalFormat; - result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - //std::cout << "Current physical stream format:" << std::endl; - //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; - //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; - //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; - //std::cout << " sample rate = " << description.mSampleRate << std::endl; - - if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { - description.mFormatID = kAudioFormatLinearPCM; - //description.mSampleRate = (Float64) sampleRate; - AudioStreamBasicDescription testDescription = description; - UInt32 formatFlags; - - // We'll try higher bit rates first and then work our way down. - std::vector< std::pair > physicalFormats; - formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; - physicalFormats.push_back( std::pair( 32, formatFlags ) ); - formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; - physicalFormats.push_back( std::pair( 32, formatFlags ) ); - physicalFormats.push_back( std::pair( 24, formatFlags ) ); // 24-bit packed - formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); - physicalFormats.push_back( std::pair( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low - formatFlags |= kAudioFormatFlagIsAlignedHigh; - physicalFormats.push_back( std::pair( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high - formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; - physicalFormats.push_back( std::pair( 16, formatFlags ) ); - physicalFormats.push_back( std::pair( 8, formatFlags ) ); - - bool setPhysicalFormat = false; - for( unsigned int i=0; iflags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( streamCount == 1 ) { - if ( stream_.nUserChannels[mode] > 1 && - stream_.userInterleaved != stream_.deviceInterleaved[mode] ) - stream_.doConvertBuffer[mode] = true; - } - else if ( monoMode && stream_.userInterleaved ) - stream_.doConvertBuffer[mode] = true; - - // Allocate our CoreHandle structure for the stream. - CoreHandle *handle = 0; - if ( stream_.apiHandle == 0 ) { - try { - handle = new CoreHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; - goto error; - } - - if ( pthread_cond_init( &handle->condition, NULL ) ) { - errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - stream_.apiHandle = (void *) handle; - } - else - handle = (CoreHandle *) stream_.apiHandle; - handle->iStream[mode] = firstStream; - handle->nStreams[mode] = streamCount; - handle->id[mode] = id; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); - memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - // If possible, we will make use of the CoreAudio stream buffers as - // "device buffers". However, we can't do this if using multiple - // streams. - if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) { - if ( streamCount > 1 ) setConvertInfo( mode, 0 ); - else setConvertInfo( mode, channelOffset ); - } - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) - // Only one callback procedure per device. - stream_.mode = DUPLEX; - else { -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); -#else - // deprecated in favor of AudioDeviceCreateIOProcID() - result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); -#endif - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; - errorText_ = errorStream_.str(); - goto error; - } - if ( stream_.mode == OUTPUT && mode == INPUT ) - stream_.mode = DUPLEX; - else - stream_.mode = mode; - } - - // Setup the device property listener for over/underload. - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -void RtApiCore :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if (handle) { - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { - errorText_ = "RtApiCore::closeStream(): error removing property listener!"; - error( RtAudioError::WARNING ); - } - } - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[0], callbackHandler ); -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); -#else - // deprecated in favor of AudioDeviceDestroyIOProcID() - AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); -#endif - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - if (handle) { - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { - errorText_ = "RtApiCore::closeStream(): error removing property listener!"; - error( RtAudioError::WARNING ); - } - } - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[1], callbackHandler ); -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); -#else - // deprecated in favor of AudioDeviceDestroyIOProcID() - AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); -#endif - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - // Destroy pthread condition variable. - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiCore :: startStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiCore::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - OSStatus result = noErr; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - result = AudioDeviceStart( handle->id[0], callbackHandler ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || - ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - - result = AudioDeviceStart( handle->id[1], callbackHandler ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - - unlock: - if ( result == noErr ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiCore :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - OSStatus result = noErr; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled - } - - 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] ) ) { - - 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: - if ( result == noErr ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiCore :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - handle->drainCounter = 2; - - stopStream(); -} - -// This function will be called by a spawned thread when the user -// callback function signals that the stream should be stopped or -// aborted. It is better to handle it this way because the -// callbackEvent() function probably should return before the AudioDeviceStop() -// function is called. -static void *coreStopStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiCore *object = (RtApiCore *) info->object; - - object->stopStream(); - pthread_exit( NULL ); -} - -bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, - const AudioBufferList *inBufferList, - const AudioBufferList *outBufferList ) -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - ThreadHandle threadId; - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == true ) - pthread_create( &threadId, NULL, coreStopStream, info ); - else // external call to stopStream() - pthread_cond_signal( &handle->condition ); - return SUCCESS; - } - - AudioDeviceID outputDevice = handle->id[0]; - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream or duplex mode AND the input/output devices are - // different AND this function is called for the input device. - if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - abortStream(); - return SUCCESS; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - if ( handle->nStreams[0] == 1 ) { - memset( outBufferList->mBuffers[handle->iStream[0]].mData, - 0, - outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); - } - else { // fill multiple streams with zeros - for ( unsigned int i=0; inStreams[0]; i++ ) { - memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, - 0, - outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); - } - } - } - else if ( handle->nStreams[0] == 1 ) { - if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer - convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, - stream_.userBuffer[0], stream_.convertInfo[0] ); - } - else { // copy from user buffer - memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, - stream_.userBuffer[0], - outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); - } - } - else { // fill multiple streams - Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; - if ( stream_.doConvertBuffer[0] ) { - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - inBuffer = (Float32 *) stream_.deviceBuffer; - } - - if ( stream_.deviceInterleaved[0] == false ) { // mono mode - UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; - for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, - (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); - } - } - else { // fill multiple multi-channel streams with interleaved data - UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; - Float32 *out, *in; - - bool inInterleaved = ( stream_.userInterleaved ) ? true : false; - UInt32 inChannels = stream_.nUserChannels[0]; - if ( stream_.doConvertBuffer[0] ) { - inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode - inChannels = stream_.nDeviceChannels[0]; - } - - if ( inInterleaved ) inOffset = 1; - else inOffset = stream_.bufferSize; - - channelsLeft = inChannels; - for ( unsigned int i=0; inStreams[0]; i++ ) { - in = inBuffer; - out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; - streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; - - outJump = 0; - // Account for possible channel offset in first stream - if ( i == 0 && stream_.channelOffset[0] > 0 ) { - streamChannels -= stream_.channelOffset[0]; - outJump = stream_.channelOffset[0]; - out += outJump; - } - - // Account for possible unfilled channels at end of the last stream - if ( streamChannels > channelsLeft ) { - outJump = streamChannels - channelsLeft; - streamChannels = channelsLeft; - } - - // Determine input buffer offsets and skips - if ( inInterleaved ) { - inJump = inChannels; - in += inChannels - channelsLeft; - } - else { - inJump = 1; - in += (inChannels - channelsLeft) * inOffset; - } - - for ( unsigned int i=0; idrainCounter ) { - handle->drainCounter++; - goto unlock; - } - - AudioDeviceID inputDevice; - inputDevice = handle->id[1]; - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { - - if ( handle->nStreams[1] == 1 ) { - if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer - convertBuffer( stream_.userBuffer[1], - (char *) inBufferList->mBuffers[handle->iStream[1]].mData, - stream_.convertInfo[1] ); - } - else { // copy to user buffer - memcpy( stream_.userBuffer[1], - inBufferList->mBuffers[handle->iStream[1]].mData, - inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); - } - } - else { // read from multiple streams - Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; - if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; - - if ( stream_.deviceInterleaved[1] == false ) { // mono mode - UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; - for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); - } - } - else { // read from multiple multi-channel streams - UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; - Float32 *out, *in; - - bool outInterleaved = ( stream_.userInterleaved ) ? true : false; - UInt32 outChannels = stream_.nUserChannels[1]; - if ( stream_.doConvertBuffer[1] ) { - outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode - outChannels = stream_.nDeviceChannels[1]; - } - - if ( outInterleaved ) outOffset = 1; - else outOffset = stream_.bufferSize; - - channelsLeft = outChannels; - for ( unsigned int i=0; inStreams[1]; i++ ) { - out = outBuffer; - in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; - streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; - - inJump = 0; - // Account for possible channel offset in first stream - if ( i == 0 && stream_.channelOffset[1] > 0 ) { - streamChannels -= stream_.channelOffset[1]; - inJump = stream_.channelOffset[1]; - in += inJump; - } - - // Account for possible unread channels at end of the last stream - if ( streamChannels > channelsLeft ) { - inJump = streamChannels - channelsLeft; - streamChannels = channelsLeft; - } - - // Determine output buffer offsets and skips - if ( outInterleaved ) { - outJump = outChannels; - out += outChannels - channelsLeft; - } - else { - outJump = 1; - out += (outChannels - channelsLeft) * outOffset; - } - - for ( unsigned int i=0; i -#include -#include - -// A structure to hold various information related to the Jack API -// implementation. -struct JackHandle { - jack_client_t *client; - jack_port_t **ports[2]; - std::string deviceName[2]; - bool xrun[2]; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - JackHandle() - :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -static void jackSilentError( const char * ) {}; - -RtApiJack :: RtApiJack() - :shouldAutoconnect_(true) { - // Nothing to do here. -#if !defined(__RTAUDIO_DEBUG__) - // Turn off Jack's internal error reporting. - jack_set_error_function( &jackSilentError ); -#endif -} - -RtApiJack :: ~RtApiJack() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiJack :: getDeviceCount( void ) -{ - // See if we can become a jack client. - jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption; - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); - if ( client == 0 ) return 0; - - const char **ports; - std::string port, previousPort; - unsigned int nChannels = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nChannels ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon + 1 ); - if ( port != previousPort ) { - nDevices++; - previousPort = port; - } - } - } while ( ports[++nChannels] ); - free( ports ); - } - - jack_client_close( client ); - return nDevices; -} - -RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; - error( RtAudioError::WARNING ); - return info; - } - - const char **ports; - std::string port, previousPort; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) info.name = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } - - if ( device >= nDevices ) { - jack_client_close( client ); - errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - // Get the current jack server sample rate. - info.sampleRates.clear(); - - info.preferredSampleRate = jack_get_sample_rate( client ); - info.sampleRates.push_back( info.preferredSampleRate ); - - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.outputChannels = nChannels; - } - - // Jack "output ports" equal RtAudio input channels. - nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.inputChannels = nChannels; - } - - if ( info.outputChannels == 0 && info.inputChannels == 0 ) { - jack_client_close(client); - errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; - error( RtAudioError::WARNING ); - return info; - } - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Jack always uses 32-bit floats. - info.nativeFormats = RTAUDIO_FLOAT32; - - // Jack doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; - - jack_client_close(client); - info.probed = true; - return info; -} - -static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - - RtApiJack *object = (RtApiJack *) info->object; - if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; - - return 0; -} - -// This function will be called by a spawned thread when the Jack -// server signals that it is shutting down. It is necessary to handle -// it this way because the jackShutdown() function must return before -// the jack_deactivate() function (in closeStream()) will return. -static void *jackCloseStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiJack *object = (RtApiJack *) info->object; - - object->closeStream(); - - pthread_exit( NULL ); -} -static void jackShutdown( void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - RtApiJack *object = (RtApiJack *) info->object; - - // Check current stream state. If stopped, then we'll assume this - // was called as a result of a call to RtApiJack::stopStream (the - // deactivation of a client handle causes this function to be called). - // If not, we'll assume the Jack server is shutting down or some - // other problem occurred and we should close the stream. - if ( object->isStreamRunning() == false ) return; - - ThreadHandle threadId; - pthread_create( &threadId, NULL, jackCloseStream, info ); - std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; -} - -static int jackXrun( void *infoPointer ) -{ - JackHandle *handle = (JackHandle *) infoPointer; - - if ( handle->ports[0] ) handle->xrun[0] = true; - if ( handle->ports[1] ) handle->xrun[1] = true; - - return 0; -} - -bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - JackHandle *handle = (JackHandle *) stream_.apiHandle; - - // Look for jack server and try to become a client (only do once per stream). - jack_client_t *client = 0; - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { - jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption; - jack_status_t *status = NULL; - if ( options && !options->streamName.empty() ) - client = jack_client_open( options->streamName.c_str(), jackoptions, status ); - else - client = jack_client_open( "RtApiJack", jackoptions, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - } - else { - // The handle must have been created on an earlier pass. - client = handle->client; - } - - const char **ports; - std::string port, previousPort, deviceName; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) deviceName = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - unsigned long flag = JackPortIsInput; - if ( mode == INPUT ) flag = JackPortIsOutput; - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - } - - // Compare the jack ports for specified client to the requested number of channels. - if ( nChannels < (channels + firstChannel) ) { - errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check the jack server sample rate. - unsigned int jackRate = jack_get_sample_rate( client ); - if ( sampleRate != jackRate ) { - jack_client_close( client ); - errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = jackRate; - - // Get the latency of the JACK port. - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports[ firstChannel ] ) { - // Added by Ge Wang - jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); - // the range (usually the min and max are equal) - jack_latency_range_t latrange; latrange.min = latrange.max = 0; - // get the latency range - jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); - // be optimistic, use the min! - stream_.latency[mode] = latrange.min; - //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); - } - free( ports ); - - // The jack server always uses 32-bit floating-point data. - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - stream_.userFormat = format; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // Jack always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; - - // Jack always provides host byte-ordered data. - stream_.doByteSwap[mode] = false; - - // Get the buffer size. The buffer size and number of buffers - // (periods) is set when the jack server is started. - stream_.bufferSize = (int) jack_get_buffer_size( client ); - *bufferSize = stream_.bufferSize; - - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate our JackHandle structure for the stream. - if ( handle == 0 ) { - try { - handle = new JackHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; - goto error; - } - - if ( pthread_cond_init(&handle->condition, NULL) ) { - errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - stream_.apiHandle = (void *) handle; - handle->client = client; - } - handle->deviceName[mode] = deviceName; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - if ( mode == OUTPUT ) - bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - else { // mode == INPUT - bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); - if ( bufferBytes < bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Allocate memory for the Jack ports (channels) identifiers. - handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); - if ( handle->ports[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; - goto error; - } - - stream_.device[mode] = device; - stream_.channelOffset[mode] = firstChannel; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; - - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up the stream for output. - stream_.mode = DUPLEX; - else { - stream_.mode = mode; - jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); - jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); - jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); - } - - // Register our ports. - char label[64]; - if ( mode == OUTPUT ) { - for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); - } - } - else { - for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); - } - } - - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - - if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false; - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - jack_client_close( handle->client ); - - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiJack :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiJack::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( handle ) { - - if ( stream_.state == STREAM_RUNNING ) - jack_deactivate( handle->client ); - - jack_client_close( handle->client ); - } - - if ( handle ) { - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiJack :: startStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiJack::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - int result = jack_activate( handle->client ); - if ( result ) { - errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; - goto unlock; - } - - const char **ports; - - // Get the list of available ports. - if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; - goto unlock; - } - - // Now make the port connections. Since RtAudio wasn't designed to - // allow the user to select particular channels of a device, we'll - // just open the first "nChannels" ports with offset. - for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting output ports!"; - goto unlock; - } - } - free(ports); - } - - if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; - goto unlock; - } - - // Now make the port connections. See note above. - for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting input ports!"; - goto unlock; - } - } - free(ports); - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - - unlock: - if ( result == 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiJack :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled - } - } - - jack_deactivate( handle->client ); - stream_.state = STREAM_STOPPED; -} - -void RtApiJack :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - handle->drainCounter = 2; - - stopStream(); -} - -// This function will be called by a spawned thread when the user -// callback function signals that the stream should be stopped or -// aborted. It is necessary to handle it this way because the -// callbackEvent() function must return before the jack_deactivate() -// function will return. -static void *jackStopStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiJack *object = (RtApiJack *) info->object; - - object->stopStream(); - pthread_exit( NULL ); -} - -bool RtApiJack :: callbackEvent( unsigned long nframes ) -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - if ( stream_.bufferSize != nframes ) { - errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - JackHandle *handle = (JackHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - ThreadHandle threadId; - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == true ) - pthread_create( &threadId, NULL, jackStopStream, info ); - else - pthread_cond_signal( &handle->condition ); - return SUCCESS; - } - - // Invoke user callback first, to get fresh output data. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - ThreadHandle id; - pthread_create( &id, NULL, jackStopStream, info ); - return SUCCESS; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - jack_default_audio_sample_t *jackbuffer; - unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memset( jackbuffer, 0, bufferBytes ); - } - - } - else if ( stream_.doConvertBuffer[0] ) { - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); - } - } - else { // no buffer conversion - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); - } - } - } - - // Don't bother draining input - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - if ( stream_.doConvertBuffer[1] ) { - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); - } - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } - else { // no buffer conversion - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); - } - } - } - - unlock: - RtApi::tickStreamTime(); - return SUCCESS; -} - //******************** End of __UNIX_JACK__ *********************// -#endif - -#if defined(__WINDOWS_ASIO__) // ASIO API on Windows - -// The ASIO API is designed around a callback scheme, so this -// implementation is similar to that used for OS-X CoreAudio and Linux -// Jack. The primary constraint with ASIO is that it only allows -// access to a single driver at a time. Thus, it is not possible to -// have more than one simultaneous RtAudio stream. -// -// This implementation also requires a number of external ASIO files -// and a few global variables. The ASIO callback scheme does not -// allow for the passing of user data, so we must create a global -// pointer to our callbackInfo structure. -// -// On unix systems, we make use of a pthread condition variable. -// Since there is no equivalent in Windows, I hacked something based -// on information found in -// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. - -#include "asiosys.h" -#include "asio.h" -#include "iasiothiscallresolver.h" -#include "asiodrivers.h" -#include - -static AsioDrivers drivers; -static ASIOCallbacks asioCallbacks; -static ASIODriverInfo driverInfo; -static CallbackInfo *asioCallbackInfo; -static bool asioXRun; - -struct AsioHandle { - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - ASIOBufferInfo *bufferInfos; - HANDLE condition; - - AsioHandle() - :drainCounter(0), internalDrain(false), bufferInfos(0) {} -}; - -// Function declarations (definitions at end of section) -static const char* getAsioErrorString( ASIOError result ); -static void sampleRateChanged( ASIOSampleRate sRate ); -static long asioMessages( long selector, long value, void* message, double* opt ); - -RtApiAsio :: RtApiAsio() -{ - // ASIO cannot run on a multi-threaded appartment. You can call - // CoInitialize beforehand, but it must be for appartment threading - // (in which case, CoInitilialize will return S_FALSE here). - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( FAILED(hr) ) { - errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; - error( RtAudioError::WARNING ); - } - coInitialized_ = true; - - drivers.removeCurrentDriver(); - driverInfo.asioVersion = 2; - - // See note in DirectSound implementation about GetDesktopWindow(). - driverInfo.sysRef = GetForegroundWindow(); -} - -RtApiAsio :: ~RtApiAsio() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); - if ( coInitialized_ ) CoUninitialize(); -} - -unsigned int RtApiAsio :: getDeviceCount( void ) -{ - return (unsigned int) drivers.asioGetNumDev(); -} - -RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - // If a stream is already open, we cannot probe other devices. Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; - error( RtAudioError::WARNING ); - return info; - } - return devices_[ device ]; - } - - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - info.name = driverName; - - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Determine the device channel information. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - info.outputChannels = outputChannels; - info.inputChannels = inputChannels; - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Determine the supported sample rates. - info.sampleRates.clear(); - for ( unsigned int i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[i]; - } - } - - // 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( RtAudioError::WARNING ); - return info; - } - - 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; - else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) - info.nativeFormats |= RTAUDIO_SINT24; - - if ( info.outputChannels > 0 ) - if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; - if ( info.inputChannels > 0 ) - if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; - - info.probed = true; - drivers.removeCurrentDriver(); - return info; -} - -static void bufferSwitch( long index, ASIOBool /*processNow*/ ) -{ - RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; - object->callbackEvent( index ); -} - -void RtApiAsio :: saveDeviceInfo( void ) -{ - devices_.clear(); - - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; isaveDeviceInfo(); - - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // keep them before any "goto error", they are used for error cleanup + goto device boundary checks - bool buffersAllocated = false; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - unsigned int nChannels; - - - // Check the device channel count. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || - ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; - errorText_ = errorStream_.str(); - goto error; - } - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = firstChannel; - - // Verify the sample rate is supported. - result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - // Get the current sample rate - ASIOSampleRate currentRate; - result = ASIOGetSampleRate( ¤tRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; - errorText_ = errorStream_.str(); - goto error; - } - - // Set the sample rate only if necessary - if ( currentRate != sampleRate ) { - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - 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 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; - errorText_ = errorStream_.str(); - goto error; - } - - // 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; - } - else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; - } - - if ( stream_.deviceFormat[mode] == 0 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - 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 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; - errorText_ = errorStream_.str(); - goto error; - } - - if ( isDuplexInput ) { - // When this is the duplex input (output was opened before), then we have to use the same - // buffersize as the output, because it might use the preferred buffer size, which most - // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, - // So instead of throwing an error, make them equal. The caller uses the reference - // to the "bufferSize" param as usual to set up processing buffers. - - *bufferSize = stream_.bufferSize; - - } else { - if ( *bufferSize == 0 ) *bufferSize = preferSize; - else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else if ( granularity == -1 ) { - // Make sure bufferSize is a power of two. - int log2_of_min_size = 0; - int log2_of_max_size = 0; - - for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { - if ( minSize & ((long)1 << i) ) log2_of_min_size = i; - if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; - } - - long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); - int min_delta_num = log2_of_min_size; - - for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { - long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); - if (current_delta < min_delta) { - min_delta = current_delta; - min_delta_num = i; - } - } - - *bufferSize = ( (unsigned int)1 << min_delta_num ); - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - } - else if ( granularity != 0 ) { - // Set to an even multiple of granularity, rounding up. - *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; - } - } - - /* - // we don't use it anymore, see above! - // Just left it here for the case... - if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { - errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; - goto error; - } - */ - - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 2; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // ASIO always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; - - // Allocate, if necessary, our AsioHandle structure for the stream. - if ( handle == 0 ) { - try { - handle = new AsioHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; - } - handle->bufferInfos = 0; - - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - - // Create the ASIO internal buffers. Since RtAudio sets up input - // and output separately, we'll have to dispose of previously - // created output buffers for a duplex stream. - if ( mode == INPUT && stream_.mode == OUTPUT ) { - ASIODisposeBuffers(); - if ( handle->bufferInfos ) free( handle->bufferInfos ); - } - - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - unsigned int i; - nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); - if ( handle->bufferInfos == NULL ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - ASIOBufferInfo *infos; - infos = handle->bufferInfos; - for ( i=0; iisInput = ASIOFalse; - infos->channelNum = i + stream_.channelOffset[0]; - infos->buffers[0] = infos->buffers[1] = 0; - } - for ( i=0; iisInput = ASIOTrue; - infos->channelNum = i + stream_.channelOffset[1]; - infos->buffers[0] = infos->buffers[1] = 0; - } - - // prepare for callbacks - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.mode = isDuplexInput ? DUPLEX : mode; - - // store this class instance before registering callbacks, that are going to use it - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - - // Set up the ASIO callback structure and create the ASIO data buffers. - asioCallbacks.bufferSwitch = &bufferSwitch; - asioCallbacks.sampleRateDidChange = &sampleRateChanged; - asioCallbacks.asioMessage = &asioMessages; - asioCallbacks.bufferSwitchTimeInfo = NULL; - result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); - if ( result != ASE_OK ) { - // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges - // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver - // in that case, let's be naïve and try that instead - *bufferSize = preferSize; - stream_.bufferSize = *bufferSize; - result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); - } - - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; - errorText_ = errorStream_.str(); - goto error; - } - buffersAllocated = true; - stream_.state = STREAM_STOPPED; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( isDuplexInput && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Determine device latencies - long inputLatency, outputLatency; - result = ASIOGetLatencies( &inputLatency, &outputLatency ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; - errorText_ = errorStream_.str(); - error( RtAudioError::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 ( !isDuplexInput ) { - // the cleanup for error in the duplex input, is done by RtApi::openStream - // So we clean up for single channel only - - if ( buffersAllocated ) - ASIODisposeBuffers(); - - drivers.removeCurrentDriver(); - - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - - delete handle; - stream_.apiHandle = 0; - } - - - if ( stream_.userBuffer[mode] ) { - free( stream_.userBuffer[mode] ); - stream_.userBuffer[mode] = 0; - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - } - - return FAILURE; -}//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// - -void RtApiAsio :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - ASIOStop(); - } - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -bool stopThreadCalled = false; - -void RtApiAsio :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAsio::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - ASIOError result = ASIOStart(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; - errorText_ = errorStream_.str(); - goto unlock; - } - - handle->drainCounter = 0; - handle->internalDrain = false; - ResetEvent( handle->condition ); - stream_.state = STREAM_RUNNING; - asioXRun = false; - - unlock: - stopThreadCalled = false; - - if ( result == ASE_OK ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAsio :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - } - } - - stream_.state = STREAM_STOPPED; - - ASIOError result = ASIOStop(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; - errorText_ = errorStream_.str(); - } - - if ( result == ASE_OK ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAsio :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - // The following lines were commented-out because some behavior was - // noted where the device buffers need to be zeroed to avoid - // continuing sound, even when the device buffers are completely - // disposed. So now, calling abort is the same as calling stop. - // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - // handle->drainCounter = 2; - stopStream(); -} - -// This function will be called by a spawned thread when the user -// callback function signals that the stream should be stopped or -// aborted. It is necessary to handle it this way because the -// callbackEvent() function must return before the ASIOStop() -// function will return. -static unsigned __stdcall asioStopStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAsio *object = (RtApiAsio *) info->object; - - object->stopStream(); - _endthreadex( 0 ); - return 0; -} - -bool RtApiAsio :: callbackEvent( long