f@0: /************************************************************************/
f@0: /*! \class RtAudio
f@0:     \brief Realtime audio i/o C++ classes.
f@0: 
f@0:     RtAudio provides a common API (Application Programming Interface)
f@0:     for realtime audio input/output across Linux (native ALSA, Jack,
f@0:     and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
f@0:     (DirectSound, ASIO and WASAPI) operating systems.
f@0: 
f@0:     RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
f@0: 
f@0:     RtAudio: realtime audio i/o C++ classes
f@0:     Copyright (c) 2001-2014 Gary P. Scavone
f@0: 
f@0:     Permission is hereby granted, free of charge, to any person
f@0:     obtaining a copy of this software and associated documentation files
f@0:     (the "Software"), to deal in the Software without restriction,
f@0:     including without limitation the rights to use, copy, modify, merge,
f@0:     publish, distribute, sublicense, and/or sell copies of the Software,
f@0:     and to permit persons to whom the Software is furnished to do so,
f@0:     subject to the following conditions:
f@0: 
f@0:     The above copyright notice and this permission notice shall be
f@0:     included in all copies or substantial portions of the Software.
f@0: 
f@0:     Any person wishing to distribute modifications to the Software is
f@0:     asked to send the modifications to the original developer so that
f@0:     they can be incorporated into the canonical version.  This is,
f@0:     however, not a binding provision of this license.
f@0: 
f@0:     THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
f@0:     EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
f@0:     MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
f@0:     IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
f@0:     ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
f@0:     CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
f@0:     WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
f@0: */
f@0: /************************************************************************/
f@0: 
f@0: // RtAudio: Version 4.1.1
f@0: #pragma once
f@0: 
f@0: 
f@0: #include "../include/RtAudio.h"
f@0: #include <iostream>
f@0: #include <cstdlib>
f@0: #include <cstring>
f@0: #include <climits>
f@0: 
f@0: template <class T> const T& max(const T& a, const T& b) {
f@0: 	return (a<b) ? b : a;     // or: return comp(a,b)?b:a; for version (2)
f@0: }
f@0: 
f@0: // Static variable definitions.
f@0: const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
f@0: const unsigned int RtApi::SAMPLE_RATES[] = {
f@0:   4000, 5512, 8000, 9600, 11025, 16000, 22050,
f@0:   32000, 44100, 48000, 88200, 96000, 176400, 192000
f@0: };
f@0: 
f@0: #if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
f@0:   #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
f@0:   #define MUTEX_DESTROY(A)    DeleteCriticalSection(A)
f@0:   #define MUTEX_LOCK(A)       EnterCriticalSection(A)
f@0:   #define MUTEX_UNLOCK(A)     LeaveCriticalSection(A)
f@0: #elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
f@0:   // pthread API
f@0:   #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
f@0:   #define MUTEX_DESTROY(A)    pthread_mutex_destroy(A)
f@0:   #define MUTEX_LOCK(A)       pthread_mutex_lock(A)
f@0:   #define MUTEX_UNLOCK(A)     pthread_mutex_unlock(A)
f@0: #else
f@0:   #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
f@0:   #define MUTEX_DESTROY(A)    abs(*A) // dummy definitions
f@0: #endif
f@0: 
f@0: // *************************************************** //
f@0: //
f@0: // RtAudio definitions.
f@0: //
f@0: // *************************************************** //
f@0: 
f@0: std::string RtAudio :: getVersion( void ) throw()
f@0: {
f@0:   return RTAUDIO_VERSION;
f@0: }
f@0: 
f@0: void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
f@0: {
f@0:   apis.clear();
f@0: 
f@0:   // The order here will control the order of RtAudio's API search in
f@0:   // the constructor.
f@0: #if defined(__UNIX_JACK__)
f@0:   apis.push_back( UNIX_JACK );
f@0: #endif
f@0: #if defined(__LINUX_ALSA__)
f@0:   apis.push_back( LINUX_ALSA );
f@0: #endif
f@0: #if defined(__LINUX_PULSE__)
f@0:   apis.push_back( LINUX_PULSE );
f@0: #endif
f@0: #if defined(__LINUX_OSS__)
f@0:   apis.push_back( LINUX_OSS );
f@0: #endif
f@0: #if defined(__WINDOWS_ASIO__)
f@0:   apis.push_back( WINDOWS_ASIO );
f@0: #endif
f@0: #if defined(__WINDOWS_WASAPI__)
f@0:   apis.push_back( WINDOWS_WASAPI );
f@0: #endif
f@0: #if defined(__WINDOWS_DS__)
f@0:   apis.push_back( WINDOWS_DS );
f@0: #endif
f@0: #if defined(__MACOSX_CORE__)
f@0:   apis.push_back( MACOSX_CORE );
f@0: #endif
f@0: #if defined(__RTAUDIO_DUMMY__)
f@0:   apis.push_back( RTAUDIO_DUMMY );
f@0: #endif
f@0: }
f@0: 
f@0: void RtAudio :: openRtApi( RtAudio::Api api )
f@0: {
f@0:   if ( rtapi_ )
f@0:     delete rtapi_;
f@0:   rtapi_ = 0;
f@0: 
f@0: #if defined(__UNIX_JACK__)
f@0:   if ( api == UNIX_JACK )
f@0:     rtapi_ = new RtApiJack();
f@0: #endif
f@0: #if defined(__LINUX_ALSA__)
f@0:   if ( api == LINUX_ALSA )
f@0:     rtapi_ = new RtApiAlsa();
f@0: #endif
f@0: #if defined(__LINUX_PULSE__)
f@0:   if ( api == LINUX_PULSE )
f@0:     rtapi_ = new RtApiPulse();
f@0: #endif
f@0: #if defined(__LINUX_OSS__)
f@0:   if ( api == LINUX_OSS )
f@0:     rtapi_ = new RtApiOss();
f@0: #endif
f@0: #if defined(__WINDOWS_ASIO__)
f@0:   if ( api == WINDOWS_ASIO )
f@0:     rtapi_ = new RtApiAsio();
f@0: #endif
f@0: #if defined(__WINDOWS_WASAPI__)
f@0:   if ( api == WINDOWS_WASAPI )
f@0:     rtapi_ = new RtApiWasapi();
f@0: #endif
f@0: #if defined(__WINDOWS_DS__)
f@0:   if ( api == WINDOWS_DS )
f@0:     rtapi_ = new RtApiDs();
f@0: #endif
f@0: #if defined(__MACOSX_CORE__)
f@0:   if ( api == MACOSX_CORE )
f@0:     rtapi_ = new RtApiCore();
f@0: #endif
f@0: #if defined(__RTAUDIO_DUMMY__)
f@0:   if ( api == RTAUDIO_DUMMY )
f@0:     rtapi_ = new RtApiDummy();
f@0: #endif
f@0: }
f@0: 
f@0: RtAudio :: RtAudio( RtAudio::Api api )
f@0: {
f@0:   rtapi_ = 0;
f@0: 
f@0:   if ( api != UNSPECIFIED ) {
f@0:     // Attempt to open the specified API.
f@0:     openRtApi( api );
f@0:     if ( rtapi_ ) return;
f@0: 
f@0:     // No compiled support for specified API value.  Issue a debug
f@0:     // warning and continue as if no API was specified.
f@0:     std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
f@0:   }
f@0: 
f@0:   // Iterate through the compiled APIs and return as soon as we find
f@0:   // one with at least one device or we reach the end of the list.
f@0:   std::vector< RtAudio::Api > apis;
f@0:   getCompiledApi( apis );
f@0:   for ( unsigned int i=0; i<apis.size(); i++ ) {
f@0:     openRtApi( apis[i] );
f@0:     if ( rtapi_->getDeviceCount() ) break;
f@0:   }
f@0: 
f@0:   if ( rtapi_ ) return;
f@0: 
f@0:   // It should not be possible to get here because the preprocessor
f@0:   // definition __RTAUDIO_DUMMY__ is automatically defined if no
f@0:   // API-specific definitions are passed to the compiler. But just in
f@0:   // case something weird happens, we'll thow an error.
f@0:   std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
f@0:   throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
f@0: }
f@0: 
f@0: RtAudio :: ~RtAudio() throw()
f@0: {
f@0:   if ( rtapi_ )
f@0:     delete rtapi_;
f@0: }
f@0: 
f@0: void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
f@0:                             RtAudio::StreamParameters *inputParameters,
f@0:                             RtAudioFormat format, unsigned int sampleRate,
f@0:                             unsigned int *bufferFrames,
f@0:                             RtAudioCallback callback, void *userData,
f@0:                             RtAudio::StreamOptions *options,
f@0:                             RtAudioErrorCallback errorCallback )
f@0: {
f@0:   return rtapi_->openStream( outputParameters, inputParameters, format,
f@0:                              sampleRate, bufferFrames, callback,
f@0:                              userData, options, errorCallback );
f@0: }
f@0: 
f@0: // *************************************************** //
f@0: //
f@0: // Public RtApi definitions (see end of file for
f@0: // private or protected utility functions).
f@0: //
f@0: // *************************************************** //
f@0: 
f@0: RtApi :: RtApi()
f@0: {
f@0:   stream_.state = STREAM_CLOSED;
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.apiHandle = 0;
f@0:   stream_.userBuffer[0] = 0;
f@0:   stream_.userBuffer[1] = 0;
f@0:   MUTEX_INITIALIZE( &stream_.mutex );
f@0:   showWarnings_ = true;
f@0:   firstErrorOccurred_ = false;
f@0: }
f@0: 
f@0: RtApi :: ~RtApi()
f@0: {
f@0:   MUTEX_DESTROY( &stream_.mutex );
f@0: }
f@0: 
f@0: void RtApi :: openStream( RtAudio::StreamParameters *oParams,
f@0:                           RtAudio::StreamParameters *iParams,
f@0:                           RtAudioFormat format, unsigned int sampleRate,
f@0:                           unsigned int *bufferFrames,
f@0:                           RtAudioCallback callback, void *userData,
f@0:                           RtAudio::StreamOptions *options,
f@0:                           RtAudioErrorCallback errorCallback )
f@0: {
f@0:   if ( stream_.state != STREAM_CLOSED ) {
f@0:     errorText_ = "RtApi::openStream: a stream is already open!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0: 
f@0:   // Clear stream information potentially left from a previously open stream.
f@0:   clearStreamInfo();
f@0: 
f@0:   if ( oParams && oParams->nChannels < 1 ) {
f@0:     errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( iParams && iParams->nChannels < 1 ) {
f@0:     errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( oParams == NULL && iParams == NULL ) {
f@0:     errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( formatBytes(format) == 0 ) {
f@0:     errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0: 
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   unsigned int oChannels = 0;
f@0:   if ( oParams ) {
f@0:     oChannels = oParams->nChannels;
f@0:     if ( oParams->deviceId >= nDevices ) {
f@0:       errorText_ = "RtApi::openStream: output device parameter value is invalid.";
f@0:       error( RtAudioError::INVALID_USE );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   unsigned int iChannels = 0;
f@0:   if ( iParams ) {
f@0:     iChannels = iParams->nChannels;
f@0:     if ( iParams->deviceId >= nDevices ) {
f@0:       errorText_ = "RtApi::openStream: input device parameter value is invalid.";
f@0:       error( RtAudioError::INVALID_USE );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   bool result;
f@0: 
f@0:   if ( oChannels > 0 ) {
f@0: 
f@0:     result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
f@0:                               sampleRate, format, bufferFrames, options );
f@0:     if ( result == false ) {
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( iChannels > 0 ) {
f@0: 
f@0:     result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
f@0:                               sampleRate, format, bufferFrames, options );
f@0:     if ( result == false ) {
f@0:       if ( oChannels > 0 ) closeStream();
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.callbackInfo.callback = (void *) callback;
f@0:   stream_.callbackInfo.userData = userData;
f@0:   stream_.callbackInfo.errorCallback = (void *) errorCallback;
f@0: 
f@0:   if ( options ) options->numberOfBuffers = stream_.nBuffers;
f@0:   stream_.state = STREAM_STOPPED;
f@0: }
f@0: 
f@0: unsigned int RtApi :: getDefaultInputDevice( void )
f@0: {
f@0:   // Should be implemented in subclasses if possible.
f@0:   return 0;
f@0: }
f@0: 
f@0: unsigned int RtApi :: getDefaultOutputDevice( void )
f@0: {
f@0:   // Should be implemented in subclasses if possible.
f@0:   return 0;
f@0: }
f@0: 
f@0: void RtApi :: closeStream( void )
f@0: {
f@0:   // MUST be implemented in subclasses!
f@0:   return;
f@0: }
f@0: 
f@0: bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
f@0:                                unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
f@0:                                RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
f@0:                                RtAudio::StreamOptions * /*options*/ )
f@0: {
f@0:   // MUST be implemented in subclasses!
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: void RtApi :: tickStreamTime( void )
f@0: {
f@0:   // Subclasses that do not provide their own implementation of
f@0:   // getStreamTime should call this function once per buffer I/O to
f@0:   // provide basic stream time support.
f@0: 
f@0:   stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
f@0: 
f@0: #if defined( HAVE_GETTIMEOFDAY )
f@0:   gettimeofday( &stream_.lastTickTimestamp, NULL );
f@0: #endif
f@0: }
f@0: 
f@0: long RtApi :: getStreamLatency( void )
f@0: {
f@0:   verifyStream();
f@0: 
f@0:   long totalLatency = 0;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
f@0:     totalLatency = stream_.latency[0];
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
f@0:     totalLatency += stream_.latency[1];
f@0: 
f@0:   return totalLatency;
f@0: }
f@0: 
f@0: double RtApi :: getStreamTime( void )
f@0: {
f@0:   verifyStream();
f@0: 
f@0: #if defined( HAVE_GETTIMEOFDAY )
f@0:   // Return a very accurate estimate of the stream time by
f@0:   // adding in the elapsed time since the last tick.
f@0:   struct timeval then;
f@0:   struct timeval now;
f@0: 
f@0:   if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
f@0:     return stream_.streamTime;
f@0: 
f@0:   gettimeofday( &now, NULL );
f@0:   then = stream_.lastTickTimestamp;
f@0:   return stream_.streamTime +
f@0:     ((now.tv_sec + 0.000001 * now.tv_usec) -
f@0:      (then.tv_sec + 0.000001 * then.tv_usec));     
f@0: #else
f@0:   return stream_.streamTime;
f@0: #endif
f@0: }
f@0: 
f@0: void RtApi :: setStreamTime( double time )
f@0: {
f@0:   verifyStream();
f@0: 
f@0:   if ( time >= 0.0 )
f@0:     stream_.streamTime = time;
f@0: }
f@0: 
f@0: unsigned int RtApi :: getStreamSampleRate( void )
f@0: {
f@0:  verifyStream();
f@0: 
f@0:  return stream_.sampleRate;
f@0: }
f@0: 
f@0: 
f@0: // *************************************************** //
f@0: //
f@0: // OS/API-specific methods.
f@0: //
f@0: // *************************************************** //
f@0: 
f@0: #if defined(__MACOSX_CORE__)
f@0: 
f@0: // The OS X CoreAudio API is designed to use a separate callback
f@0: // procedure for each of its audio devices.  A single RtAudio duplex
f@0: // stream using two different devices is supported here, though it
f@0: // cannot be guaranteed to always behave correctly because we cannot
f@0: // synchronize these two callbacks.
f@0: //
f@0: // A property listener is installed for over/underrun information.
f@0: // However, no functionality is currently provided to allow property
f@0: // listeners to trigger user handlers because it is unclear what could
f@0: // be done if a critical stream parameter (buffer size, sample rate,
f@0: // device disconnect) notification arrived.  The listeners entail
f@0: // quite a bit of extra code and most likely, a user program wouldn't
f@0: // be prepared for the result anyway.  However, we do provide a flag
f@0: // to the client callback function to inform of an over/underrun.
f@0: 
f@0: // A structure to hold various information related to the CoreAudio API
f@0: // implementation.
f@0: struct CoreHandle {
f@0:   AudioDeviceID id[2];    // device ids
f@0: #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
f@0:   AudioDeviceIOProcID procId[2];
f@0: #endif
f@0:   UInt32 iStream[2];      // device stream index (or first if using multiple)
f@0:   UInt32 nStreams[2];     // number of streams to use
f@0:   bool xrun[2];
f@0:   char *deviceBuffer;
f@0:   pthread_cond_t condition;
f@0:   int drainCounter;       // Tracks callback counts when draining
f@0:   bool internalDrain;     // Indicates if stop is initiated from callback or not.
f@0: 
f@0:   CoreHandle()
f@0:     :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
f@0: };
f@0: 
f@0: RtApiCore:: RtApiCore()
f@0: {
f@0: #if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
f@0:   // This is a largely undocumented but absolutely necessary
f@0:   // requirement starting with OS-X 10.6.  If not called, queries and
f@0:   // updates to various audio device properties are not handled
f@0:   // correctly.
f@0:   CFRunLoopRef theRunLoop = NULL;
f@0:   AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
f@0:                                           kAudioObjectPropertyScopeGlobal,
f@0:                                           kAudioObjectPropertyElementMaster };
f@0:   OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
f@0:     error( RtAudioError::WARNING );
f@0:   }
f@0: #endif
f@0: }
f@0: 
f@0: RtApiCore :: ~RtApiCore()
f@0: {
f@0:   // The subclass destructor gets called before the base class
f@0:   // destructor, so close an existing stream before deallocating
f@0:   // apiDeviceId memory.
f@0:   if ( stream_.state != STREAM_CLOSED ) closeStream();
f@0: }
f@0: 
f@0: unsigned int RtApiCore :: getDeviceCount( void )
f@0: {
f@0:   // Find out how many audio devices there are, if any.
f@0:   UInt32 dataSize;
f@0:   AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
f@0:   OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
f@0:     error( RtAudioError::WARNING );
f@0:     return 0;
f@0:   }
f@0: 
f@0:   return dataSize / sizeof( AudioDeviceID );
f@0: }
f@0: 
f@0: unsigned int RtApiCore :: getDefaultInputDevice( void )
f@0: {
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   if ( nDevices <= 1 ) return 0;
f@0: 
f@0:   AudioDeviceID id;
f@0:   UInt32 dataSize = sizeof( AudioDeviceID );
f@0:   AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
f@0:   OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
f@0:     error( RtAudioError::WARNING );
f@0:     return 0;
f@0:   }
f@0: 
f@0:   dataSize *= nDevices;
f@0:   AudioDeviceID deviceList[ nDevices ];
f@0:   property.mSelector = kAudioHardwarePropertyDevices;
f@0:   result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
f@0:     error( RtAudioError::WARNING );
f@0:     return 0;
f@0:   }
f@0: 
f@0:   for ( unsigned int i=0; i<nDevices; i++ )
f@0:     if ( id == deviceList[i] ) return i;
f@0: 
f@0:   errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
f@0:   error( RtAudioError::WARNING );
f@0:   return 0;
f@0: }
f@0: 
f@0: unsigned int RtApiCore :: getDefaultOutputDevice( void )
f@0: {
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   if ( nDevices <= 1 ) return 0;
f@0: 
f@0:   AudioDeviceID id;
f@0:   UInt32 dataSize = sizeof( AudioDeviceID );
f@0:   AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
f@0:   OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
f@0:     error( RtAudioError::WARNING );
f@0:     return 0;
f@0:   }
f@0: 
f@0:   dataSize = sizeof( AudioDeviceID ) * nDevices;
f@0:   AudioDeviceID deviceList[ nDevices ];
f@0:   property.mSelector = kAudioHardwarePropertyDevices;
f@0:   result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
f@0:     error( RtAudioError::WARNING );
f@0:     return 0;
f@0:   }
f@0: 
f@0:   for ( unsigned int i=0; i<nDevices; i++ )
f@0:     if ( id == deviceList[i] ) return i;
f@0: 
f@0:   errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
f@0:   error( RtAudioError::WARNING );
f@0:   return 0;
f@0: }
f@0: 
f@0: RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   info.probed = false;
f@0: 
f@0:   // Get device ID
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   if ( nDevices == 0 ) {
f@0:     errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   AudioDeviceID deviceList[ nDevices ];
f@0:   UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
f@0:   AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
f@0:                                           kAudioObjectPropertyScopeGlobal,
f@0:                                           kAudioObjectPropertyElementMaster };
f@0:   OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
f@0:                                                 0, NULL, &dataSize, (void *) &deviceList );
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   AudioDeviceID id = deviceList[ device ];
f@0: 
f@0:   // Get the device name.
f@0:   info.name.erase();
f@0:   CFStringRef cfname;
f@0:   dataSize = sizeof( CFStringRef );
f@0:   property.mSelector = kAudioObjectPropertyManufacturer;
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
f@0:   if ( result != noErr ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
f@0:   int length = CFStringGetLength(cfname);
f@0:   char *mname = (char *)malloc(length * 3 + 1);
f@0: #if defined( UNICODE ) || defined( _UNICODE )
f@0:   CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
f@0: #else
f@0:   CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
f@0: #endif
f@0:   info.name.append( (const char *)mname, strlen(mname) );
f@0:   info.name.append( ": " );
f@0:   CFRelease( cfname );
f@0:   free(mname);
f@0: 
f@0:   property.mSelector = kAudioObjectPropertyName;
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
f@0:   if ( result != noErr ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
f@0:   length = CFStringGetLength(cfname);
f@0:   char *name = (char *)malloc(length * 3 + 1);
f@0: #if defined( UNICODE ) || defined( _UNICODE )
f@0:   CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
f@0: #else
f@0:   CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
f@0: #endif
f@0:   info.name.append( (const char *)name, strlen(name) );
f@0:   CFRelease( cfname );
f@0:   free(name);
f@0: 
f@0:   // Get the output stream "configuration".
f@0:   AudioBufferList	*bufferList = nil;
f@0:   property.mSelector = kAudioDevicePropertyStreamConfiguration;
f@0:   property.mScope = kAudioDevicePropertyScopeOutput;
f@0:   //  property.mElement = kAudioObjectPropertyElementWildcard;
f@0:   dataSize = 0;
f@0:   result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
f@0:   if ( result != noErr || dataSize == 0 ) {
f@0:     errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Allocate the AudioBufferList.
f@0:   bufferList = (AudioBufferList *) malloc( dataSize );
f@0:   if ( bufferList == NULL ) {
f@0:     errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
f@0:   if ( result != noErr || dataSize == 0 ) {
f@0:     free( bufferList );
f@0:     errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Get output channel information.
f@0:   unsigned int i, nStreams = bufferList->mNumberBuffers;
f@0:   for ( i=0; i<nStreams; i++ )
f@0:     info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
f@0:   free( bufferList );
f@0: 
f@0:   // Get the input stream "configuration".
f@0:   property.mScope = kAudioDevicePropertyScopeInput;
f@0:   result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
f@0:   if ( result != noErr || dataSize == 0 ) {
f@0:     errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Allocate the AudioBufferList.
f@0:   bufferList = (AudioBufferList *) malloc( dataSize );
f@0:   if ( bufferList == NULL ) {
f@0:     errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
f@0:   if (result != noErr || dataSize == 0) {
f@0:     free( bufferList );
f@0:     errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Get input channel information.
f@0:   nStreams = bufferList->mNumberBuffers;
f@0:   for ( i=0; i<nStreams; i++ )
f@0:     info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
f@0:   free( bufferList );
f@0: 
f@0:   // If device opens for both playback and capture, we determine the channels.
f@0:   if ( info.outputChannels > 0 && info.inputChannels > 0 )
f@0:     info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
f@0: 
f@0:   // Probe the device sample rates.
f@0:   bool isInput = false;
f@0:   if ( info.outputChannels == 0 ) isInput = true;
f@0: 
f@0:   // Determine the supported sample rates.
f@0:   property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
f@0:   if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
f@0:   result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
f@0:   if ( result != kAudioHardwareNoError || dataSize == 0 ) {
f@0:     errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   UInt32 nRanges = dataSize / sizeof( AudioValueRange );
f@0:   AudioValueRange rangeList[ nRanges ];
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
f@0:   if ( result != kAudioHardwareNoError ) {
f@0:     errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // The sample rate reporting mechanism is a bit of a mystery.  It
f@0:   // seems that it can either return individual rates or a range of
f@0:   // rates.  I assume that if the min / max range values are the same,
f@0:   // then that represents a single supported rate and if the min / max
f@0:   // range values are different, the device supports an arbitrary
f@0:   // range of values (though there might be multiple ranges, so we'll
f@0:   // use the most conservative range).
f@0:   Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
f@0:   bool haveValueRange = false;
f@0:   info.sampleRates.clear();
f@0:   for ( UInt32 i=0; i<nRanges; i++ ) {
f@0:     if ( rangeList[i].mMinimum == rangeList[i].mMaximum )
f@0:       info.sampleRates.push_back( (unsigned int) rangeList[i].mMinimum );
f@0:     else {
f@0:       haveValueRange = true;
f@0:       if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
f@0:       if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( haveValueRange ) {
f@0:     for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
f@0:       if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
f@0:         info.sampleRates.push_back( SAMPLE_RATES[k] );
f@0:     }
f@0:   }
f@0: 
f@0:   // Sort and remove any redundant values
f@0:   std::sort( info.sampleRates.begin(), info.sampleRates.end() );
f@0:   info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
f@0: 
f@0:   if ( info.sampleRates.size() == 0 ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // CoreAudio always uses 32-bit floating point data for PCM streams.
f@0:   // Thus, any other "physical" formats supported by the device are of
f@0:   // no interest to the client.
f@0:   info.nativeFormats = RTAUDIO_FLOAT32;
f@0: 
f@0:   if ( info.outputChannels > 0 )
f@0:     if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
f@0:   if ( info.inputChannels > 0 )
f@0:     if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
f@0: 
f@0:   info.probed = true;
f@0:   return info;
f@0: }
f@0: 
f@0: static OSStatus callbackHandler( AudioDeviceID inDevice,
f@0:                                  const AudioTimeStamp* /*inNow*/,
f@0:                                  const AudioBufferList* inInputData,
f@0:                                  const AudioTimeStamp* /*inInputTime*/,
f@0:                                  AudioBufferList* outOutputData,
f@0:                                  const AudioTimeStamp* /*inOutputTime*/,
f@0:                                  void* infoPointer )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) infoPointer;
f@0: 
f@0:   RtApiCore *object = (RtApiCore *) info->object;
f@0:   if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
f@0:     return kAudioHardwareUnspecifiedError;
f@0:   else
f@0:     return kAudioHardwareNoError;
f@0: }
f@0: 
f@0: static OSStatus xrunListener( AudioObjectID /*inDevice*/,
f@0:                               UInt32 nAddresses,
f@0:                               const AudioObjectPropertyAddress properties[],
f@0:                               void* handlePointer )
f@0: {
f@0:   CoreHandle *handle = (CoreHandle *) handlePointer;
f@0:   for ( UInt32 i=0; i<nAddresses; i++ ) {
f@0:     if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
f@0:       if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
f@0:         handle->xrun[1] = true;
f@0:       else
f@0:         handle->xrun[0] = true;
f@0:     }
f@0:   }
f@0: 
f@0:   return kAudioHardwareNoError;
f@0: }
f@0: 
f@0: static OSStatus rateListener( AudioObjectID inDevice,
f@0:                               UInt32 /*nAddresses*/,
f@0:                               const AudioObjectPropertyAddress /*properties*/[],
f@0:                               void* ratePointer )
f@0: {
f@0:   Float64 *rate = (Float64 *) ratePointer;
f@0:   UInt32 dataSize = sizeof( Float64 );
f@0:   AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
f@0:                                           kAudioObjectPropertyScopeGlobal,
f@0:                                           kAudioObjectPropertyElementMaster };
f@0:   AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
f@0:   return kAudioHardwareNoError;
f@0: }
f@0: 
f@0: bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
f@0:                                    unsigned int firstChannel, unsigned int sampleRate,
f@0:                                    RtAudioFormat format, unsigned int *bufferSize,
f@0:                                    RtAudio::StreamOptions *options )
f@0: {
f@0:   // Get device ID
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   if ( nDevices == 0 ) {
f@0:     // This should not happen because a check is made before this function is called.
f@0:     errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     // This should not happen because a check is made before this function is called.
f@0:     errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   AudioDeviceID deviceList[ nDevices ];
f@0:   UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
f@0:   AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
f@0:                                           kAudioObjectPropertyScopeGlobal,
f@0:                                           kAudioObjectPropertyElementMaster };
f@0:   OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
f@0:                                                 0, NULL, &dataSize, (void *) &deviceList );
f@0:   if ( result != noErr ) {
f@0:     errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   AudioDeviceID id = deviceList[ device ];
f@0: 
f@0:   // Setup for stream mode.
f@0:   bool isInput = false;
f@0:   if ( mode == INPUT ) {
f@0:     isInput = true;
f@0:     property.mScope = kAudioDevicePropertyScopeInput;
f@0:   }
f@0:   else
f@0:     property.mScope = kAudioDevicePropertyScopeOutput;
f@0: 
f@0:   // Get the stream "configuration".
f@0:   AudioBufferList	*bufferList = nil;
f@0:   dataSize = 0;
f@0:   property.mSelector = kAudioDevicePropertyStreamConfiguration;
f@0:   result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
f@0:   if ( result != noErr || dataSize == 0 ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Allocate the AudioBufferList.
f@0:   bufferList = (AudioBufferList *) malloc( dataSize );
f@0:   if ( bufferList == NULL ) {
f@0:     errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
f@0:   if (result != noErr || dataSize == 0) {
f@0:     free( bufferList );
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Search for one or more streams that contain the desired number of
f@0:   // channels. CoreAudio devices can have an arbitrary number of
f@0:   // streams and each stream can have an arbitrary number of channels.
f@0:   // For each stream, a single buffer of interleaved samples is
f@0:   // provided.  RtAudio prefers the use of one stream of interleaved
f@0:   // data or multiple consecutive single-channel streams.  However, we
f@0:   // now support multiple consecutive multi-channel streams of
f@0:   // interleaved data as well.
f@0:   UInt32 iStream, offsetCounter = firstChannel;
f@0:   UInt32 nStreams = bufferList->mNumberBuffers;
f@0:   bool monoMode = false;
f@0:   bool foundStream = false;
f@0: 
f@0:   // First check that the device supports the requested number of
f@0:   // channels.
f@0:   UInt32 deviceChannels = 0;
f@0:   for ( iStream=0; iStream<nStreams; iStream++ )
f@0:     deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
f@0: 
f@0:   if ( deviceChannels < ( channels + firstChannel ) ) {
f@0:     free( bufferList );
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Look for a single stream meeting our needs.
f@0:   UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
f@0:   for ( iStream=0; iStream<nStreams; iStream++ ) {
f@0:     streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
f@0:     if ( streamChannels >= channels + offsetCounter ) {
f@0:       firstStream = iStream;
f@0:       channelOffset = offsetCounter;
f@0:       foundStream = true;
f@0:       break;
f@0:     }
f@0:     if ( streamChannels > offsetCounter ) break;
f@0:     offsetCounter -= streamChannels;
f@0:   }
f@0: 
f@0:   // If we didn't find a single stream above, then we should be able
f@0:   // to meet the channel specification with multiple streams.
f@0:   if ( foundStream == false ) {
f@0:     monoMode = true;
f@0:     offsetCounter = firstChannel;
f@0:     for ( iStream=0; iStream<nStreams; iStream++ ) {
f@0:       streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
f@0:       if ( streamChannels > offsetCounter ) break;
f@0:       offsetCounter -= streamChannels;
f@0:     }
f@0: 
f@0:     firstStream = iStream;
f@0:     channelOffset = offsetCounter;
f@0:     Int32 channelCounter = channels + offsetCounter - streamChannels;
f@0: 
f@0:     if ( streamChannels > 1 ) monoMode = false;
f@0:     while ( channelCounter > 0 ) {
f@0:       streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
f@0:       if ( streamChannels > 1 ) monoMode = false;
f@0:       channelCounter -= streamChannels;
f@0:       streamCount++;
f@0:     }
f@0:   }
f@0: 
f@0:   free( bufferList );
f@0: 
f@0:   // Determine the buffer size.
f@0:   AudioValueRange	bufferRange;
f@0:   dataSize = sizeof( AudioValueRange );
f@0:   property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
f@0: 
f@0:   if ( result != noErr ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
f@0:   else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
f@0:   if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
f@0: 
f@0:   // Set the buffer size.  For multiple streams, I'm assuming we only
f@0:   // need to make this setting for the master channel.
f@0:   UInt32 theSize = (UInt32) *bufferSize;
f@0:   dataSize = sizeof( UInt32 );
f@0:   property.mSelector = kAudioDevicePropertyBufferFrameSize;
f@0:   result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
f@0: 
f@0:   if ( result != noErr ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // If attempting to setup a duplex stream, the bufferSize parameter
f@0:   // MUST be the same in both directions!
f@0:   *bufferSize = theSize;
f@0:   if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   stream_.bufferSize = *bufferSize;
f@0:   stream_.nBuffers = 1;
f@0: 
f@0:   // Try to set "hog" mode ... it's not clear to me this is working.
f@0:   if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
f@0:     pid_t hog_pid;
f@0:     dataSize = sizeof( hog_pid );
f@0:     property.mSelector = kAudioDevicePropertyHogMode;
f@0:     result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     if ( hog_pid != getpid() ) {
f@0:       hog_pid = getpid();
f@0:       result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
f@0:       if ( result != noErr ) {
f@0:         errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
f@0:         errorText_ = errorStream_.str();
f@0:         return FAILURE;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   // Check and if necessary, change the sample rate for the device.
f@0:   Float64 nominalRate;
f@0:   dataSize = sizeof( Float64 );
f@0:   property.mSelector = kAudioDevicePropertyNominalSampleRate;
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
f@0:   if ( result != noErr ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Only change the sample rate if off by more than 1 Hz.
f@0:   if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
f@0: 
f@0:     // Set a property listener for the sample rate change
f@0:     Float64 reportedRate = 0.0;
f@0:     AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
f@0:     result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     nominalRate = (Float64) sampleRate;
f@0:     result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
f@0:     if ( result != noErr ) {
f@0:       AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Now wait until the reported nominal rate is what we just set.
f@0:     UInt32 microCounter = 0;
f@0:     while ( reportedRate != nominalRate ) {
f@0:       microCounter += 5000;
f@0:       if ( microCounter > 5000000 ) break;
f@0:       usleep( 5000 );
f@0:     }
f@0: 
f@0:     // Remove the property listener.
f@0:     AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
f@0: 
f@0:     if ( microCounter > 5000000 ) {
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0: 
f@0:   // Now set the stream format for all streams.  Also, check the
f@0:   // physical format of the device and change that if necessary.
f@0:   AudioStreamBasicDescription	description;
f@0:   dataSize = sizeof( AudioStreamBasicDescription );
f@0:   property.mSelector = kAudioStreamPropertyVirtualFormat;
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
f@0:   if ( result != noErr ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Set the sample rate and data format id.  However, only make the
f@0:   // change if the sample rate is not within 1.0 of the desired
f@0:   // rate and the format is not linear pcm.
f@0:   bool updateFormat = false;
f@0:   if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
f@0:     description.mSampleRate = (Float64) sampleRate;
f@0:     updateFormat = true;
f@0:   }
f@0: 
f@0:   if ( description.mFormatID != kAudioFormatLinearPCM ) {
f@0:     description.mFormatID = kAudioFormatLinearPCM;
f@0:     updateFormat = true;
f@0:   }
f@0: 
f@0:   if ( updateFormat ) {
f@0:     result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0: 
f@0:   // Now check the physical format.
f@0:   property.mSelector = kAudioStreamPropertyPhysicalFormat;
f@0:   result = AudioObjectGetPropertyData( id, &property, 0, NULL,  &dataSize, &description );
f@0:   if ( result != noErr ) {
f@0:     errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   //std::cout << "Current physical stream format:" << std::endl;
f@0:   //std::cout << "   mBitsPerChan = " << description.mBitsPerChannel << std::endl;
f@0:   //std::cout << "   aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
f@0:   //std::cout << "   bytesPerFrame = " << description.mBytesPerFrame << std::endl;
f@0:   //std::cout << "   sample rate = " << description.mSampleRate << std::endl;
f@0: 
f@0:   if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
f@0:     description.mFormatID = kAudioFormatLinearPCM;
f@0:     //description.mSampleRate = (Float64) sampleRate;
f@0:     AudioStreamBasicDescription	testDescription = description;
f@0:     UInt32 formatFlags;
f@0: 
f@0:     // We'll try higher bit rates first and then work our way down.
f@0:     std::vector< std::pair<UInt32, UInt32>  > physicalFormats;
f@0:     formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
f@0:     physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
f@0:     formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
f@0:     physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
f@0:     physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) );   // 24-bit packed
f@0:     formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
f@0:     physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
f@0:     formatFlags |= kAudioFormatFlagIsAlignedHigh;
f@0:     physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
f@0:     formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
f@0:     physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
f@0:     physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
f@0: 
f@0:     bool setPhysicalFormat = false;
f@0:     for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
f@0:       testDescription = description;
f@0:       testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
f@0:       testDescription.mFormatFlags = physicalFormats[i].second;
f@0:       if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
f@0:         testDescription.mBytesPerFrame =  4 * testDescription.mChannelsPerFrame;
f@0:       else
f@0:         testDescription.mBytesPerFrame =  testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
f@0:       testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
f@0:       result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
f@0:       if ( result == noErr ) {
f@0:         setPhysicalFormat = true;
f@0:         //std::cout << "Updated physical stream format:" << std::endl;
f@0:         //std::cout << "   mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
f@0:         //std::cout << "   aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
f@0:         //std::cout << "   bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
f@0:         //std::cout << "   sample rate = " << testDescription.mSampleRate << std::endl;
f@0:         break;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( !setPhysicalFormat ) {
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   } // done setting virtual/physical formats.
f@0: 
f@0:   // Get the stream / device latency.
f@0:   UInt32 latency;
f@0:   dataSize = sizeof( UInt32 );
f@0:   property.mSelector = kAudioDevicePropertyLatency;
f@0:   if ( AudioObjectHasProperty( id, &property ) == true ) {
f@0:     result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
f@0:     if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
f@0:     else {
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::WARNING );
f@0:     }
f@0:   }
f@0: 
f@0:   // Byte-swapping: According to AudioHardware.h, the stream data will
f@0:   // always be presented in native-endian format, so we should never
f@0:   // need to byte swap.
f@0:   stream_.doByteSwap[mode] = false;
f@0: 
f@0:   // From the CoreAudio documentation, PCM data must be supplied as
f@0:   // 32-bit floats.
f@0:   stream_.userFormat = format;
f@0:   stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
f@0: 
f@0:   if ( streamCount == 1 )
f@0:     stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
f@0:   else // multiple streams
f@0:     stream_.nDeviceChannels[mode] = channels;
f@0:   stream_.nUserChannels[mode] = channels;
f@0:   stream_.channelOffset[mode] = channelOffset;  // offset within a CoreAudio stream
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
f@0:   else stream_.userInterleaved = true;
f@0:   stream_.deviceInterleaved[mode] = true;
f@0:   if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
f@0: 
f@0:   // Set flags for buffer conversion.
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if ( stream_.userFormat != stream_.deviceFormat[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( streamCount == 1 ) {
f@0:     if ( stream_.nUserChannels[mode] > 1 &&
f@0:          stream_.userInterleaved != stream_.deviceInterleaved[mode] )
f@0:       stream_.doConvertBuffer[mode] = true;
f@0:   }
f@0:   else if ( monoMode && stream_.userInterleaved )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   // Allocate our CoreHandle structure for the stream.
f@0:   CoreHandle *handle = 0;
f@0:   if ( stream_.apiHandle == 0 ) {
f@0:     try {
f@0:       handle = new CoreHandle;
f@0:     }
f@0:     catch ( std::bad_alloc& ) {
f@0:       errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     if ( pthread_cond_init( &handle->condition, NULL ) ) {
f@0:       errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
f@0:       goto error;
f@0:     }
f@0:     stream_.apiHandle = (void *) handle;
f@0:   }
f@0:   else
f@0:     handle = (CoreHandle *) stream_.apiHandle;
f@0:   handle->iStream[mode] = firstStream;
f@0:   handle->nStreams[mode] = streamCount;
f@0:   handle->id[mode] = id;
f@0: 
f@0:   // Allocate necessary internal buffers.
f@0:   unsigned long bufferBytes;
f@0:   bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
f@0:   //  stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
f@0:   stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
f@0:   memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
f@0:   if ( stream_.userBuffer[mode] == NULL ) {
f@0:     errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
f@0:     goto error;
f@0:   }
f@0: 
f@0:   // If possible, we will make use of the CoreAudio stream buffers as
f@0:   // "device buffers".  However, we can't do this if using multiple
f@0:   // streams.
f@0:   if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
f@0: 
f@0:     bool makeBuffer = true;
f@0:     bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
f@0:     if ( mode == INPUT ) {
f@0:       if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
f@0:         unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
f@0:         if ( bufferBytes <= bytesOut ) makeBuffer = false;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( makeBuffer ) {
f@0:       bufferBytes *= *bufferSize;
f@0:       if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
f@0:       stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
f@0:       if ( stream_.deviceBuffer == NULL ) {
f@0:         errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
f@0:         goto error;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.sampleRate = sampleRate;
f@0:   stream_.device[mode] = device;
f@0:   stream_.state = STREAM_STOPPED;
f@0:   stream_.callbackInfo.object = (void *) this;
f@0: 
f@0:   // Setup the buffer conversion information structure.
f@0:   if ( stream_.doConvertBuffer[mode] ) {
f@0:     if ( streamCount > 1 ) setConvertInfo( mode, 0 );
f@0:     else setConvertInfo( mode, channelOffset );
f@0:   }
f@0: 
f@0:   if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
f@0:     // Only one callback procedure per device.
f@0:     stream_.mode = DUPLEX;
f@0:   else {
f@0: #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
f@0:     result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
f@0: #else
f@0:     // deprecated in favor of AudioDeviceCreateIOProcID()
f@0:     result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
f@0: #endif
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto error;
f@0:     }
f@0:     if ( stream_.mode == OUTPUT && mode == INPUT )
f@0:       stream_.mode = DUPLEX;
f@0:     else
f@0:       stream_.mode = mode;
f@0:   }
f@0: 
f@0:   // Setup the device property listener for over/underload.
f@0:   property.mSelector = kAudioDeviceProcessorOverload;
f@0:   property.mScope = kAudioObjectPropertyScopeGlobal;
f@0:   result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
f@0: 
f@0:   return SUCCESS;
f@0: 
f@0:  error:
f@0:   if ( handle ) {
f@0:     pthread_cond_destroy( &handle->condition );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_CLOSED;
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: void RtApiCore :: closeStream( void )
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiCore::closeStream(): no open stream to close!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     if ( stream_.state == STREAM_RUNNING )
f@0:       AudioDeviceStop( handle->id[0], callbackHandler );
f@0: #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
f@0:     AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
f@0: #else
f@0:     // deprecated in favor of AudioDeviceDestroyIOProcID()
f@0:     AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
f@0: #endif
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
f@0:     if ( stream_.state == STREAM_RUNNING )
f@0:       AudioDeviceStop( handle->id[1], callbackHandler );
f@0: #if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
f@0:     AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
f@0: #else
f@0:     // deprecated in favor of AudioDeviceDestroyIOProcID()
f@0:     AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
f@0: #endif
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   // Destroy pthread condition variable.
f@0:   pthread_cond_destroy( &handle->condition );
f@0:   delete handle;
f@0:   stream_.apiHandle = 0;
f@0: 
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.state = STREAM_CLOSED;
f@0: }
f@0: 
f@0: void RtApiCore :: startStream( void )
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiCore::startStream(): the stream is already running!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   OSStatus result = noErr;
f@0:   CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     result = AudioDeviceStart( handle->id[0], callbackHandler );
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT ||
f@0:        ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
f@0: 
f@0:     result = AudioDeviceStart( handle->id[1], callbackHandler );
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   handle->drainCounter = 0;
f@0:   handle->internalDrain = false;
f@0:   stream_.state = STREAM_RUNNING;
f@0: 
f@0:  unlock:
f@0:   if ( result == noErr ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiCore :: stopStream( void )
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   OSStatus result = noErr;
f@0:   CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     if ( handle->drainCounter == 0 ) {
f@0:       handle->drainCounter = 2;
f@0:       pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
f@0:     }
f@0: 
f@0:     result = AudioDeviceStop( handle->id[0], callbackHandler );
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
f@0: 
f@0:     result = AudioDeviceStop( handle->id[1], callbackHandler );
f@0:     if ( result != noErr ) {
f@0:       errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0: 
f@0:  unlock:
f@0:   if ( result == noErr ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiCore :: abortStream( void )
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
f@0:   handle->drainCounter = 2;
f@0: 
f@0:   stopStream();
f@0: }
f@0: 
f@0: // This function will be called by a spawned thread when the user
f@0: // callback function signals that the stream should be stopped or
f@0: // aborted.  It is better to handle it this way because the
f@0: // callbackEvent() function probably should return before the AudioDeviceStop()
f@0: // function is called.
f@0: static void *coreStopStream( void *ptr )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) ptr;
f@0:   RtApiCore *object = (RtApiCore *) info->object;
f@0: 
f@0:   object->stopStream();
f@0:   pthread_exit( NULL );
f@0: }
f@0: 
f@0: bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
f@0:                                  const AudioBufferList *inBufferList,
f@0:                                  const AudioBufferList *outBufferList )
f@0: {
f@0:   if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
f@0:     error( RtAudioError::WARNING );
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
f@0:   CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
f@0: 
f@0:   // Check if we were draining the stream and signal is finished.
f@0:   if ( handle->drainCounter > 3 ) {
f@0:     ThreadHandle threadId;
f@0: 
f@0:     stream_.state = STREAM_STOPPING;
f@0:     if ( handle->internalDrain == true )
f@0:       pthread_create( &threadId, NULL, coreStopStream, info );
f@0:     else // external call to stopStream()
f@0:       pthread_cond_signal( &handle->condition );
f@0:     return SUCCESS;
f@0:   }
f@0: 
f@0:   AudioDeviceID outputDevice = handle->id[0];
f@0: 
f@0:   // Invoke user callback to get fresh output data UNLESS we are
f@0:   // draining stream or duplex mode AND the input/output devices are
f@0:   // different AND this function is called for the input device.
f@0:   if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
f@0:     RtAudioCallback callback = (RtAudioCallback) info->callback;
f@0:     double streamTime = getStreamTime();
f@0:     RtAudioStreamStatus status = 0;
f@0:     if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
f@0:       status |= RTAUDIO_OUTPUT_UNDERFLOW;
f@0:       handle->xrun[0] = false;
f@0:     }
f@0:     if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
f@0:       status |= RTAUDIO_INPUT_OVERFLOW;
f@0:       handle->xrun[1] = false;
f@0:     }
f@0: 
f@0:     int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
f@0:                                   stream_.bufferSize, streamTime, status, info->userData );
f@0:     if ( cbReturnValue == 2 ) {
f@0:       stream_.state = STREAM_STOPPING;
f@0:       handle->drainCounter = 2;
f@0:       abortStream();
f@0:       return SUCCESS;
f@0:     }
f@0:     else if ( cbReturnValue == 1 ) {
f@0:       handle->drainCounter = 1;
f@0:       handle->internalDrain = true;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
f@0: 
f@0:     if ( handle->drainCounter > 1 ) { // write zeros to the output stream
f@0: 
f@0:       if ( handle->nStreams[0] == 1 ) {
f@0:         memset( outBufferList->mBuffers[handle->iStream[0]].mData,
f@0:                 0,
f@0:                 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
f@0:       }
f@0:       else { // fill multiple streams with zeros
f@0:         for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
f@0:           memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
f@0:                   0,
f@0:                   outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
f@0:         }
f@0:       }
f@0:     }
f@0:     else if ( handle->nStreams[0] == 1 ) {
f@0:       if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
f@0:         convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
f@0:                        stream_.userBuffer[0], stream_.convertInfo[0] );
f@0:       }
f@0:       else { // copy from user buffer
f@0:         memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
f@0:                 stream_.userBuffer[0],
f@0:                 outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
f@0:       }
f@0:     }
f@0:     else { // fill multiple streams
f@0:       Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
f@0:       if ( stream_.doConvertBuffer[0] ) {
f@0:         convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
f@0:         inBuffer = (Float32 *) stream_.deviceBuffer;
f@0:       }
f@0: 
f@0:       if ( stream_.deviceInterleaved[0] == false ) { // mono mode
f@0:         UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
f@0:         for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
f@0:           memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
f@0:                   (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
f@0:         }
f@0:       }
f@0:       else { // fill multiple multi-channel streams with interleaved data
f@0:         UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
f@0:         Float32 *out, *in;
f@0: 
f@0:         bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
f@0:         UInt32 inChannels = stream_.nUserChannels[0];
f@0:         if ( stream_.doConvertBuffer[0] ) {
f@0:           inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
f@0:           inChannels = stream_.nDeviceChannels[0];
f@0:         }
f@0: 
f@0:         if ( inInterleaved ) inOffset = 1;
f@0:         else inOffset = stream_.bufferSize;
f@0: 
f@0:         channelsLeft = inChannels;
f@0:         for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
f@0:           in = inBuffer;
f@0:           out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
f@0:           streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
f@0: 
f@0:           outJump = 0;
f@0:           // Account for possible channel offset in first stream
f@0:           if ( i == 0 && stream_.channelOffset[0] > 0 ) {
f@0:             streamChannels -= stream_.channelOffset[0];
f@0:             outJump = stream_.channelOffset[0];
f@0:             out += outJump;
f@0:           }
f@0: 
f@0:           // Account for possible unfilled channels at end of the last stream
f@0:           if ( streamChannels > channelsLeft ) {
f@0:             outJump = streamChannels - channelsLeft;
f@0:             streamChannels = channelsLeft;
f@0:           }
f@0: 
f@0:           // Determine input buffer offsets and skips
f@0:           if ( inInterleaved ) {
f@0:             inJump = inChannels;
f@0:             in += inChannels - channelsLeft;
f@0:           }
f@0:           else {
f@0:             inJump = 1;
f@0:             in += (inChannels - channelsLeft) * inOffset;
f@0:           }
f@0: 
f@0:           for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
f@0:             for ( unsigned int j=0; j<streamChannels; j++ ) {
f@0:               *out++ = in[j*inOffset];
f@0:             }
f@0:             out += outJump;
f@0:             in += inJump;
f@0:           }
f@0:           channelsLeft -= streamChannels;
f@0:         }
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   // Don't bother draining input
f@0:   if ( handle->drainCounter ) {
f@0:     handle->drainCounter++;
f@0:     goto unlock;
f@0:   }
f@0: 
f@0:   AudioDeviceID inputDevice;
f@0:   inputDevice = handle->id[1];
f@0:   if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
f@0: 
f@0:     if ( handle->nStreams[1] == 1 ) {
f@0:       if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
f@0:         convertBuffer( stream_.userBuffer[1],
f@0:                        (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
f@0:                        stream_.convertInfo[1] );
f@0:       }
f@0:       else { // copy to user buffer
f@0:         memcpy( stream_.userBuffer[1],
f@0:                 inBufferList->mBuffers[handle->iStream[1]].mData,
f@0:                 inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
f@0:       }
f@0:     }
f@0:     else { // read from multiple streams
f@0:       Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
f@0:       if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
f@0: 
f@0:       if ( stream_.deviceInterleaved[1] == false ) { // mono mode
f@0:         UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
f@0:         for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
f@0:           memcpy( (void *)&outBuffer[i*stream_.bufferSize],
f@0:                   inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
f@0:         }
f@0:       }
f@0:       else { // read from multiple multi-channel streams
f@0:         UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
f@0:         Float32 *out, *in;
f@0: 
f@0:         bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
f@0:         UInt32 outChannels = stream_.nUserChannels[1];
f@0:         if ( stream_.doConvertBuffer[1] ) {
f@0:           outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
f@0:           outChannels = stream_.nDeviceChannels[1];
f@0:         }
f@0: 
f@0:         if ( outInterleaved ) outOffset = 1;
f@0:         else outOffset = stream_.bufferSize;
f@0: 
f@0:         channelsLeft = outChannels;
f@0:         for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
f@0:           out = outBuffer;
f@0:           in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
f@0:           streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
f@0: 
f@0:           inJump = 0;
f@0:           // Account for possible channel offset in first stream
f@0:           if ( i == 0 && stream_.channelOffset[1] > 0 ) {
f@0:             streamChannels -= stream_.channelOffset[1];
f@0:             inJump = stream_.channelOffset[1];
f@0:             in += inJump;
f@0:           }
f@0: 
f@0:           // Account for possible unread channels at end of the last stream
f@0:           if ( streamChannels > channelsLeft ) {
f@0:             inJump = streamChannels - channelsLeft;
f@0:             streamChannels = channelsLeft;
f@0:           }
f@0: 
f@0:           // Determine output buffer offsets and skips
f@0:           if ( outInterleaved ) {
f@0:             outJump = outChannels;
f@0:             out += outChannels - channelsLeft;
f@0:           }
f@0:           else {
f@0:             outJump = 1;
f@0:             out += (outChannels - channelsLeft) * outOffset;
f@0:           }
f@0: 
f@0:           for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
f@0:             for ( unsigned int j=0; j<streamChannels; j++ ) {
f@0:               out[j*outOffset] = *in++;
f@0:             }
f@0:             out += outJump;
f@0:             in += inJump;
f@0:           }
f@0:           channelsLeft -= streamChannels;
f@0:         }
f@0:       }
f@0:       
f@0:       if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
f@0:         convertBuffer( stream_.userBuffer[1],
f@0:                        stream_.deviceBuffer,
f@0:                        stream_.convertInfo[1] );
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   //MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   RtApi::tickStreamTime();
f@0:   return SUCCESS;
f@0: }
f@0: 
f@0: const char* RtApiCore :: getErrorCode( OSStatus code )
f@0: {
f@0:   switch( code ) {
f@0: 
f@0:   case kAudioHardwareNotRunningError:
f@0:     return "kAudioHardwareNotRunningError";
f@0: 
f@0:   case kAudioHardwareUnspecifiedError:
f@0:     return "kAudioHardwareUnspecifiedError";
f@0: 
f@0:   case kAudioHardwareUnknownPropertyError:
f@0:     return "kAudioHardwareUnknownPropertyError";
f@0: 
f@0:   case kAudioHardwareBadPropertySizeError:
f@0:     return "kAudioHardwareBadPropertySizeError";
f@0: 
f@0:   case kAudioHardwareIllegalOperationError:
f@0:     return "kAudioHardwareIllegalOperationError";
f@0: 
f@0:   case kAudioHardwareBadObjectError:
f@0:     return "kAudioHardwareBadObjectError";
f@0: 
f@0:   case kAudioHardwareBadDeviceError:
f@0:     return "kAudioHardwareBadDeviceError";
f@0: 
f@0:   case kAudioHardwareBadStreamError:
f@0:     return "kAudioHardwareBadStreamError";
f@0: 
f@0:   case kAudioHardwareUnsupportedOperationError:
f@0:     return "kAudioHardwareUnsupportedOperationError";
f@0: 
f@0:   case kAudioDeviceUnsupportedFormatError:
f@0:     return "kAudioDeviceUnsupportedFormatError";
f@0: 
f@0:   case kAudioDevicePermissionsError:
f@0:     return "kAudioDevicePermissionsError";
f@0: 
f@0:   default:
f@0:     return "CoreAudio unknown error";
f@0:   }
f@0: }
f@0: 
f@0:   //******************** End of __MACOSX_CORE__ *********************//
f@0: #endif
f@0: 
f@0: #if defined(__UNIX_JACK__)
f@0: 
f@0: // JACK is a low-latency audio server, originally written for the
f@0: // GNU/Linux operating system and now also ported to OS-X. It can
f@0: // connect a number of different applications to an audio device, as
f@0: // well as allowing them to share audio between themselves.
f@0: //
f@0: // When using JACK with RtAudio, "devices" refer to JACK clients that
f@0: // have ports connected to the server.  The JACK server is typically
f@0: // started in a terminal as follows:
f@0: //
f@0: // .jackd -d alsa -d hw:0
f@0: //
f@0: // or through an interface program such as qjackctl.  Many of the
f@0: // parameters normally set for a stream are fixed by the JACK server
f@0: // and can be specified when the JACK server is started.  In
f@0: // particular,
f@0: //
f@0: // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
f@0: //
f@0: // specifies a sample rate of 44100 Hz, a buffer size of 512 sample
f@0: // frames, and number of buffers = 4.  Once the server is running, it
f@0: // is not possible to override these values.  If the values are not
f@0: // specified in the command-line, the JACK server uses default values.
f@0: //
f@0: // The JACK server does not have to be running when an instance of
f@0: // RtApiJack is created, though the function getDeviceCount() will
f@0: // report 0 devices found until JACK has been started.  When no
f@0: // devices are available (i.e., the JACK server is not running), a
f@0: // stream cannot be opened.
f@0: 
f@0: #include <jack/jack.h>
f@0: #include <unistd.h>
f@0: #include <cstdio>
f@0: 
f@0: // A structure to hold various information related to the Jack API
f@0: // implementation.
f@0: struct JackHandle {
f@0:   jack_client_t *client;
f@0:   jack_port_t **ports[2];
f@0:   std::string deviceName[2];
f@0:   bool xrun[2];
f@0:   pthread_cond_t condition;
f@0:   int drainCounter;       // Tracks callback counts when draining
f@0:   bool internalDrain;     // Indicates if stop is initiated from callback or not.
f@0: 
f@0:   JackHandle()
f@0:     :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
f@0: };
f@0: 
f@0: static void jackSilentError( const char * ) {};
f@0: 
f@0: RtApiJack :: RtApiJack()
f@0: {
f@0:   // Nothing to do here.
f@0: #if !defined(__RTAUDIO_DEBUG__)
f@0:   // Turn off Jack's internal error reporting.
f@0:   jack_set_error_function( &jackSilentError );
f@0: #endif
f@0: }
f@0: 
f@0: RtApiJack :: ~RtApiJack()
f@0: {
f@0:   if ( stream_.state != STREAM_CLOSED ) closeStream();
f@0: }
f@0: 
f@0: unsigned int RtApiJack :: getDeviceCount( void )
f@0: {
f@0:   // See if we can become a jack client.
f@0:   jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
f@0:   jack_status_t *status = NULL;
f@0:   jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
f@0:   if ( client == 0 ) return 0;
f@0: 
f@0:   const char **ports;
f@0:   std::string port, previousPort;
f@0:   unsigned int nChannels = 0, nDevices = 0;
f@0:   ports = jack_get_ports( client, NULL, NULL, 0 );
f@0:   if ( ports ) {
f@0:     // Parse the port names up to the first colon (:).
f@0:     size_t iColon = 0;
f@0:     do {
f@0:       port = (char *) ports[ nChannels ];
f@0:       iColon = port.find(":");
f@0:       if ( iColon != std::string::npos ) {
f@0:         port = port.substr( 0, iColon + 1 );
f@0:         if ( port != previousPort ) {
f@0:           nDevices++;
f@0:           previousPort = port;
f@0:         }
f@0:       }
f@0:     } while ( ports[++nChannels] );
f@0:     free( ports );
f@0:   }
f@0: 
f@0:   jack_client_close( client );
f@0:   return nDevices;
f@0: }
f@0: 
f@0: RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   info.probed = false;
f@0: 
f@0:   jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
f@0:   jack_status_t *status = NULL;
f@0:   jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
f@0:   if ( client == 0 ) {
f@0:     errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   const char **ports;
f@0:   std::string port, previousPort;
f@0:   unsigned int nPorts = 0, nDevices = 0;
f@0:   ports = jack_get_ports( client, NULL, NULL, 0 );
f@0:   if ( ports ) {
f@0:     // Parse the port names up to the first colon (:).
f@0:     size_t iColon = 0;
f@0:     do {
f@0:       port = (char *) ports[ nPorts ];
f@0:       iColon = port.find(":");
f@0:       if ( iColon != std::string::npos ) {
f@0:         port = port.substr( 0, iColon );
f@0:         if ( port != previousPort ) {
f@0:           if ( nDevices == device ) info.name = port;
f@0:           nDevices++;
f@0:           previousPort = port;
f@0:         }
f@0:       }
f@0:     } while ( ports[++nPorts] );
f@0:     free( ports );
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     jack_client_close( client );
f@0:     errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Get the current jack server sample rate.
f@0:   info.sampleRates.clear();
f@0:   info.sampleRates.push_back( jack_get_sample_rate( client ) );
f@0: 
f@0:   // Count the available ports containing the client name as device
f@0:   // channels.  Jack "input ports" equal RtAudio output channels.
f@0:   unsigned int nChannels = 0;
f@0:   ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
f@0:   if ( ports ) {
f@0:     while ( ports[ nChannels ] ) nChannels++;
f@0:     free( ports );
f@0:     info.outputChannels = nChannels;
f@0:   }
f@0: 
f@0:   // Jack "output ports" equal RtAudio input channels.
f@0:   nChannels = 0;
f@0:   ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
f@0:   if ( ports ) {
f@0:     while ( ports[ nChannels ] ) nChannels++;
f@0:     free( ports );
f@0:     info.inputChannels = nChannels;
f@0:   }
f@0: 
f@0:   if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
f@0:     jack_client_close(client);
f@0:     errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // If device opens for both playback and capture, we determine the channels.
f@0:   if ( info.outputChannels > 0 && info.inputChannels > 0 )
f@0:     info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
f@0: 
f@0:   // Jack always uses 32-bit floats.
f@0:   info.nativeFormats = RTAUDIO_FLOAT32;
f@0: 
f@0:   // Jack doesn't provide default devices so we'll use the first available one.
f@0:   if ( device == 0 && info.outputChannels > 0 )
f@0:     info.isDefaultOutput = true;
f@0:   if ( device == 0 && info.inputChannels > 0 )
f@0:     info.isDefaultInput = true;
f@0: 
f@0:   jack_client_close(client);
f@0:   info.probed = true;
f@0:   return info;
f@0: }
f@0: 
f@0: static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) infoPointer;
f@0: 
f@0:   RtApiJack *object = (RtApiJack *) info->object;
f@0:   if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: // This function will be called by a spawned thread when the Jack
f@0: // server signals that it is shutting down.  It is necessary to handle
f@0: // it this way because the jackShutdown() function must return before
f@0: // the jack_deactivate() function (in closeStream()) will return.
f@0: static void *jackCloseStream( void *ptr )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) ptr;
f@0:   RtApiJack *object = (RtApiJack *) info->object;
f@0: 
f@0:   object->closeStream();
f@0: 
f@0:   pthread_exit( NULL );
f@0: }
f@0: static void jackShutdown( void *infoPointer )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) infoPointer;
f@0:   RtApiJack *object = (RtApiJack *) info->object;
f@0: 
f@0:   // Check current stream state.  If stopped, then we'll assume this
f@0:   // was called as a result of a call to RtApiJack::stopStream (the
f@0:   // deactivation of a client handle causes this function to be called).
f@0:   // If not, we'll assume the Jack server is shutting down or some
f@0:   // other problem occurred and we should close the stream.
f@0:   if ( object->isStreamRunning() == false ) return;
f@0: 
f@0:   ThreadHandle threadId;
f@0:   pthread_create( &threadId, NULL, jackCloseStream, info );
f@0:   std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
f@0: }
f@0: 
f@0: static int jackXrun( void *infoPointer )
f@0: {
f@0:   JackHandle *handle = (JackHandle *) infoPointer;
f@0: 
f@0:   if ( handle->ports[0] ) handle->xrun[0] = true;
f@0:   if ( handle->ports[1] ) handle->xrun[1] = true;
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
f@0:                                    unsigned int firstChannel, unsigned int sampleRate,
f@0:                                    RtAudioFormat format, unsigned int *bufferSize,
f@0:                                    RtAudio::StreamOptions *options )
f@0: {
f@0:   JackHandle *handle = (JackHandle *) stream_.apiHandle;
f@0: 
f@0:   // Look for jack server and try to become a client (only do once per stream).
f@0:   jack_client_t *client = 0;
f@0:   if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
f@0:     jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
f@0:     jack_status_t *status = NULL;
f@0:     if ( options && !options->streamName.empty() )
f@0:       client = jack_client_open( options->streamName.c_str(), jackoptions, status );
f@0:     else
f@0:       client = jack_client_open( "RtApiJack", jackoptions, status );
f@0:     if ( client == 0 ) {
f@0:       errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
f@0:       error( RtAudioError::WARNING );
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0:   else {
f@0:     // The handle must have been created on an earlier pass.
f@0:     client = handle->client;
f@0:   }
f@0: 
f@0:   const char **ports;
f@0:   std::string port, previousPort, deviceName;
f@0:   unsigned int nPorts = 0, nDevices = 0;
f@0:   ports = jack_get_ports( client, NULL, NULL, 0 );
f@0:   if ( ports ) {
f@0:     // Parse the port names up to the first colon (:).
f@0:     size_t iColon = 0;
f@0:     do {
f@0:       port = (char *) ports[ nPorts ];
f@0:       iColon = port.find(":");
f@0:       if ( iColon != std::string::npos ) {
f@0:         port = port.substr( 0, iColon );
f@0:         if ( port != previousPort ) {
f@0:           if ( nDevices == device ) deviceName = port;
f@0:           nDevices++;
f@0:           previousPort = port;
f@0:         }
f@0:       }
f@0:     } while ( ports[++nPorts] );
f@0:     free( ports );
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Count the available ports containing the client name as device
f@0:   // channels.  Jack "input ports" equal RtAudio output channels.
f@0:   unsigned int nChannels = 0;
f@0:   unsigned long flag = JackPortIsInput;
f@0:   if ( mode == INPUT ) flag = JackPortIsOutput;
f@0:   ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
f@0:   if ( ports ) {
f@0:     while ( ports[ nChannels ] ) nChannels++;
f@0:     free( ports );
f@0:   }
f@0: 
f@0:   // Compare the jack ports for specified client to the requested number of channels.
f@0:   if ( nChannels < (channels + firstChannel) ) {
f@0:     errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Check the jack server sample rate.
f@0:   unsigned int jackRate = jack_get_sample_rate( client );
f@0:   if ( sampleRate != jackRate ) {
f@0:     jack_client_close( client );
f@0:     errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0:   stream_.sampleRate = jackRate;
f@0: 
f@0:   // Get the latency of the JACK port.
f@0:   ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
f@0:   if ( ports[ firstChannel ] ) {
f@0:     // Added by Ge Wang
f@0:     jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
f@0:     // the range (usually the min and max are equal)
f@0:     jack_latency_range_t latrange; latrange.min = latrange.max = 0;
f@0:     // get the latency range
f@0:     jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
f@0:     // be optimistic, use the min!
f@0:     stream_.latency[mode] = latrange.min;
f@0:     //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
f@0:   }
f@0:   free( ports );
f@0: 
f@0:   // The jack server always uses 32-bit floating-point data.
f@0:   stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
f@0:   stream_.userFormat = format;
f@0: 
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
f@0:   else stream_.userInterleaved = true;
f@0: 
f@0:   // Jack always uses non-interleaved buffers.
f@0:   stream_.deviceInterleaved[mode] = false;
f@0: 
f@0:   // Jack always provides host byte-ordered data.
f@0:   stream_.doByteSwap[mode] = false;
f@0: 
f@0:   // Get the buffer size.  The buffer size and number of buffers
f@0:   // (periods) is set when the jack server is started.
f@0:   stream_.bufferSize = (int) jack_get_buffer_size( client );
f@0:   *bufferSize = stream_.bufferSize;
f@0: 
f@0:   stream_.nDeviceChannels[mode] = channels;
f@0:   stream_.nUserChannels[mode] = channels;
f@0: 
f@0:   // Set flags for buffer conversion.
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if ( stream_.userFormat != stream_.deviceFormat[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
f@0:        stream_.nUserChannels[mode] > 1 )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   // Allocate our JackHandle structure for the stream.
f@0:   if ( handle == 0 ) {
f@0:     try {
f@0:       handle = new JackHandle;
f@0:     }
f@0:     catch ( std::bad_alloc& ) {
f@0:       errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     if ( pthread_cond_init(&handle->condition, NULL) ) {
f@0:       errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
f@0:       goto error;
f@0:     }
f@0:     stream_.apiHandle = (void *) handle;
f@0:     handle->client = client;
f@0:   }
f@0:   handle->deviceName[mode] = deviceName;
f@0: 
f@0:   // Allocate necessary internal buffers.
f@0:   unsigned long bufferBytes;
f@0:   bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
f@0:   stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
f@0:   if ( stream_.userBuffer[mode] == NULL ) {
f@0:     errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
f@0:     goto error;
f@0:   }
f@0: 
f@0:   if ( stream_.doConvertBuffer[mode] ) {
f@0: 
f@0:     bool makeBuffer = true;
f@0:     if ( mode == OUTPUT )
f@0:       bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
f@0:     else { // mode == INPUT
f@0:       bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
f@0:       if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
f@0:         unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
f@0:         if ( bufferBytes < bytesOut ) makeBuffer = false;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( makeBuffer ) {
f@0:       bufferBytes *= *bufferSize;
f@0:       if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
f@0:       stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
f@0:       if ( stream_.deviceBuffer == NULL ) {
f@0:         errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
f@0:         goto error;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   // Allocate memory for the Jack ports (channels) identifiers.
f@0:   handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
f@0:   if ( handle->ports[mode] == NULL )  {
f@0:     errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
f@0:     goto error;
f@0:   }
f@0: 
f@0:   stream_.device[mode] = device;
f@0:   stream_.channelOffset[mode] = firstChannel;
f@0:   stream_.state = STREAM_STOPPED;
f@0:   stream_.callbackInfo.object = (void *) this;
f@0: 
f@0:   if ( stream_.mode == OUTPUT && mode == INPUT )
f@0:     // We had already set up the stream for output.
f@0:     stream_.mode = DUPLEX;
f@0:   else {
f@0:     stream_.mode = mode;
f@0:     jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
f@0:     jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
f@0:     jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
f@0:   }
f@0: 
f@0:   // Register our ports.
f@0:   char label[64];
f@0:   if ( mode == OUTPUT ) {
f@0:     for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
f@0:       snprintf( label, 64, "outport %d", i );
f@0:       handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
f@0:                                                 JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
f@0:     }
f@0:   }
f@0:   else {
f@0:     for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
f@0:       snprintf( label, 64, "inport %d", i );
f@0:       handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
f@0:                                                 JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
f@0:     }
f@0:   }
f@0: 
f@0:   // Setup the buffer conversion information structure.  We don't use
f@0:   // buffers to do channel offsets, so we override that parameter
f@0:   // here.
f@0:   if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
f@0: 
f@0:   return SUCCESS;
f@0: 
f@0:  error:
f@0:   if ( handle ) {
f@0:     pthread_cond_destroy( &handle->condition );
f@0:     jack_client_close( handle->client );
f@0: 
f@0:     if ( handle->ports[0] ) free( handle->ports[0] );
f@0:     if ( handle->ports[1] ) free( handle->ports[1] );
f@0: 
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: void RtApiJack :: closeStream( void )
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiJack::closeStream(): no open stream to close!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   JackHandle *handle = (JackHandle *) stream_.apiHandle;
f@0:   if ( handle ) {
f@0: 
f@0:     if ( stream_.state == STREAM_RUNNING )
f@0:       jack_deactivate( handle->client );
f@0: 
f@0:     jack_client_close( handle->client );
f@0:   }
f@0: 
f@0:   if ( handle ) {
f@0:     if ( handle->ports[0] ) free( handle->ports[0] );
f@0:     if ( handle->ports[1] ) free( handle->ports[1] );
f@0:     pthread_cond_destroy( &handle->condition );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.state = STREAM_CLOSED;
f@0: }
f@0: 
f@0: void RtApiJack :: startStream( void )
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiJack::startStream(): the stream is already running!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   JackHandle *handle = (JackHandle *) stream_.apiHandle;
f@0:   int result = jack_activate( handle->client );
f@0:   if ( result ) {
f@0:     errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
f@0:     goto unlock;
f@0:   }
f@0: 
f@0:   const char **ports;
f@0: 
f@0:   // Get the list of available ports.