bufferIndex ) -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal if finished. - if ( handle->drainCounter > 3 ) { - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else { // spawn a thread to stop the stream - unsigned threadId; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, - &stream_.callbackInfo, 0, &threadId ); - } - return SUCCESS; - } - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && asioXRun == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - asioXRun = false; - } - if ( stream_.mode != OUTPUT && asioXRun == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - asioXRun = false; - } - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - unsigned threadId; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, - &stream_.callbackInfo, 0, &threadId ); - return SUCCESS; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - unsigned int nChannels, bufferBytes, i, j; - nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); - } - - } - else if ( stream_.doConvertBuffer[0] ) { - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[0], - stream_.deviceFormat[0] ); - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); - } - - } - else { - - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.userBuffer[0], - stream_.bufferSize * stream_.nUserChannels[0], - stream_.userFormat ); - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); - } - - } - } - - // Don't bother draining input - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); - - if (stream_.doConvertBuffer[1]) { - - // Always interleave ASIO input data. - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) - memcpy( &stream_.deviceBuffer[j++*bufferBytes], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } - - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[1], - stream_.deviceFormat[1] ); - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - } - else { - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { - memcpy( &stream_.userBuffer[1][bufferBytes*j++], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } - } - - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.userBuffer[1], - stream_.bufferSize * stream_.nUserChannels[1], - stream_.userFormat ); - } - } - - unlock: - // The following call was suggested by Malte Clasen. While the API - // documentation indicates it should not be required, some device - // drivers apparently do not function correctly without it. - ASIOOutputReady(); - - RtApi::tickStreamTime(); - return SUCCESS; -} - -static void sampleRateChanged( ASIOSampleRate sRate ) -{ - // The ASIO documentation says that this usually only happens during - // external sync. Audio processing is not stopped by the driver, - // actual sample rate might not have even changed, maybe only the - // sample rate status of an AES/EBU or S/PDIF digital input at the - // audio device. - - RtApi *object = (RtApi *) asioCallbackInfo->object; - try { - object->stopStream(); - } - catch ( RtAudioError &exception ) { - std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; - return; - } - - std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; -} - -static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) -{ - long ret = 0; - - switch( selector ) { - case kAsioSelectorSupported: - if ( value == kAsioResetRequest - || value == kAsioEngineVersion - || value == kAsioResyncRequest - || value == kAsioLatenciesChanged - // The following three were added for ASIO 2.0, you don't - // necessarily have to support them. - || value == kAsioSupportsTimeInfo - || value == kAsioSupportsTimeCode - || value == kAsioSupportsInputMonitor) - ret = 1L; - break; - case kAsioResetRequest: - // Defer the task and perform the reset of the driver during the - // next "safe" situation. You cannot reset the driver right now, - // as this code is called from the driver. Reset the driver is - // done by completely destruct is. I.e. ASIOStop(), - // ASIODisposeBuffers(), Destruction Afterwards you initialize the - // driver again. - std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; - ret = 1L; - break; - case kAsioResyncRequest: - // This informs the application that the driver encountered some - // non-fatal data loss. It is used for synchronization purposes - // of different media. Added mainly to work around the Win16Mutex - // problems in Windows 95/98 with the Windows Multimedia system, - // which could lose data because the Mutex was held too long by - // another thread. However a driver can issue it in other - // situations, too. - // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; - asioXRun = true; - ret = 1L; - break; - case kAsioLatenciesChanged: - // This will inform the host application that the drivers were - // latencies changed. Beware, it this does not mean that the - // buffer sizes have changed! You might need to update internal - // delay data. - std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; - ret = 1L; - break; - case kAsioEngineVersion: - // Return the supported ASIO version of the host application. If - // a host application does not implement this selector, ASIO 1.0 - // is assumed by the driver. - ret = 2L; - break; - case kAsioSupportsTimeInfo: - // Informs the driver whether the - // asioCallbacks.bufferSwitchTimeInfo() callback is supported. - // For compatibility with ASIO 1.0 drivers the host application - // should always support the "old" bufferSwitch method, too. - ret = 0; - break; - case kAsioSupportsTimeCode: - // Informs the driver whether application is interested in time - // code info. If an application does not need to know about time - // code, the driver has less work to do. - ret = 0; - break; - } - return ret; -} - -static const char* getAsioErrorString( ASIOError result ) -{ - struct Messages - { - ASIOError value; - const char*message; - }; - - static const Messages m[] = - { - { ASE_NotPresent, "Hardware input or output is not present or available." }, - { ASE_HWMalfunction, "Hardware is malfunctioning." }, - { ASE_InvalidParameter, "Invalid input parameter." }, - { ASE_InvalidMode, "Invalid mode." }, - { ASE_SPNotAdvancing, "Sample position not advancing." }, - { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, - { ASE_NoMemory, "Not enough memory to complete the request." } - }; - - for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) - if ( m[i].value == result ) return m[i].message; - - return "Unknown error."; -} - -//******************** End of __WINDOWS_ASIO__ *********************// -#endif - - -#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API - -// Authored by Marcus Tomlinson , April 2014 -// - Introduces support for the Windows WASAPI API -// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required -// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface -// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user - -#ifndef INITGUID - #define INITGUID -#endif -#include -#include -#include -#include - -//============================================================================= - -#define SAFE_RELEASE( objectPtr )\ -if ( objectPtr )\ -{\ - objectPtr->Release();\ - objectPtr = NULL;\ -} - -typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); - -//----------------------------------------------------------------------------- - -// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size. -// Therefore we must perform all necessary conversions to user buffers in order to satisfy these -// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to -// provide intermediate storage for read / write synchronization. -class WasapiBuffer -{ -public: - WasapiBuffer() - : buffer_( NULL ), - bufferSize_( 0 ), - inIndex_( 0 ), - outIndex_( 0 ) {} - - ~WasapiBuffer() { - free( buffer_ ); - } - - // sets the length of the internal ring buffer - void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { - free( buffer_ ); - - buffer_ = ( char* ) calloc( bufferSize, formatBytes ); - - bufferSize_ = bufferSize; - inIndex_ = 0; - outIndex_ = 0; - } - - // attempt to push a buffer into the ring buffer at the current "in" index - bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) - { - if ( !buffer || // incoming buffer is NULL - bufferSize == 0 || // incoming buffer has no data - bufferSize > bufferSize_ ) // incoming buffer too large - { - return false; - } - - unsigned int relOutIndex = outIndex_; - unsigned int inIndexEnd = inIndex_ + bufferSize; - if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { - relOutIndex += bufferSize_; - } - - // "in" index can end on the "out" index but cannot begin at it - if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) { - return false; // not enough space between "in" index and "out" index - } - - // copy buffer from external to internal - int fromZeroSize = inIndex_ + bufferSize - bufferSize_; - fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; - int fromInSize = bufferSize - fromZeroSize; - - switch( format ) - { - case RTAUDIO_SINT8: - memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) ); - memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) ); - break; - case RTAUDIO_SINT16: - memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) ); - memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) ); - break; - case RTAUDIO_SINT24: - memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) ); - memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) ); - break; - case RTAUDIO_SINT32: - memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) ); - memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) ); - break; - case RTAUDIO_FLOAT32: - memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) ); - memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) ); - break; - case RTAUDIO_FLOAT64: - memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) ); - memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) ); - break; - } - - // update "in" index - inIndex_ += bufferSize; - inIndex_ %= bufferSize_; - - return true; - } - - // attempt to pull a buffer from the ring buffer from the current "out" index - bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) - { - if ( !buffer || // incoming buffer is NULL - bufferSize == 0 || // incoming buffer has no data - bufferSize > bufferSize_ ) // incoming buffer too large - { - return false; - } - - unsigned int relInIndex = inIndex_; - unsigned int outIndexEnd = outIndex_ + bufferSize; - if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) { - relInIndex += bufferSize_; - } - - // "out" index can begin at and end on the "in" index - if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) { - return false; // not enough space between "out" index and "in" index - } - - // copy buffer from internal to external - int fromZeroSize = outIndex_ + bufferSize - bufferSize_; - fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; - int fromOutSize = bufferSize - fromZeroSize; - - switch( format ) - { - case RTAUDIO_SINT8: - memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) ); - memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) ); - break; - case RTAUDIO_SINT16: - memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) ); - memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) ); - break; - case RTAUDIO_SINT24: - memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) ); - memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) ); - break; - case RTAUDIO_SINT32: - memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) ); - memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) ); - break; - case RTAUDIO_FLOAT32: - memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) ); - memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) ); - break; - case RTAUDIO_FLOAT64: - memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) ); - memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) ); - break; - } - - // update "out" index - outIndex_ += bufferSize; - outIndex_ %= bufferSize_; - - return true; - } - -private: - char* buffer_; - unsigned int bufferSize_; - unsigned int inIndex_; - unsigned int outIndex_; -}; - -//----------------------------------------------------------------------------- - -// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate -// between HW and the user. The convertBufferWasapi function is used to perform this conversion -// between HwIn->UserIn and UserOut->HwOut during the stream callback loop. -// This sample rate converter works best with conversions between one rate and its multiple. -void convertBufferWasapi( char* outBuffer, - const char* inBuffer, - const unsigned int& channelCount, - const unsigned int& inSampleRate, - const unsigned int& outSampleRate, - const unsigned int& inSampleCount, - unsigned int& outSampleCount, - const RtAudioFormat& format ) -{ - // calculate the new outSampleCount and relative sampleStep - float sampleRatio = ( float ) outSampleRate / inSampleRate; - float sampleRatioInv = ( float ) 1 / sampleRatio; - float sampleStep = 1.0f / sampleRatio; - float inSampleFraction = 0.0f; - - outSampleCount = ( unsigned int ) std::roundf( inSampleCount * sampleRatio ); - - // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate - if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv ) - { - // frame-by-frame, copy each relative input sample into it's corresponding output sample - for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) - { - unsigned int inSample = ( unsigned int ) inSampleFraction; - - switch ( format ) - { - case RTAUDIO_SINT8: - memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) ); - break; - case RTAUDIO_SINT16: - memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) ); - break; - case RTAUDIO_SINT24: - memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) ); - break; - case RTAUDIO_SINT32: - memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) ); - break; - case RTAUDIO_FLOAT32: - memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) ); - break; - case RTAUDIO_FLOAT64: - memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) ); - break; - } - - // jump to next in sample - inSampleFraction += sampleStep; - } - } - else // else interpolate - { - // frame-by-frame, copy each relative input sample into it's corresponding output sample - for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) - { - unsigned int inSample = ( unsigned int ) inSampleFraction; - float inSampleDec = inSampleFraction - inSample; - unsigned int frameInSample = inSample * channelCount; - unsigned int frameOutSample = outSample * channelCount; - - switch ( format ) - { - case RTAUDIO_SINT8: - { - for ( unsigned int channel = 0; channel < channelCount; channel++ ) - { - char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ]; - char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ]; - char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec ); - ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; - } - break; - } - case RTAUDIO_SINT16: - { - for ( unsigned int channel = 0; channel < channelCount; channel++ ) - { - short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ]; - short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ]; - short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec ); - ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; - } - break; - } - case RTAUDIO_SINT24: - { - for ( unsigned int channel = 0; channel < channelCount; channel++ ) - { - int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt(); - int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt(); - int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec ); - ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; - } - break; - } - case RTAUDIO_SINT32: - { - for ( unsigned int channel = 0; channel < channelCount; channel++ ) - { - int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ]; - int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ]; - int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec ); - ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; - } - break; - } - case RTAUDIO_FLOAT32: - { - for ( unsigned int channel = 0; channel < channelCount; channel++ ) - { - float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ]; - float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ]; - float sampleDiff = ( toSample - fromSample ) * inSampleDec; - ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; - } - break; - } - case RTAUDIO_FLOAT64: - { - for ( unsigned int channel = 0; channel < channelCount; channel++ ) - { - double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ]; - double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ]; - double sampleDiff = ( toSample - fromSample ) * inSampleDec; - ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; - } - break; - } - } - - // jump to next in sample - inSampleFraction += sampleStep; - } - } -} - -//----------------------------------------------------------------------------- - -// A structure to hold various information related to the WASAPI implementation. -struct WasapiHandle -{ - IAudioClient* captureAudioClient; - IAudioClient* renderAudioClient; - IAudioCaptureClient* captureClient; - IAudioRenderClient* renderClient; - HANDLE captureEvent; - HANDLE renderEvent; - - WasapiHandle() - : captureAudioClient( NULL ), - renderAudioClient( NULL ), - captureClient( NULL ), - renderClient( NULL ), - captureEvent( NULL ), - renderEvent( NULL ) {} -}; - -//============================================================================= - -RtApiWasapi::RtApiWasapi() - : coInitialized_( false ), deviceEnumerator_( NULL ) -{ - // WASAPI can run either apartment or multi-threaded - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) - coInitialized_ = true; - - // Instantiate device enumerator - hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL, - CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ), - ( void** ) &deviceEnumerator_ ); - - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator"; - error( RtAudioError::DRIVER_ERROR ); - } -} - -//----------------------------------------------------------------------------- - -RtApiWasapi::~RtApiWasapi() -{ - if ( stream_.state != STREAM_CLOSED ) - closeStream(); - - SAFE_RELEASE( deviceEnumerator_ ); - - // If this object previously called CoInitialize() - if ( coInitialized_ ) - CoUninitialize(); -} - -//============================================================================= - -unsigned int RtApiWasapi::getDeviceCount( void ) -{ - unsigned int captureDeviceCount = 0; - unsigned int renderDeviceCount = 0; - - IMMDeviceCollection* captureDevices = NULL; - IMMDeviceCollection* renderDevices = NULL; - - // Count capture devices - errorText_.clear(); - HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; - goto Exit; - } - - hr = captureDevices->GetCount( &captureDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; - goto Exit; - } - - // Count render devices - hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; - goto Exit; - } - - hr = renderDevices->GetCount( &renderDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; - goto Exit; - } - -Exit: - // release all references - SAFE_RELEASE( captureDevices ); - SAFE_RELEASE( renderDevices ); - - if ( errorText_.empty() ) - return captureDeviceCount + renderDeviceCount; - - error( RtAudioError::DRIVER_ERROR ); - return 0; -} - -//----------------------------------------------------------------------------- - -RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - unsigned int captureDeviceCount = 0; - unsigned int renderDeviceCount = 0; - std::string defaultDeviceName; - bool isCaptureDevice = false; - - PROPVARIANT deviceNameProp; - PROPVARIANT defaultDeviceNameProp; - - IMMDeviceCollection* captureDevices = NULL; - IMMDeviceCollection* renderDevices = NULL; - IMMDevice* devicePtr = NULL; - IMMDevice* defaultDevicePtr = NULL; - IAudioClient* audioClient = NULL; - IPropertyStore* devicePropStore = NULL; - IPropertyStore* defaultDevicePropStore = NULL; - - WAVEFORMATEX* deviceFormat = NULL; - WAVEFORMATEX* closestMatchFormat = NULL; - - // probed - info.probed = false; - - // Count capture devices - errorText_.clear(); - RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; - HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; - goto Exit; - } - - hr = captureDevices->GetCount( &captureDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; - goto Exit; - } - - // Count render devices - hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; - goto Exit; - } - - hr = renderDevices->GetCount( &renderDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; - goto Exit; - } - - // validate device index - if ( device >= captureDeviceCount + renderDeviceCount ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; - errorType = RtAudioError::INVALID_USE; - goto Exit; - } - - // determine whether index falls within capture or render devices - if ( device >= renderDeviceCount ) { - hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; - goto Exit; - } - isCaptureDevice = true; - } - else { - hr = renderDevices->Item( device, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; - goto Exit; - } - isCaptureDevice = false; - } - - // get default device name - if ( isCaptureDevice ) { - hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; - goto Exit; - } - } - else { - hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; - goto Exit; - } - } - - hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; - goto Exit; - } - PropVariantInit( &defaultDeviceNameProp ); - - hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; - goto Exit; - } - - defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); - - // name - hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; - goto Exit; - } - - PropVariantInit( &deviceNameProp ); - - hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; - goto Exit; - } - - info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); - - // is default - if ( isCaptureDevice ) { - info.isDefaultInput = info.name == defaultDeviceName; - info.isDefaultOutput = false; - } - else { - info.isDefaultInput = false; - info.isDefaultOutput = info.name == defaultDeviceName; - } - - // channel count - hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; - goto Exit; - } - - hr = audioClient->GetMixFormat( &deviceFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; - goto Exit; - } - - if ( isCaptureDevice ) { - info.inputChannels = deviceFormat->nChannels; - info.outputChannels = 0; - info.duplexChannels = 0; - } - else { - info.inputChannels = 0; - info.outputChannels = deviceFormat->nChannels; - info.duplexChannels = 0; - } - - // sample rates - info.sampleRates.clear(); - - // allow support for all sample rates as we have a built-in sample rate converter - for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { - info.sampleRates.push_back( SAMPLE_RATES[i] ); - } - info.preferredSampleRate = deviceFormat->nSamplesPerSec; - - // native format - info.nativeFormats = 0; - - if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT || - ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && - ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) ) - { - if ( deviceFormat->wBitsPerSample == 32 ) { - info.nativeFormats |= RTAUDIO_FLOAT32; - } - else if ( deviceFormat->wBitsPerSample == 64 ) { - info.nativeFormats |= RTAUDIO_FLOAT64; - } - } - else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM || - ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && - ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) ) - { - if ( deviceFormat->wBitsPerSample == 8 ) { - info.nativeFormats |= RTAUDIO_SINT8; - } - else if ( deviceFormat->wBitsPerSample == 16 ) { - info.nativeFormats |= RTAUDIO_SINT16; - } - else if ( deviceFormat->wBitsPerSample == 24 ) { - info.nativeFormats |= RTAUDIO_SINT24; - } - else if ( deviceFormat->wBitsPerSample == 32 ) { - info.nativeFormats |= RTAUDIO_SINT32; - } - } - - // probed - info.probed = true; - -Exit: - // release all references - PropVariantClear( &deviceNameProp ); - PropVariantClear( &defaultDeviceNameProp ); - - SAFE_RELEASE( captureDevices ); - SAFE_RELEASE( renderDevices ); - SAFE_RELEASE( devicePtr ); - SAFE_RELEASE( defaultDevicePtr ); - SAFE_RELEASE( audioClient ); - SAFE_RELEASE( devicePropStore ); - SAFE_RELEASE( defaultDevicePropStore ); - - CoTaskMemFree( deviceFormat ); - CoTaskMemFree( closestMatchFormat ); - - if ( !errorText_.empty() ) - error( errorType ); - return info; -} - -//----------------------------------------------------------------------------- - -unsigned int RtApiWasapi::getDefaultOutputDevice( void ) -{ - for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { - if ( getDeviceInfo( i ).isDefaultOutput ) { - return i; - } - } - - return 0; -} - -//----------------------------------------------------------------------------- - -unsigned int RtApiWasapi::getDefaultInputDevice( void ) -{ - for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { - if ( getDeviceInfo( i ).isDefaultInput ) { - return i; - } - } - - return 0; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiWasapi::closeStream: No open stream to close."; - error( RtAudioError::WARNING ); - return; - } - - if ( stream_.state != STREAM_STOPPED ) - stopStream(); - - // clean up stream memory - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) - - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient ) - SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient ) - - if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ) - CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ); - - if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ) - CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ); - - delete ( WasapiHandle* ) stream_.apiHandle; - stream_.apiHandle = NULL; - - for ( int i = 0; i < 2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - // update stream state - stream_.state = STREAM_CLOSED; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::startStream( void ) -{ - verifyStream(); - - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiWasapi::startStream: The stream is already running."; - error( RtAudioError::WARNING ); - return; - } - - // update stream state - stream_.state = STREAM_RUNNING; - - // create WASAPI stream thread - stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL ); - - if ( !stream_.callbackInfo.thread ) { - errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread."; - error( RtAudioError::THREAD_ERROR ); - } - else { - SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority ); - ResumeThread( ( void* ) stream_.callbackInfo.thread ); - } -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::stopStream( void ) -{ - verifyStream(); - - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiWasapi::stopStream: The stream is already stopped."; - error( RtAudioError::WARNING ); - return; - } - - // inform stream thread by setting stream state to STREAM_STOPPING - stream_.state = STREAM_STOPPING; - - // wait until stream thread is stopped - while( stream_.state != STREAM_STOPPED ) { - Sleep( 1 ); - } - - // Wait for the last buffer to play before stopping. - Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); - - // stop capture client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - - // stop render client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - - // close thread handle - if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { - errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; - error( RtAudioError::THREAD_ERROR ); - return; - } - - stream_.callbackInfo.thread = (ThreadHandle) NULL; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::abortStream( void ) -{ - verifyStream(); - - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiWasapi::abortStream: The stream is already stopped."; - error( RtAudioError::WARNING ); - return; - } - - // inform stream thread by setting stream state to STREAM_STOPPING - stream_.state = STREAM_STOPPING; - - // wait until stream thread is stopped - while ( stream_.state != STREAM_STOPPED ) { - Sleep( 1 ); - } - - // stop capture client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - - // stop render client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - - // close thread handle - if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { - errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; - error( RtAudioError::THREAD_ERROR ); - return; - } - - stream_.callbackInfo.thread = (ThreadHandle) NULL; -} - -//----------------------------------------------------------------------------- - -bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int* bufferSize, - RtAudio::StreamOptions* options ) -{ - bool methodResult = FAILURE; - unsigned int captureDeviceCount = 0; - unsigned int renderDeviceCount = 0; - - IMMDeviceCollection* captureDevices = NULL; - IMMDeviceCollection* renderDevices = NULL; - IMMDevice* devicePtr = NULL; - WAVEFORMATEX* deviceFormat = NULL; - unsigned int bufferBytes; - stream_.state = STREAM_STOPPED; - - // create API Handle if not already created - if ( !stream_.apiHandle ) - stream_.apiHandle = ( void* ) new WasapiHandle(); - - // Count capture devices - errorText_.clear(); - RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; - HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; - goto Exit; - } - - hr = captureDevices->GetCount( &captureDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; - goto Exit; - } - - // Count render devices - hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; - goto Exit; - } - - hr = renderDevices->GetCount( &renderDeviceCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; - goto Exit; - } - - // validate device index - if ( device >= captureDeviceCount + renderDeviceCount ) { - errorType = RtAudioError::INVALID_USE; - errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; - goto Exit; - } - - // determine whether index falls within capture or render devices - if ( device >= renderDeviceCount ) { - if ( mode != INPUT ) { - errorType = RtAudioError::INVALID_USE; - errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; - goto Exit; - } - - // retrieve captureAudioClient from devicePtr - IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; - - hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; - goto Exit; - } - - hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, - NULL, ( void** ) &captureAudioClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; - goto Exit; - } - - hr = captureAudioClient->GetMixFormat( &deviceFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; - goto Exit; - } - - stream_.nDeviceChannels[mode] = deviceFormat->nChannels; - captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); - } - else { - if ( mode != OUTPUT ) { - errorType = RtAudioError::INVALID_USE; - errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device."; - goto Exit; - } - - // retrieve renderAudioClient from devicePtr - IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; - - hr = renderDevices->Item( device, &devicePtr ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; - goto Exit; - } - - hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, - NULL, ( void** ) &renderAudioClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; - goto Exit; - } - - hr = renderAudioClient->GetMixFormat( &deviceFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; - goto Exit; - } - - stream_.nDeviceChannels[mode] = deviceFormat->nChannels; - renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); - } - - // fill stream data - if ( ( stream_.mode == OUTPUT && mode == INPUT ) || - ( stream_.mode == INPUT && mode == OUTPUT ) ) { - stream_.mode = DUPLEX; - } - else { - stream_.mode = mode; - } - - stream_.device[mode] = device; - stream_.doByteSwap[mode] = false; - stream_.sampleRate = sampleRate; - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 1; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = firstChannel; - stream_.userFormat = format; - stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; - else - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] || - stream_.nUserChannels != stream_.nDeviceChannels ) - stream_.doConvertBuffer[mode] = true; - else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - if ( stream_.doConvertBuffer[mode] ) - setConvertInfo( mode, 0 ); - - // Allocate necessary internal buffers - bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat ); - - stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 ); - if ( !stream_.userBuffer[mode] ) { - errorType = RtAudioError::MEMORY_ERROR; - errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; - goto Exit; - } - - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) - stream_.callbackInfo.priority = 15; - else - stream_.callbackInfo.priority = 0; - - ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback - ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode - - methodResult = SUCCESS; - -Exit: - //clean up - SAFE_RELEASE( captureDevices ); - SAFE_RELEASE( renderDevices ); - SAFE_RELEASE( devicePtr ); - CoTaskMemFree( deviceFormat ); - - // if method failed, close the stream - if ( methodResult == FAILURE ) - closeStream(); - - if ( !errorText_.empty() ) - error( errorType ); - return methodResult; -} - -//============================================================================= - -DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) -{ - if ( wasapiPtr ) - ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); - - return 0; -} - -DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) -{ - if ( wasapiPtr ) - ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); - - return 0; -} - -DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) -{ - if ( wasapiPtr ) - ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); - - return 0; -} - -//----------------------------------------------------------------------------- - -void RtApiWasapi::wasapiThread() -{ - // as this is a new thread, we must CoInitialize it - CoInitialize( NULL ); - - HRESULT hr; - - IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; - IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; - IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient; - IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient; - HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent; - HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent; - - WAVEFORMATEX* captureFormat = NULL; - WAVEFORMATEX* renderFormat = NULL; - float captureSrRatio = 0.0f; - float renderSrRatio = 0.0f; - WasapiBuffer captureBuffer; - WasapiBuffer renderBuffer; - - // declare local stream variables - RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; - BYTE* streamBuffer = NULL; - unsigned long captureFlags = 0; - unsigned int bufferFrameCount = 0; - unsigned int numFramesPadding = 0; - unsigned int convBufferSize = 0; - bool callbackPushed = false; - bool callbackPulled = false; - bool callbackStopped = false; - int callbackResult = 0; - - // convBuffer is used to store converted buffers between WASAPI and the user - char* convBuffer = NULL; - unsigned int convBuffSize = 0; - unsigned int deviceBuffSize = 0; - - errorText_.clear(); - RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; - - // Attempt to assign "Pro Audio" characteristic to thread - HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" ); - if ( AvrtDll ) { - DWORD taskIndex = 0; - TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); - AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); - FreeLibrary( AvrtDll ); - } - - // start capture stream if applicable - if ( captureAudioClient ) { - hr = captureAudioClient->GetMixFormat( &captureFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; - goto Exit; - } - - captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); - - // initialize capture stream according to desire buffer size - float desiredBufferSize = stream_.bufferSize * captureSrRatio; - REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec ); - - if ( !captureClient ) { - hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, - AUDCLNT_STREAMFLAGS_EVENTCALLBACK, - desiredBufferPeriod, - desiredBufferPeriod, - captureFormat, - NULL ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; - goto Exit; - } - - hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), - ( void** ) &captureClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; - goto Exit; - } - - // configure captureEvent to trigger on every available capture buffer - captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); - if ( !captureEvent ) { - errorType = RtAudioError::SYSTEM_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event."; - goto Exit; - } - - hr = captureAudioClient->SetEventHandle( captureEvent ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; - goto Exit; - } - - ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; - ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; - } - - unsigned int inBufferSize = 0; - hr = captureAudioClient->GetBufferSize( &inBufferSize ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; - goto Exit; - } - - // scale outBufferSize according to stream->user sample rate ratio - unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT]; - inBufferSize *= stream_.nDeviceChannels[INPUT]; - - // set captureBuffer size - captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); - - // reset the capture stream - hr = captureAudioClient->Reset(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; - goto Exit; - } - - // start the capture stream - hr = captureAudioClient->Start(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream."; - goto Exit; - } - } - - // start render stream if applicable - if ( renderAudioClient ) { - hr = renderAudioClient->GetMixFormat( &renderFormat ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; - goto Exit; - } - - renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); - - // initialize render stream according to desire buffer size - float desiredBufferSize = stream_.bufferSize * renderSrRatio; - REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec ); - - if ( !renderClient ) { - hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, - AUDCLNT_STREAMFLAGS_EVENTCALLBACK, - desiredBufferPeriod, - desiredBufferPeriod, - renderFormat, - NULL ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; - goto Exit; - } - - hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), - ( void** ) &renderClient ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; - goto Exit; - } - - // configure renderEvent to trigger on every available render buffer - renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); - if ( !renderEvent ) { - errorType = RtAudioError::SYSTEM_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event."; - goto Exit; - } - - hr = renderAudioClient->SetEventHandle( renderEvent ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle."; - goto Exit; - } - - ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; - ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; - } - - unsigned int outBufferSize = 0; - hr = renderAudioClient->GetBufferSize( &outBufferSize ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; - goto Exit; - } - - // scale inBufferSize according to user->stream sample rate ratio - unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT]; - outBufferSize *= stream_.nDeviceChannels[OUTPUT]; - - // set renderBuffer size - renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) ); - - // reset the render stream - hr = renderAudioClient->Reset(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream."; - goto Exit; - } - - // start the render stream - hr = renderAudioClient->Start(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream."; - goto Exit; - } - } - - if ( stream_.mode == INPUT ) { - convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); - deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); - } - else if ( stream_.mode == OUTPUT ) { - convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); - deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); - } - else if ( stream_.mode == DUPLEX ) { - convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), - ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); - deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), - stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); - } - - convBuffer = ( char* ) malloc( convBuffSize ); - stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize ); - if ( !convBuffer || !stream_.deviceBuffer ) { - errorType = RtAudioError::MEMORY_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; - goto Exit; - } - - // stream process loop - while ( stream_.state != STREAM_STOPPING ) { - if ( !callbackPulled ) { - // Callback Input - // ============== - // 1. Pull callback buffer from inputBuffer - // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count - // Convert callback buffer to user format - - if ( captureAudioClient ) { - // Pull callback buffer from inputBuffer - callbackPulled = captureBuffer.pullBuffer( convBuffer, - ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT], - stream_.deviceFormat[INPUT] ); - - if ( callbackPulled ) { - // Convert callback buffer to user sample rate - convertBufferWasapi( stream_.deviceBuffer, - convBuffer, - stream_.nDeviceChannels[INPUT], - captureFormat->nSamplesPerSec, - stream_.sampleRate, - ( unsigned int ) ( stream_.bufferSize * captureSrRatio ), - convBufferSize, - stream_.deviceFormat[INPUT] ); - - if ( stream_.doConvertBuffer[INPUT] ) { - // Convert callback buffer to user format - convertBuffer( stream_.userBuffer[INPUT], - stream_.deviceBuffer, - stream_.convertInfo[INPUT] ); - } - else { - // no further conversion, simple copy deviceBuffer to userBuffer - memcpy( stream_.userBuffer[INPUT], - stream_.deviceBuffer, - stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) ); - } - } - } - else { - // if there is no capture stream, set callbackPulled flag - callbackPulled = true; - } - - // Execute Callback - // ================ - // 1. Execute user callback method - // 2. Handle return value from callback - - // if callback has not requested the stream to stop - if ( callbackPulled && !callbackStopped ) { - // Execute user callback method - callbackResult = callback( stream_.userBuffer[OUTPUT], - stream_.userBuffer[INPUT], - stream_.bufferSize, - getStreamTime(), - captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, - stream_.callbackInfo.userData ); - - // Handle return value from callback - if ( callbackResult == 1 ) { - // instantiate a thread to stop this thread - HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL ); - if ( !threadHandle ) { - errorType = RtAudioError::THREAD_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; - goto Exit; - } - else if ( !CloseHandle( threadHandle ) ) { - errorType = RtAudioError::THREAD_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; - goto Exit; - } - - callbackStopped = true; - } - else if ( callbackResult == 2 ) { - // instantiate a thread to stop this thread - HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); - if ( !threadHandle ) { - errorType = RtAudioError::THREAD_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; - goto Exit; - } - else if ( !CloseHandle( threadHandle ) ) { - errorType = RtAudioError::THREAD_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; - goto Exit; - } - - callbackStopped = true; - } - } - } - - // Callback Output - // =============== - // 1. Convert callback buffer to stream format - // 2. Convert callback buffer to stream sample rate and channel count - // 3. Push callback buffer into outputBuffer - - if ( renderAudioClient && callbackPulled ) { - if ( stream_.doConvertBuffer[OUTPUT] ) { - // Convert callback buffer to stream format - convertBuffer( stream_.deviceBuffer, - stream_.userBuffer[OUTPUT], - stream_.convertInfo[OUTPUT] ); - - } - - // Convert callback buffer to stream sample rate - convertBufferWasapi( convBuffer, - stream_.deviceBuffer, - stream_.nDeviceChannels[OUTPUT], - stream_.sampleRate, - renderFormat->nSamplesPerSec, - stream_.bufferSize, - convBufferSize, - stream_.deviceFormat[OUTPUT] ); - - // Push callback buffer into outputBuffer - callbackPushed = renderBuffer.pushBuffer( convBuffer, - convBufferSize * stream_.nDeviceChannels[OUTPUT], - stream_.deviceFormat[OUTPUT] ); - } - else { - // if there is no render stream, set callbackPushed flag - callbackPushed = true; - } - - // Stream Capture - // ============== - // 1. Get capture buffer from stream - // 2. Push capture buffer into inputBuffer - // 3. If 2. was successful: Release capture buffer - - if ( captureAudioClient ) { - // if the callback input buffer was not pulled from captureBuffer, wait for next capture event - if ( !callbackPulled ) { - WaitForSingleObject( captureEvent, INFINITE ); - } - - // Get capture buffer from stream - hr = captureClient->GetBuffer( &streamBuffer, - &bufferFrameCount, - &captureFlags, NULL, NULL ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; - goto Exit; - } - - if ( bufferFrameCount != 0 ) { - // Push capture buffer into inputBuffer - if ( captureBuffer.pushBuffer( ( char* ) streamBuffer, - bufferFrameCount * stream_.nDeviceChannels[INPUT], - stream_.deviceFormat[INPUT] ) ) - { - // Release capture buffer - hr = captureClient->ReleaseBuffer( bufferFrameCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; - goto Exit; - } - } - else - { - // Inform WASAPI that capture was unsuccessful - hr = captureClient->ReleaseBuffer( 0 ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; - goto Exit; - } - } - } - else - { - // Inform WASAPI that capture was unsuccessful - hr = captureClient->ReleaseBuffer( 0 ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; - goto Exit; - } - } - } - - // Stream Render - // ============= - // 1. Get render buffer from stream - // 2. Pull next buffer from outputBuffer - // 3. If 2. was successful: Fill render buffer with next buffer - // Release render buffer - - if ( renderAudioClient ) { - // if the callback output buffer was not pushed to renderBuffer, wait for next render event - if ( callbackPulled && !callbackPushed ) { - WaitForSingleObject( renderEvent, INFINITE ); - } - - // Get render buffer from stream - hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; - goto Exit; - } - - hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; - goto Exit; - } - - bufferFrameCount -= numFramesPadding; - - if ( bufferFrameCount != 0 ) { - hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; - goto Exit; - } - - // Pull next buffer from outputBuffer - // Fill render buffer with next buffer - if ( renderBuffer.pullBuffer( ( char* ) streamBuffer, - bufferFrameCount * stream_.nDeviceChannels[OUTPUT], - stream_.deviceFormat[OUTPUT] ) ) - { - // Release render buffer - hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; - goto Exit; - } - } - else - { - // Inform WASAPI that render was unsuccessful - hr = renderClient->ReleaseBuffer( 0, 0 ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; - goto Exit; - } - } - } - else - { - // Inform WASAPI that render was unsuccessful - hr = renderClient->ReleaseBuffer( 0, 0 ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; - goto Exit; - } - } - } - - // if the callback buffer was pushed renderBuffer reset callbackPulled flag - if ( callbackPushed ) { - callbackPulled = false; - // tick stream time - RtApi::tickStreamTime(); - } - - } - -Exit: - // clean up - CoTaskMemFree( captureFormat ); - CoTaskMemFree( renderFormat ); - - free ( convBuffer ); - - CoUninitialize(); - - // update stream state - stream_.state = STREAM_STOPPED; - - if ( errorText_.empty() ) - return; - else - error( errorType ); -} - -//******************** End of __WINDOWS_WASAPI__ *********************// -#endif - - -#if defined(__WINDOWS_DS__) // Windows DirectSound API - -// Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. -// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. -// - Auto-call CoInitialize for DSOUND and ASIO platforms. -// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 -// Changed device query structure for RtAudio 4.0.7, January 2010 - -#include -#include -#include -#include -#include - -#if defined(__MINGW32__) - // missing from latest mingw winapi -#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ -#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ -#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ -#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ -#endif - -#define MINIMUM_DEVICE_BUFFER_SIZE 32768 - -#ifdef _MSC_VER // if Microsoft Visual C++ -#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. -#endif - -static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) -{ - if ( pointer > bufferSize ) pointer -= bufferSize; - if ( laterPointer < earlierPointer ) laterPointer += bufferSize; - if ( pointer < earlierPointer ) pointer += bufferSize; - return pointer >= earlierPointer && pointer < laterPointer; -} - -// A structure to hold various information related to the DirectSound -// API implementation. -struct DsHandle { - unsigned int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - void *id[2]; - void *buffer[2]; - bool xrun[2]; - UINT bufferPointer[2]; - DWORD dsBufferSize[2]; - DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. - HANDLE condition; - - DsHandle() - :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } -}; - -// Declarations for utility functions, callbacks, and structures -// specific to the DirectSound implementation. -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ); - -static const char* getErrorString( int code ); - -static unsigned __stdcall callbackHandler( void *ptr ); - -struct DsDevice { - LPGUID id[2]; - bool validId[2]; - bool found; - std::string name; - - DsDevice() - : found(false) { validId[0] = false; validId[1] = false; } -}; - -struct DsProbeData { - bool isInput; - std::vector* dsDevices; -}; - -RtApiDs :: RtApiDs() -{ - // Dsound will run both-threaded. If CoInitialize fails, then just - // accept whatever the mainline chose for a threading model. - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) coInitialized_ = true; -} - -RtApiDs :: ~RtApiDs() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); - if ( coInitialized_ ) CoUninitialize(); // balanced call. -} - -// The DirectSound default output is always the first device. -unsigned int RtApiDs :: getDefaultOutputDevice( void ) -{ - return 0; -} - -// The DirectSound default input is always the first input device, -// which is the first capture device enumerated. -unsigned int RtApiDs :: getDefaultInputDevice( void ) -{ - return 0; -} - -unsigned int RtApiDs :: getDeviceCount( void ) -{ - // Set query flag for previously found devices to false, so that we - // can check for any devices that have disappeared. - for ( unsigned int i=0; i(dsDevices.size()); -} - -RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - if ( dsDevices.size() == 0 ) { - // Force a query of all devices - getDeviceCount(); - if ( dsDevices.size() == 0 ) { - errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - } - - if ( device >= dsDevices.size() ) { - errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - HRESULT result; - if ( dsDevices[ device ].validId[0] == false ) goto probeInput; - - LPDIRECTSOUND output; - DSCAPS outCaps; - result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto probeInput; - } - - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto probeInput; - } - - // Get output channel information. - info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - - // Get sample rate information. - info.sampleRates.clear(); - for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && - SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) { - info.sampleRates.push_back( SAMPLE_RATES[k] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - } - } - - // Get format information. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; - if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; - - output->Release(); - - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; - - if ( dsDevices[ device ].validId[1] == false ) { - info.name = dsDevices[ device ].name; - info.probed = true; - return info; - } - - probeInput: - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get input channel information. - info.inputChannels = inCaps.dwChannels; - - // Get sample rate and format information. - std::vector rates; - if ( inCaps.dwChannels >= 2 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); - } - } - else if ( inCaps.dwChannels == 1 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); - } - } - else info.inputChannels = 0; // technically, this would be an error - - input->Release(); - - if ( info.inputChannels == 0 ) return info; - - // Copy the supported rates to the info structure but avoid duplication. - bool found; - for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - if ( device == 0 ) info.isDefaultInput = true; - - // Copy name and return. - info.name = dsDevices[ device ].name; - info.probed = true; - return info; -} - -bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - if ( channels + firstChannel > 2 ) { - errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; - return FAILURE; - } - - size_t nDevices = dsDevices.size(); - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiDs::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - if ( mode == OUTPUT ) { - if ( dsDevices[ device ].validId[0] == false ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - else { // mode == INPUT - if ( dsDevices[ device ].validId[1] == false ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // According to a note in PortAudio, using GetDesktopWindow() - // instead of GetForegroundWindow() is supposed to avoid problems - // that occur when the application's window is not the foreground - // window. Also, if the application window closes before the - // DirectSound buffer, DirectSound can crash. In the past, I had - // problems when using GetDesktopWindow() but it seems fine now - // (January 2010). I'll leave it commented here. - // HWND hWnd = GetForegroundWindow(); - HWND hWnd = GetDesktopWindow(); - - // Check the numberOfBuffers parameter and limit the lowest value to - // two. This is a judgement call and a value of two is probably too - // low for capture, but it should work for playback. - int nBuffers = 0; - if ( options ) nBuffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; - if ( nBuffers < 2 ) nBuffers = 3; - - // Check the lower range of the user-specified buffer size and set - // (arbitrarily) to a lower bound of 32. - if ( *bufferSize < 32 ) *bufferSize = 32; - - // Create the wave format structure. The data format setting will - // be determined later. - WAVEFORMATEX waveFormat; - ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); - waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels + firstChannel; - waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - - // Determine the device buffer size. By default, we'll use the value - // defined above (32K), but we will grow it to make allowances for - // very large software buffer sizes. - DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE; - DWORD dsPointerLeadTime = 0; - - void *ohandle = 0, *bhandle = 0; - HRESULT result; - if ( mode == OUTPUT ) { - - LPDIRECTSOUND output; - result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - DSCAPS outCaps; - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check channel information. - if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check format information. Use 16-bit format unless not - // supported or user requests 8-bit. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && - !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > dsBufferSize ) - dsBufferSize *= 2; - - // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. - // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); - // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. - result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Even though we will write to the secondary buffer, we need to - // access the primary buffer to set the correct output format - // (since the default is 8-bit, 22 kHz!). Setup the DS primary - // buffer description. - DSBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; - - // Obtain the primary buffer - LPDIRECTSOUNDBUFFER buffer; - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the primary DS buffer sound format. - result = buffer->SetFormat( &waveFormat ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Setup the secondary DS buffer description. - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = dsBufferSize; - bufferDescription.lpwfxFormat = &waveFormat; - - // Try to create the secondary DS buffer. If that doesn't work, - // try to use software mixing. Otherwise, there's a problem. - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Get the buffer size ... might be different from what we specified. - DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof( DSBCAPS ); - result = buffer->GetCaps( &dsbcaps ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - dsBufferSize = dsbcaps.dwBufferBytes; - - // Lock the DS buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - ohandle = (void *) output; - bhandle = (void *) buffer; - } - - if ( mode == INPUT ) { - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check channel information. - if ( inCaps.dwChannels < channels + firstChannel ) { - errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; - return FAILURE; - } - - // Check format information. Use 16-bit format unless user - // requests 8-bit. - DWORD deviceFormats; - if ( channels + firstChannel == 2 ) { - deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - else { // channel == 1 - deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > dsBufferSize ) - dsBufferSize *= 2; - - // Setup the secondary DS buffer description. - DSCBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); - bufferDescription.dwFlags = 0; - bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = dsBufferSize; - bufferDescription.lpwfxFormat = &waveFormat; - - // Create the capture buffer. - LPDIRECTSOUNDCAPTUREBUFFER buffer; - result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Get the buffer size ... might be different from what we specified. - DSCBCAPS dscbcaps; - dscbcaps.dwSize = sizeof( DSCBCAPS ); - result = buffer->GetCaps( &dscbcaps ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - dsBufferSize = dscbcaps.dwBufferBytes; - - // NOTE: We could have a problem here if this is a duplex stream - // and the play and capture hardware buffer sizes are different - // (I'm actually not sure if that is a problem or not). - // Currently, we are not verifying that. - - // Lock the capture buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Zero the buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - ohandle = (void *) input; - bhandle = (void *) buffer; - } - - // Set various stream parameters - DsHandle *handle = 0; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.nUserChannels[mode] = channels; - stream_.bufferSize = *bufferSize; - stream_.channelOffset[mode] = firstChannel; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // Set flag for buffer conversion - stream_.doConvertBuffer[mode] = false; - if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) - stream_.doConvertBuffer[mode] = true; - if (stream_.userFormat != stream_.deviceFormat[mode]) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Allocate our DsHandle structures for the stream. - if ( stream_.apiHandle == 0 ) { - try { - handle = new DsHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; - } - - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - else - handle = (DsHandle *) stream_.apiHandle; - handle->id[mode] = ohandle; - handle->buffer[mode] = bhandle; - handle->dsBufferSize[mode] = dsBufferSize; - handle->dsPointerLeadTime[mode] = dsPointerLeadTime; - - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - stream_.nBuffers = nBuffers; - stream_.sampleRate = sampleRate; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup the callback thread. - if ( stream_.callbackInfo.isRunning == false ) { - unsigned threadId; - stream_.callbackInfo.isRunning = true; - stream_.callbackInfo.object = (void *) this; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, - &stream_.callbackInfo, 0, &threadId ); - if ( stream_.callbackInfo.thread == 0 ) { - errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; - goto error; - } - - // Boost DS thread priority - SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); - } - return SUCCESS; - - error: - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) buffer->Release(); - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) buffer->Release(); - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -void RtApiDs :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - // Stop the callback thread. - stream_.callbackInfo.isRunning = false; - WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE) stream_.callbackInfo.thread ); - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiDs :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiDs::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - - // Increase scheduler frequency on lesser windows (a side-effect of - // increasing timer accuracy). On greater windows (Win2K or later), - // this is already in effect. - timeBeginPeriod( 1 ); - - buffersRolling = false; - duplexPrerollBytes = 0; - - if ( stream_.mode == DUPLEX ) { - // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. - duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); - } - - HRESULT result = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - result = buffer->Start( DSCBSTART_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - handle->drainCounter = 0; - handle->internalDrain = false; - ResetEvent( handle->condition ); - stream_.state = STREAM_RUNNING; - - unlock: - if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiDs :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - HRESULT result = 0; - LPVOID audioPtr; - DWORD dataLen; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 2; - WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - } - - stream_.state = STREAM_STOPPED; - - MUTEX_LOCK( &stream_.mutex ); - - // Stop the buffer and clear memory - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // If we start playing again, we must begin at beginning of buffer. - handle->bufferPointer[0] = 0; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - audioPtr = NULL; - dataLen = 0; - - stream_.state = STREAM_STOPPED; - - if ( stream_.mode != DUPLEX ) - MUTEX_LOCK( &stream_.mutex ); - - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // If we start recording again, we must begin at beginning of buffer. - handle->bufferPointer[1] = 0; - } - - unlock: - timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - MUTEX_UNLOCK( &stream_.mutex ); - - if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiDs :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - handle->drainCounter = 2; - - stopStream(); -} - -void RtApiDs :: callbackEvent() -{ - if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { - Sleep( 50 ); // sleep 50 milliseconds - return; - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > stream_.nBuffers + 2 ) { - - stream_.state = STREAM_STOPPING; - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return; - } - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( cbReturnValue == 2 ) { - stream_.state = STREAM_STOPPING; - handle->drainCounter = 2; - abortStream(); - return; - } - else if ( cbReturnValue == 1 ) { - handle->drainCounter = 1; - handle->internalDrain = true; - } - } - - HRESULT result; - DWORD currentWritePointer, safeWritePointer; - DWORD currentReadPointer, safeReadPointer; - UINT nextWritePointer; - - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; - - char *buffer; - long bufferBytes; - - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - if ( buffersRolling == false ) { - if ( stream_.mode == DUPLEX ) { - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - // It takes a while for the devices to get rolling. As a result, - // there's no guarantee that the capture and write device pointers - // will move in lockstep. Wait here for both devices to start - // rolling, and then set our buffer pointers accordingly. - // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 - // bytes later than the write buffer. - - // Stub: a serious risk of having a pre-emptive scheduling round - // take place between the two GetCurrentPosition calls... but I'm - // really not sure how to solve the problem. Temporarily boost to - // Realtime priority, maybe; but I'm not sure what priority the - // DirectSound service threads run at. We *should* be roughly - // within a ms or so of correct. - - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - - DWORD startSafeWritePointer, startSafeReadPointer; - - result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - while ( true ) { - result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; - Sleep( 1 ); - } - - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - handle->bufferPointer[1] = safeReadPointer; - } - else if ( stream_.mode == OUTPUT ) { - - // Set the proper nextWritePosition after initial startup. - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; - if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; - } - - buffersRolling = true; - } - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - memset( stream_.userBuffer[0], 0, bufferBytes ); - } - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; - bufferBytes *= formatBytes( stream_.deviceFormat[0] ); - } - else { - buffer = stream_.userBuffer[0]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - // No byte swapping necessary in DirectSound implementation. - - // Ahhh ... windoze. 16-bit data is signed but 8-bit data is - // unsigned. So, we need to convert our signed 8-bit data here to - // unsigned. - if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) - for ( int i=0; idsBufferSize[0]; - nextWritePointer = handle->bufferPointer[0]; - - DWORD endWrite, leadPointer; - while ( true ) { - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - // We will copy our output buffer into the region between - // safeWritePointer and leadPointer. If leadPointer is not - // beyond the next endWrite position, wait until it is. - leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; - //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; - if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; - if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset - endWrite = nextWritePointer + bufferBytes; - - // Check whether the entire write region is behind the play pointer. - if ( leadPointer >= endWrite ) break; - - // If we are here, then we must wait until the leadPointer advances - // beyond the end of our next write region. We use the - // Sleep() function to suspend operation until that happens. - double millis = ( endWrite - leadPointer ) * 1000.0; - millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - } - - if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) - || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { - // We've strayed into the forbidden zone ... resync the read pointer. - handle->xrun[0] = true; - nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; - if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; - handle->bufferPointer[0] = nextWritePointer; - endWrite = nextWritePointer + bufferBytes; - } - - // Lock free space in the buffer - result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - // Copy our buffer into the DS buffer - CopyMemory( buffer1, buffer, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); - - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; - handle->bufferPointer[0] = nextWritePointer; - } - - // Don't bother draining input - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; - bufferBytes *= formatBytes( stream_.deviceFormat[1] ); - } - else { - buffer = stream_.userBuffer[1]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - long nextReadPointer = handle->bufferPointer[1]; - DWORD dsBufferSize = handle->dsBufferSize[1]; - - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPointer + bufferBytes; - - // Handling depends on whether we are INPUT or DUPLEX. - // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, - // then a wait here will drag the write pointers into the forbidden zone. - // - // In DUPLEX mode, rather than wait, we will back off the read pointer until - // it's in a safe position. This causes dropouts, but it seems to be the only - // practical way to sync up the read and write pointers reliably, given the - // the very complex relationship between phase and increment of the read and write - // pointers. - // - // In order to minimize audible dropouts in DUPLEX mode, we will - // provide a pre-roll period of 0.5 seconds in which we return - // zeros from the read buffer while the pointers sync up. - - if ( stream_.mode == DUPLEX ) { - if ( safeReadPointer < endRead ) { - if ( duplexPrerollBytes <= 0 ) { - // Pre-roll time over. Be more agressive. - int adjustment = endRead-safeReadPointer; - - handle->xrun[1] = true; - // Two cases: - // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, - // and perform fine adjustments later. - // - small adjustments: back off by twice as much. - if ( adjustment >= 2*bufferBytes ) - nextReadPointer = safeReadPointer-2*bufferBytes; - else - nextReadPointer = safeReadPointer-bufferBytes-adjustment; - - if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; - - } - else { - // In pre=roll time. Just do it. - nextReadPointer = safeReadPointer - bufferBytes; - while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; - } - endRead = nextReadPointer + bufferBytes; - } - } - else { // mode == INPUT - while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { - // See comments for playback. - double millis = (endRead - safeReadPointer) * 1000.0; - millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up and find out where we are now. - result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset - } - } - - // Lock free space in the buffer - result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - - if ( duplexPrerollBytes <= 0 ) { - // Copy our buffer into the DS buffer - CopyMemory( buffer, buffer1, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); - } - else { - memset( buffer, 0, bufferSize1 ); - if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); - duplexPrerollBytes -= bufferSize1 + bufferSize2; - } - - // Update our buffer offset and unlock sound buffer - nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - handle->bufferPointer[1] = nextReadPointer; - - // No byte swapping necessary in DirectSound implementation. - - // If necessary, convert 8-bit data from unsigned to signed. - if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) - for ( int j=0; jobject; - bool* isRunning = &info->isRunning; - - while ( *isRunning == true ) { - object->callbackEvent(); - } - - _endthreadex( 0 ); - return 0; -} - -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR /*module*/, - LPVOID lpContext ) -{ - struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; - std::vector& dsDevices = *probeInfo.dsDevices; - - HRESULT hr; - bool validDevice = false; - if ( probeInfo.isInput == true ) { - DSCCAPS caps; - LPDIRECTSOUNDCAPTURE object; - - hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) - validDevice = true; - } - object->Release(); - } - else { - DSCAPS caps; - LPDIRECTSOUND object; - hr = DirectSoundCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) - validDevice = true; - } - object->Release(); - } - - // If good device, then save its name and guid. - std::string name = convertCharPointerToStdString( description ); - //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) - if ( lpguid == NULL ) - name = "Default Device"; - if ( validDevice ) { - for ( unsigned int i=0; i -#include - - // A structure to hold various information related to the ALSA API - // implementation. -struct AlsaHandle { - snd_pcm_t *handles[2]; - bool synchronized; - bool xrun[2]; - pthread_cond_t runnable_cv; - bool runnable; - - AlsaHandle() - :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } -}; - -static void *alsaCallbackHandler( void * ptr ); - -RtApiAlsa :: RtApiAlsa() -{ - // Nothing to do here. -} - -RtApiAlsa :: ~RtApiAlsa() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiAlsa :: getDeviceCount( void ) -{ - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *handle; - - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &handle, name, 0 ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( handle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - break; - } - if ( subdevice < 0 ) - break; - nDevices++; - } - nextcard: - snd_ctl_close( handle ); - snd_card_next( &card ); - } - - result = snd_ctl_open( &handle, "default", 0 ); - if (result == 0) { - nDevices++; - snd_ctl_close( handle ); - } - - return nDevices; -} - -RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *chandle; - - // Count cards and devices - card = -1; - subdevice = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - break; - } - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - goto foundDevice; - } - nDevices++; - } - nextcard: - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - - result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); - if ( result == 0 ) { - if ( nDevices == device ) { - strcpy( name, "default" ); - goto foundDevice; - } - nDevices++; - } - - if ( nDevices == 0 ) { - errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - foundDevice: - - // If a stream is already open, we cannot probe the stream devices. - // Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED && - ( stream_.device[0] == device || stream_.device[1] == device ) ) { - snd_ctl_close( chandle ); - if ( device >= devices_.size() ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; - error( RtAudioError::WARNING ); - return info; - } - return devices_[ device ]; - } - - int openMode = SND_PCM_ASYNC; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); - snd_pcm_t *phandle; - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca( ¶ms ); - - // First try for playback unless default device (which has subdev -1) - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_stream( pcminfo, stream ); - if ( subdevice != -1 ) { - snd_pcm_info_set_device( pcminfo, subdevice ); - snd_pcm_info_set_subdevice( pcminfo, 0 ); - - result = snd_ctl_pcm_info( chandle, pcminfo ); - if ( result < 0 ) { - // Device probably doesn't support playback. - goto captureProbe; - } - } - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto captureProbe; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto captureProbe; - } - - // Get output channel information. - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - goto captureProbe; - } - info.outputChannels = value; - snd_pcm_close( phandle ); - - captureProbe: - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); - - // Now try for capture unless default device (with subdev = -1) - if ( subdevice != -1 ) { - result = snd_ctl_pcm_info( chandle, pcminfo ); - snd_ctl_close( chandle ); - if ( result < 0 ) { - // Device probably doesn't support capture. - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - } - else - snd_ctl_close( chandle ); - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - info.inputChannels = value; - snd_pcm_close( phandle ); - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // ALSA doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; - - probeParameters: - // At this point, we just need to figure out the supported data - // formats and sample rates. We'll proceed by opening the device in - // the direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. - - if ( info.outputChannels >= info.inputChannels ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Test our discrete set of sample rate values. - info.sampleRates.clear(); - for ( unsigned int i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[i]; - } - } - if ( info.sampleRates.size() == 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Probe the supported data formats ... we don't care about endian-ness just yet - snd_pcm_format_t format; - info.nativeFormats = 0; - format = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT8; - format = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT16; - format = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT24; - format = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT32; - format = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT32; - format = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT64; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Get the device name - char *cardname; - result = snd_card_get_name( card, &cardname ); - if ( result >= 0 ) { - sprintf( name, "hw:%s,%d", cardname, subdevice ); - free( cardname ); - } - info.name = name; - - // That's all ... close the device and return - snd_pcm_close( phandle ); - info.probed = true; - return info; -} - -void RtApiAlsa :: saveDeviceInfo( void ) -{ - devices_.clear(); - - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; iflags & RTAUDIO_ALSA_USE_DEFAULT ) - snprintf(name, sizeof(name), "%s", "default"); - else { - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) break; - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - snd_ctl_close( chandle ); - goto foundDevice; - } - nDevices++; - } - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - - result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); - if ( result == 0 ) { - if ( nDevices == device ) { - strcpy( name, "default" ); - goto foundDevice; - } - nDevices++; - } - - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - } - - foundDevice: - - // The getDeviceInfo() function will not work for a device that is - // already open. Thus, we'll probe the system before opening a - // stream and save the results for use by getDeviceInfo(). - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once - this->saveDeviceInfo(); - - snd_pcm_stream_t stream; - if ( mode == OUTPUT ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - - snd_pcm_t *phandle; - int openMode = SND_PCM_ASYNC; - result = snd_pcm_open( &phandle, name, stream, openMode ); - if ( result < 0 ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; - else - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca( &hw_params ); - result = snd_pcm_hw_params_any( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); - snd_pcm_hw_params_dump( hw_params, out ); -#endif - - // Set access ... check user preference. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { - stream_.userInterleaved = false; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - stream_.deviceInterleaved[mode] = true; - } - else - stream_.deviceInterleaved[mode] = false; - } - else { - stream_.userInterleaved = true; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - stream_.deviceInterleaved[mode] = false; - } - else - stream_.deviceInterleaved[mode] = true; - } - - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine how to set the device format. - stream_.userFormat = format; - snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; - - if ( format == RTAUDIO_SINT8 ) - deviceFormat = SND_PCM_FORMAT_S8; - else if ( format == RTAUDIO_SINT16 ) - deviceFormat = SND_PCM_FORMAT_S16; - else if ( format == RTAUDIO_SINT24 ) - deviceFormat = SND_PCM_FORMAT_S24; - else if ( format == RTAUDIO_SINT32 ) - deviceFormat = SND_PCM_FORMAT_S32; - else if ( format == RTAUDIO_FLOAT32 ) - deviceFormat = SND_PCM_FORMAT_FLOAT; - else if ( format == RTAUDIO_FLOAT64 ) - deviceFormat = SND_PCM_FORMAT_FLOAT64; - - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { - stream_.deviceFormat[mode] = format; - goto setFormat; - } - - // The user requested format is not natively supported by the device. - deviceFormat = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - goto setFormat; - } - - // If we get here, no supported format was found. - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - - setFormat: - result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine whether byte-swaping is necessary. - stream_.doByteSwap[mode] = false; - if ( deviceFormat != SND_PCM_FORMAT_S8 ) { - result = snd_pcm_format_cpu_endian( deviceFormat ); - if ( result == 0 ) - stream_.doByteSwap[mode] = true; - else if (result < 0) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Set the sample rate. - result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine the number of channels for this device. We support a possible - // minimum device channel number > than the value requested by the user. - stream_.nUserChannels[mode] = channels; - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); - unsigned int deviceChannels = value; - if ( result < 0 || deviceChannels < channels + firstChannel ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - deviceChannels = value; - if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; - stream_.nDeviceChannels[mode] = deviceChannels; - - // Set the device channels. - result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the buffer (or period) size. - int dir = 0; - snd_pcm_uframes_t periodSize = *bufferSize; - result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - *bufferSize = periodSize; - - // Set the buffer number, which in ALSA is referred to as the "period". - unsigned int periods = 0; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; - if ( periods < 2 ) periods = 4; // a fairly safe default value - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - stream_.bufferSize = *bufferSize; - - // Install the hardware configuration - result = snd_pcm_hw_params( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump( hw_params, out ); -#endif - - // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca( &sw_params ); - snd_pcm_sw_params_current( phandle, sw_params ); - snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); - snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); - - // The following two settings were suggested by Theo Veenker - //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); - //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); - - // here are two options for a fix - //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); - snd_pcm_uframes_t val; - snd_pcm_sw_params_get_boundary( sw_params, &val ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); - - result = snd_pcm_sw_params( phandle, sw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); - snd_pcm_sw_params_dump( sw_params, out ); -#endif - - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the ApiHandle if necessary and then save. - AlsaHandle *apiInfo = 0; - if ( stream_.apiHandle == 0 ) { - try { - apiInfo = (AlsaHandle *) new AlsaHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; - goto error; - } - - if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - - stream_.apiHandle = (void *) apiInfo; - apiInfo->handles[0] = 0; - apiInfo->handles[1] = 0; - } - else { - apiInfo = (AlsaHandle *) stream_.apiHandle; - } - apiInfo->handles[mode] = phandle; - phandle = 0; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.nBuffers = periods; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - // Link the streams if possible. - apiInfo->synchronized = false; - if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) - apiInfo->synchronized = true; - else { - errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; - error( RtAudioError::WARNING ); - } - } - else { - stream_.mode = mode; - - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority (optional). The higher priority will only take affect - // if the program is run as root or suid. Note, under Linux - // processes with CAP_SYS_NICE privilege, a user can change - // scheduling policy and priority (thus need not be root). See - // POSIX "capabilities". - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); - -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - // We previously attempted to increase the audio callback priority - // to SCHED_RR here via the attributes. However, while no errors - // were reported in doing so, it did not work. So, now this is - // done in the alsaCallbackHandler function. - stream_.callbackInfo.doRealtime = true; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - stream_.callbackInfo.priority = priority; - } -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiAlsa::error creating callback thread!"; - goto error; - } - } - - return SUCCESS; - - error: - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable_cv ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } - - if ( phandle) snd_pcm_close( phandle ); - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.state = STREAM_CLOSED; - return FAILURE; -} - -void RtApiAlsa :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) { - apiInfo->runnable = true; - pthread_cond_signal( &apiInfo->runnable_cv ); - } - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); - - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[0] ); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[1] ); - } - - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable_cv ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - 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( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - int result = 0; - snd_pcm_state_t state; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - state = snd_pcm_state( handle[0] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open - state = snd_pcm_state( handle[1] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - } - - stream_.state = STREAM_RUNNING; - - unlock: - apiInfo->runnable = true; - pthread_cond_signal( &apiInfo->runnable_cv ); - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) - result = snd_pcm_drop( handle[0] ); - else - result = snd_pcm_drain( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - apiInfo->runnable = false; // fixes high CPU usage when stopped - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = snd_pcm_drop( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - apiInfo->runnable = false; // fixes high CPU usage when stopped - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: callbackEvent() -{ - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - while ( !apiInfo->runnable ) - pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); - - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - apiInfo->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - apiInfo->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - - if ( doStopStream == 2 ) { - abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - int result; - char *buffer; - int channels; - snd_pcm_t **handle; - snd_pcm_sframes_t frames; - RtAudioFormat format; - handle = (snd_pcm_t **) apiInfo->handles; - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - channels = stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - channels = stream_.nUserChannels[1]; - format = stream_.userFormat; - } - - // Read samples from device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[1] ) - result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ixrun[1] = true; - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtAudioError::WARNING ); - goto tryOutput; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - // Check stream latency - result = snd_pcm_delay( handle[1], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; - } - - tryOutput: - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - channels = stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - channels = stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer(buffer, stream_.bufferSize * channels, format); - - // Write samples to device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[0] ) - result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ixrun[0] = true; - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - else - errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtAudioError::WARNING ); - goto unlock; - } - - // Check stream latency - result = snd_pcm_delay( handle[0], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} - -static void *alsaCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAlsa *object = (RtApiAlsa *) info->object; - bool *isRunning = &info->isRunning; - -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( info->doRealtime ) { - pthread_t tID = pthread_self(); // ID of this thread - sched_param prio = { info->priority }; // scheduling priority of thread - pthread_setschedparam( tID, SCHED_RR, &prio ); - } -#endif - - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } - - pthread_exit( NULL ); -} - -//******************** End of __LINUX_ALSA__ *********************// -#endif - -#if defined(__LINUX_PULSE__) - -// Code written by Peter Meerwald, pmeerw@pmeerw.net -// and Tristan Matthews. - -#include -#include -#include - -static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, - 44100, 48000, 96000, 0}; - -struct rtaudio_pa_format_mapping_t { - RtAudioFormat rtaudio_format; - pa_sample_format_t pa_format; -}; - -static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { - {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, - {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, - {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, - {0, PA_SAMPLE_INVALID}}; - -struct PulseAudioHandle { - pa_simple *s_play; - pa_simple *s_rec; - pthread_t thread; - pthread_cond_t runnable_cv; - bool runnable; - PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } -}; - -RtApiPulse::~RtApiPulse() -{ - if ( stream_.state != STREAM_CLOSED ) - closeStream(); -} - -unsigned int RtApiPulse::getDeviceCount( void ) -{ - return 1; -} - -RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) -{ - RtAudio::DeviceInfo info; - info.probed = true; - info.name = "PulseAudio"; - info.outputChannels = 2; - info.inputChannels = 2; - info.duplexChannels = 2; - info.isDefaultOutput = true; - info.isDefaultInput = true; - - for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) - info.sampleRates.push_back( *sr ); - - info.preferredSampleRate = 48000; - info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; - - return info; -} - -static void *pulseaudio_callback( void * user ) -{ - CallbackInfo *cbi = static_cast( user ); - RtApiPulse *context = static_cast( cbi->object ); - volatile bool *isRunning = &cbi->isRunning; - - while ( *isRunning ) { - pthread_testcancel(); - context->callbackEvent(); - } - - pthread_exit( NULL ); -} - -void RtApiPulse::closeStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - stream_.callbackInfo.isRunning = false; - if ( pah ) { - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) { - pah->runnable = true; - pthread_cond_signal( &pah->runnable_cv ); - } - MUTEX_UNLOCK( &stream_.mutex ); - - pthread_join( pah->thread, 0 ); - if ( pah->s_play ) { - pa_simple_flush( pah->s_play, NULL ); - pa_simple_free( pah->s_play ); - } - if ( pah->s_rec ) - pa_simple_free( pah->s_rec ); - - pthread_cond_destroy( &pah->runnable_cv ); - delete pah; - stream_.apiHandle = 0; - } - - if ( stream_.userBuffer[0] ) { - free( stream_.userBuffer[0] ); - stream_.userBuffer[0] = 0; - } - if ( stream_.userBuffer[1] ) { - free( stream_.userBuffer[1] ); - stream_.userBuffer[1] = 0; - } - - stream_.state = STREAM_CLOSED; - stream_.mode = UNINITIALIZED; -} - -void RtApiPulse::callbackEvent( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - while ( !pah->runnable ) - pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); - - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " - "this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], - stream_.bufferSize, streamTime, status, - stream_.callbackInfo.userData ); - - if ( doStopStream == 2 ) { - abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT]; - void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT]; - - if ( stream_.state != STREAM_RUNNING ) - goto unlock; - - int pa_error; - size_t bytes; - if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( stream_.doConvertBuffer[OUTPUT] ) { - convertBuffer( stream_.deviceBuffer, - stream_.userBuffer[OUTPUT], - stream_.convertInfo[OUTPUT] ); - bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize * - formatBytes( stream_.deviceFormat[OUTPUT] ); - } else - bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize * - formatBytes( stream_.userFormat ); - - if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX) { - if ( stream_.doConvertBuffer[INPUT] ) - bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize * - formatBytes( stream_.deviceFormat[INPUT] ); - else - bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * - formatBytes( stream_.userFormat ); - - if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - } - if ( stream_.doConvertBuffer[INPUT] ) { - convertBuffer( stream_.userBuffer[INPUT], - stream_.deviceBuffer, - stream_.convertInfo[INPUT] ); - } - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - RtApi::tickStreamTime(); - - if ( doStopStream == 1 ) - stopStream(); -} - -void RtApiPulse::startStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::startStream(): the stream is not open!"; - error( RtAudioError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiPulse::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - stream_.state = STREAM_RUNNING; - - pah->runnable = true; - pthread_cond_signal( &pah->runnable_cv ); - MUTEX_UNLOCK( &stream_.mutex ); -} - -void RtApiPulse::stopStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; - error( RtAudioError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - if ( pah && pah->s_play ) { - int pa_error; - if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::stopStream: error draining output device, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); -} - -void RtApiPulse::abortStream( void ) -{ - PulseAudioHandle *pah = static_cast( stream_.apiHandle ); - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; - error( RtAudioError::INVALID_USE ); - return; - } - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - if ( pah && pah->s_play ) { - int pa_error; - if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { - errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << - pa_strerror( pa_error ) << "."; - errorText_ = errorStream_.str(); - MUTEX_UNLOCK( &stream_.mutex ); - error( RtAudioError::SYSTEM_ERROR ); - return; - } - } - - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); -} - -bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, - unsigned int channels, unsigned int firstChannel, - unsigned int sampleRate, RtAudioFormat format, - unsigned int *bufferSize, RtAudio::StreamOptions *options ) -{ - PulseAudioHandle *pah = 0; - unsigned long bufferBytes = 0; - pa_sample_spec ss; - - if ( device != 0 ) return false; - if ( mode != INPUT && mode != OUTPUT ) return false; - if ( channels != 1 && channels != 2 ) { - errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels."; - return false; - } - ss.channels = channels; - - if ( firstChannel != 0 ) return false; - - bool sr_found = false; - for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { - if ( sampleRate == *sr ) { - sr_found = true; - stream_.sampleRate = sampleRate; - ss.rate = sampleRate; - break; - } - } - if ( !sr_found ) { - errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; - return false; - } - - bool sf_found = 0; - for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; - sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { - if ( format == sf->rtaudio_format ) { - sf_found = true; - stream_.userFormat = sf->rtaudio_format; - stream_.deviceFormat[mode] = stream_.userFormat; - ss.format = sf->pa_format; - break; - } - } - if ( !sf_found ) { // Use internal data format conversion. - stream_.userFormat = format; - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - ss.format = PA_SAMPLE_FLOAT32LE; - } - - // Set other stream parameters. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - stream_.nBuffers = 1; - stream_.doByteSwap[mode] = false; - stream_.nUserChannels[mode] = channels; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.channelOffset[mode] = 0; - std::string streamName = "RtAudio"; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers. - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - stream_.bufferSize = *bufferSize; - - 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_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.device[mode] = device; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - if ( !stream_.apiHandle ) { - PulseAudioHandle *pah = new PulseAudioHandle; - if ( !pah ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; - goto error; - } - - stream_.apiHandle = pah; - if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; - goto error; - } - } - pah = static_cast( stream_.apiHandle ); - - int error; - if ( options && !options->streamName.empty() ) streamName = options->streamName; - switch ( mode ) { - case INPUT: - pa_buffer_attr buffer_attr; - buffer_attr.fragsize = bufferBytes; - buffer_attr.maxlength = -1; - - pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error ); - if ( !pah->s_rec ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; - goto error; - } - break; - case OUTPUT: - pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); - if ( !pah->s_play ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; - goto error; - } - break; - default: - goto error; - } - - if ( stream_.mode == UNINITIALIZED ) - stream_.mode = mode; - else if ( stream_.mode == mode ) - goto error; - else - stream_.mode = DUPLEX; - - if ( !stream_.callbackInfo.isRunning ) { - stream_.callbackInfo.object = this; - stream_.callbackInfo.isRunning = true; - if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; - goto error; - } - } - - stream_.state = STREAM_STOPPED; - return true; - - error: - if ( pah && stream_.callbackInfo.isRunning ) { - pthread_cond_destroy( &pah->runnable_cv ); - delete pah; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -//******************** End of __LINUX_PULSE__ *********************// -#endif - -#if defined(__LINUX_OSS__) - -#include -#include -#include -#include -#include -#include -#include - -static void *ossCallbackHandler(void * ptr); - -// A structure to hold various information related to the OSS API -// implementation. -struct OssHandle { - int id[2]; // device ids - bool xrun[2]; - bool triggered; - pthread_cond_t runnable; - - OssHandle() - :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -RtApiOss :: RtApiOss() -{ - // Nothing to do here. -} - -RtApiOss :: ~RtApiOss() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiOss :: getDeviceCount( void ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; - error( RtAudioError::WARNING ); - return 0; - } - - oss_sysinfo sysinfo; - if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtAudioError::WARNING ); - return 0; - } - - close( mixerfd ); - return sysinfo.numaudios; -} - -RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; - error( RtAudioError::WARNING ); - return info; - } - - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtAudioError::WARNING ); - return info; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - if ( device >= nDevices ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; - error( RtAudioError::INVALID_USE ); - return info; - } - - 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( RtAudioError::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; -#ifdef AFMT_FLOAT - if ( mask & AFMT_FLOAT ) - info.nativeFormats |= RTAUDIO_FLOAT32; -#endif - if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) - info.nativeFormats |= RTAUDIO_SINT24; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - return info; - } - - // Probe the supported sample rates. - info.sampleRates.clear(); - if ( ainfo.nrates ) { - for ( unsigned int i=0; i info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - - break; - } - } - } - } - else { - // Check min and max rate values; - for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) { - info.sampleRates.push_back( SAMPLE_RATES[k] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = SAMPLE_RATES[k]; - } - } - } - - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - error( RtAudioError::WARNING ); - } - else { - info.probed = true; - info.name = ainfo.name; - } - - return info; -} - - -bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; - return FAILURE; - } - - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; - return FAILURE; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check if device supports input or output - if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || - ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - int flags = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( mode == OUTPUT ) - flags |= O_WRONLY; - else { // mode == INPUT - if (stream_.mode == OUTPUT && stream_.device[0] == device) { - // We just set the same device for playback ... close and reopen for duplex (OSS only). - close( handle->id[0] ); - handle->id[0] = 0; - if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Check that the number previously set channels is the same. - if ( stream_.nUserChannels[0] != channels ) { - errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - flags |= O_RDWR; - } - else - flags |= O_RDONLY; - } - - // Set exclusive access if specified. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; - - // Try to open the device. - int fd; - fd = open( ainfo.devnode, flags, 0 ); - if ( fd == -1 ) { - if ( errno == EBUSY ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // For duplex operation, specifically set this mode (this doesn't seem to work). - /* - if ( flags | O_RDWR ) { - result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); - if ( result == -1) { - errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - */ - - // Check the device channel support. - stream_.nUserChannels[mode] = channels; - if ( ainfo.max_channels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the number of channels. - int deviceChannels = channels + firstChannel; - result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); - if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = deviceChannels; - - // Get the data format mask - int mask; - result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine how to set the device format. - stream_.userFormat = format; - int deviceFormat = -1; - stream_.doByteSwap[mode] = false; - if ( format == RTAUDIO_SINT8 ) { - if ( mask & AFMT_S8 ) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - else if ( format == RTAUDIO_SINT16 ) { - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT24 ) { - if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT32 ) { - if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - } - - if ( deviceFormat == -1 ) { - // The user requested format is not natively supported by the device. - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S8) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - - if ( stream_.deviceFormat[mode] == 0 ) { - // This really shouldn't happen ... - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the data format. - int temp = deviceFormat; - result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); - if ( result == -1 || deviceFormat != temp ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Attempt to set the buffer size. According to OSS, the minimum - // number of buffers is two. The supposed minimum buffer size is 16 - // bytes, so that will be our lower bound. The argument to this - // call is in the form 0xMMMMSSSS (hex), where the buffer size (in - // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. - // We'll check the actual value used near the end of the setup - // procedure. - int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; - if ( ossBufferBytes < 16 ) ossBufferBytes = 16; - int buffers = 0; - if ( options ) buffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; - if ( buffers < 2 ) buffers = 3; - temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); - result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nBuffers = buffers; - - // Save buffer size (in sample frames). - *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); - stream_.bufferSize = *bufferSize; - - // Set the sample rate. - int srate = sampleRate; - result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Verify the sample rate setup worked. - if ( abs( srate - (int)sampleRate ) > 100 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = sampleRate; - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { - // We're doing duplex setup here. - stream_.deviceFormat[0] = stream_.deviceFormat[1]; - stream_.nDeviceChannels[0] = deviceChannels; - } - - // Set interleaving parameters. - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; - - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the stream handles if necessary and then save. - if ( stream_.apiHandle == 0 ) { - try { - handle = new OssHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; - goto error; - } - - if ( pthread_cond_init( &handle->runnable, NULL ) ) { - errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - - stream_.apiHandle = (void *) handle; - } - else { - handle = (OssHandle *) stream_.apiHandle; - } - handle->id[mode] = fd; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - if ( stream_.device[0] == device ) handle->id[0] = fd; - } - else { - stream_.mode = mode; - - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiOss::error creating callback thread!"; - goto error; - } - } - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiOss :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtAudioError::WARNING ); - return; - } - - OssHandle *handle = (OssHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &handle->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); - - if ( stream_.state == STREAM_RUNNING ) { - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - else - ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - stream_.state = STREAM_STOPPED; - } - - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiOss :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiOss::startStream(): the stream is already running!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - stream_.state = STREAM_RUNNING; - - // No need to do anything else here ... OSS automatically starts - // when fed samples. - - MUTEX_UNLOCK( &stream_.mutex ); - - OssHandle *handle = (OssHandle *) stream_.apiHandle; - pthread_cond_signal( &handle->runnable ); -} - -void RtApiOss :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Flush the output with zeros a few times. - char *buffer; - int samples; - RtAudioFormat format; - - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - memset( buffer, 0, samples * formatBytes(format) ); - for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); - if ( result == -1 ) { - errorText_ = "RtApiOss::stopStream: audio write error."; - error( RtAudioError::WARNING ); - } - } - - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result != -1 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiOss :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtAudioError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result != -1 ) return; - error( RtAudioError::SYSTEM_ERROR ); -} - -void RtApiOss :: callbackEvent() -{ - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &handle->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtAudioError::WARNING ); - return; - } - - // Invoke user callback to get fresh output data. - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - if ( doStopStream == 2 ) { - this->abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - int result; - char *buffer; - int samples; - RtAudioFormat format; - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( buffer, samples, format ); - - if ( stream_.mode == DUPLEX && handle->triggered == false ) { - int trig = 0; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - handle->triggered = true; - } - else - // Write samples to device. - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - - if ( result == -1 ) { - // We'll assume this is an underrun, though there isn't a - // specific means for determining that. - handle->xrun[0] = true; - errorText_ = "RtApiOss::callbackEvent: audio write error."; - error( RtAudioError::WARNING ); - // Continue on to input section. - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - samples = stream_.bufferSize * stream_.nUserChannels[1]; - format = stream_.userFormat; - } - - // Read samples from device. - result = read( handle->id[1], buffer, samples * formatBytes(format) ); - - if ( result == -1 ) { - // We'll assume this is an overrun, though there isn't a - // specific means for determining that. - handle->xrun[1] = true; - errorText_ = "RtApiOss::callbackEvent: audio read error."; - error( RtAudioError::WARNING ); - goto unlock; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, samples, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} - -static void *ossCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiOss *object = (RtApiOss *) info->object; - bool *isRunning = &info->isRunning; - - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } - - pthread_exit( NULL ); -} - -//******************** End of __LINUX_OSS__ *********************// -#endif - - -// *************************************************** // -// -// Protected common (OS-independent) RtAudio methods. -// -// *************************************************** // - -// This method can be modified to control the behavior of error -// message printing. -void RtApi :: error( RtAudioError::Type type ) -{ - errorStream_.str(""); // clear the ostringstream - - RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; - if ( errorCallback ) { - // abortStream() can generate new error messages. Ignore them. Just keep original one. - - if ( firstErrorOccurred_ ) - return; - - firstErrorOccurred_ = true; - const std::string errorMessage = errorText_; - - if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { - stream_.callbackInfo.isRunning = false; // exit from the thread - abortStream(); - } - - errorCallback( type, errorMessage ); - firstErrorOccurred_ = false; - return; - } - - if ( type == RtAudioError::WARNING && showWarnings_ == true ) - std::cerr << '\n' << errorText_ << "\n\n"; - else if ( type != RtAudioError::WARNING ) - throw( RtAudioError( errorText_, type ) ); -} - -void RtApi :: verifyStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApi:: a stream is not open!"; - error( RtAudioError::INVALID_USE ); - } -} - -void RtApi :: clearStreamInfo() -{ - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; - stream_.sampleRate = 0; - stream_.bufferSize = 0; - stream_.nBuffers = 0; - stream_.userFormat = 0; - stream_.userInterleaved = true; - stream_.streamTime = 0.0; - stream_.apiHandle = 0; - stream_.deviceBuffer = 0; - stream_.callbackInfo.callback = 0; - stream_.callbackInfo.userData = 0; - stream_.callbackInfo.isRunning = false; - stream_.callbackInfo.errorCallback = 0; - for ( int i=0; i<2; i++ ) { - stream_.device[i] = 11111; - stream_.doConvertBuffer[i] = false; - stream_.deviceInterleaved[i] = true; - stream_.doByteSwap[i] = false; - stream_.nUserChannels[i] = 0; - stream_.nDeviceChannels[i] = 0; - stream_.channelOffset[i] = 0; - stream_.deviceFormat[i] = 0; - stream_.latency[i] = 0; - stream_.userBuffer[i] = 0; - stream_.convertInfo[i].channels = 0; - stream_.convertInfo[i].inJump = 0; - stream_.convertInfo[i].outJump = 0; - stream_.convertInfo[i].inFormat = 0; - stream_.convertInfo[i].outFormat = 0; - stream_.convertInfo[i].inOffset.clear(); - stream_.convertInfo[i].outOffset.clear(); - } -} - -unsigned int RtApi :: formatBytes( RtAudioFormat format ) -{ - if ( format == RTAUDIO_SINT16 ) - return 2; - else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 ) - return 4; - else if ( format == RTAUDIO_FLOAT64 ) - return 8; - else if ( format == RTAUDIO_SINT24 ) - return 3; - else if ( format == RTAUDIO_SINT8 ) - return 1; - - errorText_ = "RtApi::formatBytes: undefined format."; - error( RtAudioError::WARNING ); - - return 0; -} - -void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) -{ - if ( mode == INPUT ) { // convert device to user buffer - stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; - stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; - stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; - stream_.convertInfo[mode].outFormat = stream_.userFormat; - } - else { // convert user to device buffer - stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; - stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; - stream_.convertInfo[mode].inFormat = stream_.userFormat; - stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; - } - - if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; - else - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; - - // Set up the interleave/deinterleave offsets. - if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { - if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || - ( mode == INPUT && stream_.userInterleaved ) ) { - for ( int k=0; k 0 ) { - if ( stream_.deviceInterleaved[mode] ) { - if ( mode == OUTPUT ) { - for ( int k=0; k> 8); - //out[info.outOffset[j]] >>= 8; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 16) & 0x0000ffff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8) & 0x00ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int24 *in = (Int24 *)inBuffer; - for (unsigned int i=0; i> 16); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 24) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i>8) | (x<<8); } -//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } -//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } - -void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) -{ - char val; - char *ptr; - - ptr = buffer; - if ( format == RTAUDIO_SINT16 ) { - for ( unsigned int i=0; i +#include +#include +#include +#include +#include + +// Static variable definitions. +const unsigned int RtApi::MAX_SAMPLE_RATES = 14; +const unsigned int RtApi::SAMPLE_RATES[] = { + 4000, 5512, 8000, 9600, 11025, 16000, 22050, + 32000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) + #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) + #define MUTEX_DESTROY(A) DeleteCriticalSection(A) + #define MUTEX_LOCK(A) EnterCriticalSection(A) + #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) + + #include "tchar.h" + + static std::string convertCharPointerToStdString(const char *text) + { + return std::string(text); + } + + static std::string convertCharPointerToStdString(const wchar_t *text) + { + int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); + std::string s( length-1, '\0' ); + WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); + return s; + } + +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) + // pthread API + #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) + #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) + #define MUTEX_LOCK(A) pthread_mutex_lock(A) + #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) +#else + #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions + #define MUTEX_DESTROY(A) abs(*A) // dummy definitions +#endif + +// *************************************************** // +// +// RtAudio definitions. +// +// *************************************************** // + +std::string RtAudio :: getVersion( void ) +{ + return RTAUDIO_VERSION; +} + +void RtAudio :: getCompiledApi( std::vector &apis ) +{ + 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_PULSE__) + apis.push_back( LINUX_PULSE ); +#endif +#if defined(__LINUX_OSS__) + apis.push_back( LINUX_OSS ); +#endif +#if defined(__WINDOWS_ASIO__) + apis.push_back( WINDOWS_ASIO ); +#endif +#if defined(__WINDOWS_WASAPI__) + apis.push_back( WINDOWS_WASAPI ); +#endif +#if defined(__WINDOWS_DS__) + apis.push_back( WINDOWS_DS ); +#endif +#if defined(__MACOSX_CORE__) + apis.push_back( MACOSX_CORE ); +#endif +#if defined(__RTAUDIO_DUMMY__) + apis.push_back( RTAUDIO_DUMMY ); +#endif +} + +void RtAudio :: openRtApi( RtAudio::Api api ) +{ + if ( rtapi_ ) + delete rtapi_; + rtapi_ = 0; + +#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_PULSE__) + if ( api == LINUX_PULSE ) + rtapi_ = new RtApiPulse(); +#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_WASAPI__) + if ( api == WINDOWS_WASAPI ) + rtapi_ = new RtApiWasapi(); +#endif +#if defined(__WINDOWS_DS__) + if ( api == WINDOWS_DS ) + rtapi_ = new RtApiDs(); +#endif +#if defined(__MACOSX_CORE__) + if ( api == MACOSX_CORE ) + rtapi_ = new RtApiCore(); +#endif +#if defined(__RTAUDIO_DUMMY__) + if ( api == RTAUDIO_DUMMY ) + rtapi_ = new RtApiDummy(); +#endif +} + +RtAudio :: RtAudio( RtAudio::Api api ) +{ + rtapi_ = 0; + + if ( api != UNSPECIFIED ) { + // Attempt to open the specified API. + openRtApi( api ); + if ( rtapi_ ) return; + + // No compiled support for specified API value. Issue a debug + // warning and continue as if no API was specified. + std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; + } + + // Iterate through the compiled APIs and return as soon as we find + // one with at least one device or we reach the end of the list. + std::vector< RtAudio::Api > apis; + getCompiledApi( apis ); + for ( unsigned int i=0; igetDeviceCount() ) break; + } + + if ( rtapi_ ) return; + + // It should not be possible to get here because the preprocessor + // definition __RTAUDIO_DUMMY__ is automatically defined if no + // API-specific definitions are passed to the compiler. But just in + // case something weird happens, we'll thow an error. + std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n"; + throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) ); +} + +RtAudio :: ~RtAudio() +{ + if ( rtapi_ ) + delete rtapi_; +} + +void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, + RtAudio::StreamParameters *inputParameters, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + return rtapi_->openStream( outputParameters, inputParameters, format, + sampleRate, bufferFrames, callback, + userData, options, errorCallback ); +} + +// *************************************************** // +// +// Public RtApi definitions (see end of file for +// private or protected utility functions). +// +// *************************************************** // + +RtApi :: RtApi() +{ + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; + stream_.apiHandle = 0; + stream_.userBuffer[0] = 0; + stream_.userBuffer[1] = 0; + MUTEX_INITIALIZE( &stream_.mutex ); + showWarnings_ = true; + firstErrorOccurred_ = false; +} + +RtApi :: ~RtApi() +{ + MUTEX_DESTROY( &stream_.mutex ); +} + +void RtApi :: openStream( RtAudio::StreamParameters *oParams, + RtAudio::StreamParameters *iParams, + RtAudioFormat format, unsigned int sampleRate, + unsigned int *bufferFrames, + RtAudioCallback callback, void *userData, + RtAudio::StreamOptions *options, + RtAudioErrorCallback errorCallback ) +{ + if ( stream_.state != STREAM_CLOSED ) { + errorText_ = "RtApi::openStream: a stream is already open!"; + error( RtAudioError::INVALID_USE ); + return; + } + + // Clear stream information potentially left from a previously open stream. + clearStreamInfo(); + + if ( oParams && oParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( iParams && iParams->nChannels < 1 ) { + errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( oParams == NULL && iParams == NULL ) { + errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; + error( RtAudioError::INVALID_USE ); + return; + } + + if ( formatBytes(format) == 0 ) { + errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; + error( RtAudioError::INVALID_USE ); + return; + } + + 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( RtAudioError::INVALID_USE ); + return; + } + } + + unsigned int iChannels = 0; + if ( iParams ) { + iChannels = iParams->nChannels; + if ( iParams->deviceId >= nDevices ) { + errorText_ = "RtApi::openStream: input device parameter value is invalid."; + error( RtAudioError::INVALID_USE ); + return; + } + } + + bool result; + + if ( oChannels > 0 ) { + + result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + if ( iChannels > 0 ) { + + result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, + sampleRate, format, bufferFrames, options ); + if ( result == false ) { + if ( oChannels > 0 ) closeStream(); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.callbackInfo.callback = (void *) callback; + stream_.callbackInfo.userData = userData; + stream_.callbackInfo.errorCallback = (void *) errorCallback; + + if ( options ) options->numberOfBuffers = stream_.nBuffers; + stream_.state = STREAM_STOPPED; +} + +unsigned int RtApi :: getDefaultInputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +unsigned int RtApi :: getDefaultOutputDevice( void ) +{ + // Should be implemented in subclasses if possible. + return 0; +} + +void RtApi :: closeStream( void ) +{ + // MUST be implemented in subclasses! + return; +} + +bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, + unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, + RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, + RtAudio::StreamOptions * /*options*/ ) +{ + // MUST be implemented in subclasses! + return FAILURE; +} + +void RtApi :: tickStreamTime( void ) +{ + // Subclasses that do not provide their own implementation of + // getStreamTime should call this function once per buffer I/O to + // provide basic stream time support. + + stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); + +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +long RtApi :: getStreamLatency( void ) +{ + verifyStream(); + + long totalLatency = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + totalLatency = stream_.latency[0]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + totalLatency += stream_.latency[1]; + + return totalLatency; +} + +double RtApi :: getStreamTime( void ) +{ + verifyStream(); + +#if defined( HAVE_GETTIMEOFDAY ) + // Return a very accurate estimate of the stream time by + // adding in the elapsed time since the last tick. + struct timeval then; + struct timeval now; + + if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + return stream_.streamTime; + + gettimeofday( &now, NULL ); + then = stream_.lastTickTimestamp; + return stream_.streamTime + + ((now.tv_sec + 0.000001 * now.tv_usec) - + (then.tv_sec + 0.000001 * then.tv_usec)); +#else + return stream_.streamTime; +#endif +} + +void RtApi :: setStreamTime( double time ) +{ + verifyStream(); + + if ( time >= 0.0 ) + stream_.streamTime = time; +#if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); +#endif +} + +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + + +// *************************************************** // +// +// OS/API-specific methods. +// +// *************************************************** // + +#if defined(__MACOSX_CORE__) + +// The OS X CoreAudio API is designed to use a separate callback +// procedure for each of its audio devices. A single RtAudio duplex +// stream using two different devices is supported here, though it +// cannot be guaranteed to always behave correctly because we cannot +// synchronize these two callbacks. +// +// A property listener is installed for over/underrun information. +// However, no functionality is currently provided to allow property +// listeners to trigger user handlers because it is unclear what could +// be done if a critical stream parameter (buffer size, sample rate, +// device disconnect) notification arrived. The listeners entail +// quite a bit of extra code and most likely, a user program wouldn't +// be prepared for the result anyway. However, we do provide a flag +// to the client callback function to inform of an over/underrun. + +// A structure to hold various information related to the CoreAudio API +// implementation. +struct CoreHandle { + AudioDeviceID id[2]; // device ids +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceIOProcID procId[2]; +#endif + UInt32 iStream[2]; // device stream index (or first if using multiple) + UInt32 nStreams[2]; // number of streams to use + bool xrun[2]; + char *deviceBuffer; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + CoreHandle() + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiCore:: RtApiCore() +{ +#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER ) + // This is a largely undocumented but absolutely necessary + // requirement starting with OS-X 10.6. If not called, queries and + // updates to various audio device properties are not handled + // correctly. + CFRunLoopRef theRunLoop = NULL; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop); + if ( result != noErr ) { + errorText_ = "RtApiCore::RtApiCore: error setting run loop property!"; + error( RtAudioError::WARNING ); + } +#endif +} + +RtApiCore :: ~RtApiCore() +{ + // The subclass destructor gets called before the base class + // destructor, so close an existing stream before deallocating + // apiDeviceId memory. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiCore :: getDeviceCount( void ) +{ + // Find out how many audio devices there are, if any. + UInt32 dataSize; + AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; + error( RtAudioError::WARNING ); + return 0; + } + + return dataSize / sizeof( AudioDeviceID ); +} + +unsigned int RtApiCore :: getDefaultInputDevice( void ) +{ + unsigned int nDevices = getDeviceCount(); + if ( nDevices <= 1 ) return 0; + + AudioDeviceID id; + UInt32 dataSize = sizeof( AudioDeviceID ); + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; + error( RtAudioError::WARNING ); + return 0; + } + + dataSize *= nDevices; + AudioDeviceID deviceList[ nDevices ]; + property.mSelector = kAudioHardwarePropertyDevices; + result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; + error( RtAudioError::WARNING ); + return 0; + } + + for ( unsigned int i=0; i= nDevices ) { + errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; + error( RtAudioError::WARNING ); + return info; + } + + AudioDeviceID id = deviceList[ device ]; + + // Get the device name. + info.name.erase(); + CFStringRef cfname; + dataSize = sizeof( CFStringRef ); + property.mSelector = kAudioObjectPropertyManufacturer; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + int length = CFStringGetLength(cfname); + char *mname = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)mname, strlen(mname) ); + info.name.append( ": " ); + CFRelease( cfname ); + free(mname); + + property.mSelector = kAudioObjectPropertyName; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() ); + length = CFStringGetLength(cfname); + char *name = (char *)malloc(length * 3 + 1); +#if defined( UNICODE ) || defined( _UNICODE ) + CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8); +#else + CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding()); +#endif + info.name.append( (const char *)name, strlen(name) ); + CFRelease( cfname ); + free(name); + + // Get the output stream "configuration". + AudioBufferList *bufferList = nil; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + property.mScope = kAudioDevicePropertyScopeOutput; + // property.mElement = kAudioObjectPropertyElementWildcard; + dataSize = 0; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if ( result != noErr || dataSize == 0 ) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get output channel information. + unsigned int i, nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // Get the input stream "configuration". + property.mScope = kAudioDevicePropertyScopeInput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; + error( RtAudioError::WARNING ); + return info; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get input channel information. + nStreams = bufferList->mNumberBuffers; + for ( i=0; imBuffers[i].mNumberChannels; + free( bufferList ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Probe the device sample rates. + bool isInput = false; + if ( info.outputChannels == 0 ) isInput = true; + + // Determine the supported sample rates. + property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates; + if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != kAudioHardwareNoError || dataSize == 0 ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + UInt32 nRanges = dataSize / sizeof( AudioValueRange ); + AudioValueRange rangeList[ nRanges ]; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList ); + if ( result != kAudioHardwareNoError ) { + errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // The sample rate reporting mechanism is a bit of a mystery. It + // seems that it can either return individual rates or a range of + // rates. I assume that if the min / max range values are the same, + // then that represents a single supported rate and if the min / max + // range values are different, the device supports an arbitrary + // range of values (though there might be multiple ranges, so we'll + // use the most conservative range). + Float64 minimumRate = 1.0, maximumRate = 10000000000.0; + bool haveValueRange = false; + info.sampleRates.clear(); + for ( UInt32 i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = tmpSr; + + } else { + haveValueRange = true; + if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum; + if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum; + } + } + + if ( haveValueRange ) { + for ( unsigned int k=0; k= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + // Sort and remove any redundant values + std::sort( info.sampleRates.begin(), info.sampleRates.end() ); + info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() ); + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // CoreAudio always uses 32-bit floating point data for PCM streams. + // Thus, any other "physical" formats supported by the device are of + // no interest to the client. + info.nativeFormats = RTAUDIO_FLOAT32; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + return info; +} + +static OSStatus callbackHandler( AudioDeviceID inDevice, + const AudioTimeStamp* /*inNow*/, + const AudioBufferList* inInputData, + const AudioTimeStamp* /*inInputTime*/, + AudioBufferList* outOutputData, + const AudioTimeStamp* /*inOutputTime*/, + void* infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiCore *object = (RtApiCore *) info->object; + if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) + return kAudioHardwareUnspecifiedError; + else + return kAudioHardwareNoError; +} + +static OSStatus xrunListener( AudioObjectID /*inDevice*/, + UInt32 nAddresses, + const AudioObjectPropertyAddress properties[], + void* handlePointer ) +{ + CoreHandle *handle = (CoreHandle *) handlePointer; + for ( UInt32 i=0; ixrun[1] = true; + else + handle->xrun[0] = true; + } + } + + return kAudioHardwareNoError; +} + +static OSStatus rateListener( AudioObjectID inDevice, + UInt32 /*nAddresses*/, + const AudioObjectPropertyAddress /*properties*/[], + void* ratePointer ) +{ + Float64 *rate = (Float64 *) ratePointer; + UInt32 dataSize = sizeof( Float64 ); + AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate ); + return kAudioHardwareNoError; +} + +bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + AudioDeviceID deviceList[ nDevices ]; + UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, + 0, NULL, &dataSize, (void *) &deviceList ); + if ( result != noErr ) { + errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; + return FAILURE; + } + + AudioDeviceID id = deviceList[ device ]; + + // Setup for stream mode. + bool isInput = false; + if ( mode == INPUT ) { + isInput = true; + property.mScope = kAudioDevicePropertyScopeInput; + } + else + property.mScope = kAudioDevicePropertyScopeOutput; + + // Get the stream "configuration". + AudioBufferList *bufferList = nil; + dataSize = 0; + property.mSelector = kAudioDevicePropertyStreamConfiguration; + result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize ); + if ( result != noErr || dataSize == 0 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Allocate the AudioBufferList. + bufferList = (AudioBufferList *) malloc( dataSize ); + if ( bufferList == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; + return FAILURE; + } + + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList ); + if (result != noErr || dataSize == 0) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Search for one or more streams that contain the desired number of + // channels. CoreAudio devices can have an arbitrary number of + // streams and each stream can have an arbitrary number of channels. + // For each stream, a single buffer of interleaved samples is + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; + bool foundStream = false; + + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels >= channels + offsetCounter ) { + firstStream = iStream; + channelOffset = offsetCounter; + foundStream = true; + break; + } + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. + if ( foundStream == false ) { + monoMode = true; + offsetCounter = firstChannel; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; + } + } + + free( bufferList ); + + // Determine the buffer size. + AudioValueRange bufferRange; + dataSize = sizeof( AudioValueRange ); + property.mSelector = kAudioDevicePropertyBufferFrameSizeRange; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; + else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; + + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. + UInt32 theSize = (UInt32) *bufferSize; + dataSize = sizeof( UInt32 ); + property.mSelector = kAudioDevicePropertyBufferFrameSize; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize ); + + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + *bufferSize = theSize; + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + + // Try to set "hog" mode ... it's not clear to me this is working. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) { + pid_t hog_pid; + dataSize = sizeof( hog_pid ); + property.mSelector = kAudioDevicePropertyHogMode; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( hog_pid != getpid() ) { + hog_pid = getpid(); + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + } + + // Check and if necessary, change the sample rate for the device. + Float64 nominalRate; + dataSize = sizeof( Float64 ); + property.mSelector = kAudioDevicePropertyNominalSampleRate; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Only change the sample rate if off by more than 1 Hz. + if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) { + + // Set a property listener for the sample rate change + Float64 reportedRate = 0.0; + AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; + result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + nominalRate = (Float64) sampleRate; + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate ); + if ( result != noErr ) { + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Now wait until the reported nominal rate is what we just set. + UInt32 microCounter = 0; + while ( reportedRate != nominalRate ) { + microCounter += 5000; + if ( microCounter > 5000000 ) break; + usleep( 5000 ); + } + + // Remove the property listener. + AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate ); + + if ( microCounter > 5000000 ) { + errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now set the stream format for all streams. Also, check the + // physical format of the device and change that if necessary. + AudioStreamBasicDescription description; + dataSize = sizeof( AudioStreamBasicDescription ); + property.mSelector = kAudioStreamPropertyVirtualFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the sample rate and data format id. However, only make the + // change if the sample rate is not within 1.0 of the desired + // rate and the format is not linear pcm. + bool updateFormat = false; + if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) { + description.mSampleRate = (Float64) sampleRate; + updateFormat = true; + } + + if ( description.mFormatID != kAudioFormatLinearPCM ) { + description.mFormatID = kAudioFormatLinearPCM; + updateFormat = true; + } + + if ( updateFormat ) { + result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Now check the physical format. + property.mSelector = kAudioStreamPropertyPhysicalFormat; + result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + //std::cout << "Current physical stream format:" << std::endl; + //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl; + //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl; + //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl; + //std::cout << " sample rate = " << description.mSampleRate << std::endl; + + if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) { + description.mFormatID = kAudioFormatLinearPCM; + //description.mSampleRate = (Float64) sampleRate; + AudioStreamBasicDescription testDescription = description; + UInt32 formatFlags; + + // We'll try higher bit rates first and then work our way down. + std::vector< std::pair > physicalFormats; + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger; + physicalFormats.push_back( std::pair( 32, formatFlags ) ); + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair( 32, formatFlags ) ); + physicalFormats.push_back( std::pair( 24, formatFlags ) ); // 24-bit packed + formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh ); + physicalFormats.push_back( std::pair( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low + formatFlags |= kAudioFormatFlagIsAlignedHigh; + physicalFormats.push_back( std::pair( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high + formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat; + physicalFormats.push_back( std::pair( 16, formatFlags ) ); + physicalFormats.push_back( std::pair( 8, formatFlags ) ); + + bool setPhysicalFormat = false; + for( unsigned int i=0; iflags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our CoreHandle structure for the stream. + CoreHandle *handle = 0; + if ( stream_.apiHandle == 0 ) { + try { + handle = new CoreHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->condition, NULL ) ) { + errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + } + else + handle = (CoreHandle *) stream_.apiHandle; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; + handle->id[mode] = id; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) ); + memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + // If possible, we will make use of the CoreAudio stream buffers as + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) { + if ( streamCount > 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) + // Only one callback procedure per device. + stream_.mode = DUPLEX; + else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() + result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif + if ( result != noErr ) { + errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; + errorText_ = errorStream_.str(); + goto error; + } + if ( stream_.mode == OUTPUT && mode == INPUT ) + stream_.mode = DUPLEX; + else + stream_.mode = mode; + } + + // Setup the device property listener for over/underload. + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle ); + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiCore :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + } + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[0], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); +#endif + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + if (handle) { + AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, + kAudioObjectPropertyScopeGlobal, + kAudioObjectPropertyElementMaster }; + + property.mSelector = kAudioDeviceProcessorOverload; + property.mScope = kAudioObjectPropertyScopeGlobal; + if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { + errorText_ = "RtApiCore::closeStream(): error removing property listener!"; + error( RtAudioError::WARNING ); + } + } + if ( stream_.state == STREAM_RUNNING ) + AudioDeviceStop( handle->id[1], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() + AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); +#endif + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + // Destroy pthread condition variable. + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiCore :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiCore::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + result = AudioDeviceStart( handle->id[0], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || + ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { + + result = AudioDeviceStart( handle->id[1], callbackHandler ); + if ( result != noErr ) { + errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + if ( result == noErr ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiCore :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + OSStatus result = noErr; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + + 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] ) ) { + + 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: + if ( result == noErr ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiCore :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is better to handle it this way because the +// callbackEvent() function probably should return before the AudioDeviceStop() +// function is called. +static void *coreStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiCore *object = (RtApiCore *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + +bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, + const AudioBufferList *inBufferList, + const AudioBufferList *outBufferList ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + CoreHandle *handle = (CoreHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, coreStopStream, info ); + else // external call to stopStream() + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + AudioDeviceID outputDevice = handle->id[0]; + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream or duplex mode AND the input/output devices are + // different AND this function is called for the input device. + if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + if ( handle->nStreams[0] == 1 ) { + memset( outBufferList->mBuffers[handle->iStream[0]].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + else { // fill multiple streams with zeros + for ( unsigned int i=0; inStreams[0]; i++ ) { + memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, + 0, + outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); + } + } + } + else if ( handle->nStreams[0] == 1 ) { + if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer + convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], stream_.convertInfo[0] ); + } + else { // copy from user buffer + memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], + outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); + } + } + else { // fill multiple streams + Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + inBuffer = (Float32 *) stream_.deviceBuffer; + } + + if ( stream_.deviceInterleaved[0] == false ) { // mono mode + UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, + (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); + } + } + else { // fill multiple multi-channel streams with interleaved data + UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; + Float32 *out, *in; + + bool inInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 inChannels = stream_.nUserChannels[0]; + if ( stream_.doConvertBuffer[0] ) { + inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + inChannels = stream_.nDeviceChannels[0]; + } + + if ( inInterleaved ) inOffset = 1; + else inOffset = stream_.bufferSize; + + channelsLeft = inChannels; + for ( unsigned int i=0; inStreams[0]; i++ ) { + in = inBuffer; + out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; + streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; + + outJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[0] > 0 ) { + streamChannels -= stream_.channelOffset[0]; + outJump = stream_.channelOffset[0]; + out += outJump; + } + + // Account for possible unfilled channels at end of the last stream + if ( streamChannels > channelsLeft ) { + outJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine input buffer offsets and skips + if ( inInterleaved ) { + inJump = inChannels; + in += inChannels - channelsLeft; + } + else { + inJump = 1; + in += (inChannels - channelsLeft) * inOffset; + } + + for ( unsigned int i=0; idrainCounter ) { + handle->drainCounter++; + goto unlock; + } + + AudioDeviceID inputDevice; + inputDevice = handle->id[1]; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { + + if ( handle->nStreams[1] == 1 ) { + if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer + convertBuffer( stream_.userBuffer[1], + (char *) inBufferList->mBuffers[handle->iStream[1]].mData, + stream_.convertInfo[1] ); + } + else { // copy to user buffer + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + } + } + else { // read from multiple streams + Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; + if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; + + if ( stream_.deviceInterleaved[1] == false ) { // mono mode + UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; + for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); + } + } + else { // read from multiple multi-channel streams + UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; + Float32 *out, *in; + + bool outInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 outChannels = stream_.nUserChannels[1]; + if ( stream_.doConvertBuffer[1] ) { + outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode + outChannels = stream_.nDeviceChannels[1]; + } + + if ( outInterleaved ) outOffset = 1; + else outOffset = stream_.bufferSize; + + channelsLeft = outChannels; + for ( unsigned int i=0; inStreams[1]; i++ ) { + out = outBuffer; + in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; + streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; + + inJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[1] > 0 ) { + streamChannels -= stream_.channelOffset[1]; + inJump = stream_.channelOffset[1]; + in += inJump; + } + + // Account for possible unread channels at end of the last stream + if ( streamChannels > channelsLeft ) { + inJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine output buffer offsets and skips + if ( outInterleaved ) { + outJump = outChannels; + out += outChannels - channelsLeft; + } + else { + outJump = 1; + out += (outChannels - channelsLeft) * outOffset; + } + + for ( unsigned int i=0; i +#include +#include + +// A structure to hold various information related to the Jack API +// implementation. +struct JackHandle { + jack_client_t *client; + jack_port_t **ports[2]; + std::string deviceName[2]; + bool xrun[2]; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + JackHandle() + :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +static void jackSilentError( const char * ) {}; + +RtApiJack :: RtApiJack() + :shouldAutoconnect_(true) { + // Nothing to do here. +#if !defined(__RTAUDIO_DEBUG__) + // Turn off Jack's internal error reporting. + jack_set_error_function( &jackSilentError ); +#endif +} + +RtApiJack :: ~RtApiJack() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiJack :: getDeviceCount( void ) +{ + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); + if ( client == 0 ) return 0; + + const char **ports; + std::string port, previousPort; + unsigned int nChannels = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nChannels ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon + 1 ); + if ( port != previousPort ) { + nDevices++; + previousPort = port; + } + } + } while ( ports[++nChannels] ); + free( ports ); + } + + jack_client_close( client ); + return nDevices; +} + +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; + error( RtAudioError::WARNING ); + return info; + } + + const char **ports; + std::string port, previousPort; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) info.name = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + jack_client_close( client ); + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // Get the current jack server sample rate. + info.sampleRates.clear(); + + info.preferredSampleRate = jack_get_sample_rate( client ); + info.sampleRates.push_back( info.preferredSampleRate ); + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.outputChannels = nChannels; + } + + // Jack "output ports" equal