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     result = 1;
f@0:     ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
f@0:     if ( ports == NULL) {
f@0:       errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Now make the port connections.  Since RtAudio wasn't designed to
f@0:     // allow the user to select particular channels of a device, we'll
f@0:     // just open the first "nChannels" ports with offset.
f@0:     for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
f@0:       result = 1;
f@0:       if ( ports[ stream_.channelOffset[0] + i ] )
f@0:         result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
f@0:       if ( result ) {
f@0:         free( ports );
f@0:         errorText_ = "RtApiJack::startStream(): error connecting output ports!";
f@0:         goto unlock;
f@0:       }
f@0:     }
f@0:     free(ports);
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0:     result = 1;
f@0:     ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
f@0:     if ( ports == NULL) {
f@0:       errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Now make the port connections.  See note above.
f@0:     for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
f@0:       result = 1;
f@0:       if ( ports[ stream_.channelOffset[1] + i ] )
f@0:         result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
f@0:       if ( result ) {
f@0:         free( ports );
f@0:         errorText_ = "RtApiJack::startStream(): error connecting input ports!";
f@0:         goto unlock;
f@0:       }
f@0:     }
f@0:     free(ports);
f@0:   }
f@0: 
f@0:   handle->drainCounter = 0;
f@0:   handle->internalDrain = false;
f@0:   stream_.state = STREAM_RUNNING;
f@0: 
f@0:  unlock:
f@0:   if ( result == 0 ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiJack :: stopStream( void )
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   JackHandle *handle = (JackHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     if ( handle->drainCounter == 0 ) {
f@0:       handle->drainCounter = 2;
f@0:       pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
f@0:     }
f@0:   }
f@0: 
f@0:   jack_deactivate( handle->client );
f@0:   stream_.state = STREAM_STOPPED;
f@0: }
f@0: 
f@0: void RtApiJack :: abortStream( void )
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   JackHandle *handle = (JackHandle *) stream_.apiHandle;
f@0:   handle->drainCounter = 2;
f@0: 
f@0:   stopStream();
f@0: }
f@0: 
f@0: // This function will be called by a spawned thread when the user
f@0: // callback function signals that the stream should be stopped or
f@0: // aborted.  It is necessary to handle it this way because the
f@0: // callbackEvent() function must return before the jack_deactivate()
f@0: // function will return.
f@0: static void *jackStopStream( void *ptr )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) ptr;
f@0:   RtApiJack *object = (RtApiJack *) info->object;
f@0: 
f@0:   object->stopStream();
f@0:   pthread_exit( NULL );
f@0: }
f@0: 
f@0: bool RtApiJack :: callbackEvent( unsigned long nframes )
f@0: {
f@0:   if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
f@0:     error( RtAudioError::WARNING );
f@0:     return FAILURE;
f@0:   }
f@0:   if ( stream_.bufferSize != nframes ) {
f@0:     errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
f@0:     error( RtAudioError::WARNING );
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
f@0:   JackHandle *handle = (JackHandle *) stream_.apiHandle;
f@0: 
f@0:   // Check if we were draining the stream and signal is finished.
f@0:   if ( handle->drainCounter > 3 ) {
f@0:     ThreadHandle threadId;
f@0: 
f@0:     stream_.state = STREAM_STOPPING;
f@0:     if ( handle->internalDrain == true )
f@0:       pthread_create( &threadId, NULL, jackStopStream, info );
f@0:     else
f@0:       pthread_cond_signal( &handle->condition );
f@0:     return SUCCESS;
f@0:   }
f@0: 
f@0:   // Invoke user callback first, to get fresh output data.
f@0:   if ( handle->drainCounter == 0 ) {
f@0:     RtAudioCallback callback = (RtAudioCallback) info->callback;
f@0:     double streamTime = getStreamTime();
f@0:     RtAudioStreamStatus status = 0;
f@0:     if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
f@0:       status |= RTAUDIO_OUTPUT_UNDERFLOW;
f@0:       handle->xrun[0] = false;
f@0:     }
f@0:     if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
f@0:       status |= RTAUDIO_INPUT_OVERFLOW;
f@0:       handle->xrun[1] = false;
f@0:     }
f@0:     int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
f@0:                                   stream_.bufferSize, streamTime, status, info->userData );
f@0:     if ( cbReturnValue == 2 ) {
f@0:       stream_.state = STREAM_STOPPING;
f@0:       handle->drainCounter = 2;
f@0:       ThreadHandle id;
f@0:       pthread_create( &id, NULL, jackStopStream, info );
f@0:       return SUCCESS;
f@0:     }
f@0:     else if ( cbReturnValue == 1 ) {
f@0:       handle->drainCounter = 1;
f@0:       handle->internalDrain = true;
f@0:     }
f@0:   }
f@0: 
f@0:   jack_default_audio_sample_t *jackbuffer;
f@0:   unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     if ( handle->drainCounter > 1 ) { // write zeros to the output stream
f@0: 
f@0:       for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
f@0:         jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
f@0:         memset( jackbuffer, 0, bufferBytes );
f@0:       }
f@0: 
f@0:     }
f@0:     else if ( stream_.doConvertBuffer[0] ) {
f@0: 
f@0:       convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
f@0: 
f@0:       for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
f@0:         jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
f@0:         memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
f@0:       }
f@0:     }
f@0:     else { // no buffer conversion
f@0:       for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
f@0:         jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
f@0:         memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   // Don't bother draining input
f@0:   if ( handle->drainCounter ) {
f@0:     handle->drainCounter++;
f@0:     goto unlock;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     if ( stream_.doConvertBuffer[1] ) {
f@0:       for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
f@0:         jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
f@0:         memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
f@0:       }
f@0:       convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
f@0:     }
f@0:     else { // no buffer conversion
f@0:       for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
f@0:         jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
f@0:         memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   RtApi::tickStreamTime();
f@0:   return SUCCESS;
f@0: }
f@0:   //******************** End of __UNIX_JACK__ *********************//
f@0: #endif
f@0: 
f@0: #if defined(__WINDOWS_ASIO__) // ASIO API on Windows
f@0: 
f@0: // The ASIO API is designed around a callback scheme, so this
f@0: // implementation is similar to that used for OS-X CoreAudio and Linux
f@0: // Jack.  The primary constraint with ASIO is that it only allows
f@0: // access to a single driver at a time.  Thus, it is not possible to
f@0: // have more than one simultaneous RtAudio stream.
f@0: //
f@0: // This implementation also requires a number of external ASIO files
f@0: // and a few global variables.  The ASIO callback scheme does not
f@0: // allow for the passing of user data, so we must create a global
f@0: // pointer to our callbackInfo structure.
f@0: //
f@0: // On unix systems, we make use of a pthread condition variable.
f@0: // Since there is no equivalent in Windows, I hacked something based
f@0: // on information found in
f@0: // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
f@0: 
f@0: #include "asiosys.h"
f@0: #include "asio.h"
f@0: #include "iasiothiscallresolver.h"
f@0: #include "asiodrivers.h"
f@0: #include <cmath>
f@0: 
f@0: static AsioDrivers drivers;
f@0: static ASIOCallbacks asioCallbacks;
f@0: static ASIODriverInfo driverInfo;
f@0: static CallbackInfo *asioCallbackInfo;
f@0: static bool asioXRun;
f@0: 
f@0: struct AsioHandle {
f@0:   int drainCounter;       // Tracks callback counts when draining
f@0:   bool internalDrain;     // Indicates if stop is initiated from callback or not.
f@0:   ASIOBufferInfo *bufferInfos;
f@0:   HANDLE condition;
f@0: 
f@0:   AsioHandle()
f@0:     :drainCounter(0), internalDrain(false), bufferInfos(0) {}
f@0: };
f@0: 
f@0: // Function declarations (definitions at end of section)
f@0: static const char* getAsioErrorString( ASIOError result );
f@0: static void sampleRateChanged( ASIOSampleRate sRate );
f@0: static long asioMessages( long selector, long value, void* message, double* opt );
f@0: 
f@0: RtApiAsio :: RtApiAsio()
f@0: {
f@0:   // ASIO cannot run on a multi-threaded appartment. You can call
f@0:   // CoInitialize beforehand, but it must be for appartment threading
f@0:   // (in which case, CoInitilialize will return S_FALSE here).
f@0:   coInitialized_ = false;
f@0:   HRESULT hr = CoInitialize( NULL ); 
f@0:   if ( FAILED(hr) ) {
f@0:     errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
f@0:     error( RtAudioError::WARNING );
f@0:   }
f@0:   coInitialized_ = true;
f@0: 
f@0:   drivers.removeCurrentDriver();
f@0:   driverInfo.asioVersion = 2;
f@0: 
f@0:   // See note in DirectSound implementation about GetDesktopWindow().
f@0:   driverInfo.sysRef = GetForegroundWindow();
f@0: }
f@0: 
f@0: RtApiAsio :: ~RtApiAsio()
f@0: {
f@0:   if ( stream_.state != STREAM_CLOSED ) closeStream();
f@0:   if ( coInitialized_ ) CoUninitialize();
f@0: }
f@0: 
f@0: unsigned int RtApiAsio :: getDeviceCount( void )
f@0: {
f@0:   return (unsigned int) drivers.asioGetNumDev();
f@0: }
f@0: 
f@0: RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   info.probed = false;
f@0: 
f@0:   // Get device ID
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   if ( nDevices == 0 ) {
f@0:     errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // If a stream is already open, we cannot probe other devices.  Thus, use the saved results.
f@0:   if ( stream_.state != STREAM_CLOSED ) {
f@0:     if ( device >= devices_.size() ) {
f@0:       errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
f@0:       error( RtAudioError::WARNING );
f@0:       return info;
f@0:     }
f@0:     return devices_[ device ];
f@0:   }
f@0: 
f@0:   char driverName[32];
f@0:   ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
f@0:   if ( result != ASE_OK ) {
f@0:     errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   info.name = driverName;
f@0: 
f@0:   if ( !drivers.loadDriver( driverName ) ) {
f@0:     errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   result = ASIOInit( &driverInfo );
f@0:   if ( result != ASE_OK ) {
f@0:     errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Determine the device channel information.
f@0:   long inputChannels, outputChannels;
f@0:   result = ASIOGetChannels( &inputChannels, &outputChannels );
f@0:   if ( result != ASE_OK ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   info.outputChannels = outputChannels;
f@0:   info.inputChannels = inputChannels;
f@0:   if ( info.outputChannels > 0 && info.inputChannels > 0 )
f@0:     info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
f@0: 
f@0:   // Determine the supported sample rates.
f@0:   info.sampleRates.clear();
f@0:   for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
f@0:     result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
f@0:     if ( result == ASE_OK )
f@0:       info.sampleRates.push_back( SAMPLE_RATES[i] );
f@0:   }
f@0: 
f@0:   // Determine supported data types ... just check first channel and assume rest are the same.
f@0:   ASIOChannelInfo channelInfo;
f@0:   channelInfo.channel = 0;
f@0:   channelInfo.isInput = true;
f@0:   if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
f@0:   result = ASIOGetChannelInfo( &channelInfo );
f@0:   if ( result != ASE_OK ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   info.nativeFormats = 0;
f@0:   if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
f@0:     info.nativeFormats |= RTAUDIO_SINT16;
f@0:   else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
f@0:     info.nativeFormats |= RTAUDIO_SINT32;
f@0:   else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
f@0:     info.nativeFormats |= RTAUDIO_FLOAT32;
f@0:   else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
f@0:     info.nativeFormats |= RTAUDIO_FLOAT64;
f@0:   else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
f@0:     info.nativeFormats |= RTAUDIO_SINT24;
f@0: 
f@0:   if ( info.outputChannels > 0 )
f@0:     if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
f@0:   if ( info.inputChannels > 0 )
f@0:     if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
f@0: 
f@0:   info.probed = true;
f@0:   drivers.removeCurrentDriver();
f@0:   return info;
f@0: }
f@0: 
f@0: static void bufferSwitch( long index, ASIOBool /*processNow*/ )
f@0: {
f@0:   RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
f@0:   object->callbackEvent( index );
f@0: }
f@0: 
f@0: void RtApiAsio :: saveDeviceInfo( void )
f@0: {
f@0:   devices_.clear();
f@0: 
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   devices_.resize( nDevices );
f@0:   for ( unsigned int i=0; i<nDevices; i++ )
f@0:     devices_[i] = getDeviceInfo( i );
f@0: }
f@0: 
f@0: bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
f@0:                                    unsigned int firstChannel, unsigned int sampleRate,
f@0:                                    RtAudioFormat format, unsigned int *bufferSize,
f@0:                                    RtAudio::StreamOptions *options )
f@0: {
f@0:   // For ASIO, a duplex stream MUST use the same driver.
f@0:   if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
f@0:     errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   char driverName[32];
f@0:   ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
f@0:   if ( result != ASE_OK ) {
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Only load the driver once for duplex stream.
f@0:   if ( mode != INPUT || stream_.mode != OUTPUT ) {
f@0:     // The getDeviceInfo() function will not work when a stream is open
f@0:     // because ASIO does not allow multiple devices to run at the same
f@0:     // time.  Thus, we'll probe the system before opening a stream and
f@0:     // save the results for use by getDeviceInfo().
f@0:     this->saveDeviceInfo();
f@0: 
f@0:     if ( !drivers.loadDriver( driverName ) ) {
f@0:       errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     result = ASIOInit( &driverInfo );
f@0:     if ( result != ASE_OK ) {
f@0:       errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0: 
f@0:   // Check the device channel count.
f@0:   long inputChannels, outputChannels;
f@0:   result = ASIOGetChannels( &inputChannels, &outputChannels );
f@0:   if ( result != ASE_OK ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
f@0:        ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0:   stream_.nDeviceChannels[mode] = channels;
f@0:   stream_.nUserChannels[mode] = channels;
f@0:   stream_.channelOffset[mode] = firstChannel;
f@0: 
f@0:   // Verify the sample rate is supported.
f@0:   result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
f@0:   if ( result != ASE_OK ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Get the current sample rate
f@0:   ASIOSampleRate currentRate;
f@0:   result = ASIOGetSampleRate( &currentRate );
f@0:   if ( result != ASE_OK ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Set the sample rate only if necessary
f@0:   if ( currentRate != sampleRate ) {
f@0:     result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
f@0:     if ( result != ASE_OK ) {
f@0:       drivers.removeCurrentDriver();
f@0:       errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0: 
f@0:   // Determine the driver data type.
f@0:   ASIOChannelInfo channelInfo;
f@0:   channelInfo.channel = 0;
f@0:   if ( mode == OUTPUT ) channelInfo.isInput = false;
f@0:   else channelInfo.isInput = true;
f@0:   result = ASIOGetChannelInfo( &channelInfo );
f@0:   if ( result != ASE_OK ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Assuming WINDOWS host is always little-endian.
f@0:   stream_.doByteSwap[mode] = false;
f@0:   stream_.userFormat = format;
f@0:   stream_.deviceFormat[mode] = 0;
f@0:   if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:     if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
f@0:   }
f@0:   else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_SINT32;
f@0:     if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
f@0:   }
f@0:   else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
f@0:     if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
f@0:   }
f@0:   else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
f@0:     if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
f@0:   }
f@0:   else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_SINT24;
f@0:     if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
f@0:   }
f@0: 
f@0:   if ( stream_.deviceFormat[mode] == 0 ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Set the buffer size.  For a duplex stream, this will end up
f@0:   // setting the buffer size based on the input constraints, which
f@0:   // should be ok.
f@0:   long minSize, maxSize, preferSize, granularity;
f@0:   result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
f@0:   if ( result != ASE_OK ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
f@0:   else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
f@0:   else if ( granularity == -1 ) {
f@0:     // Make sure bufferSize is a power of two.
f@0:     int log2_of_min_size = 0;
f@0:     int log2_of_max_size = 0;
f@0: 
f@0:     for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
f@0:       if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
f@0:       if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
f@0:     }
f@0: 
f@0:     long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
f@0:     int min_delta_num = log2_of_min_size;
f@0: 
f@0:     for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
f@0:       long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
f@0:       if (current_delta < min_delta) {
f@0:         min_delta = current_delta;
f@0:         min_delta_num = i;
f@0:       }
f@0:     }
f@0: 
f@0:     *bufferSize = ( (unsigned int)1 << min_delta_num );
f@0:     if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
f@0:     else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
f@0:   }
f@0:   else if ( granularity != 0 ) {
f@0:     // Set to an even multiple of granularity, rounding up.
f@0:     *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
f@0:   }
f@0: 
f@0:   if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
f@0:     drivers.removeCurrentDriver();
f@0:     errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   stream_.bufferSize = *bufferSize;
f@0:   stream_.nBuffers = 2;
f@0: 
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
f@0:   else stream_.userInterleaved = true;
f@0: 
f@0:   // ASIO always uses non-interleaved buffers.
f@0:   stream_.deviceInterleaved[mode] = false;
f@0: 
f@0:   // Allocate, if necessary, our AsioHandle structure for the stream.
f@0:   AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
f@0:   if ( handle == 0 ) {
f@0:     try {
f@0:       handle = new AsioHandle;
f@0:     }
f@0:     catch ( std::bad_alloc& ) {
f@0:       //if ( handle == NULL ) {    
f@0:       drivers.removeCurrentDriver();
f@0:       errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
f@0:       return FAILURE;
f@0:     }
f@0:     handle->bufferInfos = 0;
f@0: 
f@0:     // Create a manual-reset event.
f@0:     handle->condition = CreateEvent( NULL,   // no security
f@0:                                      TRUE,   // manual-reset
f@0:                                      FALSE,  // non-signaled initially
f@0:                                      NULL ); // unnamed
f@0:     stream_.apiHandle = (void *) handle;
f@0:   }
f@0: 
f@0:   // Create the ASIO internal buffers.  Since RtAudio sets up input
f@0:   // and output separately, we'll have to dispose of previously
f@0:   // created output buffers for a duplex stream.
f@0:   long inputLatency, outputLatency;
f@0:   if ( mode == INPUT && stream_.mode == OUTPUT ) {
f@0:     ASIODisposeBuffers();
f@0:     if ( handle->bufferInfos ) free( handle->bufferInfos );
f@0:   }
f@0: 
f@0:   // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
f@0:   bool buffersAllocated = false;
f@0:   unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
f@0:   handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
f@0:   if ( handle->bufferInfos == NULL ) {
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     goto error;
f@0:   }
f@0: 
f@0:   ASIOBufferInfo *infos;
f@0:   infos = handle->bufferInfos;
f@0:   for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
f@0:     infos->isInput = ASIOFalse;
f@0:     infos->channelNum = i + stream_.channelOffset[0];
f@0:     infos->buffers[0] = infos->buffers[1] = 0;
f@0:   }
f@0:   for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
f@0:     infos->isInput = ASIOTrue;
f@0:     infos->channelNum = i + stream_.channelOffset[1];
f@0:     infos->buffers[0] = infos->buffers[1] = 0;
f@0:   }
f@0: 
f@0:   // Set up the ASIO callback structure and create the ASIO data buffers.
f@0:   asioCallbacks.bufferSwitch = &bufferSwitch;
f@0:   asioCallbacks.sampleRateDidChange = &sampleRateChanged;
f@0:   asioCallbacks.asioMessage = &asioMessages;
f@0:   asioCallbacks.bufferSwitchTimeInfo = NULL;
f@0:   result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
f@0:   if ( result != ASE_OK ) {
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
f@0:     errorText_ = errorStream_.str();
f@0:     goto error;
f@0:   }
f@0:   buffersAllocated = true;
f@0: 
f@0:   // Set flags for buffer conversion.
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if ( stream_.userFormat != stream_.deviceFormat[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
f@0:        stream_.nUserChannels[mode] > 1 )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   // Allocate necessary internal buffers
f@0:   unsigned long bufferBytes;
f@0:   bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
f@0:   stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
f@0:   if ( stream_.userBuffer[mode] == NULL ) {
f@0:     errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
f@0:     goto error;
f@0:   }
f@0: 
f@0:   if ( stream_.doConvertBuffer[mode] ) {
f@0: 
f@0:     bool makeBuffer = true;
f@0:     bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
f@0:     if ( mode == INPUT ) {
f@0:       if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
f@0:         unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
f@0:         if ( bufferBytes <= bytesOut ) makeBuffer = false;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( makeBuffer ) {
f@0:       bufferBytes *= *bufferSize;
f@0:       if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
f@0:       stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
f@0:       if ( stream_.deviceBuffer == NULL ) {
f@0:         errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
f@0:         goto error;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.sampleRate = sampleRate;
f@0:   stream_.device[mode] = device;
f@0:   stream_.state = STREAM_STOPPED;
f@0:   asioCallbackInfo = &stream_.callbackInfo;
f@0:   stream_.callbackInfo.object = (void *) this;
f@0:   if ( stream_.mode == OUTPUT && mode == INPUT )
f@0:     // We had already set up an output stream.
f@0:     stream_.mode = DUPLEX;
f@0:   else
f@0:     stream_.mode = mode;
f@0: 
f@0:   // Determine device latencies
f@0:   result = ASIOGetLatencies( &inputLatency, &outputLatency );
f@0:   if ( result != ASE_OK ) {
f@0:     errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING); // warn but don't fail
f@0:   }
f@0:   else {
f@0:     stream_.latency[0] = outputLatency;
f@0:     stream_.latency[1] = inputLatency;
f@0:   }
f@0: 
f@0:   // Setup the buffer conversion information structure.  We don't use
f@0:   // buffers to do channel offsets, so we override that parameter
f@0:   // here.
f@0:   if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
f@0: 
f@0:   return SUCCESS;
f@0: 
f@0:  error:
f@0:   if ( buffersAllocated )
f@0:     ASIODisposeBuffers();
f@0:   drivers.removeCurrentDriver();
f@0: 
f@0:   if ( handle ) {
f@0:     CloseHandle( handle->condition );
f@0:     if ( handle->bufferInfos )
f@0:       free( handle->bufferInfos );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: void RtApiAsio :: closeStream()
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     stream_.state = STREAM_STOPPED;
f@0:     ASIOStop();
f@0:   }
f@0:   ASIODisposeBuffers();
f@0:   drivers.removeCurrentDriver();
f@0: 
f@0:   AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
f@0:   if ( handle ) {
f@0:     CloseHandle( handle->condition );
f@0:     if ( handle->bufferInfos )
f@0:       free( handle->bufferInfos );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.state = STREAM_CLOSED;
f@0: }
f@0: 
f@0: bool stopThreadCalled = false;
f@0: 
f@0: void RtApiAsio :: startStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiAsio::startStream(): the stream is already running!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
f@0:   ASIOError result = ASIOStart();
f@0:   if ( result != ASE_OK ) {
f@0:     errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
f@0:     errorText_ = errorStream_.str();
f@0:     goto unlock;
f@0:   }
f@0: 
f@0:   handle->drainCounter = 0;
f@0:   handle->internalDrain = false;
f@0:   ResetEvent( handle->condition );
f@0:   stream_.state = STREAM_RUNNING;
f@0:   asioXRun = false;
f@0: 
f@0:  unlock:
f@0:   stopThreadCalled = false;
f@0: 
f@0:   if ( result == ASE_OK ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiAsio :: stopStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     if ( handle->drainCounter == 0 ) {
f@0:       handle->drainCounter = 2;
f@0:       WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0: 
f@0:   ASIOError result = ASIOStop();
f@0:   if ( result != ASE_OK ) {
f@0:     errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
f@0:     errorText_ = errorStream_.str();
f@0:   }
f@0: 
f@0:   if ( result == ASE_OK ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiAsio :: abortStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   // The following lines were commented-out because some behavior was
f@0:   // noted where the device buffers need to be zeroed to avoid
f@0:   // continuing sound, even when the device buffers are completely
f@0:   // disposed.  So now, calling abort is the same as calling stop.
f@0:   // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
f@0:   // handle->drainCounter = 2;
f@0:   stopStream();
f@0: }
f@0: 
f@0: // This function will be called by a spawned thread when the user
f@0: // callback function signals that the stream should be stopped or
f@0: // aborted.  It is necessary to handle it this way because the
f@0: // callbackEvent() function must return before the ASIOStop()
f@0: // function will return.
f@0: static unsigned __stdcall asioStopStream( void *ptr )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) ptr;
f@0:   RtApiAsio *object = (RtApiAsio *) info->object;
f@0: 
f@0:   object->stopStream();
f@0:   _endthreadex( 0 );
f@0:   return 0;
f@0: }
f@0: 
f@0: bool RtApiAsio :: callbackEvent( long bufferIndex )
f@0: {
f@0:   if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
f@0:     error( RtAudioError::WARNING );
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
f@0:   AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
f@0: 
f@0:   // Check if we were draining the stream and signal if finished.
f@0:   if ( handle->drainCounter > 3 ) {
f@0: 
f@0:     stream_.state = STREAM_STOPPING;
f@0:     if ( handle->internalDrain == false )
f@0:       SetEvent( handle->condition );
f@0:     else { // spawn a thread to stop the stream
f@0:       unsigned threadId;
f@0:       stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
f@0:                                                     &stream_.callbackInfo, 0, &threadId );
f@0:     }
f@0:     return SUCCESS;
f@0:   }
f@0: 
f@0:   // Invoke user callback to get fresh output data UNLESS we are
f@0:   // draining stream.
f@0:   if ( handle->drainCounter == 0 ) {
f@0:     RtAudioCallback callback = (RtAudioCallback) info->callback;
f@0:     double streamTime = getStreamTime();
f@0:     RtAudioStreamStatus status = 0;
f@0:     if ( stream_.mode != INPUT && asioXRun == true ) {
f@0:       status |= RTAUDIO_OUTPUT_UNDERFLOW;
f@0:       asioXRun = false;
f@0:     }
f@0:     if ( stream_.mode != OUTPUT && asioXRun == true ) {
f@0:       status |= RTAUDIO_INPUT_OVERFLOW;
f@0:       asioXRun = false;
f@0:     }
f@0:     int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
f@0:                                      stream_.bufferSize, streamTime, status, info->userData );
f@0:     if ( cbReturnValue == 2 ) {
f@0:       stream_.state = STREAM_STOPPING;
f@0:       handle->drainCounter = 2;
f@0:       unsigned threadId;
f@0:       stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
f@0:                                                     &stream_.callbackInfo, 0, &threadId );
f@0:       return SUCCESS;
f@0:     }
f@0:     else if ( cbReturnValue == 1 ) {
f@0:       handle->drainCounter = 1;
f@0:       handle->internalDrain = true;
f@0:     }
f@0:   }
f@0: 
f@0:   unsigned int nChannels, bufferBytes, i, j;
f@0:   nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
f@0: 
f@0:     if ( handle->drainCounter > 1 ) { // write zeros to the output stream
f@0: 
f@0:       for ( i=0, j=0; i<nChannels; i++ ) {
f@0:         if ( handle->bufferInfos[i].isInput != ASIOTrue )
f@0:           memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
f@0:       }
f@0: 
f@0:     }
f@0:     else if ( stream_.doConvertBuffer[0] ) {
f@0: 
f@0:       convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
f@0:       if ( stream_.doByteSwap[0] )
f@0:         byteSwapBuffer( stream_.deviceBuffer,
f@0:                         stream_.bufferSize * stream_.nDeviceChannels[0],
f@0:                         stream_.deviceFormat[0] );
f@0: 
f@0:       for ( i=0, j=0; i<nChannels; i++ ) {
f@0:         if ( handle->bufferInfos[i].isInput != ASIOTrue )
f@0:           memcpy( handle->bufferInfos[i].buffers[bufferIndex],
f@0:                   &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
f@0:       }
f@0: 
f@0:     }
f@0:     else {
f@0: 
f@0:       if ( stream_.doByteSwap[0] )
f@0:         byteSwapBuffer( stream_.userBuffer[0],
f@0:                         stream_.bufferSize * stream_.nUserChannels[0],
f@0:                         stream_.userFormat );
f@0: 
f@0:       for ( i=0, j=0; i<nChannels; i++ ) {
f@0:         if ( handle->bufferInfos[i].isInput != ASIOTrue )
f@0:           memcpy( handle->bufferInfos[i].buffers[bufferIndex],
f@0:                   &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
f@0:       }
f@0: 
f@0:     }
f@0:   }
f@0: 
f@0:   // Don't bother draining input
f@0:   if ( handle->drainCounter ) {
f@0:     handle->drainCounter++;
f@0:     goto unlock;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
f@0: 
f@0:     if (stream_.doConvertBuffer[1]) {
f@0: 
f@0:       // Always interleave ASIO input data.
f@0:       for ( i=0, j=0; i<nChannels; i++ ) {
f@0:         if ( handle->bufferInfos[i].isInput == ASIOTrue )
f@0:           memcpy( &stream_.deviceBuffer[j++*bufferBytes],
f@0:                   handle->bufferInfos[i].buffers[bufferIndex],
f@0:                   bufferBytes );
f@0:       }
f@0: 
f@0:       if ( stream_.doByteSwap[1] )
f@0:         byteSwapBuffer( stream_.deviceBuffer,
f@0:                         stream_.bufferSize * stream_.nDeviceChannels[1],
f@0:                         stream_.deviceFormat[1] );
f@0:       convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
f@0: 
f@0:     }
f@0:     else {
f@0:       for ( i=0, j=0; i<nChannels; i++ ) {
f@0:         if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
f@0:           memcpy( &stream_.userBuffer[1][bufferBytes*j++],
f@0:                   handle->bufferInfos[i].buffers[bufferIndex],
f@0:                   bufferBytes );
f@0:         }
f@0:       }
f@0: 
f@0:       if ( stream_.doByteSwap[1] )
f@0:         byteSwapBuffer( stream_.userBuffer[1],
f@0:                         stream_.bufferSize * stream_.nUserChannels[1],
f@0:                         stream_.userFormat );
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   // The following call was suggested by Malte Clasen.  While the API
f@0:   // documentation indicates it should not be required, some device
f@0:   // drivers apparently do not function correctly without it.
f@0:   ASIOOutputReady();
f@0: 
f@0:   RtApi::tickStreamTime();
f@0:   return SUCCESS;
f@0: }
f@0: 
f@0: static void sampleRateChanged( ASIOSampleRate sRate )
f@0: {
f@0:   // The ASIO documentation says that this usually only happens during
f@0:   // external sync.  Audio processing is not stopped by the driver,
f@0:   // actual sample rate might not have even changed, maybe only the
f@0:   // sample rate status of an AES/EBU or S/PDIF digital input at the
f@0:   // audio device.
f@0: 
f@0:   RtApi *object = (RtApi *) asioCallbackInfo->object;
f@0:   try {
f@0:     object->stopStream();
f@0:   }
f@0:   catch ( RtAudioError &exception ) {
f@0:     std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
f@0:     return;
f@0:   }
f@0: 
f@0:   std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
f@0: }
f@0: 
f@0: static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
f@0: {
f@0:   long ret = 0;
f@0: 
f@0:   switch( selector ) {
f@0:   case kAsioSelectorSupported:
f@0:     if ( value == kAsioResetRequest
f@0:          || value == kAsioEngineVersion
f@0:          || value == kAsioResyncRequest
f@0:          || value == kAsioLatenciesChanged
f@0:          // The following three were added for ASIO 2.0, you don't
f@0:          // necessarily have to support them.
f@0:          || value == kAsioSupportsTimeInfo
f@0:          || value == kAsioSupportsTimeCode
f@0:          || value == kAsioSupportsInputMonitor)
f@0:       ret = 1L;
f@0:     break;
f@0:   case kAsioResetRequest:
f@0:     // Defer the task and perform the reset of the driver during the
f@0:     // next "safe" situation.  You cannot reset the driver right now,
f@0:     // as this code is called from the driver.  Reset the driver is
f@0:     // done by completely destruct is. I.e. ASIOStop(),
f@0:     // ASIODisposeBuffers(), Destruction Afterwards you initialize the
f@0:     // driver again.
f@0:     std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
f@0:     ret = 1L;
f@0:     break;
f@0:   case kAsioResyncRequest:
f@0:     // This informs the application that the driver encountered some
f@0:     // non-fatal data loss.  It is used for synchronization purposes
f@0:     // of different media.  Added mainly to work around the Win16Mutex
f@0:     // problems in Windows 95/98 with the Windows Multimedia system,
f@0:     // which could lose data because the Mutex was held too long by
f@0:     // another thread.  However a driver can issue it in other
f@0:     // situations, too.
f@0:     // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
f@0:     asioXRun = true;
f@0:     ret = 1L;
f@0:     break;
f@0:   case kAsioLatenciesChanged:
f@0:     // This will inform the host application that the drivers were
f@0:     // latencies changed.  Beware, it this does not mean that the
f@0:     // buffer sizes have changed!  You might need to update internal
f@0:     // delay data.
f@0:     std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
f@0:     ret = 1L;
f@0:     break;
f@0:   case kAsioEngineVersion:
f@0:     // Return the supported ASIO version of the host application.  If
f@0:     // a host application does not implement this selector, ASIO 1.0
f@0:     // is assumed by the driver.
f@0:     ret = 2L;
f@0:     break;
f@0:   case kAsioSupportsTimeInfo:
f@0:     // Informs the driver whether the
f@0:     // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
f@0:     // For compatibility with ASIO 1.0 drivers the host application
f@0:     // should always support the "old" bufferSwitch method, too.
f@0:     ret = 0;
f@0:     break;
f@0:   case kAsioSupportsTimeCode:
f@0:     // Informs the driver whether application is interested in time
f@0:     // code info.  If an application does not need to know about time
f@0:     // code, the driver has less work to do.
f@0:     ret = 0;
f@0:     break;
f@0:   }
f@0:   return ret;
f@0: }
f@0: 
f@0: static const char* getAsioErrorString( ASIOError result )
f@0: {
f@0:   struct Messages 
f@0:   {
f@0:     ASIOError value;
f@0:     const char*message;
f@0:   };
f@0: 
f@0:   static const Messages m[] = 
f@0:     {
f@0:       {   ASE_NotPresent,    "Hardware input or output is not present or available." },
f@0:       {   ASE_HWMalfunction,  "Hardware is malfunctioning." },
f@0:       {   ASE_InvalidParameter, "Invalid input parameter." },
f@0:       {   ASE_InvalidMode,      "Invalid mode." },
f@0:       {   ASE_SPNotAdvancing,     "Sample position not advancing." },
f@0:       {   ASE_NoClock,            "Sample clock or rate cannot be determined or is not present." },
f@0:       {   ASE_NoMemory,           "Not enough memory to complete the request." }
f@0:     };
f@0: 
f@0:   for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
f@0:     if ( m[i].value == result ) return m[i].message;
f@0: 
f@0:   return "Unknown error.";
f@0: }
f@0: 
f@0: //******************** End of __WINDOWS_ASIO__ *********************//
f@0: #endif
f@0: 
f@0: 
f@0: #if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
f@0: 
f@0: // Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
f@0: // - Introduces support for the Windows WASAPI API
f@0: // - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
f@0: // - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
f@0: // - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
f@0: 
f@0: #ifndef INITGUID
f@0:   #define INITGUID
f@0: #endif
f@0: #include <audioclient.h>
f@0: #include <avrt.h>
f@0: #include <mmdeviceapi.h>
f@0: #include <functiondiscoverykeys_devpkey.h>
f@0: 
f@0: //=============================================================================
f@0: 
f@0: #define SAFE_RELEASE( objectPtr )\
f@0: if ( objectPtr )\
f@0: {\
f@0:   objectPtr->Release();\
f@0:   objectPtr = NULL;\
f@0: }
f@0: 
f@0: typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: // WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
f@0: // Therefore we must perform all necessary conversions to user buffers in order to satisfy these
f@0: // requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
f@0: // provide intermediate storage for read / write synchronization.
f@0: class WasapiBuffer
f@0: {
f@0: public:
f@0:   WasapiBuffer()
f@0:     : buffer_( NULL ),
f@0:       bufferSize_( 0 ),
f@0:       inIndex_( 0 ),
f@0:       outIndex_( 0 ) {}
f@0: 
f@0:   ~WasapiBuffer() {
f@0:     delete buffer_;
f@0:   }
f@0: 
f@0:   // sets the length of the internal ring buffer
f@0:   void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
f@0:     delete buffer_;
f@0: 
f@0:     buffer_ = ( char* ) calloc( bufferSize, formatBytes );
f@0: 
f@0:     bufferSize_ = bufferSize;
f@0:     inIndex_ = 0;
f@0:     outIndex_ = 0;
f@0:   }
f@0: 
f@0:   // attempt to push a buffer into the ring buffer at the current "in" index
f@0:   bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
f@0:   {
f@0:     if ( !buffer ||                 // incoming buffer is NULL
f@0:          bufferSize == 0 ||         // incoming buffer has no data
f@0:          bufferSize > bufferSize_ ) // incoming buffer too large
f@0:     {
f@0:       return false;
f@0:     }
f@0: 
f@0:     unsigned int relOutIndex = outIndex_;
f@0:     unsigned int inIndexEnd = inIndex_ + bufferSize;
f@0:     if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
f@0:       relOutIndex += bufferSize_;
f@0:     }
f@0: 
f@0:     // "in" index can end on the "out" index but cannot begin at it
f@0:     if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
f@0:       return false; // not enough space between "in" index and "out" index
f@0:     }
f@0: 
f@0:     // copy buffer from external to internal
f@0:     int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
f@0:     fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
f@0:     int fromInSize = bufferSize - fromZeroSize;
f@0: 
f@0:     switch( format )
f@0:       {
f@0:       case RTAUDIO_SINT8:
f@0:         memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
f@0:         memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
f@0:         break;
f@0:       case RTAUDIO_SINT16:
f@0:         memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
f@0:         memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
f@0:         break;
f@0:       case RTAUDIO_SINT24:
f@0:         memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
f@0:         memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
f@0:         break;
f@0:       case RTAUDIO_SINT32:
f@0:         memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
f@0:         memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
f@0:         break;
f@0:       case RTAUDIO_FLOAT32:
f@0:         memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
f@0:         memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
f@0:         break;
f@0:       case RTAUDIO_FLOAT64:
f@0:         memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
f@0:         memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
f@0:         break;
f@0:     }
f@0: 
f@0:     // update "in" index
f@0:     inIndex_ += bufferSize;
f@0:     inIndex_ %= bufferSize_;
f@0: 
f@0:     return true;
f@0:   }
f@0: 
f@0:   // attempt to pull a buffer from the ring buffer from the current "out" index
f@0:   bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
f@0:   {
f@0:     if ( !buffer ||                 // incoming buffer is NULL
f@0:          bufferSize == 0 ||         // incoming buffer has no data
f@0:          bufferSize > bufferSize_ ) // incoming buffer too large
f@0:     {
f@0:       return false;
f@0:     }
f@0: 
f@0:     unsigned int relInIndex = inIndex_;
f@0:     unsigned int outIndexEnd = outIndex_ + bufferSize;
f@0:     if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
f@0:       relInIndex += bufferSize_;
f@0:     }
f@0: 
f@0:     // "out" index can begin at and end on the "in" index
f@0:     if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
f@0:       return false; // not enough space between "out" index and "in" index
f@0:     }
f@0: 
f@0:     // copy buffer from internal to external
f@0:     int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
f@0:     fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
f@0:     int fromOutSize = bufferSize - fromZeroSize;
f@0: 
f@0:     switch( format )
f@0:     {
f@0:       case RTAUDIO_SINT8:
f@0:         memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
f@0:         memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
f@0:         break;
f@0:       case RTAUDIO_SINT16:
f@0:         memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
f@0:         memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
f@0:         break;
f@0:       case RTAUDIO_SINT24:
f@0:         memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
f@0:         memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
f@0:         break;
f@0:       case RTAUDIO_SINT32:
f@0:         memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
f@0:         memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
f@0:         break;
f@0:       case RTAUDIO_FLOAT32:
f@0:         memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
f@0:         memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
f@0:         break;
f@0:       case RTAUDIO_FLOAT64:
f@0:         memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
f@0:         memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
f@0:         break;
f@0:     }
f@0: 
f@0:     // update "out" index
f@0:     outIndex_ += bufferSize;
f@0:     outIndex_ %= bufferSize_;
f@0: 
f@0:     return true;
f@0:   }
f@0: 
f@0: private:
f@0:   char* buffer_;
f@0:   unsigned int bufferSize_;
f@0:   unsigned int inIndex_;
f@0:   unsigned int outIndex_;
f@0: };
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
f@0: // between HW and the user. The convertBufferWasapi function is used to perform this conversion
f@0: // between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
f@0: // This sample rate converter favors speed over quality, and works best with conversions between
f@0: // one rate and its multiple.
f@0: void convertBufferWasapi( char* outBuffer,
f@0:                           const char* inBuffer,
f@0:                           const unsigned int& channelCount,
f@0:                           const unsigned int& inSampleRate,
f@0:                           const unsigned int& outSampleRate,
f@0:                           const unsigned int& inSampleCount,
f@0:                           unsigned int& outSampleCount,
f@0:                           const RtAudioFormat& format )
f@0: {
f@0:   // calculate the new outSampleCount and relative sampleStep
f@0:   float sampleRatio = ( float ) outSampleRate / inSampleRate;
f@0:   float sampleStep = 1.0f / sampleRatio;
f@0:   float inSampleFraction = 0.0f;
f@0: 
f@0:   outSampleCount = ( unsigned int ) ( inSampleCount * sampleRatio );
f@0: 
f@0:   // frame-by-frame, copy each relative input sample into it's corresponding output sample
f@0:   for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
f@0:   {
f@0:     unsigned int inSample = ( unsigned int ) inSampleFraction;
f@0: 
f@0:     switch ( format )
f@0:     {
f@0:       case RTAUDIO_SINT8:
f@0:         memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
f@0:         break;
f@0:       case RTAUDIO_SINT16:
f@0:         memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
f@0:         break;
f@0:       case RTAUDIO_SINT24:
f@0:         memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
f@0:         break;
f@0:       case RTAUDIO_SINT32:
f@0:         memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
f@0:         break;
f@0:       case RTAUDIO_FLOAT32:
f@0:         memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
f@0:         break;
f@0:       case RTAUDIO_FLOAT64:
f@0:         memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
f@0:         break;
f@0:     }
f@0: 
f@0:     // jump to next in sample
f@0:     inSampleFraction += sampleStep;
f@0:   }
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: // A structure to hold various information related to the WASAPI implementation.
f@0: struct WasapiHandle
f@0: {
f@0:   IAudioClient* captureAudioClient;
f@0:   IAudioClient* renderAudioClient;
f@0:   IAudioCaptureClient* captureClient;
f@0:   IAudioRenderClient* renderClient;
f@0:   HANDLE captureEvent;
f@0:   HANDLE renderEvent;
f@0: 
f@0:   WasapiHandle()
f@0:   : captureAudioClient( NULL ),
f@0:     renderAudioClient( NULL ),
f@0:     captureClient( NULL ),
f@0:     renderClient( NULL ),
f@0:     captureEvent( NULL ),
f@0:     renderEvent( NULL ) {}
f@0: };
f@0: 
f@0: //=============================================================================
f@0: 
f@0: RtApiWasapi::RtApiWasapi()
f@0:   : coInitialized_( false ), deviceEnumerator_( NULL )
f@0: {
f@0:   // WASAPI can run either apartment or multi-threaded
f@0:   HRESULT hr = CoInitialize( NULL );
f@0:   if ( !FAILED( hr ) )
f@0:     coInitialized_ = true;
f@0: 
f@0:   // Instantiate device enumerator
f@0:   hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
f@0:                          CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
f@0:                          ( void** ) &deviceEnumerator_ );
f@0: 
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
f@0:     error( RtAudioError::DRIVER_ERROR );
f@0:   }
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: RtApiWasapi::~RtApiWasapi()
f@0: {
f@0:   if ( stream_.state != STREAM_CLOSED )
f@0:     closeStream();
f@0: 
f@0:   SAFE_RELEASE( deviceEnumerator_ );
f@0: 
f@0:   // If this object previously called CoInitialize()
f@0:   if ( coInitialized_ )
f@0:     CoUninitialize();
f@0: }
f@0: 
f@0: //=============================================================================
f@0: 
f@0: unsigned int RtApiWasapi::getDeviceCount( void )
f@0: {
f@0:   unsigned int captureDeviceCount = 0;
f@0:   unsigned int renderDeviceCount = 0;
f@0: 
f@0:   IMMDeviceCollection* captureDevices = NULL;
f@0:   IMMDeviceCollection* renderDevices = NULL;
f@0: 
f@0:   // Count capture devices
f@0:   errorText_.clear();
f@0:   HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   hr = captureDevices->GetCount( &captureDeviceCount );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // Count render devices
f@0:   hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   hr = renderDevices->GetCount( &renderDeviceCount );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0: Exit:
f@0:   // release all references
f@0:   SAFE_RELEASE( captureDevices );
f@0:   SAFE_RELEASE( renderDevices );
f@0: 
f@0:   if ( errorText_.empty() )
f@0:     return captureDeviceCount + renderDeviceCount;
f@0: 
f@0:   error( RtAudioError::DRIVER_ERROR );
f@0:   return 0;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   unsigned int captureDeviceCount = 0;
f@0:   unsigned int renderDeviceCount = 0;
f@0:   std::wstring deviceName;
f@0:   std::string defaultDeviceName;
f@0:   bool isCaptureDevice = false;
f@0: 
f@0:   PROPVARIANT deviceNameProp;
f@0:   PROPVARIANT defaultDeviceNameProp;
f@0: 
f@0:   IMMDeviceCollection* captureDevices = NULL;
f@0:   IMMDeviceCollection* renderDevices = NULL;
f@0:   IMMDevice* devicePtr = NULL;
f@0:   IMMDevice* defaultDevicePtr = NULL;
f@0:   IAudioClient* audioClient = NULL;
f@0:   IPropertyStore* devicePropStore = NULL;
f@0:   IPropertyStore* defaultDevicePropStore = NULL;
f@0: 
f@0:   WAVEFORMATEX* deviceFormat = NULL;
f@0:   WAVEFORMATEX* closestMatchFormat = NULL;
f@0: 
f@0:   // probed
f@0:   info.probed = false;
f@0: 
f@0:   // Count capture devices
f@0:   errorText_.clear();
f@0:   RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
f@0:   HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   hr = captureDevices->GetCount( &captureDeviceCount );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // Count render devices
f@0:   hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   hr = renderDevices->GetCount( &renderDeviceCount );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // validate device index
f@0:   if ( device >= captureDeviceCount + renderDeviceCount ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
f@0:     errorType = RtAudioError::INVALID_USE;
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // determine whether index falls within capture or render devices
f@0:   if ( device >= renderDeviceCount ) {
f@0:     hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
f@0:       goto Exit;
f@0:     }
f@0:     isCaptureDevice = true;
f@0:   }
f@0:   else {
f@0:     hr = renderDevices->Item( device, &devicePtr );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
f@0:       goto Exit;
f@0:     }
f@0:     isCaptureDevice = false;
f@0:   }
f@0: 
f@0:   // get default device name
f@0:   if ( isCaptureDevice ) {
f@0:     hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
f@0:       goto Exit;
f@0:     }
f@0:   }
f@0:   else {
f@0:     hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
f@0:       goto Exit;
f@0:     }
f@0:   }
f@0: 
f@0:   hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
f@0:     goto Exit;
f@0:   }
f@0:   PropVariantInit( &defaultDeviceNameProp );
f@0: 
f@0:   hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   deviceName = defaultDeviceNameProp.pwszVal;
f@0:   defaultDeviceName = std::string( deviceName.begin(), deviceName.end() );
f@0: 
f@0:   // name
f@0:   hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   PropVariantInit( &deviceNameProp );
f@0: 
f@0:   hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   deviceName = deviceNameProp.pwszVal;
f@0:   info.name = std::string( deviceName.begin(), deviceName.end() );
f@0: 
f@0:   // is default
f@0:   if ( isCaptureDevice ) {
f@0:     info.isDefaultInput = info.name == defaultDeviceName;
f@0:     info.isDefaultOutput = false;
f@0:   }
f@0:   else {
f@0:     info.isDefaultInput = false;
f@0:     info.isDefaultOutput = info.name == defaultDeviceName;
f@0:   }
f@0: 
f@0:   // channel count
f@0:   hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   hr = audioClient->GetMixFormat( &deviceFormat );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   if ( isCaptureDevice ) {
f@0:     info.inputChannels = deviceFormat->nChannels;
f@0:     info.outputChannels = 0;
f@0:     info.duplexChannels = 0;
f@0:   }
f@0:   else {
f@0:     info.inputChannels = 0;
f@0:     info.outputChannels = deviceFormat->nChannels;
f@0:     info.duplexChannels = 0;
f@0:   }
f@0: 
f@0:   // sample rates
f@0:   info.sampleRates.clear();
f@0: 
f@0:   // allow support for all sample rates as we have a built-in sample rate converter
f@0:   for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
f@0:     info.sampleRates.push_back( SAMPLE_RATES[i] );
f@0:   }
f@0: 
f@0:   // native format
f@0:   info.nativeFormats = 0;
f@0: 
f@0:   if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
f@0:        ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
f@0:          ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
f@0:   {
f@0:     if ( deviceFormat->wBitsPerSample == 32 ) {
f@0:       info.nativeFormats |= RTAUDIO_FLOAT32;
f@0:     }
f@0:     else if ( deviceFormat->wBitsPerSample == 64 ) {
f@0:       info.nativeFormats |= RTAUDIO_FLOAT64;
f@0:     }
f@0:   }
f@0:   else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
f@0:            ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
f@0:              ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
f@0:   {
f@0:     if ( deviceFormat->wBitsPerSample == 8 ) {
f@0:       info.nativeFormats |= RTAUDIO_SINT8;
f@0:     }
f@0:     else if ( deviceFormat->wBitsPerSample == 16 ) {
f@0:       info.nativeFormats |= RTAUDIO_SINT16;
f@0:     }
f@0:     else if ( deviceFormat->wBitsPerSample == 24 ) {
f@0:       info.nativeFormats |= RTAUDIO_SINT24;
f@0:     }
f@0:     else if ( deviceFormat->wBitsPerSample == 32 ) {
f@0:       info.nativeFormats |= RTAUDIO_SINT32;
f@0:     }
f@0:   }
f@0: 
f@0:   // probed
f@0:   info.probed = true;
f@0: 
f@0: Exit:
f@0:   // release all references
f@0:   PropVariantClear( &deviceNameProp );
f@0:   PropVariantClear( &defaultDeviceNameProp );
f@0: 
f@0:   SAFE_RELEASE( captureDevices );
f@0:   SAFE_RELEASE( renderDevices );
f@0:   SAFE_RELEASE( devicePtr );
f@0:   SAFE_RELEASE( defaultDevicePtr );
f@0:   SAFE_RELEASE( audioClient );
f@0:   SAFE_RELEASE( devicePropStore );
f@0:   SAFE_RELEASE( defaultDevicePropStore );
f@0: 
f@0:   CoTaskMemFree( deviceFormat );
f@0:   CoTaskMemFree( closestMatchFormat );
f@0: 
f@0:   if ( !errorText_.empty() )
f@0:     error( errorType );
f@0:   return info;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: unsigned int RtApiWasapi::getDefaultOutputDevice( void )
f@0: {
f@0:   for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
f@0:     if ( getDeviceInfo( i ).isDefaultOutput ) {
f@0:       return i;
f@0:     }
f@0:   }
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: unsigned int RtApiWasapi::getDefaultInputDevice( void )
f@0: {
f@0:   for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
f@0:     if ( getDeviceInfo( i ).isDefaultInput ) {
f@0:       return i;
f@0:     }
f@0:   }
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: void RtApiWasapi::closeStream( void )
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( stream_.state != STREAM_STOPPED )
f@0:     stopStream();
f@0: 
f@0:   // clean up stream memory
f@0:   SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
f@0:   SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
f@0: 
f@0:   SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
f@0:   SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
f@0: 
f@0:   if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
f@0:     CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
f@0: 
f@0:   if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
f@0:     CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
f@0: 
f@0:   delete ( WasapiHandle* ) stream_.apiHandle;
f@0:   stream_.apiHandle = NULL;
f@0: 
f@0:   for ( int i = 0; i < 2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   // update stream state
f@0:   stream_.state = STREAM_CLOSED;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: void RtApiWasapi::startStream( void )
f@0: {
f@0:   verifyStream();
f@0: 
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiWasapi::startStream: The stream is already running.";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   // update stream state
f@0:   stream_.state = STREAM_RUNNING;
f@0: 
f@0:   // create WASAPI stream thread
f@0:   stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
f@0: 
f@0:   if ( !stream_.callbackInfo.thread ) {
f@0:     errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
f@0:     error( RtAudioError::THREAD_ERROR );
f@0:   }
f@0:   else {
f@0:     SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
f@0:     ResumeThread( ( void* ) stream_.callbackInfo.thread );
f@0:   }
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: void RtApiWasapi::stopStream( void )
f@0: {
f@0:   verifyStream();
f@0: 
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   // inform stream thread by setting stream state to STREAM_STOPPING
f@0:   stream_.state = STREAM_STOPPING;
f@0: 
f@0:   // wait until stream thread is stopped
f@0:   while( stream_.state != STREAM_STOPPED ) {
f@0:     Sleep( 1 );
f@0:   }
f@0: 
f@0:   // Wait for the last buffer to play before stopping.
f@0:   Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
f@0: 
f@0:   // stop capture client if applicable
f@0:   if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
f@0:     HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
f@0:       error( RtAudioError::DRIVER_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   // stop render client if applicable
f@0:   if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
f@0:     HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
f@0:       error( RtAudioError::DRIVER_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   // close thread handle
f@0:   if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
f@0:     errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
f@0:     error( RtAudioError::THREAD_ERROR );
f@0:     return;
f@0:   }
f@0: 
f@0:   stream_.callbackInfo.thread = (ThreadHandle) NULL;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: void RtApiWasapi::abortStream( void )
f@0: {
f@0:   verifyStream();
f@0: 
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   // inform stream thread by setting stream state to STREAM_STOPPING
f@0:   stream_.state = STREAM_STOPPING;
f@0: 
f@0:   // wait until stream thread is stopped
f@0:   while ( stream_.state != STREAM_STOPPED ) {
f@0:     Sleep( 1 );
f@0:   }
f@0: 
f@0:   // stop capture client if applicable
f@0:   if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
f@0:     HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
f@0:       error( RtAudioError::DRIVER_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   // stop render client if applicable
f@0:   if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
f@0:     HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
f@0:       error( RtAudioError::DRIVER_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   // close thread handle
f@0:   if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
f@0:     errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
f@0:     error( RtAudioError::THREAD_ERROR );
f@0:     return;
f@0:   }
f@0: 
f@0:   stream_.callbackInfo.thread = (ThreadHandle) NULL;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
f@0:                                    unsigned int firstChannel, unsigned int sampleRate,
f@0:                                    RtAudioFormat format, unsigned int* bufferSize,
f@0:                                    RtAudio::StreamOptions* options )
f@0: {
f@0:   bool methodResult = FAILURE;
f@0:   unsigned int captureDeviceCount = 0;
f@0:   unsigned int renderDeviceCount = 0;
f@0: 
f@0:   IMMDeviceCollection* captureDevices = NULL;
f@0:   IMMDeviceCollection* renderDevices = NULL;
f@0:   IMMDevice* devicePtr = NULL;
f@0:   WAVEFORMATEX* deviceFormat = NULL;
f@0:   unsigned int bufferBytes;
f@0:   stream_.state = STREAM_STOPPED;
f@0: 
f@0:   // create API Handle if not already created
f@0:   if ( !stream_.apiHandle )
f@0:     stream_.apiHandle = ( void* ) new WasapiHandle();
f@0: 
f@0:   // Count capture devices
f@0:   errorText_.clear();
f@0:   RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
f@0:   HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   hr = captureDevices->GetCount( &captureDeviceCount );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // Count render devices
f@0:   hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   hr = renderDevices->GetCount( &renderDeviceCount );
f@0:   if ( FAILED( hr ) ) {
f@0:     errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // validate device index
f@0:   if ( device >= captureDeviceCount + renderDeviceCount ) {
f@0:     errorType = RtAudioError::INVALID_USE;
f@0:     errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // determine whether index falls within capture or render devices
f@0:   if ( device >= renderDeviceCount ) {
f@0:     if ( mode != INPUT ) {
f@0:       errorType = RtAudioError::INVALID_USE;
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     // retrieve captureAudioClient from devicePtr
f@0:     IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
f@0: 
f@0:     hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
f@0:                               NULL, ( void** ) &captureAudioClient );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     hr = captureAudioClient->GetMixFormat( &deviceFormat );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
f@0:     captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
f@0:   }
f@0:   else {
f@0:     if ( mode != OUTPUT ) {
f@0:       errorType = RtAudioError::INVALID_USE;
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     // retrieve renderAudioClient from devicePtr
f@0:     IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
f@0: 
f@0:     hr = renderDevices->Item( device, &devicePtr );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
f@0:                               NULL, ( void** ) &renderAudioClient );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     hr = renderAudioClient->GetMixFormat( &deviceFormat );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
f@0:     renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
f@0:   }
f@0: 
f@0:   // fill stream data
f@0:   if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
f@0:        ( stream_.mode == INPUT && mode == OUTPUT ) ) {
f@0:     stream_.mode = DUPLEX;
f@0:   }
f@0:   else {
f@0:     stream_.mode = mode;
f@0:   }
f@0: 
f@0:   stream_.device[mode] = device;
f@0:   stream_.doByteSwap[mode] = false;
f@0:   stream_.sampleRate = sampleRate;
f@0:   stream_.bufferSize = *bufferSize;
f@0:   stream_.nBuffers = 1;
f@0:   stream_.nUserChannels[mode] = channels;
f@0:   stream_.channelOffset[mode] = firstChannel;
f@0:   stream_.userFormat = format;
f@0:   stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
f@0: 
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
f@0:     stream_.userInterleaved = false;
f@0:   else
f@0:     stream_.userInterleaved = true;
f@0:   stream_.deviceInterleaved[mode] = true;
f@0: 
f@0:   // Set flags for buffer conversion.
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if ( stream_.userFormat != stream_.deviceFormat[mode] ||
f@0:        stream_.nUserChannels != stream_.nDeviceChannels )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
f@0:             stream_.nUserChannels[mode] > 1 )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   if ( stream_.doConvertBuffer[mode] )
f@0:     setConvertInfo( mode, 0 );
f@0: 
f@0:   // Allocate necessary internal buffers
f@0:   bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
f@0: 
f@0:   stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
f@0:   if ( !stream_.userBuffer[mode] ) {
f@0:     errorType = RtAudioError::MEMORY_ERROR;
f@0:     errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
f@0:     stream_.callbackInfo.priority = 15;
f@0:   else
f@0:     stream_.callbackInfo.priority = 0;
f@0: 
f@0:   ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
f@0:   ///! TODO: RTAUDIO_HOG_DEVICE       // Exclusive mode
f@0: 
f@0:   methodResult = SUCCESS;
f@0: 
f@0: Exit:
f@0:   //clean up
f@0:   SAFE_RELEASE( captureDevices );
f@0:   SAFE_RELEASE( renderDevices );
f@0:   SAFE_RELEASE( devicePtr );
f@0:   CoTaskMemFree( deviceFormat );
f@0: 
f@0:   // if method failed, close the stream
f@0:   if ( methodResult == FAILURE )
f@0:     closeStream();
f@0: 
f@0:   if ( !errorText_.empty() )
f@0:     error( errorType );
f@0:   return methodResult;
f@0: }
f@0: 
f@0: //=============================================================================
f@0: 
f@0: DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
f@0: {
f@0:   if ( wasapiPtr )
f@0:     ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
f@0: {
f@0:   if ( wasapiPtr )
f@0:     ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
f@0: {
f@0:   if ( wasapiPtr )
f@0:     ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: //-----------------------------------------------------------------------------
f@0: 
f@0: void RtApiWasapi::wasapiThread()
f@0: {
f@0:   // as this is a new thread, we must CoInitialize it
f@0:   CoInitialize( NULL );
f@0: 
f@0:   HRESULT hr;
f@0: 
f@0:   IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
f@0:   IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
f@0:   IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
f@0:   IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
f@0:   HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
f@0:   HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
f@0: 
f@0:   WAVEFORMATEX* captureFormat = NULL;
f@0:   WAVEFORMATEX* renderFormat = NULL;
f@0:   float captureSrRatio = 0.0f;
f@0:   float renderSrRatio = 0.0f;
f@0:   WasapiBuffer captureBuffer;
f@0:   WasapiBuffer renderBuffer;
f@0: 
f@0:   // declare local stream variables
f@0:   RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
f@0:   BYTE* streamBuffer = NULL;
f@0:   unsigned long captureFlags = 0;
f@0:   unsigned int bufferFrameCount = 0;
f@0:   unsigned int numFramesPadding = 0;
f@0:   unsigned int convBufferSize = 0;
f@0:   bool callbackPushed = false;
f@0:   bool callbackPulled = false;
f@0:   bool callbackStopped = false;
f@0:   int callbackResult = 0;
f@0: 
f@0:   // convBuffer is used to store converted buffers between WASAPI and the user
f@0:   char* convBuffer = NULL;
f@0:   unsigned int convBuffSize = 0;
f@0:   unsigned int deviceBuffSize = 0;
f@0: 
f@0:   errorText_.clear();
f@0:   RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
f@0: 
f@0:   // Attempt to assign "Pro Audio" characteristic to thread
f@0:   HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
f@0:   if ( AvrtDll ) {
f@0:     DWORD taskIndex = 0;
f@0:     TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
f@0:     AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
f@0:     FreeLibrary( AvrtDll );
f@0:   }
f@0: 
f@0:   // start capture stream if applicable
f@0:   if ( captureAudioClient ) {
f@0:     hr = captureAudioClient->GetMixFormat( &captureFormat );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
f@0: 
f@0:     // initialize capture stream according to desire buffer size
f@0:     float desiredBufferSize = stream_.bufferSize * captureSrRatio;
f@0:     REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
f@0: 
f@0:     if ( !captureClient ) {
f@0:       hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
f@0:                                            AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
f@0:                                            desiredBufferPeriod,
f@0:                                            desiredBufferPeriod,
f@0:                                            captureFormat,
f@0:                                            NULL );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
f@0:                                            ( void** ) &captureClient );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       // configure captureEvent to trigger on every available capture buffer
f@0:       captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
f@0:       if ( !captureEvent ) {
f@0:         errorType = RtAudioError::SYSTEM_ERROR;
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       hr = captureAudioClient->SetEventHandle( captureEvent );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
f@0:       ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
f@0:     }
f@0: 
f@0:     unsigned int inBufferSize = 0;
f@0:     hr = captureAudioClient->GetBufferSize( &inBufferSize );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     // scale outBufferSize according to stream->user sample rate ratio
f@0:     unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
f@0:     inBufferSize *= stream_.nDeviceChannels[INPUT];
f@0: 
f@0:     // set captureBuffer size
f@0:     captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
f@0: 
f@0:     // reset the capture stream
f@0:     hr = captureAudioClient->Reset();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     // start the capture stream
f@0:     hr = captureAudioClient->Start();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
f@0:       goto Exit;
f@0:     }
f@0:   }
f@0: 
f@0:   // start render stream if applicable
f@0:   if ( renderAudioClient ) {
f@0:     hr = renderAudioClient->GetMixFormat( &renderFormat );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
f@0: 
f@0:     // initialize render stream according to desire buffer size
f@0:     float desiredBufferSize = stream_.bufferSize * renderSrRatio;
f@0:     REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
f@0: 
f@0:     if ( !renderClient ) {
f@0:       hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
f@0:                                           AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
f@0:                                           desiredBufferPeriod,
f@0:                                           desiredBufferPeriod,
f@0:                                           renderFormat,
f@0:                                           NULL );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
f@0:                                           ( void** ) &renderClient );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       // configure renderEvent to trigger on every available render buffer
f@0:       renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
f@0:       if ( !renderEvent ) {
f@0:         errorType = RtAudioError::SYSTEM_ERROR;
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       hr = renderAudioClient->SetEventHandle( renderEvent );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
f@0:       ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
f@0:     }
f@0: 
f@0:     unsigned int outBufferSize = 0;
f@0:     hr = renderAudioClient->GetBufferSize( &outBufferSize );
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     // scale inBufferSize according to user->stream sample rate ratio
f@0:     unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
f@0:     outBufferSize *= stream_.nDeviceChannels[OUTPUT];
f@0: 
f@0:     // set renderBuffer size
f@0:     renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
f@0: 
f@0:     // reset the render stream
f@0:     hr = renderAudioClient->Reset();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
f@0:       goto Exit;
f@0:     }
f@0: 
f@0:     // start the render stream
f@0:     hr = renderAudioClient->Start();
f@0:     if ( FAILED( hr ) ) {
f@0:       errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
f@0:       goto Exit;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT ) {
f@0:     convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
f@0:     deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
f@0:   }
f@0:   else if ( stream_.mode == OUTPUT ) {
f@0:     convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
f@0:     deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
f@0:   }
f@0:   else if ( stream_.mode == DUPLEX ) {
f@0:     convBuffSize = max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
f@0:                              ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
f@0:     deviceBuffSize = max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
f@0:                                stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
f@0:   }
f@0: 
f@0:   convBuffer = ( char* ) malloc( convBuffSize );
f@0:   stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
f@0:   if ( !convBuffer || !stream_.deviceBuffer ) {
f@0:     errorType = RtAudioError::MEMORY_ERROR;
f@0:     errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
f@0:     goto Exit;
f@0:   }
f@0: 
f@0:   // stream process loop
f@0:   while ( stream_.state != STREAM_STOPPING ) {
f@0:     if ( !callbackPulled ) {
f@0:       // Callback Input
f@0:       // ==============
f@0:       // 1. Pull callback buffer from inputBuffer
f@0:       // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
f@0:       //                          Convert callback buffer to user format
f@0: 
f@0:       if ( captureAudioClient ) {
f@0:         // Pull callback buffer from inputBuffer
f@0:         callbackPulled = captureBuffer.pullBuffer( convBuffer,
f@0:                                                    ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
f@0:                                                    stream_.deviceFormat[INPUT] );
f@0: 
f@0:         if ( callbackPulled ) {
f@0:           // Convert callback buffer to user sample rate
f@0:           convertBufferWasapi( stream_.deviceBuffer,
f@0:                                convBuffer,
f@0:                                stream_.nDeviceChannels[INPUT],
f@0:                                captureFormat->nSamplesPerSec,
f@0:                                stream_.sampleRate,
f@0:                                ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
f@0:                                convBufferSize,
f@0:                                stream_.deviceFormat[INPUT] );
f@0: 
f@0:           if ( stream_.doConvertBuffer[INPUT] ) {
f@0:             // Convert callback buffer to user format
f@0:             convertBuffer( stream_.userBuffer[INPUT],
f@0:                            stream_.deviceBuffer,
f@0:                            stream_.convertInfo[INPUT] );
f@0:           }
f@0:           else {
f@0:             // no further conversion, simple copy deviceBuffer to userBuffer
f@0:             memcpy( stream_.userBuffer[INPUT],
f@0:                     stream_.deviceBuffer,
f@0:                     stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
f@0:           }
f@0:         }
f@0:       }
f@0:       else {
f@0:         // if there is no capture stream, set callbackPulled flag
f@0:         callbackPulled = true;
f@0:       }
f@0: 
f@0:       // Execute Callback
f@0:       // ================
f@0:       // 1. Execute user callback method
f@0:       // 2. Handle return value from callback
f@0: 
f@0:       // if callback has not requested the stream to stop
f@0:       if ( callbackPulled && !callbackStopped ) {
f@0:         // Execute user callback method
f@0:         callbackResult = callback( stream_.userBuffer[OUTPUT],
f@0:                                    stream_.userBuffer[INPUT],
f@0:                                    stream_.bufferSize,
f@0:                                    getStreamTime(),
f@0:                                    captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
f@0:                                    stream_.callbackInfo.userData );
f@0: 
f@0:         // Handle return value from callback
f@0:         if ( callbackResult == 1 ) {
f@0:           // instantiate a thread to stop this thread
f@0:           HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
f@0:           if ( !threadHandle ) {
f@0:             errorType = RtAudioError::THREAD_ERROR;
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
f@0:             goto Exit;
f@0:           }
f@0:           else if ( !CloseHandle( threadHandle ) ) {
f@0:             errorType = RtAudioError::THREAD_ERROR;
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
f@0:             goto Exit;
f@0:           }
f@0: 
f@0:           callbackStopped = true;
f@0:         }
f@0:         else if ( callbackResult == 2 ) {
f@0:           // instantiate a thread to stop this thread
f@0:           HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
f@0:           if ( !threadHandle ) {
f@0:             errorType = RtAudioError::THREAD_ERROR;
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
f@0:             goto Exit;
f@0:           }
f@0:           else if ( !CloseHandle( threadHandle ) ) {
f@0:             errorType = RtAudioError::THREAD_ERROR;
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
f@0:             goto Exit;
f@0:           }
f@0: 
f@0:           callbackStopped = true;
f@0:         }
f@0:       }
f@0:     }
f@0: 
f@0:     // Callback Output
f@0:     // ===============
f@0:     // 1. Convert callback buffer to stream format
f@0:     // 2. Convert callback buffer to stream sample rate and channel count
f@0:     // 3. Push callback buffer into outputBuffer
f@0: 
f@0:     if ( renderAudioClient && callbackPulled ) {
f@0:       if ( stream_.doConvertBuffer[OUTPUT] ) {
f@0:         // Convert callback buffer to stream format
f@0:         convertBuffer( stream_.deviceBuffer,
f@0:                        stream_.userBuffer[OUTPUT],
f@0:                        stream_.convertInfo[OUTPUT] );
f@0: 
f@0:       }
f@0: 
f@0:       // Convert callback buffer to stream sample rate
f@0:       convertBufferWasapi( convBuffer,
f@0:                            stream_.deviceBuffer,
f@0:                            stream_.nDeviceChannels[OUTPUT],
f@0:                            stream_.sampleRate,
f@0:                            renderFormat->nSamplesPerSec,
f@0:                            stream_.bufferSize,
f@0:                            convBufferSize,
f@0:                            stream_.deviceFormat[OUTPUT] );
f@0: 
f@0:       // Push callback buffer into outputBuffer
f@0:       callbackPushed = renderBuffer.pushBuffer( convBuffer,
f@0:                                                 convBufferSize * stream_.nDeviceChannels[OUTPUT],
f@0:                                                 stream_.deviceFormat[OUTPUT] );
f@0:     }
f@0:     else {
f@0:       // if there is no render stream, set callbackPushed flag
f@0:       callbackPushed = true;
f@0:     }
f@0: 
f@0:     // Stream Capture
f@0:     // ==============
f@0:     // 1. Get capture buffer from stream
f@0:     // 2. Push capture buffer into inputBuffer
f@0:     // 3. If 2. was successful: Release capture buffer
f@0: 
f@0:     if ( captureAudioClient ) {
f@0:       // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
f@0:       if ( !callbackPulled ) {
f@0:         WaitForSingleObject( captureEvent, INFINITE );
f@0:       }
f@0: 
f@0:       // Get capture buffer from stream
f@0:       hr = captureClient->GetBuffer( &streamBuffer,
f@0:                                      &bufferFrameCount,
f@0:                                      &captureFlags, NULL, NULL );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       if ( bufferFrameCount != 0 ) {
f@0:         // Push capture buffer into inputBuffer
f@0:         if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
f@0:                                        bufferFrameCount * stream_.nDeviceChannels[INPUT],
f@0:                                        stream_.deviceFormat[INPUT] ) )
f@0:         {
f@0:           // Release capture buffer
f@0:           hr = captureClient->ReleaseBuffer( bufferFrameCount );
f@0:           if ( FAILED( hr ) ) {
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
f@0:             goto Exit;
f@0:           }
f@0:         }
f@0:         else
f@0:         {
f@0:           // Inform WASAPI that capture was unsuccessful
f@0:           hr = captureClient->ReleaseBuffer( 0 );
f@0:           if ( FAILED( hr ) ) {
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
f@0:             goto Exit;
f@0:           }
f@0:         }
f@0:       }
f@0:       else
f@0:       {
f@0:         // Inform WASAPI that capture was unsuccessful
f@0:         hr = captureClient->ReleaseBuffer( 0 );
f@0:         if ( FAILED( hr ) ) {
f@0:           errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
f@0:           goto Exit;
f@0:         }
f@0:       }
f@0:     }
f@0: 
f@0:     // Stream Render
f@0:     // =============
f@0:     // 1. Get render buffer from stream
f@0:     // 2. Pull next buffer from outputBuffer
f@0:     // 3. If 2. was successful: Fill render buffer with next buffer
f@0:     //                          Release render buffer
f@0: 
f@0:     if ( renderAudioClient ) {
f@0:       // if the callback output buffer was not pushed to renderBuffer, wait for next render event
f@0:       if ( callbackPulled && !callbackPushed ) {
f@0:         WaitForSingleObject( renderEvent, INFINITE );
f@0:       }
f@0: 
f@0:       // Get render buffer from stream
f@0:       hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
f@0:       if ( FAILED( hr ) ) {
f@0:         errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
f@0:         goto Exit;
f@0:       }
f@0: 
f@0:       bufferFrameCount -= numFramesPadding;
f@0: 
f@0:       if ( bufferFrameCount != 0 ) {
f@0:         hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
f@0:         if ( FAILED( hr ) ) {
f@0:           errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
f@0:           goto Exit;
f@0:         }
f@0: 
f@0:         // Pull next buffer from outputBuffer
f@0:         // Fill render buffer with next buffer
f@0:         if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
f@0:                                       bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
f@0:                                       stream_.deviceFormat[OUTPUT] ) )
f@0:         {
f@0:           // Release render buffer
f@0:           hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
f@0:           if ( FAILED( hr ) ) {
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
f@0:             goto Exit;
f@0:           }
f@0:         }
f@0:         else
f@0:         {
f@0:           // Inform WASAPI that render was unsuccessful
f@0:           hr = renderClient->ReleaseBuffer( 0, 0 );
f@0:           if ( FAILED( hr ) ) {
f@0:             errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
f@0:             goto Exit;
f@0:           }
f@0:         }
f@0:       }
f@0:       else
f@0:       {
f@0:         // Inform WASAPI that render was unsuccessful
f@0:         hr = renderClient->ReleaseBuffer( 0, 0 );
f@0:         if ( FAILED( hr ) ) {
f@0:           errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
f@0:           goto Exit;
f@0:         }
f@0:       }
f@0:     }
f@0: 
f@0:     // if the callback buffer was pushed renderBuffer reset callbackPulled flag
f@0:     if ( callbackPushed ) {
f@0:       callbackPulled = false;
f@0:     }
f@0: 
f@0:     // tick stream time
f@0:     RtApi::tickStreamTime();
f@0:   }
f@0: 
f@0: Exit:
f@0:   // clean up
f@0:   CoTaskMemFree( captureFormat );
f@0:   CoTaskMemFree( renderFormat );
f@0: 
f@0:   free ( convBuffer );
f@0: 
f@0:   CoUninitialize();
f@0: 
f@0:   // update stream state
f@0:   stream_.state = STREAM_STOPPED;
f@0: 
f@0:   if ( errorText_.empty() )
f@0:     return;
f@0:   else
f@0:     error( errorType );
f@0: }
f@0: 
f@0: //******************** End of __WINDOWS_WASAPI__ *********************//
f@0: #endif
f@0: 
f@0: 
f@0: #if defined(__WINDOWS_DS__) // Windows DirectSound API
f@0: 
f@0: // Modified by Robin Davies, October 2005
f@0: // - Improvements to DirectX pointer chasing. 