RtAudio input channels. + nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.inputChannels = nChannels; + } + + if ( info.outputChannels == 0 && info.inputChannels == 0 ) { + jack_client_close(client); + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtAudioError::WARNING ); + return info; + } + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; + + // Jack doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + jack_client_close(client); + info.probed = true; + return info; +} + +static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + + return 0; +} + +// This function will be called by a spawned thread when the Jack +// server signals that it is shutting down. It is necessary to handle +// it this way because the jackShutdown() function must return before +// the jack_deactivate() function (in closeStream()) will return. +static void *jackCloseStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->closeStream(); + + pthread_exit( NULL ); +} +static void jackShutdown( void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; + + // Check current stream state. If stopped, then we'll assume this + // was called as a result of a call to RtApiJack::stopStream (the + // deactivation of a client handle causes this function to be called). + // If not, we'll assume the Jack server is shutting down or some + // other problem occurred and we should close the stream. + if ( object->isStreamRunning() == false ) return; + + ThreadHandle threadId; + pthread_create( &threadId, NULL, jackCloseStream, info ); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; +} + +static int jackXrun( void *infoPointer ) +{ + JackHandle *handle = (JackHandle *) infoPointer; + + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; + + return 0; +} + +bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Look for jack server and try to become a client (only do once per stream). + jack_client_t *client = 0; + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption; + jack_status_t *status = NULL; + if ( options && !options->streamName.empty() ) + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); + else + client = jack_client_open( "RtApiJack", jackoptions, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + } + else { + // The handle must have been created on an earlier pass. + client = handle->client; + } + + const char **ports; + std::string port, previousPort, deviceName; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) deviceName = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } + + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + unsigned long flag = JackPortIsInput; + if ( mode == INPUT ) flag = JackPortIsOutput; + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + } + + // Compare the jack ports for specified client to the requested number of channels. + if ( nChannels < (channels + firstChannel) ) { + errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check the jack server sample rate. + unsigned int jackRate = jack_get_sample_rate( client ); + if ( sampleRate != jackRate ) { + jack_client_close( client ); + errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = jackRate; + + // Get the latency of the JACK port. + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports[ firstChannel ] ) { + // Added by Ge Wang + jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); + // the range (usually the min and max are equal) + jack_latency_range_t latrange; latrange.min = latrange.max = 0; + // get the latency range + jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange ); + // be optimistic, use the min! + stream_.latency[mode] = latrange.min; + //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + } + free( ports ); + + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; + + // Get the buffer size. The buffer size and number of buffers + // (periods) is set when the jack server is started. + stream_.bufferSize = (int) jack_get_buffer_size( client ); + *bufferSize = stream_.bufferSize; + + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our JackHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new JackHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; + goto error; + } + + if ( pthread_cond_init(&handle->condition, NULL) ) { + errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + handle->client = client; + } + handle->deviceName[mode] = deviceName; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + if ( mode == OUTPUT ) + bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + else { // mode == INPUT + bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); + if ( bufferBytes < bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate memory for the Jack ports (channels) identifiers. + handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); + if ( handle->ports[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; + goto error; + } + + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; + + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up the stream for output. + stream_.mode = DUPLEX; + else { + stream_.mode = mode; + jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); + jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); + } + + // Register our ports. + char label[64]; + if ( mode == OUTPUT ) { + for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); + } + } + else { + for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + } + } + + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + + if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false; + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); + + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiJack :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { + + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); + + jack_client_close( handle->client ); + } + + if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiJack :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; + goto unlock; + } + + const char **ports; + + // Get the list of available ports. + if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; + goto unlock; + } + + // Now make the port connections. Since RtAudio wasn't designed to + // allow the user to select particular channels of a device, we'll + // just open the first "nChannels" ports with offset. + for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting output ports!"; + goto unlock; + } + } + free(ports); + } + + if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; + goto unlock; + } + + // Now make the port connections. See note above. + for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting input ports!"; + goto unlock; + } + } + free(ports); + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + if ( result == 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiJack :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + } + + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; +} + +void RtApiJack :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the jack_deactivate() +// function will return. +static void *jackStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiJack *object = (RtApiJack *) info->object; + + object->stopStream(); + pthread_exit( NULL ); +} + +bool RtApiJack :: callbackEvent( unsigned long nframes ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + ThreadHandle threadId; + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) + pthread_create( &threadId, NULL, jackStopStream, info ); + else + pthread_cond_signal( &handle->condition ); + return SUCCESS; + } + + // Invoke user callback first, to get fresh output data. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + ThreadHandle id; + pthread_create( &id, NULL, jackStopStream, info ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // no buffer conversion + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + if ( stream_.doConvertBuffer[1] ) { + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); + } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + else { // no buffer conversion + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); + } + } + } + + unlock: + RtApi::tickStreamTime(); + return SUCCESS; +} + //******************** End of __UNIX_JACK__ *********************// +#endif + +#if defined(__WINDOWS_ASIO__) // ASIO API on Windows + +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS-X CoreAudio and Linux +// Jack. The primary constraint with ASIO is that it only allows +// access to a single driver at a time. Thus, it is not possible to +// have more than one simultaneous RtAudio stream. +// +// This implementation also requires a number of external ASIO files +// and a few global variables. The ASIO callback scheme does not +// allow for the passing of user data, so we must create a global +// pointer to our callbackInfo structure. +// +// On unix systems, we make use of a pthread condition variable. +// Since there is no equivalent in Windows, I hacked something based +// on information found in +// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. + +#include "asiosys.h" +#include "asio.h" +#include "iasiothiscallresolver.h" +#include "asiodrivers.h" +#include + +static AsioDrivers drivers; +static ASIOCallbacks asioCallbacks; +static ASIODriverInfo driverInfo; +static CallbackInfo *asioCallbackInfo; +static bool asioXRun; + +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; + + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; + +// Function declarations (definitions at end of section) +static const char* getAsioErrorString( ASIOError result ); +static void sampleRateChanged( ASIOSampleRate sRate ); +static long asioMessages( long selector, long value, void* message, double* opt ); + +RtApiAsio :: RtApiAsio() +{ + // ASIO cannot run on a multi-threaded appartment. You can call + // CoInitialize beforehand, but it must be for appartment threading + // (in which case, CoInitilialize will return S_FALSE here). + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( FAILED(hr) ) { + errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; + error( RtAudioError::WARNING ); + } + coInitialized_ = true; + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} + +RtApiAsio :: ~RtApiAsio() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); +} + +unsigned int RtApiAsio :: getDeviceCount( void ) +{ + return (unsigned int) drivers.asioGetNumDev(); +} + +RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + // If a stream is already open, we cannot probe other devices. Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.name = driverName; + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + info.outputChannels = outputChannels; + info.inputChannels = inputChannels; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } + } + + // 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( RtAudioError::WARNING ); + return info; + } + + 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; + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) + info.nativeFormats |= RTAUDIO_SINT24; + + if ( info.outputChannels > 0 ) + if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true; + if ( info.inputChannels > 0 ) + if ( getDefaultInputDevice() == device ) info.isDefaultInput = true; + + info.probed = true; + drivers.removeCurrentDriver(); + return info; +} + +static void bufferSwitch( long index, ASIOBool /*processNow*/ ) +{ + RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; + object->callbackEvent( index ); +} + +void RtApiAsio :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; isaveDeviceInfo(); + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // keep them before any "goto error", they are used for error cleanup + goto device boundary checks + bool buffersAllocated = false; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + unsigned int nChannels; + + + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || + ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; + errorText_ = errorStream_.str(); + goto error; + } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + // Get the current sample rate + ASIOSampleRate currentRate; + result = ASIOGetSampleRate( ¤tRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; + errorText_ = errorStream_.str(); + goto error; + } + + // Set the sample rate only if necessary + if ( currentRate != sampleRate ) { + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + 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 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; + errorText_ = errorStream_.str(); + goto error; + } + + // 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; + } + else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true; + } + + if ( stream_.deviceFormat[mode] == 0 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + 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 ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; + errorText_ = errorStream_.str(); + goto error; + } + + if ( isDuplexInput ) { + // When this is the duplex input (output was opened before), then we have to use the same + // buffersize as the output, because it might use the preferred buffer size, which most + // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, + // So instead of throwing an error, make them equal. The caller uses the reference + // to the "bufferSize" param as usual to set up processing buffers. + + *bufferSize = stream_.bufferSize; + + } else { + if ( *bufferSize == 0 ) *bufferSize = preferSize; + else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } + + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; + + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } + } + + *bufferSize = ( (unsigned int)1 << min_delta_num ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + } + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + } + } + + /* + // we don't use it anymore, see above! + // Just left it here for the case... + if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + goto error; + } + */ + + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // ASIO always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; + + // Allocate, if necessary, our AsioHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new AsioHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + handle->bufferInfos = 0; + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + if ( mode == INPUT && stream_.mode == OUTPUT ) { + ASIODisposeBuffers(); + if ( handle->bufferInfos ) free( handle->bufferInfos ); + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + unsigned int i; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + if ( handle->bufferInfos == NULL ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } + + ASIOBufferInfo *infos; + infos = handle->bufferInfos; + for ( i=0; iisInput = ASIOFalse; + infos->channelNum = i + stream_.channelOffset[0]; + infos->buffers[0] = infos->buffers[1] = 0; + } + for ( i=0; iisInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; + } + + // prepare for callbacks + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.mode = isDuplexInput ? DUPLEX : mode; + + // store this class instance before registering callbacks, that are going to use it + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + + // Set up the ASIO callback structure and create the ASIO data buffers. + asioCallbacks.bufferSwitch = &bufferSwitch; + asioCallbacks.sampleRateDidChange = &sampleRateChanged; + asioCallbacks.asioMessage = &asioMessages; + asioCallbacks.bufferSwitchTimeInfo = NULL; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + if ( result != ASE_OK ) { + // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges + // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver + // in that case, let's be naïve and try that instead + *bufferSize = preferSize; + stream_.bufferSize = *bufferSize; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + } + + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; + errorText_ = errorStream_.str(); + goto error; + } + buffersAllocated = true; + stream_.state = STREAM_STOPPED; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( isDuplexInput && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Determine device latencies + long inputLatency, outputLatency; + result = ASIOGetLatencies( &inputLatency, &outputLatency ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; + errorText_ = errorStream_.str(); + error( RtAudioError::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 ( !isDuplexInput ) { + // the cleanup for error in the duplex input, is done by RtApi::openStream + // So we clean up for single channel only + + if ( buffersAllocated ) + ASIODisposeBuffers(); + + drivers.removeCurrentDriver(); + + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + + delete handle; + stream_.apiHandle = 0; + } + + + if ( stream_.userBuffer[mode] ) { + free( stream_.userBuffer[mode] ); + stream_.userBuffer[mode] = 0; + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + } + + return FAILURE; +}//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +void RtApiAsio :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +bool stopThreadCalled = false; + +void RtApiAsio :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; + errorText_ = errorStream_.str(); + goto unlock; + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + asioXRun = false; + + unlock: + stopThreadCalled = false; + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + } + + stream_.state = STREAM_STOPPED; + + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } + + if ( result == ASE_OK ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAsio :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + // The following lines were commented-out because some behavior was + // noted where the device buffers need to be zeroed to avoid + // continuing sound, even when the device buffers are completely + // disposed. So now, calling abort is the same as calling stop. + // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // handle->drainCounter = 2; + stopStream(); +} + +// This function will be called by a spawned thread when the user +// callback function signals that the stream should be stopped or +// aborted. It is necessary to handle it this way because the +// callbackEvent() function must return before the ASIOStop() +// function will return. +static unsigned __stdcall asioStopStream( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAsio *object = (RtApiAsio *) info->object; + + object->stopStream(); + _endthreadex( 0 ); + return 0; +} + +bool RtApiAsio :: callbackEvent( long bufferIndex ) +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return FAILURE; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal if finished. + if ( handle->drainCounter > 3 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else { // spawn a thread to stop the stream + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + } + return SUCCESS; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && asioXRun == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + asioXRun = false; + } + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + unsigned threadId; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, + &stream_.callbackInfo, 0, &threadId ); + return SUCCESS; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); + } + + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } + + } + else { + + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); + } + + } + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + + if (stream_.doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) + memcpy( &stream_.deviceBuffer[j++*bufferBytes], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + } + else { + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } + } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); + } + } + + unlock: + // The following call was suggested by Malte Clasen. While the API + // documentation indicates it should not be required, some device + // drivers apparently do not function correctly without it. + ASIOOutputReady(); + + RtApi::tickStreamTime(); + return SUCCESS; +} + +static void sampleRateChanged( ASIOSampleRate sRate ) +{ + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. + + RtApi *object = (RtApi *) asioCallbackInfo->object; + try { + object->stopStream(); + } + catch ( RtAudioError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } + + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; +} + +static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ ) +{ + long ret = 0; + + switch( selector ) { + case kAsioSelectorSupported: + if ( value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver whether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; +} + +static const char* getAsioErrorString( ASIOError result ) +{ + struct Messages + { + ASIOError value; + const char*message; + }; + + static const Messages m[] = + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; + + for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) + if ( m[i].value == result ) return m[i].message; + + return "Unknown error."; +} + +//******************** End of __WINDOWS_ASIO__ *********************// +#endif + + +#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API + +// Authored by Marcus Tomlinson , April 2014 +// - Introduces support for the Windows WASAPI API +// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required +// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface +// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user + +#ifndef INITGUID + #define INITGUID +#endif +#include +#include +#include +#include + +//============================================================================= + +#define SAFE_RELEASE( objectPtr )\ +if ( objectPtr )\ +{\ + objectPtr->Release();\ + objectPtr = NULL;\ +} + +typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex ); + +//----------------------------------------------------------------------------- + +// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size. +// Therefore we must perform all necessary conversions to user buffers in order to satisfy these +// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to +// provide intermediate storage for read / write synchronization. +class WasapiBuffer +{ +public: + WasapiBuffer() + : buffer_( NULL ), + bufferSize_( 0 ), + inIndex_( 0 ), + outIndex_( 0 ) {} + + ~WasapiBuffer() { + free( buffer_ ); + } + + // sets the length of the internal ring buffer + void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { + free( buffer_ ); + + buffer_ = ( char* ) calloc( bufferSize, formatBytes ); + + bufferSize_ = bufferSize; + inIndex_ = 0; + outIndex_ = 0; + } + + // attempt to push a buffer into the ring buffer at the current "in" index + bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relOutIndex = outIndex_; + unsigned int inIndexEnd = inIndex_ + bufferSize; + if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) { + relOutIndex += bufferSize_; + } + + // "in" index can end on the "out" index but cannot begin at it + if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) { + return false; // not enough space between "in" index and "out" index + } + + // copy buffer from external to internal + int fromZeroSize = inIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromInSize = bufferSize - fromZeroSize; + + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) ); + memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) ); + memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) ); + memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) ); + memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) ); + memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) ); + memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) ); + break; + } + + // update "in" index + inIndex_ += bufferSize; + inIndex_ %= bufferSize_; + + return true; + } + + // attempt to pull a buffer from the ring buffer from the current "out" index + bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format ) + { + if ( !buffer || // incoming buffer is NULL + bufferSize == 0 || // incoming buffer has no data + bufferSize > bufferSize_ ) // incoming buffer too large + { + return false; + } + + unsigned int relInIndex = inIndex_; + unsigned int outIndexEnd = outIndex_ + bufferSize; + if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) { + relInIndex += bufferSize_; + } + + // "out" index can begin at and end on the "in" index + if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) { + return false; // not enough space between "out" index and "in" index + } + + // copy buffer from internal to external + int fromZeroSize = outIndex_ + bufferSize - bufferSize_; + fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize; + int fromOutSize = bufferSize - fromZeroSize; + + switch( format ) + { + case RTAUDIO_SINT8: + memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) ); + memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) ); + memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) ); + memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) ); + memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) ); + memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) ); + memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) ); + break; + } + + // update "out" index + outIndex_ += bufferSize; + outIndex_ %= bufferSize_; + + return true; + } + +private: + char* buffer_; + unsigned int bufferSize_; + unsigned int inIndex_; + unsigned int outIndex_; +}; + +//----------------------------------------------------------------------------- + +// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate +// between HW and the user. The convertBufferWasapi function is used to perform this conversion +// between HwIn->UserIn and UserOut->HwOut during the stream callback loop. +// This sample rate converter works best with conversions between one rate and its multiple. +void convertBufferWasapi( char* outBuffer, + const char* inBuffer, + const unsigned int& channelCount, + const unsigned int& inSampleRate, + const unsigned int& outSampleRate, + const unsigned int& inSampleCount, + unsigned int& outSampleCount, + const RtAudioFormat& format ) +{ + // calculate the new outSampleCount and relative sampleStep + float sampleRatio = ( float ) outSampleRate / inSampleRate; + float sampleRatioInv = ( float ) 1 / sampleRatio; + float sampleStep = 1.0f / sampleRatio; + float inSampleFraction = 0.0f; + + outSampleCount = ( unsigned int ) std::roundf( inSampleCount * sampleRatio ); + + // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate + if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv ) + { + // frame-by-frame, copy each relative input sample into it's corresponding output sample + for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) + { + unsigned int inSample = ( unsigned int ) inSampleFraction; + + switch ( format ) + { + case RTAUDIO_SINT8: + memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) ); + break; + case RTAUDIO_SINT16: + memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) ); + break; + case RTAUDIO_SINT24: + memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) ); + break; + case RTAUDIO_SINT32: + memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) ); + break; + case RTAUDIO_FLOAT32: + memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) ); + break; + case RTAUDIO_FLOAT64: + memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) ); + break; + } + + // jump to next in sample + inSampleFraction += sampleStep; + } + } + else // else interpolate + { + // frame-by-frame, copy each relative input sample into it's corresponding output sample + for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) + { + unsigned int inSample = ( unsigned int ) inSampleFraction; + float inSampleDec = inSampleFraction - inSample; + unsigned int frameInSample = inSample * channelCount; + unsigned int frameOutSample = outSample * channelCount; + + switch ( format ) + { + case RTAUDIO_SINT8: + { + for ( unsigned int channel = 0; channel < channelCount; channel++ ) + { + char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ]; + char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ]; + char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec ); + ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; + } + break; + } + case RTAUDIO_SINT16: + { + for ( unsigned int channel = 0; channel < channelCount; channel++ ) + { + short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ]; + short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ]; + short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec ); + ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; + } + break; + } + case RTAUDIO_SINT24: + { + for ( unsigned int channel = 0; channel < channelCount; channel++ ) + { + int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt(); + int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt(); + int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec ); + ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; + } + break; + } + case RTAUDIO_SINT32: + { + for ( unsigned int channel = 0; channel < channelCount; channel++ ) + { + int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ]; + int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ]; + int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec ); + ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; + } + break; + } + case RTAUDIO_FLOAT32: + { + for ( unsigned int channel = 0; channel < channelCount; channel++ ) + { + float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ]; + float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ]; + float sampleDiff = ( toSample - fromSample ) * inSampleDec; + ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; + } + break; + } + case RTAUDIO_FLOAT64: + { + for ( unsigned int channel = 0; channel < channelCount; channel++ ) + { + double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ]; + double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ]; + double sampleDiff = ( toSample - fromSample ) * inSampleDec; + ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff; + } + break; + } + } + + // jump to next in sample + inSampleFraction += sampleStep; + } + } +} + +//----------------------------------------------------------------------------- + +// A structure to hold various information related to the WASAPI implementation. +struct WasapiHandle +{ + IAudioClient* captureAudioClient; + IAudioClient* renderAudioClient; + IAudioCaptureClient* captureClient; + IAudioRenderClient* renderClient; + HANDLE captureEvent; + HANDLE renderEvent; + + WasapiHandle() + : captureAudioClient( NULL ), + renderAudioClient( NULL ), + captureClient( NULL ), + renderClient( NULL ), + captureEvent( NULL ), + renderEvent( NULL ) {} +}; + +//============================================================================= + +RtApiWasapi::RtApiWasapi() + : coInitialized_( false ), deviceEnumerator_( NULL ) +{ + // WASAPI can run either apartment or multi-threaded + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) + coInitialized_ = true; + + // Instantiate device enumerator + hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL, + CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ), + ( void** ) &deviceEnumerator_ ); + + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator"; + error( RtAudioError::DRIVER_ERROR ); + } +} + +//----------------------------------------------------------------------------- + +RtApiWasapi::~RtApiWasapi() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); + + SAFE_RELEASE( deviceEnumerator_ ); + + // If this object previously called CoInitialize() + if ( coInitialized_ ) + CoUninitialize(); +} + +//============================================================================= + +unsigned int RtApiWasapi::getDeviceCount( void ) +{ + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + + // Count capture devices + errorText_.clear(); + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count."; + goto Exit; + } + +Exit: + // release all references + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + + if ( errorText_.empty() ) + return captureDeviceCount + renderDeviceCount; + + error( RtAudioError::DRIVER_ERROR ); + return 0; +} + +//----------------------------------------------------------------------------- + +RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + std::string defaultDeviceName; + bool isCaptureDevice = false; + + PROPVARIANT deviceNameProp; + PROPVARIANT defaultDeviceNameProp; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + IMMDevice* defaultDevicePtr = NULL; + IAudioClient* audioClient = NULL; + IPropertyStore* devicePropStore = NULL; + IPropertyStore* defaultDevicePropStore = NULL; + + WAVEFORMATEX* deviceFormat = NULL; + WAVEFORMATEX* closestMatchFormat = NULL; + + // probed + info.probed = false; + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index."; + errorType = RtAudioError::INVALID_USE; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle."; + goto Exit; + } + isCaptureDevice = true; + } + else { + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle."; + goto Exit; + } + isCaptureDevice = false; + } + + // get default device name + if ( isCaptureDevice ) { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle."; + goto Exit; + } + } + else { + hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle."; + goto Exit; + } + } + + hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store."; + goto Exit; + } + PropVariantInit( &defaultDeviceNameProp ); + + hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); + + // name + hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store."; + goto Exit; + } + + PropVariantInit( &deviceNameProp ); + + hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName."; + goto Exit; + } + + info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); + + // is default + if ( isCaptureDevice ) { + info.isDefaultInput = info.name == defaultDeviceName; + info.isDefaultOutput = false; + } + else { + info.isDefaultInput = false; + info.isDefaultOutput = info.name == defaultDeviceName; + } + + // channel count + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client."; + goto Exit; + } + + hr = audioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + goto Exit; + } + + if ( isCaptureDevice ) { + info.inputChannels = deviceFormat->nChannels; + info.outputChannels = 0; + info.duplexChannels = 0; + } + else { + info.inputChannels = 0; + info.outputChannels = deviceFormat->nChannels; + info.duplexChannels = 0; + } + + // sample rates + info.sampleRates.clear(); + + // allow support for all sample rates as we have a built-in sample rate converter + for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) { + info.sampleRates.push_back( SAMPLE_RATES[i] ); + } + info.preferredSampleRate = deviceFormat->nSamplesPerSec; + + // native format + info.nativeFormats = 0; + + if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) ) + { + if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_FLOAT32; + } + else if ( deviceFormat->wBitsPerSample == 64 ) { + info.nativeFormats |= RTAUDIO_FLOAT64; + } + } + else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM || + ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE && + ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) ) + { + if ( deviceFormat->wBitsPerSample == 8 ) { + info.nativeFormats |= RTAUDIO_SINT8; + } + else if ( deviceFormat->wBitsPerSample == 16 ) { + info.nativeFormats |= RTAUDIO_SINT16; + } + else if ( deviceFormat->wBitsPerSample == 24 ) { + info.nativeFormats |= RTAUDIO_SINT24; + } + else if ( deviceFormat->wBitsPerSample == 32 ) { + info.nativeFormats |= RTAUDIO_SINT32; + } + } + + // probed + info.probed = true; + +Exit: + // release all references + PropVariantClear( &deviceNameProp ); + PropVariantClear( &defaultDeviceNameProp ); + + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + SAFE_RELEASE( defaultDevicePtr ); + SAFE_RELEASE( audioClient ); + SAFE_RELEASE( devicePropStore ); + SAFE_RELEASE( defaultDevicePropStore ); + + CoTaskMemFree( deviceFormat ); + CoTaskMemFree( closestMatchFormat ); + + if ( !errorText_.empty() ) + error( errorType ); + return info; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultOutputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultOutput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +unsigned int RtApiWasapi::getDefaultInputDevice( void ) +{ + for ( unsigned int i = 0; i < getDeviceCount(); i++ ) { + if ( getDeviceInfo( i ).isDefaultInput ) { + return i; + } + } + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiWasapi::closeStream: No open stream to close."; + error( RtAudioError::WARNING ); + return; + } + + if ( stream_.state != STREAM_STOPPED ) + stopStream(); + + // clean up stream memory + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) + + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient ) + SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient ) + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent ); + + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ) + CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent ); + + delete ( WasapiHandle* ) stream_.apiHandle; + stream_.apiHandle = NULL; + + for ( int i = 0; i < 2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + // update stream state + stream_.state = STREAM_CLOSED; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::startStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiWasapi::startStream: The stream is already running."; + error( RtAudioError::WARNING ); + return; + } + + // update stream state + stream_.state = STREAM_RUNNING; + + // create WASAPI stream thread + stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL ); + + if ( !stream_.callbackInfo.thread ) { + errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread."; + error( RtAudioError::THREAD_ERROR ); + } + else { + SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority ); + ResumeThread( ( void* ) stream_.callbackInfo.thread ); + } +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::stopStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::stopStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } + + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; + + // wait until stream thread is stopped + while( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } + + // Wait for the last buffer to play before stopping. + Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); + + // stop capture client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // stop render client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::abortStream( void ) +{ + verifyStream(); + + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiWasapi::abortStream: The stream is already stopped."; + error( RtAudioError::WARNING ); + return; + } + + // inform stream thread by setting stream state to STREAM_STOPPING + stream_.state = STREAM_STOPPING; + + // wait until stream thread is stopped + while ( stream_.state != STREAM_STOPPED ) { + Sleep( 1 ); + } + + // stop capture client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // stop render client if applicable + if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { + HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream."; + error( RtAudioError::DRIVER_ERROR ); + return; + } + } + + // close thread handle + if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { + errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; + error( RtAudioError::THREAD_ERROR ); + return; + } + + stream_.callbackInfo.thread = (ThreadHandle) NULL; +} + +//----------------------------------------------------------------------------- + +bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int* bufferSize, + RtAudio::StreamOptions* options ) +{ + bool methodResult = FAILURE; + unsigned int captureDeviceCount = 0; + unsigned int renderDeviceCount = 0; + + IMMDeviceCollection* captureDevices = NULL; + IMMDeviceCollection* renderDevices = NULL; + IMMDevice* devicePtr = NULL; + WAVEFORMATEX* deviceFormat = NULL; + unsigned int bufferBytes; + stream_.state = STREAM_STOPPED; + + // create API Handle if not already created + if ( !stream_.apiHandle ) + stream_.apiHandle = ( void* ) new WasapiHandle(); + + // Count capture devices + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection."; + goto Exit; + } + + hr = captureDevices->GetCount( &captureDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count."; + goto Exit; + } + + // Count render devices + hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection."; + goto Exit; + } + + hr = renderDevices->GetCount( &renderDeviceCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count."; + goto Exit; + } + + // validate device index + if ( device >= captureDeviceCount + renderDeviceCount ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index."; + goto Exit; + } + + // determine whether index falls within capture or render devices + if ( device >= renderDeviceCount ) { + if ( mode != INPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device."; + goto Exit; + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + hr = captureDevices->Item( device - renderDeviceCount, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + else { + if ( mode != OUTPUT ) { + errorType = RtAudioError::INVALID_USE; + errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device."; + goto Exit; + } + + // retrieve renderAudioClient from devicePtr + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + + hr = renderDevices->Item( device, &devicePtr ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle."; + goto Exit; + } + + hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, + NULL, ( void** ) &renderAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + goto Exit; + } + + hr = renderAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + goto Exit; + } + + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // fill stream data + if ( ( stream_.mode == OUTPUT && mode == INPUT ) || + ( stream_.mode == INPUT && mode == OUTPUT ) ) { + stream_.mode = DUPLEX; + } + else { + stream_.mode = mode; + } + + stream_.device[mode] = device; + stream_.doByteSwap[mode] = false; + stream_.sampleRate = sampleRate; + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 1; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; + stream_.userFormat = format; + stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats; + + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + else + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] || + stream_.nUserChannels != stream_.nDeviceChannels ) + stream_.doConvertBuffer[mode] = true; + else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + if ( stream_.doConvertBuffer[mode] ) + setConvertInfo( mode, 0 ); + + // Allocate necessary internal buffers + bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat ); + + stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 ); + if ( !stream_.userBuffer[mode] ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory."; + goto Exit; + } + + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) + stream_.callbackInfo.priority = 15; + else + stream_.callbackInfo.priority = 0; + + ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback + ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode + + methodResult = SUCCESS; + +Exit: + //clean up + SAFE_RELEASE( captureDevices ); + SAFE_RELEASE( renderDevices ); + SAFE_RELEASE( devicePtr ); + CoTaskMemFree( deviceFormat ); + + // if method failed, close the stream + if ( methodResult == FAILURE ) + closeStream(); + + if ( !errorText_.empty() ) + error( errorType ); + return methodResult; +} + +//============================================================================= + +DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->stopStream(); + + return 0; +} + +DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr ) +{ + if ( wasapiPtr ) + ( ( RtApiWasapi* ) wasapiPtr )->abortStream(); + + return 0; +} + +//----------------------------------------------------------------------------- + +void RtApiWasapi::wasapiThread() +{ + // as this is a new thread, we must CoInitialize it + CoInitialize( NULL ); + + HRESULT hr; + + IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient; + IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient; + HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent; + HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent; + + WAVEFORMATEX* captureFormat = NULL; + WAVEFORMATEX* renderFormat = NULL; + float captureSrRatio = 0.0f; + float renderSrRatio = 0.0f; + WasapiBuffer captureBuffer; + WasapiBuffer renderBuffer; + + // declare local stream variables + RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; + BYTE* streamBuffer = NULL; + unsigned long captureFlags = 0; + unsigned int bufferFrameCount = 0; + unsigned int numFramesPadding = 0; + unsigned int convBufferSize = 0; + bool callbackPushed = false; + bool callbackPulled = false; + bool callbackStopped = false; + int callbackResult = 0; + + // convBuffer is used to store converted buffers between WASAPI and the user + char* convBuffer = NULL; + unsigned int convBuffSize = 0; + unsigned int deviceBuffSize = 0; + + errorText_.clear(); + RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR; + + // Attempt to assign "Pro Audio" characteristic to thread + HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" ); + if ( AvrtDll ) { + DWORD taskIndex = 0; + TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); + AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); + FreeLibrary( AvrtDll ); + } + + // start capture stream if applicable + if ( captureAudioClient ) { + hr = captureAudioClient->GetMixFormat( &captureFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); + + // initialize capture stream according to desire buffer size + float desiredBufferSize = stream_.bufferSize * captureSrRatio; + REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec ); + + if ( !captureClient ) { + hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + desiredBufferPeriod, + desiredBufferPeriod, + captureFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; + goto Exit; + } + + hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ), + ( void** ) &captureClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle."; + goto Exit; + } + + // configure captureEvent to trigger on every available capture buffer + captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !captureEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event."; + goto Exit; + } + + hr = captureAudioClient->SetEventHandle( captureEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; + } + + unsigned int inBufferSize = 0; + hr = captureAudioClient->GetBufferSize( &inBufferSize ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; + goto Exit; + } + + // scale outBufferSize according to stream->user sample rate ratio + unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT]; + inBufferSize *= stream_.nDeviceChannels[INPUT]; + + // set captureBuffer size + captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); + + // reset the capture stream + hr = captureAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; + goto Exit; + } + + // start the capture stream + hr = captureAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream."; + goto Exit; + } + } + + // start render stream if applicable + if ( renderAudioClient ) { + hr = renderAudioClient->GetMixFormat( &renderFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + goto Exit; + } + + renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); + + // initialize render stream according to desire buffer size + float desiredBufferSize = stream_.bufferSize * renderSrRatio; + REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec ); + + if ( !renderClient ) { + hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, + AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + desiredBufferPeriod, + desiredBufferPeriod, + renderFormat, + NULL ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; + goto Exit; + } + + hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ), + ( void** ) &renderClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; + goto Exit; + } + + // configure renderEvent to trigger on every available render buffer + renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); + if ( !renderEvent ) { + errorType = RtAudioError::SYSTEM_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event."; + goto Exit; + } + + hr = renderAudioClient->SetEventHandle( renderEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; + ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; + } + + unsigned int outBufferSize = 0; + hr = renderAudioClient->GetBufferSize( &outBufferSize ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; + goto Exit; + } + + // scale inBufferSize according to user->stream sample rate ratio + unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT]; + outBufferSize *= stream_.nDeviceChannels[OUTPUT]; + + // set renderBuffer size + renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) ); + + // reset the render stream + hr = renderAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream."; + goto Exit; + } + + // start the render stream + hr = renderAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream."; + goto Exit; + } + } + + if ( stream_.mode == INPUT ) { + convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + } + else if ( stream_.mode == OUTPUT ) { + convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ); + } + else if ( stream_.mode == DUPLEX ) { + convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ); + } + + convBuffer = ( char* ) malloc( convBuffSize ); + stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize ); + if ( !convBuffer || !stream_.deviceBuffer ) { + errorType = RtAudioError::MEMORY_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; + goto Exit; + } + + // stream process loop + while ( stream_.state != STREAM_STOPPING ) { + if ( !callbackPulled ) { + // Callback Input + // ============== + // 1. Pull callback buffer from inputBuffer + // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count + // Convert callback buffer to user format + + if ( captureAudioClient ) { + // Pull callback buffer from inputBuffer + callbackPulled = captureBuffer.pullBuffer( convBuffer, + ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ); + + if ( callbackPulled ) { + // Convert callback buffer to user sample rate + convertBufferWasapi( stream_.deviceBuffer, + convBuffer, + stream_.nDeviceChannels[INPUT], + captureFormat->nSamplesPerSec, + stream_.sampleRate, + ( unsigned int ) ( stream_.bufferSize * captureSrRatio ), + convBufferSize, + stream_.deviceFormat[INPUT] ); + + if ( stream_.doConvertBuffer[INPUT] ) { + // Convert callback buffer to user format + convertBuffer( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.convertInfo[INPUT] ); + } + else { + // no further conversion, simple copy deviceBuffer to userBuffer + memcpy( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) ); + } + } + } + else { + // if there is no capture stream, set callbackPulled flag + callbackPulled = true; + } + + // Execute Callback + // ================ + // 1. Execute user callback method + // 2. Handle return value from callback + + // if callback has not requested the stream to stop + if ( callbackPulled && !callbackStopped ) { + // Execute user callback method + callbackResult = callback( stream_.userBuffer[OUTPUT], + stream_.userBuffer[INPUT], + stream_.bufferSize, + getStreamTime(), + captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, + stream_.callbackInfo.userData ); + + // Handle return value from callback + if ( callbackResult == 1 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; + goto Exit; + } + + callbackStopped = true; + } + else if ( callbackResult == 2 ) { + // instantiate a thread to stop this thread + HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); + if ( !threadHandle ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; + goto Exit; + } + else if ( !CloseHandle( threadHandle ) ) { + errorType = RtAudioError::THREAD_ERROR; + errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; + goto Exit; + } + + callbackStopped = true; + } + } + } + + // Callback Output + // =============== + // 1. Convert callback buffer to stream format + // 2. Convert callback buffer to stream sample rate and channel count + // 3. Push callback buffer into outputBuffer + + if ( renderAudioClient && callbackPulled ) { + if ( stream_.doConvertBuffer[OUTPUT] ) { + // Convert callback buffer to stream format + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + + } + + // Convert callback buffer to stream sample rate + convertBufferWasapi( convBuffer, + stream_.deviceBuffer, + stream_.nDeviceChannels[OUTPUT], + stream_.sampleRate, + renderFormat->nSamplesPerSec, + stream_.bufferSize, + convBufferSize, + stream_.deviceFormat[OUTPUT] ); + + // Push callback buffer into outputBuffer + callbackPushed = renderBuffer.pushBuffer( convBuffer, + convBufferSize * stream_.nDeviceChannels[OUTPUT], + stream_.deviceFormat[OUTPUT] ); + } + else { + // if there is no render stream, set callbackPushed flag + callbackPushed = true; + } + + // Stream Capture + // ============== + // 1. Get capture buffer from stream + // 2. Push capture buffer into inputBuffer + // 3. If 2. was successful: Release capture buffer + + if ( captureAudioClient ) { + // if the callback input buffer was not pulled from captureBuffer, wait for next capture event + if ( !callbackPulled ) { + WaitForSingleObject( captureEvent, INFINITE ); + } + + // Get capture buffer from stream + hr = captureClient->GetBuffer( &streamBuffer, + &bufferFrameCount, + &captureFlags, NULL, NULL ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; + goto Exit; + } + + if ( bufferFrameCount != 0 ) { + // Push capture buffer into inputBuffer + if ( captureBuffer.pushBuffer( ( char* ) streamBuffer, + bufferFrameCount * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ) ) + { + // Release capture buffer + hr = captureClient->ReleaseBuffer( bufferFrameCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that capture was unsuccessful + hr = captureClient->ReleaseBuffer( 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + goto Exit; + } + } + } + + // Stream Render + // ============= + // 1. Get render buffer from stream + // 2. Pull next buffer from outputBuffer + // 3. If 2. was successful: Fill render buffer with next buffer + // Release render buffer + + if ( renderAudioClient ) { + // if the callback output buffer was not pushed to renderBuffer, wait for next render event + if ( callbackPulled && !callbackPushed ) { + WaitForSingleObject( renderEvent, INFINITE ); + } + + // Get render buffer from stream + hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; + goto Exit; + } + + hr = renderAudioClient->GetCurrentPadding( &numFramesPadding ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; + goto Exit; + } + + bufferFrameCount -= numFramesPadding; + + if ( bufferFrameCount != 0 ) { + hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; + goto Exit; + } + + // Pull next buffer from outputBuffer + // Fill render buffer with next buffer + if ( renderBuffer.pullBuffer( ( char* ) streamBuffer, + bufferFrameCount * stream_.nDeviceChannels[OUTPUT], + stream_.deviceFormat[OUTPUT] ) ) + { + // Release render buffer + hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + else + { + // Inform WASAPI that render was unsuccessful + hr = renderClient->ReleaseBuffer( 0, 0 ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + goto Exit; + } + } + } + + // if the callback buffer was pushed renderBuffer reset callbackPulled flag + if ( callbackPushed ) { + callbackPulled = false; + // tick stream time + RtApi::tickStreamTime(); + } + + } + +Exit: + // clean up + CoTaskMemFree( captureFormat ); + CoTaskMemFree( renderFormat ); + + free ( convBuffer ); + + CoUninitialize(); + + // update stream state + stream_.state = STREAM_STOPPED; + + if ( errorText_.empty() ) + return; + else + error( errorType ); +} + +//******************** End of __WINDOWS_WASAPI__ *********************// +#endif + + +#if defined(__WINDOWS_DS__) // Windows DirectSound API + +// Modified by Robin Davies, October 2005 +// - Improvements to DirectX pointer chasing. +// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. +// - Auto-call CoInitialize for DSOUND and ASIO platforms. +// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 +// Changed device query structure for RtAudio 4.0.7, January 2010 + +#include +#include +#include +#include +#include + +#if defined(__MINGW32__) + // missing from latest mingw winapi +#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ +#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ +#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ +#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ +#endif + +#define MINIMUM_DEVICE_BUFFER_SIZE 32768 + +#ifdef _MSC_VER // if Microsoft Visual C++ +#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. +#endif + +static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; +} + +// A structure to hold various information related to the DirectSound +// API implementation. +struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; + + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } +}; + +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); + +static const char* getErrorString( int code ); + +static unsigned __stdcall callbackHandler( void *ptr ); + +struct DsDevice { + LPGUID id[2]; + bool validId[2]; + bool found; + std::string name; + + DsDevice() + : found(false) { validId[0] = false; validId[1] = false; } +}; + +struct DsProbeData { + bool isInput; + std::vector* dsDevices; +}; + +RtApiDs :: RtApiDs() +{ + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; +} + +RtApiDs :: ~RtApiDs() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); // balanced call. +} + +// The DirectSound default output is always the first device. +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + return 0; +} + +// The DirectSound default input is always the first input device, +// which is the first capture device enumerated. +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + return 0; +} + +unsigned int RtApiDs :: getDeviceCount( void ) +{ + // Set query flag for previously found devices to false, so that we + // can check for any devices that have disappeared. + for ( unsigned int i=0; i(dsDevices.size()); +} + +RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + if ( dsDevices.size() == 0 ) { + // Force a query of all devices + getDeviceCount(); + if ( dsDevices.size() == 0 ) { + errorText_ = "RtApiDs::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + } + + if ( device >= dsDevices.size() ) { + errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + HRESULT result; + if ( dsDevices[ device ].validId[0] == false ) goto probeInput; + + LPDIRECTSOUND output; + DSCAPS outCaps; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto probeInput; + } + + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto probeInput; + } + + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + + output->Release(); + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + + if ( dsDevices[ device ].validId[1] == false ) { + info.name = dsDevices[ device ].name; + info.probed = true; + return info; + } + + probeInput: + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get input channel information. + info.inputChannels = inCaps.dwChannels; + + // Get sample rate and format information. + std::vector rates; + if ( inCaps.dwChannels >= 2 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 ); + } + } + else if ( inCaps.dwChannels == 1 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 ); + } + } + else info.inputChannels = 0; // technically, this would be an error + + input->Release(); + + if ( info.inputChannels == 0 ) return info; + + // Copy the supported rates to the info structure but avoid duplication. + bool found; + for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + if ( device == 0 ) info.isDefaultInput = true; + + // Copy name and return. + info.name = dsDevices[ device ].name; + info.probed = true; + return info; +} + +bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + return FAILURE; + } + + size_t nDevices = dsDevices.size(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiDs::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + if ( mode == OUTPUT ) { + if ( dsDevices[ device ].validId[0] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + else { // mode == INPUT + if ( dsDevices[ device ].validId[1] == false ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // window. Also, if the application window closes before the + // DirectSound buffer, DirectSound can crash. In the past, I had + // problems when using GetDesktopWindow() but it seems fine now + // (January 2010). I'll leave it commented here. + // HWND hWnd = GetForegroundWindow(); + HWND hWnd = GetDesktopWindow(); + + // Check the numberOfBuffers parameter and limit the lowest value to + // two. This is a judgement call and a value of two is probably too + // low for capture, but it should work for playback. + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; + + // Check the lower range of the user-specified buffer size and set + // (arbitrarily) to a lower bound of 32. + if ( *bufferSize < 32 ) *bufferSize = 32; + + // Create the wave format structure. The data format setting will + // be determined later. + WAVEFORMATEX waveFormat; + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels + firstChannel; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + + // Determine the device buffer size. By default, we'll use the value + // defined above (32K), but we will grow it to make allowances for + // very large software buffer sizes. + DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE; + DWORD dsPointerLeadTime = 0; + + void *ohandle = 0, *bhandle = 0; + HRESULT result; + if ( mode == OUTPUT ) { + + LPDIRECTSOUND output; + result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCAPS outCaps; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check format information. Use 16-bit format unless not + // supported or user requests 8-bit. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && + !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; + + // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. + // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. + result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format + // (since the default is 8-bit, 22 kHz!). Setup the DS primary + // buffer description. + DSBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + + // Obtain the primary buffer + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the primary DS buffer sound format. + result = buffer->SetFormat( &waveFormat ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Setup the secondary DS buffer description. + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; + + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = dsbcaps.dwBufferBytes; + + // Lock the DS buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + ohandle = (void *) output; + bhandle = (void *) buffer; + } + + if ( mode == INPUT ) { + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; + return FAILURE; + } + + // Check format information. Use 16-bit format unless user + // requests 8-bit. + DWORD deviceFormats; + if ( channels + firstChannel == 2 ) { + deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + else { // channel == 1 + deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > dsBufferSize ) + dsBufferSize *= 2; + + // Setup the secondary DS buffer description. + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = dsBufferSize; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the buffer size ... might be different from what we specified. + DSCBCAPS dscbcaps; + dscbcaps.dwSize = sizeof( DSCBCAPS ); + result = buffer->GetCaps( &dscbcaps ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + dsBufferSize = dscbcaps.dwBufferBytes; + + // NOTE: We could have a problem here if this is a duplex stream + // and the play and capture hardware buffer sizes are different + // (I'm actually not sure if that is a problem or not). + // Currently, we are not verifying that. + + // Lock the capture buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Zero the buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + ohandle = (void *) input; + bhandle = (void *) buffer; + } + + // Set various stream parameters + DsHandle *handle = 0; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.nUserChannels[mode] = channels; + stream_.bufferSize = *bufferSize; + stream_.channelOffset[mode] = firstChannel; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Set flag for buffer conversion + stream_.doConvertBuffer[mode] = false; + if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + stream_.doConvertBuffer[mode] = true; + if (stream_.userFormat != stream_.deviceFormat[mode]) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + // Allocate our DsHandle structures for the stream. + if ( stream_.apiHandle == 0 ) { + try { + handle = new DsHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; + } + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + else + handle = (DsHandle *) stream_.apiHandle; + handle->id[mode] = ohandle; + handle->buffer[mode] = bhandle; + handle->dsBufferSize[mode] = dsBufferSize; + handle->dsPointerLeadTime[mode] = dsPointerLeadTime; + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + stream_.nBuffers = nBuffers; + stream_.sampleRate = sampleRate; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup the callback thread. + if ( stream_.callbackInfo.isRunning == false ) { + unsigned threadId; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, + &stream_.callbackInfo, 0, &threadId ); + if ( stream_.callbackInfo.thread == 0 ) { + errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; + goto error; + } + + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); + } + return SUCCESS; + + error: + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) buffer->Release(); + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) buffer->Release(); + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiDs :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // Increase scheduler frequency on lesser windows (a side-effect of + // increasing timer accuracy). On greater windows (Win2K or later), + // this is already in effect. + timeBeginPeriod( 1 ); + + buffersRolling = false; + duplexPrerollBytes = 0; + + if ( stream_.mode == DUPLEX ) { + // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. + duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); + } + + HRESULT result = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + result = buffer->Start( DSCBSTART_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + ResetEvent( handle->condition ); + stream_.state = STREAM_RUNNING; + + unlock: + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + HRESULT result = 0; + LPVOID audioPtr; + DWORD dataLen; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 2; + WaitForSingleObject( handle->condition, INFINITE ); // block until signaled + } + + stream_.state = STREAM_STOPPED; + + MUTEX_LOCK( &stream_.mutex ); + + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; + + stream_.state = STREAM_STOPPED; + + if ( stream_.mode != DUPLEX ) + MUTEX_LOCK( &stream_.mutex ); + + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } + + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 2; + + stopStream(); +} + +void RtApiDs :: callbackEvent() +{ + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { + Sleep( 50 ); // sleep 50 milliseconds + return; + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > stream_.nBuffers + 2 ) { + + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return; + } + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( cbReturnValue == 2 ) { + stream_.state = STREAM_STOPPING; + handle->drainCounter = 2; + abortStream(); + return; + } + else if ( cbReturnValue == 1 ) { + handle->drainCounter = 1; + handle->internalDrain = true; + } + } + + HRESULT result; + DWORD currentWritePointer, safeWritePointer; + DWORD currentReadPointer, safeReadPointer; + UINT nextWritePointer; + + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; + + char *buffer; + long bufferBytes; + + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + if ( buffersRolling == false ) { + if ( stream_.mode == DUPLEX ) { + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + // It takes a while for the devices to get rolling. As a result, + // there's no guarantee that the capture and write device pointers + // will move in lockstep. Wait here for both devices to start + // rolling, and then set our buffer pointers accordingly. + // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 + // bytes later than the write buffer. + + // Stub: a serious risk of having a pre-emptive scheduling round + // take place between the two GetCurrentPosition calls... but I'm + // really not sure how to solve the problem. Temporarily boost to + // Realtime priority, maybe; but I'm not sure what priority the + // DirectSound service threads run at. We *should* be roughly + // within a ms or so of correct. + + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + + DWORD startSafeWritePointer, startSafeReadPointer; + + result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break; + Sleep( 1 ); + } + + //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + handle->bufferPointer[1] = safeReadPointer; + } + else if ( stream_.mode == OUTPUT ) { + + // Set the proper nextWritePosition after initial startup. + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0]; + if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0]; + } + + buffersRolling = true; + } + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + memset( stream_.userBuffer[0], 0, bufferBytes ); + } + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; + bufferBytes *= formatBytes( stream_.deviceFormat[0] ); + } + else { + buffer = stream_.userBuffer[0]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + // No byte swapping necessary in DirectSound implementation. + + // Ahhh ... windoze. 16-bit data is signed but 8-bit data is + // unsigned. So, we need to convert our signed 8-bit data here to + // unsigned. + if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) + for ( int i=0; idsBufferSize[0]; + nextWritePointer = handle->bufferPointer[0]; + + DWORD endWrite, leadPointer; + while ( true ) { + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // We will copy our output buffer into the region between + // safeWritePointer and leadPointer. If leadPointer is not + // beyond the next endWrite position, wait until it is. + leadPointer = safeWritePointer + handle->dsPointerLeadTime[0]; + //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl; + if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize; + if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset + endWrite = nextWritePointer + bufferBytes; + + // Check whether the entire write region is behind the play pointer. + if ( leadPointer >= endWrite ) break; + + // If we are here, then we must wait until the leadPointer advances + // beyond the end of our next write region. We use the + // Sleep() function to suspend operation until that happens. + double millis = ( endWrite - leadPointer ) * 1000.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + } + + if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + handle->xrun[0] = true; + nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; + if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + endWrite = nextWritePointer + bufferBytes; + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); + + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePointer; + } + + // Don't bother draining input + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; + bufferBytes *= formatBytes( stream_.deviceFormat[1] ); + } + else { + buffer = stream_.userBuffer[1]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPointer = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPointer + bufferBytes; + + // Handling depends on whether we are INPUT or DUPLEX. + // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, + // then a wait here will drag the write pointers into the forbidden zone. + // + // In DUPLEX mode, rather than wait, we will back off the read pointer until + // it's in a safe position. This causes dropouts, but it seems to be the only + // practical way to sync up the read and write pointers reliably, given the + // the very complex relationship between phase and increment of the read and write + // pointers. + // + // In order to minimize audible dropouts in DUPLEX mode, we will + // provide a pre-roll period of 0.5 seconds in which we return + // zeros from the read buffer while the pointers sync up. + + if ( stream_.mode == DUPLEX ) { + if ( safeReadPointer < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPointer; + + handle->xrun[1] = true; + // Two cases: + // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, + // and perform fine adjustments later. + // - small adjustments: back off by twice as much. + if ( adjustment >= 2*bufferBytes ) + nextReadPointer = safeReadPointer-2*bufferBytes; + else + nextReadPointer = safeReadPointer-bufferBytes-adjustment; + + if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + + } + else { + // In pre=roll time. Just do it. + nextReadPointer = safeReadPointer - bufferBytes; + while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize; + } + endRead = nextReadPointer + bufferBytes; + } + } + else { // mode == INPUT + while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) { + // See comments for playback. + double millis = (endRead - safeReadPointer) * 1000.