f@0: // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
f@0: // - Auto-call CoInitialize for DSOUND and ASIO platforms.
f@0: // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
f@0: // Changed device query structure for RtAudio 4.0.7, January 2010
f@0: 
f@0: #include <dsound.h>
f@0: #include <assert.h>
f@0: #include <algorithm>
f@0: 
f@0: #if defined(__MINGW32__)
f@0:   // missing from latest mingw winapi
f@0: #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
f@0: #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
f@0: #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
f@0: #define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
f@0: #endif
f@0: 
f@0: #define MINIMUM_DEVICE_BUFFER_SIZE 32768
f@0: 
f@0: #ifdef _MSC_VER // if Microsoft Visual C++
f@0: #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
f@0: #endif
f@0: 
f@0: static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
f@0: {
f@0:   if ( pointer > bufferSize ) pointer -= bufferSize;
f@0:   if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
f@0:   if ( pointer < earlierPointer ) pointer += bufferSize;
f@0:   return pointer >= earlierPointer && pointer < laterPointer;
f@0: }
f@0: 
f@0: // A structure to hold various information related to the DirectSound
f@0: // API implementation.
f@0: struct DsHandle {
f@0:   unsigned int drainCounter; // Tracks callback counts when draining
f@0:   bool internalDrain;        // Indicates if stop is initiated from callback or not.
f@0:   void *id[2];
f@0:   void *buffer[2];
f@0:   bool xrun[2];
f@0:   UINT bufferPointer[2];  
f@0:   DWORD dsBufferSize[2];
f@0:   DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
f@0:   HANDLE condition;
f@0: 
f@0:   DsHandle()
f@0:     :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; }
f@0: };
f@0: 
f@0: // Declarations for utility functions, callbacks, and structures
f@0: // specific to the DirectSound implementation.
f@0: static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
f@0:                                           LPCTSTR description,
f@0:                                           LPCTSTR module,
f@0:                                           LPVOID lpContext );
f@0: 
f@0: static const char* getErrorString( int code );
f@0: 
f@0: static unsigned __stdcall callbackHandler( void *ptr );
f@0: 
f@0: struct DsDevice {
f@0:   LPGUID id[2];
f@0:   bool validId[2];
f@0:   bool found;
f@0:   std::string name;
f@0: 
f@0:   DsDevice()
f@0:   : found(false) { validId[0] = false; validId[1] = false; }
f@0: };
f@0: 
f@0: struct DsProbeData {
f@0:   bool isInput;
f@0:   std::vector<struct DsDevice>* dsDevices;
f@0: };
f@0: 
f@0: RtApiDs :: RtApiDs()
f@0: {
f@0:   // Dsound will run both-threaded. If CoInitialize fails, then just
f@0:   // accept whatever the mainline chose for a threading model.
f@0:   coInitialized_ = false;
f@0:   HRESULT hr = CoInitialize( NULL );
f@0:   if ( !FAILED( hr ) ) coInitialized_ = true;
f@0: }
f@0: 
f@0: RtApiDs :: ~RtApiDs()
f@0: {
f@0:   if ( coInitialized_ ) CoUninitialize(); // balanced call.
f@0:   if ( stream_.state != STREAM_CLOSED ) closeStream();
f@0: }
f@0: 
f@0: // The DirectSound default output is always the first device.
f@0: unsigned int RtApiDs :: getDefaultOutputDevice( void )
f@0: {
f@0:   return 0;
f@0: }
f@0: 
f@0: // The DirectSound default input is always the first input device,
f@0: // which is the first capture device enumerated.
f@0: unsigned int RtApiDs :: getDefaultInputDevice( void )
f@0: {
f@0:   return 0;
f@0: }
f@0: 
f@0: unsigned int RtApiDs :: getDeviceCount( void )
f@0: {
f@0:   // Set query flag for previously found devices to false, so that we
f@0:   // can check for any devices that have disappeared.
f@0:   for ( unsigned int i=0; i<dsDevices.size(); i++ )
f@0:     dsDevices[i].found = false;
f@0: 
f@0:   // Query DirectSound devices.
f@0:   struct DsProbeData probeInfo;
f@0:   probeInfo.isInput = false;
f@0:   probeInfo.dsDevices = &dsDevices;
f@0:   HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
f@0:   if ( FAILED( result ) ) {
f@0:     errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:   }
f@0: 
f@0:   // Query DirectSoundCapture devices.
f@0:   probeInfo.isInput = true;
f@0:   result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
f@0:   if ( FAILED( result ) ) {
f@0:     errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:   }
f@0: 
f@0:   // Clean out any devices that may have disappeared.
f@0:   std::vector< int > indices;
f@0:   for ( unsigned int i=0; i<dsDevices.size(); i++ )
f@0:     if ( dsDevices[i].found == false ) indices.push_back( i );
f@0:   //unsigned int nErased = 0;
f@0:   for ( unsigned int i=0; i<indices.size(); i++ )
f@0:     dsDevices.erase( dsDevices.begin()+indices[i] );
f@0:   //dsDevices.erase( dsDevices.begin()-nErased++ );
f@0: 
f@0:   return static_cast<unsigned int>(dsDevices.size());
f@0: }
f@0: 
f@0: RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   info.probed = false;
f@0: 
f@0:   if ( dsDevices.size() == 0 ) {
f@0:     // Force a query of all devices
f@0:     getDeviceCount();
f@0:     if ( dsDevices.size() == 0 ) {
f@0:       errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
f@0:       error( RtAudioError::INVALID_USE );
f@0:       return info;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( device >= dsDevices.size() ) {
f@0:     errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   HRESULT result;
f@0:   if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
f@0: 
f@0:   LPDIRECTSOUND output;
f@0:   DSCAPS outCaps;
f@0:   result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
f@0:   if ( FAILED( result ) ) {
f@0:     errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     goto probeInput;
f@0:   }
f@0: 
f@0:   outCaps.dwSize = sizeof( outCaps );
f@0:   result = output->GetCaps( &outCaps );
f@0:   if ( FAILED( result ) ) {
f@0:     output->Release();
f@0:     errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     goto probeInput;
f@0:   }
f@0: 
f@0:   // Get output channel information.
f@0:   info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
f@0: 
f@0:   // Get sample rate information.
f@0:   info.sampleRates.clear();
f@0:   for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
f@0:     if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
f@0:          SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
f@0:       info.sampleRates.push_back( SAMPLE_RATES[k] );
f@0:   }
f@0: 
f@0:   // Get format information.
f@0:   if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:   if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
f@0: 
f@0:   output->Release();
f@0: 
f@0:   if ( getDefaultOutputDevice() == device )
f@0:     info.isDefaultOutput = true;
f@0: 
f@0:   if ( dsDevices[ device ].validId[1] == false ) {
f@0:     info.name = dsDevices[ device ].name;
f@0:     info.probed = true;
f@0:     return info;
f@0:   }
f@0: 
f@0:  probeInput:
f@0: 
f@0:   LPDIRECTSOUNDCAPTURE input;
f@0:   result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
f@0:   if ( FAILED( result ) ) {
f@0:     errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   DSCCAPS inCaps;
f@0:   inCaps.dwSize = sizeof( inCaps );
f@0:   result = input->GetCaps( &inCaps );
f@0:   if ( FAILED( result ) ) {
f@0:     input->Release();
f@0:     errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Get input channel information.
f@0:   info.inputChannels = inCaps.dwChannels;
f@0: 
f@0:   // Get sample rate and format information.
f@0:   std::vector<unsigned int> rates;
f@0:   if ( inCaps.dwChannels >= 2 ) {
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0: 
f@0:     if ( info.nativeFormats & RTAUDIO_SINT16 ) {
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
f@0:     }
f@0:     else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
f@0:     }
f@0:   }
f@0:   else if ( inCaps.dwChannels == 1 ) {
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0:     if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
f@0: 
f@0:     if ( info.nativeFormats & RTAUDIO_SINT16 ) {
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
f@0:     }
f@0:     else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
f@0:       if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
f@0:     }
f@0:   }
f@0:   else info.inputChannels = 0; // technically, this would be an error
f@0: 
f@0:   input->Release();
f@0: 
f@0:   if ( info.inputChannels == 0 ) return info;
f@0: 
f@0:   // Copy the supported rates to the info structure but avoid duplication.
f@0:   bool found;
f@0:   for ( unsigned int i=0; i<rates.size(); i++ ) {
f@0:     found = false;
f@0:     for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
f@0:       if ( rates[i] == info.sampleRates[j] ) {
f@0:         found = true;
f@0:         break;
f@0:       }
f@0:     }
f@0:     if ( found == false ) info.sampleRates.push_back( rates[i] );
f@0:   }
f@0:   std::sort( info.sampleRates.begin(), info.sampleRates.end() );
f@0: 
f@0:   // If device opens for both playback and capture, we determine the channels.
f@0:   if ( info.outputChannels > 0 && info.inputChannels > 0 )
f@0:     info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
f@0: 
f@0:   if ( device == 0 ) info.isDefaultInput = true;
f@0: 
f@0:   // Copy name and return.
f@0:   info.name = dsDevices[ device ].name;
f@0:   info.probed = true;
f@0:   return info;
f@0: }
f@0: 
f@0: bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
f@0:                                  unsigned int firstChannel, unsigned int sampleRate,
f@0:                                  RtAudioFormat format, unsigned int *bufferSize,
f@0:                                  RtAudio::StreamOptions *options )
f@0: {
f@0:   if ( channels + firstChannel > 2 ) {
f@0:     errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   size_t nDevices = dsDevices.size();
f@0:   if ( nDevices == 0 ) {
f@0:     // This should not happen because a check is made before this function is called.
f@0:     errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     // This should not happen because a check is made before this function is called.
f@0:     errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   if ( mode == OUTPUT ) {
f@0:     if ( dsDevices[ device ].validId[0] == false ) {
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0:   else { // mode == INPUT
f@0:     if ( dsDevices[ device ].validId[1] == false ) {
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0: 
f@0:   // According to a note in PortAudio, using GetDesktopWindow()
f@0:   // instead of GetForegroundWindow() is supposed to avoid problems
f@0:   // that occur when the application's window is not the foreground
f@0:   // window.  Also, if the application window closes before the
f@0:   // DirectSound buffer, DirectSound can crash.  In the past, I had
f@0:   // problems when using GetDesktopWindow() but it seems fine now
f@0:   // (January 2010).  I'll leave it commented here.
f@0:   // HWND hWnd = GetForegroundWindow();
f@0:   HWND hWnd = GetDesktopWindow();
f@0: 
f@0:   // Check the numberOfBuffers parameter and limit the lowest value to
f@0:   // two.  This is a judgement call and a value of two is probably too
f@0:   // low for capture, but it should work for playback.
f@0:   int nBuffers = 0;
f@0:   if ( options ) nBuffers = options->numberOfBuffers;
f@0:   if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
f@0:   if ( nBuffers < 2 ) nBuffers = 3;
f@0: 
f@0:   // Check the lower range of the user-specified buffer size and set
f@0:   // (arbitrarily) to a lower bound of 32.
f@0:   if ( *bufferSize < 32 ) *bufferSize = 32;
f@0: 
f@0:   // Create the wave format structure.  The data format setting will
f@0:   // be determined later.
f@0:   WAVEFORMATEX waveFormat;
f@0:   ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
f@0:   waveFormat.wFormatTag = WAVE_FORMAT_PCM;
f@0:   waveFormat.nChannels = channels + firstChannel;
f@0:   waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
f@0: 
f@0:   // Determine the device buffer size. By default, we'll use the value
f@0:   // defined above (32K), but we will grow it to make allowances for
f@0:   // very large software buffer sizes.
f@0:   DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
f@0:   DWORD dsPointerLeadTime = 0;
f@0: 
f@0:   void *ohandle = 0, *bhandle = 0;
f@0:   HRESULT result;
f@0:   if ( mode == OUTPUT ) {
f@0: 
f@0:     LPDIRECTSOUND output;
f@0:     result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     DSCAPS outCaps;
f@0:     outCaps.dwSize = sizeof( outCaps );
f@0:     result = output->GetCaps( &outCaps );
f@0:     if ( FAILED( result ) ) {
f@0:       output->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Check channel information.
f@0:     if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
f@0:       errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Check format information.  Use 16-bit format unless not
f@0:     // supported or user requests 8-bit.
f@0:     if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
f@0:          !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
f@0:       waveFormat.wBitsPerSample = 16;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:     }
f@0:     else {
f@0:       waveFormat.wBitsPerSample = 8;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT8;
f@0:     }
f@0:     stream_.userFormat = format;
f@0: 
f@0:     // Update wave format structure and buffer information.
f@0:     waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
f@0:     waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
f@0:     dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
f@0: 
f@0:     // If the user wants an even bigger buffer, increase the device buffer size accordingly.
f@0:     while ( dsPointerLeadTime * 2U > dsBufferSize )
f@0:       dsBufferSize *= 2;
f@0: 
f@0:     // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
f@0:     // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
f@0:     // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
f@0:     result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
f@0:     if ( FAILED( result ) ) {
f@0:       output->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Even though we will write to the secondary buffer, we need to
f@0:     // access the primary buffer to set the correct output format
f@0:     // (since the default is 8-bit, 22 kHz!).  Setup the DS primary
f@0:     // buffer description.
f@0:     DSBUFFERDESC bufferDescription;
f@0:     ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
f@0:     bufferDescription.dwSize = sizeof( DSBUFFERDESC );
f@0:     bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
f@0: 
f@0:     // Obtain the primary buffer
f@0:     LPDIRECTSOUNDBUFFER buffer;
f@0:     result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
f@0:     if ( FAILED( result ) ) {
f@0:       output->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Set the primary DS buffer sound format.
f@0:     result = buffer->SetFormat( &waveFormat );
f@0:     if ( FAILED( result ) ) {
f@0:       output->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Setup the secondary DS buffer description.
f@0:     ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
f@0:     bufferDescription.dwSize = sizeof( DSBUFFERDESC );
f@0:     bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
f@0:                                   DSBCAPS_GLOBALFOCUS |
f@0:                                   DSBCAPS_GETCURRENTPOSITION2 |
f@0:                                   DSBCAPS_LOCHARDWARE );  // Force hardware mixing
f@0:     bufferDescription.dwBufferBytes = dsBufferSize;
f@0:     bufferDescription.lpwfxFormat = &waveFormat;
f@0: 
f@0:     // Try to create the secondary DS buffer.  If that doesn't work,
f@0:     // try to use software mixing.  Otherwise, there's a problem.
f@0:     result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
f@0:     if ( FAILED( result ) ) {
f@0:       bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
f@0:                                     DSBCAPS_GLOBALFOCUS |
f@0:                                     DSBCAPS_GETCURRENTPOSITION2 |
f@0:                                     DSBCAPS_LOCSOFTWARE );  // Force software mixing
f@0:       result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
f@0:       if ( FAILED( result ) ) {
f@0:         output->Release();
f@0:         errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
f@0:         errorText_ = errorStream_.str();
f@0:         return FAILURE;
f@0:       }
f@0:     }
f@0: 
f@0:     // Get the buffer size ... might be different from what we specified.
f@0:     DSBCAPS dsbcaps;
f@0:     dsbcaps.dwSize = sizeof( DSBCAPS );
f@0:     result = buffer->GetCaps( &dsbcaps );
f@0:     if ( FAILED( result ) ) {
f@0:       output->Release();
f@0:       buffer->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     dsBufferSize = dsbcaps.dwBufferBytes;
f@0: 
f@0:     // Lock the DS buffer
f@0:     LPVOID audioPtr;
f@0:     DWORD dataLen;
f@0:     result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       output->Release();
f@0:       buffer->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Zero the DS buffer
f@0:     ZeroMemory( audioPtr, dataLen );
f@0: 
f@0:     // Unlock the DS buffer
f@0:     result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       output->Release();
f@0:       buffer->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     ohandle = (void *) output;
f@0:     bhandle = (void *) buffer;
f@0:   }
f@0: 
f@0:   if ( mode == INPUT ) {
f@0: 
f@0:     LPDIRECTSOUNDCAPTURE input;
f@0:     result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     DSCCAPS inCaps;
f@0:     inCaps.dwSize = sizeof( inCaps );
f@0:     result = input->GetCaps( &inCaps );
f@0:     if ( FAILED( result ) ) {
f@0:       input->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Check channel information.
f@0:     if ( inCaps.dwChannels < channels + firstChannel ) {
f@0:       errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Check format information.  Use 16-bit format unless user
f@0:     // requests 8-bit.
f@0:     DWORD deviceFormats;
f@0:     if ( channels + firstChannel == 2 ) {
f@0:       deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
f@0:       if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
f@0:         waveFormat.wBitsPerSample = 8;
f@0:         stream_.deviceFormat[mode] = RTAUDIO_SINT8;
f@0:       }
f@0:       else { // assume 16-bit is supported
f@0:         waveFormat.wBitsPerSample = 16;
f@0:         stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:       }
f@0:     }
f@0:     else { // channel == 1
f@0:       deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
f@0:       if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
f@0:         waveFormat.wBitsPerSample = 8;
f@0:         stream_.deviceFormat[mode] = RTAUDIO_SINT8;
f@0:       }
f@0:       else { // assume 16-bit is supported
f@0:         waveFormat.wBitsPerSample = 16;
f@0:         stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:       }
f@0:     }
f@0:     stream_.userFormat = format;
f@0: 
f@0:     // Update wave format structure and buffer information.
f@0:     waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
f@0:     waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
f@0:     dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
f@0: 
f@0:     // If the user wants an even bigger buffer, increase the device buffer size accordingly.
f@0:     while ( dsPointerLeadTime * 2U > dsBufferSize )
f@0:       dsBufferSize *= 2;
f@0: 
f@0:     // Setup the secondary DS buffer description.
f@0:     DSCBUFFERDESC bufferDescription;
f@0:     ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
f@0:     bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
f@0:     bufferDescription.dwFlags = 0;
f@0:     bufferDescription.dwReserved = 0;
f@0:     bufferDescription.dwBufferBytes = dsBufferSize;
f@0:     bufferDescription.lpwfxFormat = &waveFormat;
f@0: 
f@0:     // Create the capture buffer.
f@0:     LPDIRECTSOUNDCAPTUREBUFFER buffer;
f@0:     result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
f@0:     if ( FAILED( result ) ) {
f@0:       input->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Get the buffer size ... might be different from what we specified.
f@0:     DSCBCAPS dscbcaps;
f@0:     dscbcaps.dwSize = sizeof( DSCBCAPS );
f@0:     result = buffer->GetCaps( &dscbcaps );
f@0:     if ( FAILED( result ) ) {
f@0:       input->Release();
f@0:       buffer->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     dsBufferSize = dscbcaps.dwBufferBytes;
f@0: 
f@0:     // NOTE: We could have a problem here if this is a duplex stream
f@0:     // and the play and capture hardware buffer sizes are different
f@0:     // (I'm actually not sure if that is a problem or not).
f@0:     // Currently, we are not verifying that.
f@0: 
f@0:     // Lock the capture buffer
f@0:     LPVOID audioPtr;
f@0:     DWORD dataLen;
f@0:     result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       input->Release();
f@0:       buffer->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     // Zero the buffer
f@0:     ZeroMemory( audioPtr, dataLen );
f@0: 
f@0:     // Unlock the buffer
f@0:     result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       input->Release();
f@0:       buffer->Release();
f@0:       errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     ohandle = (void *) input;
f@0:     bhandle = (void *) buffer;
f@0:   }
f@0: 
f@0:   // Set various stream parameters
f@0:   DsHandle *handle = 0;
f@0:   stream_.nDeviceChannels[mode] = channels + firstChannel;
f@0:   stream_.nUserChannels[mode] = channels;
f@0:   stream_.bufferSize = *bufferSize;
f@0:   stream_.channelOffset[mode] = firstChannel;
f@0:   stream_.deviceInterleaved[mode] = true;
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
f@0:   else stream_.userInterleaved = true;
f@0: 
f@0:   // Set flag for buffer conversion
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if (stream_.userFormat != stream_.deviceFormat[mode])
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
f@0:        stream_.nUserChannels[mode] > 1 )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   // Allocate necessary internal buffers
f@0:   long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
f@0:   stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
f@0:   if ( stream_.userBuffer[mode] == NULL ) {
f@0:     errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
f@0:     goto error;
f@0:   }
f@0: 
f@0:   if ( stream_.doConvertBuffer[mode] ) {
f@0: 
f@0:     bool makeBuffer = true;
f@0:     bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
f@0:     if ( mode == INPUT ) {
f@0:       if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
f@0:         unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
f@0:         if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( makeBuffer ) {
f@0:       bufferBytes *= *bufferSize;
f@0:       if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
f@0:       stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
f@0:       if ( stream_.deviceBuffer == NULL ) {
f@0:         errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
f@0:         goto error;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   // Allocate our DsHandle structures for the stream.
f@0:   if ( stream_.apiHandle == 0 ) {
f@0:     try {
f@0:       handle = new DsHandle;
f@0:     }
f@0:     catch ( std::bad_alloc& ) {
f@0:       errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     // Create a manual-reset event.
f@0:     handle->condition = CreateEvent( NULL,   // no security
f@0:                                      TRUE,   // manual-reset
f@0:                                      FALSE,  // non-signaled initially
f@0:                                      NULL ); // unnamed
f@0:     stream_.apiHandle = (void *) handle;
f@0:   }
f@0:   else
f@0:     handle = (DsHandle *) stream_.apiHandle;
f@0:   handle->id[mode] = ohandle;
f@0:   handle->buffer[mode] = bhandle;
f@0:   handle->dsBufferSize[mode] = dsBufferSize;
f@0:   handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
f@0: 
f@0:   stream_.device[mode] = device;
f@0:   stream_.state = STREAM_STOPPED;
f@0:   if ( stream_.mode == OUTPUT && mode == INPUT )
f@0:     // We had already set up an output stream.
f@0:     stream_.mode = DUPLEX;
f@0:   else
f@0:     stream_.mode = mode;
f@0:   stream_.nBuffers = nBuffers;
f@0:   stream_.sampleRate = sampleRate;
f@0: 
f@0:   // Setup the buffer conversion information structure.
f@0:   if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
f@0: 
f@0:   // Setup the callback thread.
f@0:   if ( stream_.callbackInfo.isRunning == false ) {
f@0:     unsigned threadId;
f@0:     stream_.callbackInfo.isRunning = true;
f@0:     stream_.callbackInfo.object = (void *) this;
f@0:     stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
f@0:                                                   &stream_.callbackInfo, 0, &threadId );
f@0:     if ( stream_.callbackInfo.thread == 0 ) {
f@0:       errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     // Boost DS thread priority
f@0:     SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
f@0:   }
f@0:   return SUCCESS;
f@0: 
f@0:  error:
f@0:   if ( handle ) {
f@0:     if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
f@0:       LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
f@0:       LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
f@0:       if ( buffer ) buffer->Release();
f@0:       object->Release();
f@0:     }
f@0:     if ( handle->buffer[1] ) {
f@0:       LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
f@0:       LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
f@0:       if ( buffer ) buffer->Release();
f@0:       object->Release();
f@0:     }
f@0:     CloseHandle( handle->condition );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_CLOSED;
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: void RtApiDs :: closeStream()
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiDs::closeStream(): no open stream to close!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   // Stop the callback thread.
f@0:   stream_.callbackInfo.isRunning = false;
f@0:   WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
f@0:   CloseHandle( (HANDLE) stream_.callbackInfo.thread );
f@0: 
f@0:   DsHandle *handle = (DsHandle *) stream_.apiHandle;
f@0:   if ( handle ) {
f@0:     if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
f@0:       LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
f@0:       LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
f@0:       if ( buffer ) {
f@0:         buffer->Stop();
f@0:         buffer->Release();
f@0:       }
f@0:       object->Release();
f@0:     }
f@0:     if ( handle->buffer[1] ) {
f@0:       LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
f@0:       LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
f@0:       if ( buffer ) {
f@0:         buffer->Stop();
f@0:         buffer->Release();
f@0:       }
f@0:       object->Release();
f@0:     }
f@0:     CloseHandle( handle->condition );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.state = STREAM_CLOSED;
f@0: }
f@0: 
f@0: void RtApiDs :: startStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiDs::startStream(): the stream is already running!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   DsHandle *handle = (DsHandle *) stream_.apiHandle;
f@0: 
f@0:   // Increase scheduler frequency on lesser windows (a side-effect of
f@0:   // increasing timer accuracy).  On greater windows (Win2K or later),
f@0:   // this is already in effect.
f@0:   timeBeginPeriod( 1 ); 
f@0: 
f@0:   buffersRolling = false;
f@0:   duplexPrerollBytes = 0;
f@0: 
f@0:   if ( stream_.mode == DUPLEX ) {
f@0:     // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
f@0:     duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
f@0:   }
f@0: 
f@0:   HRESULT result = 0;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
f@0:     result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
f@0:     result = buffer->Start( DSCBSTART_LOOPING );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   handle->drainCounter = 0;
f@0:   handle->internalDrain = false;
f@0:   ResetEvent( handle->condition );
f@0:   stream_.state = STREAM_RUNNING;
f@0: 
f@0:  unlock:
f@0:   if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiDs :: stopStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   HRESULT result = 0;
f@0:   LPVOID audioPtr;
f@0:   DWORD dataLen;
f@0:   DsHandle *handle = (DsHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     if ( handle->drainCounter == 0 ) {
f@0:       handle->drainCounter = 2;
f@0:       WaitForSingleObject( handle->condition, INFINITE );  // block until signaled
f@0:     }
f@0: 
f@0:     stream_.state = STREAM_STOPPED;
f@0: 
f@0:     MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:     // Stop the buffer and clear memory
f@0:     LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
f@0:     result = buffer->Stop();
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Lock the buffer and clear it so that if we start to play again,
f@0:     // we won't have old data playing.
f@0:     result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Zero the DS buffer
f@0:     ZeroMemory( audioPtr, dataLen );
f@0: 
f@0:     // Unlock the DS buffer
f@0:     result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // If we start playing again, we must begin at beginning of buffer.
f@0:     handle->bufferPointer[0] = 0;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0:     LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
f@0:     audioPtr = NULL;
f@0:     dataLen = 0;
f@0: 
f@0:     stream_.state = STREAM_STOPPED;
f@0: 
f@0:     if ( stream_.mode != DUPLEX )
f@0:       MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:     result = buffer->Stop();
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Lock the buffer and clear it so that if we start to play again,
f@0:     // we won't have old data playing.
f@0:     result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Zero the DS buffer
f@0:     ZeroMemory( audioPtr, dataLen );
f@0: 
f@0:     // Unlock the DS buffer
f@0:     result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // If we start recording again, we must begin at beginning of buffer.
f@0:     handle->bufferPointer[1] = 0;
f@0:   }
f@0: 
f@0:  unlock:
f@0:   timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiDs :: abortStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   DsHandle *handle = (DsHandle *) stream_.apiHandle;
f@0:   handle->drainCounter = 2;
f@0: 
f@0:   stopStream();
f@0: }
f@0: 
f@0: void RtApiDs :: callbackEvent()
f@0: {
f@0:   if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
f@0:     Sleep( 50 ); // sleep 50 milliseconds
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
f@0:   DsHandle *handle = (DsHandle *) stream_.apiHandle;
f@0: 
f@0:   // Check if we were draining the stream and signal is finished.
f@0:   if ( handle->drainCounter > stream_.nBuffers + 2 ) {
f@0: 
f@0:     stream_.state = STREAM_STOPPING;
f@0:     if ( handle->internalDrain == false )
f@0:       SetEvent( handle->condition );
f@0:     else
f@0:       stopStream();
f@0:     return;
f@0:   }
f@0: 
f@0:   // Invoke user callback to get fresh output data UNLESS we are
f@0:   // draining stream.
f@0:   if ( handle->drainCounter == 0 ) {
f@0:     RtAudioCallback callback = (RtAudioCallback) info->callback;
f@0:     double streamTime = getStreamTime();
f@0:     RtAudioStreamStatus status = 0;
f@0:     if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
f@0:       status |= RTAUDIO_OUTPUT_UNDERFLOW;
f@0:       handle->xrun[0] = false;
f@0:     }
f@0:     if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
f@0:       status |= RTAUDIO_INPUT_OVERFLOW;
f@0:       handle->xrun[1] = false;
f@0:     }
f@0:     int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
f@0:                                   stream_.bufferSize, streamTime, status, info->userData );
f@0:     if ( cbReturnValue == 2 ) {
f@0:       stream_.state = STREAM_STOPPING;
f@0:       handle->drainCounter = 2;
f@0:       abortStream();
f@0:       return;
f@0:     }
f@0:     else if ( cbReturnValue == 1 ) {
f@0:       handle->drainCounter = 1;
f@0:       handle->internalDrain = true;
f@0:     }
f@0:   }
f@0: 
f@0:   HRESULT result;
f@0:   DWORD currentWritePointer, safeWritePointer;
f@0:   DWORD currentReadPointer, safeReadPointer;
f@0:   UINT nextWritePointer;
f@0: 
f@0:   LPVOID buffer1 = NULL;
f@0:   LPVOID buffer2 = NULL;
f@0:   DWORD bufferSize1 = 0;
f@0:   DWORD bufferSize2 = 0;
f@0: 
f@0:   char *buffer;
f@0:   long bufferBytes;
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     MUTEX_UNLOCK( &stream_.mutex );
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( buffersRolling == false ) {
f@0:     if ( stream_.mode == DUPLEX ) {
f@0:       //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
f@0: 
f@0:       // It takes a while for the devices to get rolling. As a result,
f@0:       // there's no guarantee that the capture and write device pointers
f@0:       // will move in lockstep.  Wait here for both devices to start
f@0:       // rolling, and then set our buffer pointers accordingly.
f@0:       // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
f@0:       // bytes later than the write buffer.
f@0: 
f@0:       // Stub: a serious risk of having a pre-emptive scheduling round
f@0:       // take place between the two GetCurrentPosition calls... but I'm
f@0:       // really not sure how to solve the problem.  Temporarily boost to
f@0:       // Realtime priority, maybe; but I'm not sure what priority the
f@0:       // DirectSound service threads run at. We *should* be roughly
f@0:       // within a ms or so of correct.
f@0: 
f@0:       LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
f@0:       LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
f@0: 
f@0:       DWORD startSafeWritePointer, startSafeReadPointer;
f@0: 
f@0:       result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
f@0:       if ( FAILED( result ) ) {
f@0:         errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
f@0:         errorText_ = errorStream_.str();
f@0:         error( RtAudioError::SYSTEM_ERROR );
f@0:         return;
f@0:       }
f@0:       result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
f@0:       if ( FAILED( result ) ) {
f@0:         errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
f@0:         errorText_ = errorStream_.str();
f@0:         error( RtAudioError::SYSTEM_ERROR );
f@0:         return;
f@0:       }
f@0:       while ( true ) {
f@0:         result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
f@0:         if ( FAILED( result ) ) {
f@0:           errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
f@0:           errorText_ = errorStream_.str();
f@0:           error( RtAudioError::SYSTEM_ERROR );
f@0:           return;
f@0:         }
f@0:         result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
f@0:         if ( FAILED( result ) ) {
f@0:           errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
f@0:           errorText_ = errorStream_.str();
f@0:           error( RtAudioError::SYSTEM_ERROR );
f@0:           return;
f@0:         }
f@0:         if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
f@0:         Sleep( 1 );
f@0:       }
f@0: 
f@0:       //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
f@0: 
f@0:       handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
f@0:       if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
f@0:       handle->bufferPointer[1] = safeReadPointer;
f@0:     }
f@0:     else if ( stream_.mode == OUTPUT ) {
f@0: 
f@0:       // Set the proper nextWritePosition after initial startup.
f@0:       LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
f@0:       result = dsWriteBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );
f@0:       if ( FAILED( result ) ) {
f@0:         errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
f@0:         errorText_ = errorStream_.str();
f@0:         error( RtAudioError::SYSTEM_ERROR );
f@0:         return;
f@0:       }
f@0:       handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
f@0:       if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
f@0:     }
f@0: 
f@0:     buffersRolling = true;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     
f@0:     LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
f@0: 
f@0:     if ( handle->drainCounter > 1 ) { // write zeros to the output stream
f@0:       bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
f@0:       bufferBytes *= formatBytes( stream_.userFormat );
f@0:       memset( stream_.userBuffer[0], 0, bufferBytes );
f@0:     }
f@0: 
f@0:     // Setup parameters and do buffer conversion if necessary.
f@0:     if ( stream_.doConvertBuffer[0] ) {
f@0:       buffer = stream_.deviceBuffer;
f@0:       convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
f@0:       bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
f@0:       bufferBytes *= formatBytes( stream_.deviceFormat[0] );
f@0:     }
f@0:     else {
f@0:       buffer = stream_.userBuffer[0];
f@0:       bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
f@0:       bufferBytes *= formatBytes( stream_.userFormat );
f@0:     }
f@0: 
f@0:     // No byte swapping necessary in DirectSound implementation.
f@0: 
f@0:     // Ahhh ... windoze.  16-bit data is signed but 8-bit data is
f@0:     // unsigned.  So, we need to convert our signed 8-bit data here to
f@0:     // unsigned.
f@0:     if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
f@0:       for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
f@0: 
f@0:     DWORD dsBufferSize = handle->dsBufferSize[0];
f@0:     nextWritePointer = handle->bufferPointer[0];
f@0: 
f@0:     DWORD endWrite, leadPointer;
f@0:     while ( true ) {
f@0:       // Find out where the read and "safe write" pointers are.
f@0:       result = dsBuffer->GetCurrentPosition( &currentWritePointer, &safeWritePointer );
f@0:       if ( FAILED( result ) ) {
f@0:         errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
f@0:         errorText_ = errorStream_.str();
f@0:         error( RtAudioError::SYSTEM_ERROR );
f@0:         return;
f@0:       }
f@0: 
f@0:       // We will copy our output buffer into the region between
f@0:       // safeWritePointer and leadPointer.  If leadPointer is not
f@0:       // beyond the next endWrite position, wait until it is.
f@0:       leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
f@0:       //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
f@0:       if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
f@0:       if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
f@0:       endWrite = nextWritePointer + bufferBytes;
f@0: 
f@0:       // Check whether the entire write region is behind the play pointer.
f@0:       if ( leadPointer >= endWrite ) break;
f@0: 
f@0:       // If we are here, then we must wait until the leadPointer advances
f@0:       // beyond the end of our next write region. We use the
f@0:       // Sleep() function to suspend operation until that happens.
f@0:       double millis = ( endWrite - leadPointer ) * 1000.0;
f@0:       millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
f@0:       if ( millis < 1.0 ) millis = 1.0;
f@0:       Sleep( (DWORD) millis );
f@0:     }
f@0: 
f@0:     if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
f@0:          || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { 
f@0:       // We've strayed into the forbidden zone ... resync the read pointer.
f@0:       handle->xrun[0] = true;
f@0:       nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
f@0:       if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
f@0:       handle->bufferPointer[0] = nextWritePointer;
f@0:       endWrite = nextWritePointer + bufferBytes;
f@0:     }
f@0: 
f@0:     // Lock free space in the buffer
f@0:     result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
f@0:                              &bufferSize1, &buffer2, &bufferSize2, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0: 
f@0:     // Copy our buffer into the DS buffer
f@0:     CopyMemory( buffer1, buffer, bufferSize1 );
f@0:     if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
f@0: 
f@0:     // Update our buffer offset and unlock sound buffer
f@0:     dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0:     nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
f@0:     handle->bufferPointer[0] = nextWritePointer;
f@0:   }
f@0: 
f@0:   // Don't bother draining input
f@0:   if ( handle->drainCounter ) {
f@0:     handle->drainCounter++;
f@0:     goto unlock;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     // Setup parameters.
f@0:     if ( stream_.doConvertBuffer[1] ) {
f@0:       buffer = stream_.deviceBuffer;
f@0:       bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
f@0:       bufferBytes *= formatBytes( stream_.deviceFormat[1] );
f@0:     }
f@0:     else {
f@0:       buffer = stream_.userBuffer[1];
f@0:       bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
f@0:       bufferBytes *= formatBytes( stream_.userFormat );
f@0:     }
f@0: 
f@0:     LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
f@0:     long nextReadPointer = handle->bufferPointer[1];
f@0:     DWORD dsBufferSize = handle->dsBufferSize[1];
f@0: 
f@0:     // Find out where the write and "safe read" pointers are.
f@0:     result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0: 
f@0:     if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
f@0:     DWORD endRead = nextReadPointer + bufferBytes;
f@0: 
f@0:     // Handling depends on whether we are INPUT or DUPLEX. 
f@0:     // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
f@0:     // then a wait here will drag the write pointers into the forbidden zone.
f@0:     // 
f@0:     // In DUPLEX mode, rather than wait, we will back off the read pointer until 
f@0:     // it's in a safe position. This causes dropouts, but it seems to be the only 
f@0:     // practical way to sync up the read and write pointers reliably, given the 
f@0:     // the very complex relationship between phase and increment of the read and write 
f@0:     // pointers.
f@0:     //
f@0:     // In order to minimize audible dropouts in DUPLEX mode, we will
f@0:     // provide a pre-roll period of 0.5 seconds in which we return
f@0:     // zeros from the read buffer while the pointers sync up.
f@0: 
f@0:     if ( stream_.mode == DUPLEX ) {
f@0:       if ( safeReadPointer < endRead ) {
f@0:         if ( duplexPrerollBytes <= 0 ) {
f@0:           // Pre-roll time over. Be more agressive.
f@0:           int adjustment = endRead-safeReadPointer;
f@0: 
f@0:           handle->xrun[1] = true;
f@0:           // Two cases:
f@0:           //   - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
f@0:           //     and perform fine adjustments later.
f@0:           //   - small adjustments: back off by twice as much.
f@0:           if ( adjustment >= 2*bufferBytes )
f@0:             nextReadPointer = safeReadPointer-2*bufferBytes;
f@0:           else
f@0:             nextReadPointer = safeReadPointer-bufferBytes-adjustment;
f@0: 
f@0:           if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
f@0: 
f@0:         }
f@0:         else {
f@0:           // In pre=roll time. Just do it.
f@0:           nextReadPointer = safeReadPointer - bufferBytes;
f@0:           while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
f@0:         }
f@0:         endRead = nextReadPointer + bufferBytes;
f@0:       }
f@0:     }
f@0:     else { // mode == INPUT
f@0:       while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
f@0:         // See comments for playback.
f@0:         double millis = (endRead - safeReadPointer) * 1000.0;
f@0:         millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
f@0:         if ( millis < 1.0 ) millis = 1.0;
f@0:         Sleep( (DWORD) millis );
f@0: 
f@0:         // Wake up and find out where we are now.
f@0:         result = dsBuffer->GetCurrentPosition( &currentReadPointer, &safeReadPointer );
f@0:         if ( FAILED( result ) ) {
f@0:           errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
f@0:           errorText_ = errorStream_.str();
f@0:           error( RtAudioError::SYSTEM_ERROR );
f@0:           return;
f@0:         }
f@0:       
f@0:         if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
f@0:       }
f@0:     }
f@0: 
f@0:     // Lock free space in the buffer
f@0:     result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
f@0:                              &bufferSize1, &buffer2, &bufferSize2, 0 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0: 
f@0:     if ( duplexPrerollBytes <= 0 ) {
f@0:       // Copy our buffer into the DS buffer
f@0:       CopyMemory( buffer, buffer1, bufferSize1 );
f@0:       if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
f@0:     }
f@0:     else {
f@0:       memset( buffer, 0, bufferSize1 );
f@0:       if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
f@0:       duplexPrerollBytes -= bufferSize1 + bufferSize2;
f@0:     }
f@0: 
f@0:     // Update our buffer offset and unlock sound buffer
f@0:     nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
f@0:     dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
f@0:     if ( FAILED( result ) ) {
f@0:       errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0:     handle->bufferPointer[1] = nextReadPointer;
f@0: 
f@0:     // No byte swapping necessary in DirectSound implementation.
f@0: 
f@0:     // If necessary, convert 8-bit data from unsigned to signed.
f@0:     if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
f@0:       for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
f@0: 
f@0:     // Do buffer conversion if necessary.
f@0:     if ( stream_.doConvertBuffer[1] )
f@0:       convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
f@0:   }
f@0: 
f@0:  unlock:
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0:   RtApi::tickStreamTime();
f@0: }
f@0: 
f@0: // Definitions for utility functions and callbacks
f@0: // specific to the DirectSound implementation.
f@0: 
f@0: static unsigned __stdcall callbackHandler( void *ptr )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) ptr;
f@0:   RtApiDs *object = (RtApiDs *) info->object;
f@0:   bool* isRunning = &info->isRunning;
f@0: 
f@0:   while ( *isRunning == true ) {
f@0:     object->callbackEvent();
f@0:   }
f@0: 
f@0:   _endthreadex( 0 );
f@0:   return 0;
f@0: }
f@0: 
f@0: #include "tchar.h"
f@0: 
f@0: static std::string convertTChar( LPCTSTR name )
f@0: {
f@0: #if defined( UNICODE ) || defined( _UNICODE )
f@0:   int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL);
f@0:   std::string s( length-1, '\0' );
f@0:   WideCharToMultiByte(CP_UTF8, 0, name, -1, &s[0], length, NULL, NULL);
f@0: #else
f@0:   std::string s( name );
f@0: #endif
f@0: 
f@0:   return s;
f@0: }
f@0: 
f@0: static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
f@0:                                           LPCTSTR description,
f@0:                                           LPCTSTR /*module*/,
f@0:                                           LPVOID lpContext )
f@0: {
f@0:   struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
f@0:   std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
f@0: 
f@0:   HRESULT hr;
f@0:   bool validDevice = false;
f@0:   if ( probeInfo.isInput == true ) {
f@0:     DSCCAPS caps;
f@0:     LPDIRECTSOUNDCAPTURE object;
f@0: 
f@0:     hr = DirectSoundCaptureCreate(  lpguid, &object,   NULL );
f@0:     if ( hr != DS_OK ) return TRUE;
f@0: 
f@0:     caps.dwSize = sizeof(caps);
f@0:     hr = object->GetCaps( &caps );
f@0:     if ( hr == DS_OK ) {
f@0:       if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
f@0:         validDevice = true;
f@0:     }
f@0:     object->Release();
f@0:   }
f@0:   else {
f@0:     DSCAPS caps;
f@0:     LPDIRECTSOUND object;
f@0:     hr = DirectSoundCreate(  lpguid, &object,   NULL );
f@0:     if ( hr != DS_OK ) return TRUE;
f@0: 
f@0:     caps.dwSize = sizeof(caps);
f@0:     hr = object->GetCaps( &caps );
f@0:     if ( hr == DS_OK ) {
f@0:       if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
f@0:         validDevice = true;
f@0:     }
f@0:     object->Release();
f@0:   }
f@0: 
f@0:   // If good device, then save its name and guid.
f@0:   std::string name = convertTChar( description );
f@0:   //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
f@0:   if ( lpguid == NULL )
f@0:     name = "Default Device";
f@0:   if ( validDevice ) {
f@0:     for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
f@0:       if ( dsDevices[i].name == name ) {
f@0:         dsDevices[i].found = true;
f@0:         if ( probeInfo.isInput ) {
f@0:           dsDevices[i].id[1] = lpguid;
f@0:           dsDevices[i].validId[1] = true;
f@0:         }
f@0:         else {
f@0:           dsDevices[i].id[0] = lpguid;
f@0:           dsDevices[i].validId[0] = true;
f@0:         }
f@0:         return TRUE;
f@0:       }
f@0:     }
f@0: 
f@0:     DsDevice device;
f@0:     device.name = name;
f@0:     device.found = true;
f@0:     if ( probeInfo.isInput ) {
f@0:       device.id[1] = lpguid;
f@0:       device.validId[1] = true;
f@0:     }
f@0:     else {
f@0:       device.id[0] = lpguid;
f@0:       device.validId[0] = true;
f@0:     }
f@0:     dsDevices.push_back( device );
f@0:   }
f@0: 
f@0:   return TRUE;
f@0: }
f@0: 
f@0: static const char* getErrorString( int code )
f@0: {
f@0:   switch ( code ) {
f@0: 
f@0:   case DSERR_ALLOCATED:
f@0:     return "Already allocated";
f@0: 
f@0:   case DSERR_CONTROLUNAVAIL:
f@0:     return "Control unavailable";
f@0: 
f@0:   case DSERR_INVALIDPARAM:
f@0:     return "Invalid parameter";
f@0: 
f@0:   case DSERR_INVALIDCALL:
f@0:     return "Invalid call";
f@0: 
f@0:   case DSERR_GENERIC:
f@0:     return "Generic error";
f@0: 
f@0:   case DSERR_PRIOLEVELNEEDED:
f@0:     return "Priority level needed";
f@0: 
f@0:   case DSERR_OUTOFMEMORY:
f@0:     return "Out of memory";
f@0: 
f@0:   case DSERR_BADFORMAT:
f@0:     return "The sample rate or the channel format is not supported";
f@0: 
f@0:   case DSERR_UNSUPPORTED:
f@0:     return "Not supported";
f@0: 
f@0:   case DSERR_NODRIVER:
f@0:     return "No driver";
f@0: 
f@0:   case DSERR_ALREADYINITIALIZED:
f@0:     return "Already initialized";
f@0: 
f@0:   case DSERR_NOAGGREGATION:
f@0:     return "No aggregation";
f@0: 
f@0:   case DSERR_BUFFERLOST:
f@0:     return "Buffer lost";
f@0: 
f@0:   case DSERR_OTHERAPPHASPRIO:
f@0:     return "Another application already has priority";
f@0: 
f@0:   case DSERR_UNINITIALIZED:
f@0:     return "Uninitialized";
f@0: 
f@0:   default:
f@0:     return "DirectSound unknown error";
f@0:   }
f@0: }
f@0: //******************** End of __WINDOWS_DS__ *********************//
f@0: #endif
f@0: 
f@0: 
f@0: #if defined(__LINUX_ALSA__)
f@0: 
f@0: #include <alsa/asoundlib.h>
f@0: #include <unistd.h>
f@0: 
f@0:   // A structure to hold various information related to the ALSA API
f@0:   // implementation.
f@0: struct AlsaHandle {
f@0:   snd_pcm_t *handles[2];
f@0:   bool synchronized;
f@0:   bool xrun[2];
f@0:   pthread_cond_t runnable_cv;
f@0:   bool runnable;
f@0: 
f@0:   AlsaHandle()
f@0:     :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
f@0: };
f@0: 
f@0: static void *alsaCallbackHandler( void * ptr );
f@0: 
f@0: RtApiAlsa :: RtApiAlsa()
f@0: {
f@0:   // Nothing to do here.
f@0: }
f@0: 
f@0: RtApiAlsa :: ~RtApiAlsa()
f@0: {
f@0:   if ( stream_.state != STREAM_CLOSED ) closeStream();
f@0: }
f@0: 
f@0: unsigned int RtApiAlsa :: getDeviceCount( void )
f@0: {
f@0:   unsigned nDevices = 0;
f@0:   int result, subdevice, card;
f@0:   char name[64];
f@0:   snd_ctl_t *handle;
f@0: 
f@0:   // Count cards and devices
f@0:   card = -1;
f@0:   snd_card_next( &card );
f@0:   while ( card >= 0 ) {
f@0:     sprintf( name, "hw:%d", card );
f@0:     result = snd_ctl_open( &handle, name, 0 );
f@0:     if ( result < 0 ) {
f@0:       errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::WARNING );
f@0:       goto nextcard;
f@0:     }
f@0:     subdevice = -1;
f@0:     while( 1 ) {
f@0:       result = snd_ctl_pcm_next_device( handle, &subdevice );
f@0:       if ( result < 0 ) {
f@0:         errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
f@0:         errorText_ = errorStream_.str();
f@0:         error( RtAudioError::WARNING );
f@0:         break;
f@0:       }
f@0:       if ( subdevice < 0 )
f@0:         break;
f@0:       nDevices++;
f@0:     }
f@0:   nextcard:
f@0:     snd_ctl_close( handle );
f@0:     snd_card_next( &card );
f@0:   }
f@0: 
f@0:   result = snd_ctl_open( &handle, "default", 0 );
f@0:   if (result == 0) {
f@0:     nDevices++;
f@0:     snd_ctl_close( handle );
f@0:   }
f@0: 
f@0:   return nDevices;
f@0: }
f@0: 
f@0: RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   info.probed = false;
f@0: 
f@0:   unsigned nDevices = 0;
f@0:   int result, subdevice, card;
f@0:   char name[64];
f@0:   snd_ctl_t *chandle;
f@0: 
f@0:   // Count cards and devices
f@0:   card = -1;
f@0:   snd_card_next( &card );
f@0:   while ( card >= 0 ) {
f@0:     sprintf( name, "hw:%d", card );
f@0:     result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
f@0:     if ( result < 0 ) {
f@0:       errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::WARNING );
f@0:       goto nextcard;
f@0:     }
f@0:     subdevice = -1;
f@0:     while( 1 ) {
f@0:       result = snd_ctl_pcm_next_device( chandle, &subdevice );
f@0:       if ( result < 0 ) {
f@0:         errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
f@0:         errorText_ = errorStream_.str();
f@0:         error( RtAudioError::WARNING );
f@0:         break;
f@0:       }
f@0:       if ( subdevice < 0 ) break;
f@0:       if ( nDevices == device ) {
f@0:         sprintf( name, "hw:%d,%d", card, subdevice );
f@0:         goto foundDevice;
f@0:       }
f@0:       nDevices++;
f@0:     }
f@0:   nextcard:
f@0:     snd_ctl_close( chandle );
f@0:     snd_card_next( &card );
f@0:   }
f@0: 
f@0:   result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
f@0:   if ( result == 0 ) {
f@0:     if ( nDevices == device ) {
f@0:       strcpy( name, "default" );
f@0:       goto foundDevice;
f@0:     }
f@0:     nDevices++;
f@0:   }
f@0: 
f@0:   if ( nDevices == 0 ) {
f@0:     errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:  foundDevice:
f@0: 
f@0:   // If a stream is already open, we cannot probe the stream devices.
f@0:   // Thus, use the saved results.
f@0:   if ( stream_.state != STREAM_CLOSED &&
f@0:        ( stream_.device[0] == device || stream_.device[1] == device ) ) {
f@0:     snd_ctl_close( chandle );
f@0:     if ( device >= devices_.size() ) {
f@0:       errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
f@0:       error( RtAudioError::WARNING );
f@0:       return info;
f@0:     }
f@0:     return devices_[ device ];
f@0:   }
f@0: 
f@0:   int openMode = SND_PCM_ASYNC;
f@0:   snd_pcm_stream_t stream;
f@0:   snd_pcm_info_t *pcminfo;
f@0:   snd_pcm_info_alloca( &pcminfo );
f@0:   snd_pcm_t *phandle;
f@0:   snd_pcm_hw_params_t *params;
f@0:   snd_pcm_hw_params_alloca( &params );
f@0: 
f@0:   // First try for playback unless default device (which has subdev -1)
f@0:   stream = SND_PCM_STREAM_PLAYBACK;
f@0:   snd_pcm_info_set_stream( pcminfo, stream );
f@0:   if ( subdevice != -1 ) {
f@0:     snd_pcm_info_set_device( pcminfo, subdevice );
f@0:     snd_pcm_info_set_subdevice( pcminfo, 0 );
f@0: 
f@0:     result = snd_ctl_pcm_info( chandle, pcminfo );
f@0:     if ( result < 0 ) {
f@0:       // Device probably doesn't support playback.
f@0:       goto captureProbe;
f@0:     }
f@0:   }
f@0: 
f@0:   result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
f@0:   if ( result < 0 ) {
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     goto captureProbe;
f@0:   }
f@0: 
f@0:   // The device is open ... fill the parameter structure.
f@0:   result = snd_pcm_hw_params_any( phandle, params );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     goto captureProbe;
f@0:   }
f@0: 
f@0:   // Get output channel information.
f@0:   unsigned int value;
f@0:   result = snd_pcm_hw_params_get_channels_max( params, &value );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     goto captureProbe;
f@0:   }
f@0:   info.outputChannels = value;
f@0:   snd_pcm_close( phandle );
f@0: 
f@0:  captureProbe:
f@0:   stream = SND_PCM_STREAM_CAPTURE;
f@0:   snd_pcm_info_set_stream( pcminfo, stream );
f@0: 
f@0:   // Now try for capture unless default device (with subdev = -1)
f@0:   if ( subdevice != -1 ) {
f@0:     result = snd_ctl_pcm_info( chandle, pcminfo );
f@0:     snd_ctl_close( chandle );
f@0:     if ( result < 0 ) {
f@0:       // Device probably doesn't support capture.
f@0:       if ( info.outputChannels == 0 ) return info;
f@0:       goto probeParameters;
f@0:     }
f@0:   }
f@0:   else
f@0:     snd_ctl_close( chandle );
f@0: 
f@0:   result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
f@0:   if ( result < 0 ) {
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     if ( info.outputChannels == 0 ) return info;
f@0:     goto probeParameters;
f@0:   }
f@0: 
f@0:   // The device is open ... fill the parameter structure.
f@0:   result = snd_pcm_hw_params_any( phandle, params );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     if ( info.outputChannels == 0 ) return info;
f@0:     goto probeParameters;
f@0:   }
f@0: 
f@0:   result = snd_pcm_hw_params_get_channels_max( params, &value );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     if ( info.outputChannels == 0 ) return info;
f@0:     goto probeParameters;
f@0:   }
f@0:   info.inputChannels = value;
f@0:   snd_pcm_close( phandle );
f@0: 
f@0:   // If device opens for both playback and capture, we determine the channels.
f@0:   if ( info.outputChannels > 0 && info.inputChannels > 0 )
f@0:     info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
f@0: 
f@0:   // ALSA doesn't provide default devices so we'll use the first available one.
f@0:   if ( device == 0 && info.outputChannels > 0 )
f@0:     info.isDefaultOutput = true;
f@0:   if ( device == 0 && info.inputChannels > 0 )
f@0:     info.isDefaultInput = true;
f@0: 
f@0:  probeParameters:
f@0:   // At this point, we just need to figure out the supported data
f@0:   // formats and sample rates.  We'll proceed by opening the device in
f@0:   // the direction with the maximum number of channels, or playback if
f@0:   // they are equal.  This might limit our sample rate options, but so
f@0:   // be it.
f@0: 
f@0:   if ( info.outputChannels >= info.inputChannels )
f@0:     stream = SND_PCM_STREAM_PLAYBACK;
f@0:   else
f@0:     stream = SND_PCM_STREAM_CAPTURE;
f@0:   snd_pcm_info_set_stream( pcminfo, stream );
f@0: 
f@0:   result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
f@0:   if ( result < 0 ) {
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // The device is open ... fill the parameter structure.
f@0:   result = snd_pcm_hw_params_any( phandle, params );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Test our discrete set of sample rate values.
f@0:   info.sampleRates.clear();
f@0:   for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
f@0:     if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
f@0:       info.sampleRates.push_back( SAMPLE_RATES[i] );
f@0:   }
f@0:   if ( info.sampleRates.size() == 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Probe the supported data formats ... we don't care about endian-ness just yet
f@0:   snd_pcm_format_t format;
f@0:   info.nativeFormats = 0;
f@0:   format = SND_PCM_FORMAT_S8;
f@0:   if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
f@0:     info.nativeFormats |= RTAUDIO_SINT8;
f@0:   format = SND_PCM_FORMAT_S16;
f@0:   if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
f@0:     info.nativeFormats |= RTAUDIO_SINT16;
f@0:   format = SND_PCM_FORMAT_S24;
f@0:   if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
f@0:     info.nativeFormats |= RTAUDIO_SINT24;
f@0:   format = SND_PCM_FORMAT_S32;
f@0:   if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
f@0:     info.nativeFormats |= RTAUDIO_SINT32;
f@0:   format = SND_PCM_FORMAT_FLOAT;
f@0:   if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
f@0:     info.nativeFormats |= RTAUDIO_FLOAT32;
f@0:   format = SND_PCM_FORMAT_FLOAT64;
f@0:   if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
f@0:     info.nativeFormats |= RTAUDIO_FLOAT64;
f@0: 
f@0:   // Check that we have at least one supported format
f@0:   if ( info.nativeFormats == 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Get the device name
f@0:   char *cardname;
f@0:   result = snd_card_get_name( card, &cardname );
f@0:   if ( result >= 0 ) {
f@0:     sprintf( name, "hw:%s,%d", cardname, subdevice );
f@0:     free( cardname );
f@0:   }
f@0:   info.name = name;
f@0: 
f@0:   // That's all ... close the device and return
f@0:   snd_pcm_close( phandle );
f@0:   info.probed = true;
f@0:   return info;
f@0: }
f@0: 
f@0: void RtApiAlsa :: saveDeviceInfo( void )
f@0: {
f@0:   devices_.clear();
f@0: 
f@0:   unsigned int nDevices = getDeviceCount();
f@0:   devices_.resize( nDevices );
f@0:   for ( unsigned int i=0; i<nDevices; i++ )
f@0:     devices_[i] = getDeviceInfo( i );
f@0: }
f@0: 
f@0: bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
f@0:                                    unsigned int firstChannel, unsigned int sampleRate,
f@0:                                    RtAudioFormat format, unsigned int *bufferSize,
f@0:                                    RtAudio::StreamOptions *options )
f@0: 
f@0: {
f@0: #if defined(__RTAUDIO_DEBUG__)
f@0:   snd_output_t *out;
f@0:   snd_output_stdio_attach(&out, stderr, 0);
f@0: #endif
f@0: 
f@0:   // I'm not using the "plug" interface ... too much inconsistent behavior.
f@0: 
f@0:   unsigned nDevices = 0;
f@0:   int result, subdevice, card;
f@0:   char name[64];
f@0:   snd_ctl_t *chandle;
f@0: 
f@0:   if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
f@0:     snprintf(name, sizeof(name), "%s", "default");
f@0:   else {
f@0:     // Count cards and devices
f@0:     card = -1;
f@0:     snd_card_next( &card );
f@0:     while ( card >= 0 ) {
f@0:       sprintf( name, "hw:%d", card );
f@0:       result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
f@0:       if ( result < 0 ) {
f@0:         errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
f@0:         errorText_ = errorStream_.str();
f@0:         return FAILURE;
f@0:       }
f@0:       subdevice = -1;
f@0:       while( 1 ) {
f@0:         result = snd_ctl_pcm_next_device( chandle, &subdevice );
f@0:         if ( result < 0 ) break;
f@0:         if ( subdevice < 0 ) break;
f@0:         if ( nDevices == device ) {
f@0:           sprintf( name, "hw:%d,%d", card, subdevice );
f@0:           snd_ctl_close( chandle );
f@0:           goto foundDevice;
f@0:         }
f@0:         nDevices++;
f@0:       }
f@0:       snd_ctl_close( chandle );
f@0:       snd_card_next( &card );
f@0:     }
f@0: 
f@0:     result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
f@0:     if ( result == 0 ) {
f@0:       if ( nDevices == device ) {
f@0:         strcpy( name, "default" );
f@0:         goto foundDevice;
f@0:       }
f@0:       nDevices++;
f@0:     }
f@0: 
f@0:     if ( nDevices == 0 ) {
f@0:       // This should not happen because a check is made before this function is called.
f@0:       errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
f@0:       return FAILURE;
f@0:     }
f@0: 
f@0:     if ( device >= nDevices ) {
f@0:       // This should not happen because a check is made before this function is called.
f@0:       errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0: 
f@0:  foundDevice:
f@0: 
f@0:   // The getDeviceInfo() function will not work for a device that is
f@0:   // already open.  Thus, we'll probe the system before opening a
f@0:   // stream and save the results for use by getDeviceInfo().
f@0:   if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
f@0:     this->saveDeviceInfo();
f@0: 
f@0:   snd_pcm_stream_t stream;
f@0:   if ( mode == OUTPUT )
f@0:     stream = SND_PCM_STREAM_PLAYBACK;
f@0:   else
f@0:     stream = SND_PCM_STREAM_CAPTURE;
f@0: 
f@0:   snd_pcm_t *phandle;
f@0:   int openMode = SND_PCM_ASYNC;
f@0:   result = snd_pcm_open( &phandle, name, stream, openMode );
f@0:   if ( result < 0 ) {
f@0:     if ( mode == OUTPUT )
f@0:       errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
f@0:     else
f@0:       errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Fill the parameter structure.
f@0:   snd_pcm_hw_params_t *hw_params;
f@0:   snd_pcm_hw_params_alloca( &hw_params );
f@0:   result = snd_pcm_hw_params_any( phandle, hw_params );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0: #if defined(__RTAUDIO_DEBUG__)
f@0:   fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
f@0:   snd_pcm_hw_params_dump( hw_params, out );
f@0: #endif
f@0: 
f@0:   // Set access ... check user preference.