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up and find out where we are now. + result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset + } + } + + // Lock free space in the buffer + result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + + if ( duplexPrerollBytes <= 0 ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer, buffer1, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + } + else { + memset( buffer, 0, bufferSize1 ); + if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); + duplexPrerollBytes -= bufferSize1 + bufferSize2; + } + + // Update our buffer offset and unlock sound buffer + nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize; + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + handle->bufferPointer[1] = nextReadPointer; + + // No byte swapping necessary in DirectSound implementation. + + // If necessary, convert 8-bit data from unsigned to signed. + if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) + for ( int j=0; jobject; + bool* isRunning = &info->isRunning; + + while ( *isRunning == true ) { + object->callbackEvent(); + } + + _endthreadex( 0 ); + return 0; +} + +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR /*module*/, + LPVOID lpContext ) +{ + struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext; + std::vector& dsDevices = *probeInfo.dsDevices; + + HRESULT hr; + bool validDevice = false; + if ( probeInfo.isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; + + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) + validDevice = true; + } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + validDevice = true; + } + object->Release(); + } + + // If good device, then save its name and guid. + std::string name = convertCharPointerToStdString( description ); + //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" ) + if ( lpguid == NULL ) + name = "Default Device"; + if ( validDevice ) { + for ( unsigned int i=0; i +#include + + // A structure to hold various information related to the ALSA API + // implementation. +struct AlsaHandle { + snd_pcm_t *handles[2]; + bool synchronized; + bool xrun[2]; + pthread_cond_t runnable_cv; + bool runnable; + + AlsaHandle() + :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; } +}; + +static void *alsaCallbackHandler( void * ptr ); + +RtApiAlsa :: RtApiAlsa() +{ + // Nothing to do here. +} + +RtApiAlsa :: ~RtApiAlsa() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiAlsa :: getDeviceCount( void ) +{ + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *handle; + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &handle, name, 0 ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( handle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) + break; + nDevices++; + } + nextcard: + snd_ctl_close( handle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &handle, "default", 0 ); + if (result == 0) { + nDevices++; + snd_ctl_close( handle ); + } + + return nDevices; +} + +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *chandle; + + // Count cards and devices + card = -1; + subdevice = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + break; + } + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + goto foundDevice; + } + nDevices++; + } + nextcard: + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + nDevices++; + } + + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + foundDevice: + + // If a stream is already open, we cannot probe the stream devices. + // Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED && + ( stream_.device[0] == device || stream_.device[1] == device ) ) { + snd_ctl_close( chandle ); + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtAudioError::WARNING ); + return info; + } + return devices_[ device ]; + } + + int openMode = SND_PCM_ASYNC; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); + snd_pcm_t *phandle; + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca( ¶ms ); + + // First try for playback unless default device (which has subdev -1) + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_stream( pcminfo, stream ); + if ( subdevice != -1 ) { + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + + result = snd_ctl_pcm_info( chandle, pcminfo ); + if ( result < 0 ) { + // Device probably doesn't support playback. + goto captureProbe; + } + } + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + + // Get output channel information. + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); + + captureProbe: + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + // Now try for capture unless default device (with subdev = -1) + if ( subdevice != -1 ) { + result = snd_ctl_pcm_info( chandle, pcminfo ); + snd_ctl_close( chandle ); + if ( result < 0 ) { + // Device probably doesn't support capture. + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + } + else + snd_ctl_close( chandle ); + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; + snd_pcm_close( phandle ); + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // ALSA doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + probeParameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if ( info.outputChannels >= info.inputChannels ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); + + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[i]; + } + } + if ( info.sampleRates.size() == 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe the supported data formats ... we don't care about endian-ness just yet + snd_pcm_format_t format; + info.nativeFormats = 0; + format = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT8; + format = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT16; + format = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT24; + format = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_SINT32; + format = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT32; + format = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) + info.nativeFormats |= RTAUDIO_FLOAT64; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Get the device name + char *cardname; + result = snd_card_get_name( card, &cardname ); + if ( result >= 0 ) { + sprintf( name, "hw:%s,%d", cardname, subdevice ); + free( cardname ); + } + info.name = name; + + // That's all ... close the device and return + snd_pcm_close( phandle ); + info.probed = true; + return info; +} + +void RtApiAlsa :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; iflags & RTAUDIO_ALSA_USE_DEFAULT ) + snprintf(name, sizeof(name), "%s", "default"); + else { + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) break; + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + snd_ctl_close( chandle ); + goto foundDevice; + } + nDevices++; + } + snd_ctl_close( chandle ); + snd_card_next( &card ); + } + + result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK ); + if ( result == 0 ) { + if ( nDevices == device ) { + strcpy( name, "default" ); + goto foundDevice; + } + nDevices++; + } + + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + } + + foundDevice: + + // The getDeviceInfo() function will not work for a device that is + // already open. Thus, we'll probe the system before opening a + // stream and save the results for use by getDeviceInfo(). + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once + this->saveDeviceInfo(); + + snd_pcm_stream_t stream; + if ( mode == OUTPUT ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + + snd_pcm_t *phandle; + int openMode = SND_PCM_ASYNC; + result = snd_pcm_open( &phandle, name, stream, openMode ); + if ( result < 0 ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; + else + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca( &hw_params ); + result = snd_pcm_hw_params_any( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set access ... check user preference. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { + stream_.userInterleaved = false; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + stream_.deviceInterleaved[mode] = true; + } + else + stream_.deviceInterleaved[mode] = false; + } + else { + stream_.userInterleaved = true; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + stream_.deviceInterleaved[mode] = false; + } + else + stream_.deviceInterleaved[mode] = true; + } + + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + + if ( format == RTAUDIO_SINT8 ) + deviceFormat = SND_PCM_FORMAT_S8; + else if ( format == RTAUDIO_SINT16 ) + deviceFormat = SND_PCM_FORMAT_S16; + else if ( format == RTAUDIO_SINT24 ) + deviceFormat = SND_PCM_FORMAT_S24; + else if ( format == RTAUDIO_SINT32 ) + deviceFormat = SND_PCM_FORMAT_S32; + else if ( format == RTAUDIO_FLOAT32 ) + deviceFormat = SND_PCM_FORMAT_FLOAT; + else if ( format == RTAUDIO_FLOAT64 ) + deviceFormat = SND_PCM_FORMAT_FLOAT64; + + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { + stream_.deviceFormat[mode] = format; + goto setFormat; + } + + // The user requested format is not natively supported by the device. + deviceFormat = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + goto setFormat; + } + + // If we get here, no supported format was found. + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + + setFormat: + result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine whether byte-swaping is necessary. + stream_.doByteSwap[mode] = false; + if ( deviceFormat != SND_PCM_FORMAT_S8 ) { + result = snd_pcm_format_cpu_endian( deviceFormat ); + if ( result == 0 ) + stream_.doByteSwap[mode] = true; + else if (result < 0) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + + // Set the sample rate. + result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine the number of channels for this device. We support a possible + // minimum device channel number > than the value requested by the user. + stream_.nUserChannels[mode] = channels; + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); + unsigned int deviceChannels = value; + if ( result < 0 || deviceChannels < channels + firstChannel ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + deviceChannels = value; + if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; + stream_.nDeviceChannels[mode] = deviceChannels; + + // Set the device channels. + result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the buffer (or period) size. + int dir = 0; + snd_pcm_uframes_t periodSize = *bufferSize; + result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + *bufferSize = periodSize; + + // Set the buffer number, which in ALSA is referred to as the "period". + unsigned int periods = 0; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers; + if ( periods < 2 ) periods = 4; // a fairly safe default value + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + stream_.bufferSize = *bufferSize; + + // Install the hardware configuration + result = snd_pcm_hw_params( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump( hw_params, out ); +#endif + + // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca( &sw_params ); + snd_pcm_sw_params_current( phandle, sw_params ); + snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); + snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + + result = snd_pcm_sw_params( phandle, sw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + +#if defined(__RTAUDIO_DEBUG__) + fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); + snd_pcm_sw_params_dump( sw_params, out ); +#endif + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the ApiHandle if necessary and then save. + AlsaHandle *apiInfo = 0; + if ( stream_.apiHandle == 0 ) { + try { + apiInfo = (AlsaHandle *) new AlsaHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; + goto error; + } + + if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) apiInfo; + apiInfo->handles[0] = 0; + apiInfo->handles[1] = 0; + } + else { + apiInfo = (AlsaHandle *) stream_.apiHandle; + } + apiInfo->handles[mode] = phandle; + phandle = 0; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.sampleRate = sampleRate; + stream_.nBuffers = periods; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + // Link the streams if possible. + apiInfo->synchronized = false; + if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) + apiInfo->synchronized = true; + else { + errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; + error( RtAudioError::WARNING ); + } + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); + +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + // We previously attempted to increase the audio callback priority + // to SCHED_RR here via the attributes. However, while no errors + // were reported in doing so, it did not work. So, now this is + // done in the alsaCallbackHandler function. + stream_.callbackInfo.doRealtime = true; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + stream_.callbackInfo.priority = priority; + } +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiAlsa::error creating callback thread!"; + goto error; + } + } + + return SUCCESS; + + error: + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + if ( phandle) snd_pcm_close( phandle ); + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.state = STREAM_CLOSED; + return FAILURE; +} + +void RtApiAlsa :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[0] ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[1] ); + } + + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable_cv ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + 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( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + snd_pcm_state_t state; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + state = snd_pcm_state( handle[0] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open + state = snd_pcm_state( handle[1] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + } + + stream_.state = STREAM_RUNNING; + + unlock: + apiInfo->runnable = true; + pthread_cond_signal( &apiInfo->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( apiInfo->synchronized ) + result = snd_pcm_drop( handle[0] ); + else + result = snd_pcm_drain( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = snd_pcm_drop( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + apiInfo->runnable = false; // fixes high CPU usage when stopped + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: callbackEvent() +{ + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !apiInfo->runnable ) + pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + apiInfo->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + apiInfo->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + channels = stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto tryOutput; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; + } + + tryOutput: + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + + // Write samples to device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[0] ) + result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + else + errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtAudioError::WARNING ); + goto unlock; + } + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +static void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; + +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( info->doRealtime ) { + pthread_t tID = pthread_self(); // ID of this thread + sched_param prio = { info->priority }; // scheduling priority of thread + pthread_setschedparam( tID, SCHED_RR, &prio ); + } +#endif + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_ALSA__ *********************// +#endif + +#if defined(__LINUX_PULSE__) + +// Code written by Peter Meerwald, pmeerw@pmeerw.net +// and Tristan Matthews. + +#include +#include +#include + +static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, + 44100, 48000, 96000, 0}; + +struct rtaudio_pa_format_mapping_t { + RtAudioFormat rtaudio_format; + pa_sample_format_t pa_format; +}; + +static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { + {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, + {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, + {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, + {0, PA_SAMPLE_INVALID}}; + +struct PulseAudioHandle { + pa_simple *s_play; + pa_simple *s_rec; + pthread_t thread; + pthread_cond_t runnable_cv; + bool runnable; + PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } +}; + +RtApiPulse::~RtApiPulse() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); +} + +unsigned int RtApiPulse::getDeviceCount( void ) +{ + return 1; +} + +RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) +{ + RtAudio::DeviceInfo info; + info.probed = true; + info.name = "PulseAudio"; + info.outputChannels = 2; + info.inputChannels = 2; + info.duplexChannels = 2; + info.isDefaultOutput = true; + info.isDefaultInput = true; + + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + info.sampleRates.push_back( *sr ); + + info.preferredSampleRate = 48000; + info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; + + return info; +} + +static void *pulseaudio_callback( void * user ) +{ + CallbackInfo *cbi = static_cast( user ); + RtApiPulse *context = static_cast( cbi->object ); + volatile bool *isRunning = &cbi->isRunning; + + while ( *isRunning ) { + pthread_testcancel(); + context->callbackEvent(); + } + + pthread_exit( NULL ); +} + +void RtApiPulse::closeStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + stream_.callbackInfo.isRunning = false; + if ( pah ) { + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + + pthread_join( pah->thread, 0 ); + if ( pah->s_play ) { + pa_simple_flush( pah->s_play, NULL ); + pa_simple_free( pah->s_play ); + } + if ( pah->s_rec ) + pa_simple_free( pah->s_rec ); + + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + stream_.apiHandle = 0; + } + + if ( stream_.userBuffer[0] ) { + free( stream_.userBuffer[0] ); + stream_.userBuffer[0] = 0; + } + if ( stream_.userBuffer[1] ) { + free( stream_.userBuffer[1] ); + stream_.userBuffer[1] = 0; + } + + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; +} + +void RtApiPulse::callbackEvent( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !pah->runnable ) + pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " + "this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT], + stream_.bufferSize, streamTime, status, + stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT]; + void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT]; + + if ( stream_.state != STREAM_RUNNING ) + goto unlock; + + int pa_error; + size_t bytes; + if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.doConvertBuffer[OUTPUT] ) { + convertBuffer( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.convertInfo[OUTPUT] ); + bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[OUTPUT] ); + } else + bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX) { + if ( stream_.doConvertBuffer[INPUT] ) + bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.deviceFormat[INPUT] ); + else + bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * + formatBytes( stream_.userFormat ); + + if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + if ( stream_.doConvertBuffer[INPUT] ) { + convertBuffer( stream_.userBuffer[INPUT], + stream_.deviceBuffer, + stream_.convertInfo[INPUT] ); + } + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); + + if ( doStopStream == 1 ) + stopStream(); +} + +void RtApiPulse::startStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::startStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiPulse::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + stream_.state = STREAM_RUNNING; + + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::stopStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::stopStream: error draining output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::abortStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; + error( RtAudioError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtAudioError::SYSTEM_ERROR ); + return; + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, + unsigned int channels, unsigned int firstChannel, + unsigned int sampleRate, RtAudioFormat format, + unsigned int *bufferSize, RtAudio::StreamOptions *options ) +{ + PulseAudioHandle *pah = 0; + unsigned long bufferBytes = 0; + pa_sample_spec ss; + + if ( device != 0 ) return false; + if ( mode != INPUT && mode != OUTPUT ) return false; + if ( channels != 1 && channels != 2 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels."; + return false; + } + ss.channels = channels; + + if ( firstChannel != 0 ) return false; + + bool sr_found = false; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { + if ( sampleRate == *sr ) { + sr_found = true; + stream_.sampleRate = sampleRate; + ss.rate = sampleRate; + break; + } + } + if ( !sr_found ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; + return false; + } + + bool sf_found = 0; + for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; + sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { + if ( format == sf->rtaudio_format ) { + sf_found = true; + stream_.userFormat = sf->rtaudio_format; + stream_.deviceFormat[mode] = stream_.userFormat; + ss.format = sf->pa_format; + break; + } + } + if ( !sf_found ) { // Use internal data format conversion. + stream_.userFormat = format; + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + ss.format = PA_SAMPLE_FLOAT32LE; + } + + // Set other stream parameters. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + stream_.nBuffers = 1; + stream_.doByteSwap[mode] = false; + stream_.nUserChannels[mode] = channels; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.channelOffset[mode] = 0; + std::string streamName = "RtAudio"; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers. + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + stream_.bufferSize = *bufferSize; + + 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_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.device[mode] = device; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + if ( !stream_.apiHandle ) { + PulseAudioHandle *pah = new PulseAudioHandle; + if ( !pah ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; + goto error; + } + + stream_.apiHandle = pah; + if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; + goto error; + } + } + pah = static_cast( stream_.apiHandle ); + + int error; + if ( options && !options->streamName.empty() ) streamName = options->streamName; + switch ( mode ) { + case INPUT: + pa_buffer_attr buffer_attr; + buffer_attr.fragsize = bufferBytes; + buffer_attr.maxlength = -1; + + pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error ); + if ( !pah->s_rec ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; + goto error; + } + break; + case OUTPUT: + pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + if ( !pah->s_play ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; + goto error; + } + break; + default: + goto error; + } + + if ( stream_.mode == UNINITIALIZED ) + stream_.mode = mode; + else if ( stream_.mode == mode ) + goto error; + else + stream_.mode = DUPLEX; + + if ( !stream_.callbackInfo.isRunning ) { + stream_.callbackInfo.object = this; + stream_.callbackInfo.isRunning = true; + if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; + goto error; + } + } + + stream_.state = STREAM_STOPPED; + return true; + + error: + if ( pah && stream_.callbackInfo.isRunning ) { + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +//******************** End of __LINUX_PULSE__ *********************// +#endif + +#if defined(__LINUX_OSS__) + +#include +#include +#include +#include +#include +#include +#include + +static void *ossCallbackHandler(void * ptr); + +// A structure to hold various information related to the OSS API +// implementation. +struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; + pthread_cond_t runnable; + + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; + +RtApiOss :: RtApiOss() +{ + // Nothing to do here. +} + +RtApiOss :: ~RtApiOss() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiOss :: getDeviceCount( void ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return 0; + } + + oss_sysinfo sysinfo; + if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtAudioError::WARNING ); + return 0; + } + + close( mixerfd ); + return sysinfo.numaudios; +} + +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; + + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtAudioError::WARNING ); + return info; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtAudioError::WARNING ); + return info; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtAudioError::INVALID_USE ); + return info; + } + + 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( RtAudioError::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; +#ifdef AFMT_FLOAT + if ( mask & AFMT_FLOAT ) + info.nativeFormats |= RTAUDIO_FLOAT32; +#endif + if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) + info.nativeFormats |= RTAUDIO_SINT24; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + return info; + } + + // Probe the supported sample rates. + info.sampleRates.clear(); + if ( ainfo.nrates ) { + for ( unsigned int i=0; i info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + + break; + } + } + } + } + else { + // Check min and max rate values; + for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) { + info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) + info.preferredSampleRate = SAMPLE_RATES[k]; + } + } + } + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + error( RtAudioError::WARNING ); + } + else { + info.probed = true; + info.name = ainfo.name; + } + + return info; +} + + +bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; + } + + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; + return FAILURE; + } + + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; + return FAILURE; + } + + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check if device supports input or output + if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || + ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + int flags = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( mode == OUTPUT ) + flags |= O_WRONLY; + else { // mode == INPUT + if (stream_.mode == OUTPUT && stream_.device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close( handle->id[0] ); + handle->id[0] = 0; + if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // Check that the number previously set channels is the same. + if ( stream_.nUserChannels[0] != channels ) { + errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + flags |= O_RDWR; + } + else + flags |= O_RDONLY; + } + + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; + + // Try to open the device. + int fd; + fd = open( ainfo.devnode, flags, 0 ); + if ( fd == -1 ) { + if ( errno == EBUSY ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // For duplex operation, specifically set this mode (this doesn't seem to work). + /* + if ( flags | O_RDWR ) { + result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); + if ( result == -1) { + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + */ + + // Check the device channel support. + stream_.nUserChannels[mode] = channels; + if ( ainfo.max_channels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the number of channels. + int deviceChannels = channels + firstChannel; + result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); + if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = deviceChannels; + + // Get the data format mask + int mask; + result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Determine how to set the device format. + stream_.userFormat = format; + int deviceFormat = -1; + stream_.doByteSwap[mode] = false; + if ( format == RTAUDIO_SINT8 ) { + if ( mask & AFMT_S8 ) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT24 ) { + if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + } + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + } + + if ( deviceFormat == -1 ) { + // The user requested format is not natively supported by the device. + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S8) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + } + + if ( stream_.deviceFormat[mode] == 0 ) { + // This really shouldn't happen ... + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the data format. + int temp = deviceFormat; + result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); + if ( result == -1 || deviceFormat != temp ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; + if ( ossBufferBytes < 16 ) ossBufferBytes = 16; + int buffers = 0; + if ( options ) buffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; + if ( buffers < 2 ) buffers = 3; + temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); + result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nBuffers = buffers; + + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; + + // Set the sample rate. + int srate = sampleRate; + result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Verify the sample rate setup worked. + if ( abs( srate - (int)sampleRate ) > 100 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = sampleRate; + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { + // We're doing duplex setup here. + stream_.deviceFormat[0] = stream_.deviceFormat[1]; + stream_.nDeviceChannels[0] = deviceChannels; + } + + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the stream handles if necessary and then save. + if ( stream_.apiHandle == 0 ) { + try { + handle = new OssHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; + } + + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + handle->id[mode] = fd; + + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority. The higher priority will only take affect if the + // program is run as root or suid. + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiOss::error creating callback thread!"; + goto error; + } + } + + return SUCCESS; + + error: + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + return FAILURE; +} + +void RtApiOss :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtAudioError::WARNING ); + return; + } + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); + + if ( stream_.state == STREAM_RUNNING ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + else + ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + stream_.state = STREAM_STOPPED; + } + + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiOss :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + stream_.state = STREAM_RUNNING; + + // No need to do anything else here ... OSS automatically starts + // when fed samples. + + MUTEX_UNLOCK( &stream_.mutex ); + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); +} + +void RtApiOss :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + memset( buffer, 0, samples * formatBytes(format) ); + for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); + if ( result == -1 ) { + errorText_ = "RtApiOss::stopStream: audio write error."; + error( RtAudioError::WARNING ); + } + } + + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiOss :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtAudioError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; + } + + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtAudioError::SYSTEM_ERROR ); +} + +void RtApiOss :: callbackEvent() +{ + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtAudioError::WARNING ); + return; + } + + // Invoke user callback to get fresh output data. + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; + + int result; + char *buffer; + int samples; + RtAudioFormat format; + + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( buffer, samples, format ); + + if ( stream_.mode == DUPLEX && handle->triggered == false ) { + int trig = 0; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + handle->triggered = true; + } + else + // Write samples to device. + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an underrun, though there isn't a + // specific means for determining that. + handle->xrun[0] = true; + errorText_ = "RtApiOss::callbackEvent: audio write error."; + error( RtAudioError::WARNING ); + // Continue on to input section. + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + samples = stream_.bufferSize * stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device. + result = read( handle->id[1], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an overrun, though there isn't a + // specific means for determining that. + handle->xrun[1] = true; + errorText_ = "RtApiOss::callbackEvent: audio read error."; + error( RtAudioError::WARNING ); + goto unlock; + } + + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); + + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +static void *ossCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; + + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } + + pthread_exit( NULL ); +} + +//******************** End of __LINUX_OSS__ *********************// +#endif + + +// *************************************************** // +// +// Protected common (OS-independent) RtAudio methods. +// +// *************************************************** // + +// This method can be modified to control the behavior of error +// message printing. +void RtApi :: error( RtAudioError::Type type ) +{ + errorStream_.str(""); // clear the ostringstream + + RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback; + if ( errorCallback ) { + // abortStream() can generate new error messages. Ignore them. Just keep original one. + + if ( firstErrorOccurred_ ) + return; + + firstErrorOccurred_ = true; + const std::string errorMessage = errorText_; + + if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) { + stream_.callbackInfo.isRunning = false; // exit from the thread + abortStream(); + } + + errorCallback( type, errorMessage ); + firstErrorOccurred_ = false; + return; + } + + if ( type == RtAudioError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else if ( type != RtAudioError::WARNING ) + throw( RtAudioError( errorText_, type ) ); +} + +void RtApi :: verifyStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtAudioError::INVALID_USE ); + } +} + +void RtApi :: clearStreamInfo() +{ + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; + stream_.sampleRate = 0; + stream_.bufferSize = 0; + stream_.nBuffers = 0; + stream_.userFormat = 0; + stream_.userInterleaved = true; + stream_.streamTime = 0.0; + stream_.apiHandle = 0; + stream_.deviceBuffer = 0; + stream_.callbackInfo.callback = 0; + stream_.callbackInfo.userData = 0; + stream_.callbackInfo.isRunning = false; + stream_.callbackInfo.errorCallback = 0; + for ( int i=0; i<2; i++ ) { + stream_.device[i] = 11111; + stream_.doConvertBuffer[i] = false; + stream_.deviceInterleaved[i] = true; + stream_.doByteSwap[i] = false; + stream_.nUserChannels[i] = 0; + stream_.nDeviceChannels[i] = 0; + stream_.channelOffset[i] = 0; + stream_.deviceFormat[i] = 0; + stream_.latency[i] = 0; + stream_.userBuffer[i] = 0; + stream_.convertInfo[i].channels = 0; + stream_.convertInfo[i].inJump = 0; + stream_.convertInfo[i].outJump = 0; + stream_.convertInfo[i].inFormat = 0; + stream_.convertInfo[i].outFormat = 0; + stream_.convertInfo[i].inOffset.clear(); + stream_.convertInfo[i].outOffset.clear(); + } +} + +unsigned int RtApi :: formatBytes( RtAudioFormat format ) +{ + if ( format == RTAUDIO_SINT16 ) + return 2; + else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 ) + return 4; + else if ( format == RTAUDIO_FLOAT64 ) + return 8; + else if ( format == RTAUDIO_SINT24 ) + return 3; + else if ( format == RTAUDIO_SINT8 ) + return 1; + + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtAudioError::WARNING ); + + return 0; +} + +void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) +{ + if ( mode == INPUT ) { // convert device to user buffer + stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; + stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; + stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; + stream_.convertInfo[mode].outFormat = stream_.userFormat; + } + else { // convert user to device buffer + stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; + stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; + stream_.convertInfo[mode].inFormat = stream_.userFormat; + stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; + } + + if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; + else + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; + + // Set up the interleave/deinterleave offsets. + if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { + if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || + ( mode == INPUT && stream_.userInterleaved ) ) { + for ( int k=0; k 0 ) { + if ( stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; k> 8); + //out[info.outOffset[j]] >>= 8; + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x00ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int24 *in = (Int24 *)inBuffer; + for (unsigned int i=0; i> 16); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 24) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; + } + } + else if (info.inFormat == RTAUDIO_FLOAT32) { + Float32 *in = (Float32 *)inBuffer; + for (unsigned int i=0; i>8) | (x<<8); } +//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } +//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } + +void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) +{ + char val; + char *ptr; + + ptr = buffer; + if ( format == RTAUDIO_SINT16 ) { + for ( unsigned int i=0; i