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
f@0:     stream_.userInterleaved = false;
f@0:     result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
f@0:     if ( result < 0 ) {
f@0:       result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
f@0:       stream_.deviceInterleaved[mode] =  true;
f@0:     }
f@0:     else
f@0:       stream_.deviceInterleaved[mode] = false;
f@0:   }
f@0:   else {
f@0:     stream_.userInterleaved = true;
f@0:     result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
f@0:     if ( result < 0 ) {
f@0:       result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
f@0:       stream_.deviceInterleaved[mode] =  false;
f@0:     }
f@0:     else
f@0:       stream_.deviceInterleaved[mode] =  true;
f@0:   }
f@0: 
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Determine how to set the device format.
f@0:   stream_.userFormat = format;
f@0:   snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
f@0: 
f@0:   if ( format == RTAUDIO_SINT8 )
f@0:     deviceFormat = SND_PCM_FORMAT_S8;
f@0:   else if ( format == RTAUDIO_SINT16 )
f@0:     deviceFormat = SND_PCM_FORMAT_S16;
f@0:   else if ( format == RTAUDIO_SINT24 )
f@0:     deviceFormat = SND_PCM_FORMAT_S24;
f@0:   else if ( format == RTAUDIO_SINT32 )
f@0:     deviceFormat = SND_PCM_FORMAT_S32;
f@0:   else if ( format == RTAUDIO_FLOAT32 )
f@0:     deviceFormat = SND_PCM_FORMAT_FLOAT;
f@0:   else if ( format == RTAUDIO_FLOAT64 )
f@0:     deviceFormat = SND_PCM_FORMAT_FLOAT64;
f@0: 
f@0:   if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
f@0:     stream_.deviceFormat[mode] = format;
f@0:     goto setFormat;
f@0:   }
f@0: 
f@0:   // The user requested format is not natively supported by the device.
f@0:   deviceFormat = SND_PCM_FORMAT_FLOAT64;
f@0:   if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
f@0:     goto setFormat;
f@0:   }
f@0: 
f@0:   deviceFormat = SND_PCM_FORMAT_FLOAT;
f@0:   if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
f@0:     goto setFormat;
f@0:   }
f@0: 
f@0:   deviceFormat = SND_PCM_FORMAT_S32;
f@0:   if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_SINT32;
f@0:     goto setFormat;
f@0:   }
f@0: 
f@0:   deviceFormat = SND_PCM_FORMAT_S24;
f@0:   if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_SINT24;
f@0:     goto setFormat;
f@0:   }
f@0: 
f@0:   deviceFormat = SND_PCM_FORMAT_S16;
f@0:   if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:     goto setFormat;
f@0:   }
f@0: 
f@0:   deviceFormat = SND_PCM_FORMAT_S8;
f@0:   if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
f@0:     stream_.deviceFormat[mode] = RTAUDIO_SINT8;
f@0:     goto setFormat;
f@0:   }
f@0: 
f@0:   // If we get here, no supported format was found.
f@0:   snd_pcm_close( phandle );
f@0:   errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
f@0:   errorText_ = errorStream_.str();
f@0:   return FAILURE;
f@0: 
f@0:  setFormat:
f@0:   result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Determine whether byte-swaping is necessary.
f@0:   stream_.doByteSwap[mode] = false;
f@0:   if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
f@0:     result = snd_pcm_format_cpu_endian( deviceFormat );
f@0:     if ( result == 0 )
f@0:       stream_.doByteSwap[mode] = true;
f@0:     else if (result < 0) {
f@0:       snd_pcm_close( phandle );
f@0:       errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       return FAILURE;
f@0:     }
f@0:   }
f@0: 
f@0:   // Set the sample rate.
f@0:   result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Determine the number of channels for this device.  We support a possible
f@0:   // minimum device channel number > than the value requested by the user.
f@0:   stream_.nUserChannels[mode] = channels;
f@0:   unsigned int value;
f@0:   result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
f@0:   unsigned int deviceChannels = value;
f@0:   if ( result < 0 || deviceChannels < channels + firstChannel ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0:   deviceChannels = value;
f@0:   if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
f@0:   stream_.nDeviceChannels[mode] = deviceChannels;
f@0: 
f@0:   // Set the device channels.
f@0:   result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Set the buffer (or period) size.
f@0:   int dir = 0;
f@0:   snd_pcm_uframes_t periodSize = *bufferSize;
f@0:   result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0:   *bufferSize = periodSize;
f@0: 
f@0:   // Set the buffer number, which in ALSA is referred to as the "period".
f@0:   unsigned int periods = 0;
f@0:   if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
f@0:   if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
f@0:   if ( periods < 2 ) periods = 4; // a fairly safe default value
f@0:   result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // If attempting to setup a duplex stream, the bufferSize parameter
f@0:   // MUST be the same in both directions!
f@0:   if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   stream_.bufferSize = *bufferSize;
f@0: 
f@0:   // Install the hardware configuration
f@0:   result = snd_pcm_hw_params( phandle, hw_params );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0: #if defined(__RTAUDIO_DEBUG__)
f@0:   fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
f@0:   snd_pcm_hw_params_dump( hw_params, out );
f@0: #endif
f@0: 
f@0:   // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
f@0:   snd_pcm_sw_params_t *sw_params = NULL;
f@0:   snd_pcm_sw_params_alloca( &sw_params );
f@0:   snd_pcm_sw_params_current( phandle, sw_params );
f@0:   snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
f@0:   snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
f@0:   snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
f@0: 
f@0:   // The following two settings were suggested by Theo Veenker
f@0:   //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
f@0:   //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
f@0: 
f@0:   // here are two options for a fix
f@0:   //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
f@0:   snd_pcm_uframes_t val;
f@0:   snd_pcm_sw_params_get_boundary( sw_params, &val );
f@0:   snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
f@0: 
f@0:   result = snd_pcm_sw_params( phandle, sw_params );
f@0:   if ( result < 0 ) {
f@0:     snd_pcm_close( phandle );
f@0:     errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0: #if defined(__RTAUDIO_DEBUG__)
f@0:   fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
f@0:   snd_pcm_sw_params_dump( sw_params, out );
f@0: #endif
f@0: 
f@0:   // Set flags for buffer conversion
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if ( stream_.userFormat != stream_.deviceFormat[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
f@0:        stream_.nUserChannels[mode] > 1 )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   // Allocate the ApiHandle if necessary and then save.
f@0:   AlsaHandle *apiInfo = 0;
f@0:   if ( stream_.apiHandle == 0 ) {
f@0:     try {
f@0:       apiInfo = (AlsaHandle *) new AlsaHandle;
f@0:     }
f@0:     catch ( std::bad_alloc& ) {
f@0:       errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
f@0:       errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     stream_.apiHandle = (void *) apiInfo;
f@0:     apiInfo->handles[0] = 0;
f@0:     apiInfo->handles[1] = 0;
f@0:   }
f@0:   else {
f@0:     apiInfo = (AlsaHandle *) stream_.apiHandle;
f@0:   }
f@0:   apiInfo->handles[mode] = phandle;
f@0:   phandle = 0;
f@0: 
f@0:   // Allocate necessary internal buffers.
f@0:   unsigned long bufferBytes;
f@0:   bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
f@0:   stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
f@0:   if ( stream_.userBuffer[mode] == NULL ) {
f@0:     errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
f@0:     goto error;
f@0:   }
f@0: 
f@0:   if ( stream_.doConvertBuffer[mode] ) {
f@0: 
f@0:     bool makeBuffer = true;
f@0:     bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
f@0:     if ( mode == INPUT ) {
f@0:       if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
f@0:         unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
f@0:         if ( bufferBytes <= bytesOut ) makeBuffer = false;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( makeBuffer ) {
f@0:       bufferBytes *= *bufferSize;
f@0:       if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
f@0:       stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
f@0:       if ( stream_.deviceBuffer == NULL ) {
f@0:         errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
f@0:         goto error;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.sampleRate = sampleRate;
f@0:   stream_.nBuffers = periods;
f@0:   stream_.device[mode] = device;
f@0:   stream_.state = STREAM_STOPPED;
f@0: 
f@0:   // Setup the buffer conversion information structure.
f@0:   if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
f@0: 
f@0:   // Setup thread if necessary.
f@0:   if ( stream_.mode == OUTPUT && mode == INPUT ) {
f@0:     // We had already set up an output stream.
f@0:     stream_.mode = DUPLEX;
f@0:     // Link the streams if possible.
f@0:     apiInfo->synchronized = false;
f@0:     if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
f@0:       apiInfo->synchronized = true;
f@0:     else {
f@0:       errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
f@0:       error( RtAudioError::WARNING );
f@0:     }
f@0:   }
f@0:   else {
f@0:     stream_.mode = mode;
f@0: 
f@0:     // Setup callback thread.
f@0:     stream_.callbackInfo.object = (void *) this;
f@0: 
f@0:     // Set the thread attributes for joinable and realtime scheduling
f@0:     // priority (optional).  The higher priority will only take affect
f@0:     // if the program is run as root or suid. Note, under Linux
f@0:     // processes with CAP_SYS_NICE privilege, a user can change
f@0:     // scheduling policy and priority (thus need not be root). See
f@0:     // POSIX "capabilities".
f@0:     pthread_attr_t attr;
f@0:     pthread_attr_init( &attr );
f@0:     pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
f@0: 
f@0: #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
f@0:     if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
f@0:       // We previously attempted to increase the audio callback priority
f@0:       // to SCHED_RR here via the attributes.  However, while no errors
f@0:       // were reported in doing so, it did not work.  So, now this is
f@0:       // done in the alsaCallbackHandler function.
f@0:       stream_.callbackInfo.doRealtime = true;
f@0:       int priority = options->priority;
f@0:       int min = sched_get_priority_min( SCHED_RR );
f@0:       int max = sched_get_priority_max( SCHED_RR );
f@0:       if ( priority < min ) priority = min;
f@0:       else if ( priority > max ) priority = max;
f@0:       stream_.callbackInfo.priority = priority;
f@0:     }
f@0: #endif
f@0: 
f@0:     stream_.callbackInfo.isRunning = true;
f@0:     result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
f@0:     pthread_attr_destroy( &attr );
f@0:     if ( result ) {
f@0:       stream_.callbackInfo.isRunning = false;
f@0:       errorText_ = "RtApiAlsa::error creating callback thread!";
f@0:       goto error;
f@0:     }
f@0:   }
f@0: 
f@0:   return SUCCESS;
f@0: 
f@0:  error:
f@0:   if ( apiInfo ) {
f@0:     pthread_cond_destroy( &apiInfo->runnable_cv );
f@0:     if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
f@0:     if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
f@0:     delete apiInfo;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   if ( phandle) snd_pcm_close( phandle );
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_CLOSED;
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: void RtApiAlsa :: closeStream()
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
f@0:   stream_.callbackInfo.isRunning = false;
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     apiInfo->runnable = true;
f@0:     pthread_cond_signal( &apiInfo->runnable_cv );
f@0:   }
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0:   pthread_join( stream_.callbackInfo.thread, NULL );
f@0: 
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     stream_.state = STREAM_STOPPED;
f@0:     if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
f@0:       snd_pcm_drop( apiInfo->handles[0] );
f@0:     if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
f@0:       snd_pcm_drop( apiInfo->handles[1] );
f@0:   }
f@0: 
f@0:   if ( apiInfo ) {
f@0:     pthread_cond_destroy( &apiInfo->runnable_cv );
f@0:     if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
f@0:     if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
f@0:     delete apiInfo;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.state = STREAM_CLOSED;
f@0: }
f@0: 
f@0: void RtApiAlsa :: startStream()
f@0: {
f@0:   // This method calls snd_pcm_prepare if the device isn't already in that state.
f@0: 
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   int result = 0;
f@0:   snd_pcm_state_t state;
f@0:   AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
f@0:   snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     state = snd_pcm_state( handle[0] );
f@0:     if ( state != SND_PCM_STATE_PREPARED ) {
f@0:       result = snd_pcm_prepare( handle[0] );
f@0:       if ( result < 0 ) {
f@0:         errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
f@0:         errorText_ = errorStream_.str();
f@0:         goto unlock;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
f@0:     result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
f@0:     state = snd_pcm_state( handle[1] );
f@0:     if ( state != SND_PCM_STATE_PREPARED ) {
f@0:       result = snd_pcm_prepare( handle[1] );
f@0:       if ( result < 0 ) {
f@0:         errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
f@0:         errorText_ = errorStream_.str();
f@0:         goto unlock;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_RUNNING;
f@0: 
f@0:  unlock:
f@0:   apiInfo->runnable = true;
f@0:   pthread_cond_signal( &apiInfo->runnable_cv );
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   if ( result >= 0 ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiAlsa :: stopStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   int result = 0;
f@0:   AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
f@0:   snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     if ( apiInfo->synchronized ) 
f@0:       result = snd_pcm_drop( handle[0] );
f@0:     else
f@0:       result = snd_pcm_drain( handle[0] );
f@0:     if ( result < 0 ) {
f@0:       errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
f@0:     result = snd_pcm_drop( handle[1] );
f@0:     if ( result < 0 ) {
f@0:       errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   apiInfo->runnable = false; // fixes high CPU usage when stopped
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   if ( result >= 0 ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiAlsa :: abortStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   int result = 0;
f@0:   AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
f@0:   snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     result = snd_pcm_drop( handle[0] );
f@0:     if ( result < 0 ) {
f@0:       errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
f@0:     result = snd_pcm_drop( handle[1] );
f@0:     if ( result < 0 ) {
f@0:       errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   apiInfo->runnable = false; // fixes high CPU usage when stopped
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   if ( result >= 0 ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiAlsa :: callbackEvent()
f@0: {
f@0:   AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     MUTEX_LOCK( &stream_.mutex );
f@0:     while ( !apiInfo->runnable )
f@0:       pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
f@0: 
f@0:     if ( stream_.state != STREAM_RUNNING ) {
f@0:       MUTEX_UNLOCK( &stream_.mutex );
f@0:       return;
f@0:     }
f@0:     MUTEX_UNLOCK( &stream_.mutex );
f@0:   }
f@0: 
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   int doStopStream = 0;
f@0:   RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
f@0:   double streamTime = getStreamTime();
f@0:   RtAudioStreamStatus status = 0;
f@0:   if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
f@0:     status |= RTAUDIO_OUTPUT_UNDERFLOW;
f@0:     apiInfo->xrun[0] = false;
f@0:   }
f@0:   if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
f@0:     status |= RTAUDIO_INPUT_OVERFLOW;
f@0:     apiInfo->xrun[1] = false;
f@0:   }
f@0:   doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
f@0:                            stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
f@0: 
f@0:   if ( doStopStream == 2 ) {
f@0:     abortStream();
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   // The state might change while waiting on a mutex.
f@0:   if ( stream_.state == STREAM_STOPPED ) goto unlock;
f@0: 
f@0:   int result;
f@0:   char *buffer;
f@0:   int channels;
f@0:   snd_pcm_t **handle;
f@0:   snd_pcm_sframes_t frames;
f@0:   RtAudioFormat format;
f@0:   handle = (snd_pcm_t **) apiInfo->handles;
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     // Setup parameters.
f@0:     if ( stream_.doConvertBuffer[1] ) {
f@0:       buffer = stream_.deviceBuffer;
f@0:       channels = stream_.nDeviceChannels[1];
f@0:       format = stream_.deviceFormat[1];
f@0:     }
f@0:     else {
f@0:       buffer = stream_.userBuffer[1];
f@0:       channels = stream_.nUserChannels[1];
f@0:       format = stream_.userFormat;
f@0:     }
f@0: 
f@0:     // Read samples from device in interleaved/non-interleaved format.
f@0:     if ( stream_.deviceInterleaved[1] )
f@0:       result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
f@0:     else {
f@0:       void *bufs[channels];
f@0:       size_t offset = stream_.bufferSize * formatBytes( format );
f@0:       for ( int i=0; i<channels; i++ )
f@0:         bufs[i] = (void *) (buffer + (i * offset));
f@0:       result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
f@0:     }
f@0: 
f@0:     if ( result < (int) stream_.bufferSize ) {
f@0:       // Either an error or overrun occured.
f@0:       if ( result == -EPIPE ) {
f@0:         snd_pcm_state_t state = snd_pcm_state( handle[1] );
f@0:         if ( state == SND_PCM_STATE_XRUN ) {
f@0:           apiInfo->xrun[1] = true;
f@0:           result = snd_pcm_prepare( handle[1] );
f@0:           if ( result < 0 ) {
f@0:             errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
f@0:             errorText_ = errorStream_.str();
f@0:           }
f@0:         }
f@0:         else {
f@0:           errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
f@0:           errorText_ = errorStream_.str();
f@0:         }
f@0:       }
f@0:       else {
f@0:         errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
f@0:         errorText_ = errorStream_.str();
f@0:       }
f@0:       error( RtAudioError::WARNING );
f@0:       goto tryOutput;
f@0:     }
f@0: 
f@0:     // Do byte swapping if necessary.
f@0:     if ( stream_.doByteSwap[1] )
f@0:       byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
f@0: 
f@0:     // Do buffer conversion if necessary.
f@0:     if ( stream_.doConvertBuffer[1] )
f@0:       convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
f@0: 
f@0:     // Check stream latency
f@0:     result = snd_pcm_delay( handle[1], &frames );
f@0:     if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
f@0:   }
f@0: 
f@0:  tryOutput:
f@0: 
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     // Setup parameters and do buffer conversion if necessary.
f@0:     if ( stream_.doConvertBuffer[0] ) {
f@0:       buffer = stream_.deviceBuffer;
f@0:       convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
f@0:       channels = stream_.nDeviceChannels[0];
f@0:       format = stream_.deviceFormat[0];
f@0:     }
f@0:     else {
f@0:       buffer = stream_.userBuffer[0];
f@0:       channels = stream_.nUserChannels[0];
f@0:       format = stream_.userFormat;
f@0:     }
f@0: 
f@0:     // Do byte swapping if necessary.
f@0:     if ( stream_.doByteSwap[0] )
f@0:       byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
f@0: 
f@0:     // Write samples to device in interleaved/non-interleaved format.
f@0:     if ( stream_.deviceInterleaved[0] )
f@0:       result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
f@0:     else {
f@0:       void *bufs[channels];
f@0:       size_t offset = stream_.bufferSize * formatBytes( format );
f@0:       for ( int i=0; i<channels; i++ )
f@0:         bufs[i] = (void *) (buffer + (i * offset));
f@0:       result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
f@0:     }
f@0: 
f@0:     if ( result < (int) stream_.bufferSize ) {
f@0:       // Either an error or underrun occured.
f@0:       if ( result == -EPIPE ) {
f@0:         snd_pcm_state_t state = snd_pcm_state( handle[0] );
f@0:         if ( state == SND_PCM_STATE_XRUN ) {
f@0:           apiInfo->xrun[0] = true;
f@0:           result = snd_pcm_prepare( handle[0] );
f@0:           if ( result < 0 ) {
f@0:             errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
f@0:             errorText_ = errorStream_.str();
f@0:           }
f@0:         }
f@0:         else {
f@0:           errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
f@0:           errorText_ = errorStream_.str();
f@0:         }
f@0:       }
f@0:       else {
f@0:         errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
f@0:         errorText_ = errorStream_.str();
f@0:       }
f@0:       error( RtAudioError::WARNING );
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Check stream latency
f@0:     result = snd_pcm_delay( handle[0], &frames );
f@0:     if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
f@0:   }
f@0: 
f@0:  unlock:
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   RtApi::tickStreamTime();
f@0:   if ( doStopStream == 1 ) this->stopStream();
f@0: }
f@0: 
f@0: static void *alsaCallbackHandler( void *ptr )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) ptr;
f@0:   RtApiAlsa *object = (RtApiAlsa *) info->object;
f@0:   bool *isRunning = &info->isRunning;
f@0: 
f@0: #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
f@0:   if ( &info->doRealtime ) {
f@0:     pthread_t tID = pthread_self();	 // ID of this thread
f@0:     sched_param prio = { info->priority }; // scheduling priority of thread
f@0:     pthread_setschedparam( tID, SCHED_RR, &prio );
f@0:   }
f@0: #endif
f@0: 
f@0:   while ( *isRunning == true ) {
f@0:     pthread_testcancel();
f@0:     object->callbackEvent();
f@0:   }
f@0: 
f@0:   pthread_exit( NULL );
f@0: }
f@0: 
f@0: //******************** End of __LINUX_ALSA__ *********************//
f@0: #endif
f@0: 
f@0: #if defined(__LINUX_PULSE__)
f@0: 
f@0: // Code written by Peter Meerwald, pmeerw@pmeerw.net
f@0: // and Tristan Matthews.
f@0: 
f@0: #include <pulse/error.h>
f@0: #include <pulse/simple.h>
f@0: #include <cstdio>
f@0: 
f@0: static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
f@0:                                                       44100, 48000, 96000, 0};
f@0: 
f@0: struct rtaudio_pa_format_mapping_t {
f@0:   RtAudioFormat rtaudio_format;
f@0:   pa_sample_format_t pa_format;
f@0: };
f@0: 
f@0: static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
f@0:   {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
f@0:   {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
f@0:   {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
f@0:   {0, PA_SAMPLE_INVALID}};
f@0: 
f@0: struct PulseAudioHandle {
f@0:   pa_simple *s_play;
f@0:   pa_simple *s_rec;
f@0:   pthread_t thread;
f@0:   pthread_cond_t runnable_cv;
f@0:   bool runnable;
f@0:   PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
f@0: };
f@0: 
f@0: RtApiPulse::~RtApiPulse()
f@0: {
f@0:   if ( stream_.state != STREAM_CLOSED )
f@0:     closeStream();
f@0: }
f@0: 
f@0: unsigned int RtApiPulse::getDeviceCount( void )
f@0: {
f@0:   return 1;
f@0: }
f@0: 
f@0: RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   info.probed = true;
f@0:   info.name = "PulseAudio";
f@0:   info.outputChannels = 2;
f@0:   info.inputChannels = 2;
f@0:   info.duplexChannels = 2;
f@0:   info.isDefaultOutput = true;
f@0:   info.isDefaultInput = true;
f@0: 
f@0:   for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
f@0:     info.sampleRates.push_back( *sr );
f@0: 
f@0:   info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
f@0: 
f@0:   return info;
f@0: }
f@0: 
f@0: static void *pulseaudio_callback( void * user )
f@0: {
f@0:   CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
f@0:   RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
f@0:   volatile bool *isRunning = &cbi->isRunning;
f@0: 
f@0:   while ( *isRunning ) {
f@0:     pthread_testcancel();
f@0:     context->callbackEvent();
f@0:   }
f@0: 
f@0:   pthread_exit( NULL );
f@0: }
f@0: 
f@0: void RtApiPulse::closeStream( void )
f@0: {
f@0:   PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
f@0: 
f@0:   stream_.callbackInfo.isRunning = false;
f@0:   if ( pah ) {
f@0:     MUTEX_LOCK( &stream_.mutex );
f@0:     if ( stream_.state == STREAM_STOPPED ) {
f@0:       pah->runnable = true;
f@0:       pthread_cond_signal( &pah->runnable_cv );
f@0:     }
f@0:     MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:     pthread_join( pah->thread, 0 );
f@0:     if ( pah->s_play ) {
f@0:       pa_simple_flush( pah->s_play, NULL );
f@0:       pa_simple_free( pah->s_play );
f@0:     }
f@0:     if ( pah->s_rec )
f@0:       pa_simple_free( pah->s_rec );
f@0: 
f@0:     pthread_cond_destroy( &pah->runnable_cv );
f@0:     delete pah;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   if ( stream_.userBuffer[0] ) {
f@0:     free( stream_.userBuffer[0] );
f@0:     stream_.userBuffer[0] = 0;
f@0:   }
f@0:   if ( stream_.userBuffer[1] ) {
f@0:     free( stream_.userBuffer[1] );
f@0:     stream_.userBuffer[1] = 0;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_CLOSED;
f@0:   stream_.mode = UNINITIALIZED;
f@0: }
f@0: 
f@0: void RtApiPulse::callbackEvent( void )
f@0: {
f@0:   PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
f@0: 
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     MUTEX_LOCK( &stream_.mutex );
f@0:     while ( !pah->runnable )
f@0:       pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
f@0: 
f@0:     if ( stream_.state != STREAM_RUNNING ) {
f@0:       MUTEX_UNLOCK( &stream_.mutex );
f@0:       return;
f@0:     }
f@0:     MUTEX_UNLOCK( &stream_.mutex );
f@0:   }
f@0: 
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
f@0:       "this shouldn't happen!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
f@0:   double streamTime = getStreamTime();
f@0:   RtAudioStreamStatus status = 0;
f@0:   int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
f@0:                                stream_.bufferSize, streamTime, status,
f@0:                                stream_.callbackInfo.userData );
f@0: 
f@0:   if ( doStopStream == 2 ) {
f@0:     abortStream();
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0:   void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
f@0:   void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
f@0: 
f@0:   if ( stream_.state != STREAM_RUNNING )
f@0:     goto unlock;
f@0: 
f@0:   int pa_error;
f@0:   size_t bytes;
f@0:   if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     if ( stream_.doConvertBuffer[OUTPUT] ) {
f@0:         convertBuffer( stream_.deviceBuffer,
f@0:                        stream_.userBuffer[OUTPUT],
f@0:                        stream_.convertInfo[OUTPUT] );
f@0:         bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
f@0:                 formatBytes( stream_.deviceFormat[OUTPUT] );
f@0:     } else
f@0:         bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
f@0:                 formatBytes( stream_.userFormat );
f@0: 
f@0:     if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
f@0:       errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
f@0:         pa_strerror( pa_error ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::WARNING );
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
f@0:     if ( stream_.doConvertBuffer[INPUT] )
f@0:       bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
f@0:         formatBytes( stream_.deviceFormat[INPUT] );
f@0:     else
f@0:       bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
f@0:         formatBytes( stream_.userFormat );
f@0:             
f@0:     if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
f@0:       errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
f@0:         pa_strerror( pa_error ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       error( RtAudioError::WARNING );
f@0:     }
f@0:     if ( stream_.doConvertBuffer[INPUT] ) {
f@0:       convertBuffer( stream_.userBuffer[INPUT],
f@0:                      stream_.deviceBuffer,
f@0:                      stream_.convertInfo[INPUT] );
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0:   RtApi::tickStreamTime();
f@0: 
f@0:   if ( doStopStream == 1 )
f@0:     stopStream();
f@0: }
f@0: 
f@0: void RtApiPulse::startStream( void )
f@0: {
f@0:   PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
f@0: 
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiPulse::startStream(): the stream is not open!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiPulse::startStream(): the stream is already running!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   stream_.state = STREAM_RUNNING;
f@0: 
f@0:   pah->runnable = true;
f@0:   pthread_cond_signal( &pah->runnable_cv );
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: }
f@0: 
f@0: void RtApiPulse::stopStream( void )
f@0: {
f@0:   PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
f@0: 
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   if ( pah && pah->s_play ) {
f@0:     int pa_error;
f@0:     if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
f@0:       errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
f@0:         pa_strerror( pa_error ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       MUTEX_UNLOCK( &stream_.mutex );
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: }
f@0: 
f@0: void RtApiPulse::abortStream( void )
f@0: {
f@0:   PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
f@0: 
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return;
f@0:   }
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   if ( pah && pah->s_play ) {
f@0:     int pa_error;
f@0:     if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
f@0:       errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
f@0:         pa_strerror( pa_error ) << ".";
f@0:       errorText_ = errorStream_.str();
f@0:       MUTEX_UNLOCK( &stream_.mutex );
f@0:       error( RtAudioError::SYSTEM_ERROR );
f@0:       return;
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: }
f@0: 
f@0: bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
f@0:                                   unsigned int channels, unsigned int firstChannel,
f@0:                                   unsigned int sampleRate, RtAudioFormat format,
f@0:                                   unsigned int *bufferSize, RtAudio::StreamOptions *options )
f@0: {
f@0:   PulseAudioHandle *pah = 0;
f@0:   unsigned long bufferBytes = 0;
f@0:   pa_sample_spec ss;
f@0: 
f@0:   if ( device != 0 ) return false;
f@0:   if ( mode != INPUT && mode != OUTPUT ) return false;
f@0:   if ( channels != 1 && channels != 2 ) {
f@0:     errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
f@0:     return false;
f@0:   }
f@0:   ss.channels = channels;
f@0: 
f@0:   if ( firstChannel != 0 ) return false;
f@0: 
f@0:   bool sr_found = false;
f@0:   for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
f@0:     if ( sampleRate == *sr ) {
f@0:       sr_found = true;
f@0:       stream_.sampleRate = sampleRate;
f@0:       ss.rate = sampleRate;
f@0:       break;
f@0:     }
f@0:   }
f@0:   if ( !sr_found ) {
f@0:     errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
f@0:     return false;
f@0:   }
f@0: 
f@0:   bool sf_found = 0;
f@0:   for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
f@0:         sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
f@0:     if ( format == sf->rtaudio_format ) {
f@0:       sf_found = true;
f@0:       stream_.userFormat = sf->rtaudio_format;
f@0:       stream_.deviceFormat[mode] = stream_.userFormat;
f@0:       ss.format = sf->pa_format;
f@0:       break;
f@0:     }
f@0:   }
f@0:   if ( !sf_found ) { // Use internal data format conversion.
f@0:     stream_.userFormat = format;
f@0:     stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
f@0:     ss.format = PA_SAMPLE_FLOAT32LE;
f@0:   }
f@0: 
f@0:   // Set other stream parameters.
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
f@0:   else stream_.userInterleaved = true;
f@0:   stream_.deviceInterleaved[mode] = true;
f@0:   stream_.nBuffers = 1;
f@0:   stream_.doByteSwap[mode] = false;
f@0:   stream_.nUserChannels[mode] = channels;
f@0:   stream_.nDeviceChannels[mode] = channels + firstChannel;
f@0:   stream_.channelOffset[mode] = 0;
f@0:   std::string streamName = "RtAudio";
f@0: 
f@0:   // Set flags for buffer conversion.
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if ( stream_.userFormat != stream_.deviceFormat[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   // Allocate necessary internal buffers.
f@0:   bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
f@0:   stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
f@0:   if ( stream_.userBuffer[mode] == NULL ) {
f@0:     errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
f@0:     goto error;
f@0:   }
f@0:   stream_.bufferSize = *bufferSize;
f@0: 
f@0:   if ( stream_.doConvertBuffer[mode] ) {
f@0: 
f@0:     bool makeBuffer = true;
f@0:     bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
f@0:     if ( mode == INPUT ) {
f@0:       if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
f@0:         unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
f@0:         if ( bufferBytes <= bytesOut ) makeBuffer = false;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( makeBuffer ) {
f@0:       bufferBytes *= *bufferSize;
f@0:       if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
f@0:       stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
f@0:       if ( stream_.deviceBuffer == NULL ) {
f@0:         errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
f@0:         goto error;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.device[mode] = device;
f@0: 
f@0:   // Setup the buffer conversion information structure.
f@0:   if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
f@0: 
f@0:   if ( !stream_.apiHandle ) {
f@0:     PulseAudioHandle *pah = new PulseAudioHandle;
f@0:     if ( !pah ) {
f@0:       errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     stream_.apiHandle = pah;
f@0:     if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
f@0:       errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
f@0:       goto error;
f@0:     }
f@0:   }
f@0:   pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
f@0: 
f@0:   int error;
f@0:   if ( !options->streamName.empty() ) streamName = options->streamName;
f@0:   switch ( mode ) {
f@0:   case INPUT:
f@0:     pa_buffer_attr buffer_attr;
f@0:     buffer_attr.fragsize = bufferBytes;
f@0:     buffer_attr.maxlength = -1;
f@0: 
f@0:     pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
f@0:     if ( !pah->s_rec ) {
f@0:       errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
f@0:       goto error;
f@0:     }
f@0:     break;
f@0:   case OUTPUT:
f@0:     pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
f@0:     if ( !pah->s_play ) {
f@0:       errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
f@0:       goto error;
f@0:     }
f@0:     break;
f@0:   default:
f@0:     goto error;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == UNINITIALIZED )
f@0:     stream_.mode = mode;
f@0:   else if ( stream_.mode == mode )
f@0:     goto error;
f@0:   else
f@0:     stream_.mode = DUPLEX;
f@0: 
f@0:   if ( !stream_.callbackInfo.isRunning ) {
f@0:     stream_.callbackInfo.object = this;
f@0:     stream_.callbackInfo.isRunning = true;
f@0:     if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
f@0:       errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
f@0:       goto error;
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.state = STREAM_STOPPED;
f@0:   return true;
f@0:  
f@0:  error:
f@0:   if ( pah && stream_.callbackInfo.isRunning ) {
f@0:     pthread_cond_destroy( &pah->runnable_cv );
f@0:     delete pah;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: //******************** End of __LINUX_PULSE__ *********************//
f@0: #endif
f@0: 
f@0: #if defined(__LINUX_OSS__)
f@0: 
f@0: #include <unistd.h>
f@0: #include <sys/ioctl.h>
f@0: #include <unistd.h>
f@0: #include <fcntl.h>
f@0: #include <sys/soundcard.h>
f@0: #include <errno.h>
f@0: #include <math.h>
f@0: 
f@0: static void *ossCallbackHandler(void * ptr);
f@0: 
f@0: // A structure to hold various information related to the OSS API
f@0: // implementation.
f@0: struct OssHandle {
f@0:   int id[2];    // device ids
f@0:   bool xrun[2];
f@0:   bool triggered;
f@0:   pthread_cond_t runnable;
f@0: 
f@0:   OssHandle()
f@0:     :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
f@0: };
f@0: 
f@0: RtApiOss :: RtApiOss()
f@0: {
f@0:   // Nothing to do here.
f@0: }
f@0: 
f@0: RtApiOss :: ~RtApiOss()
f@0: {
f@0:   if ( stream_.state != STREAM_CLOSED ) closeStream();
f@0: }
f@0: 
f@0: unsigned int RtApiOss :: getDeviceCount( void )
f@0: {
f@0:   int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
f@0:   if ( mixerfd == -1 ) {
f@0:     errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
f@0:     error( RtAudioError::WARNING );
f@0:     return 0;
f@0:   }
f@0: 
f@0:   oss_sysinfo sysinfo;
f@0:   if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
f@0:     close( mixerfd );
f@0:     errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
f@0:     error( RtAudioError::WARNING );
f@0:     return 0;
f@0:   }
f@0: 
f@0:   close( mixerfd );
f@0:   return sysinfo.numaudios;
f@0: }
f@0: 
f@0: RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
f@0: {
f@0:   RtAudio::DeviceInfo info;
f@0:   info.probed = false;
f@0: 
f@0:   int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
f@0:   if ( mixerfd == -1 ) {
f@0:     errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   oss_sysinfo sysinfo;
f@0:   int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
f@0:   if ( result == -1 ) {
f@0:     close( mixerfd );
f@0:     errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   unsigned nDevices = sysinfo.numaudios;
f@0:   if ( nDevices == 0 ) {
f@0:     close( mixerfd );
f@0:     errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     close( mixerfd );
f@0:     errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:     return info;
f@0:   }
f@0: 
f@0:   oss_audioinfo ainfo;
f@0:   ainfo.dev = device;
f@0:   result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
f@0:   close( mixerfd );
f@0:   if ( result == -1 ) {
f@0:     errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Probe channels
f@0:   if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
f@0:   if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
f@0:   if ( ainfo.caps & PCM_CAP_DUPLEX ) {
f@0:     if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
f@0:       info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
f@0:   }
f@0: 
f@0:   // Probe data formats ... do for input
f@0:   unsigned long mask = ainfo.iformats;
f@0:   if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
f@0:     info.nativeFormats |= RTAUDIO_SINT16;
f@0:   if ( mask & AFMT_S8 )
f@0:     info.nativeFormats |= RTAUDIO_SINT8;
f@0:   if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
f@0:     info.nativeFormats |= RTAUDIO_SINT32;
f@0:   if ( mask & AFMT_FLOAT )
f@0:     info.nativeFormats |= RTAUDIO_FLOAT32;
f@0:   if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
f@0:     info.nativeFormats |= RTAUDIO_SINT24;
f@0: 
f@0:   // Check that we have at least one supported format
f@0:   if ( info.nativeFormats == 0 ) {
f@0:     errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:     return info;
f@0:   }
f@0: 
f@0:   // Probe the supported sample rates.
f@0:   info.sampleRates.clear();
f@0:   if ( ainfo.nrates ) {
f@0:     for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
f@0:       for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
f@0:         if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
f@0:           info.sampleRates.push_back( SAMPLE_RATES[k] );
f@0:           break;
f@0:         }
f@0:       }
f@0:     }
f@0:   }
f@0:   else {
f@0:     // Check min and max rate values;
f@0:     for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
f@0:       if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
f@0:         info.sampleRates.push_back( SAMPLE_RATES[k] );
f@0:     }
f@0:   }
f@0: 
f@0:   if ( info.sampleRates.size() == 0 ) {
f@0:     errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     error( RtAudioError::WARNING );
f@0:   }
f@0:   else {
f@0:     info.probed = true;
f@0:     info.name = ainfo.name;
f@0:   }
f@0: 
f@0:   return info;
f@0: }
f@0: 
f@0: 
f@0: bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
f@0:                                   unsigned int firstChannel, unsigned int sampleRate,
f@0:                                   RtAudioFormat format, unsigned int *bufferSize,
f@0:                                   RtAudio::StreamOptions *options )
f@0: {
f@0:   int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
f@0:   if ( mixerfd == -1 ) {
f@0:     errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   oss_sysinfo sysinfo;
f@0:   int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
f@0:   if ( result == -1 ) {
f@0:     close( mixerfd );
f@0:     errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   unsigned nDevices = sysinfo.numaudios;
f@0:   if ( nDevices == 0 ) {
f@0:     // This should not happen because a check is made before this function is called.
f@0:     close( mixerfd );
f@0:     errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   if ( device >= nDevices ) {
f@0:     // This should not happen because a check is made before this function is called.
f@0:     close( mixerfd );
f@0:     errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   oss_audioinfo ainfo;
f@0:   ainfo.dev = device;
f@0:   result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
f@0:   close( mixerfd );
f@0:   if ( result == -1 ) {
f@0:     errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Check if device supports input or output
f@0:   if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
f@0:        ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
f@0:     if ( mode == OUTPUT )
f@0:       errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
f@0:     else
f@0:       errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   int flags = 0;
f@0:   OssHandle *handle = (OssHandle *) stream_.apiHandle;
f@0:   if ( mode == OUTPUT )
f@0:     flags |= O_WRONLY;
f@0:   else { // mode == INPUT
f@0:     if (stream_.mode == OUTPUT && stream_.device[0] == device) {
f@0:       // We just set the same device for playback ... close and reopen for duplex (OSS only).
f@0:       close( handle->id[0] );
f@0:       handle->id[0] = 0;
f@0:       if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
f@0:         errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
f@0:         errorText_ = errorStream_.str();
f@0:         return FAILURE;
f@0:       }
f@0:       // Check that the number previously set channels is the same.
f@0:       if ( stream_.nUserChannels[0] != channels ) {
f@0:         errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
f@0:         errorText_ = errorStream_.str();
f@0:         return FAILURE;
f@0:       }
f@0:       flags |= O_RDWR;
f@0:     }
f@0:     else
f@0:       flags |= O_RDONLY;
f@0:   }
f@0: 
f@0:   // Set exclusive access if specified.
f@0:   if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
f@0: 
f@0:   // Try to open the device.
f@0:   int fd;
f@0:   fd = open( ainfo.devnode, flags, 0 );
f@0:   if ( fd == -1 ) {
f@0:     if ( errno == EBUSY )
f@0:       errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
f@0:     else
f@0:       errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // For duplex operation, specifically set this mode (this doesn't seem to work).
f@0:   /*
f@0:     if ( flags | O_RDWR ) {
f@0:     result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
f@0:     if ( result == -1) {
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:     }
f@0:     }
f@0:   */
f@0: 
f@0:   // Check the device channel support.
f@0:   stream_.nUserChannels[mode] = channels;
f@0:   if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Set the number of channels.
f@0:   int deviceChannels = channels + firstChannel;
f@0:   result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
f@0:   if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0:   stream_.nDeviceChannels[mode] = deviceChannels;
f@0: 
f@0:   // Get the data format mask
f@0:   int mask;
f@0:   result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
f@0:   if ( result == -1 ) {
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Determine how to set the device format.
f@0:   stream_.userFormat = format;
f@0:   int deviceFormat = -1;
f@0:   stream_.doByteSwap[mode] = false;
f@0:   if ( format == RTAUDIO_SINT8 ) {
f@0:     if ( mask & AFMT_S8 ) {
f@0:       deviceFormat = AFMT_S8;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT8;
f@0:     }
f@0:   }
f@0:   else if ( format == RTAUDIO_SINT16 ) {
f@0:     if ( mask & AFMT_S16_NE ) {
f@0:       deviceFormat = AFMT_S16_NE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:     }
f@0:     else if ( mask & AFMT_S16_OE ) {
f@0:       deviceFormat = AFMT_S16_OE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:       stream_.doByteSwap[mode] = true;
f@0:     }
f@0:   }
f@0:   else if ( format == RTAUDIO_SINT24 ) {
f@0:     if ( mask & AFMT_S24_NE ) {
f@0:       deviceFormat = AFMT_S24_NE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT24;
f@0:     }
f@0:     else if ( mask & AFMT_S24_OE ) {
f@0:       deviceFormat = AFMT_S24_OE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT24;
f@0:       stream_.doByteSwap[mode] = true;
f@0:     }
f@0:   }
f@0:   else if ( format == RTAUDIO_SINT32 ) {
f@0:     if ( mask & AFMT_S32_NE ) {
f@0:       deviceFormat = AFMT_S32_NE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT32;
f@0:     }
f@0:     else if ( mask & AFMT_S32_OE ) {
f@0:       deviceFormat = AFMT_S32_OE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT32;
f@0:       stream_.doByteSwap[mode] = true;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( deviceFormat == -1 ) {
f@0:     // The user requested format is not natively supported by the device.
f@0:     if ( mask & AFMT_S16_NE ) {
f@0:       deviceFormat = AFMT_S16_NE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:     }
f@0:     else if ( mask & AFMT_S32_NE ) {
f@0:       deviceFormat = AFMT_S32_NE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT32;
f@0:     }
f@0:     else if ( mask & AFMT_S24_NE ) {
f@0:       deviceFormat = AFMT_S24_NE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT24;
f@0:     }
f@0:     else if ( mask & AFMT_S16_OE ) {
f@0:       deviceFormat = AFMT_S16_OE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT16;
f@0:       stream_.doByteSwap[mode] = true;
f@0:     }
f@0:     else if ( mask & AFMT_S32_OE ) {
f@0:       deviceFormat = AFMT_S32_OE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT32;
f@0:       stream_.doByteSwap[mode] = true;
f@0:     }
f@0:     else if ( mask & AFMT_S24_OE ) {
f@0:       deviceFormat = AFMT_S24_OE;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT24;
f@0:       stream_.doByteSwap[mode] = true;
f@0:     }
f@0:     else if ( mask & AFMT_S8) {
f@0:       deviceFormat = AFMT_S8;
f@0:       stream_.deviceFormat[mode] = RTAUDIO_SINT8;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceFormat[mode] == 0 ) {
f@0:     // This really shouldn't happen ...
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Set the data format.
f@0:   int temp = deviceFormat;
f@0:   result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
f@0:   if ( result == -1 || deviceFormat != temp ) {
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Attempt to set the buffer size.  According to OSS, the minimum
f@0:   // number of buffers is two.  The supposed minimum buffer size is 16
f@0:   // bytes, so that will be our lower bound.  The argument to this
f@0:   // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
f@0:   // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
f@0:   // We'll check the actual value used near the end of the setup
f@0:   // procedure.
f@0:   int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
f@0:   if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
f@0:   int buffers = 0;
f@0:   if ( options ) buffers = options->numberOfBuffers;
f@0:   if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
f@0:   if ( buffers < 2 ) buffers = 3;
f@0:   temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
f@0:   result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
f@0:   if ( result == -1 ) {
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0:   stream_.nBuffers = buffers;
f@0: 
f@0:   // Save buffer size (in sample frames).
f@0:   *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
f@0:   stream_.bufferSize = *bufferSize;
f@0: 
f@0:   // Set the sample rate.
f@0:   int srate = sampleRate;
f@0:   result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
f@0:   if ( result == -1 ) {
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0: 
f@0:   // Verify the sample rate setup worked.
f@0:   if ( abs( srate - sampleRate ) > 100 ) {
f@0:     close( fd );
f@0:     errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
f@0:     errorText_ = errorStream_.str();
f@0:     return FAILURE;
f@0:   }
f@0:   stream_.sampleRate = sampleRate;
f@0: 
f@0:   if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
f@0:     // We're doing duplex setup here.
f@0:     stream_.deviceFormat[0] = stream_.deviceFormat[1];
f@0:     stream_.nDeviceChannels[0] = deviceChannels;
f@0:   }
f@0: 
f@0:   // Set interleaving parameters.
f@0:   stream_.userInterleaved = true;
f@0:   stream_.deviceInterleaved[mode] =  true;
f@0:   if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
f@0:     stream_.userInterleaved = false;
f@0: 
f@0:   // Set flags for buffer conversion
f@0:   stream_.doConvertBuffer[mode] = false;
f@0:   if ( stream_.userFormat != stream_.deviceFormat[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0:   if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
f@0:        stream_.nUserChannels[mode] > 1 )
f@0:     stream_.doConvertBuffer[mode] = true;
f@0: 
f@0:   // Allocate the stream handles if necessary and then save.
f@0:   if ( stream_.apiHandle == 0 ) {
f@0:     try {
f@0:       handle = new OssHandle;
f@0:     }
f@0:     catch ( std::bad_alloc& ) {
f@0:       errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     if ( pthread_cond_init( &handle->runnable, NULL ) ) {
f@0:       errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
f@0:       goto error;
f@0:     }
f@0: 
f@0:     stream_.apiHandle = (void *) handle;
f@0:   }
f@0:   else {
f@0:     handle = (OssHandle *) stream_.apiHandle;
f@0:   }
f@0:   handle->id[mode] = fd;
f@0: 
f@0:   // Allocate necessary internal buffers.
f@0:   unsigned long bufferBytes;
f@0:   bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
f@0:   stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
f@0:   if ( stream_.userBuffer[mode] == NULL ) {
f@0:     errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
f@0:     goto error;
f@0:   }
f@0: 
f@0:   if ( stream_.doConvertBuffer[mode] ) {
f@0: 
f@0:     bool makeBuffer = true;
f@0:     bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
f@0:     if ( mode == INPUT ) {
f@0:       if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
f@0:         unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
f@0:         if ( bufferBytes <= bytesOut ) makeBuffer = false;
f@0:       }
f@0:     }
f@0: 
f@0:     if ( makeBuffer ) {
f@0:       bufferBytes *= *bufferSize;
f@0:       if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
f@0:       stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
f@0:       if ( stream_.deviceBuffer == NULL ) {
f@0:         errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
f@0:         goto error;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   stream_.device[mode] = device;
f@0:   stream_.state = STREAM_STOPPED;
f@0: 
f@0:   // Setup the buffer conversion information structure.
f@0:   if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
f@0: 
f@0:   // Setup thread if necessary.
f@0:   if ( stream_.mode == OUTPUT && mode == INPUT ) {
f@0:     // We had already set up an output stream.
f@0:     stream_.mode = DUPLEX;
f@0:     if ( stream_.device[0] == device ) handle->id[0] = fd;
f@0:   }
f@0:   else {
f@0:     stream_.mode = mode;
f@0: 
f@0:     // Setup callback thread.
f@0:     stream_.callbackInfo.object = (void *) this;
f@0: 
f@0:     // Set the thread attributes for joinable and realtime scheduling
f@0:     // priority.  The higher priority will only take affect if the
f@0:     // program is run as root or suid.
f@0:     pthread_attr_t attr;
f@0:     pthread_attr_init( &attr );
f@0:     pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
f@0: #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
f@0:     if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
f@0:       struct sched_param param;
f@0:       int priority = options->priority;
f@0:       int min = sched_get_priority_min( SCHED_RR );
f@0:       int max = sched_get_priority_max( SCHED_RR );
f@0:       if ( priority < min ) priority = min;
f@0:       else if ( priority > max ) priority = max;
f@0:       param.sched_priority = priority;
f@0:       pthread_attr_setschedparam( &attr, &param );
f@0:       pthread_attr_setschedpolicy( &attr, SCHED_RR );
f@0:     }
f@0:     else
f@0:       pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
f@0: #else
f@0:     pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
f@0: #endif
f@0: 
f@0:     stream_.callbackInfo.isRunning = true;
f@0:     result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
f@0:     pthread_attr_destroy( &attr );
f@0:     if ( result ) {
f@0:       stream_.callbackInfo.isRunning = false;
f@0:       errorText_ = "RtApiOss::error creating callback thread!";
f@0:       goto error;
f@0:     }
f@0:   }
f@0: 
f@0:   return SUCCESS;
f@0: 
f@0:  error:
f@0:   if ( handle ) {
f@0:     pthread_cond_destroy( &handle->runnable );
f@0:     if ( handle->id[0] ) close( handle->id[0] );
f@0:     if ( handle->id[1] ) close( handle->id[1] );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   return FAILURE;
f@0: }
f@0: 
f@0: void RtApiOss :: closeStream()
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiOss::closeStream(): no open stream to close!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   OssHandle *handle = (OssHandle *) stream_.apiHandle;
f@0:   stream_.callbackInfo.isRunning = false;
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0:   if ( stream_.state == STREAM_STOPPED )
f@0:     pthread_cond_signal( &handle->runnable );
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0:   pthread_join( stream_.callbackInfo.thread, NULL );
f@0: 
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
f@0:       ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
f@0:     else
f@0:       ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
f@0:     stream_.state = STREAM_STOPPED;
f@0:   }
f@0: 
f@0:   if ( handle ) {
f@0:     pthread_cond_destroy( &handle->runnable );
f@0:     if ( handle->id[0] ) close( handle->id[0] );
f@0:     if ( handle->id[1] ) close( handle->id[1] );
f@0:     delete handle;
f@0:     stream_.apiHandle = 0;
f@0:   }
f@0: 
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     if ( stream_.userBuffer[i] ) {
f@0:       free( stream_.userBuffer[i] );
f@0:       stream_.userBuffer[i] = 0;
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.deviceBuffer ) {
f@0:     free( stream_.deviceBuffer );
f@0:     stream_.deviceBuffer = 0;
f@0:   }
f@0: 
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.state = STREAM_CLOSED;
f@0: }
f@0: 
f@0: void RtApiOss :: startStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_RUNNING ) {
f@0:     errorText_ = "RtApiOss::startStream(): the stream is already running!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   stream_.state = STREAM_RUNNING;
f@0: 
f@0:   // No need to do anything else here ... OSS automatically starts
f@0:   // when fed samples.
f@0: 
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   OssHandle *handle = (OssHandle *) stream_.apiHandle;
f@0:   pthread_cond_signal( &handle->runnable );
f@0: }
f@0: 
f@0: void RtApiOss :: stopStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   // The state might change while waiting on a mutex.
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     MUTEX_UNLOCK( &stream_.mutex );
f@0:     return;
f@0:   }
f@0: 
f@0:   int result = 0;
f@0:   OssHandle *handle = (OssHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     // Flush the output with zeros a few times.
f@0:     char *buffer;
f@0:     int samples;
f@0:     RtAudioFormat format;
f@0: 
f@0:     if ( stream_.doConvertBuffer[0] ) {
f@0:       buffer = stream_.deviceBuffer;
f@0:       samples = stream_.bufferSize * stream_.nDeviceChannels[0];
f@0:       format = stream_.deviceFormat[0];
f@0:     }
f@0:     else {
f@0:       buffer = stream_.userBuffer[0];
f@0:       samples = stream_.bufferSize * stream_.nUserChannels[0];
f@0:       format = stream_.userFormat;
f@0:     }
f@0: 
f@0:     memset( buffer, 0, samples * formatBytes(format) );
f@0:     for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
f@0:       result = write( handle->id[0], buffer, samples * formatBytes(format) );
f@0:       if ( result == -1 ) {
f@0:         errorText_ = "RtApiOss::stopStream: audio write error.";
f@0:         error( RtAudioError::WARNING );
f@0:       }
f@0:     }
f@0: 
f@0:     result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
f@0:     if ( result == -1 ) {
f@0:       errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:     handle->triggered = false;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
f@0:     result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
f@0:     if ( result == -1 ) {
f@0:       errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   if ( result != -1 ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiOss :: abortStream()
f@0: {
f@0:   verifyStream();
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   // The state might change while waiting on a mutex.
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     MUTEX_UNLOCK( &stream_.mutex );
f@0:     return;
f@0:   }
f@0: 
f@0:   int result = 0;
f@0:   OssHandle *handle = (OssHandle *) stream_.apiHandle;
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0:     result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
f@0:     if ( result == -1 ) {
f@0:       errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:     handle->triggered = false;
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
f@0:     result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
f@0:     if ( result == -1 ) {
f@0:       errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
f@0:       errorText_ = errorStream_.str();
f@0:       goto unlock;
f@0:     }
f@0:   }
f@0: 
f@0:  unlock:
f@0:   stream_.state = STREAM_STOPPED;
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   if ( result != -1 ) return;
f@0:   error( RtAudioError::SYSTEM_ERROR );
f@0: }
f@0: 
f@0: void RtApiOss :: callbackEvent()
f@0: {
f@0:   OssHandle *handle = (OssHandle *) stream_.apiHandle;
f@0:   if ( stream_.state == STREAM_STOPPED ) {
f@0:     MUTEX_LOCK( &stream_.mutex );
f@0:     pthread_cond_wait( &handle->runnable, &stream_.mutex );
f@0:     if ( stream_.state != STREAM_RUNNING ) {
f@0:       MUTEX_UNLOCK( &stream_.mutex );
f@0:       return;
f@0:     }
f@0:     MUTEX_UNLOCK( &stream_.mutex );
f@0:   }
f@0: 
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
f@0:     error( RtAudioError::WARNING );
f@0:     return;
f@0:   }
f@0: 
f@0:   // Invoke user callback to get fresh output data.
f@0:   int doStopStream = 0;
f@0:   RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
f@0:   double streamTime = getStreamTime();
f@0:   RtAudioStreamStatus status = 0;
f@0:   if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
f@0:     status |= RTAUDIO_OUTPUT_UNDERFLOW;
f@0:     handle->xrun[0] = false;
f@0:   }
f@0:   if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
f@0:     status |= RTAUDIO_INPUT_OVERFLOW;
f@0:     handle->xrun[1] = false;
f@0:   }
f@0:   doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
f@0:                            stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
f@0:   if ( doStopStream == 2 ) {
f@0:     this->abortStream();
f@0:     return;
f@0:   }
f@0: 
f@0:   MUTEX_LOCK( &stream_.mutex );
f@0: 
f@0:   // The state might change while waiting on a mutex.
f@0:   if ( stream_.state == STREAM_STOPPED ) goto unlock;
f@0: 
f@0:   int result;
f@0:   char *buffer;
f@0:   int samples;
f@0:   RtAudioFormat format;
f@0: 
f@0:   if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     // Setup parameters and do buffer conversion if necessary.
f@0:     if ( stream_.doConvertBuffer[0] ) {
f@0:       buffer = stream_.deviceBuffer;
f@0:       convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
f@0:       samples = stream_.bufferSize * stream_.nDeviceChannels[0];
f@0:       format = stream_.deviceFormat[0];
f@0:     }
f@0:     else {
f@0:       buffer = stream_.userBuffer[0];
f@0:       samples = stream_.bufferSize * stream_.nUserChannels[0];
f@0:       format = stream_.userFormat;
f@0:     }
f@0: 
f@0:     // Do byte swapping if necessary.
f@0:     if ( stream_.doByteSwap[0] )
f@0:       byteSwapBuffer( buffer, samples, format );
f@0: 
f@0:     if ( stream_.mode == DUPLEX && handle->triggered == false ) {
f@0:       int trig = 0;
f@0:       ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
f@0:       result = write( handle->id[0], buffer, samples * formatBytes(format) );
f@0:       trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
f@0:       ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
f@0:       handle->triggered = true;
f@0:     }
f@0:     else
f@0:       // Write samples to device.
f@0:       result = write( handle->id[0], buffer, samples * formatBytes(format) );
f@0: 
f@0:     if ( result == -1 ) {
f@0:       // We'll assume this is an underrun, though there isn't a
f@0:       // specific means for determining that.
f@0:       handle->xrun[0] = true;
f@0:       errorText_ = "RtApiOss::callbackEvent: audio write error.";
f@0:       error( RtAudioError::WARNING );
f@0:       // Continue on to input section.
f@0:     }
f@0:   }
f@0: 
f@0:   if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
f@0: 
f@0:     // Setup parameters.
f@0:     if ( stream_.doConvertBuffer[1] ) {
f@0:       buffer = stream_.deviceBuffer;
f@0:       samples = stream_.bufferSize * stream_.nDeviceChannels[1];
f@0:       format = stream_.deviceFormat[1];
f@0:     }
f@0:     else {
f@0:       buffer = stream_.userBuffer[1];
f@0:       samples = stream_.bufferSize * stream_.nUserChannels[1];
f@0:       format = stream_.userFormat;
f@0:     }
f@0: 
f@0:     // Read samples from device.
f@0:     result = read( handle->id[1], buffer, samples * formatBytes(format) );
f@0: 
f@0:     if ( result == -1 ) {
f@0:       // We'll assume this is an overrun, though there isn't a
f@0:       // specific means for determining that.
f@0:       handle->xrun[1] = true;
f@0:       errorText_ = "RtApiOss::callbackEvent: audio read error.";
f@0:       error( RtAudioError::WARNING );
f@0:       goto unlock;
f@0:     }
f@0: 
f@0:     // Do byte swapping if necessary.
f@0:     if ( stream_.doByteSwap[1] )
f@0:       byteSwapBuffer( buffer, samples, format );
f@0: 
f@0:     // Do buffer conversion if necessary.
f@0:     if ( stream_.doConvertBuffer[1] )
f@0:       convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
f@0:   }
f@0: 
f@0:  unlock:
f@0:   MUTEX_UNLOCK( &stream_.mutex );
f@0: 
f@0:   RtApi::tickStreamTime();
f@0:   if ( doStopStream == 1 ) this->stopStream();
f@0: }
f@0: 
f@0: static void *ossCallbackHandler( void *ptr )
f@0: {
f@0:   CallbackInfo *info = (CallbackInfo *) ptr;
f@0:   RtApiOss *object = (RtApiOss *) info->object;
f@0:   bool *isRunning = &info->isRunning;
f@0: 
f@0:   while ( *isRunning == true ) {
f@0:     pthread_testcancel();
f@0:     object->callbackEvent();
f@0:   }
f@0: 
f@0:   pthread_exit( NULL );
f@0: }
f@0: 
f@0: //******************** End of __LINUX_OSS__ *********************//
f@0: #endif
f@0: 
f@0: 
f@0: // *************************************************** //
f@0: //
f@0: // Protected common (OS-independent) RtAudio methods.
f@0: //
f@0: // *************************************************** //
f@0: 
f@0: // This method can be modified to control the behavior of error
f@0: // message printing.
f@0: void RtApi :: error( RtAudioError::Type type )
f@0: {
f@0:   errorStream_.str(""); // clear the ostringstream
f@0: 
f@0:   RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
f@0:   if ( errorCallback ) {
f@0:     // abortStream() can generate new error messages. Ignore them. Just keep original one.
f@0: 
f@0:     if ( firstErrorOccurred_ )
f@0:       return;
f@0: 
f@0:     firstErrorOccurred_ = true;
f@0:     const std::string errorMessage = errorText_;
f@0: 
f@0:     if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
f@0:       stream_.callbackInfo.isRunning = false; // exit from the thread
f@0:       abortStream();
f@0:     }
f@0: 
f@0:     errorCallback( type, errorMessage );
f@0:     firstErrorOccurred_ = false;
f@0:     return;
f@0:   }
f@0: 
f@0:   if ( type == RtAudioError::WARNING && showWarnings_ == true )
f@0:     std::cerr << '\n' << errorText_ << "\n\n";
f@0:   else if ( type != RtAudioError::WARNING )
f@0:     throw( RtAudioError( errorText_, type ) );
f@0: }
f@0: 
f@0: void RtApi :: verifyStream()
f@0: {
f@0:   if ( stream_.state == STREAM_CLOSED ) {
f@0:     errorText_ = "RtApi:: a stream is not open!";
f@0:     error( RtAudioError::INVALID_USE );
f@0:   }
f@0: }
f@0: 
f@0: void RtApi :: clearStreamInfo()
f@0: {
f@0:   stream_.mode = UNINITIALIZED;
f@0:   stream_.state = STREAM_CLOSED;
f@0:   stream_.sampleRate = 0;
f@0:   stream_.bufferSize = 0;
f@0:   stream_.nBuffers = 0;
f@0:   stream_.userFormat = 0;
f@0:   stream_.userInterleaved = true;
f@0:   stream_.streamTime = 0.0;
f@0:   stream_.apiHandle = 0;
f@0:   stream_.deviceBuffer = 0;
f@0:   stream_.callbackInfo.callback = 0;
f@0:   stream_.callbackInfo.userData = 0;
f@0:   stream_.callbackInfo.isRunning = false;
f@0:   stream_.callbackInfo.errorCallback = 0;
f@0:   for ( int i=0; i<2; i++ ) {
f@0:     stream_.device[i] = 11111;
f@0:     stream_.doConvertBuffer[i] = false;
f@0:     stream_.deviceInterleaved[i] = true;
f@0:     stream_.doByteSwap[i] = false;
f@0:     stream_.nUserChannels[i] = 0;
f@0:     stream_.nDeviceChannels[i] = 0;
f@0:     stream_.channelOffset[i] = 0;
f@0:     stream_.deviceFormat[i] = 0;
f@0:     stream_.latency[i] = 0;
f@0:     stream_.userBuffer[i] = 0;
f@0:     stream_.convertInfo[i].channels = 0;
f@0:     stream_.convertInfo[i].inJump = 0;
f@0:     stream_.convertInfo[i].outJump = 0;
f@0:     stream_.convertInfo[i].inFormat = 0;
f@0:     stream_.convertInfo[i].outFormat = 0;
f@0:     stream_.convertInfo[i].inOffset.clear();
f@0:     stream_.convertInfo[i].outOffset.clear();
f@0:   }
f@0: }
f@0: 
f@0: unsigned int RtApi :: formatBytes( RtAudioFormat format )
f@0: {
f@0:   if ( format == RTAUDIO_SINT16 )
f@0:     return 2;
f@0:   else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
f@0:     return 4;
f@0:   else if ( format == RTAUDIO_FLOAT64 )
f@0:     return 8;
f@0:   else if ( format == RTAUDIO_SINT24 )
f@0:     return 3;
f@0:   else if ( format == RTAUDIO_SINT8 )
f@0:     return 1;
f@0: 
f@0:   errorText_ = "RtApi::formatBytes: undefined format.";
f@0:   error( RtAudioError::WARNING );
f@0: 
f@0:   return 0;
f@0: }
f@0: 
f@0: void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
f@0: {
f@0:   if ( mode == INPUT ) { // convert device to user buffer
f@0:     stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
f@0:     stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
f@0:     stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
f@0:     stream_.convertInfo[mode].outFormat = stream_.userFormat;
f@0:   }
f@0:   else { // convert user to device buffer
f@0:     stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
f@0:     stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
f@0:     stream_.convertInfo[mode].inFormat = stream_.userFormat;
f@0:     stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
f@0:   }
f@0: 
f@0:   if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
f@0:     stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
f@0:   else
f@0:     stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
f@0: 
f@0:   // Set up the interleave/deinterleave offsets.
f@0:   if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
f@0:     if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
f@0:          ( mode == INPUT && stream_.userInterleaved ) ) {
f@0:       for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
f@0:         stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
f@0:         stream_.convertInfo[mode].outOffset.push_back( k );
f@0:         stream_.convertInfo[mode].inJump = 1;
f@0:       }
f@0:     }
f@0:     else {
f@0:       for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
f@0:         stream_.convertInfo[mode].inOffset.push_back( k );
f@0:         stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
f@0:         stream_.convertInfo[mode].outJump = 1;
f@0:       }
f@0:     }
f@0:   }
f@0:   else { // no (de)interleaving
f@0:     if ( stream_.userInterleaved ) {
f@0:       for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
f@0:         stream_.convertInfo[mode].inOffset.push_back( k );
f@0:         stream_.convertInfo[mode].outOffset.push_back( k );
f@0:       }
f@0:     }
f@0:     else {
f@0:       for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
f@0:         stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
f@0:         stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
f@0:         stream_.convertInfo[mode].inJump = 1;
f@0:         stream_.convertInfo[mode].outJump = 1;
f@0:       }
f@0:     }
f@0:   }
f@0: 
f@0:   // Add channel offset.
f@0:   if ( firstChannel > 0 ) {
f@0:     if ( stream_.deviceInterleaved[mode] ) {
f@0:       if ( mode == OUTPUT ) {
f@0:         for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
f@0:           stream_.convertInfo[mode].outOffset[k] += firstChannel;
f@0:       }
f@0:       else {
f@0:         for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
f@0:           stream_.convertInfo[mode].inOffset[k] += firstChannel;
f@0:       }
f@0:     }
f@0:     else {
f@0:       if ( mode == OUTPUT ) {
f@0:         for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
f@0:           stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
f@0:       }
f@0:       else {
f@0:         for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
f@0:           stream_.convertInfo[mode].inOffset[k] += ( firstChannel  * stream_.bufferSize );
f@0:       }
f@0:     }
f@0:   }
f@0: }
f@0: 
f@0: void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
f@0: {
f@0:   // This function does format conversion, input/output channel compensation, and
f@0:   // data interleaving/deinterleaving.  24-bit integers are assumed to occupy
f@0:   // the lower three bytes of a 32-bit integer.
f@0: 
f@0:   // Clear our device buffer when in/out duplex device channels are different
f@0:   if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
f@0:        ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
f@0:     memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
f@0: 
f@0:   int j;
f@0:   if (info.outFormat == RTAUDIO_FLOAT64) {
f@0:     Float64 scale;
f@0:     Float64 *out = (Float64 *)outBuffer;
f@0: 
f@0:     if (info.inFormat == RTAUDIO_SINT8) {
f@0:       signed char *in = (signed char *)inBuffer;
f@0:       scale = 1.0 / 127.5;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT16) {
f@0:       Int16 *in = (Int16 *)inBuffer;
f@0:       scale = 1.0 / 32767.5;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT24) {
f@0:       Int24 *in = (Int24 *)inBuffer;
f@0:       scale = 1.0 / 8388607.5;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT32) {
f@0:       Int32 *in = (Int32 *)inBuffer;
f@0:       scale = 1.0 / 2147483647.5;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT32) {
f@0:       Float32 *in = (Float32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT64) {
f@0:       // Channel compensation and/or (de)interleaving only.
f@0:       Float64 *in = (Float64 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:   }
f@0:   else if (info.outFormat == RTAUDIO_FLOAT32) {
f@0:     Float32 scale;
f@0:     Float32 *out = (Float32 *)outBuffer;
f@0: 
f@0:     if (info.inFormat == RTAUDIO_SINT8) {
f@0:       signed char *in = (signed char *)inBuffer;
f@0:       scale = (Float32) ( 1.0 / 127.5 );
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT16) {
f@0:       Int16 *in = (Int16 *)inBuffer;
f@0:       scale = (Float32) ( 1.0 / 32767.5 );
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT24) {
f@0:       Int24 *in = (Int24 *)inBuffer;
f@0:       scale = (Float32) ( 1.0 / 8388607.5 );
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT32) {
f@0:       Int32 *in = (Int32 *)inBuffer;
f@0:       scale = (Float32) ( 1.0 / 2147483647.5 );
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] += 0.5;
f@0:           out[info.outOffset[j]] *= scale;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT32) {
f@0:       // Channel compensation and/or (de)interleaving only.
f@0:       Float32 *in = (Float32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT64) {
f@0:       Float64 *in = (Float64 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:   }
f@0:   else if (info.outFormat == RTAUDIO_SINT32) {
f@0:     Int32 *out = (Int32 *)outBuffer;
f@0:     if (info.inFormat == RTAUDIO_SINT8) {
f@0:       signed char *in = (signed char *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] <<= 24;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT16) {
f@0:       Int16 *in = (Int16 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] <<= 16;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT24) {
f@0:       Int24 *in = (Int24 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
f@0:           out[info.outOffset[j]] <<= 8;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT32) {
f@0:       // Channel compensation and/or (de)interleaving only.
f@0:       Int32 *in = (Int32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT32) {
f@0:       Float32 *in = (Float32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT64) {
f@0:       Float64 *in = (Float64 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:   }
f@0:   else if (info.outFormat == RTAUDIO_SINT24) {
f@0:     Int24 *out = (Int24 *)outBuffer;
f@0:     if (info.inFormat == RTAUDIO_SINT8) {
f@0:       signed char *in = (signed char *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
f@0:           //out[info.outOffset[j]] <<= 16;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT16) {
f@0:       Int16 *in = (Int16 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
f@0:           //out[info.outOffset[j]] <<= 8;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT24) {
f@0:       // Channel compensation and/or (de)interleaving only.
f@0:       Int24 *in = (Int24 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT32) {
f@0:       Int32 *in = (Int32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
f@0:           //out[info.outOffset[j]] >>= 8;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT32) {
f@0:       Float32 *in = (Float32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT64) {
f@0:       Float64 *in = (Float64 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:   }
f@0:   else if (info.outFormat == RTAUDIO_SINT16) {
f@0:     Int16 *out = (Int16 *)outBuffer;
f@0:     if (info.inFormat == RTAUDIO_SINT8) {
f@0:       signed char *in = (signed char *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
f@0:           out[info.outOffset[j]] <<= 8;
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT16) {
f@0:       // Channel compensation and/or (de)interleaving only.
f@0:       Int16 *in = (Int16 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT24) {
f@0:       Int24 *in = (Int24 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT32) {
f@0:       Int32 *in = (Int32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT32) {
f@0:       Float32 *in = (Float32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT64) {
f@0:       Float64 *in = (Float64 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:   }
f@0:   else if (info.outFormat == RTAUDIO_SINT8) {
f@0:     signed char *out = (signed char *)outBuffer;
f@0:     if (info.inFormat == RTAUDIO_SINT8) {
f@0:       // Channel compensation and/or (de)interleaving only.
f@0:       signed char *in = (signed char *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = in[info.inOffset[j]];
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     if (info.inFormat == RTAUDIO_SINT16) {
f@0:       Int16 *in = (Int16 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT24) {
f@0:       Int24 *in = (Int24 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_SINT32) {
f@0:       Int32 *in = (Int32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT32) {
f@0:       Float32 *in = (Float32 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:     else if (info.inFormat == RTAUDIO_FLOAT64) {
f@0:       Float64 *in = (Float64 *)inBuffer;
f@0:       for (unsigned int i=0; i<stream_.bufferSize; i++) {
f@0:         for (j=0; j<info.channels; j++) {
f@0:           out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
f@0:         }
f@0:         in += info.inJump;
f@0:         out += info.outJump;
f@0:       }
f@0:     }
f@0:   }
f@0: }
f@0: 
f@0: //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
f@0: //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
f@0: //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
f@0: 
f@0: void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
f@0: {
f@0:   register char val;
f@0:   register char *ptr;
f@0: 
f@0:   ptr = buffer;
f@0:   if ( format == RTAUDIO_SINT16 ) {
f@0:     for ( unsigned int i=0; i<samples; i++ ) {
f@0:       // Swap 1st and 2nd bytes.
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+1);
f@0:       *(ptr+1) = val;
f@0: 
f@0:       // Increment 2 bytes.
f@0:       ptr += 2;
f@0:     }
f@0:   }
f@0:   else if ( format == RTAUDIO_SINT32 ||
f@0:             format == RTAUDIO_FLOAT32 ) {
f@0:     for ( unsigned int i=0; i<samples; i++ ) {
f@0:       // Swap 1st and 4th bytes.
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+3);
f@0:       *(ptr+3) = val;
f@0: 
f@0:       // Swap 2nd and 3rd bytes.
f@0:       ptr += 1;
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+1);
f@0:       *(ptr+1) = val;
f@0: 
f@0:       // Increment 3 more bytes.
f@0:       ptr += 3;
f@0:     }
f@0:   }
f@0:   else if ( format == RTAUDIO_SINT24 ) {
f@0:     for ( unsigned int i=0; i<samples; i++ ) {
f@0:       // Swap 1st and 3rd bytes.
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+2);
f@0:       *(ptr+2) = val;
f@0: 
f@0:       // Increment 2 more bytes.
f@0:       ptr += 2;
f@0:     }
f@0:   }
f@0:   else if ( format == RTAUDIO_FLOAT64 ) {
f@0:     for ( unsigned int i=0; i<samples; i++ ) {
f@0:       // Swap 1st and 8th bytes
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+7);
f@0:       *(ptr+7) = val;
f@0: 
f@0:       // Swap 2nd and 7th bytes
f@0:       ptr += 1;
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+5);
f@0:       *(ptr+5) = val;
f@0: 
f@0:       // Swap 3rd and 6th bytes
f@0:       ptr += 1;
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+3);
f@0:       *(ptr+3) = val;
f@0: 
f@0:       // Swap 4th and 5th bytes
f@0:       ptr += 1;
f@0:       val = *(ptr);
f@0:       *(ptr) = *(ptr+1);
f@0:       *(ptr+1) = val;
f@0: 
f@0:       // Increment 5 more bytes.
f@0:       ptr += 5;
f@0:     }
f@0:   }
f@0: }
f@0: 
f@0:   // Indentation settings for Vim and Emacs
f@0:   //
f@0:   // Local Variables:
f@0:   // c-basic-offset: 2
f@0:   // indent-tabs-mode: nil
f@0:   // End:
f@0:   //
f@0:   // vim: et sts=2 sw=2
f@0: